First Order Crossovers: Pros and Cons

The output of a crossover network is a vector sum with real and imaginary components in polar coordinates. What you have in a 1st order at the crossover freq is one vector at .707, +45, and the other at .707, -45, which adds to unity in vector space with a combined phase shift of zero. But because the vectors rotate together with frequency, they are always 90 degrees apart, and they always add to unity voltage and zero phase in vector space, no matter what the frequency.

To say it differently, the combined output of the two drivers is always unity at zero phase, even though the two vectors are always 90 degrees apart. This is difficult to conceptualize, but the math behind it is relatively simple.

For obvious reasons, this is called a constant-voltage minimum-phase transfer function, and the first-order is the only crossover type that has this characteristic. I should note that this presumes identical drivers mounted very closely together and resistive loading, which is hard to achieve in the real world. But with some effort, one can come close, and the effort is well rewarded in the listening.

The exchange between Roy and Golix above leads me to bring up something I have been thinking about for a long time – the phasing of the recorded music before it even reaches your stereo system.

I have read enough about 1st order crossovers and phasing and to convince myself that this design philosophy is valid and this approach is the only one that can achieved a time coherent design. To me the $20,000 question is – does it matter enough to outweigh all the other compromises that speaker designers must make?

Others have asked if time coherence, or lack there of, audible. Let’s assume the answer is yes. The next question, is it preferable? Obviously not to everyone. Many have auditioned Vandersteens, Theils, and Meadowlarks and still chosen other brands.

Time coherence designs with first order crossovers are the only design that can come close to preserving the original waveform, most easily seen by measuring its step response. I totally agree with this. But music is not a square wave, so tracking something that doesn’t exist in natural music doesn’t prove you can make better music. Transients can be sharp, but they are not infinite slop step functions.

Going back to my first paragraph, what about the phasing of the music recorded on that CD or LP? Even if a stereo system from source to speaker was perfectly able to preserve the original waveform, what are we trying to preserve? I know very little about the recording process, but I can bet many an album has been processed by recording engineers in ways that destroys the phasing of the instruments used to make the music. Some genre’s may be better than others. Some studios may be better than others. But how are we, the music buyer, supposed to know if the waveform we are buying is worth preserving? I would love to see some discussion on this HUGE factor. I have read Richard Vandersteen tests his speakers with his own recordings. I am sure those recording have their phasing preserved. But doesn’t this say something about the phasing of recorded music in general?

Finally I would also like to see discussions on the real world trade-offs of using 1st order crossovers. Like small sweet spots. Is that inherent to all 1st order designs or do all good imaging direct radiating speakers have that? And is the sound outside the sweet spot worse for time coherent designs? If so why? I have only listened to Vandersteens (of the 1st order designs) and the difference in sound from sitting in the time coherence zone to standing up is quite alarming. There is something special going on in the sweet spot, but the Vandersteens sound flat when I stand up – treble drops right off. I know they just lost their time coherence, but when I perform the stand/sit test with my home speakers, the difference is far less dramatic. Do 1st order designs sound extra good in the sweet spot and extra bad everywhere else? Comments please.

Other inherent trade-offs of 1st order designs? Thanks in advance.

It seems Roy is correct, as usual.

But I do want to chime in on one point: There are companies which design and advertise phase coherent speakers without making claims of time alignment, such as VMPS or Fried. There is nothing wrong with doing this, although it's understood that Roy Johnson would not pursue or endorse this design decision.

Then there are companies whose ads specifically claim time coherence, when their multi driver speakers do not have first order crossovers and the baffles are not stepped back in any way. These ads are lying consciously, or someone in the engineering or marketing department is confused. Dali is the most recent and eggregious example of this phenomenon, which unlike VMPS or Fried, amounts to snake oil.

Applejelly:

You are indeed correct that recorded phase is a serious problem, and one which is utterly ignored by many recording engineers. However, some of them do take it seriously, along with many other aspects of their craft. So, in my opinion, it is well worth having a system that preserves time and phase. It can't fix the bad recordings, but at least it doesn't screw up the good ones.

About the step response: You are correct that music is not a square wave, but I will make the argument that the square wave (or step function) is the best test signal yet devised for predicting musical fidelity in a loudspeaker. The square wave, after all, is simply the mathematical summation of an infinite number of phase-matched sine waves in a specific frequency progression. As such, any transducer that passes a clean square wave is eminently qualified to perform well on any musical signal it will ever encounter.

The reason that first-order designs change sound when going from sitting to standing has to do with lobing patterns, as discussed in previous posts. First-order designs have more overlap in output between the two drivers, so the lobing effects are more noticeable than with higher-order crossovers. However, there is an important point to keep in mind: The higher-order crossovers have non-uniform phase characteristics by definition, resulting in audible distortions of a different kind, even when listening on-axis. In other words, first-order is bad if you want to listen critically while standing up; higher-order is bad no matter where you are.

I think that many people who listen to Vandys (etc.) and end up buying something else are doing so because of issues other than time-and-phase coherence. This is not the only thing that matters, not by a long shot. The Vandys are built to a specific budget point, and in their price range, there are other speakers that give higher resolution, tighter bass, more treble extension, a more neutral tonal balance, etc. Anyone who values one of these specific things highly could easily decide to buy something else.

However, having said that, it is hard to imagine a better "real-world" compromise at a given price point than what Vandersteen has achieved. As I am fond of saying, the 2C may not be "the best" in any one area, but it is the cheapest speaker ever made that can truly lay claim to having addressed all of the important fundamentals in loudspeaker design. Its status as the best-selling high-end speaker of all time supports this conclusion.

Holy cow, this is really a great discussion.

Karls, you said it well. Thank you. I should have included that for the benefit of those familiar with vector addition.

Sean, the answers to what you ask in your first two paragraphs lie in Karls' contribution, and the information you present in your last three paragraphs is correct. Yes, we are trying for a linear phase response across the board, which is not possible as you approach the frequency extremes, without digital correction (which has not, in my experience, been applied correctly to any speaker). There will be some thoughts on that on our new website by the time it is published, and if not, in the weeks following.

Skrivis, I think some of the reason Stereophile does not draw any conclusions is related to what Paul Candy says about that in his sixmoons.com review of our Callisto speakers, about alienating most of their speaker manufacturers. Part of it (again, in my opinion) is that their tests only extend down through the mid and tweeter crossover range, never down into the woofer/mid crossover range, which is where more of the music lies.

Finally, one must know what time-coherence really sounds like, perhaps by going to hear live music up close and personal, for hundreds of hours. Ever hear a string quartet practice for weeks in one's living room, and then get up and walk around them? Or a bluegrass band, or a wind band, or a Fender Twin Reverb, or a soprano, or a Steinway played with expert hands? From four feet away? How about experiencing a 100-voice chorale, on stage from 20 feet? Across weeks of rehearsals? I know this is what a speaker designer needs to do.

Those are not common experiences for anyone, let alone to have spent hundreds of hours in studios learning what the mic hears and how the studio must alter its sound, so that we think it is a) pleasing, and b) realistic.

Golix,
You say "Also a speaker which has the drivers on a vertical axis can only be phase coherent at one point in space. This of course is a purely geometrical problem independent of any electrical feature."
True. And in a two-way speaker, that point can be aligned to anywhere via tilting the speaker and with a small change in your ear height for the final touch.

For a three way, one has to move the top two drivers relative to the woofer. Our Continuum 1 three-way offered adjustable driver positions in 1995, and was reviewed in Audio Ideas Guide in `97. That adjustability has become a standard feature in all our three-way designs, and for several new models coming into production over the next many weeks. We call this "adjusting the Soundfield Convergence(tm)".

Golix, please do reconsider your notion that
"Basically a phase coherent speaker is one that is not only [coherent] in time but also in phase; a time coherent speaker is one that's [coherent] in time but not in phase."

To be correct, that should read nearly the opposite:
"Basically a phase coherent speaker is one that is ONLY coherent in phase; a time coherent speaker is one that's coherent in time AND in phase. All the math, all the physics supports that.

And we speaker designers get to screw that up! Nowhere in the recording chain, nor in the playback chain is the timing split between highs and lows, or the polarity, or both. This is purely a speaker phenomenon/distortion. One can learn to recognize that, the way an amplifier designer can hear if someone's amplifier needs a bigger transformer- it's a unique sound distortion.

Thus, Golix, please understand that while those Tannoys are indeed smoothly phase coherent, they are not time coherent. A step function would show this: Perfection in a step function looks like the plus-half of a single square wave, rising up quickly, leveling off and then going on forever- like a single stair step. There would be no ringing or rounding over at the initial corner, and the top would stay level forever, never returning to zero.

In that Tannoy, the first energy to arrive from that step-input is upwards-going, as it should be. A moment later, the late-arriving, inverted-polarity tweeter shoves (sucks) that initial positive air-pressure-increase down into the negative-air-pressure portion of the graph.

The tweeter's dome then returns to rest from its full "-" excursion, because the crossover cannot pass the "DC" to tell it to "hold your position, albeit sucked in". The air pressure then returns to the positive from the midrange tones' positive-pressure continuing to arrive.

Finally, no speaker ever then "levels off" and holds that air-pressure "positive" interminably, because the room leaks that pressure away. So the step droops back to zero, even though you see the woofer still shoved "out".

With regard to our measurements- we have those measurements supplied by the driver manufacturers, such as MorelUSA, taken in their chambers. But those measurements are usually taken in a half-anechoic chamber. Go to the Scanspeak website here:

http://www.d-s-t.com/link/scs/data/d2008_851200c.htm
and click the frequency response graph to see their measurement setup.
Can anyone say that is a realistic test of the direct sound from a tweeter? You will see a picture come up of a woofer mounted in this fully-reflective wall of that test chamber- their tweeters are tested in that same position.

For our own testing, we are still in the analog days here, not for want of trying to go digital, so I cannot show you hard copy. I will be working to present this information on our website as we continue to grow.

I can tell you that our anechoic chamber is outdoors when required from 200Hz on up, which covers the woofer/mid crossover region. Testing indoors, in an average room, is fine for looking what happens from 800Hz on up. There are also certain ways to combine very nearfield measurements, that I must decline to describe, which obviate the need for a chamber.

Digital test-gear has not shown us what we need to know any more accurately, and I have extensively used/leased all of the well-known systems available. I do know that it is easier to perform many more misleading tests in the digital domain. One has to scrutinize for many problems, with very specific measurements, either by analog or digital means. There is no one, or two, or three measurements, or even "dozens", that tell anyone how a speaker will actually sound. It takes many more than that, with an experienced ear listening for suspected deviations that physics is pointing out, and a working knowledge of what "a suspected (and/or measured) deviation or problem" should sound like.

In analog measurements, we look directly at the `scope. In particular, we look at the moment of first arrival of a burst of 4-8 cycles of a single sine-wave tone, taken all the way up the frequency scale. Perfection in time-arrival means that each of those tone bursts, at each frequency tested, starts upwards from the zero-axis at the very same L-R position on the `scope face.

The Tannoys would show a left-to-right motion of that starting point (which is the time delay creeping in) in the crossover range, and then the tone burst smoothly flips upside down in the tweeter range. Ours stand still from 200Hz to 8kHz, and always have the same polarity.

What does +/-2 degrees mean at 200Hz? It is +/- 1/180th of a 200Hz wave's period, or +/- 1/180th of 1/200th of a second, or +/- 0.3millseconds, which is readable on a `scope face. This, for the lower midrange, amounts to a front-to-rear shift of the mid-driver's location by +/- 0.4 inches, relative to the woofer. I can hear when the focus becomes as sharp in that crossover range as it is away from that range. So has every person for whom I have demo'd this.

The audible change from moving that mid back and forth, even an eighth of an inch, cannot be explained by wave-cancellation math, nor can it be explained by any change in the cabinet-face or wall-surface reflections in my designs.

We hear the difference as a loss of sharpness, or definition, of a sound's location from front-to-rear. Depth is time delay, and the sharpness of the image begins at its front-most element (the singer's mouth). If that initial location is smeared from front-to-rear, then the depth "behind" that voice is also smeared over by that initial information, and the depth itself is also smeared in time.

This is all information audible by WHEN it arrives. If that initial location is smeared, we also hear a loss of attack, which is a leading-edge phenomenon- another time-domain aberration. There exist many more ways the ears can guide time-domain measurements, and vice versa.

For the 8kHz point, that +/- 2 degrees amounts to a spatial shift of +/- one one-hundredth of an inch- tough to measure: One can easily have the microphone inadvertently jiggling from floor vibrations, by that small amount. It can be heard however, as an overall clarity of the top end, because there are a lot of frequencies nearby that 8kHz- notably the ones all the way down to 4kHz- only one octave, one "undertone" away.

I can hear when the ribbon supertweeter in our previous Imago flagship-design is a 1/32nd of an inch too close or too far away- it was crossed over from the Dynaudio dome tweeter at 8500Hz:
What moment did the stick strike a small bell or a triangle (which creates a very sharp and brief transient) relative to when did that instrument's actual tones emerge? They should have begun after the stick hit and then was removed from the metal body, right? Yet, the timing can be warped just enough so that one hears the stick-hit occurring AFTER the tones start. Now that is an un-natural sound anyone can identify! And the time-delay from this small offset of that supertweeter? Millionths of a second.

The same thing happens when judging the firmness of the felt on a mallet on a tympani or vibraphone. Or no felt at all- just the sound of hard wood, or a large-diameter mallet head or a small one. They each make their own sound, which a time-coherent speaker easily reveals, even in the midst of an entire orchestra reaching its crescendo around them.

In the usual mid-to-tweeter crossover range, achieving precise focus lets us hear exactly when the singer's tongue leaves the roof of her mouth- important to her shaping that note. Or to the definition of any other instrument that requires half-mid and half-tweeter, such as tambourine, trumpet, guitar, piano...it's a long list that includes non-instrument wideband-sounds, such as applause and film "noises". Then include the distinctive sound of each one's ambience directly behind those events- there is much to listen for, that leads to more musicianship being heard.

Also, it is possible for nearfield, tweeter-only measurements to have a standing wave build up between the microphone and the tweeter's dome, on sinewave tones, which totally fouls up anything we are trying to measure. Changing that test-tone's frequency by just a few percent, or moving the microphone back just a 1/4 of an inch, makes a huge change in the sound pressure level at the microphone (again, on a sine wave).

This is somewhat related to the how the notion of first-order speakers having comb-filtering effects comes about- from applying the math, and measuring, with specific single tones. Which do not occur in music, especially if that particular frequency lies between the tones of the musical scale. I do agree with all of what Karls goes on to say in his post right above, including his analysis of lobing. However, I find lobing is exaggerated when the cabinet-face, or even the area right around the tweeter, is contributing many reflections.

Comb filtering, from simple, "fewest possible drivers", first-order speakers, is not apparent to me, or at least objectionable on music. The ultimate audible test was comparing what is heard out of a speaker's mid/tweeter crossover range, with what is heard in that tone range from a small, say 6" square, electrostatic panel, or a plasma tweeter. We have done that, and found no significant differences that we can say were from the comb-filtering effects that must indeed arise from having two drivers producing the same range.

Multiple drivers in the same tone range present a lot of different frequencies to cancel out, because those six tweeters, for example, each arrive later than the one nearest your ear. That leads directly to lobing, which is a frequency-dense form of comb filtering. What you want to call it depends on how you measure it.

Stereophile thoroughly tested our original Diamante model in April 1994, and showed how its step response aligned quite well between mid and tweeter as the microphone was moved down to their time-coherent axis. JA was really nice to us by also showing how the corresponding step response also changed (for the better) as that time-coherence was achieved. He then showed how the overall phase response measured, which was pretty good. Ten years later, our deviation from zero is far less. Also, the Diamante tweeter's tone balance on that "best" axis was not flat for him, because at 50" away, the tweeter was well above the mic (the mic was far off the tweeter's axis). In the same issue, examine the B&W Silver Signature two-way monitor's step response. Not even close to being a step at all...The Diamante review is not archived on-line, unfortunately. Maybe those measurements are- I have not searched for those in JA's database he graciously offers.

Returning for a moment to the use of tone-bursts: One can also look at the envelope shape of each burst, from which many things can be seen, such as cones and cabinets flexing. If there is cone-breakup/ringing, then that energy was stolen from the initial input of energy into the cone. That is something that can be seen at the beginning of the envelope, as the output failed to reach full height on the very first cycles. The cone flexing absorbed that energy, only to give it back later. Of course the cone could be highly damped, then it never gives it back as audible sound, but just leaves the initial dynamic-rise blunted. Too "laid back" you would hear. Think about the dynamic response heard from soft plastic cones...

One can see a returning echo from inside the cabinet, after the end of the pulse, which can be fixed. A flex in a cabinet wall can be detected, and that can be stiffened. A reflection off the cabinet-face can be seen, and that can be absorbed or avoided. One does not need an anechoic chamber to perform those tests.

There is digital hope for us: This Summer, I look forward to working with Agilent Technologies in developing a system that will do what we need. A few years ago, the computing power was also not available for certain tests I have always wanted to make in the digital domain. Now it looks like it is.

My apology that this is so long, but I do not see this basic information published elsewhere. When (and if) you re-read it, it does seem to fit together. There were also a lot of good questions posed one after the other. Arnie of Audiogon, thank you so much for publishing this.

Applejelly:
You ask, "Even if a stereo system from source to speaker was perfectly able to preserve the original waveform, what are we trying to preserve? I know very little about the recording process, but I can bet many an album has been processed by recording engineers in ways that destroys the phasing of the instruments used to make the music."

The answer, yes, they destroy the phasing, just as you say. But please note that no studio effect ever splits the time-coherence of the signal anything like a speaker can. You are indeed trying to preserve the "original waveform", for nothing more than to reduce another audible distortion we don't need to hear. I hope that helps, because yours is a valid question. Karls, I would say that time coherence not only helps great recordings, but is very necessary to avoid "chewing up" distorted recordings. Think about how "distorted distortion" would sound.

Applejelly, you also ask, "And is the sound outside the sweet spot worse for time coherent designs?". Yes and no. It is better than severely phase-shifted speakers, because when you stand, you are not moving spatially as far off alignment as the other crossovers delayed the signals. And by direct comparison, the other speakers are scrambled even sitting down, and so your added "positional" phase shift does not add that much more. Karls says something on this, above.

You hear the difference on first-order speakers precisely because you have at least a focal point to compare, as you physically move away from it. I know that with proper attention to the speaker's design, at ten feet or more away, it does not feel like your head must be in a vise- I know now that is one artifact of the drivers "not quite being in full alignment- just very close". As that broadband alignment is widened and sharpened, the sweet spot relaxes.

About the highs going away when you stand? That is what happens with a particular design you heard. This is not indigenous to "being a first-order speaker", but only "that particular first-order speaker" you auditioned. What one can say with certainty is that when you stand, you always hear less depth to the image.

Also, you did hear the tweeter's sound emerge first, which is not natural, but it is emerging first by far less of a time "advance" than what higher-order speakers do. And if it emerges even more "too soon" when you stand up (still less "too soon" than with high-order speakers), then it can reach a relative location that lets it cancel the mid's output in the range above the mid's crossover point, and that means "less highs."

Thanks for the compliment, Suits_me. I don't think I have made any mis-statement, but please let me know if I foul up. Since this is my profession, I deeply feel I owe every bit of science, and knowledge of the sound of real music, and of how studios work, and how we hear, to my designs and our customers.

Thanks to all for reading through this. I hope you found it worth your time. I wish that I had someone tell me all of this when I started designing in 1973! My hope is that someone young picks up the ball and runs with it, to see what we have from them in thirty years, `cause it probably won't be from me! It is part of what is behind my mention of a "Foundation" in sixmoons' Callisto review's Q&A at its end. Also there are all the topics I consider important to a speaker's design, before we even strap on a crossover.

Best regards,
Roy Johnson
Founder and Designer
Green Mountain Audio

Wow. Thanks for your post. It's going to take me a while to absorb all of your comments. An excellent contribution.

Regards,

Roy, you use an upward facing bass port on your C-3's - can you tell us if any of the air pressure principles you outline in Tannoy's design would have any negative bearing on the C-3?

Also there are some "officianados" on this site that claim ported speakers do not reproduce bass frequencies with the same level of integrity as sealed designs. The claim is that many designers use ports to exagerate bass output because; either a.) a sealed box has to be so much larger to reproduce a similar frequency level, which ultimately leads to increased cost and WAF issues b.) the designer cut costs by crafting an inferior cabinet to hold the bass driver.

At first glance there seems to be some merit to their sealed design argument, however to your knowledge is this position supported by math or physics? Did you come to a crossroads in your design theory regarding sealed vs ported designs and if so, how did you arrive at a ported design for your speakers?

"I have only listened to Vandersteens (of the 1st order designs) and the difference in sound from sitting in the time coherence zone to standing up is quite alarming. There is something special going on in the sweet spot, but the Vandersteens sound flat when I stand up – treble drops right off"

Applejelly - what model Vandersteens did you listen to and were they properly set up. I did not experience anything "alarming" at all when I demoed the 3A Sigs at two different dealers here in Los Aangeles, nor did the "sound flat" when I stood up.

I bought the 3A Sigs and after my dealer delivered them and properly set them up, I'm not experiencing these issues at home either. Although I listen in the "sweet spot" alot, I've had many friends over for listening, many standing in the kitchen area behind the sweet spot, me standing in this area as well and they marvel at the sound they hear.

How far back were you seated and then standing where you heard this effect?

"This is not indigenous to "being a first-order speaker", but only "that particular first-order speaker" you auditioned"

Roy - your response was an great read, and like everyone commend you on how valuable it is to have you particiapate here.

In Applejelly's comments however, without knowing what model speaker, where he listened, and if the speakers were properly set-up, is your comment above still correct? I've heard several Vandersteen demo's were set up was optimal (including what I think is my room), and I do not hear this. There is/was not a "dramatic loss of highs" upon standing. Sure there is less depth to the image when standing, but isn't that true with every speaker?

A previous post asked whether preserving the waveform was more important than other aspects of speaker design (I am guessing that other aspects are flat frequency response, radiation pattern, and input impedance curve, dynamics and ability to handle high SPLs).

This led me to wonder whether the real catalyst for the increasing number of 1st order designs is that the newer generation of drivers is allowing speaker designers to offer 1st order designs, without having to make great sacrifices elsewhere. I remember reading an interview with Jon Bau of Spica fame where he said he would have liked a stronger bass response from the Angelus, and would have liked a design to handle higher SPL but that drivers to achieve that and also achieve his other design goals were not available at the time within his price constraints.

Looking at the drivers on the green mountain speakers, the Morel HF unit and the Aurasound LF unit I did a little research on the units and found that they appear to offer very high performance for relatively little money. The Morel tweeter is able to reproduce relatively low frequencies, and the aurasound woofer has a very lightweight, but quite rigid cone, allowing it to produce quite high frequencies before it breaks up. These low(ish) cost wideband, high sensitivity drivers are the enabler for a first order 2 way design. Perhaps they just didn't exist 10 years ago, and perhaps that is why 1st order designs have become more popular of late.

That's not to take away from the skills of designers like Roy, but it does seem that he has some great raw materials to work with now that Jon Bau and others may not previously have had access to.

I'm not convinced that amplifiers have made great strides in the last 20 years, but I am convinced that speaker technology has.

I haven't noticed a huge difference in sound on my GMA speakers between the sweet spot and the standing position. There is SOME difference, but I would not say it was a big difference.

Good thread and great technical contributions from Roy and Karl.

Good points also brought up by Seandtaylor99 i.e. parts quality has improved in terms of speaker technology, but how well that has been implimented in most designs may be another story. Just getting some of the basics right in older designs places them miles ahead of newer designs using higher quality parts in many cases. It is too bad that some of the "speaker industry giants" aren't around now to take advantage of the better quality drivers that are available to use now.

Sean's comments about "vintage" electronics is also true i.e. i've often said that older products with updated componentry can many times outperform newer products for a LOT less money. Sean
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I hear what you are saying about xovers, Roy.
But the fact remains that in there own pass bands tweeter and woofer are
working 90deg apart. If I now want to replay say a large cymbal with a fundamental at 440Hz and strong harmonics all the way beyond our hearing range (in our case past the xover point) it still means the fundamental will be 90deg out of phase with (some of) its harmonics.
I don't really care what vectors do as I don't hear vectors but I hear phasing.
We all do since, with the exception stereo recordings, all our spatial information derives from phase differences. You can test that next time you have a bad headcold that cloggs up one ear: Listen to your stereo and its like mono, go outside and you can easily tell where a noise comes from. This also works with a small ball of cotton wool, if you haven't got a cold handy.
But anyway, what I understand as phase coherent means that the entire output is in phase ideally independent of listening position.
The only speakers capable of this are full-range, single driver designs.
But Tannoy makes an acceptable(to me) compromise. Seperate drivers on a vertical line are just one step too far for me.

Still can't accept your time/phase 'explanations' it goes against everything I have ever learned and would directly contradict my two relevant college lecturers, my Professor at the Technical University Berlin ,
Guy R. Fountain ( Founder Tannoy)
Peter Walker ( Founder, Quad)
Peter Voigt ( Lowther )
and pretty much everybody else I know who's worked with AC current and/or acoustics. I don't think your lone voice is enough for me to budge on that one.Again lets look at sinewaves and lets only regard 3 points (max., null point and minimum)of it curve for the moment: to be time coherent 2 sine waves need only to start at the same moment, they could start at any of our 3 points: max and falling; min and rising; nullpoint either rising or falling.
Thus we have 4 ways in which our two sine waves can be in time.
To be in phase our 2 waves have not only start at the same moment but also the same point. There are now 3 ways in which our 2 waves are in time but not in phase.

You mention some distortion regarding my Tannoys, fair enough they distort. So do all speakers, but of course total distortion is very easily measured and mine measure up thus: for 90dB SPL, 50Hz-20kHz less than 1%;for 110dB less than 3%. How do yours do?

The thing with your test tones is quite amusing since you should be using pink noise or white noise to measure for phase coherence. Its not difficult to find a driver thats in phase with itself and one single tone from another but that does not make it phase coherent.

I'm sure you could hear the comb-filtering going on if you'd honestly compare to speakers which do not exhibit this particular problem.
I can and, compared to some people, my hearing isn't that good.
Its the comb filter effect thats (partially) responsible for the sweet spot ie the sweet spot is the area where the comb filtering is at its minimum. With speakers that emulate the point-source ideal (planars,Tannoy DC's and full-range drivers) this is much less pronounced although fr-drivers teend to produce their own version of the sweet spot due to beaming.

Some great points from Golix and i agree with some of the points that he's making here too. This is the reason that i love my modified Ohm F's, warts and all, and why i've said what i have about them. You've got one driver that is both phase and time coherent, covering the entire audible range with great bass weight and a phenomenally spacious radiation pattern. Other than that, and as i've mentioned before, any other attempt at loudspeaker design becomes extremely complex with multitudes of trade-off's involved. Juggling the trade-off's boils down to the personal preferences of the design engineer and the individual buying / listening to the speakers. As such, it is a no-win argument, just a discussion of various beliefs and preferences. There is only one way to achieve specific levels of performance, and at this time, even that approach has limitations. Sean
>

Golix: I have to back up Roy on this. In a first-order, both drivers are at zero phase in their PASSbands, and at 90 degrees in their STOPbands. (Close, anyway. The only places either of them truly reach 0 or 90 is at DC and at infinity, both of which are well outside the audioband.)

However, when either driver is at 90, the output amplitude is ZERO, by definition, so it contributes nothing to the sound. Its major contribution comes within its PASSband, where its phase is close to zero.

Now, in beween DC and infinity, both drivers make a contribution depending on the frequency relative to the crossover frequency. In a first-order, they are ALWAYS 90 degrees out of phase, regardless of the frequency, and they ALWAYS sum to unity and zero phase. If this isn't clear, you need to look into the math (including complex variables and vector addition).

At the crossover point, for example, one driver is at .707, +45, and the other is at .707, -45, as I stated previously. Due to the fact that this is vector addition, they sum to unity at zero phase. And they do this not only at the crossover frequency, but at every single point from DC to infinity. The first-order is the only crossover that does this.

If you wish to prove this to yourself, it is easily proved by doing some math. If you want to avoid the math, it can still be proved by simply drawing sine waves. First, draw a single sine wave with amplitude of 1.0 and any phase you choose. Next, draw two identical sine waves, each of amplitude .707, one shifted 1/8 wavelength to the left of the original one, and one shifted 1/8 wavelength to the right. Now simply sum their values. What you will find in that the summation is an EXACT replica of the original sine wave, in both phase and amplitude.

This principle works the same way with speakers. As long as your ear is equidistant from the two drivers, you will be utterly unable to distinguish the crossover. This is because the output summation IN THE AIR is identical to the original signal (in both time and amplitude), no matter what the frequency. (Again, this is true of the first-order only!)

Now, it must be said that the real world is not so perfect, and most drivers have rolloff-related and impedance-related phase shifts that add into the equation, giving a less-than-perfect end result. For this reason, designing a good first-order crossover is harder than it sounds.

>But anyway, what I understand as phase coherent means that the entire output is in phase ideally independent of listening position. The only speakers capable of this are full-range, single driver designs.

Since that last sentence is not true in all cases, nor in the majority of cases if we want to talk about phase and time alignment, which we do want to in this thread, primarily. There are other threads to talk just about phase coherent speakers which are not time coherent.

Note that I am agnostic on whether first order crossovers and stepped baffles are always the best selection of tradeoffs or not. I've mentioned near field listening situations as one situation to consider.

"At the crossover point, for example, one driver is at .707, +45, and the other is at .707, -45, as I stated previously. Due to the fact that this is vector addition, they sum to unity at zero phase. And they do this not only at the crossover frequency, but at every single point from DC to infinity. The first-order is the only crossover that does this."

Karls, thanks for that explanation ... now I completely understand why the first order crossover can work in amplitude and phase terms through the region where both drivers are contributing to the sound. The power output of both driver is 3dB down at this point (amplitude is reduced by 1/(square root of 2), and they are 90 degrees out of phase, but the vector addition of these two waves results in a sine wave that is in phase and 0dB down in amplitude.

Let me explain my porting question on GMA's C-3's relative to time and phase coherence. I have no idea if there is an audible issue with the C-3's design and based on Roy's published specifications there is nothing to suggest what I'm about ask actually takes place, nevertheless I thought I'd throw it out there to elicit a response regarding the porting theory behind this design.

In GMA's C-3 the bass port is firing in an upward direction, directly below and slightly behind both the midrange and tweeter, with a clearance of perhaps 4-5 inches. I'm wondering if conceptually there could be some type of Doppler effect taking place with the placement of the port relative to the midrange and tweeter that could slightly alter the phasing or timing of these drivers? Although there is no high-frequency whizzer cone riding on top the woofer; as in the Tannoy, in theory as the woofer moves backward and air is pushed out of the port, is it possible that this change in air pressure could somehow modulate the midrange and tweeter response by disturbing their wave lengths? Alternatively, when the woofer pushes forward, does the port suck in enough air to also disturb the wave length of the midrange driver and tweeter, thereby throwing off time and phase coherency?

Just to add to the info on vector diagrams, there's a very good explanation of what's going on at the Rane site, particularly in http://www.rane.com/note119.html

The most telling part of that is "The 1st-order case is ideal when summed. It yields a piece of wire. Since the responses are the exact mirror images of each other, they cancel when summed, thus behaving as if neither was there in the first place. Unfortunately, all optimized higher order versions yield flat voltage/power response, group delay or phase shift, but not all at once. Hence, the existence of different alignments and resultant compromises."

Skrivis, great link. As always, a graph is worth a thousand words.

An even more telling part of the article regarding the group delay of a 8th order L-R crossover.
"Is It Audible?

The conservative answer says it is not audible to the overwhelming majority of audio professionals. Under laboratory conditions, some people hear a difference on non-musical tones (clicks and square waves).

The practical answer says it is not audible to anyone for real sound systems reproducing real audio signals."

It always amuses me when someone makes the claim "It's not audible. Well, not audible to most people most of the time, anyway. Or at least not audible to some people some of the time....."

This is fundamentally no different than the claims that "all amplifiers sound the same" or "lamp cord is perfectly good for speaker wire." Anyone with good ears can only shake his head at such statements. Maybe they hold true in the world of budget-fi, because at some point those differences get swamped by the colorations of the rest of the system. But in a good system, with good music, the differences are plain as day, and time/phase coherence is no exception.

Now, it is not much of a surprise that Rane (and others) would downplay the audibility of time-and-phase coherence, given that they are in the business of selling high-slope crossovers. And in the pro audio world, this is indeed likely the best overall compromise, given that power bandwidth is a serious consideration. I mean that with total sincerity-- if I were designing a pro system with active crossovers, 4th or 8th order L-R would be my first choice, no question. But that doesn't necessarily make it the best for ultra-high-end home audio playback, where other priorities (fidelity to the musical signal in both time and amplitude, for example) take on much higher importance.

The most telling truth is that once someone has lived with a really good time/phase coherent system for some time, he finds it impossible to ever "go back". The lack of coherence in high-order systems, while potentially ignorable if one has never tried anything else, is nonetheless a major step backwards once one has heard the possibilities of an electrostat or a good first-order design. And since the vast majority of systems on the market are still non-coherent, it is quite possible that the majority of audiophiles have never actually lived long-term with a time/phase coherent system, and simply don't know what they're missing.

Luckily, forums such as these allow the minority not only to make our voices heard, but more importantly, to plant seeds of inquiry in the minds of those who may have simply never thought about such subjects before. For to me, there's nothing more satisfying than seeing that light bulb go off, and hearing someone say, "Wow, it sounds like real music!"

Best,
Karl

Is it audible ? I think the best answer is for everyone to demo for themselves and answer for themselves.

I was only quoting from the same source that was cited, since the earlier quote gave an impression that was different than the articles author intended.

Musical realism is the goal of every high end audio designer.

While visiting our local library last week, I heard some symphonic music and immediately thought "that sounds right!"
So I went to see what could sound like an orchestra in a library, and found....a live orchestra practicing in a library!

Each and every design approach has its advantages and its trade-offs. Getting a realistic sound from a system is a complex topic, and a sucessful design is the result of many decisions. The crossover slope is not the sole determining factor in the sound of the speakers. Examining the crossover slope in isolation is impossible outside the digital realm -
you are always introducing variables as the drivers interact and new radiation patterns are established.

I sympathize with the minority given voice on the forums. After all, I am the only manufacturer with Infinite Slope crossovers! I think its important to look at the acoustical wave interference and lobing patterns of the speaker - but since most other companies have no solution to these problems it is seldom talked about except on these forums.

Jeff

I gave the source as an explanation of vector addition. Then since the thread is about 1st-order crossovers, I included a quote that shows the superiority of 1st-order crossovers. There can be no argument with that. "It yields a piece of wire."

You might argue that some drivers won't tolerate 1st-order crossovers. Ok, that's valid. You can then either look for "better" drivers, or you can compromise with a higher-order crossover.

Lobing? If you're looking for certain types of directivity or power response, then that could be a valid concern. I would certainly look with favor upon a high-order L-R crossover for sound reinforcement use, and this is one of the reasons.

There is much concern with flat frequency and power response. I'm a bit skeptical about their importance. (I feel there are other more important problems. Besides that, to paraphrase Pat McGinty, "Once you solve the transient response, power and frequency response fall right into line."

A number of people have suggested ways to test whether phase coherence is audible, and there are indeed studies with conflicting results.

Rane's suggestion of passing a signal through a 4th-order L-R crossover and then summing the output makes sense. (Linkwitz suggests a similar thing.)

But, in order for the test to be valid, we need to play the summed signal via a transducer with no phase distortions of its own.

A panel speaker would seem to be out, since it has widely spaced sources of the same frequency. In the near field, it smears transients.

Speakers with high-order crossovers are out because they're doing the very thing we're trying to test for and there would be no possibility of a control in the experiment.

Speakers approximating a point source, with 1st-order crossovers might be suitable, but I feel the best bet is headphones. It completely removes any phase distortion due to the room as well.

I do plan on performing this test at some point, but at present I can only say that I find the sound of speakers designed to be time and phase correct to be more realistic than those that are not correct.

Looking at this logically, we can say that a speaker that can pass a transient is better than one that can't. We expect the same from other components in the chain.

The compromises needed to build a speaker that will pass transients is where argument arises. That, or the compromise of building a speaker that won't pass transients. :-)

As for "Infinite Slope," I question whether it is beneficial. What is the phase and time behavior like? Don't sharp filters like these ring? Do most drivers actually need such steep slopes? What kind of load does it present to the amplifier?

The NHT Xd would seem to offer more of an "Infinite Slope" than JosephAudio does. :-)

If one is going to go with steep slopes, then the approach taken by DEQX seems attractive. It corrects some of the problems.

As for the intent of the Rane article, for their purposes the high-order L-R alignment is ideal. They sell such crossovers, so their intent was to sell more of them. However, they certainly cannot say "It yields a piece of wire" about their crossovers.

I did not quote out of context, because I did not change the meaning of what I quoted by quoting only it. As for the author's intent, I don't actually care a fig for his intent. The quote I made stands alone.

The rest of the article deals with: "Are 1st-order crossovers more accurate?" "Yes, but..." The author's intent lies within the "Yes, but..."

Would I choose Rane crossovers for sound reinforcement use? You betcha. I wish I had had them. They offer a superior product for that use.

Karl,Roy&Skrivis:SERIES v PARALLEL 1st Order XO, What are the pos&cons of each in a two way and three way system also in a three way what is more important the T&M or the M&W interface?Would a HEIL be suitable for a 1st order tweeter if OX at about 3k with the new PEERLESS 134 HDS NOMEX paper cone? And A BIG THANK YOU GUYS for a great thread.

Thanks,
Ben

There's some disagreement on the benefits of series vs. parallel. I feel series is superior, but I'm not the last word on things. :-)

You can find a fair bit of info on the net about series and parallel crossovers. Rod Elliott has some good info, for example. http://sound.westhost.com/parallel-series.htm

I don't see that you can say that either the W/M or M/T interface is more important. Both will affect the operation of the mid driver, and that's where your ears are most sensitive.

I don't know enough about the Heil to render an opinion, sorry.

As per Skrivis, my own (unsollicited) opinion would be slightly in favour of a series -- IF you can easily get the drivers' electrical parametres VERY close (or near identical). Again, I'm just a hobbyist -- not a professional.

As to the Heil, 3kHz seems quite high -- which model are you using? As I know nothing about the Peerless, I really can't offer an opinion as to that particular match.
However, the Heil is dipole, so are you considering a 3-way with an open baffle mid -- or are you going closed cab after the Heil? If so, getting your system radiation pattern acceptable 3kHz downward could be a bit tricky!

ASAIK, the guy who makes "Heil" speakers crosses his AMT's about two octaves lower onto an upward firing peerless (quite a big unit if I remember correctly...). Frankly speaking the result is excellent down to the peerless: then things get a bit messy BUT that's just my opinion.
Cheers
Cheers

Ooops, no "edit" function anymore. Well, the BIG Heil is dipole, of course, the other one has a back chamber...

Skrivis, I must applaud you on the link you provided!

Also, along with the series crossover being "self correcting" in terms of driver variation, it exhibits the same characteristics for variances in the crossover components. For example, as we know, using a 5% capacitor in the network can result in a fair bit of variation from speaker to speaker. While painstaking matching of all components is a solution, the cost effects (time, testing, and parts) cannot be dismissed. The series network yields some very positive advantages here.

However, as the article points out, in the end, there is no free lunch. But, we have always known that and come to this conclusion for a lot of things. Otherwise, there wouldn't need to be much variation in crossover design.

Rod puts a lot of thought into the articles he writes. :-)

I feel series crossovers are better than Rod makes them out to be, but they're not the greatest thing to ever happen to speakers - as some people claim. :-)

Bud had an article that he was making available that contained a summary of his knowledge on series crossovers. However, I was told that this (and other) articles are no longer available. Perhaps someone will make it available on the web so I can read it.

Where I disagreed with Bud (and the current Fried Products) is the claim: "Properly implemented series networks provide superior driver coherence, increased dynamic range and introduce a Doppler effect similar to live music that increases the sense of realism."

"Doppler effect?" Sounds like pseudo-science to me, and it also would qualify as distortion if it exists.

Perhaps it simply strikes me as odd because DiAural were making some unfounded claims about "Doppler" effects, but they were using series crossovers to reduce them. :-)

Nevertheless, I'd like to see some proof for this "Doppler effect" and how it "increases the sense of realism."

Some series xover related pages: Graddon, Murray H, K Lehma, Jon Risch.

Of course, any ref list on xovers would be incomplete without the frd consortium.

Cheers

So what makes a time and phase coherent 1st order speaker better than a single driver design like Jordan JX92s or Fostex F200A? Jordan has NO crossover, no components in the signal path to create distortion, no driver blending problems, etc. etc.
Seems the more people talk about their 1st order T & P designs the more I like the single driver approach. Are the only benefits broader frequency range and louder volumes?

Cdc -- there may be numerous advantages to multi-driver spkrs but spl is not necessarily one of them (think of those 22.000 gauss Lowthers with a front horn).

I love wide-range drivers. Single wide-range drivers are actually rare; remember the whizzers on most such drive units. Limited frequency range, beaming, dispersion, IM, (response peaks & valleys)... are some of the most annoying (to me) problems. {BTW, you DO use a circuit on the Jordan & it actually sounds good}.

BUT, a single wide-range unit has immediacy, reasonable response in a critical region (200-4kHz, most will do 8kHz for you, some will actually hit F6 @ 20kHz withOUT a whizzer!!), phase & the like are out... it's marvellous. Extension can be had using a supertweet (not easy to match) and, better still, using a stereo subwoof.
Tough to beat.

Ultimately though, these are EXPENSIVE spkrs. Driver cost alone for a high level full-range biamped design can easily top $6k (that's $35-50k in commercial equivalent).

Gregm, that's interesting about the Jordans. Is this a baffle step compensation if they are not put up against the wall?
In my limited experience with single driver speakers, the biggest shortcoming IMHO, and this is highly listener dependent; is lack of detail especially in the higher frequencies. They are there but at a very low volume level. Maybe the driver jsut can't move this fast to reproduce at full volume level.
The Fostex F200A is an 8" driver which measures maybe -2dB at 20Khz. Wish I could give that a listen.

Well, a single "perfect" driver would be large in diameter so as to achieve good coupling with the air load so that it could reproduce low frequencies well. Unfortunately, that leads to problems with directivity and also possibly smearing of the high frequencies from widely spaced sources of the same signal. (Think panel speakers.)

So we'll go with a smaller driver and increase excursion. Then we start seeing problems with IM.

Since real drivers aren't perfectly stiff, they don't maintain pistonic motion at all frequencies. (Real drivers also have mass, and that changes things too.)

So, a cone driver will "want" to become a smaller cone at higher frequencies, and we start seeing breakup modes. This can be damped to some extent by the surround.

Ted Jordan explicitly allows for this behavior. He claims to control it in his drivers. But he's still using metal cones, and there's going to be a nasty breakup mode. the only question is how well controlled it is.

I suspect that some "full-range" drivers actually use these breakup modes to increase output at high frequencies and provide some impression of treble. Not what you'd call accuracy. :-)

So what's wrong with a multi-way system with drivers that are more likely to be able to display pistonic motion throughout their passband? Crossovers and the physical spacing of the drivers. This causes some problems and makes it harder to provide an accurate "re-assembly" of the waveform at your ear.

1st-order crossovers screw things up less than steeper slope crossovers, but place higher demands upon the drivers. (But still not as high as the demands upon full-range drivers.)

Physical placement is harder to cope with, but you can control some of it, particularly since we're only listening from one point at a time.

It's almost a Catch 22. Real full-range drivers have major problems. To solve those problems we use multiple drivers, which introduces other problems. I submit that the problems with multiple drivers can be solved more easily and more fully than those with full-range drivers, given the current state of technology.

(One possible solution is to use full-range drivers at very low power levels. This works with headphones, but not so well for normal speaker systems.)

There's just no free lunch. :-)

Cdc-- I only dimly remember the Jordan circuit, but it looked like a contour rather than BSC (it was across the driver -- but don't hold me on this!).

Skrivis:
Re: first order+ drivers. Why not use a wide-range drivers -- i.e. a 8" + supertweet, then a hefty subwoof. You'd have to biamp (at least) but, as you note, there's no free lunch!

Gregm> "Why not use a wide-range drivers -- i.e. a 8" + supertweet, then a hefty subwoof"

Well, an 8" driver is generally going to be getting pretty far away from a pure piston at treble frequencies.

Perhaps a smaller mid-woofer crossed at 3-4K to a tweeter, and then a woofer or sub below it, crossed at maybe 100 Hz?

Thanks Skrivis and Gregm. I have tried twice to respond but both times, 15 minutes into my response my computer crashes from doing Ebay auction at the same time. So I give up.

Cdc: forget the perfect piston. It'd be nice to have -- but let's just dream for now:)
An 8" driver has the advantage of being able to cope with mid & low-mid frequencies w/out extreme excursion. A smaller driver would be straining. OTOH, it WILL beam, as you hint, higher up. That'll narrow the sweet spot but, on the brighter side, IF the dispersion changes in a reasonably controlled manner, you reduce reflections... (at least that's s/thing). If the full-range can take it elegantly, I would try cutting in the tweet, 1st order, higher around 8kHz.
It's a pain to align the drivers -- but once it's done, it sounds good.
Cheers

Greg, I like the concept. The equation I use is 1,100/ freq = wavelength (ft). So a 2" driver would start to beam at 8K. How do you align the drivers? If you use a 1" tweeter I would think you get a flare if you drop it in at under 13K.
My concept is to use a 4" Tang-Band or new WR125S from Creative Sounds. Both roll off over 10K so you use the natural rolloff of the driver and add a tweeter with a capacitor to cut frequency below 10K. I'm sure my inexperience is showing here but this is what I'd like to do. Minimal x-over and F-R to 20k. Beaming would be a minor issue.
On your 8K concept there is Hammer Dynamics 12" but I think the main driver has a whizzer which I am against using. Here are Other single drivers
Have fun. Let me know what you think.

Diaural v. Acoustic-Reality what is the differance? and how do they differ from other series crossovers?

Driver aligment isn;t easy to do well (wavelengths are so short up there). A Supravox 215 fron mounted on the baffle and a tweet back mounted with a makeshift waveguide, blended quite well at ~8-9kHz. All open baffle.
bastanissells an open baffle kit called "prometheus airforce" that's reputedly very good (someone does sell import it in the US).
Do check the Tangbang's excursion capabilities -- I think you'd need to cross the Tang to a woof quite high up to avoid hitting xmax too soon. Also run a quick test to see if a small contour wouldn't be indicated (quite a few people at diyaudio have worked with the Tang).

As to the Hammer, I haven't heard it but given the many accolades (esp with a fostex supertweet) it may be an option. Good discussion at melhuish's site (super 12.
Cheers

Skrivis - Srajan Ebaen has recently written about the "Doppler Effect" in his quasi review of the Zu Cable Druid speaker. The following is an excerpt in which he mentions the Doppler Effect.

"Conceptually, single-driver loudspeakers (this one's technically a 1.5-way) are phase and time coherent though the Doppler effect could be cited when you consider how the high-frequency whizzer cone rides atop the woofer. The day-to-day observable Doppler effect occurs with police or fire sirens. They sound higher pitched as they approach (wavelengths shorten), then successively lower as they pass us and recede into the distance. Theoretically, each time the Druids' woofers move forward, they modulate the tweeter response. Once you do the math and consider the average stroke of this 10" driver -- to calculate possible tweeter response deviations in terms of how woofer distance traveled equates to wave length -- it seems more of a conceptual than audible problem. Still, it's only fair to mention in this context and avoid painting a picture of theoretical perfection. Clearly, if the single-driver ideal were the one perfect solution, nobody would bother with multi-driver designs. The market place rather demolishes any such notions in one brief instance. As usual, it's about priorities. What type of compromises are acceptable to facilitate certain concrete gains that matter more to you than that which is sacrificed?"

Here is the link: http://www.6moons.com/audioreviews/zu/druid.html

Srajan doesn't quite grasp the concept either. :-)

To many Doppelspaten's while trying to figure out the Doppler Effect! ;-)

Perhaps the single drive IS the perfect solution. They just don't want people to know. Satisfied people don't need to upgrade evry few months.

Perhaps the single driver IS the perfect solution. They just don't want people to know. Satisfied people don't need to upgrade every few months.

What happened to the "edit my post" feature so I don't have to post twice.

I don't think I've ever seen a perfect driver, let alone a perfect full-range driver.

Given the limitations of current materials, I think that it's possible to do less damage by using multiple drivers, even though you need x-overs.

The end result seems to be better than what you get with single-driver speakers. Even the most savvy (IMO) of the single-driver designers (Ted Jordan) doesn't really recommend single-drivers. He recommends line arrays of mid-tweeters, along with subwoofers.