Sloped baffle

"I still wonder out loud whether speaker inductance as a function of frequency response in fact remains constant within the speaker's pass band. Indeed ... even if speaker inductance remains constant as a function of frequency, wouldn't that also impact phase coherency?"
Speaker inductance is one of the main factors that we measure to build a impedance/phase correction circuitry. The more that impedance and phase can be controlled, the easier it is on amplifiers (especially tubed).. Tubes can handle inductive loads reasonably well, but crap on themselves trying to drive capacitive loads.
Here is a thread where Al does a good job of explaining it...
http://forum.audiogon.com/cgi-bin/fr.pl?htech&1377551562&read&3&zzlMesch&&

Bombaywalla, sounds_real_audio didn't ask if the were on a sloped front or a flat front, He simply asked, If they leave the source at the same time, would they end up at the listener at the same time...
I believe his real question is "do all frequencies move at the same speed"
I'm sorry if I mis understood the question, but as it was proposed, the answer is Yes.... I'm not try to start an argument, only to head off confusion.
Tim

Bombaywalla, yes, your post is responsive and I get it. I still wonder out loud whether speaker inductance as a function of frequency response in fact remains constant within the speaker's pass band. Indeed ... even if speaker inductance remains constant as a function of frequency, wouldn't that also impact phase coherency?

I gather from your prior posts that the answer is "no" as long as inductance doesn't change. Then there will be no impact on phase coherency. Instead, phase coherency is effected only when there is a change in X-over reactance, albeit whether it is capacitive or inductive.

Al ... if you're catching any of this, please chime in. I think this is an important issue. Put it to you this way, my sense is that even if proponents and opponents of the importance (or not) of phase coherence want to argue yay or nay on the issue, it seems to me that phase shifting can't be good factor ... at best neutral.

BIF

07-06-14: Sounds_real_audio

I have a question. If an 18kHz sound left its' source and a 30Hz sound wave left its source at the same time. would they both get to the listener at the same time?

No, they would not.
If you have the driver creating the 18KHz signal & the other driver creating the 30Hz signal mounted on a perfectly vertical plane, the acoustical center of the 18KHz driver would be in front of the acoustical center of the 30Hz driver. Due to this, the 18KHz signal would get a head-start & would reach your ear 1st.
You hear this all the time at shows - the music is always "tipped up". You hear way too much high freq & the bass seems to be missing. The speakers are not time-coherent & often the drivers are not time-aligned.
If I remember Roy Johnson's paper, the woofer driver has a 90 degree phase lag in its pass band meaning that it starts to produce the 30Hz signal 1/4 wavelength of the x-over frequency later than the tweeter driver.
That's why you see sloped baffles with the tweeter on top - the furtherest away from the listener's ear. This aligns the acoustical centers of the tweeter, mid & woofer drivers to give them a chance to arrive at your ear at the same time.
hope this helps.....

Let's assume we have a single dynamic cone speaker with a pass band of 35Hz to 20K Hz. Let's forget about high frequency beaming and cone breakup. Just assume this hypothetical speaker has a flat frequency response within its pass band, as measure on axis.

Bifwynne, I agree with Ngjockey here that if your hypothetical speaker has a flat freq response between 35Hz & 20KHz then all signals in this frequency region will pass thru minimally unaltered. That's the meaning of "pass band" - frequency passes thru minmimally altered. This, of course, means that in the 35Hz-20KHz the effect of the speaker coil moving inside the magnetic field poses no issues. So, there should be almost zero phase shift in the 35Hz-20KHz region.

Is there a frequency range where a speaker is phase coherent

yes, its phase coherent inside its pass-band. In the case of your hypothetical speaker it's phase coherent within 35Hz - 20KHz.

or does phase nonlinearity increase as a function of frequency ... period??

yes, it does. And, in the case of your hypothetical speaker, phase coherency degrades below 35Hz & above 20KHz both of which are outside the pass-band of the speaker/driver.

If the answers to all of these questions are -- yes, then it seems to me using 1st order X-overs and sloped baffles is at best a rough justice engineering response to a problem that is inherent with dynamic speakers that use voice coils.

Bifwynne, I'm not sure that you realize what the benefit is of using 1st-order x-over? The benefit of 1st-order x-over is that the PHASE DIFFERENCE (not talking about the absolute phase of a certain frequency) among all the signals in the audio band (20Hz-20KHz) is constant.
So, you have a music signal coming into the speaker. This music signal is a complex mixture of many frequencies. All these frequencies have some absolute phase that is different from each other. Further, each frequency has some non-zero phase difference with another frequency in this complex music signal. So, this whole complex music signal now goes into a time-coherent speaker as an electrical signal & comes out as a sound pressure wave. The phase difference amongst all the frequencies in this complex music signal do not change (i.e. remain the same) if the speaker used a 1st-order x-over. This means that the timbre & harmonic structure of the music remained unchanged as it passed thru the speaker. No other higher order x-over can achieve this i.e. higher order x-overs change the phase diference among the many frequencies of the music signal as it (music signal) passes thru these higher order x-overs.
So, ifffffff, the solution is a moving target (as you wrote) a time-coherent, first-order x-over speaker is the least damaging (IOW, the best compromise solution to a moving target problem).
hope that this helps some.....

"If an 18kHz sound left its' source and a 30Hz sound wave left its source at the same time. would they both get to the listener at the same time?"
Yep, its not the speed of frequency, its the speed of sound. All frequencies travel at around 1000 ft per second... I'd have to look it up to be exact, but it also varies by sea level. The difference is how many times a wave will hit you.

How fast is the train travelling?

@Bifwynne

For the first part of your question, you misunderstand. Pass band is the part of the frequency the driver is covering, unattenuated, within the filter. Actually, I used the term technically incorrectly in the BSC context since that is attenuated long before the crossover point. Driver rolloff caused by inductance usually occurs out of the pass band but is still important. If a driver could, realistically cover from 35 to 20 KHz, than it would require very little inductance. There are drivers with little inductance, relatively, like the Satori MP16, but the numbers you mention are bordering on some AVR brochures :O

The second part is beyond me, even if I could understand the question.

Bombaywalla, I reread Roy's White Papers. He speaks to time and phase effects caused by speaker cone mass, suspension elasticity and damping. Nothing about phase shifting (if any) that may be caused by the inductive reactance of the driver itself, namely the voice coil moving in a magnetic field and producing back EMF. Perhaps Roy will catch my Q and share some thoughts.

If the driver's inherent inductance, as a stand alone factor, causes or contributes to nonlinear phase shifting, the challenge becomes a moving target.

Any ESLs out there that don't use X-overs??

Bombaywalla,

The DSP signal processing is touching your music signal in a very fundamental way in that the entire music signal has to go thru the DSP before you can hear it. Same deal with the passive x-over. But the difference is that you change the quality of the cap or the inductor or the hook-up wire & you can change the sound to your liking. It appears that it's not that simple with the DSP software - you cant go in there & change the code. Or, maybe I'm not thinking of this correctly?

I think you have it right; at least from my perspective. However, I don't have the skills to change caps or wire...so I'm basically stuck with what I get. Actually, software provides more flexibility here, in my case.

So, if I'm envisioning this correctly - computer digital out runs into the DSP software which breaks the audio signal into highs, mids, bass. You get 3 digital streams now. You feed these 3 streams into 3 identical DACs or 1 Pro quality DAC able to output 3 analog streams (one box would be better as everything sits in 1 chassis & has a better chance of being matched to the other analog stream). Then 3 analog streams into 3 power amps - you need to match these very well too: same input sensitivity, same gain, same sort of clipping algorithm, same dynamic headroom extension, same power output capability, same current source/sink capability.

Here I would say, not exactly. Inside the computer being used as audio server the DSP software runs as well. On the DSP software (eg, Acourate) you set up XO frequencies, slopes, delays, etc, and perform the driver measurements, do the adjustments, etc, perform digital room correction, and eventually get a sort of digital filter. Then you apply this through a convolver to the audio player software (eg, JRMC). Now the computer is outputting through USB several channels. Eight in my case/plan. A multichannel DAC, such as the exaSound e28 takes USB in and decodes into the 8 channels and outputs 8 analog signals. Simple 1-box solution!

Also the amps don't need to be identical. You adjust gain at the software level. Take a look at the article by Mitchco I linked before. It's an easy read and provides a nice view of his setup.

I have in the past toyed with the idea of multiamping, but always in the analog domain. It always seemed it was too cumbersome, needed too many boxes, and was creating new problems. This newer technology seems to be bridging that gap. Or maybe it's me convincing myself?

Thanks for the clarifications regarding driver time-coherency. Conceptually I understand it. My gut feeling is, though, that lack of coherency is at least one order of magnitude smaller than that introduced by passive XOs. Right? If so, most of the issue would be solved with said software/approach.

Thanks Bombaywalla. I read Roy's White Paper, but will re-read the sections you suggested.

Meanwhile, I just checked Stereophile's bench test report of the Maggie 3.5R and see that it is not time coherent. In fact, JA speculated that the midrange was connected in reverse polarity to the tweeter and woofer. I assume similar characteristics for the 3.7i.

Bifwynne,
I would very much like Roy J to jump in here & answer your question.....
Meanwhile, have you read Roy's white paper on "Time & Phase Coherence" on his website?
http://greenmountainaudio.com/time-and-phase-coherence/
when you read this paper, skip the initial part & read this section titled "Time Coherent Speakers". You'll see the response of the individual driver & how they add up in a time coherent speaker.
Then scroll past the rest of the material & read the section titled "Where a speaker goes wrong". I *think* you might get many answers (maybe not all) to your questions. Thanks.

@Ngjockey ... let me try to unpack what you just wrote. Let's assume we have a single dynamic cone speaker with a pass band of 35Hz to 20K Hz. Let's forget about high frequency beaming and cone breakup. Just assume this hypothetical speaker has a flat frequency response within its pass band, as measure on axis. Obviously no X-over needed here.

Now ... like all dynamic drivers, we have a voice coil, a spider, magnets, and so forth. Let's focus on your comment about the voice coil being inherently inductive. Makes sense. After all, we have a wire coil moving in a magnetic field, producing voltage and its own magnetic field. The faster it moves, presumably, the more voltage and back inductive reactance to the input signal.

Now, if a complex signal was fed into the speaker, would there be phase shifting with respect to the higher frequencies as compared to the low order fundamentals? To be more specific, say the signal was composed of a 100 Hz fundamental, plus "n" number of harmonics into the high treble. I assume this complex signal could be visually reproduced on an oscilloscope.

If the driver's output was compared to the input signal, would there be some sort of harmonic difference between input and output signals? Would the speaker's lack of inherent phase coherence be the cause of this distortion? Would this phase nonlinearity be caused by the inductance resulting from the voice coil moving in the speaker motor's magnetic field??

Let's assume the answers to my questions are -- yes?? Is there a frequency range where a speaker is phase coherent, or does phase nonlinearity increase as a function of frequency ... period??

If the answers to all of these questions are -- yes, then it seems to me using 1st order X-overs and sloped baffles is at best a rough justice engineering response to a problem that is inherent with dynamic speakers that use voice coils.

So ... where do we go from here?? Magneplaners, ESLs??

Cheers.

P.S. Bombaywalla and Al, feel free to chime in. I think I'm getting tangled up in my shoe-laces.

For individual drivers, cone woofers have voice coils and are inductive. So, yes, they do have phase shift as frequencies increase. Some are more inductive than others. Even dome tweeters have some degree of phase shift.

A first order, parallel low pass is an inductor coil with phase shift, typically 90 degrees in the pass band and more beyond. They're cumulative and that's called acoustic slope. In a 2-way, there's also baffle step compensation, which inolves a bigger inductor well into the pass band, causing even more phase shift, maybe another 90 degrees more or less. And that's just first order. Add another 90 degrees for every order over that. Basics 101.

In the next class, we'll discuss capacitors, high pass filters, zobels, notch and contour filters, all involving various degrees of phase shift. Then, on to impedance phase and reactance. Your homework is expected and there will be a test.

I have a question. If an 18kHz sound left its' source and a 30Hz sound wave left its source at the same time. would they both get to the listener at the same time?

07-06-14: Bifwynne
......The tweeter uses an ultra low mass.....
......Similarly, the mid driver uses an extremely light and strong cobolt/aluminum cone.....

Bifwynne, do you see what's happening here in the Paradigm drivers?? they are being made light-weight, rigid. Which other driver by the very physics of is light-weight? An electro-static panel driver. You make it rigid by putting a stator around it (like Martin Logan & SoundLab). You'll find that the ESL drivers are linear (flat freq response) over a very wide freq range & that really helps make ESL time-coherent speakers. Not all of them but many of them. The cone drivers are all aspiring to become like ESL drivers - light-weight, rigid.
The hope is that the drivers are out of the pix when the signal gets crossed-over.

@Bombaywalla -- got a Q. Do most drivers remain linear through their selected pass-band with respect to time delay. In other words, when pulse testing a speaker, is it just the X-over that causes the tweeter to respond first, followed by the midrange, and then the woofer?

Bifwynne, the x-over is electrical & the drivers are mechanical (the spring & weight analog that was in one of Roy Johnson's papers that Almarg pointed all of us to in a post w-a-y earlier). So, there is some phase delay thru the electrical x-over as the signal gets low-passed, band-passed & high-passed but there are delays thru the drivers themselves as well. The fastest to respond is the tweeter. More delay thru the mid & the most delay thru the woofer driver. Every driver is flat over a certain freq range before it rolls off. How wide that freq range is depends on the driver was made by the manufacturer.

is there anyway to compensate for the time delay phase distortion through the pass-bands of the drivers? Or is that analogous to unscrambling an egg. That is the damage is done ... no fixing it with more passives.

no, I believe that there is no way to fix this - once the transducer has converted the electrical signal to sound pressure it has already imparted its signature onto the sound pressure wave. The damage is done - I cant grab the air in the room & push it back onto the driver to give it one more go-around nor can I take that air in the room & convert back to an electrical signal & push it back into the amplifier for another go-around. Impossible to do. Your analogy of unscrambling an egg is a good one.

Not sure if this hit the point, but I own a self powered Paradigm subwoofer. The sub permits adjustments for loudness and frequency cut-off. But of relevance here, the sub permits phase alignment adjustments and I assure you ... it makes a big difference. Suck-out or no suck-out at the X-over point (35 Hz).

Bingo!! So, you have experienced some effects of phase alignment & seen the dramatic effect of it. You've been holding out on us, Bifwynne! LOL!! :-) OK, so you now know just how important phase is to the bass response. Imagine doing this over the entire audio band? You are now trending towards a time-coherent speaker....
You see something like this in speaker time-domain response measurements in Stereophile & SoundStage where the woofer is in phase or out-of-phase with the tweeter. you can see the suck-out in the impedance & phase curves.

@Nrenter ... you make a fair point. My non-techie surmise that the "non-linearity" you described is the reason why manufacturers use multiple drivers. I suspect that in making design trade-offs, the characteristics of the particular driver are chosen to optimize performance within the chosen pass-band.

Sorry to be tooting Paradigm's figurative horn again, but from a non-techie's perspective the drivers used in their Signature line might address some of the nonlinear concerns you mentioned. The tweeter uses an ultra low mass and hard beryllium dome. The motor uses neodymium magnets rated at 20,000 gauss at the voice coil gap .... (btw, is that a lot??). Plus ferro-fluid for cooling and low distortion.

Similarly, the mid driver uses an extremely light and strong cobolt/aluminum cone. The motor uses neodymium magnets rated at 15,000 gauss at the voice coil gap. Plus ferro-fluid for cooling and low distortion.

And that's enough tooting for Paradigm. I don't work for them and they certainly don't pay me.

But another manufacturer who seems to put a lot of thought and effort into their drivers is Magico. Been doing a lot of reading about their S speaker line. Build quality seems superb. And there are many other fine manufacturers who put their heart and soul into what they design and build.

@Bombaywalla -- got a Q. Do most drivers remain linear through their selected pass-band with respect to time delay. In other words, when pulse testing a speaker, is it just the X-over that causes the tweeter to respond first, followed by the midrange, and then the woofer?

Regardless of the answer, if a manufacturer chooses to use a high order X-over for design considerations, is there anyway to compensate for the time delay phase distortion through the pass-bands of the drivers? Or is that analogous to unscrambling an egg. That is the damage is done ... no fixing it with more passives.

Not sure if this hit the point, but I own a self powered Paradigm subwoofer. The sub permits adjustments for loudness and frequency cut-off. But of relevance here, the sub permits phase alignment adjustments and I assure you ... it makes a big difference. Suck-out or no suck-out at the X-over point (35 Hz).

Cheers,

Bruce

Lewinskih01
....What would be premium brands sound-wise?....

Well not sure how many of these brands are still accessible to the public but.....Scanspeak makes some very good drivers, another excellent brand for woofers is Audax & Peerless, yet another brand for mids & tweeters is Morel & Eton. I'm sure that there are many others.

BTW, I never thought about drivers not being time-coherent..... I thought time misalignment was between/among drivers.

yes, you are right - drivers in & of themselves are not time-coherent. Drivers are linear (wide frequency range of operation) well above the frequency at which you cross them over. Using such drivers greatly helps to manuf time-coherent speakers because the driver itself does not come into play, it's just the electrical x-over (or in your case the electronic x-over since you will be using DSP).
yes, you are correct - time misalignment is between/among drivers.

But you made me remember about Meridian's approach. I will look into it. I believe they deliver a digital signal to the speaker and then convert it to analog inside the amp. I'll check if they have processors that deliver multiple analog channels,

:-) that's the point of these forums. Yes, you are correct - they do deliver a digital signal to their speaker & convert it to analog inside the speaker box. Pretty complicated stuff w.r.t. all the signal processing they do. How long has Meridian in business? I would say some 40 years. How many people own & appreciate Meridian speakers? I personally don't know any. Doesn't mean that there aren't any/many. Also check into Emerald Physics' methodology.
Just a thing to be aware that you are putting all your trust into that DSP software & the handles it gives you to vary x-overs & slopes, etc. I hope that you like the exact flexibility that is given to you & that you are not saying "I wish this software had this other XYZ flexibility".
The DSP signal processing is touching your music signal in a very fundamental way in that the entire music signal has to go thru the DSP before you can hear it. Same deal with the passive x-over. But the difference is that you change the quality of the cap or the inductor or the hook-up wire & you can change the sound to your liking. It appears that it's not that simple with the DSP software - you cant go in there & change the code. Or, maybe I'm not thinking of this correctly?
So, if I'm envisioning this correctly - computer digital out runs into the DSP software which breaks the audio signal into highs, mids, bass. You get 3 digital streams now. You feed these 3 streams into 3 identical DACs or 1 Pro quality DAC able to output 3 analog streams (one box would be better as everything sits in 1 chassis & has a better chance of being matched to the other analog stream). Then 3 analog streams into 3 power amps - you need to match these very well too: same input sensitivity, same gain, same sort of clipping algorithm, same dynamic headroom extension, same power output capability, same current source/sink capability.
There are a lot of variables to contend with. That's why I wrote earlier that it's a huge undertaking. You'll have to become very savvy with DSP (which is discrete-time) & analog signal processing (which is continuous-time).
There aren't many people who make speakers using this technique - Meridian, Linkwitz DIY, Emerald Physics that I can think of. maybe this is the wave of the future???

BTW, I never thought about drivers not being time-coherent. What does that mean? I thought time misalignment was between/among drivers.

Let's look at just 2 of the physical components of a driver (imagine a midrange driver): the cone and the suspension (the surround and the spider).

The cone itself has mass, and because of the kinetic energy of a cone moving outward, it sometimes just can't reverse direction as quickly as the signal is asking. Sure it responds, but with a very slight delay. This delay is one source of phase distortion.

The suspension not only centers the coil in the gap, but defines the resting position of the coil within the gap. Don't believe me? Gently push in on a mid-woofer, and let go. The speaker pushes back out to the resting position as defined by the suspension.

Now, small movements around the "resting point" all experience the same (minute) amount of resistance from the suspension. This is by design and is where the driver operates with minimal effective phase distortion.

Now, midrange drivers require larger strokes to produce lower frequencies, and the suspension applies more (nonlinear) resistance against the movement of the driver as the driver approaches X-max, imparting another set of phase distortions.

This is an oversimplification of the subject. But shows that (good) time-and-phase aligned speakers require more than just a sloped baffle and first-order crossovers.

what is your definition of "premium drivers"? Cost of the driver? Cost of a commercial speaker using this driver? The marketing hype surrounding that driver that makes you believe it must be the best?
From the little I know, some Scanspeak drivers are very good performance that would qualify for time-coherence.
Accuton drivers need not apply for time-coherence.
I don't know much about Raal.
Be careful how you choose your drivers - don't let cost be the judge - look at their freq bandwidth & where you intend to cross them over.

Bombaywalla,

Very fair point. Honestly, I have not done much research on drivers yet. I was trying to provide examples to show what I meant and used the brands that are usually mentioned in reviews as premium. I did look into their prices, and in a way I was think along the lines of price. What would be premium brands sound-wise?

BTW, I never thought about drivers not being time-coherent. What does that mean? I thought time misalignment was between/among drivers.

I have not even started looking into building a speaker. I am now thinking through / coming to terms with moving away from a nice digital and analog chain (Audiophilleo with PurePower going into a Metrum Octave, going into a Lamm LL2) and into a multichannel DAC driving multiple amps directly and using software for volume control. I'm sure many can relate to having second, third, and forth thoughts on such a move.

The benefit of driver time and phase alignment seems to be significant. The benefit of multiamping I believe is well documented, but the challenge is on the implementation. Digital room correction also makes sense to me.

After taking this plunge next step will be thinking which amps to get to drive my existing speakers. And later I will look into building my DIY speakers. Nevertheless I wanted to chime in here to provide a different approach for achieving time alignment.

BTW, yes, in a way this is similar to the Meridian path. But I do like tubes!!! So my idea is a SS or class D amp for woofers and tubes for midrange and tweeters. And I also have two power subs. But you made me remember about Meridian's approach. I will look into it. I believe they deliver a digital signal to the speaker and then convert it to analog inside the amp. I'll check if they have processors that deliver multiple analog channels, but I'm also wary of spending big on digital components considering it is not yet mature and hence evolving so fast.

As I said before, great thread!

Lewinski... great post. I was impressed by the various digital analyses and corrections that the software was able to effect. Hopefully, speaker manufacturers will be able to achieve greater time and phase alignment, driver linearity and low distortion by designing better X-overs. It may be that the best solution will be an active crossover that can effect the various functionalities that were the subject of your post.
...
After all, not everybody is a former NASA rocket scientist or a Steven Jobs/Bill Gates computer genius.

Thanks Bifwynne.

Not sure if you were implying I'm a computer genius. But to clarify it just in case: I'm certainly not!!! Not even very savvy, honestly! It looks a lot harder than what it is. I just have a dedicated server with JRiver and the Audiophile Optimizer running in it. The hardware is optimized. Amazing sound. And it was a set it and forget it setup.

Cheers!

The DIY forums and this one have something in common. Very rarely does anybody listen to any advice or criticism that they don't want to hear.

Lewinski... great post. I was impressed by the various digital analyses and corrections that the software was able to effect. Hopefully, speaker manufacturers will be able to achieve greater time and phase alignment, driver linearity and low distortion by designing better X-overs. It may be that the best solution will be an active crossover that can effect the various functionalities that were the subject of your post.

I realize that Richard Vandersteen does this on his high end speakers to some degree, but it sure would be nice if it would be more plug and play. After all, not everybody is a former NASA rocket scientist or a Steven Jobs/Bill Gates computer genius.

Thanks.

I'm a fan of time-and-phase coherent speakers (and a GMA owner). Just want to throw that out there...

One correction to the conversation - single-driver speakers are NOT time-and-phase coherent. They would be if the speaker operates as a perfect-piston over the entire frequency range, but this is not reality. At certain frequencies, the driver can't respond as quickly as a perfect-piston, and the phase starts to lag.

Proving that even those who totally believe in time and phase correct speakers can love a non-coherent speaker, read the following.

The Tannoys are not time and phase coherent. The higher order crossover prevents phase coherency, and that they are not time correct shows in every review where there is an impulse test, such as the review that these comments by John Atkinson were in:

"In the time domain, the Tannoy's impulse response (fig.6) looks typical of a design that uses a high-order crossover. Indeed, the step response (fig.7) confirms my suspicion from the impulse response that the Churchill is not time-coherent, despite its use of a coaxial drive-unit that places the tweeter diaphragm close to the acoustic center of the woofer. The tweeter output arrives at the measuring microphone first, followed by the woofer output."

Note that guy who argues heavily for time and phase correctness, Lewinskih, previously wrote, on this forum:

" the Tannoy studio monitors sound simply superb. They sound cohesive like the sound is cut from a single piece of cloth. The concentric horn-loaded tweeter is superbly integrated with regular cone woofer. The sound is very real. I've paired it with a tube amp & this combination seems to be a winner to my ears. The dispersion pattern of the speaker is 90 degrees the way the woofer is made & because of the horn loaded tweeter. Hence these speaker care much less whether they are mounted high up or sitting on the floor. I've actually tested this when they sat on the floor - the images were all up at my seated ear level!
I've tried these speakers with my s.s amp as well & they sound very good there as well. Realistic sound, excellent imaging, extended highs"

Since he waxes poetic about a design that has never time nor phase alignment / coherency, one can only conclude that these characteristics are not the "be all, end all" that many espouse.

Lewinskih01,
it appears that you are going down the path of (Boothroyd-Stuart) Meridian (the UK-based company). If you were able to stuff your amps into your speaker, you'd have an active loudspeaker like Meridian's along with your DSP x-over. OK, so now your are not listening to passive x-over components; you are now listening to the sound of your DSP software which is processing your music signal & creating delays to align the sonics at your ear/destination.
when it comes to using a DSP x-over another company called Emerald Physics is also using this concept. I've listening to their CS2 & CS3 speakers quite a bit - both at shows & at a dealer's place. Somehow I never took to their sonics. It also did not help that a new revision of the DSP x-over arrived every week or every couple of weeks with the pledge that it was an improvement over the prev rev.
IMO, with DSP x-over you sonics will be heavily influenced by the software (very much akin to having an oversampled/upsampled DAC - here again, the quality of the sonics is heavily dependent on the upsampling/oversampling algorithm. You already know for yourself that there are some oversampling/upsampling DACs you like & others you do not).
I personally think that it's much easier to overcome the sonic short-comings of passive x-over components than it is of the DSP software.
At any rate this post was to cite the trade-offs (which I'm sure you already know).
I applaud your effort, which is a big one - biamping or triamping & getting all delays & phase of the music signal correctly lined up. I sincerely wish you all the best. Do keep us Audiogon members posted on your progress.

It seems premium driver (top Raal, Accuton, scanspeak, etc) can be had for relative low prices (compared to speakers that carry them).

what is your definition of "premium drivers"? Cost of the driver? Cost of a commercial speaker using this driver? The marketing hype surrounding that driver that makes you believe it must be the best?
From the little I know, some Scanspeak drivers are very good performance that would qualify for time-coherence.
Accuton drivers need not apply for time-coherence.
I don't know much about Raal.
Be careful how you choose your drivers - don't let cost be the judge - look at their freq bandwidth & where you intend to cross them over. FWIW.

07-05-14: Kiddman
Yes, I do doubt your experience and you sure sound like a guy with no technical education and little technical aptitude. Anyone who is fixated on one aspect of design and thinks it guarantees something is usually one who has little technical experience or knowledge. Someone who has experience and physics and engineering in his background always knows designs never hinge on one parameter or feature.

listen, Kidboy, if you think that I have no technical education or background then you are deeply negative in that area! I had a good laugh when I read the above...
the more you write, the more you put your foot into your mouth. At this point you've swallowed your 1st foot & your 2nd foot is well on its way down. Like I wrote before, you are totally clueless on this subject matter.
Time-coherence is not a "parameter or feature" of speaker design; it's a speaker design philiosophy. The designer 1st decides if his/her speaker is going to be time-coherent or not. Based on this decision, he/she selects drivers, x-over topology & then determines to solve all the other issues in designing that speaker under the umbrella of time-coherence.
You are far from getting that this concept. I suggest that you change your moniker to 'more_than_clueless' (BTW, you are the one who started insulting various Audiogon members & I'm just returning the favour as I wont sit back & take your sh$$. you are a most unsavoury fellow who doesn't know how to debate a topic without insulting people. That's why I wrote - if you are going to uncivil, go find another place to waste your time. Other Audiogon members do have disagreements but we all try our best to remain civil).

the Vandersteen 2 are time and phase conherent.

And that surely does not make it a state of the art speaker, like it makes no speaker state of the art.

And, look at your depth of knowledge on display here to the rest of the A'gon community! Your writings repeatedly say that just because the Vandersteen 2 model sounds bad that selecting time-cohrerent as a design "parameter" will not make any speaker sound its best. Wow! diffident mentality here. The Vandy 2 is a really old model speaker & it's very possible that Vandersteen was limited by the driver technology available back then. It's only recently that he started drivers made to his spec - maybe he realized the limitations of what was available to him commercially? I know that a lot of the manuf who make very good drivers have stopped selling them to the public. I had a friend who owned a pair of Vandy 2 which I heard for a short period of time & long ago & not enough to make a judgement on their sound.
Once again, time-coherence is design philosophy & not a design parameter. have you heard any other time-coherent speaker? Or, are you basing all this on the Vandy 2 speaker?

Time coherent speakers are not easy to make esp. with cone drivers that's why you have very few manuf in this arena. Your pee-wee brain has informed you that it's because time-coherent speakers don't sound good so manuf have dropped the idea. Wow! Perhaps it could be these speaker manuf incompetence in understanding time-coherence & translating that to a product that can be sold that has prevented them from manuf a time-coherence speaker?? Nah, that possibly cannot be the case, right??

Lewinskih01, your plan is great. There is so much info about making speakers in real texts, you will be surprised that it is not magic. First thing, yes, use the best drivers you can. Check out Audio Technology, they are some of the absolute best.

And sure, the prices are low compared to finished speakers.

Cabinets are time consuming, finishing is time consuming, this labor has to be accounted for to the tune of $100 per hour or so, all parts have to have markups, there is dealer markup. Without any gouging, prices escalate quickly.

You will learn so much in a diy endeavor, and you will end up with a good set of speakers if you research and execute well.

Start reading the DIY forum. You will find a number of folks who really know what they are talking about. Fewer "know it alls", But lots of guys who really do things.

Just saw this thread. Wonderful to have so many knowledgeable folks chime in, plus the links to very good past discussions.

I'm certainly not up to par with my two cents here, but Psag might find it useful. Uli Brueggemann, the man behind Acourate DSP/DRC software, wrote this article on crossovers you are likely to find enlightening. It is in layman's terms: http://files.computeraudiophile.com/2013/1202/XOWhitePaper.pdf

Not sure what your system configuration is. Mine is based 100% on a computer server as source, which allows a neat approach - in my view, of course:

One way to achieve time and phase alignment is to use a multi-amped system (someone already said this above), having one amp directly driving a driver (no passive crossover used), and having a multichannel DAC and DSP software such as Acourate. Acourate allows to set digital crossovers and set time delays. So you can achieve time alignment without a sloped baffle.
Here's a great setup article http://www.computeraudiophile.com/content/556-advanced-acourate-digital-xo-time-alignment-driver-linearization-walkthrough/

I'm starting to go down this route, although I'm still coming to terms with the notion of the benefits of a time and phase aligned system where the amps are driven directly by a DAC (with the drawbacks of the latter) outdoing the benefits of my Lamm preamp driving the amp.

BTW, would like to ask a side question taking the advantage of so many knowledgeable guys reading this thread: following the above, my thoughts are of eventually replacing my speakers with DIY speakers using premium drivers, without passive XO, and enclosed in a DIY cabinet (I'm rather skilled at that). It seems premium driver (top Raal, Accuton, scanspeak, etc) can be had for relative low prices (compared to speakers that carry them). Does this sound like a good plan, or am I missing a significant issue??

Great thread!

Bombaywanker, the Vandersteen 2 are time and phase conherent.

And that surely does not make it a state of the art speaker, like it makes no speaker state of the art.

Yes, I do doubt your experience and you sure sound like a guy with no technical education and little technical aptitude. Anyone who is fixated on one aspect of design and thinks it guarantees something is usually one who has little technical experience or knowledge. Someone who has experience and physics and engineering in his background always knows designs never hinge on one parameter or feature.

Usermanual and some others have it right, they recognize that this thread is only talking about one aspect of speaker design.

Bifwynne, yes, I think you have the list.
Single-driver speakers are also time coherent (since they dont have a x-over to begin with) but they might not have the freq range extension you are looking for.

Some of the latest generation Martin-Logans might also be time-coherent (they claim to have made big strides in integrating their woofer with their ESL panel) & the full-range CLX.

Quad speakers are also time-coherent such as the ESL-2085 & they might other models (ESL-989?)

Another brand is Eminent Technology LFT 8. They might have a latter rev of this model, not sure.

Yet another brand would be Sanders Sound Systems 10C & 11 ESLs. You'll find measurements of the Innersound Kaya & Eros Mk3 speakers on Stereophile if you search. Innersound speakers were basically made by the same person who owns Sanders Sound Systems today. I realize that I'm extrapolating since Innersound Kayas were time-coherent that Sanders Sound Systems 10C/11 will also be. This is based on a reasonable assumption that the same designer has not changed his philosophy when he started his new company. Atleast I did not get this impression when I spoke to him in Dec 2013/Jan 2014.

I'm almost willing to say that SoundLab ESLs are also time-coherent but I might be wrong here. Not sure.

That's all I can think of right now. If I think of more brands/models I shall post. Thanks.

Bombaywalla, please list the major speaker brands that are time and phase coherent. At this point, I am aware of three brands: Vandersteen, Thiel and GMA. Are there others?

The reason I ask is because I'd like to check area dealers who sell time and phase coherent models and maybe do some comparative auditioning. The other alternative is audio shows.

Thanks

07-05-14: Kiddman
For a ubitiquous speaker that shows good time alignment, look no further than Vandersteen Model 2............
So there you have a great example: a manufacturer that makes a barely passable (to my standards) time coherent speaker........

Kiddman, you are screwing up again!!
In your 1st sentence you wrote that the Vandy Model 2 has TIME ALIGNMENT.
In a sentence much you lament by saying that the Vandy Model 2 is TIME COHERENT (which is wrong) & how could it be so bad sounding.
The Vandy Model 2 is time-aligned & that's it. The Vandy Model 2 (therefore) is NOT time-coherent.
Time aligned speakers are NOT necessarily time-coherent.
The other way is true - time-coherent speakers are time-aligned.

Ever since you participated in this thread, you have been NOTHING but negative - casting doubt on this subject matter & being insulting - and, yet, you have contributed NOTHING & no information to this thread/subject matter. By reading your posts, other Audiogon members gain no new information except determine that you are a stubborn 'nay-sayer' with perhaps little experience. If you have no positive contribution to make, go find another place to spend your time rather than driving off the other members who come here to learn something new & different. Your negativism benefits nobody....
And, don't cast doubt on my experience & education, you jerk!

Kiddman,

You are absolutely right. Solving one issue (If at all, in this case) , while creating many others is a far cry from good engineering, or good sounding loudspeakers. People who get stuck behind “critical” issues, usually do not see the entire forest. Move on, you are wasting good ink.

For a ubitiquous speaker that shows good time alignment, look no further than Vandersteen Model 2. A fair speaker for the price, but a hooded, somewhat grainy sound in the mids and highs with bass that sounds like a cardboard box. So time alignment it has. OK sound for the price. But nothing more than OK. If time alignment were so important, how can this speaker sound so ordinary, so mediocre?

Because extension matters, driver resonance matters, driver distortion matters, driver symmetry of motion matters, overall harmonic distortion matters, intermodulation distortion matters, box colorations matter......and we can go on and on.

So there you have a great example: a manufacturer that makes a barely passable (to my standards) time coherent speaker that I would never own, and he makes a fantastic, state of the art speaker that I would be happy to own. Any more demonstration needed that time coherence is not the most driving factor in the sound?

Wasn't Jon Dalqhuist's DQ-10 an early attempt at time coherence? And, ditto, Wilson's Franken-speaker, the Whamm?
It's been so long since I heard either speaker I couldn't say how either stands up today. My concern, in theory, would be that multiple drivers, with a bunch of different crossovers, adds more complications to the affair. But, I guess, as they say, in practice, theory and practice aren't the same.

I don't doubt his sincerity and efforts either. Some great products are made by such sincere guys making large efforts. And many more lousy ones are.

I'm going to seek them out for a listen.

I have no doubt Roy is sincere and his efforts are genuine. I don't have enough information to intelligently debate his views either way. Beep is an interesting phenomena in itself.

Bombaywalla, you've never designed anything, have you, or you would know what I said is true. A very poor speaker can be made that is still time coherent, and if you can't get that far in your brain you have little experience and education.

You may feel insulted....but that does not mean I'm intending to insult you.

I can reprint papers on mixture flow in internal combuston engines, but that does not mean the heads I flow are perfect. It only means I can write theory. Self-published graphs and dyno runs done by me don't prove that they were the runs for that motor, and that the science I can read, then write papers about, ensured that my engine is the best.

I would love for your speakers to be the best, that would represent an improvement. Which upcoming show will you be playing them at? Which top electronics manufacturers are using them? Surely they must be making a splash in the industry if they are that great. I simply can't wait to hear them. Tell me where.

Kiddman

07-04-14: Kiddman
........I have to submit that time coherence is not the driving factor in speaker sound.

Put another way: an absolutely horrible, highly distorted speaker that is time coherent could easily be made,.....

Kiddman, I'm afraid you are quite clueless & remain so. you really have no idea, do you? The more you write, the more ignorance you show in this matter...

Kiddman, no one benefits from your insults.

If you did the math, or at least read my technical papers and relevant papers in the AES Journals, and above all hear what everyone we know hears, you would agree with me, no doubt.

You could read my letter to six moons describing the problems that measuring speakers presents, and WHY each measurement technique has particular problems. No Roy opinions there-- just scientifically-tested facts accepted by the AES.

The complete sets of measurements we post on our website for our speakers are far more detailed than any others anywhere out there.

You are wrong about being able to make a highly-distorted speaker somehow time-coherent. Its drivers themselves would not even be minimum-phase to begin with (that is, well-behaved) to be able to employ the required first-order crossover.

Best regards,
Roy

Remember, that long post is essentially a non-scientific, non-specific, biased piece of salesmanship by a guy who builds products that allegedly conform to this behavior.

I have no horse in the race. But I do like some speakers in each camp...those that conform (proven by measurements) and those that don't. I have to submit that time coherence is not the driving factor in speaker sound.

Put another way: an absolutely horrible, highly distorted speaker that is time coherent could easily be made, and great ones that are not are also made.

Again: listen with your own ears.

Thanks for chiming in Roy. :-) Thanks for the explanation in your long post.

Thank you!

Hi all,

Sorry to have delayed this post-- unexpected duties arose.

I hope the majority will be served by some words below, along with a close study of the diagram I've posted at

http://s1374.photobucket.com/user/greenmountainaudio/media/TimeCoherenceDiagram_zpse8c92f2a.jpg.html?filters[user]=140737398&filters[recent]=1&sort=1&o=0

For whatever reason, I cannot get this to post as a clickable link here, sorry. Perhaps someone else can! At least Cut and Paste works, so please open this image in a new window/tab and magnify, as it illustrates much of what takes too many words to explain.

Here we go:
When a speaker spits out a brief piece of sound, making just a "Beep" then falling silent, what is moving towards us is a chunk of higherthenlower ("wavering") air pressure. The air itself is not going anywhere.

I encourage you to conceive of this as a traveling packet of sound, silent before and silent after, a packet that contains perhaps six wavelengths (six cycles) of a pure tone just like what is emitted from a tuning fork.

However, that one "Beep", high or low ("Boop"), is not a perfectly accurate example for a 'pure tone'. We must imagine instead that "Beep" stripped of its B and p, leaving only the eee or ooo.

That simple burst of an 'eee' or 'ooo' still conveys useful information- perhaps to warn of a car door ajar. Its message comes from its possession of just two characteristics:

1) It has a unique tone, high or low on the scale,
2) Lasting for a unique period of time.

Thus, to make and then hear any message takes both tone and time. Time is important to our sense of hearing.

Non-time-coherent speakers delay bass tones more than voice range tones, and those tones more than its highs. Another way to state this is their highs always come out first/too soon. This cannot be completely fixed by digital delays nor by stepping the tweeter back from the plane of the mid, because the amount of time delay is DIFFERENT at EACH frequency, which also leads to serious measurement difficulties for most designers.

Perfectly time-coherent speakers do not delay ANY tones whether bass, voice or treble.

Designers of non-time coherent speakers quote studies showing we cannot hear less than a 2 millisecond difference in arrival between the voice range and the high treble.

This means they believe it is OK for the mid's voice range to arrive up to ~60cm (two feet!) after the tweeter's highest tones. And even greater offsets/longer-delays are OK in the bass.

When those tests first came out, I saw they misled by only using tones that do not mimic the complex sounds of music, nor even resemble sounds anyone has grown up experiencing everyday naturally, so untrained listeners instinctively would know how 'it' is supposed to sound.

For myself, after years of being intimately near to the artistry of very many world-class musicians and singers, each for long hours, sometimes for days on end (rehearsals), I think "Who are engineers to say time delays are OK-- that somehow those don't screw up the music?"

I have no doubt everyone has watched even simple music bounce along on the computer screen. Know that we are observing only a miniscule faction of that music's waveform.

When a loudspeaker injects time delay, the shape of that waveform changes. The 'wave envelope' changes.

What is in that one shape, inside that one envelope? A zillion different sounds, each occurring with its own unique loudness, tonality, onset, duration, and decay. In that shape also lies the ebb and flow, the give and take, the emotion of the music, and the texture of each sound, its unique timbre ('tam-burr').

And we only get to see and record that one complex wave-shape per channel. So I think best to strive not to change it and hear what happens.

When this time-delay-as-we-go-lower is progressively removed from a speaker's design, along with the sonic reflections off its cabinet surfaces, then we always hear from any recording more and more the sound of people playing music 'over there'. There are no microphones. Our attention is no longer drawn to "the details" such as "the sound of the bass", "the airiness of the highs" or the sharpness of images.

Instead, even an inexperienced listener soon focuses instead on HOW the bass player is responding to the others (and thus WHY). All of the hi-fi 'details' are still there but now serve to shape the tones, to give each sound precision and purity (or perhaps a wandering pace and a fuzzy tone) RELATIVE to all the other musicians' sounds. Also, each musician remains distinctly separate in space, regardless of the music's complexity.

Ebb and flow, sudden changes- all are part of music, and hearing those makes sense to the ear. A much wider range of music is enjoyed, with little effort. Music FEELS good, just as if you were a teenager again, before becoming caught up in hearing all the very cool and entertaining 'hi-fi details'. You also do not need to turn it up.

By the way, speaker designs have become far more time-incoherent since the 1970's. Marketing pressures combined with the appeal to designers of 'new technology' has led to the use of many drivers that require high-order crossovers to operate.

This has given the majority of audiophiles, reviewers and designers loudspeakers that present "details" instead of music ("I can't stand loud rock recordings from these expensive speakers!"). They do not hear this as being a problem due to time-delays for any or all of several reasons:

a) They have not gone to enough intimate concerts, live theater or recitals, or sat in a living room for hours listening to singing or an acoustic guitar, clarinet, piano, a violin, played superbly. In that intimate environment, such sounds are to die for; far more breathtaking that us mere mortals can imagine- until we hear 'it'.

b) They play only a limited variety of recordings to evaluate gear, much of those recordings electronically manipulated, even of acoustic instruments (I read the reviews).

c) They have collected some non-musical gear and cables, as there is a lot of it out there, some very expensive. Very clean, but sterile, devoid of musical flow.

d) They have been told over and over again that time-coherence does not matter. No editor wants to piss off any large speaker manufacturer that uses only high-order crossovers.

e) They have lived with only very time-incoherent speakers, never with original Quads, nor electrostatic or planar headphones.

f) They do not know how to work the math of speaker design from a time domain perspective, and I don't like it either.

g) They have ears of cloth, for which no amount of exposure to live music can help. Fortunately, I "had to take piano lessons" as a youth, enjoying it and eventually playing much music well enough as an adult, from Mozart to Bartok to Joplin, to know how terribly bad I still was compared to any prodigy! Now I am long out of practice, as the piano needs new strings and new action ($$$)-- an Everett upright grand from 1887, weighing 600 pounds with a bronze frame holding a spruce sounding board. Sorry to digress.


So, we have that "eee" from the speaker coming towards us at 343 meters per second, no matter whether it's a bass tone or a treble tone. It will arrive at our chair in about 10 milliseconds, to begin to push or pull on our eardrums, because we are sitting about 3 meters away. Unless the speaker delayed when this sound came out.

A jet liner up high is cruising at ~85% of this speed. Sound is about ten times faster than cars on a distant highway. It is the speed of that blast of pressure coming at us from an atomic bomb. We could see ordinary sound traveling if the air weren't so clear.

Non-time-coherent speakers create time delays by their choice of drivers, those drivers' locations from your ear, and the type of crossover circuits used.

Drivers have both mechanical time delays and electrical delays, as they are Transducers, which operate in both domains.

A tweeter may be stepped back from the mid, to "put it perfectly in phase with the mid" at the crossover point only. This still does not make for time-coherent operation (study my diagram and my other Audiogon posts in the links others supplied above, thank you).

ALL crossover circuits introduce time delays, but only first-order crossover circuits create time delays that naturally offset each other, when crossing from the tweeter to the mid, and mid to woofer, thus producing no RELATIVE time delay between the drivers, which is the (my) goal.

I hope this helps! For those wondering what the step response or the impulse response indicate in Stereophile, I advise you to study John Atkinson's explanations of them and to remember that, in those tests, nothing nearly as low 'C' above middle 'C' is shown. That takes a large, expensive anechoic chamber or careful measurement outdoors with the speaker up on a pole, far away from the ground, using a very loud pulse, one that likely damages any tweeter.

The detractors of first-order speakers talk of power-handling issues caused by the slow rolloff of the crossover allowing bass to get into a driver, even a tweeter, making it distort or melt. Maybe the mid's cone would ring in the tweeter's range, because of its strong resonance at a high frequency. For us, there has been no problem because we use the best drivers, by anyone's standards.

Detractors also claim there will be off-axis cancellations between drivers, leading to a weird tone balance for a listener off-axis. Not true with proper crossover points, slender cabinets, and a lack of cabinet reflections. Never is the math behind those claims shown, as it never supports them.

To the original poster- thank you for this opportunity, and know that one reason a tweeter is placed behind the plane of a mid is because the sound from that mid emerges first from down deep in the center of its cone, no matter the crossover slopes used nor the mid driver's design.

Best regards,
Roy Johnson
Designer
Green Mountain Audio

^+1:-)!!!!!
That's exactly how I've feel.

I have a little experience with owning Thiel 1.2's. I think they were suppose to be phase and time coherent. At home I have never heard a speaker do most things as well as them. It seemed to me the musical cloth was all from one piece. No discontinuities. I could follow the scale up and down the piano without a noticeable bulge or reticence on a certain frequency, and not imagine it was real, because it sounded as real as it could get as a recording, even though that speaker was not state of the art even in Thiel's lineup. That allowed me to enjoy the flow and nuances of the musical presentation better than anything I have ever had. Some speakers convey an instrument truer in some senses than my Theils to me, and some have conveyed the recording venue better, or spacing of images, macro dynamics, and other attributes better, but not the whole musical package. To me the Thiel's presented a realistic presentation of the whole. It was to me, like everything presented fit. It doesn't seem to do it justice to break it down in descriptive terms but it seemed relatively speaking it didn't get more real the that.
So I have always been curious to hear the sound of Green Mtn spkrs. Maybe someday. I have heard Vandersten 2ce's, so while I didn't care for the warmness of the spkr I did notice a whole cloth sound that was easy to see the whole picture of the musical presentation.

Hi Ngjockey,
OK, thanks for the clarification. :-)

In GMA's specs, they state phase shift acoustically over given frequencies. Does that mean impedance phase (reactance) or total phase? Either way, impressive.

this is really a question for Roy Johnson (who designs these speakers) but from my many detailed chats with him & from my ownership of his speakers, I believe that he is citing total phase shift - it's acoustical & electrical combined. The driver selection is critical to achieve this kind of minimal phase response.
If you read that same Audio Ideas Guide article he clearly states that driver selection is key & I quote

.....What he means is that the drivers must be well-behaved far beyond their crossover points to be used with a first-order circuit, because this circuit allows the drivers to overlap across a wide range. To be used with a first-order crossover, only the best drivers need apply.

The "he" in the above quote is referring to Siegfried Linkwitz, just FYI.

Roy J: you might want to chime in to clarify your speaker spec. Thanks.