Yeah, the original Ohm had a lot going for it. Magazines replaced square-wave tests with the computerized MLS test, which can interpolate the phase response and any ringing from the MLS psuedo-noise (but not in great detail- as most of what you see are the averages of 20Hz-wide frequency bins).
Some of the first-order speakers currently marketed do well on square waves, but manufacturers see no reason to publish the test, for marketing reasons, so the competition cannot find out easily, and because this test is not the only one to be passed for good sound, as I'm sure you suspect.
What you heard, good or bad, in the Ohm lays far deeper than what the square wave can reveal- for two reasons:
--the square wave is composed of only odd-order harmonics plus the fundamental (any even-order content seen on the `scope is distortion). Thus it only tests certain tones, not all tones.
--the square wave's dynamic range is far different than music- it does not stress the drivers enough, nor last long enough to excite the woofer.
A square wave is a guide- if you can find out where the little departures from a flat-topped characteristic come from, and then fix them, great! However, there are better tests for the problems behind those squiggles, ones which a smart manufacturer is not going to reveal, nor a poor one reveal that they don't perform!
You raise valid points- not a very professional industry is it? Becoming a better listener and gaining some technological understanding seems to be the only way to find something decent!
Best,
Roy |
Phasecorrect-
If you use pink noise or MLS or swept sine waves (common for testing dispersion), most of what we see on the printout does not explain what we hear. Dispersion, as heard on music, depends on some very definite factors:
--the diameter of the driver (not so much its shape) vs. the wavelength.
--does that driver remain a rigid piston in its operating band? Most do not.
--how the reflections from the enclosure's front and sides, and reflections off the other drivers' surfaces smear transients.
--how the crossover disturbs those transients.
The real problem is that we listen to music. Look at a musical waveform on a `scope- what do you see? Do you see any sine waves, or square waves, or sharp, stand-alone impulses? No. You see an ever-changing wave "form" that has more dynamic range than the face of the scope can reveal. It REPRESENTS how the mic diaphragm moved in and out, and how our ear drum is supposed to.
The music we hear- all its tones, rhythmic interplays, harmonies, imaging- our minds interpret from that complex "wave envelope". It is this unpredictable envelope's shape that counts. When a designer focuses on the theoretical "sine wave" components only, then the shape of the envelope has become immaterial to him.
Except to the ear. Which is why time coherence, and lack of cabinet problems, and linear drivers, and fewer crappy crossover parts, and proper crossover points are all important. Those all affect dispersion AS HEARD ON MUSIC.
Phasecorrect- you asked, "if time/phase accuracy is indeed retained...why do all time/phase coherent speakers sound different?" Because they are basing their claims of accuracy upon flawed measurements. The measurements don't pick up on all that we hear.
Ever wonder why we can't often play poor recordings? Everyone blames the studios, but it's the speaker's time-domain problems that are further distorting that distortion, contributing to unlistenabilty. Test: play a poor recording on phase-coherent headphones (Grado, Stax, others) then play it on a high-order crossover speaker just as loud.
Music is about time as much as tone and loudness. If you only test for two out of three, you won't be designing- only shoving parts into a box.
Best,
Roy |
Eldartford-
Have a look at this link- let it load all its photos-
http://melhuish.org/audio/response.htm
You'll see how bad many drivers are at impulse response. You'll see ringing- just count the time period for each "cycle"- the ring is 1/period. You will see corresponding ripples in the impedance curve for each of those rings, and also see cabinet/floor reflections arrive, depending on the test and the enclosure.
Best regards,
Roy |
Roy...thanks for the response....maybe i should rephrase my question: "if a speaker is fundamentally accurate(time/phase domain)....and can even reproduce a square wave...does this mean it automatically sounds good? I understand that accuracy is a concern...but should it overide overall musical abilities? I have heard both poor examples(phase correct) and non that sounded amazing..... |
Just because it can do a square wave does not guarantee it will sound excellent on music. A nice square-wave response does not reveal many other problems that affect what we hear. However, it is a GIANT step in the right direction.
Roy |
Can a ported or transmission line design be truly time/phase coherent? For that matter can a dipole or bipole with their trailing backwave be time/phase coherent? I suppose the same question can be asked of omni-directional designs as well. Is a point source design mandatory? Are sealed boxes required? How important is amplitutde linear(ty?) to sucessfull speaker accuracy, especially with regard to the vacillating ability of power delivery by amplifiers and the speakers dependency upon them? |
A ported or transmission line speaker can never be time coherent, nor dipole, bipole or omni, OR SEALED. See my 1/16/03 post, about why any moving system has a natural time delay down at its resonance. The ported or "transmission line" (which is a port variation and NOT a transmission line) designs have the SAME phase shift as a sealed box for the sound leaving the front of the cone, and the port opening's output has additional time delay AND polarity inversion.
For living room use and mix monitoring, I believe point-source design techniques are the best way to achieve fidelity- primarily to avoid hearing time-delayed output from more distant drivers, or from more distant panel regions.
Amplitude linearity is most important when the speakers are of minimum-phase design, as then one can hear small deviations from amplitude linearity. But if the speaker has lots of phase shift, that skews the harmonic structure of the music, which puts those harmonics out of phase and thus alters the perceived timbre of the instrument or voice. Which makes it harder to "accept" what your test microphone is saying is "flat" amplitude response. The warped phase response keeps amplitude deviations from being noticed as much. In those designs, phase and amplitude are not independant parameters. When you remove phase nonlinearities, then amplitude response IS an independant parameter.
Amplifiers have problems- if a speaker has phase shift, it will change the sound of those problems, usually making them worse. It is distorting distortion. The amplifier has its lowest distortion working into a flat impedance curve from the speaker, but many speaker designers try for that by adding extra parts in the crossover- which act to flatten the impedance curve, but reduce clarity. And some of those impedance-flattening techniques do more harm than good- especially the ones for the woofer resonance- what a mistake!
Good questions, Mr. Unsound.
Best,
Roy |
Roy..according to your "definition"...Meadowlark,Thiel,and Dunlavy would all be incapabable of true time/phase accuracy... and since Vandersteen incorporates a T-line in their designs...I guess they would also be eliminated... |
Vandersteens are definitely NOT T-lines, marketing hype aside. They are ported, end of story, and passive radiators are just a variation of porting. True T-lines are VERY big and VERY difficult to build. And I will take exception to Roy's (and Martin Colloms') lumping of T-lines in with ported designs, as they are not the same. The crucial difference is that a properly filled T-line has the best and most uniformly damped impedance curve you will see in any speaker anywhere, better than most sealed boxes and the exact opposite of all ported boxes. Impedance peaks are due to resonance, plain and simple, and the sharper and higher the peaks, the worse the resonance Q and the higher the stored energy. This is directly related to bass transient performance and that is why properly damped T-lines have a legendary reputation in the deep bass (and why ported boxes suck). I say this as a huge fan of sealed-box loading, often the best "real-world" compromise. But beware, a claim by the manufacturer of "T-line" loading is not sufficient to achieve this level of performance. It takes a lot more work than writing ad copy. |
Karls, I think the issue with transmission lines in this thread may have more to do with time. |
There is NO port on the Vandersteen 2,3 and 5 series speakers. They are not claimed to be trasmission lines in literature or anywhere else. They have an 8" woofer and a 10" driver (operates below 35hz) in a SEALED enclosure. The 10" driver is active, not passive. The Vandersteens are also as close to time aligned speakers as there is out there. They are also phase correct. Also, they used a baffleless design that does away with reflections from the front of the cabinet and the drivers are staggered for alignment. Vandersteen has covered all the bases in his designs. All drivers operate in the same acoustic phase(something a lot of speakers don't)
I don't know where you got your information from but you need to recheck it. End of story! |
Karls, Phasecorrect, no speaker is without time delay in its lowest three octaves, because a moving mass on a spring has 90 degrees of wave-period time delay at its primary resonant frequency.
Also, any port's output is time delayed, from the interior time-of-travel, and from the exterior extra time-of-travel from that opening. Port outputs are also polarity inverted.
The specific reasons a properly-engineered t-line (which are few) has low distortion bass:
--The port opening is reproducing the lowest bass and so the cone is not moving very much at that resonance point- this is part of the definition of a "ported" speaker.
--The upper impedance peak always seen in a ported speaker is mostly absent- a peak due to the port's air mass bouncing off the compliance of the air in the enclosure. This, Karls, is what you are referring to. Why is it not there? From the proper application of the wool stuffing and the shape of the small enclosure right behind the woofer.
--The t-line cabinet CAN be shaped so that its rear wall generates less echo directly behind the cone- but not via the usual tapered short horn leading into the line. A smooth taper only efficiently loads the returning third harmonic (of the t-line's fundamental resonance) back into the rear of the cone, causing a serious dip in the cone's output at 3X the fundamental t-line resonance.
Also, resonance is not always accompanied by an impedance peak- there's always resonance at an impedance minima, which even a t-line has. That's the frequency where the cone is driving the port or t-line most efficiently. So a t-line is a resonator, and no more well-damped at THAT resonant frequency than a ported speaker. The ported speaker has trouble at the next resonance- its upper impedance peak, as noted above.
A t-line also often uses a very low resonant-frequency woofer. In combination with the actual cubic volume contained in that t-line, this leads to a really high impedance peak at a very low frequency, usually well below 20Hz, which can be hard on an amplifier's power supply when excited.
T-lines are less efficient, ONLY because the woofer chosen has a longer voice coil, for more stroke to reach down to that impeance minima. A longer VC means greater moving mass. It is not because they are "more well-damped behind the cone", "which sucks energy from the cone". Utter nonsense, if the t-line is properly designed, as shown 35 years ago in the AES papers, available from Old Colony Sound Labs.
Wool is used in a t-line A) to make it an acoustically longer line (saves floor space), and B) to suppress upper-bass resonances. Wool is transparent to the lowest bass- it offers very little attenuation, which means that the low bass is no better damped. This is in the AES papers as well.
The best way to think of a t-line is as a very small enclosure with a very long port, needed to tune that enclosure to resonate at a low frequency.
A ported speaker is a medium-size cabinet with a modest port length, but without much acoustic stuffing, which would close off the volume of air needed to drive the port. So, with less stuffing, the ported enclosure is "noisier". A t-line enclosure is usually much quieter in the upper bass than a ported speaker's enclosure, and often much quieter than poorly-designed sealed boxes.
From a properly-done t-line (like the old IMF's), you hear extended, low distortion bass. Which arrives so much later than the upper bass, it sounds like it came from another part of the house. And because it took a while to get up to full amplitude, it takes the same amount of time to stop. Which means this resonance puts its signature on different recordings. Which is why sealed-box woofers offer better sound- still putting their own signature out there to hear, just less of one.
A transmission line, by definition means "transmitting energy without reflection". Except that "t-lines" in speakers reflect energy back to the cone, taking several cycles to build to full resonance at the impedance minima. So a t-line speaker is not a transmission line, as the energy came back to the cone.
The only true transmission lines for speakers would be A) an infinite horn (energy goes one way w/o reflection), and B) a muffler (energy goes away and cannot return). A t-line is neither.
Best,
Roy |
Bigtree,
Vandersteen 2's and 3's have an unusual arrangement where the rear driver is both active AND passively driven. It is an active woofer below ~100HZ and is also a passive radiator reacting to the cubic air volume driven by the front woofer- which is ingenious. The 5's woofer is in an enclosure with an amplifier and a lot of EQ to make up for being mounted so far away from the upper woofer, and to compensate for its small enclosure.
Vandersteen has reduced the baffle size greatly, but that does not mean that these speakers are free of baffle reflections. There are still large amounts of reflected sound:
from the tweeter impinging on the mid (a little),
from the mid onto the front woofer and onto the entire cabinet,
from the front woofer reflecting off the entire cabinet.
Why? The felt applied to the face does not absorb much below 800Hz. And below that frequency is the range where the mid and front woofer also both want to be fully omni, thus reflecting off the entire cabinet and each other, and the felt does not prevent that. If you would like to see some of the math behind this, read my latter postings at this site in Europe, called The Vinyl Engine:
http://www.nakedresource.com//yabb/cgi-bin/yabb/YaBB.pl?board=general;action=display;num=1038342561;start=
I respect what Mr. Vandersteen has accomplished- his engineering is far better in many obvious ways, and in many subtle ways, than virtually all other speakers, which is why he has so many satisfied customers who can play just about any kind of music with them. But he has not banished reflections, just reduced them. Those still cause abberations in the total sound output. This is one reason the crossover circuit is complex- to make the final measurements read OK in spite of the reflections.
Best,
Roy |
Roy,
I have not investigated T-lines thoroughly enough to have all the answers, but I will say that much of the received wisdom is downright wrong. Examples: (1) The commonly bandied-about equation describing speed of sound changes based on stuffing density is patently wrong on its face, and almost no one seems to notice. (2) There are all these theories about how the stuffing works, from the air causing movement in the stuffing to adiabatic/isothermal changes to who knows what else. From what I have seen, these are 90+% BS. Viscous damping due to air movement past the fibers is almost all you need to understand stuffing.
This is why the impedance curves come out so flat. In an undamped line, you have a whole series of sharp impedance peaks at n/4 for all odd n. (Note that these are pipe resonances just like in an organ, and that this is very different from a ported box, which has only two peaks which are compliance/mass resonances.) The stuffing removes these peaks entirely at even midbass frequencies and damps them extremely effectively at lower frequencies (including at the lowest 1/4 lambda resonance). On the other hand, ported design is specifically intended to function without damping, for all practical purposes.
In addition, although this isn't discussed much, T-line woofers have their fundamental resonance frequency dramatically reduced when placed in the line (as opposed to a sealed box, which always drives it upward). This is most likely due to the effect that at the lowest frequencies, the entire mass of the air in the line becomes coupled almost 1:1 to the cone. This is a very substantial increase in effective mass. An argument could be made, however, that due to compressibility, the initial attack at higher frequencies is much faster than if an equivalent real mass were added. Contrast this to a sealed box, where the only way to drive the resonance down is to add real mass, which hurts the transient response at higher frequencies. (And in addition, an argument could be made that decay at all frequencies occurs much faster as well, due to the high level of damping the stuffing provides.) This increase in effective mass at low frequencies is very nearly "something for nothing", and is probably why T-lines seem to have both "speed" and "weight".
I cannot disagree about the delay of the back wave, but I question whether it is an audible effect at the very lowest frequencies (because, again, a properly stuffed line will absorb everything from the lower midbass on up). The question becomes whether an 8-ft delay is audible at 35 Hz. I can't say because I don't honestly know. It could well be.
I am not trying to disparage the quality of a low-Q sealed box in any way, as I too think it is often the best real-world solution, but I think that there is a lot more going on in a "T-line" than is commonly appreciated, and worse, a lot of plain misinformation floating around.
Cheers,
Karl |
Roy...in your own designs...how do you approach low frequency reproduction while retaining time/phase integrity?This seems to be the biggest challenge(there are many)...somehow containing the rear energy wave...and then knowing what to do with it...which I assume even in a sealed design...there is a high degree of stored energy...
ALso...although many frown on ported designs(myself included)...not all designs are created equal...and yes the number of poorly designed port speakers(often due to cost restraints) far outweigh the few that seem to "get it right"....lets face it...ports are found on many costly, highly regarded speakers(especially in hi-end monitors)....which brings me to this:very good speakers with ports exist...which leads me to believe it is the execution of the design rather than the design itself that is paramount...and that time/phase relationships are one of many concerns a designer must face ...in short...there is more than one way to skin a cat... |
Roy, I certainly agree that all reflections are not removed from the Vandersteen's minimal baffle designs. However, it is much better to attempt a solution that helps minimize these reflections than the way a lot of designers have basically ignored them using a large area baffle. When you look at the front of a Vandersteen, you see very, very little cabinet structure around the drivers. I think Vandersteen has attempted to address a lot of issues with sound engineering in a very reasonably priced product.
I was actually attempting to respond to Karls statement that the Vandersteen's were ported designs which they are not with the exception of the 1 series which is stated to be a transmision line of sorts.
I did know that the woofer arrangement was unique in its implementation. However, again, I was responding to Karls since the driver(s) are active, not passive, although, as you stated, the front woofer will move the back woofer, etc since they share the same sealed chamber.
I have certainly enjoyed reading your posts. They are very informative. Its nice to cut through the hype and get to the point. |
Hi Karls-
Much of what you say I agree with. And to answer you, I need to qoute your statements below. However, there is a lot of peer-reviewed published research which disputes a couple of points you have taken as fact:
You say, "(1)The commonly bandied-about equation describing speed of sound changes based on stuffing density is patently wrong on its face, and almost no one seems to notice."
Yes, the EQUATION is probably wrong, but there are many research articles which have directly measured the dramatic DECREASE in the speed of sound with decreasing frequency. I have never seen an experiment that shows the opposite, I'm sorry. Some are in the AES Anthology reprints from Old Colony Sound Labs. I have studied them for many years and see no mistakes in the many methods used to make that measurement. I'd be interested in seeing any research that demonstrates the speed of sound does NOT change dramatically. But I can believe that a particular equation would be wrong.
"(2)There are all these theories about how the stuffing works, from the air causing movement in the stuffing to adiabatic/isothermal changes to who knows what else. From what I have seen, these are 90+% BS. Viscous damping due to air movement past the fibers is almost all you need to understand stuffing."
These theories differ, along with their perceived BS content, because we don't really know how the fibers behave under all types of signals- transient or continuous, loud or soft...
For example, fibers can couple as a unified mass at certain frequencies, depending on the type of fiber, its packing density, the orientation of its fibers, the length of each line segment, the loudness of the sound and its duration. If the fibers do lock together under certain signals, then quite simply there must be less frictional loss as the fibers cannot rub against each other (because they are locked together). And that means little attenuation. Furthermore, they would then behave as a mass/spring system on that signal- which means resonance. So, viscous damping is not all we need to know, as sound in fibers does not always encounter viscous damping.
As far as adiabatic/isothermal arguments- The pressure throughout the t-line (or any box) is subtantially constant. When it does change, it does not last long enough to initiate a temperature change. Those are two important numerical values to know, so one can use them to come up with a theory (and a decent equation) for why the speed of sound does indeed change- a theory and equation that fit the experimental data and fit all preceding theories and equations which other experimental data validated.
So, why is the pressure substantially constant at all points in the t-line or any box? As some air molecules collide with the fibers, that scrubs off some of their velocity with each impact- some kinetic energy is lost to frictional heat as the fibers are made to rub together.
Fiberglass is rough, microscopically.
Wool is lubricated by its lanolin, and also smoother. Experiments show that fiberglass absorbs some low bass and a fair amount of midbass, and that wool absorbs very little low bass and just a little midbass. So the connection to roughness/smoothness seems obvious. But the real kicker comes after calculating the actual pressure differentials (which are responsible for velocities) at any point in the line:
The speed of sound, even if lowered inside the line, is still really fast- if the woofer starts to compress that air at 50Hz (taking 1/200 second, 5 millseconds, to reach its max stroke), upon reaching that max 1/4" stroke, the initial sound pressure is already 4+ feet distant! That means the pressure EVERYWHERE in the line changes nearly instantaneously- there is no/little air flow! It also means the woofer cone is moving far slower than the speed of sound down at 50Hz, ~2.7mph down there (1/4" in 1/200th second).
And since (not IF) the pressure everywhere rises and falls at once, there is little pressure differential to be found across any one region, so any local velocities cannot be high- there's little molecular velocity to scrub off. Which is exactly why materials of all sorts fail to absorb anywhere near 100% in the lowest bass- the individual air molecules just aren't moving much, so there's little velocity to scrub off.
In fact, with the pressure high, the molecules are mostly colliding with each other, as they're orders of magnitude closer together than the distance between the fibers- experiencing far more lossless collisions with their neighbors than lossy ones with the fibers. With the pressure lower, they do travel at an average higher velocity before hitting their neighbors, which are still `way closer than the distance over to a fiber.
You say "[Viscous damping] is why the impedance curves come out so flat."
Yes, by supressing the upper resonance peak that the undamped line would normally generate, as explained in my previous post.
"In an undamped line, you have a whole series of sharp impedance peaks at n/4 for all odd n. (Note that these are pipe resonances just like in an organ, and that this is very different from a ported box, which has only two peaks which are compliance/mass resonances.) The stuffing removes these peaks entirely at even midbass frequencies,"
Well, not entirely removed, but otherwise I agree with you on most every point. Except that a ported box has three resonances- at the two peaks and at the minima between them. And so does a transmission line, but the upper one is usually pretty-well damped, and we still have both the impedance minima resonance and an ultra-low frequency resonance (record-warp range) which most tests don't bother to measure, so the impedance curve looks flat. But here, flat does not mean non-resonant.
"The stuffing... damps them extremely effectively at lower frequencies (including at the lowest 1/4 lambda resonance)."
I have never seen any experimental evidence of that at the lowest 1/4 wave resonance, unless the line is stuffed like a bell pepper.
"On the other hand, ported design is specifically intended to function without damping, for all practical purposes."
Yes. The trick is to have the damping "turn on" at the higher bass frequencies, so it doesn't sound like a box... and Phasecorrect, that is my answer to your first paragraph in your last post- that I found a way to keep the box really quiet above resonance, whether sealed or ported, after years of building t-lines and all other designs. The only energy storage is at resonance, an amount which comes with a critically-damped system. It is not a lot, nor is it "stored" for longer than a half-cycle of the LF resonance (= critically damped).
"In addition, although this isn't discussed much, T-line woofers have their fundamental resonance frequency dramatically reduced when placed in the line... the entire mass of the air in the line becomes coupled almost 1:1 to the cone... a very substantial increase in effective mass."
Yes on all points Karls, and that will also reduce efficiency. This coupled mass can disconnect though, but only at VERY soft SPLs and VERY high ones, for viscosity reasons beyond the scope of this thread, but in any fluid dynamics text.
"An argument could be made, however, that due to compressibility, the initial attack at higher frequencies is much faster than if an equivalent real mass were added."
Yes that argument could be made, as this would be the de-coupling I mentioned above, which does not happen at our "ordinary" SPL's. But if that "less mass" effect was indeed true at the woofer's higher frequencies, then all that would do is allow the same input voltage to come out LOUDER in the higher tone range below the crossover point- because there's less mass to push for the same voltage. But the rise time would not be any different- that was already limited by the crossover. Which means the woofer is not "faster".
"Speed" is determined by the woofer quality, and also how sloppy the designer was in allowing any fibers to come too close to its cone, as there is an invisible layer of air that remains attached to the cone several inches deep. If the fibers reach into that zone, they drag down the cone's velocity.
"Contrast this to a sealed box, where the only way to drive the resonance down is to add real mass,"
Not the only way. One could increase the woofer's compliance only, or increase compliance AND mass (plus change the box' size).
"[added mass] hurts the transient response at higher frequencies."
Intuitive, but wrong. Added mass hurts only efficiency, not the transient response (which is the same as high-frequency response). Transient response is determined/limited by the crossover point. This was in shown the AES Journals- clearly, nearly 70 years ago. Added mass would hurt the transient response (HF respopnse) if we were allowing that woofer to go up to say, 1kHz or higher. If the added mass is a high %, that would tilt the woofer's tone balance downwards from 100Hz on up, but that can be balanced back to "flat" by designing the woofer to have higher compliance (also clearly explained in the AES Journals). So considering the woofer is probably crossed over below 300Hz, then its rise time is not affected by additional mass, just its loudness and maybe the tone balance of the woofer on the way to the mid.
"And in addition, an argument could be made that decay at all frequencies occurs much faster as well, due to the high level of damping the stuffing provides."
That is a very good way of describing any quieter box. If quieter, then there's less energy returning to the cone, so the decay time on an impulse would indeed be less. But we cannot measure that directly- only indirectly, and by listening via before/after comparisons.
"This increase in effective mass at low frequencies is very nearly "something for nothing", and is probably why T-lines seem to have both "speed" and "weight".
It's not for nothing if you have to slave over a far more complicated and heavy cabinet- especially in production! However, t-lines sound like they have more "speed" because the designer probably picked a better, more linear-motored woofer that has a vented voice coil and spider, and also has kept the stuffing away from the cone's vicinity. "Speed" is also a result of a properly-built box, and t-lines are always strong cabinets. Sealed and ported boxes are usually weak, as their designers don't know woodworking well enough to make the strongest joints using less wood. And neither do their cabinetmakers, as they are not mechanical engineers.
The sensation of "weight" comes from the lower distortion of the better woofer and from the woofer stroking less at the impedance minima, and from extra low-bass output from the port. And from the port's LF time delay, as those delayed LF's also linger on longer, longer than the rest of the music- making them audible on their own. Which is not entirely amusical!
You wrote, "I cannot disagree about the [time] delay of the back wave [out the t-line port opening], but I question whether it is an audible effect at the very lowest frequencies (because, again, a properly stuffed line will absorb everything from the lower midbass on up).."
Karls, I don't understand your comparision here between 'lower midbass' and the 'very lowest frequencies', but...
"The question becomes whether an 8-ft delay is audible at 35 Hz. I can't say because I don't honestly know. It could well be."
It is, especially when you can compare it to a sealed system which goes that low without the t-line resonance. Remember- it is not just an 8' delay from the t-line output- the woofer has its own delay in getting moving, often equivalent to ~another 8' delay. So the t-line output is actually ~16' behind the midband.
"I am not trying to disparage the quality of a low-Q sealed box in any way, as I too think it is often the best real-world solution, but I think that there is a lot more going on in a "T-line" than is commonly appreciated, and worse, a lot of plain misinformation floating around."
You're right on all points. To learn more, refer to those AES papers, and others in overseas journals whose titles escape me, but for which I have copies on file if you want them- experiments done by experienced scientists who had nothing to gain from seeing the results come out one way or another- just performing basic research, then trying to come up with theories that fit that experimental data- which any theory must, or it is just conjecture.
Thanks for your thoughts, Karls- you make some very good points. It sounds like you have done a lot of reading and made many speakers.
Bigtee- thanks for your thoughts. You are right about what have just said about Vandersteen.
And Phasecorrect, you are generally right about everything in your last paragraph! With regards to the question in your first paragraph- I cross over high enough to the mid that the woofer has stopped changing phase due to its own mechanical/acoustic rolloff down at ~40Hz, and has not started changing phase due to its HF mechanical rolloff. I also figured out how to put an aperiodic damping on the back of our mids to keep their 70-100Hz impedance peak from "turning off" my simple first-order electrical crossover up at the 300-400Hz crossover points, and to keep its own LF phase shift from adding to the desired x-over phase shift. And all of that holds for what I did for our mids/tweeters at their 2.8-3khz crossover points. And then I minimized our cabinets (but not too much), separated the drivers so they reflect much less off each other, developed the cast marble recipe we use, and figured out how to make strong, yet slender woofer cabinets, as their walls are not 2-3" thick. Go hear them.
Too many speakers jam a crossover point onto the woofer just an octave or two above its LF resonance, so the unavoidable woofer mechanical/acoustic phase shift adds to that crossover's phase shift. Thus, nothing comes out right, and we also hear room positioning become critical. And to make up for their losses due to the phase cancellation between woofer and mid, we see those woofers measure `way too loud, which makes John Atkinson scratch his head, because "it doesn't sound like it measures!" Right. Because time is being left out of that measurement.
Please read carefully my previous post and look into the link I gave in it to understand more, because I don't know how much more I can explain about time coherence. The link I gave is much more about the mechanical limitations of the transducers- no reason to duplicate that here!
Mr. Bischoff, this is all your fault.
Signing off in the big snow,
Roy
Green Mountain Audio |
If there were a such thing as the best way to design a speaker, they would all be the same.To each his own,end of story! |
And your point would be... not to discuss the many approachs to design, and the pros and cons THAT EACH MAN MADE DEVICE FACES?
Or would it be that every speaker is good? Then why are you reading...
Oh well,
Roy |
Since ports are being picked on...I think there should be some clarification..."port" has become a generic term for any bass reflex type enclosure regardless of construction...and as such carries some negative conotations...it is all too easy to state "ports suck"...as there are numerous poorly designed ones that really are nothing more than a hole in a speaker...however...it is possible to obtain steller bass reproduction and overall coherency through such a design which leads me to believe that a)a high degree of phase integrity is obtained or B)time/phase relationships are not the endall in speaker design...
however this depends on the following: a)the design team is world class...B)the entire speaker is made in-house to hi end standards...which allows the enclosure itself(considering it is well made) to act as a "tuning" mechanism vs. a elaborate "correctional" crossover network that degrades the original signal....
and since most "ported" designs look the same from the outside...and many sound poor...it is all too easy to dismiss the whole lot...however...it is the internal construction of hi-end designs...often more of a elaborate sound "chamber"...that distinquishes hifi from midfi... |
Roy, my post wasn't towards you!I try not to generalize the design of speakers.I see some that do instead of allowing their ears to lead the way.If it sounds right than that's what I look at.Not whether it has ports or not.My hats off to all of you designers for bringing that magic to my ears and others.This is one of the few hobbies that is truly personal.Each man has his favorite and the one he prefers.Talking about a man's gear is just about like talking about his wife.Ha Ha HA Best regards |
Roy,
Thanks for the long response. Here are a couple links that show interesting data on the above topics:
www.t-linespeakers.org/projects/tlB/radresponse.html
This has impedance data which shows a remarkable impedance flattening at the 1/4 lambda frequency at quite reasonable stuffing densities, in addition to a dramatic reduction in the driver resonance peak itself.
www.t-linespeakers.org/projects/martin/focal/test_line.html
This also shows a dramatic drop in the 1/4 lambda resonance at normal stuffing levels (using Dacron), and also has several other interesting results. One is that the reduction in speed of sound is far less than Bradbury etc's data on wool and fiberglass. You are likely correct that the microscopic fiber characteristics have a major role in this. Also, note that at the higher frequency peaks, the experimental data show near-perfect correspondence with the theoretical numbers, suggesting that there is effectively NO air-mass coupling to the cone at these frequencies. This one plot is what convinced me that there is indeed a strongly frequency-dependent air-mass coupling.
I will still take issue with your (implied) statement that added mass is not a problem. I understand the games that can be played with mass and compliance, but only at the expense of cabinet size and/or efficiency. I also understand that one can say that "you can always make the magnet bigger." But therein lies the real-world problem: you would like to keep the efficiency as high as possible (within reasonable limits), and the cabinet at a reasonable size, while being limited by the reality of the relatively weak magnetic fields achievable with fixed magnets. So added mass does not come without penalty. In addition, my passion for a long time now has been for 2-way systems, so my perspective tends to be skewed by that reality without my realizing the need to state it, and I should have prefaced my comments with it.
Again, thanks for the extraordinary effort you have put into this thread. It has been very enjoyable. |
Gmood1-
I am glad to know you do not generalize based upon the design of the speakers. Ears should lead the way- So play an extremely wide variety of music and recordings (old/new/audiophile/distorted) until you hear what the speaker cannot do, as if you don't find those faults in the store, you will find them in the home at some point.
And when you find a flaw- such as "too peaky sounding on bluegrass", that means not only can you not play bluegrass, you'll find you cannot stand the sound of the massed, slightly dissonant strings that a 20th-century composer such as Samuel Barber or Morton Gould used to great effect, or soprano voices, or a Vienna Boys Choir disc, or realize the effect which comes over you hearing a Rachmaninoff piano concerto at full tilt, or appreciate more fully the genius of Hendrix, or the delicacy of touch required for ragtime piano, or Dixieland, or the inflections of Billie Holiday, or Janis Joplin, or Creedence Clearwater, or Chris Whitley, or King Crimson, or No Doubt, or Massive Attack, or Metallica, or Screamin' Jay Hawkins, or appreciate the real differences between...
So you play only the 'approved' audiophile recordings, of rather bland music.
You are wrong however, when you say there is no "best way to design a speaker" I could assume you are talking about basic decisions like woofer size, port or transmission line, six tweeters or one, but actually I really don't know what you mean with that statement.
There is a best WAY to design a speaker, which I'm sure you hadn't known, nor would I expect anyone to. It's the scientific methodology used to think through and then test and build and test and... And that method is for the designer to always start with the listener's location and the room around him and the SPL required and the bandwidth desired and the coverage angles. Those are exactly the parameters any professional concert-sound designer starts with. Then he's paid to work backwards to the drivers which will deliver that desired sound. Time coherence is only part of the equation, an important part.
And this approach to design is contrary to the way most all home speakers are designed- most of their designers got a wild hair and said something like, "the d'Appolito configuration is the way to go!" and never went beyond that, into understanding what happens because of that decision out at the listener's location. They began their designs at the cabinets instead of at your ears. This explains why so many high-end speakers are bought and then sold- the dissatisfaction.
So again, I'm not sure what you meant- maybe it was "don't trust any designer". Fine- in fact an excellent idea! But as you use your ears, don't do yourself a major disservice by listening to only audiophile recordings to find the best speakers. Happy listening!
Karls, thanks for the links- I've had a look, but will not respond here, as this is not the thread for that, and I probably have not the time to say anything useful. I do see, at first glance, what appear to be some wrong assumptions about what the impedance curve peaks mean vs. the 1/4-wave line lengths. But I'm likely wrong- their measurements do not go low enough below 20Hz to reveal the errors.
Phasecorrect- you make some good points about design execution, and bear in mind most speaker designers are nowhere near fully trained. Fortunately, no permanent harm comes from bad speakers, so those designers can "get away with it". You just wouldn't want them to engineer your car or medicines or food-handling machinery or house.
If I seem mean-spirited or overly critical- I am sorry, but what I've said about poor design methods is true- I have spoken to `way too many designers, while great guys, well-intentioned, smart and hard-working, simply never slogged through the graduate calculus and fluid dynamics, thermodynamics and the mechanical engineering it takes to make a speaker that performs well on most all music, in most rooms, with most amplifiers. And reviewers support those halfway design decisions saying, "These speakers really need tubes." or "They really can't play a distorted recording." While those are accurate statements, they put the blame on something else in the chain, and not the speaker. What a disservice to you, the listener! But then reviewers are usually not technically trained, so it's only natural. I would hope that anyone reading these submissions of mine here and on that European link I gave will see how basic physics applies to speakers and how that explains what we hear and also the discrepencies between measurement and hearing.
Best regards,
Roy |
Roy,
Whats up with the Green Mountain Audio website??? |
There's been no spare time to finish the last few pages (for the new speakers), as we take care of orders/existing customers. However, it shouldn't be too much longer `till we can get back to it and get it published. Thanks for checking!
Best,
Roy |
New Speakers?
Are you designing or upgrading any new bookshelfs?........ |
The Europa was upgraded many months ago, and I have been working on other designs, both smaller and larger. However, I feel uncomfortable using this forum for promotion, but I do appreciate your asking. Such info will be on the website, and anyone may email me or call and I'd be happy to give them an idea what is to come, prior to the site's publication.
But the first shipments of the new Continuum 3 (replaces the C-2) must come before finishing the site's last pages. The increased sales of existing models had slowed the C-3 production down, but we are almost there. I will post a notice here, if the moderator feels that is appropriate, when the website gets up and running.
For a couple of months, I had wanted to offer my summary of this "time coherence" debate, because while I had made some points about the effect on sound quality and the physical limits to "perfect" time-coherence in any design, it still seemed to be hard for some to visualize what a time coherent speaker did for the sound, for the "waveform".
I believe part of that comes from the way the electronic age has let us visualize sound- still struggling to somehow see it directly, like we can with a water wave.
We have filmed the movement of the ear drum- we know it moves in and out with local changes in air pressure. Is that a linear response- air pressure to mental response? No. But on what we each hear from a given stimulation, we will generally agree. We don't call that stimulation a "wave"- we call it a sound. Maybe that's mom, or Mozart, a large bus passing by, or something we've never heard before.
We know a mic diaphragm also moves in and out- we can see the corresponding voltage rise and fall on the `scope. The `scope face freezes for us a 10th of a second of the diaphragm's motion- and we see there's a wave pattern going up and down.!
But to the mind- all we know is the pressure on the body went up and down.
Time-coherent speaker design means preserving that sequence of pressure changes- the same sequence of pressure variations the mic turned into voltage variations.
Non-time coherent design means the "wave pattern" we see on the scope does not matter- we can present the eardrum with a new sequence of pressure variations, and expect somehow that this will sound the same, or nearly so.
Why do some think that pattern can be changed? When they claim, "Mathematically, and audibly, that is just a collection of particular tones. The ear doesn't care EXACTLY when they arrive- just as long as they do, sort of near their original sequence- say, within a couple of wavelengths or so (~720 degrees of phase shift)."
And there is the mistake. We know in our heads those are a bunch of individual tones- we can hear them, and point to their source. We know that mathematically as well, and can now see this via a computer's FFT.
They are heard as tones, seen as tones... they are NOT tones when impinging on the body- they are a series of apparently random pressure fluctuations imposed upon it.
For me there is no choice; a speaker must be capable of making that original sequence of pressure variations traverse your body. This is an event, a series of events- described by the frozen-in-time "envelope" or "wave packet" seen on a `scope. Our minds decode the tones out of that sequence. And when they occur. And how loud they are.
Most speaker designers throw out the "When" and use tests that ignore any distortions of "When".
A sinewave frequecy response test doesn't care "when".
An FFT throws out the "when".
A pink noise test ignores the "when".
I think keeping track of distortions in the time domain, the "when" of every moment-by-moment variation in pressure, is as important as preserving the pressure (loudness or amplitude) of any particular variation.
Speaker design is easy when you ignore the time domain, but once you hear the difference, you cannot ignore the importance of reproducing the "when". Perhaps one of the non-time-coherent speaker designers could tell us why they believe the "when" is unimportant.
But to do so, they will somehow ignore the most fundamental concept- that Sound is air pressure changing in time. That is all it is. Pressure vs. time, a particular pressure change at a particular moment.
I believe reproduction approaches "hi-fi" when any change in the air pressure next to our bodies as time flows is dictated more by the sound/by the music, than the loudspeaker.
Best regards,
Roy Johnson
Green Mountain Audio |
NSM, Role Audio, and maybe Jordan single driver speakers are time and phase coherent. NSM states "Time coherent" at their website here: NSMAny comments Roy? |
Cdc, NSM is blowing smoke up your ... There is no way on heaven or earth that the speaker shown in your link is time coherent. As I posted early on in this thread, very few speakers are actually time coherent, and I've never ever seen one with drivers mounted to a flat vertical baffle that comes even close. Sure, many manufacturers would love to have you believe it, but one look at the step response is all you need to tell otherwise.
Roy, great summary of philosophy, which I agree with 100%.
Meadowlark has a simple and easy-to-visualize example of this on their website, www.meadowlarkaudio.com. |
I've said straight out what I find wrong with the sound of higher-order crossovers and the philosophy of that sort of design approach, but have not critiqued the finer points of any design. That holds here, but I do appreciate Cdc's wanting to know more. I have commented on Audio Asylum about the Jordan approach, and can add no more here. I can say that at least NSM is implementing many good things! Karls is right though about the flat baffle, but not exactly because it is flat and also perpendicular to the floor. Remember that any speaker design which permits cabinet reflections reduces much of its potential, including any attempt at time-coherence. Reflections always cause any crossover design to be "fudged" and the driver choices to be less than optimal. In first-order crossover speakers, all those lead to a low speaker height (tweeter too close), because that's the only way to produce a decent tone balance from what is still a twisted phase response. I will not present the math for that here (couldn't even begin to), nor would I want to give it away. That height is also too low to offset the flat-baffle/tweeter-too-close placement, which is Karls' point. One also measures a greater +/- dB response variation through the crossover region, even when sitting at that "proper" height. Finally, the listener is often encouraged to not toe-in these speakers, so less reflections are heard in that best seat. But this reduces the strength of the center-fill, so the speakers are moved closer together. All of this makes the head position even more twitchy. It is the reflections that steer a designer to all these choices, and away from true time coherency. But if you read the NSM and Jordan-designs reviews, you can see what coming closer to time coherence accomplishes, including clarity, dynamics, ease of placement, freedom to play all music, and to use any component, especially when the 1st-order crossover circuit employed is simple and made with high quality parts. I am aware of NSM's/Leo Massi's/Audio Physics/Michael Green's room placement guidelines, and I do not recommend those for any speaker. Ours are closer to Cardas', as are most firms, and described in our manuals. Got a link for you to try: Doo Wop HorsesIf the sound "clicks", erase your browser's caches and re-visit when there's less web traffic. Takes about 30 seconds to load thru a 56k modem. Let it load completely (wait for the fence to appear), then click on each horse. Whoever wrote this is quite accomplished! Makes you remember why we want fidelity! Best, Roy Green Mountain Audio |
So I guess I should pass over the NSM because the drivers are too close together? I'm not surprised they aren't time coherent. I guess this is the A-A thread; Roy's comments on JordanJordan driver speakers now don't sound like they are worth the $1,500 after reading about their limitations. |
Cdc, I'd listen and decide. There are numerous people on here that have the Carolina Audio speakers that use Jordan drivers and like them. A one-way speaker has much to recommend it. Nothing is perfect.
I quote from Roy's comments on AA,
"I require more dynamic headroom, less IM at loud levels/on complex music, and more bass extension- not a knock on the good design work being done with Jordans, just a recognition of the physical limitations Jordans impose for me."
Those requirements may not be your requirements. Those are a very good description of reasons to select multi-way speakers. I use single-driver(Lowther) speakers for other reasons, and I prefer them over the compromises that multi-way speakers make. Not because they are perfect, but because I can live with their compromises in order to get their strengths.
Everything in speaker building is a compromise. What may be a "requirement" for some people, could be seen as a "detriment" to others. Every speaker designer's product is a statement about the designer's point of view on how he thinks a speaker should be made.
Make your decisions based upon what YOU need and what YOU like to hear. I recommend auditioning. |
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http://svt.se/hogafflahage/hogafflaHage_site/Kor/hestekor.swf
He used one too may http's.....
Nice link!!!!!!! |
Cdc, if you can listen to a Jordan-driver based system- do so. It is a real treat- I probably should have made it more clear that I greatly respect their level of performance. I worked with them a lot in the 1970's when the 2" modules became available. All of Twl's advice is sound, and I am sure that the $1500 is more than a fair price- so you should listen.
As far as the NSM drivers being too close together? That just means you get quite loud "tweeter low-end" reflections off the mid/woofer cone. It is those and also the reflections off the cabinet surface which force NSM's hand in "voicing" their crossovers- which then changes the phase response and leads to recommending the tweeter be at ear level- which is then not time coherent with the midrange.
NSM would have a hard time measuring their changed phase-response and may still think "it's close enough" to claim time-coherency, because of the clutter that those close-order reflections off the cabinet impose on their measurements. They are really hard to separate distinctly from the direct sound with a microphone and computer.
Also, even if they had put heavy felt around that tweeter, ala Vandersteen and Dunlavy, and then re-voiced the crossover (which would really help), then the tweeter's direct sound would no longer be flat, but would exhibit a rising response, from that particular tweeter's low-frequency tuning- its raw Qts being too low for any artificial "free field" mounting. I hope that makes sense the way I wrote that- it's been a long day.
Let me try this another way: Basically, the tweeter- most tweeters, are overdamped by their large magnets to roll-off what would normally be a rising low-end response caused by all the reflections off the cabinet face, and indeed off just the usual 3.5"-4"dia tweeter mounting flange.
This low-end rise is exactly the same as seen when comparing a woofer flush in a wall to the woofer outdoors, relative to the woofer's upper range at 300Hz. But unlike the low bass where we hear the nearby reflections blend, we hear this reflection in the tweeter as a "splash", so the reflections are worth removing. But it takes a tweeter with a higher Qts to do so, and those are rare, especially with enough stroke and thermal capacity to stand up to a first-order crossover.
Twl- the main reason I "require" more dynamic headroom is that our speakers are sold into many different environments, including really large rooms and for large home theaters. Plus, after years of pro-sound work and recording, from classical to reggae, I knew what it really meant to hear something at live levels- it is a different experience than you expect when you can hear it the way the musicians "felt" it. And so I wanted to be able to rock the house in any size room, and on any type of music.
Sorry I screwed up that link address- there was a sale on extra http's that day. Thanks Albert, thanks Gsselling. It is a nice one- send it to all your women and children. I have not looked back into that site yet- wonder what they do there??
We did go ahead and post a simple home page today at greenmountainaudio.com, if the moderators will permit me to mention that.
Best to all,
Roy |
Roy,
I have read the entire thread again (whew), and I have a few questions:
1) the detractors of time coherent designs almost always mention: dispersion characteristics, smoothness of power response, distortion,wave interference, off-axis lobing, and compression. I think you touched on wave interference and off axis lobing, how about the rest? Can a 1st order crossover based speaker be good in these areas or are they mutually exclusive? Can a 1st order crossover based speaker compete with the best 4th order based speakers in these parameters?
2) someone in this thread mentioned that crossovers, 1st order and others, can be implemented in series or parallel. Can you talk a bit about the pro's and con's of either implementation in a 1st order crossover?
Thanks. This is one of the best threads on the Gon! My audio wish, lol, would involve getting other knowledgeable people like yourself involved in this thread. Here is my dream team, in no particular order:
1) Roger Sanders
2) Richard Vandersteen and/or Pat McGinty
3) Alvert Von Schweikert
4) Joe D'Appolito or Floyd Toole
Now THAT would be a discussion. Who can make this happen? Maybe we can collectively e-mail these individuals to encourage a dialog to help eductate the supporters of this wonderful hobby???? I would pay to see/read it :) |
Dolphin ... add John Bau (Spica) to that list. |
Dolphin..also add Jim Winey of Maggie fame...hey...have to have a planar guy in a speaker discussion...even if his products are "out of phase"! |
hello, perhaps this is somewhat of an off topic question, concerning crossovers I have been under the impression that a 4th order linkwitz-riley electronic crossover was a far superior choice for auido production/reproduction of music, especially due to phase and time matching between adjacent drivers. For an excellent source of info on this: http://www.rane.com/note107.html I realise that the thread has been mainly about less complex systems, such as a simple pair of commercially available speakers, or the same with a sub/s. But even a 5 way crossed over system comprised of 2 stacks of loaded cabs is really nothing but a complex 5 way speaker system. Any comments will be well appreciated, thanks |
4th order LR does have the benefits described. It should be noted that your particular link seems to be talking mainly about ACTIVE crossovers, meaning (typically) op-amp based electronic filters placed upstream of the power amps. This is an absolute piece of cake compared to designing proper passive crossovers, but very few speakers on the market go this route as it requires an amp for every driver, and makes the overall package pretty spendy. ATC does a great job at this, but their market share is pretty low overall because most people want to be able to mix-n-match their amps and speakers (I'm not saying this is a better approach, quite the opposite, merely that it's what most people seem to want). Implementing an ACCURATE 4th order L-R in a passive crossover with real-world drivers and all their problems is a giant headache, to say the least. Lots of manufacturers will claim to be using this type, and they are in a theoretical sense, but in fact the end result isn't a truly accurate 4th order L-R due to the frequency response/ impedance/ phase variations in the drivers themselves. They can sometimes come close, but the end result is never ideal like it can be when using op-amps. This is why active crossovers are such a piece of cake by comparison. |
Dolphin, you asked The detractors of time coherent designs almost always mention: dispersion characteristics, smoothness of power response, distortion, wave interference, off-axis lobing, and compression. I think you touched on wave interference and off axis lobing, how about the rest? See below... Can a 1st order crossover based speaker be good in these areas or are they mutually exclusive? As good in some, better in most. Can a 1st order crossover based speaker compete with the best 4th order based speakers in these parameters? As good or better- ask any of our owners or dealers. Someone in this thread mentioned that crossovers, 1st order and others, can be implemented in series or parallel. Can you talk a bit about the pro's and con's of either implementation in a 1st order crossover? After considering a response for quite a while, I should not do the series-designers any favors by explaining the results of our research. We use parallel circuits. Think about damping factor and also the distortions passed on, to begin with. So, to answer your original questions, with respect to first-order crossovers- Dispersion characteristics: No problem with us. This does become a complicated issue when cabinet reflections are considered, which we avoid. Do note, however, you have never read of dispersion problems with most any 1st-order speaker design. The math is `way too involved to show why here, but there will be info on this on our website. Smoothness of power response: About the same, although this is usually botched by choice of crossover point (any style crossover) and by using spaced, double drivers in one frequency range. Biggest deviation we see when "power response" is poor, is a hollow-sounding voice range past about 30 degrees off axis to the sides. For those not familiar with this term, it was coined to describe how it might be good in some circumstances for a speaker to put out a "smooth amount" of acoustic power per frequency into the room- pretty vague, considering the "results" were an integration of the output at various angles over a complete hemisphere, which could be skewed by having a tweeter very bright on-axis and dull elsewhere- just to mention one of the flaws in "integrating". This method was championed first by the AR LST, Design Acoustics, and the Walsh driver, and now the current omni designs. It is better to say that we want a speaker to have a smooth dispersion w/frequency off to the sides (no holes), tilted downwards in the highs so we don't send too many highest-highs to the sidewalls or wall behind the speakers. Distortion: Harmonic- depends on the design of the drivers. The best drivers have NO problem in any type of living-room use, with an appropriate crossover point that respects the dispersion pattern and the radiation impedance seen by each driver. IM distortion- depends on the drivers again AND also the crossover points AND the woofer excursion allowed below 50Hz. Wave interference: Here's a concept- THERE ARE NO WAVES. We heard sound only AT our ears; an air pressure fluctuation- rising and falling minutely, UNPREDICTABLY. Unless you listen to pure, single tones, sustained, like from a tuning fork- then the wave concept is useful. But only as a solution to that very simple, eighth-grade wave math. It does not describe very much about how we will hear music. - Speakers designed only via sine wave analysis sound radically different from each other, and do not measure like they sound, because of their designers' preferences and interpretations about what sine-wave measurements mean. - Speakers designed via the time domain approach (uses extraordinarily difficult math which can keep track of the music signal's demands), include sinewave analysis automatically (the converse is never true). And guess what- these designs sound `way more similar than different, and their measurements- no matter how performed, consistently more correlate with what we hear on music. Off-axis lobing: Audible only on selected sine waves. You must realize, of course, that cancellation arguments depend on "relative distance to the drivers is now different when you are standing up". And thus to get a cancellation of a particular sinewave, you must be exactly a half-wavelength farther away/closer to the tweeter compared to the mid. Which is 180 degrees. Which is 4.5" closer/farther at 1.5kHz, and 2.25" at 3kHz, and 1.5" at 4.5kHz. So pick your frequency for cancellation. If you are standing up, remaining motionless at one spot, there is only one distance difference, say 1.5", which would then put a null on sine waves at 4.5kHz, 2.25kHz, and 9kHz. And also create partial nulls beginning within +/- 20% of that primary 4.5kHz frequency (as the distance difference reaches less/more than that 180 degrees). Which means a general dip from 3600Hz to 5400Hz. Which is less than a half octave- a few notes on the piano- only its harmonics go that high. A dip which could be "covered up" (usually is) by tweeter "splash" off a flat cabinet face. So then move around the room a little (why else are you standing?), and the null frequencies move to different tone ranges- usually higher as you move away. So you hear a different tonality/tone balance/timbre in the treble. Is it unpleasant because of the transient distortion? (see below) Yeah, if you play it at >95dB on music with a lot of treble information. You did want complete honesty, right?? The sine wave math for these nulls is not inaccurate- it is just useful for sine waves and on pink noise. What we hear from first-order speaker designs on music with its varied tones and timbres (i.e., no sustained single, pure no-harmonic-content tones) is a reduction in the treble, a compression of the depth of the image, an accentuation of the leading-edge of the low-treble sibilants (`cause tweeter is closer), a blurring of the dynamic contrasts, and a reduction in the clarity of separate performers singing the same line (think chorale and massed strings). All because of the time delay imposed by being at that "1.5-inch" distance offset. Which is a constant 1/2250 second of time DELAY (= 4.5kHz half-period = .11 milliseconds ). Contrast that with the constantly-changing HUNDREDS of degrees of time DELAY that the higher-order designs impose, no matter where you sit or stand. Ten to several hundred times longer time delays than the .11msec above!!! Delay times that also VARY with EACH frequency no matter where you sit or stand, unlike first-order designs which give you only that ONE, constant, time delay at every frequency when you stand up. And this gross amount of varying time delay creates far worse distortions of the same kind mentioned in the paragraph above. Not to mention that some of those designs (many) also invert the polarity of the mid vs. the tweeter and woofer, so the initial transients are also warped by one driver sucking in, while the others push out. This is a POLARITY INVERSION, which many try to tell you "well that is just 180 degrees". Yeah- on sine waves. Tell the drummer pounding outwards on the kick-drum skin that you are going to make his snare drum whack SUCK IN, and also that his kick drum will get there THREE FEET late because the speaker has the woofers around the side of the cabinet, time delayed even more by the crossover. And then try to explain that NONE of that kick drum's harmonics will be arriving three feet late- only the lowest fundamental. The higher tones will arrive sooner, so his pulsing rhythm will sound lagging, and less powerful. And that his "sucking snare" will likely sound hollow. And the crack of his stick on the snare head will be of positive polarity, AND arriving a few inches sooner than the sound from the sucking-in skin... Thus, I do not see the point in warping the time-domain for critical home or studio listening, especially since, for the last 15 years, we have had drivers that will handle the power and excursions required. So, it is (not only) my humble opinion, that high-order crossovers screw up the music's timbre, dynamics, rhythm, transients and imaging, because they warp the time domain so grossly, and differently, at each and every frequency. And so on them, certainly it does not matter much where you sit or stand, or measure- it is always "out of phase", far more than standing up on a first-order speaker. To demonstrate this, play a particularly poor recording on high-order speakers vs. 1st-order speakers. And then hear it over decent headphones- which also have little time-domain distortion (better to call it that than "phase shift"). And finally compression: Not a problem for home or studio nearfield with the best drivers out there. Most drivers are not very linear in terms of power compression (from voice-coil temperature increases and from magnetic-field non-linearites vs. stroke). And a high-order crossover protects those drivers, and sounds high tech, and needs to be "computer designed" for the "best" results. Which is also good for advertising. And for which is easy to present the "benefits" via sine-waves. Karls and Vettemanbc- You are both correct, on all points you make. Those crossovers are the proper way to go if you cannot take the 1st-order route, such as in pro-sound, because they screw things up AT THE CROSSOVER POINT by injecting EXACTLY 360 degrees of delay at THAT frequency. Which means they "sound OK for PA speakers". And "EXACT" is best achieved via electronic crossovers for the reasons Karls states. However, Vettemanbc, you say "especially due to phase and time matching between adjacent drivers". Phase, yes. Time- no. A common mis-interpretation, based again on sine wave analysis. If Rane said it wrong, shame on them (of course maybe that is what they wanted you to believe). Seandtaylor99- Sorry Spicas, although easy to listen to, are not time coherent. They are instead smoothly time-delayed as the music moves into the treble- but this does compress the image from front-to-rear, and make for laid-back dynamics. It was indeed a higher-order circuit, necessary to protect the drivers he had available back then. The circuit he used warps the time-domain to much less degree and more gradually from frequency-to-frequency than the highest-order crossovers. He was among the first to do that. Celestion did it some in their old Ditton 33 10" 3-way bookshelf model from the mid `70's. Kef is trying that again, I believe in their new series. Will any designer out there show me where I am off base? I am not about attacking them- I would like them to justify why they believe we cannot hear time-domain distortion. The numbers I give above are not in dispute- just their audibility and the need for "waveform fidelity" (Technics, 1976). In fact, I will absolutely refrain from any comment or question until others have posed their questions to those designers and had them answered. Hope this helps. I cannot seem to explain the benefits of a time-coherent approach to design any more easily. Tried many times. Best regards, Roy Johnson Green Mountain Audio greenmountainaudio.com |
Roy, I'd like to commend you for a continued good show of the understanding of speaker systems, and sharing it with the people on this board. Many designers would not do as you are doing. I think it does a world of good for people to get a deeper look into things, than the usual marketing brochure hype.
I have to admit that even though I have done many DIY speaker systems and studied alot, there are some things you mention that I have not considered, or I simply just was not aware of. I have found your posts very informative. |
Thanks Roy ... I was not trying to suggest that Spicas are true time-coherent, because I did not know whether they were or were not. I had an idea that they were designed to approach time coherency, and if that is what makes them image the way they do then I'm quite interested to hear a pair of Europas to see if they will give me more of what I like about the Spicas with a bit more dynamics, and less of a recessed midbass. To date I've heard speakers that have better bass than the spicas, but none to touch their overall sound for under $1000. |
Sean...take the SPicas...increase their legendary imaging,soundstaging, and transparency abilities...add an octave of bass... extend the highs a little...and welcome to the GMA sound...granted this comes with a higher pricetag than the SPicas(probably double)...but as a former tc-50 owner... not that much out there for under a 1k...but the Europas changed all that...they are the best kept secret in audio... |
Roy,
I am not exactly sure if my question is related to this thread (but it might if the answer has something to do with crossover topology :) but I would be interested to hear your reply nonetheless. You touched on the fact that your speakers can be cranked at loud levels, which is nice from my vantage point. However, what makes a speaker good at low volumes? Very few speakers IMHO sound good at low levels. |
Lack of phase shift is one criteria I know helps a lot- for clarity, texture/timbre, subtle dynamic contrasts and sharp imaging. But none of those comes through unless three things are present:
- the drivers have suspensions designed for high compliance at micro-amounts of stroke.
- the crossover parts, primarily capacitors, can pass very faint signals (most caps cannot).
- a very quiet cabinet on the inside.
We can rule out (as first-order causes) the linearity of the magnetic fields around the voice coils- there is no voice-coil stroke occurring.
We can rule out voice-coil venting and high-temperature voice-coil construction- as there's no stroke to create any air pressure to be vented, and little power input to have thermal changes in the voice coil.
We can rule out "extreme" cabinet rigidity, because of the low levels of energy input.
We can rule out cone rigidity, for the same reason.
We can rule out the way the enclosure is tuned (ported/sealed/T-line) as those become non-linear with INCREASES in SPL, if they are going to mis-behave.
Of course, I'm sure you know a lot of gear isn't that great at soft levels (especially interconnects- which is why I recommend the Audio Magic Sorcerer cables before any component upgrade). In fact, I know of some amplifiers which have a decidedly "off-on" type of sound that actually gives speakers with poor low-level response more "jump". Of course, an amplifier which does have excellent low-level response is termed "laid back" when auditioned/reviewed with those speakers- too suave and graceful and subtle for those speakers.
To the others- thank you for your kind compliments.
Best,
Roy |
Dolphin...bass at low volumes is another key area...most people increase the volume for this reason alone...so full range (40 hz) frequency coverage would be one of my criterias for listening at lower volumes...or adding a sub...cheers |
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Roy, will you please tell us what the impedance ranges, sensitivity, amplitude response and power recomendations are for each of your speakers? |
Now, as opposed to then, I'd have to guess that the execution of a speaker's chosen methodology and quality of materials is significantly more important than simply the methodology. There are some important questions about speakers (phase shift perhaps), but I don't think time and/or phase coherent to be among the make or break methodologies.
For example, the Vandersteen Model 5 and 5A are phase and time coherrent and come very highly rated by some and retail for about $15k. Some have even claimed the Model 5's to be among the very best speakers available at any price. (Although, I've always been bothered with the fact that the Model 5's have their own built-in bass amps. As if to say nobody can build a better amp than Vandersteen).
On the other hand, there's the Von Schweikert VR4 Gen III SE's that (I believe) are neither time nor phase coherent and employ 4th order crossovers. Some here may puke at the thought of these ingredients and methodology. Yet, there's at least 2 reviewers and some to many happy owners who claim the VR4 Gen III SE's to be the very best speaker available at the $20k and under price range. And they retail for only $6k.
And remember that the Vandersteen Model 5a's retail for $15k and therefore fall into the $20k and under category.
No matter how you look at it, that's quite a statement for a $6k speaker.
I've listened to neither and that is not the point as I'm all for have the best speakers possible within one's given budget. But personally, I think people put way too much stock into speaker design, etc. when in fact, it's the amplifiers where the more serious deficiencies lie.
It all comes down to priorities. And you can only have 1 top priority. And all other priorities will suffer for the number one priority.
Therefore, I'll take the best amplifier/mediocre full-range speakers combo over a mediocre amplifier/best full-range speaker combo any day of the week.
And, without a doubt in my mind, this chosen path would reign superior sonics every single time.
-IMO (of course) |
