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 |
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 Horses If 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 |
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 |
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 |
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 |
We have several dozen dealers- a list is available. Admittedly not a large number, but they do cover a fair portion of the US, some of Canada, and Puerto Rico. There are some international distributors too. Many have been with us for more than a decade. Most serve their local markets- they are not involved in the Stereophile/TAS product of the month club. Our few, select dealers that do significant internet business can be located easily. If we seem to move too slowly... a long time ago we saw that if we grew too fast, quality and ergonomic issues would arise with the cast marble process- another reason to avoid the sales fluctuations created by magazine-based retailers. I can understand how all this gives the impression we're resistant to new dealers. I think it's important for the health of our industry to patronize retailers who go out of their way to listen to everything, and avoid those who primarily sell whatever is new or hot. Our acoustic memories are short. So are our consumer memories- one can hardly count the number of audio maufacturers that no longer exist, once well-known. And many of them failed after a few years of marketing to only the big mail-order retailers, finding themselves dropped in favor of the new. This of course, means that those products, and their designers never had much of a chance to evolve, so today we see a lot of "me-too" product- I think a result of slip-shod R&D efforts, even from quite large firms, because product is being rushed to the market- selling because it's new for this Fall, not because it's a significant improvement in the art. I feel strongly that non-time coherent speaker design is one indicator of this lack of extended R&D effort in the speaker community. Another is the ignorance of what "radiation resistance" means. That is one of the most important parameters of speaker design that determines the sound you hear over at your chair. Radiation resistance is determined by the quantity and the sizes of the drivers chosen and their crossover points. It is the term for the acoustic load on the driver, and is why a larger woofer has more "slam" than a smaller one which goes as low on steady test tones- the bigger woofer sees a higher radiation resistance down there. That is just the LF range- the effect of varying or inadequate radiation resistance higher up the scale means that the tone balance and dynamic response change with listening distance. This is not my opinion, but a principle of operation firmly grounded in physics and explained in the peer-reviewed AES papers re-printed by Old Colony Sound Labs. Best, Roy Johnson Green Mountain Audio |
