Sloped baffle

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....

Thanks, Tim. Good points.

Best,
Roy

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

@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

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,

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 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

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!

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

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 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

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

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!!

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

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

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.

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!

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 trying to be a troll here. I restated my DEQX point because I already have speakers. However, as Lewinskih01 kinda alluded to above, the device may not be perfect ... for all the reasons Roy mentioned, but it may get me to a much better place.

Trying to arrange for a DEQX audition.

Kudos to Roy for his well written posts and dedication to our hobby.

Roy,

Thanks for your fantastic contribution here. We can only hope for more really knowledgeable folks like you to take the time and educate us hobbyists.

On the 1st or 2nd page I posted about a way I was intending to get at this. I have, at least for this purpose, the advantage of having only an optimized computer as audio source. My plan is to use Acourate software on the server, a multichannel DAC, and independent amps connected directly to each driver, without passive crossovers. Acourate allows the use of a variety of digital crossovers, and allows for time alignment of the drivers. BUT it is limited to a single time delay between any pair of drivers, much like the limitations you describe for DEQX (which I previously considered too, but a needlessly expensive option if the only source is a computer).
Clearly this will not solve 100% of the problem - something I learnt from you. But what's your very educated guess: will it solve maybe 80% of the problem vs a non-time-aligned 3-way speaker?

BTW, would love to get your thoughts about this XO white paper by Dr Uli Bruggemann, the guy behind Acourate.

As of now I'm using B&W 804S. Obviously not time-aligned. Probably not even phase-coherent. So the setup described above would first be used with these speakers. And eventually I'm thinking of building my own speakers using top-notch drivers, the Loudspeaker Cookbook as guide. I'm a mechanical engineer and handy building stuff. Assuming I do a good job selecting drivers and building the cabinets...sounds like I'll end up with very good speakers in terms of bang for buck...what do you think?

Why not DEQX?
Bifwynne

Bifwynne,Roy has already answered your question some time back on this page 4. Did you miss reading it??
Here's a cut & paste from his 7/16/14 post:

Bifwynne,
DEQX seems fine in theory, and certainly makes a positive difference. For me, it has serious limitations because it cannot measure exactly what needs to be corrected. This leads to results that depend on the music being played and sometimes a limitation in one's seating position.

In particular, DEQX cannot see the immediate reflections from the cabinet surface surrounding the tweeter. It cannot correct properly for anything happening below middle C because of floor-bounce effects on the microphone are not the same as they are to our ears on music.
There are other issues, but to me, those are the two largest ones. I find that a much higher level of coherence is achievable passively.

here's more info from Roy on DEQX in his 7/17/14 psot to the whole group

This varying time-delay is what DEQX-type components are trying to correct, and what regular digital crossover circuits never attempt to correct (offering only fixed time delays, such as one millisecond). To correct the varying time delay, a heck of a computer is required, hence the high cost of DEQX type of gear.

Measurement issues and limitations still confuse DEQX type of gear, for two reasons- we cannot (yet) program that computer to how we actually hear on music, and that a measurement microphone cannot resolve the (countless) reflections off the front of a cabinet. If I had spent money on a DEQX, I would first place an "F-11" pure wool felt all around the tweeter, and then run the calibration routine.

Best,
Roy

more info on why DEQX has limitations from Roy's 7/19/14 post:

Putting the measuring mic for DEQX up close to a speaker is pointless (except for fixing up a subwoofer), as what the mic would then be hearing is coming from drivers at much different path-length-differences to the mic compared to the path-lengths to an ear ten feet away. We all know how walking up to a speaker changes everything we hear. Perhaps they are suggesting this for fixing one driver at a time. That has problems too, because any driver's tone balance is different at ten feet away vs. ten inches away.

Bifwynne, this is plenty of info for you to understand why DEQX has limitations & is not a panacea for time-INcoherent speakers. Don't you think?

Thanks Roy. I think I have the instinct, but not the brains for the math and physics. B'li neder (not a vow), I will teach myself the math and physics when I retire as Caeser's tax collector.

So ... I'm all set up. Nice electronics, good music ... listening to some hi-rez redbook CDs right now, ... good looking wife and ok speakers. Why not just go for the DEQX and call it a day?

Btw, just anecdotally and IMO, I think my Paradigm S8s (v3) are made of decent kit: beryllium tweets, aluminum-cobalt alloy mids, good woofies, tweets and mids are ferro-fluid cooled and damped, super neodymium magnets in the tweets and mids, 20,000 gauss magnetic flux density in the tweets and 15K gauss magnetic flux density in mids and woofies. What am I missing except time coherence?

Why not DEQX?

Here are my answers to important questions posed earlier, and some clarifications.

To the OP: Psag, you originally asked if a sloped baffle is important. Speaker designs that avoid this are instead using the phase shifts of their crossovers to make sure there are no cancellations/suckouts in frequency response. That is about all their designers look for/measure during the design phase, since they do not make any measurements in the time domain.

I think those designers would have an easier time developing their high-order crossovers if their drivers were first stepped back from each other, as on a sloped baffle, and they got rid of the sonic reflections off their front surfaces.

===

Bifwynne, at the beginning, you asked "perhaps someone could explain in layman's terms what causes speaker to operate out of phase. Does it have something to do with the use of caps and chokes in the x-over? Or perhaps the attribute of a dynamic speaker creating its own back EMF by reason of the voice coil moving in a magnetic field??

Incidentally, do all these electrical dynamics operating in tandem cause the electrical phase shifting that gives most amps a headache? "

Let us begin with the phase definition. If a speaker's woofer and tweeter were out of phase more than 'a bit', they would show a dip or even a complete suckout in frequency response, at or near their crossover point, with the microphone placed where your ear would be. As we see from Stereophile's tests, most speakers do not have this issue. So all of those must be "in phase", "phase coherent", "phase linear", or "phased aligned" As I explained earlier, that does not mean they are time-coherent speakers. As a reminder, the opposite IS true: time-coherent speakers are always phase coherent.

What makes the phase go weird?
-- In the speaker cabinet, it is from the drivers' locations/no stepped baffle, and having too many drivers per frequency range.
-- Any crossover circuit's inductors and capacitors delay the signal or advance it, respectively. Resistors do neither. A simple first-order crossover circuit has an inductor going to the woofer, and a capacitor on the way to its tweeter. At their crossover point AND ALL other frequencies, the time-delay created by the woofer's inductor is precisely offset by the time-advance created by the tweeter's capacitor. This is not possible with higher-order crossovers, because the values of their more-numerous inductors and capacitors cannot offset each other.
-- The back-emf from any driver is also a contributor to time-delay in its lower-range, whether woofer or tweeter. Thank you for pointing this out. I should have mentioned this earlier. That back-emf situation is altered by the type and size of the cabinet behind a woofer, and the size of any rear-chamber on a tweeter, and from ferrofluid in its magnet gap.
-- Any cone or dome breakups change the arrival-time as we go up the scale, but mostly we would hear ringing, sibilance, maybe 'dirt' being added to the music. Regardless, the best cones will not show a loud ringing at some frequency (as with most metal cones available in 2014) nor have a ragged frequency response in their upper ranges.
-- And yes, all these phase shifts will talk back to the amp. However, the crossover circuit's design is the primary cause of large swings in a speaker's impedance curve, above 100Hz. Those variations are 'electrical phase shifts' only. These swings in impedance do not reflect the acoustic phase at one's ear- no direct correlation.
The amp gets a headache because large swings in impedance means its output voltage (the pressure it puts on its electrons) is no longer sync'd up with WHEN those electrons are allowed to move by the crossover parts (inductors and capacitors). When the values of those caps and inductors do not offset each other, the result is exactly like pushing a child on a swing at the WRONG time.

===

Bifwynne, on the first page, you speculated on the effects of mics, of recording and mastering, processing, playback, etc.

Each of those areas has unique problems, which do not sound like phase shift from a speaker. Each process produces a time delay in the highs and sometimes the lows, but only a speaker can put phase shifts (plural) across the main tone range. Also, whatever that signal is, I see no reason for home- and studio-speaker designs to distort it more.

On that same page you asked
"How are small speaker manufacturers able to design speakers without the benefit of the R&D budget, engineers, and testing facilities that some of the larger manufacturers have at their disposal?"

For me, it's been knowledge, education, and longer, much wider experience. My talent seems to have been expressed as an ability to make the cognitive leap between seemingly unrelated factors, which then made one more link to hearing vs. measurement. All of this has led to me not needing an anechoic chamber (I can always go outdoors for that). I also found the fancy digital test gear gave misleading and often incorrect numbers, compared to analog test gear.

When a designer does not really understand the fundamental physics of how and why drivers move and respond as they do, nor how crossovers delay the signals, then their only recourse TO IMPRESS their board of directors, is the anechoic chamber/digital route, for that is what the AES and any university would also advise those board members responsible for hiring 'a great designer'. Such a designer then blames the sonic differences between his and other speakers as 'we all hear differently'. His board of directors and all reviewers and editors gladly go along with that bullcrap.

We all certainly listen for different things. But here we have found, that as a speaker is made more and more time-coherent, everyone AGREES on the sounds heard in each and every tome range. They all hear 'the bass' in the same way, etc.

===

Ohlala, on page one, the possibilities of off-axis cancellations you mention turn out to be non-issues on music, especially when the cabinet is not large, and has little sonic reflections from its surface.

===

Timlub, on page one, your speaker design is only phase coherent at its crossover point, not time coherent, as you may know. Your electrical crossover slopes work well because they are combining with the phase shifts of your particular woofer and tweeter, which I am sure you suspect. Thank you for sharing your experiences! Appreciated.

===

Bifwynne, you ask too many (good) questions! On page one you ask,

"here the ultimate Q. How can one tell whether a speaker is time and phase coherent? Critical listening? Reviewer comments? Bench test?

If critical listening is that important, the real challenge for us is, as many have written, that it is not easy to meaningfully audition speakers. So what's a person to do?

I'll ask again, how important is time and phase coherence? FWIW, ... really more as an FYI, ... Paradigm's web site states that its 'speakers have phase coherent crossovers designed so that the summed output of the drivers is completely and accurately rejoined.' Is that hype? It is true at all frequencies?"

On my website, I have suggestions on how to audition speakers. I know these work. They are simple, taking only time and effort. The time-coherence part of the audition sounds like clarity and depth, and when time-coherent speakers are designed with the best parts, the musicality is greatly improved.
With the very best, you find yourself never, ever thinking about 'the sound of the bass' or 'the highs'. Instead, you subconsciously always focus on the music and how it is being played, and its emotional and physical connection to you.
When a speaker is time-INcoherent, the music is fragmented, leaving you to hear only 'the details' and 'the soundstage' or 'the air', or 'the impact'. Right now, I see only Green Mountain Audio, certain models from Thiel, and Vandersteeen as making time-coherent speakers. The Audio Machina company is part-way there. With any others claiming time-coherence, I've seen no proof on their websites, or in Stereophile tests.

===

Ivan_nosnibor, I appreciated your thoughts, thanks. However, the time delays in your digital crossover circuits are fixed time delays for each driver, when the real problem is the amount of time delays are different at each frequency. You remark on hearing perhaps the highs 'imaging closer to you' on non-time-coherenet speakers, with the mids 'not projecting as far into the room', and so on.

I have found instead it is about the lack of depth in the highs, caused by the smearing of a late-arriving mid, and so on down the musical scale. WHEN the highs arrive is not WHEN you hear the image, but only a portion of that image. One example is hearing the esses and tees of the singer's voice arrive from the tweeter's location above the mid, not from the mid driver's location, where the main part of her voice comes from, listening with eyes closed. That is one sound of a tweeter arriving too soon. It can also sound like the band is leaning forward, for want of a better word. It can sound like the rhythm section is behind the beat (as they would be in those speakers).

===
Almarg,

Your described a square wave as "the summation of an infinite number of sine waves, one being at its 'fundamental frequency"' (the frequency with which its pulses repeat), plus others at every odd multiple of that frequency (i.e., the 3rd, 5th, 7th, etc. harmonics). The amplitude of each harmonic decreasing as its order (i.e., its frequency) increases." This is all true, but only of an ongoing series of square waves. The analysis is somewhat different when we examine just the first up-cycle, without even the first down-cycle following it. Just an FYI, seemingly never mentioned on the internet nor in textbooks.

===

Mofimadness, the Loudspeaker Design Cookbook is generally excellent, but all previous issues got the concepts of phase time-coherence somewhat wrong. It has been awhile since I looked over a copy, so I can't remember where the problems showed up. I advise to take its advice with a modicum of salt.

===

Bfwynne, the Revel 2 and Magico have oodles of phase shift, mostly from their crossovers. What you are seeing in the Stereophile tests is just as John Atkinson says- the mid and woofer take longer for their sounds to arrive. What is not readily apparent is how the phase (time delay) is changing at EVERY frequency. Otherwise, one could fix the Magico and Revel 'problems' by moving their tweeters back, etc. Actually, Almarg gave you a very excellent answer.

===

Usermanual, you ask about us proving we are time-coherent.
1) This would not change our sales.
2) It cannot be done in a singular graph or 'scope image useful to a layman, by anyone including us. This is not a case of sour grapes- please read my letter to sixmoons regarding the issues with measurements. Note some of my graphs do not line up correctly with my text on their website.

In the 1994 Stereophile test on our Diamante three-way, remember JA always measures at 50 inches, right in front of a speaker's tweeter. That makes ANYONE'S mid and woofer too far away, relative to the tweeter.

JA then moved his mic straight down, to get farther from our tweeter, closer to the mid and woofer, looking for our claim of time coherence. You see our step response get sharper, more compact. But our frequency response goes to heck because he is now going VERY far off-axis of both mid and tweeter. Again, this test was done in 1994. In the intervening twenty years, every aspect of our sound, and of any measured performance, has improved.

Above all, trust your ears more than measurements and reviewers. My letter to sixmoons shows why this has to be so.

===

This covers page one, I think. Perhaps page two will be much, much shorter.

Best,
roy

That's how I was presented this subject. A good link, thank you. Made me flash back to all the horrible homework involved. And then, as the math of physics became ever more advanced during grad school, one wound up using this math daily...

Best,
Roy

In light of Roy's feedback to the sound.westhost.com material, here is (what seems) a much better website to read up on quadrature signals.
The article deals with complex signals but it is not complicated - the graphics make it much easier to understand.
if you don't want to read the article, scroll to Fig 10 directly & you will see why 2 signals in quadrature (i.e. separated by 90 degrees of phase) add up to a constant i.e. adding 2 signals in quadrature does not give you another signal; rather it gives you a scalar/just a number.
As Roy was saying earlier on - this can happen ONLY with a 1st order cross-over where the phase difference between tweeter-mid, mid-woofer is 90 degrees & when these signals add up at the listener's ear they appear as tho' there was no additional delay thru the x-over.
http://www.dsprelated.com/showarticle/192.php

Ngjockey,

I have looked at this site for very many years. The Soundwest site has enough errors to mislead someone relying upon it for 'basic information' and a bit of the math.

Specifically:
In its Section one, the author does not understand a tweeter is still not time coherent when its wires are flipped over to invert its polarity (paragraph 3). He goes on to mis-represent the amount and degree of cancellations between mid and tweeter when the tweeter is not in the right position (below Fig. 5). What he presents instead is a graph showing TWO IDENTICAL, PERFECT, FULL-RANGE DRIVERS interfering, not a graph of one mid crossing over to one tweeter.

In its Section two, the information in the paragraph below Fig. 10, about phase shift and its audibility on square waves, is just plain wrong (even stating we can't hear it, then giving real examples of how we can hear it).

In Section four, on the audibility of phase distortion, not only is he wrong about its audibility, but he goes on to present an argument based on sound coming from live instruments.
He does not get it that we want to PRESERVE whatever phase relationships exist in the music, no matter where we sat, no matter where the recording microphones were placed. Can you spot the big flaw in his argument based on hearing live music? I have seen this exact bad-logic presented on many other forums as the main reason not to bother with making speakers time coherent.

In his Conclusions, he claims the room acoustics and bad recordings will hide much of what should be gained from making the speakers time coherent. To me, that makes it obvious he's never lived with time-coherent speakers for any length of time.
He mentions how a little pair of speakers in his workshop will reproduce a square wave at one frequency if he holds the mic in just the right place. I can see he does not recognize those speakers likely still have a phase shift of 360 degrees at some frequency, and how that will make a CONTINUOUS square-wave signal still appear square.
He does not remember that 360 degrees of shift at some frequency means the previous square-cycle is then projecting/delaying some of its frequency-components INTO THE NEXT CYCLE, and so on. He should have been examining only the first half of the very first square-wave cycle-- its first up-and-down only, to figure out what a speaker is doing.

===

His are the answers I find quite common on the web, but not in most of the professionally-reviewed papers published by the AES. Their important papers on speaker design can be purchased by anyone as their three Audio Anthologies books. There are still errors in too many of those, but one must know calculus and physics quite well to find them.


I think the general public should not take a writer's claims about audio design for granted, unless the writer also presents the scientific concepts and logic behind those concepts, and WHY those have to be correct. Which is what I've endeavored to do.

Best regards,
Roy

Looked up quadrature and unless you're masochisticly inclined to imaginary vectors, this might be easier...

http://sound.westhost.com/ptd.htm

^Please forgive me, I was confusing DEQX with Holm Acoustics.

Unsound, not sure I understand your question. You ask why the DEQX "system approach includes analog conversion rather than keeping the whole stream in the digital domain?" Perhaps ... it is because the device is inserted in between the source components and the power amp. Seems that at some point the device has to go digital to analogue in order to drive the amp.

Am I misunderstanding your question?

I hope to hear back from DEQX this week. I've got to nail this time coherence issue one way or the other.

Btw, I been shopping around for insulated 10 gauge solid core copper wire. I intend to make my own speaker cables. May cost me $15 bucks ... gasp!

Duh, I forgot to add Duntech to the list.

You are right Unsound, Audiomachina's website seems to be up-to-date (I saw one post dated June 9, 2014) so it seems that they are alive & well producing time-coherent speakers. Thanx for the correction.

And, BTW, you guys have to read these "Pearls" (of speaker deisgn) from the owner/designer of Audiomachina speakers - esp. for determined naysayers. These notes echo practically 100% what Roy Johnson has been trying to educate us with since 2002 when that time-coherence thread started by RBischoff appeared on Audiogon. Just like Roy's texts, this essay on speaker design is very well written (& I can see time & again the points that Roy has tried to drive home into us re. 1st order x-overs & time-coherence) - a superb & easy read:
http://audiomachina.com/pearls/

Hey guys I found this wonderful, very simple, animated essay from Pat McGinty (of Meadowlark) explaining time-coherence (in which he believed sincerely). Please read this (in the beginning he goes off the deep end relating the birth of the stars to time-coherence but please bear with him - he's trying to make a point:

http://www.patmcginty.com/Dbench2.htm

Bombaywalla, you are correct we lost two great proponents of time coherent speakers and both gracious gentlemen as well, John Dunlavy and Jim Thiel. R.I.P.

Bifwynne, Unsound,
found this website of Meadowlark/Pat McGinty showing off all the Meadowlark speakers. I loved the looks of the Meadowlarks due to their superb wood finishes - always such a pleasure to see!
http://www.patmcginty.com/index.html

Bombaywalla, unless this:

http://audiomachina.com/philosophy/

is out of date, I think they still are.

Bifwynne,
unfortunately all of the companies stated in Unsound's post do NOT still make time-coherent speakers. (John) Dunlavy quit making his speakers long time back & I believe that he, unfortunately, is not with us anymore (correct me if I'm wrong but that's what I remember reading in one thread).
And, Meadowlark also does not make speakers anymore. Unfortunately there was something very unsavoury that went down w/ Pat McGinty (owner/designer of Meadowlark), his company & the location where he was making speakers.
Audiomachina also does not make time-coherent speakers anymore.
I was reading an old Meadowlark Osprey speaker review on enjoythemusic.com

The Meadowlark Mantra

Three design principles underlie every Meadowlark loudspeaker: time coherence, first-order crossovers and transmission line bass............

McGinty argues articulately on his web site that time coherence is essential to long-term musical satisfaction and avoidance of listener fatigue.....

http://www.enjoythemusic.com/magazine/equipment/0204/meadowlarkosprey.htm link provided if anyone is interested in the full review.

Bifwynne, if you're in contact with DEQX, if you wouldn't mind, perhaps you might ask them why their system approach includes analog conversion rather than keeping the whole stream in the digital domain?

Bifwynne, Dunlavy and Meadowlark are gone. I'm not sure if the newer Quads qualify. Some Thiels are, for the time being...the future looks glum. I have doubts that the newer Ohms qualify. I'd hazard a guess that the HHR ExoticSpeakers do, but I haven't seen verification. I would say the same for the German Physiks and similar Huffs, at least for the models that use their DDD drivers exclusively, I have doubts that their full range models qualify. FWIW, I have confidence that the others still do.

Unsound, are all of the companies listed in your last post still making time coherent speakers. Of course, Vandersteen, Thiel and GMA are obvious. Not sure about the others.

Btw, been in contact with DEQX. Will try and arrange a home demo. If I can pull it off, I'll report back.

^oops!
I forgot the original Ohms with genuine Walsh drivers
and perhaps their progeny from HHR-ExoticSpeakers, German Physiks and Huff might qualify too.

I just remembered the name of the old poster that started a time coherent speaker company; Karl Shuemann. To be fair to all here's a list of those companies that have made time coherent speakers that I'm aware; of starting with Karl's (which fortuitously falls into alphabetical order):
Audiomachina
Dunlavy
Greenmountain
Meadowlark
Quad
Thiel
Vandersteen

Roy. thanks again. ^ Wouldn't using 2nd order cross-overs compromise the whole effort?

Certainly, but a lot of other problems would be fixed.

Roy. thanks again. ^ Wouldn't using 2nd order cross-overs compromise the whole effort?

Sounds Real,
For an eyeball estimate, the acoustic center is approximately where the voice coil former meets a cone or dome- the glue joint. But this is true only in the upper-middle range of any driver, whether tweeter, mid or woofer, where each one's frequency response is still flat.

To measure it (within +/- 1/8th inch at best for a tweeter, much more for a woofer), one sends an impulse, a click, to a driver having no crossover.

On a `scope, one examines when that click arrived compared to when the `scope's sweep was triggered by the click electrically.

Now, what we are looking for as markers will not be the top of those two spikes that click generated. We are looking for when each spike just begins to turn upwards from 0.0 at its bottom-- when each just begins to rise up. That is a very difficult transition to judge, which leads to inaccuracy.

Regardless, that time-difference times the speed of sound is your distance from the mic to the acoustic center in a driver's upper-middle range. Compare that to the tape measure distance and you often get close to the eyeball estimate I mentioned above. Of course, the test mic will be expensive, not a $200 special, for those cheaper ones have their own phase shifts in the audible range. Figure $1000 for a proper mic, plus a $1000 wideband mic preamp. Even so, the results will still be rather inaccurate. I was able to find ways around this, fortunately.

===

Unsound,
It is not that the two woofers are equidistant from their surfaces but the fact that we have two (four in stereo) woofers rather near two surfaces with you living in between.

Do have a look at that new drawing I posted to get an idea where "the bass source really is", which is my red dot in that drawing. Imagine what standing waves would then occur in between a red dot on the ceiling and one on the floor when the measuring mic/your ear is placed somewhere in between. Double trouble has been my experience.

No doubt about the outer horn-surfaces making reflections. But those reflections would go mostly upwards, and we can apply wool felt or acoustic foam to minimize most of them. I still think the biggest problem to be getting far enough away from the speakers so stand up/sit down differences would not drive us crazy- a large living room, say 30 x 40 feet is probably enough.

I would like to hear a Klipschorn corner horn triamped with time-delays applied to its mid and tweeter, since the woofer is so far back inside (~4 milliseconds) and the tweeter is so far in front of the mid driver (~2ms). Again, one would be stuck with using second-order crossovers on the drivers with the mid driver in inverted polarity.

Best,
Roy

Hi Roy, I don't mean to beat a dead horse, but I think that in a W M T M W array, it would be very rare indeed for any of the drivers to be equi-distant to floor and ceiling, and those surfaces would typically be different enough to reflect some frequencies differently as well.
I would have guessed that in an attempt at a staggered driver time coherent horn design that the horns themselves would get in the way of each other with the surrounding horns causing early reflections. Wouldn't a deep throat coincidental multi-driver have more early reflections than a flatter design?

Hi Unsound,

Thank you for your thoughts. The use of multiple subs does smooth out standing-wave issues. The math used for the theory behind that is formed from adding together the simple sinewave/wavelength equations for standing waves you have seen for bass tones and room modes before.

That is fine for long-running test tones, for movie sound effects, and certainly for a pipe organ. The test tones used to adjust those multiple subs are long-running, and not found in music.

When the time-arrivals at the ear between multiple subs are 'excessively different', you would think we'd hear stumbling or mumbling on string bass, drum kits and perhaps even cello. But if those subs are not allowed to go above ~40Hz, those issues are bypassed.

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WMTMW bass problems arise from both woofers being close to the bottom and top surfaces of our room. This is a 'very symmetrical' situation, which always produces the strongest standing waves. Another 'very symmetrical' layout would be subs placed in every corner.

Have a look at this drawing: Reflections

Also, do note that WMTMW woofers operate to 150 or even up to 300Hz, which is above middle 'C' on the piano. In these upper ranges, changes are very audible standing vs. sitting vs. walking into the kitchen.

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You ask about the over/under head effect of an image jumping when hearing live sound from vertically-large concert speakers. Good question. I can say I've never heard that problem, including from long line-source speakers. Remember, most concert sound systems are mixed close to mono, so everyone hears everything. And in most live situations, sound from a tall concert speaker comes to you from a narrower vertical angle than when at home listening to a six-foot tall speaker ten feet away.

Also, I probably did not make it clear enough before that the over/under head leakage of sound to the opposite ear is caused by the WMTMW use of double mids, not double woofers, because of those shorter wavelengths vs. the size of our skulls.

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We get reflections off any hard surface-- it matters little that a Thiel's mid surface might be flat or corrugated around its coax tweeter. This is because any 1" tweeter, without a several-inch deep horn around it, is omnidirectional below 5kHz. That means it pushes waves between ~1kHz and ~5kHz across the face of the cabinet, since they cannot escape to the rear.
So those pressures escape to the front as they move across the face of the cabinet.
Hence, reflections.

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Putting the measuring mic for DEQX up close to a speaker is pointless (except for fixing up a subwoofer), as what the mic would then be hearing is coming from drivers at much different path-length-differences to the mic compared to the path-lengths to an ear ten feet away. We all know how walking up to a speaker changes everything we hear. Perhaps they are suggesting this for fixing one driver at a time. That has problems too, because any driver's tone balance is different at ten feet away vs. ten inches away.

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Horn speakers can be made time coherent, but our best technology leads to that speaker being at least a four-way if not a five-way design, to stay far enough away from horn cutoff points on the low-end of each driver, and the high-frequency breakups which come from running a large mid high into the upper voice range, and a compression driver with a 4-inch diaphragm into the high treble. Also, with 4 to 5 horns stacked up, their vertical height would make for very strong changes as one stood up or even just sat higher.

The nicest sound I ever achieved on horns was to use the lowest order of electronic crossover possible (12dB/octave, 'second-order') on a three-way horn system. The tweeter horn was moved far back on top of the mid's horn, and mid horn `way back on top of the woofer's folded horn, to equalize the driver-to-ear distances for people twenty+ feet away. This describes a system I put together for Taj Mahal. I had to add a small amount of EQ to smooth the mids, boost the ultra-highs, and for flat output to 40Hz. Of course I had to reverse the polarity on the mid horn because 12dB/oct. crossovers need that to avoid cancellations at the crossover points.

Since everyone was 20 to 70 feet away from either the left or right speaker (mixed to mono), everyone heard a smooth blend from a speaker whether seated of standing. Sure there was phase shift from those speakers, but it was far less severe than any higher-order crossovers would have been. I received very many compliments on the ease and clarity of the sound.

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I hope everyone sees my answers are lengthy because I include WHY something is audible or will measure a certain way, so you finally get a proper technical perspective on the VARIABLES that must be considered, and also HOW they must be considered. Magazines and reviews leave out all these variables-- make of that what you will.

Best,
Roy

Hi Roy, thank you for your response.
At the risk of appearing argumentative; here on Audiogon another speaker designer has suggested that placing multiple woofers in a room at different distances could be beneficial towards evening out standing waves, if that were the case; wouldn't W M T M W speaker arrays have some advantages? I'm not aware of any speakers that are touted be time coherent having more than one tweeter per channel, are there any?
As for sound bouncing differently above and below ones head, wouldn't that be typical during live musical performances? Wouldn't the wave size from midranges and woofers (and live performers) be large enough to extend above and below a typically seated listener's head?
Thiel's concentric drivers appear to be flat, so reflections should be minimized, no?
I have no direct experience, but I seem to recall that DEQX suggests that speaker correction should be first done close to the speakers and then followed with room correction at the listening positions.
Another question if I may; could horn loaded speakers be time coherent?
Thanks again!

If you were to try to line up the drivers how would you find the " acoustic center" of a driver. In a midrange or woofer would that be about half way out from the center or would it be further out because the majority of the area of a larger driver would be closer to the perimeter?

Sounds Real-
You are indeed right about 'just tilting back the front face'. That can be enough to line up the acoustic centers of woofer/mid and tweeter, so that the drivers are possibly in their best positions to combine properly at your ears, no matter what crossover design is used. The high-order crossover circuits then put more and more time delay on the signal the lower and lower down the scale we go. That cannot be fixed.

And then to make the first order crossover work correctly, one must choose the correct drivers to begin with. I hope this clarifies a bit more for you.

Omsed-
You ask "though the difference between the woof and mid remain constant, there is a difference, yes? And that means that the wave launch of a transient will not be the same for the 2 drivers, correct? The are not time aligned, it would seem. Even if the sum of the outputs through the crossover point remains correct, are we not stuck with the constant time differential between the 2 drivers?

Could you tell me what I am missing? "

Yes, I agree. Again, with the right drivers, Zobels, and first-order crossover design, there will always be a time difference created by that constant 90-degree differential, a constant difference 'in degrees only' at every frequency we examine.

90 degrees is one-fourth of any sinewave's period. At a 3kHz crossover point, that wave's period is 1/3000 of a second. One fourth is 1/12000 of a second. This is the time-difference between the mid and tweeter at this frequency. If we choose 1000 Hz instead, the time difference would be three times longer, 1/4000 of a second.

I can only tell you that the math of "two waves of the same tone traveling out of phase with each other by 90 degrees" will measure and sound like one wave having no time delays. Perhaps you must do the math yourself to see this-- I certainly understand that feeling! Again, the key words to look up are "operating in quadrature".

Bombaywalla-
I apologize if I gave the wrong impression. T/S parameters are quite important, as they tell us a great deal about how the driver will perform in any box.

They just do not give the exact box size, which was the hope. The error can be 10 to 20% off of the correct box volume.

A real test box's performance is determined by listening and then measuring its impedance curve and resonant frequency, to find out the Qts and Fs. That tells us how close we came to meeting the T/S ideals with that test box. Then build another...

Best,
Roy