Sloped baffle

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???

@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

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.

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?

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.

@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.

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.

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.

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.

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??

@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.

How fast is the train travelling?

"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.

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.....

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.....

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

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

"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&&

Hi Timlub,
Ok, no problem.

I believe his real question is "do all frequencies move at the same speed"

yes, all freq travel at the same speed. But the answer to his question is still "No, they do not arrive at the listener's ear at the same time". I tried to explain that in my post - looks like you missed it? I'll cut & paste here again for your convenience:
" 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."
if you do nothing to compensate for the fact that the acoustical centers of the 2 drivers are different, the highs arrive earlier.
hope that this clarifies.

07-07-14: Bifwynne
.....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?

Bifwynne, here is some more material for you to read that will address the phase coherency question you had yesterday (7/6/14):
http://greenmountainaudio.com/speaker-time-phase-coherence/
this is an article that Roy J wrote for Audio Ideas Guide back in 1997. A small cut & paste from this article:

"The causes of phase distortion

Time delay is the natural consequence of making something vibrate, whether it's electric fields or material objects. In speakers, only three things can cause time delays:

◾The moving elements (the drivers -- woofers, midranges, tweeters);

◾Their distances through the air to the listener; and

◾The crossover circuit.

Let's go over the cause of motion-based time delays first. Different drivers (round, square, flat) have an inherent amount of phase shift, related only to each one's natural resonant frequency. One analogy is a weight hanging from a spring. If you move the other end of the spring up and down very slowly, the spring does not stretch and the weight follows your motion exactly. The phase shift between your applied force and the weight's motion is zero. The moving system is in a 'minimum-phase' mode. If you move more rapidly, the spring starts to stretch and contract -- and the weight no longer follows your driving force. It moves with a different phase.
......"

there's a lot more to read but I believe that you should read the section "The Causes of Phase Distortion" to answer your question....

I have never heard Roy's speakers, but I would love to, I've never seen any speaker that was plus or minus just a few degrees throughout its frequency range... It would probably be easier to read a book than try to get all the info from a forum, so much skipped over, so many half truths... very difficult... We have only really discussed baffle step compensation... I have tried several times to electrically time align a woofer rather that the tweeter. It can be done, but throws so many other things out of wack that I've never really been successful.
One thing that I would say... it is possible to get a great sounding speaker without very good alignment, but most speakers that "just aren't right" do not have good alignment characteristics.... Any speaker that has very good alignment characteristics, you will find very listenable, it may not be your cup of tea, but it will do most things well. At least that is my experience.
Tim

Thanks again Bombaywalla. I caught the article. It doesn't speak to the impact of the speaker's electrical characteristics on phase coherence. Maybe it's just a non-issue.

@Tim ... based on the various posts in this thread, I gather there are not very many conventional speaker brands that are time and phase coherent. Vandersteen, Thiel and GMA come to mind.

As to your point about flat impedance and phase angle plots, take a look at the stats on the Magico S5 here:

http://www.soundstagenetwork.com/index.php?option=com_content&view=article&id=1043:nrc-measurements-magico-s5-loudspeakers&catid=77:loudspeaker-measurements&Itemid=153

Not saying the S5 is an easy speaker to drive because its impedance plot ranges for the most part between 3 and 4 ohms and its phase angle goes negative in the low bass, but overall pretty flat plots. I think my ARC Ref 150 could drive it ok off its 4 ohm taps, especially since my amp has a pretty muscular power supply - 1040 joules. OTOH, I would not try to drive the S5s with a low power SET amp. :)

I find the phase coherence issue to be extremely interesting. It's frustrating because without doing critical listening, it's hard to get one's "ears" around the issue.

Thanks Psag for bringing this important issue to our attention. Just not sure what to do with it. :)

BIF

Those S5 plots are actually quite smooth. A sign of well matched, quality drivers, a good cabinet and unobtrusive crossovers. That phase dip in the bass is fairly standard and hard to avoid. It's from the woofer, not the xover. Other than the low impedance, it would be a relatively easy load. Of course, there's always ways to make those plots look better but it might not sound any better.

Ngjockey ... I concur with your take on the S5 stats and build quality. As far as the X-over is concerned, not sure what you mean by unobtrusive, but I'd be willing to bet we're talking about 2nd and 3rd order X-overs here.

Fyi, I traded e mails with Magico tech folks about phase coherency. They freely admitted that the S5 is not phase coherent, but that attribute was a trade off in order to achieve other design objectives.

Gotta give these pups a listen!

07-07-14: Bifwynne
Thanks again Bombaywalla. I caught the article. It doesn't speak to the impact of the speaker's electrical characteristics on phase coherence.

I am totally confused here Bifwynne!! I don't understand your question - what do you mean by "the impact of the speaker's electrical characteristics on phase coherence"??
I've pointed you to Roy's article that talked about the impact of the electrical x-over on time-coherency.
I've pointed you to Roy's article that talked about the impact of the driver construction on time-coherency.

The x-over is the electrical part. the driver construction is the mechanical part.

These 2 articles should have covered the info you were looking for.....

Bombaywalla, sorry for the confusion. I'm referring to a driver's electrical, not mechanical, attributes. Rather than go off on a tangent, if Al catches these last few posts, he might be able to untangle what I'm trying to say. In the meantime, I'll just assume that the only relevant driver attributes that affect phase coherency are the mechanical points Roy discussed in his White Papers.

OP-

Yes. It is important to have a sloped baffle loudspeaker.
Today's drivers are very advanced in design and construction. Therefore, we (listener) want to get the maximum potential out of them during sessions IMO.

Regarding digital signal processing as it relates to these issues, this from the DEQX website:

"In addition to frequency-response errors DEQX’s biggest strength is restoring phase and time-domain coherence by delaying faster-arriving frequencies until slower-arriving frequencies catch up for a coherent Impulse-response. DEQX even corrects timing delays in frequency groups within the drivers themselves rather than just time-aligning one driver to the next."

Thanks for this info Psag. The DSP software definitely considers having phase & time-coherence as an important aspect of its signal processing so as to have cohesive sound. This should tell us something about the importance of time-coherence in speaker design :-)
Looks like DSP might be the way in the future...

Practically, headphones are the best for time and phase coherence. Even good quality cheap ones. Use those as a reference to help decide how good speakers sound in this regard. Then check the measurements if available to see if things correlate.

Bombaywalla, I use the DEQX, and I can tell you its tranformative. Because it 'corrects' the drivers, it has a way of making different speakers sound more similar, which would no doubt be disturbing to some potential users. Also, it makes some recordings sound somewhat different than we are used to hearing them, which is something that also takes some getting used to.

Hi Psag,
What you are quoting makes absolute sense... If you pull the crossovers and make everything perfectly phase and time aligned along with perfect frequency response, then the only difference is sound between speakers is the materials themselves...ie, how does the box sound, what does a Kevlar cone vs a paper or aluminum cone sound like etc.... So you are hearing first hand, (by correction) how important a flat response along with phase and time alignment can be.

Even with DSP, bet there's still a market for $300 Revelators vs. a $20 Silver Flute and vice versa.

Psag ... just took a quick peek at the DEQX web site. Very interesting.

Problem is that it's not cheap and where is the vendor located? What recourse if it doesn't work well.

Also, it obviously entails inserting an artifact into the signal path, presumably between source components (e.g., CDP, DAC and phone pre) and linestage. Oftentimes, not the best thing to do. How does it work if one has an integrated amp with built-in phono section??

Sure wish I could try the device on approval.

Cheers,

BIF

07-08-14: Bifwynne
Bombaywalla, sorry for the confusion. I'm referring to a driver's electrical, not mechanical, attributes. Rather than go off on a tangent, if Al catches these last few posts, he might be able to untangle what I'm trying to say.

Bruce (Bifwynne) raises a good question, to which I suspect there is a good answer, but I don't know precisely what that answer may be :-) But I'll reformulate what I interpret to be the question, and perhaps one of the others who are participating can address it.

Consider a simple two-way speaker having a first order crossover consisting of a capacitor in series with the tweeter, and an inductor in series with the woofer. For each driver that will result in well behaved 6 db/octave rolloff characteristics, which will result in time and phase coherence if other aspects of the design are also supportive, **IF** the impedances of the woofer and tweeter are purely resistive.

However I believe Bruce has been alluding to the fact that the impedances of the drivers are not purely resistive. And it would be more accurate (if still somewhat oversimplified) to electrically model them as consisting of a resistor and an inductor in series.

So the question then becomes: Doesn't the presence of that inductive component of the driver impedance (especially in the case of the tweeter) cause a deviation from first order 6 db/octave behavior? And if so, to a degree that may audibly compromise phase and time coherence? And if so, is that or can that be compensated for in other aspects of the speaker's design?

As I said, I don't know the answers, but those strike me as good questions.

BTW, Tim (Timlub), thanks for providing the link to my post about impedance phase angle.

Best regards,
-- Al

"So the question then becomes: Doesn't the presence of that inductive component of the driver impedance (especially in the case of the tweeter) cause a deviation from first order 6 db/octave behavior? And if so, to a degree that may audibly compromise phase and time coherence? And if so, is that or can that be compensated for in other aspects of the speaker's design?"
ABSOLUTELY, I had stated earlier that a simple cap seldom produces a 6db slope for that exact reason, no speaker that I've seen shows purely resistive(other than a ribbon). In the crossover, we can only lower impedance of a driver with compensation circuitry. A simple pad in series on a tweeter will raise impedance, but also cause you to need a new crossover. All this is the difference between electrical vs acoustical crossovers, cause & effect. 6, 12, 18 or 24 db per octave crossovers on paper, often end up being larger slopes because of the natural inductance, impedance and capacitance of a driver that must be taken into consideration in the design itself. In my experience without impedance compensation work, a driver with 6 db filters are still TYPICALLY 10 to 20 degrees out of phase... Any speaker that is with + or - 15 degrees of phase for its useable response curves in my mind IS phase coherent.

Bifwynne, I'd say that the cost is substantial but fair. The vendor is located in Colorado, and he came to my home in Arizona for the calibration and installation. I don't know if there is a trial program. The DEQX corrects the room acoustics, and it also corrects the speaker, as outlined in my previous post. The unit resides between my conventional preamp and the amps, but it also has full preamp capabilities if needed. A less expensive option deletes the preamp functions.

Why does so much discussion center around how "perfect" 6db per octave is? There is 90 phase shift at the crossover point and we can add on electrical and mechanical impedance changes in the drivers across their frequency range. So can't we fully expect some pretty significant deviations from "perfect phase" in the performance of any speaker, even with the first order crossover?

To answer Al's rephrased question or Bruces's origional, I'll give the example of the speakers I'm working with lately. The midwoofer is an inexpensive 5" that has a fairly nasty and fairly normal breakup around 5KHz and rolls off rapidly after that. The design I'm using it in is basically a flat baffle MTM (+++) with a low crossover point of about 1500 Hz. BSC is handled separately, but we won't get into that because that gets complex. The low pass is a first order (electrical) with an inductor.

By itself, no crossover, the woofer shows 30 degrees of phase shift by 1500 Hz even though it is nowhere close to rolling off. With the crossover, it's 120 degrees at the same point and nearly 180 degrees by 3000 Hz. It's the combined acoustic slope that matters and that's measured in Hz and dB. Phase is along for the ride.

Things get a bit more complicated. To attenuate the cone breakup 5K, which would still be audible, I added a "tweeked" Zobel. By that, I mean I oversized the cap and undersized the resistor so that it falls somewhere between a filter and impedance compensation. It also comes in handy to get phase dialed in. Tried but couldn't get a notch filter to work well in this case. Got it about 20 dB down.

As some of you might have guessed, for a tweeter to cross that low, it has to be particularly rugged and there's only a few I know that capable, particularly with only a second order high pass. Didn't want the crossover that low but that's where the combination wanted to be. It's already 6 dB down by the crossover point, which gets summed back, when drivers are in phase. The tweeter, with crossover, has begun rolling off from around 5000 Hz. Another "trick" was utilized to round the knee. By the tweeter's Fs (resonant frequency) it's down 20 dB. The tweeter's phase shift from 1500 Hz to 20K, before any baffle diffraction and with crossover, is only 60 degrees. Essentially, little to no phase shift without crossover. If you're still paying attention, you might think something's wrong with my math. Shouldn't second order shift 180 degrees? For a high pass, the phase shift is caused by capacitance, not inductance.

In order to get the driver's phase aligned, I needed to invert the polarity of the tweeter. By the way, this sims out to a 45 dB reverse null at 2m, so I think it's pretty much on target. Nicer is that it's consistent over a wide vertical and horizontal range and listening distances. Gently sloped plateau on the impedance phase to +30 (inductive) degrees maximum, which by most standards, is quite good.

That's a very simple example, even for a two-way. You should see what happens with real woofers. Remember the old spinning plates act, where a guy balanced plates on poles and ran around to keep them going while he added more plates? Now, tie the poles together with strings and springs and rods and hinges and that's speakers.

Next week, I'm planning to do some fact finding about the DEQX device. Short of outboard active crossovers, I'm starting to get the sense that mechanical (e.g., sloped baffles) and electrical (i.e., 1st order X-overs), at best, does rough justice. I'll be back.

Bifwynne, the DEQX unit that I have also has active crossover capability, although I do not use that feature. Unfortunately, the designer of my speakers had no interest in helping me to disengage the internal passive crossovers.

Yes, the DEQX units do seem very intriguing. And their pricing does seem very fair, as Psag indicated, especially given that they can be had with both preamp and DAC functionality.

I believe that the Trinnov Amethyst offers somewhat comparable functionality, but is way more expensive. And I believe that Trinnov's somewhat older ST2 model is significantly more expensive as well.

Lyngdorf offers a room correction processor including preamp and DAC, but it appears that its focus is just on correcting room-related frequency response issues. Also, its analog input impedance is only 10K, which would rule it out for use with some tube-based sources.

One question I would have about the DEQX products is if the company is set up to make repairs in the USA, given that they are based in Australia.

But I'd have to say that it's definitely something I'll be considering when I next feel motivated to make a major change to my system, although that won't be particularly soon. One reason it might be especially applicable in my case being that my room treatment options are very limited, since the system is in my living room.

Psag, thanks for calling DEQX to our attention.

Best regards,
-- Al

Psag ... so you say the device is inserted in between the pre amp/linestage and amp?? Well here's a little quandary I may have.

My ARC REf 5 SE linestage has to see a combined impedance of not less than 20K ohms. As currently configured, the Ref 5 sees 300K ohms off Main 1 - imput imp. of amp and 337K ohms off Main 2 - imput imp. of custom made subwoofer/impedance buffer/channel summing gizmo. The combined impedance is about 157K ohms. Well north of 20K ohms.

Any idea what the DEQX input impedance is. As you can see .. kinda important.

Thnx

Bruce, all of the DEQX models are indicated at their website as having 50K input impedances, for their balanced and unbalanced analog inputs.

Best,
-- Al

Holm Acoustics also offers a similar product but it's more expensive. Without the preamp feature, there's miniDSP, Ground Sound modules and what I consider the best of the 'pro' units, Xilica, which is also used on the top-of-the-line Legacy speakers. Almost forgot the McIntosh MEN220.

You still gotta know what you're doin', so they're really no easier to use than designing a passive system.

Thanks Al ... caught that stat after I read your post. Didn't do the math, but I think that the DEQX would likely present an ok input impedance for my linestage even with the impedance buffer (330K ohms). Something north of 30K ohms if my "Jethro Bodeine double knot head cyphering" is right.

Still very fuzzy about this whole phase coherence conundrum. And even if its real, whether inserting the DEQX device in my signal path will hurt more than it helps.

The real problem is that there are so few B&M stores around, especially those that carry the gear in which I am interested, its hard to do serious listening and make rational decisions. Maybe an audio show??

I hate this hobby.

Al, You are welcome. The U.S. distributor, based in Colorado, knows the DEQX unit inside and out, so service is absolutely not an issue.
Bifwynne, the input impedance is 30-40 K and the output impedance is about 100 ohm on each signal line. My line stage is also ARC (Reference 10). I have paired subwoofers in my system that receive signal from the DEQX. The only output from the ARC is directly to the DEQX. The DEQX handles everything after that.
Every once in a while I take the DEXQ out of the system to reassure myself that its truly transparent when in bypass mode, most recently last week. To my ears, despite the extra A/D and D/A conversion (analog source) it is transparent.

Short of outboard active crossovers, I'm starting to get the sense that mechanical (e.g., sloped baffles) and electrical (i.e., 1st order X-overs),....

cool, I like that! :-)
I believe that the user community should demand more time-coherent speakers from the various manufs. Many people pay a pretty penny (incl you if you go the Magico-S route) for their resp. speakers & I really doubt that anybody wants to listen to added (speaker) distortion after having paid so much....

let us know what you discover about DEQX - like Ngjockey indicated, I doubt that DEQX is simply plug-n-play. I think that the user will have to know something about the physics behind the usage scenario (freq response, phase response, x-over slopes, phase coherency at x-over freq, amplitude of freq response at higher freq, etc) to bring out the best in DEQX. This is my guess. Meanwhile we will await word from you on this subject. Might want to start a new thread.
Of course, do not forget to search the Audiogon & Audioasylum archives for existing chatter on DEQX - might give you a jump-start.
Thanks.

There is also some chatter on DEQX on computeraudiophile.com. Actually, I've been looking into it as well, along the lines of my posts above (waay above).

The DEQX HDP-4, the most expensive unit, has good DAC inside, 3-way crossovers, and digital volume control. And room/driver correction. What I researched was for use instead of my DAC and preamp, so getting the signal from my computer server and running 3 amps per side to drive speaker drivers directly, avoiding passive crossovers. I found a guy in Texas with very nice and expensive system, such as YG speakers, say he replaced a $30k DAC with it, so the DAC section must be good. He's using it in the same fashion I'm interested, so can't speak to the ADC section - but that has been clarified by Psag above.
He's only caveat was the unit only allowed for one subwoofer out and he's using two in mono but needed different time delays on each to address a room mode so he uses a Xilica unit for that.
BTW, he also uses Dirac room correction software on his server. That piece I don't fully understand why as the HDP-4 does room correction too...

The HDP-4 is very interesting to me. Where I start wondering if it is the best path is for people like me who only use a computer as server. In that case you could use a Lynx Hilo plus Acourate software to achieve the same, but for $3k instead of 5, or use an exaSound e28 plus Acourate for $4k and have 8 channels available (hi/mid/bass for L&R, plus 2 subs) and be able to time/phase align all of them.

But I realize the majority of users here aren't running servers as their main source. Yet the HDP-4 can take 2 analogue inputs and several digital inputs, so you could still connect a phono section and a CD transport.

BTW, Psag, how good is the HDP-4 volume control? When bypassing your preamp, what changes do you notice?

I think a call to the US Distributor may be in order. Lew... I agree that the DEQX is likely NOT plug and play. From what I picked up from the DEQX web site, one can pay extra for a remote professional set up. To me ... that is just part of the cost.

Honestly, I am a frustrated scientist. This stuff is very interesting to me. Problem is my IQ isn't high enough to get the math and science. My math skills are just a little north of the "Jethro Bodeine double-knot head cyphering" level.

Here's a guess ... I surmise that if the DEQX's hype is fairly stated, it may do more for my rig than stepping up to $25+ Magicos S speakers. There is nothing wrong with the drivers in my Paradigm S8s. We're talking about a low distortion beryllium tweeter, an aluminum/cobalt mid and polypropylene/mineral infused woofers. I have a sub to augment bass roll-off. The basics are all there.

I'll report back.

Cheers.

P.S. I have a better chance of sneaking the DEQX into my house than new speakers. That factor alone weighs heavily in favor of the DEQX. :)

Al -- you are always the voice of reason. What are your thoughts? Could this be transformational or is that un likely?