Sloped baffle

Lewinskih01,
I took the time to read the XO White Paper by Dr. Uli. In many places he makes the same points that Roy is making i.e. lower order x-over ckts are better than higher order x-over ckts. He talks about the time delays getting worse with higher order x-over ckts - same point that Roy has made many times.
Dr. Uli talks about using minimum phase filters for the analog x-overs &
using linear phase filters (which are digital FIR filters. there is no equivalent in the analog domain) for his Acourate digital x-over software.
Dr. Uli makes a general statement that low-order minimum phase filters used in analog x-overs have limitations & create time distortions & cannot be used......
BUT he conveniently starts off with a 2nd-order x-over ckt while completely glossing over a 1st-order x-over ckt. Does the 1st-order x-over ckt have the same limitations as the 2nd-order x-over Dr. Uli discussed? Dr. Uli would like you to think so but I don't think so......

I created a simple 1-order network for a tweeter, midrange & woofer. I assumed a 6 Ohms resistance for each of the 3 drivers (totally arbitrary). I arbitrarily chose x-over frequencies of 300Hz & 2KHz. I simulated the frequency, phase & step responses of this 1-order x-over. I've labeled the curves in each of the 3 graphs so you can see which curve belongs to which driver. I've also put markers on various curves so you can see the phase shift at the x-over frequency.

For the frequency response - look at the sum of the frequency responses. There's only a 2dB hump at the x-over points.

For the phase response - look at the sum of all the phase responses/ There's a phase shift of only +/- 8 degrees over the entire audio band of 20Hz - 20KHz.

And, for the step response - you can clearly see that all 3 drivers act in unison to create unified step response (rather than the spikes you see in time-Incoherent speakers where the tweeter acts first, the mid second & the woofer third).
From these simulations, a 1st-order passive x-over looks quite good.
And, I don't have the music signal going thru somebody's DSP algorithm which is doing a great deal of signal processing to massage the music signal thereby imparting its sonic signature to the music signal.
Sure the passive x-over components are also imparting their signature to the music signal but by using top quality components I can minimize this.
In the DSP software, if I don't know what I'm doing, I can botch thing pretty badly because the music signal is so heavily modified by the DSP algorithm.

Here is the link to the simulations, if anyone is interested:
https://picasaweb.google.com/bombaywalla9/FirstOrderXOverFreqPhaseStepResponses?authkey=Gv1sRgCOz6xv6RnMDeUA#

In the XO White Paper, Dr. Uli says that "So the crossover has to be selected so that the good properties of the driver are used ! If the driver does not have a good behaviour we should not use it."
I am assuming the "good properties" of a driver are that it has flat freq response over its passband & rolls off at a frequency beyond the x-over freq chosen in Acourate by the user. BUT............
The degree that Acourate can compensate for any driver depends on how well you can characterize the driver. And, we of course, do not know if the drivers in our existing speakers have these "good properties" or not.....

Bombaywalla,

Not sure what to take away from your post. 1st order passive crossovers are better than any type of digital crossover? That would be in line with Roy's explanation. But I wasn't arguing otherwise...

I'll take as given that Roy's approach is the best one could hope for. My question to him is how close to that would my described approach get me.

Cheers!

hi Lewinskih01,
yes, with some engineering proof, that's what I was trying to say. And, the reason that seemed to make sense to me is that signal processing is happening correctly, real-time thru the passive x-over components without any intervention by a human-being. In a time-coherent loudspeaker with passive x-overs, drivers with "good properties" have already been selected & the x-over designed around them & the whole system would be working to benefit the user.

With digital x-overs the correction is as good as the skill of the user to characterize the drivers & to come up with the appropriate filter response to yield a time-coherent delivery. And, from reading Roy's letter to Six Moons - the link to which he provided earlier on - it's no easy feat to characterize a driver in the room. One cannot use 1 type of test tone, one needs to use many different types. And, one needs to measure the driver response in many ways to get an accurate characterization of the driver. Otherwise, the DEQX or Acourate correction will be (very) limited leading to less than stellar benefits.

I don't think that Roy can tell you how well DEQX or Acourate will solve your problem because the answer lies in how skilled you are in understanding the science behind how the driver response is affected by your room,
how skilled you are in DSP algorithms to come up with a filter that corrects for your room & your particular choice of drivers
how skilled you are in understanding the science behind reflections of drivers off the front baffle,
how skilled you are in understanding what the requirements are for selecting a microphone to do the driver characterization,
how skilled you are in compensating for this mic's own frequency response so that you don't misunderstand the mic's response to be that of your driver's,
etc, etc.

My understanding is that if you room correct like HT Receivers do & plug in the correction into some pre-designed filter in the software, you'll get a correction that's average at best & you might not like the results.
The thing that Roy has been saying all along is that we don't listen to test tones (which is what the room correction tones are) - we listen to music which is a bunch of partial wavelengths of various frequencies.
You use full cycle tones to characterize the driver then do the correction & then play partial wavelengths of various frequencies thru that driver - the correction to the driver, in my understanding, is invalid.
Of course, I could be totally off-base here....
FWIW.

Good questions.

I do agree with what Bombaywalla just posted- knowledge and experience in many different areas is required. I know of no way out of that, to simplify a home-designer's life.

Driver selection is by far the most important factor. If all we care about is making the best sound, instead of spending money on the newest technology (usually inferior, I find), then here are the important questions to ask before selecting any drivers:

- How far away will I be from the speakers?
- What kinds of music will I play most?
- How loud will I play, even if only on occasion?
- How large is my room?
- How low in the bass do I want the speakers to go? Here, it is best to use 'body feel' as your guide. If you want to shake the house and your lower pants legs on electric bass, then the speakers need to have good output to 40Hz, but not any lower.

Listening at ten feet away in a room that is not entirely open into the rest of the home, this amount of low-bass output requires a low-distortion eight-inch woofer with a large-diameter bass port tuned to ~40 Hz, or a sealed-box ten-inch woofer, flat to 40Hz (good luck finding that in today's marketplace), at the minimum. There is no reason to use multiple 8 or 10-inch woofers per cabinet.

Which means this will be a three-way design to be able to use a first-order crossover, since no 8 or 10-inch woofer can meet a tweeter.

On the top end, choose ~1" dome tweeter, not one made of metal nor of 'ring radiator' design. That means ~3kHz crossover point. The eight or ten inch woofer means ~300Hz crossover point, or slightly higher. And that means using a 4 to 5-inch mid driver showing no cone breakup nor the HF resonance of metal-cone drivers.

All these drivers need very flat frequency responses. Avoid drivers with impedance-curve wiggles, as those indicate resonances and cone breakups. Avoid molded plastic cones and metal cones.

Sorry- got carried away. I cannot put out my version of the Loudspeaker Design Cookbook here.

Do know that, by careful manipulation of the Zobel parts in my passive crossovers, I can fine-tune the time-coherence between drivers (their individual phase responses), for a better blend. This cannot be achieved digitally without custom programming and the consequent extra signal processing (assuming the right measurements can be made, which is not likely).

But you can always listen to your adjustments, and for that process, I recommend you listen to only your left speaker, but not in mono. Start with getting that speaker's voice range right, such as on a older Diana Krall recording. And get rid of cabinet reflections with wool felt for at least the tweeter, or you are screwed from the beginning.

For a home designer, the results with a simple passive crossover with Zobels or with a digital first-order crossover/EQ/time delay setup will be satisfying on most music. However, the sound would still 'not be quite right' on enough other music to make you think there's something wrong with your source or room or cables or amplifiers.

That turns out to be the residual phase shift of the speakers, which is what I finally fixed .

I will continue to think about questions Bfwynne and Lewinskih01 posed and get back to you.

Best,
Roy

My goodness, I just glanced through the XO paper by Dr. Brüggemann. With all due respect, he is not right in many ways about how crossovers work!

The technical details are far too lengthy for here, but I will point out that, in Fig. 8 on his page seven, he described 'lining up the peaks' from a woofer, mid and tweeter. Instead, what must be done is to line up WHEN each driver's pulse JUST BEGINS to turn upwards from Zero. That's a point easily judged for the beginning of a tweeter's spike, but not on a woofer's slow rise (hence a measurement problem). Thus I advise not bothering with his paper, sorry.

The diagrams from Bombaywalla on his Picasa page DO get that starting alignment correct, although I see some problems:
- The scale used shows a definite starting point to the woofer's pulse. That point is not well-defined when the horizontal scale is expanded.
- The loudness of the mid driver seems low, but I could be wrong.
- The summation pulse is not close enough to the ideal.

But it is late now, and no one is paying me to analyze what may be wrong there- just wanted to point out some suspicious items.

Best,
Roy

Hey Roy.

Thanks for the thoughts again.

Thanks for pointing out that mistake in the paper, about aligning start times vs peaks. Seems something easily fixable by setting different delays in the software. So the software approach still is limited by all the previously mentioned aspects, but not an additional one :-)

I spent good time reading your website, particularly the development of the Calypso HD. Very interesting too.

In reality my system would be 4-way, as I have a pair of subwoofers I intend to continue to use. They are 12" Rythmiks in a sealed, DIY and very heavy enclosure. So below 80Hz I wouldn't need the woofers to get there, hopefully making their selection easier. Maybe an 8" woofer in a sealed enclosure does it?

I have by no means studied this at all so what follows has the goal of providing real-world examples rather than representing what I think might be best. I spent some time at Madisound.com to skim over the drivers they carry. These 3 woofers, non-metallic, from known brands. Sure, price was a simplistic way of focusing...I know it's wrong, but for this purpose...

Scanspeak Classic
Scanspeak Revelator
Accuton Ceramic

None of them is really flat down to 80Hz, let alone well below that. But they are quite flat to 100Hz, so the "problem area" seems to be rather narrow in the 80-100 Hz...hopefully not a huge deal.
Both Scanspeaks seem to be able to work well for a crossover around 500Hz. The Accuton maybe at 1kHz?
None showing wiggles on the impedance curve within these ranges.

The midrange was more difficult than I expected. VERY few drivers are flat within their expected range. Here are two looking good:

Accuton. This one looks as it could be used higher up, up to 5kHz per their recommendation.

SEAS. This one is a lot cheaper, but good on paper.

What's your take on ribbon tweeters? Clearly, you prefer non-metal dome tweeters, and non-ring-radiators. But why not ribbons? Or AMTs, such as Mundorf's? Their frequency responses look very good, and they extend well beyond 20kHz pretty flat...

I realize A LOT more thought needs to go into proper driver selection. But I am taking away that such selection is critical. Since I won't have the skills to design a proper passive XO, it could make sense embarking in all of this if the Acourate approach was good enough.

I won't get tired of saying it: thanks for the fantastic food for thought, and taking the time!!

Here's a simple un-tweak that may have helped just a little tweak (pun)... dunno.

For the longest time, I lifted the back of my speakers so they tilted forward. Here the old thought process:

My listening position is below the level of the tweeters. The speakers are about 44 inches high and my listening position is about 10 feet back. But my couch sits very low. I thought that by tilting the speakers forward, the tweeters would beam directly at me and treble would be improved.

Here's my current thinking, courtesy of this thread:

Lifting the back of the speakers as described may have augmented treble response, but the tweeter voice coils are even more forward of the mid and woofer driver voice coils than before the tilt forward. So ... to the extent there was time incoherence before, I'm just augmenting it.

So, at the expense of maybe losing a little treble, I attenuated an already non-optimal time incoherent situation just a tad.

Bottom line: it's probably in my head, but I think the speakers sound a little better. Little less bassey, a tad more coherent and invisible.

Btw, a couple of weeks ago, I switched back to the 4 ohm taps on my amp. There's definitely a noticeable change in coloration because the output impedance off the 4 ohm taps is lower -- and output voltage regulation is tighter. Bass is tighter and more extended. Upper mids/low treble are less bright.

But I also think the amp is "happier" with the load presentation because a good part of the speaker's power delivery demands are in the bass/low midrange region which specs at 4 ohms (70 Hz to 700 Hz). IOW, better impedance matching with the amp where it counts the most.

Still want to check out the DEQX.

Cheers,

BIF

You are quite welcome.
I know what I write doesn't pose questions to you all. Instead, I've mostly laid out the facts and some science. It's up to you to use those to develop your own questions. This is how I proceeded back in the early 1970's, by reading all of the AES papers and many others on speaker design in old and current magazines, on acoustics, studied basic physics, calculus, and psychoacoustics. Later, I returned to university to master all the math, and to learn more about how materials behave when vibrations exist and when electromagnetic fields pass by/pass through.

Sometimes I would find an error in the logic or math of someone's research paper. Usually, I used a paper as a springboard, expanding upon the author's thoughts and test methods, to better look at 'something' in detail.

To choose that 'something' to examine, to fix, or even to ignore, I first had to understand the very basics of WHY and HOW that 'something' would be important to what we hear, and then learn WHY and HOW 'it' occurs. This included how and why cabinets vibrate, cones break up, critical damping is achieved, a tweeter can fail to move on very tiny sounds, the air itself distorts... countless questions.

The most important ones are addressed in the Audio Engineering Society's Audio Anthology 3-book set.
Also, one should get The Audio Cyclopedia, even a twenty-year old copy. It is full of important info on acoustics, speaker design and recording methods, found nowhere else. Make sure you get one that's not falling apart in its binding.
Another book, out of print, is Elements of Acoustics by Temkin. You need to know calculus to get the most from it, but it's readable without that.
Finally, the Theory of Sound by Rayleigh, from Dover Press, is exactly like reading Isaac Newton's original papers. Get both volumes one and two, first published in the 1880's.

If you are interested in design but will never build your own speakers, these books are full of the very best information found nowhere else, and are written well enough to make for good, casual reading.

In these books, you get to see how others approached issues and usually find out WHY they did, along with what had been tried before then and WHY.

Knowing WHY is the most important factor in making better speakers. I can tell you most current speaker designs say to me that their designers know no more than what was mastered by 1979. If you read over the topics presented in those AES books, you'd see this for yourself, darn it.

At this point, I see nowhere on the internet any guidelines on how to select the proper woofer, etc. While I cannot help you directly with that, I can point out the principle differences in the drivers you selected, and leave you to have a good weekend!

- The Classic Scanspeak woofer has ALL of the right numbers for a sealed box. I wish it were more efficient.
- The more expensive Scanspeak woofer will not go as low in its proper sealed box. And unless you are stroking the heck out of it (not likely), it has no less bass distortion than the less expensive Scanspeak. However, it would be very slightly clearer in the lower-voice, high bass range. But then it goes nuts above 1kHz, all from its harder cone. Its first resonance at 1kHz is from its heavier rubber surround bouncing back, like a ripple in a flag, and then vibrating the cone running around its rim, like a church bell's 'first mode' of ringing `round-the-mouth vibration. The big spike above 1khz is its harder cone ringing like crazy.
- The Accuton woofer is a lot of $$, has high bass distortion, and will not go as low as the Classic Scanspeak.

- The Accuton mid driver has many wrong numbers and is not quite efficient enough.
- The Scan mid has the right numbers, its cone breakup is under control, and it has a vented suspension like the Scan woofer. Cross it over at ~300Hz. Read my Continuum 3 and Calypso speaker design papers for more info on using a mid.

- The only ribbons worth using, for sonic quality and which will not break for our purposes, are from RAAL. Excellent products, the best by far. You will need to create a Zobel to offset its inductance. Cross it over at 3kHz. Use their smallest model, for the best highs.

- I advise you fade in the subwoofer(s) below 40Hz, leaving the main three-way to run 'full range'.

So now you face a zillion other questions. Get the AES books above and the Audio Cyclopedia at the minimum for both guidance and answers, compared to the Loudspeaker Design Cookbook.

The Acourate approach is not right. I advise anyone hopefully learn what 'the numbers mean' for any driver, then use the parts I like above to fine-tune your own passive crossovers, with woofer mid and tweeter in their own boxes so you can move each one back and forth.

- You only need to build one speaker, as I posted before.
- You need a $100 voltmeter, a $200 fairly-low-distortion sinewave generator, a decent measuring mic with preamp, to run into some kind of third-octave spectrum analyzer for looking at pink noise.
- And a pocket calculator (scientific), especially to calculate real "L-Pads" for mid and tweeter using the best wire-wound resistors.
That's about it for tools, IF you go through the AES books.

When 'designers' do not understand in depth the extensive research from the past, they rely upon digital test gear. And then get many wrong answers since they do not understand 'the basics'. They have purchased an expensive tool that does not help solve the real problems. But they don't know-- they just stick a mic up in the air and tweak their crossovers to 'get the right curve' for each driver, which is soooo wrong.

And then they hear something 'not quite right', to then tweak the circuits by ear, so their favored recordings sound 'right'. And of course then brag about how carefully their gifted designer listened, how much money they (Revel/Harman) spent on a robotic speaker-comparison room or anechoic chamber (Paradigm/Canadian government). Hey, this isn't the space program where people get killed. This is an unsupervised field of endeavor, with no university program for it, requiring money more so than any real technical education. They always claim, "Well, we all just hear differently." Pooh.

And do get rid of/prevent any cabinet reflections for your mid and tweeter (get the mid's box away from the woofer's and tweeter's boxes, vertically). Put wool felt near the tweeter's dome.

Hope this gives you food for thought!

Best,
Roy

Roy, do you think my "un-tweak" re tipping my speakers back could have attenuated my speakers' time incoherency as I described above? Or is it just wishful hearing?

You are looking to reduce your time incoherence, is how I would say it. And yes, moving the tweeter closer may have increased your Paradigm's incoherence. But the only way to tell is to have a friend help you swing, quite literally, an arc between where your ear is and the location of where each driver's cone or dome meets its voice coil. Those should lie along the same arc.

Because you must keep the string or tape measure pulled tight, you would find you cannot just hold that string against your ear. I recommend you tape a dowel rod to a camera tripod, to mark your ear's location.

Also, get out your calculator to find out how far you are off axis. However, do not listen for tone balance, but for 'depth', for each instrument and voice to appear more and more whole, right there in front of you. The opposite is the tweeter and woofer becoming audible on their own, audibly separated away from the mid. The mid's tone range must be our reference point for someone's location, because that's the main tone range we hear every day.

It has been proven to very many people's satisfaction that the ear is not as sensitive to variations in frequency response as we would like to believe- not to say a flat response is unimportant. However, this must be true, as we never get to hear 'the best frequency response' from any source in real life, because we are never in 'the perfect spot'.

However, when you do get the Paradigm speakers into the right tape measure position/arc, the sound may be worse, because that is not 'the position' they intended. So again, always trust your ears.

In that case, have your friend tilt your left speaker back and forth while listening to Diana Krall's voice on just that left speaker. But not in mono. Her well-recorded voice is already in mono, because she and her piano were panned to the center, which means she and the piano are equal in left and right channels. You do not want the distractions of left-right information, but only the depth info and to hear a sharper focus on her voice, which one speaker can deliver.

Best,
Roy

Hey Roy.

Surely enough, I have a gazillion questions, but I won't keep asking! Well...not the gazillion anyway :-)

Thanks for the names of the publications worth reading. I am in the process of getting the first AES papers to get started. Being in Argentina doesn't help in sourcing!!

I do want to ask back about two specific comments you made:

1) why do you advise to cross over the subwoofers at 40Hz? Wouldn't the Classic Scanspeak driver listed above, for example, have an easier time if it had to reproduce down to 60 or 80Hz instead of 40?

2) you state "the Acourate approach is not right". But WHY? I'm following you other advise: to understand why? ;-)
Seriously, I realize it is not "completely" right, like with your passive network. But doesn't it get me closer to "right" than a middle of the road, non-time coherent passive XO?

I carefully re-read your paper on the Calypso HD development. I would say I studied it more that just reading. Lots of fantastic info there. I can see myself following your guidelines to build my DIY cabinets (plus what I hope to learn from the books, of course), and to get it mostly right in choosing drivers. But it would be just too arrogant on my part to assume I will be that good with XO design, and if that's the only path then it might become a deal-breaker for me. That would be a pitty!

Hi Lewinskih01,

Thank you for the questions.
1) The Classic woofer would go naturally to ~40Hz, then you would just fade in the subwoofer using its own its built-in crossover. There would be no crossover on the main speakers, which is a good thing. Also, you would then not have as much phase shift above 40Hz, compared to the main speakers needing a sub up at 60 or 80Hz- there you would hear the sub all of the time. This Scanspeak woofer would not have "an easier time" unless you are going to blast your music screaming loud.

2) on the Acourate approach, the first claim on their home page is
"The powerful software enables you
- to measure your audio system."
Yes, as do other measurement programs. None are doing anything wrong, but their measurement techniques do not match what we hear. A user will be misled by the limitations of its measurement techniques, unless he studies in detail the subjects I touch upon in that measurement-letter I wrote to sixmoons. No calculations can be right when they rely upon measurements that are wrong. An analogy is measuring a car's straight-line performance to tell how it corners.

On the other hand, I do know that after each 'good driver' gets a 'good Zobel' from you, a pocket calculator can then design your crossover. You verify its -3dB points on pink noise with spectrum-analyzer software, by measuring each driver up close. It does not matter if your microphone curve is weird, from your mic being so close, because you are only looking for what happens with and without your crossover.

The Acourate home page also claims you can use their software:
"- to display, interpret and process measurement data."
A novice user will not know how data is to be interpreted, compared to what is being heard.

It also claims
"- to establish correction filters for speaker drivers and the listening room"
For any 'correction', the software will be relying upon measurements having large flaws, as I explained in an earlier post. This includes it not being able to measure cabinet-surface reflections around the tweeter, and not being able to measure the floor reflections between you and the speakers in the same way as you perceive them.

Furthermore, no measurement made at your chair will be accurate below 500Hz, because of room reflections from your floor, the sidewalls, the wall between the two speakers, in that order. And since 500Hz is nearly an octave above middle 'C' on the piano, you are not measuring accurately much of the musical range.


I cannot see the need for expensive measurement software that gives inaccurate results, compared to how we hear. You will get far more use out of the analog test-gear I mentioned above, using less-expensive computer software as your spectrum analyzer, such as software sold by PartsExpress.com

Best,
Roy

Roy,

I looked up Zobel circuits on Wiki, but the theory got beyond me. Can you please explain what a Zobel circuit does in the context of speakers.

For example, my speakers have an impedance peak of 20 ohms at the mid/tweeter x-over point. Would a Zobel change the impedance presented to the amp.

At what cost? Less efficiency? Distortion? Does the Zobel introduce something into the circuit that wasn't there before. Nothing good comes at a cost of nothing bad.

Thanks

Bruce

Hi Bruce,

A Zobel circuit for any driver makes its crossover circuit perform more to 'spec'. Zobels can result in a flat impedance curve, making life easier for an amp, but this does not always happen.

The Zobel circuit for any voice coil is just a capacitor and a resistor placed in parallel with the driver, before any crossover is added. It is there to make that driver's impedance curve appear flat to its crossover parts, so that they work as you would want them to, in terms of 'rate of rolloff' and for your actual -3dB crossover frequency.

To determine the values for its cap and resistor, you can use an inaccurate pocket calculator equation, or you can measure the impedance curve of the driver as you try different values. This takes a sinewave signal generator and a good voltmeter. The driver under test is not in its cabinet nor hooked up to its crossover.

There are likely some internet sources for how to hook up the voltmeter and sinewave generator to measure the impedance of the driver + Zobel at each frequency. You can either plot the values on graph paper or in a spreadsheet, or just write them down.

The value at which that impedance levels off is what you then plug into a crossover-parts calculation as 'your driver's impedance'. Despite how carefully you measure, that impedance will be wrong to some degree.

That error happens because your Zobel circuit was used to flatten what you thought was that driver's electrical impedance, but you've been measuring instead its electrical + mechanical impedance(s). Therefore, you must adjust any Zobel to get what you want.

It would not affect your speakers' 20 Ohm peak because that was created by a higher-order crossover. Your speakers may have Zobel circuits built in to their crossovers, perhaps not visible in a schematic, as they may have been 'wrapped in' with the values of other crossover parts, through 'computer modeling' of that crossover.

If you added Zobel circuits to an existing speaker, most all of its crossover parts would then need to be changed. The end result may lower the impedance presented to an amp, but not enough to be of any concern. The speaker can become easier to drive, since the amp could see a more resistive load at all frequencies (= a less 'reactive' load that stores energy).

But then again, using any high-order crossover circuit in that speaker will more than negate this, because these crossovers make their own impedance curve. Smart designers can add more parts to make the final impedance curve look flat to an amplifier and to a magazine reviewer, but that's an illusion, as a complex crossover still lays between the amp and drivers.

The cost per Zobel is 'not much' and there is no loss of efficiency. No penalty at all comes from using Zobel circuits. Distortion is not increased if you use the best parts you can afford. A Zobel is 'all good'.

Best,
Roy

@Roy: Gotcha. As you say, the Paradigm Sig 8 (v3) 20 ohm peak may be high because Paradigm uses a 3rd order x-over at the mid/tweeter driver x-over point. What's that? About 24 db/octave??? Practically a brick wall filter.

Incidentally, it occurred to me that if the x-over screws up timing between the various drivers, with a 24 db x-over will frequencies covered by JUST the tweeter be out of phase within the pass-band covered by the tweeter, i.e., 2000 Hz to 45K Hz??

If not ... that's a large part of the acoustic spectrum that (2K to 20K Hz) IS phase coherent.

Just asking.

Btw, when I play my stereo, the neighborhood dogs sit on my front lawn and howl. I guess the high frequencies drive them crazy. Even my poor wife howls. LOL

Btw, btw, quality beryllium tweeter break up in the high ultrasonic range. They are super brittle, super light, and super fast. Don't know how the Be tweets compare to ribbons, but the Be tweets might give a good ribbon a run for its money.

I recall that you buy some of your drivers from Denmark. I think SEAS and/or ScanSpeak makes Be tweets. Have you ever considered using them for your speakers? Only problem I can think of is that Be is toxic. That's why I use a gas mask when I listen to my rig. HaHa. LOL

Well, I know you guys are looking to Roy, but to add just a bit to what he said... The idea of z-compensation is to control rising impedances. You can add resistance to tweeters to make higher impedances with some success as long as you figure the added impedance in your crossover, but z comp or zobel controls the amount of rise of impedance. The main importance is that your speaker is crossed at a given frequency at a given impedance (say 2k @ 8 ohms)... impedance swings can throw that out of whack... so it is possible for your impedance to actually vary as it plays. Next z comp also makes a consistent load to your amplifier. Also, everytime I have used Zcomp, it improved phasing also. I hope this helps, Tim

Thanks, Tim. Good points.

Best,
Roy

At risk of opening up a hornets nest of comment, I have just read through all of the posts on this quite lengthy and interesting thread and feel I have to add something

I do not consider myself particularly technically minded but I am first and foremost a music lover and have spent a lifetime regularly listening to live, mostly amplified music (aren't we all MUSIC lovers and hi-fi gear is just a means to an end?). Sorry about this but I get frustrated with lengthy academic debates about theory whereas surely the acid test is actually listening to MUSIC at home and referencing that to a live performance. I rarely see that word mentioned in many posts on forums like this and I post quite rarely because I enjoy listening to it too much

On that basis, I discovered DEQX back in 2012 and it has totally transformed the realism of my system. You may well judge my response as some type of placebo effect, however I know what I hear and it is like nothing else I have experienced in many years of seeking hi-fi nirvana

I am quite an experienced DEQX user and would like to comment on one or two things I have read here and maybe clarify any misconceptions (from the viewpoint of a user)

Here is a pretty basic description of the process. It can be as simple or as complex as you choose. I started off quite simply... with surprisingly good results. Now I understand the many complexities and interactions, the final result can only be described as stunning (whilst listening to music)

Firstly (actually... first check the weather forecast!) - measurement and automatic timing/phase correction of each speaker should be ideally performed outdoors in a large open space. The speakers in my case were raised around 1 metre above grass and the microphone positioned at tweeter/normal listening height at 1 metre distance away. A calibrated microphone is used of course

For subwoofers I also measure at 1m height above ground but in this case the mic is centred 150mm in front of the single driver centre

DEQX produces a lengthy rising frequency tone (for each speaker or sub, not by means of music as I had read in one of the posts) which repeats multiple times to average out other noises (ie birdsong etc...be aware that any neighbours may believe that aliens have landed nearby as this should be at around 90db which is pretty loud, with subs it is best to remove false teeth or limbs first)

This tone can be extended over many minutes if required and there is also a facility to repeat the exercise after DEQX has made the corrections and see the original frequency or phase/timing/group delay plots adjusted. At this point the frequency response is automatically corrected to 'flat' within the parameters of the drivers in that speaker. DEQX claim this correction is carried out thousands of times within the frequency response. From the finished result (see end) I can see no reason to disagree

Then repeat for the matching stereo paired speaker

All drivers in both main speakers are then calibrated by yourself as a stereo group using the frequency plot produced together with your chosen type and slope of crossover - at a point on the plot before the first reflection (using the method I describe, there is hardly any apparent reflection but easy to see it when it eventually appears)

Then, choose the type of setup you will be using, up to 6 channels in total from a single speaker, bi-amp, tri-amp or with up to 2 separate subs. Then 'create' virtual speakers on the PC with appropriate crossovers and load these into the DEQX processor in 4 separate switchable (on-the-fly) filters

Finally, at the listening position, place the mic at head position and re measure using the same rising frequency tones. The subsequent plots show firstly the damage any room will do to your once 'perfect' and flat corrected frequency response and allows subtle equalisation (I have no need for anything more than +/- 2db and only below 250hz)

Secondly, you are able to see the time response of each speaker and sub and slow the main speakers (either the whole speaker or separate driver sets) to align exactly with the subs. This makes an incredible difference to the final result once you hit the spot. Using the four presets or even changing the timing in real-time listening to MUSIC (that word again) allows you to get this spot on and you know when you are there. It just sounds holographically real

As I said, I do not go into any of the theory of sloped baffle or other means to achieve time alignment but if you take the time to fully appreciate what something like DEQX can do it, the final results certainly bear out the claims made. In fact, reading the multiple reviews replicated on their website this former cynic has to agree wholeheartedly

Now back to the real reason I did all this, but I won't annoy you all with that word again....