Al -- you are always the voice of reason. What are your thoughts? Could this be transformational or is that un likely? Thanks, Bruce. I would certainly expect it to make a very major difference, and hence be "transformational." Having no experience with it or comparable products, I of course can't say how the pluses and minuses (if any) would be likely to net out. But given Psag's comments (and I know from other threads that he is into very high quality equipment), and the writeups at the website, which strike me as confidence inspiring, as I indicated earlier it's certainly something I'll be considering whenever I next look into making a major change to my system. Best regards, -- Al |
The Holm site has an interesting, albeit monotone, video tutorial how these things work. Specifically his, but generally for all. |
Al's last post is what I needed to take the next step. A call into DEQX is in order. For some reason, my gut tells me I might get more bang for the buck going in this direction than by dropping a wad of cash into expensive speakers. I'll be back.
Al -- off topic, but there's another speaker cable thread running. I asked a serious question. If you catch the thread, please share your thoughts there. If not, my question is why wouldn't heavy gauge romex (say 10 or 12 gauge) make for good speaker cables?? What electrical properties would expensive designer cables have that heavy romex lacks. Isn't this all about resistance, inductance and capacitance?? Is there some other electrical voodoo or snake oil I am missing??? |
Bombaywalla, the DEQX is not plug and play. The correction requires separate measurements of the drivers and the room. Generating the appropriate corrections is not a trivial matter. For me, having the distributor come to my home to do the installation was the only way to go, but some do it themselves. Once a year or so, we will meet up in a remote session over the internet to remeasure and tweak the curves.
Levinskih01, the volume control is superb, although it does not have as many steps as my ARC preamp. One nice thing is that I can keep the ARC at its optimal volume level and do the volume adjustments with the DEXQ. For digital sources, I bypass the preamp. Although some tube richness is lost, I want to get the maximum resolution possible from the digital source. With an analogue source I slightly prefer having the tube preamp before the DEQX, but I don't consider it essential. |
Hello to all,
I am happy to answer some questions on design. Before that, I need everyone to truly understand the simple difference between the definitions of phase and time coherence.
Please open that earlier link to my illustration, and study the differences between waves combining. Then consider the following:
1) Time coherence-
Send the speaker a 'beep' near the crossover frequency between mid and tweeter.
Unplug the mid.
The beep coming from just the tweeter arrives at the ear 'X' milliseconds after the signal left the tweeter.
Now, turn on the mid; unplug the tweeter.
If the beep from the mid arrives at THAT SAME INSTANT, the mid and tweeter outputs are TIME COHERENT at that one frequency (perhaps not at others).
2) Phase coherence-
The mid's beep arrives at least one FULL cycle later than the tweeter's. Thus, their peaks and valleys still line up, making the two drivers IN PHASE. Yet they are not time coherent since the two beeps' beginnings and endings did not line up.
Any decent speaker is PHASE COHERENT. If not, cancellations occur at its crossover points. To market that a speaker is 'phase coherent' or 'phase linear' is only a 'feel good', to impress those who know little of speaker design.
Referring to my illustration, note that a time-coherent speaker is automatically phase coherent. It is redundant to write that a speaker is "time- and phase-coherent."
A speaker that is 'phase coherent' or 'linear phase' you can bet is not time coherent. Several speakers companies claim to make time coherent designs, but Stereophile tests reveal those claims to be completely false.
By the way, metal dome tweeters are no lighter than soft-dome tweeters. Visit Madisound.com to examine the specs of the best tweeters for yourself.
There are other misunderstandings I would like to address, but first, everyone must have a clear understanding of what IN PHASE means versus what TIME COHERENT means (hint- the latter always involves a stopwatch). For your own edification, I strongly suggest all of you discuss some examples such as:
Two cars traveling along the highway, one always fifty feet behind the other. As they travel, their RELATIVE phase is UNCHANGING (their phase relationship remains CONSTANT).
Two cars start off at the same instant, and travel along side by side. They are again in phase, since their relative positions are unchanging, and they are also now time-coherent.
===========
Two bicyclists crank their pedals at the same RPM, in the same gear. Thus, they travel at the same speed.
But notice when one rider's pedals are UP and DOWN, the other rider's are always in some other position. Because both riders' RPMs are staying constant, the pedals' RELATIVE PHASE remains constant. But these two sets of pedals are not "IN PHASE" with each other, since their peaks and valleys (their ups and downs) are not happening at the same time.
Time coherence, at its most fundamental, is about beginnings and endings lining up. Phase is only about peaks and valleys of any REPETITIVE cycles lining up. Music has BOTH characteristics.
Hope this helps.
Best,
Roy
Green Mountain Audio |
Thanks Roy. For those of us time and phase dummies who have already invested in expensive gear, and who are confounded by the difficulties associated with meaningfully auditioning speakers, what are your thoughts about using the DEQX unit to unscramble the time coherence egg?
As an fyi, which I may have mentioned in this thread or elsewhere, I auditioned a pair of Vandies and another speaker (Brand X). All conditions were held constant. Notably, the amp was the same model as my former amp. I struggled to get happy with the Vandy Treos. Spent close to 90 minutes with them. Played all kinds of music. Moved my listening chair forward, backwards, sideways. Even listened backwards. :)
I was getting ready to walkout and the dealer offered to compare another speaker that is similar to mine. Same set up, amp, etc. Brand X ate the Vandy's lunch. It wasn't even close.
Maybe I like sonic swill??? Dunno
Back to my Q: Any thoughts or comments about the DEQX?? |
Thanks Roy. Again, you beeped. I'll counter with pop, bang, whack, pow, and all the other 60's Batman fight words that seem to have something else in common. Not arguing you're goal of recreating as accurately as possible but the words describe multiple, changing tones that appear to define our hearing ability more than the actual sound. So, my rhetorical question becomes: Is the b and p really in beep or is that our imagination? |
Hi Roy, thanks for coming back. Your contributions are most welcome.
I have few questions for you.
Could you explain the pros and cons of making a speaker time coherent either by analog/digital/active/passive means?
If one were to use digital eq that only deals with room considerations and not speaker refinement, would there be a chance (and if so how much) of altering the time coherence of speakers such as yours?
The late John Dunlavy seemed to be somewhat unique (there might be another but for the life of me I can't remember the name of the manufacturer that was once a regular contributor here) in that he used W M T M W driver, as well as down firing woofer configurations. He told me that because of his previous experience in military antenna array technology he had more experience than most in wave propagation technology. Others who tout their speakers as time coherent seem to stick with more traditional W M T arrays. Is this due to size and marketing considerations, or something else?
Though most manufactures of speakers designed for time coherence seem to make fairly similar placement suggestions, they do vary a bit, from equi-T, to equilateral triangle, to wider than near, etc.. Why would that be?
It would seem to me that ideally a speaker designed for time coherence would have a sealed box, yet none of your current offerings seem to be designed that way. Am I wrong? If not, why aren't they?
|
Again, I look forward to people coming to understand the concepts behind my waveform illustrations. This understanding is necessary to our discussion here, because we then have agreed on the nature of these concepts at hand and also on some vocabulary.
And I wish my answers could be shorter, but that would leave out necessary details- the same ones glossed over/ignored by the press.
Bifwynne,
DEQX seems fine in theory, and certainly makes a positive difference. For me, it has serious limitations because it cannot measure exactly what needs to be corrected. This leads to results that depend on the music being played and sometimes a limitation in one's seating position.
In particular, DEQX cannot see the immediate reflections from the cabinet surface surrounding the tweeter. It cannot correct properly for anything happening below middle C because of floor-bounce effects on the microphone are not the same as they are to our ears on music.
There are other issues, but to me, those are the two largest ones. I find that a much higher level of coherence is achievable passively.
Regarding Treos-- I've not heard them and it's never good for me to comment on the sound of other speakers. I say to trust your ears above all. And you are right to listen backwards and from another room if possible! I can point out Treo, like other Vandersteens and most speakers for that matter, has a terribly complex crossover circuit, made of what I know are not the most transparent parts.
Ngjockey, thank you for your comments. Our experiences with everyday sounds and noises allows us to use them to imagine the SHAPE of a sound, which means its starting and stopping, and what happens in between. The next step up the chain is to imagine the combination of that sound with another, or literally just hear it via programming a synthesizer.
Remember, in the word 'beep', the opening 'b' has its own shape, since the lips are opening. That 'b' is is a CHANGE that happens along a certain TIMELINE, and we recognize its waveform's CHANGING SHAPE as unique to the letter 'b'. The same happens at the end when the lips close, but like 'p' instead. When you imagine hearing only the the middle 'eee', that is exactly what comes out of a sinewave test tone being switched on and then off. Which is exactly what I illustrate in my drawing.
Unsound, I am glad you find my comments useful. Thanks! Using digital EQ to treat room problems seems like a good idea, but again, just know that what you will measure is not what you are hearing-- not to say there will not be some or even a good amount of improvement. If it is used just for subwoofer correction, there are issues in most every sub's design that look exactly like room problems to the measurement microphone.
I think it best to first measure and correct a sub up close with the mic, then use that correction as your basis for further corrections YOU HEAR out in the room, listening to string bass run the scale and to kick drum (the first for tone balance; the second for transient alignment with the main speakers).
Mr. Dunlavy decided early on that driver symmetry about the ear was important, since it was important to his microphone and to his antenna-derived math. Turns out that when you are seated, the MTM arrangement is not important to the ear. In particular, you hear the tweeter's sound come from the mid when just one tweeter and one mid operate time-coherently without cabinet-surface reflections.
An MTM arrangement, including the infamous D'Appolito arrangement, always places one mid above our heads. This causes the image to be unstable with small head movements, and just plain poor for anyone off center. Why would this be? Because we have a head between our ears and a chest below them. Thus, with a small head movement to the left, much more middle-range sound literally leaks over the TOP of the head to the right ear than it does from the mid placed below the head, so the image jumps to the left speaker. This is also true for any large sound-source, such as a panel speaker or a so-called 'line-array', for the same reason.
When only one mid is used time-coherently with one tweeter, and then placed right in front of us, that single source of sound then leaks over the top of the head by the same amount no matter how much we rotate, move sideways or even stand backwards! So the image remains stable, even when far off-center, if and only if the speaker baffle is also narrow and reflections are not allowed from around the tweeter and mid.
MTM also leads to room placement issues, since sidewall reflections throughout the voice range are more complicated than when only one time-coherent tweeter and mid are used.
Time-coherent Coax operation ala Thiel would be best, except there's no way to avoid the intense tweeter reflections off the mid's cone. Also, a terribly complex crossover is required to get the tweeter's timing right. There are other limitations.
Finally, with two woofers placed high and low, for a WMTMW alignment, the bass response in any room becomes unpredictable, since you are driving bass near TWO boundaries, with your ears trapped in between.
Since everything is a compromise, a one-woofer arrangement works best when the woofer is a certain distance from the floor, in medium-size rooms, with a certain crossover frequency. But in those rooms, the bass output will then be predictable, which helps me. Nothing wrong with having the extra 'slam' from two large woofers- it just requires a very large room to make them perform as one. Then again, a very large room I find uncommon.
Speaker placement/spread is similar for very many speakers using slender front baffles, regardless of their crossover design, when these speakers are placed in 'good' rooms. This is because we need to hear a certain amount of crosstalk for the image to be continuous.
Sidewall reflections and reflections off all the fancy gear piled up between the speakers affects the final spread and the toe-in. Speakers having a large amount of reflections off their fronts are sometimes used with less toe-in, so those reflections are not shot as directly into one's ears. When there are many center-reflections (from that gear or off a video screen), toe-in is reduced. When a speaker is not time-coherent, its particular phase shift may mean those speakers sound best placed close together, pointed nearly straight ahead.
Sealed box is the best for woofers, but the market prefers more efficiency and compact enclosures, so our woofers are smaller, requiring a port. Our new three-way coming out uses twin 6.5-inch woofers, each ported at 40Hz in its own enclosure, for a sensitivity of 91dB with the same cone area as one 11-inch woofer. A single ten-inch sealed-box woofer would be in a cabinet half again larger, with only an 88dB sensitivity (requiring twice the power). The mid and tweeter would also need to be turned down by 3dB -not a great solution.
Again, I hope this helps! I realize other questions still remain, posed earlier in this tread, but I thought it best to get these out of the way right now, so I can look forward to folks' thoughts on my waveform illustrations. I will endeavor to cover the other questions soon.
Best,
Roy |
Hi Roy, good to note that you are back on this thread & have been kind enough to give us your time on this subject. Thanks! yes, I personally have looked at the waveforms (on the photobucket.com website) you pointed us to. I understand it much better now thanks to your recent post where you explained the diff between time-coherency & phase-coherency. I was looking at the waveforms but did not draw that conclusion; now I have! Also, the 2 cars & 2 cyclists analogy was very helpful. I have no particular question for you but I'm hoping that many other members who are on the fence re. time-coherence & others you are determined nay-sayers of time-coherence will take this opportunity of your being on this thread to ask their questions.... Bifwynne had a question re. the electrical properties of a driver & how that translated into distortion. Almarg enunciated the issue quite well & I've cut & paste his text below:
"Consider a simple two-way speaker having a first order crossover consisting of a capacitor in series with the tweeter, and an inductor in series with the woofer. For each driver that will result in well behaved 6 db/octave rolloff characteristics, which will result in time and phase coherence if other aspects of the design are also supportive, **IF** the impedances of the woofer and tweeter are purely resistive.
However I believe Bruce has been alluding to the fact that the impedances of the drivers are not purely resistive. And it would be more accurate (if still somewhat oversimplified) to electrically model them as consisting of a resistor and an inductor in series.
So the question then becomes: Doesn't the presence of that inductive component of the driver impedance (especially in the case of the tweeter) cause a deviation from first order 6 db/octave behavior? And if so, to a degree that may audibly compromise phase and time coherence? And if so, is that or can that be compensated for in other aspects of the speaker's design?"
Can you please address this question for us? thank you. (My understanding of this question was that the driver is resistive in its pass-band frequency range where its response is flat. I understood that it could be flat response in its pass-band only if it was linear i.e. resistive over that range of frequencies but I could be totally wrong). |
Roy, what about the 90 degree phase shift of a single pole crossover? |
Consider a simple two-way speaker having a first order crossover consisting of a capacitor in series with the tweeter, and an inductor in series with the woofer. For each driver that will result in well behaved 6 db/octave rolloff characteristics, which will result in time and phase coherence if other aspects of the design are also supportive, **IF** the impedances of the woofer and tweeter are purely resistive.
However I believe Bruce has been alluding to the fact that the impedances of the drivers are not purely resistive. And it would be more accurate (if still somewhat oversimplified) to electrically model them as consisting of a resistor and an inductor in series.
So the question then becomes: Doesn't the presence of that inductive component of the driver impedance (especially in the case of the tweeter) cause a deviation from first order 6 db/octave behavior? And if so, to a degree that may audibly compromise phase and time coherence? And if so, is that or can that be compensated for in other aspects of the speaker's design? thanx to the above question posed by Bifwynne & well enunciated by Almarg, I did some research to try to understand what the issue might be. As I stated in my prev post, my understanding is that if the driver is used within its flat frequency response range of frequencies then that particular driver can be considered linear or purely resistive. And, some research seemed to indicated exactly that! Allow me to share: When Bifwynne asked the above question, I discovered that it took the me towards understanding the Thiel/Small (or T/S) parameters of loudspeaker drivers. There's much to learn here but that's a subject for another day. Some info that I thought is pertinent to this subject: There is a T/S parameter called Re (R-little-e) & a cut/paste from Wikipedia " Re Measured in ohms (Ω), this is the DC resistance (DCR) of the voice coil, best measured with the cone blocked, or prevented from moving or vibrating because otherwise the pickup of ambient sounds can cause the measurement to be unreliable. Re should not be confused with the rated driver impedance, Re can be tightly controlled by the manufacturer, while rated impedance values are often approximate at best.. American EIA standard RS-299A specifies that Re (or DCR) should be at least 80% of the rated driver impedance, so an 8-ohm rated driver should have a DC resistance of at least 6.4 ohms, and a 4-ohm unit should measure 3.2 ohms minimum. This standard is voluntary, and many 8 ohm drivers have resistances of ~5.5 ohms, and proportionally lower for lower rated impedances. " there's also a T/S parameter called Le (L-little-e) " Le Measured in millihenries (mH), this is the inductance of the voice coil. The coil is a lossy inductor, in part due to losses in the pole piece, so the apparent inductance changes with frequency. Large Le values limit the high frequency output of the driver and cause response changes near cutoff. Simple modeling software often neglects Le, and so does not include its consequences. Inductance varies with excursion because the voice coil moves relative to the polepiece, which acts as a sliding inductor core, increasing inductance on the inward stroke and decreasing it on the outward stroke in typical overhung coil arrangements. This inductance modulation is an important source of nonlinearity (distortion) in loudspeakers. Including a copper cap on the pole piece, or a copper shorting ring on it, can reduce the increase in impedance seen at higher frequencies in typical drivers, and also reduce the nonlinearity due to inductance modulation. " So, it looks like a significant source of distortion is due to voice-coil inductance modulation (variation) & not so much the fact that the voice-coil has actually a DC resistance associated with it (as Bifwynne & Almarg were thinking). So, how to tell when viewing/reading a driver's specifications that this inductance modulation is an issue? I don't really know but I took up Roy's advice to look at driver specs on Madisound. On the Madisound I randomly selected "Seas Prestige" - Seas makes good drivers, "Prestige" seems like its upper-end line. Here's the link to one of their 8" woofer drivers: http://www.madisoundspeakerstore.com/approx-8-woofers/seas-prestige-8-woofer-cd22rn4x-h1192 Lots of good info on this page but reading the specs might be Greek to most of us - I wanted to call your attention to the graph which shows SPL (left vertical axis) vs freq & impedance (right vertical axis) vs freq. From a Wikipedia page on Speaker Electrical Characterisitics I learnt ".....the effective electrical impedance of the speaker to be at its maximum at Fs, shown as Zmax in the graph. For frequencies just below resonance, the impedance rises rapidly as the frequency approaches Fs and is inductive in nature. At resonance, the impedance is purely resistive and beyond it—as the impedance drops—it behaves capacitively. The impedance reaches a minimum value (Zmin) at some frequency where the behaviour is fairly (but not perfectly) resistive over some range. A speaker's rated or nominal impedance (Znom) is derived from this Zmin value (see below)." This Seas driver seems to have a 6.1 Ohms impedance at, say, 150Hz. Using the info from the Wikipedia site, the driver must be mostly resistive at 150Hz to give its minimum impedance at that frequency. Look at this driver's frequency response from 90Hz - 400Hz: practically ruler flat & look at the impedance variation over this same range - goes from 6.1 Ohms to 10 Ohms on both sides of 6.1Ohms, which is a small change in driver impedance compared to the change over the entire 20Hz-20KHz. The driver appears to be mostly resistive in this frequency range. I *think* the answer to Bifwynne's question is that if you use this driver in the 90Hz-400Hz range, you will get a mostly resistive driver whose impedance varies very little (between 6-10 Ohms), it's frequency response will be flat/linear & the phase distortion will be minimal meaning that the voice-coil inductance modulation/variation (which is a significant source of distortion) will be negligible. Roy, please correct me if I'm wrong. Thanks. |
Hi to all,
Bombaywalla, you ask-
"So the question then becomes: Doesn't the presence of that inductive component of the driver impedance (especially in the case of the tweeter) cause a deviation from first order 6 db/octave behavior? And if so, to a degree that may audibly compromise phase and time coherence? And if so, is that or can that be compensated for in other aspects of the speaker's design?"
My answer is YES, but we and others use simple Zobel networks on woofer, mid and tweeter. These offset the changes in impedance at high frequencies.
Normally, the impedance of a woofer, mid or tweeter (= 'driver') becomes 9, 12, 30 Ohms as we go higher and higher up the scale (= "inductance"), instead of staying constant at say, a steady six Ohms, which is what any type of crossover circuit wants to see-- a flat impedance 'curve', so that all of its capacitors and inductors do what they are supposed to do. Any change in impedance literally turns some of those circuit parts off, with the result 'not measuring right' to the microphone.
A Zobel circuit offsets a voice-coil's rise in impedance with increasing frequency, and is quite simple to construct: a capacitor is connected to a resistor (= 'in series'). Those two are then placed in parallel with the driver's +/- wires, the capacitor connected usually to the "+" and the resistor to the "-". I hope that is clear!
With those two Zobel parts placed 'across' a woofer, mid or tweeter, the result is literally a 'Y-adaptor' to the signal coming from that driver's crossover, because two paths now exist for the signal. One goes through the driver back to the crossover as usual, and the other through that Zobel cap, then its resistor, and thence back to the crossover.
That woofer, mid or tweeter's impedance is still going up and up the higher up the scale we go, but our Zobel's capacitor has an impedance that is going down and down by the same amount/at the same rate. This is its 'correction', with the resistor limiting/shaping the amount of correction provided.
When those two paths are made to change by 'equal and opposite amounts', that driver's crossover circuit then sees 'no change in the impedance at any frequency', so goes the theory.
Where most every designer goes wrong is by making the assumption that the electrical impedance curve of the voice coil is what one is measuring and correcting. Not so, unfortunately. There are many other impedance curves that overlay, thus hide, the real electrical curve one is looking to flatten. These other impedances include:
- the mechanical impedance of the driver's suspension and any ferrofluid used.
- the acoustic impedance from how each driver is coupled to the air in front of it, and behind.
- any cone/dome flexing (= mechanical impedance changes).
- the mechanical impedance changes caused by the size of the boxes used for woofer and mid, and a tweeter's rear-chamber.
- what happens in the various types of fibers placed behind drivers to absorb their rear waves.
Then (!) most of those change with loudness, especially with 'average' drivers. Some of those also change when a voice coil is moving inwards versus out, again especially on 'average' drivers. Visit the Klippel company's website to see some of their measurements for these problems, now done automatically by their unique computer and programming- such a smart designer! I had to perform them manually, darn it. On our website, I describe much of what can be done to minimize or avoid these last issues.
Finally, a few years ago, I found a way to use the values of cap and resistor in that simple Zobel circuit to perfect our final acoustic phase down to near Zero at all frequencies. While I still must update our website about this, it's never been accomplished by anyone, as seen in the values they still use in their Zobels. If we sought a patent, I'd have to reveal how to come up with 'the right numbers'.
====
FYI, it is a big mistake to flatten the low-frequency change in impedance caused by a woofer or mid's box-size and a tweeter's rear-chamber size, by using a different type of Zobel circuit. Those who do this type of correction do not understand what these impedance rises at resonance actually represent electrically, mechanically and acoustically. The result is poor sound, and usually a very difficult speaker to drive in the bass.
===
Addressing another question of yours right above: Thiel/Small parameters are a guide only to box tuning, nothing else. However, those equations turned out to give, at best, only an approximation of the correct box size, because the impedance values plugged into it are not 'right' because we have left out all of those other impedances I just described.
Box-modeling software relies on those simple equations, so they cannot give you the exact box size for a woofer or a mid. One must build several test boxes to determine the actual 'best size'.
FYI, Seas' best drivers are their Excel line. Even so, have a look at the high-frequency cone-breakup resonance in their best metal-cone woofer. Designers believe a high-order crossover and a notch filter 'fix' that problem. Not true, as that cone resonance is also triggered by lower-frequency sounds and 'noises'.
Example- your car's dashboard buzzes from the low-frequency 'thump' of a pothole. This concept and its math are taught in high-school level physics, which is what most speaker designers never study. It is why those metal-cone drivers still sound 'metallic'. Stereo magazines and reviewing websites never mention these facts, but then again, it makes sense how they do not want to upset any advertisers!
For tweeters having a strong ultrasonic resonance from their metal dome breakup, the same thing still happens,with the ultrasonic HF resonance ringing out. However, what is heard instead is its effect on the audible-treble tones. That is a 'modulation distortion', and sounds like perhaps a 'zing' to the treble, or again, a metallic sound. These are all factual statements supported by physics theory and math, and by measurements. They are not 'Roy's opinions'.
===
Omsed,
The phase shift of that one inductor used on a woofer with its Zobel produces -45 degrees of phase shift (= time delay) at the crossover point. The single capacitor used for the tweeter's first-order crossover gives the opposite shift of +45 degrees (= time 'advance'). The difference between these two is 90 degrees.
When a website or text makes this mistake, that writer had never looked at the simple math involved, which any competent electrical engineer should have learned in their first Filter Theory class. Only hearsay is being passed on to you, including the non-existent 'downwards tilt to a first-order speaker's radiation pattern'. A totally bogus claim. There, the math was completely mis-interpreted.
The important aspects of this 90-degree DIFFERENTIAL produced by a simple first-order crossover, proper Zobels and really good drivers are
a) it remains a CONSTANT 90-degree difference between those two drivers as we go up or down the scale, and
b) that constant difference of 90 degrees allows the sonic outputs of those two drivers to always add up to the one original wave, having no added time delay, which is totally non-intuitive.
A 'perfect summation' happens because those two drivers are operating 'in quadrature' (90 degrees being one-fourth of 360) at every frequency. The math involved shows their outputs, one lagging, one leading, really do combine to make only the one original wave having neither lag nor lead. Weird.
No higher-order crossovers can maintain this CONSTANT phase differential, so they produce a time delay, a phase shift, that changes with frequency, perhaps 'linearly' but always changing.
This varying time-delay is what DEQX-type components are trying to correct, and what regular digital crossover circuits never attempt to correct (offering only fixed time delays, such as one millisecond). To correct the varying time delay, a heck of a computer is required, hence the high cost of DEQX type of gear.
Measurement issues and limitations still confuse DEQX type of gear, for two reasons- we cannot (yet) program that computer to how we actually hear on music, and that a measurement microphone cannot resolve the (countless) reflections off the front of a cabinet. If I had spent money on a DEQX, I would first place an "F-11" pure wool felt all around the tweeter, and then run the calibration routine.
Best,
Roy
|
I forgot to mention a couple of things:
When a driver is being run full-range with no crossover or Zobel, its changing impedance curve has no effect on its tone balance when using solid-state amps, but only on tube amps via interaction with their much higher output impedances. For a tube amp running a 'full range' driver, a voice-coil Zobel circuit on that driver would return its tone balance to 'factory spec'.
When a speaker has a flat impedance curve, that does not indicate if this speaker is time-coherent. From the outside, all we can see is how the many different impedance curves I described above combine into one curve.
Best,
Roy |
Wouldn't it be easier just to tilt the speaker slightly backwards? I just want to listen to good music |
Hi Roy,
Thanks for the discussion, but though the difference between the woof and mid remain constant, there is a difference, yes? And that means that the wave launch of a transient will not be the same for the 2 drivers, correct? The are not time aligned, it would seem. Even if the sum of the outputs through the crossover point remains correct, are we not stuck with the constant time differential between the 2 drivers?
Could you tell me what I am missing? |
07-17-14: Sounds_real_audio
Wouldn't it be easier just to tilt the speaker slightly backwards? tilting the speaker backwards attempts to merely align the acoustical centers of the drivers such that the sound from all the drivers reaches your ears at the same time. But what about the damage done by the higher-order x-over to the phase & time coherency of the music signal? This damage is the phase distortion that Roy is talking about all along. That cannot be corrected by merely tilting the speaker backwards.
I just want to listen to good music 'good' is a relative term - your music selection is best for your taste in music. Others might not find it 'good' at all... OR, did you mean 'good' as in well reproduced playback sonics?? |
Thanks Roy for the detailed explanation. :-) Was very helpful, as always.
OK, I wont put as much emphasis on the T/S parameters any more. I thought that I could read them & determine something about the quality of the driver. Not so, it seems.... |
Sounds Real-
You are indeed right about 'just tilting back the front face'. That can be enough to line up the acoustic centers of woofer/mid and tweeter, so that the drivers are possibly in their best positions to combine properly at your ears, no matter what crossover design is used. The high-order crossover circuits then put more and more time delay on the signal the lower and lower down the scale we go. That cannot be fixed.
And then to make the first order crossover work correctly, one must choose the correct drivers to begin with. I hope this clarifies a bit more for you.
Omsed-
You ask "though the difference between the woof and mid remain constant, there is a difference, yes? And that means that the wave launch of a transient will not be the same for the 2 drivers, correct? The are not time aligned, it would seem. Even if the sum of the outputs through the crossover point remains correct, are we not stuck with the constant time differential between the 2 drivers?
Could you tell me what I am missing? "
Yes, I agree. Again, with the right drivers, Zobels, and first-order crossover design, there will always be a time difference created by that constant 90-degree differential, a constant difference 'in degrees only' at every frequency we examine.
90 degrees is one-fourth of any sinewave's period. At a 3kHz crossover point, that wave's period is 1/3000 of a second. One fourth is 1/12000 of a second. This is the time-difference between the mid and tweeter at this frequency. If we choose 1000 Hz instead, the time difference would be three times longer, 1/4000 of a second.
I can only tell you that the math of "two waves of the same tone traveling out of phase with each other by 90 degrees" will measure and sound like one wave having no time delays. Perhaps you must do the math yourself to see this-- I certainly understand that feeling! Again, the key words to look up are "operating in quadrature".
Bombaywalla-
I apologize if I gave the wrong impression. T/S parameters are quite important, as they tell us a great deal about how the driver will perform in any box.
They just do not give the exact box size, which was the hope. The error can be 10 to 20% off of the correct box volume.
A real test box's performance is determined by listening and then measuring its impedance curve and resonant frequency, to find out the Qts and Fs. That tells us how close we came to meeting the T/S ideals with that test box. Then build another...
Best,
Roy |
If you were to try to line up the drivers how would you find the " acoustic center" of a driver. In a midrange or woofer would that be about half way out from the center or would it be further out because the majority of the area of a larger driver would be closer to the perimeter? |
Hi Roy, thank you for your response.
At the risk of appearing argumentative; here on Audiogon another speaker designer has suggested that placing multiple woofers in a room at different distances could be beneficial towards evening out standing waves, if that were the case; wouldn't W M T M W speaker arrays have some advantages? I'm not aware of any speakers that are touted be time coherent having more than one tweeter per channel, are there any?
As for sound bouncing differently above and below ones head, wouldn't that be typical during live musical performances? Wouldn't the wave size from midranges and woofers (and live performers) be large enough to extend above and below a typically seated listener's head?
Thiel's concentric drivers appear to be flat, so reflections should be minimized, no?
I have no direct experience, but I seem to recall that DEQX suggests that speaker correction should be first done close to the speakers and then followed with room correction at the listening positions.
Another question if I may; could horn loaded speakers be time coherent?
Thanks again! |
Hi Unsound, Thank you for your thoughts. The use of multiple subs does smooth out standing-wave issues. The math used for the theory behind that is formed from adding together the simple sinewave/wavelength equations for standing waves you have seen for bass tones and room modes before. That is fine for long-running test tones, for movie sound effects, and certainly for a pipe organ. The test tones used to adjust those multiple subs are long-running, and not found in music. When the time-arrivals at the ear between multiple subs are 'excessively different', you would think we'd hear stumbling or mumbling on string bass, drum kits and perhaps even cello. But if those subs are not allowed to go above ~40Hz, those issues are bypassed. === WMTMW bass problems arise from both woofers being close to the bottom and top surfaces of our room. This is a 'very symmetrical' situation, which always produces the strongest standing waves. Another 'very symmetrical' layout would be subs placed in every corner. Have a look at this drawing: ReflectionsAlso, do note that WMTMW woofers operate to 150 or even up to 300Hz, which is above middle 'C' on the piano. In these upper ranges, changes are very audible standing vs. sitting vs. walking into the kitchen. === You ask about the over/under head effect of an image jumping when hearing live sound from vertically-large concert speakers. Good question. I can say I've never heard that problem, including from long line-source speakers. Remember, most concert sound systems are mixed close to mono, so everyone hears everything. And in most live situations, sound from a tall concert speaker comes to you from a narrower vertical angle than when at home listening to a six-foot tall speaker ten feet away. Also, I probably did not make it clear enough before that the over/under head leakage of sound to the opposite ear is caused by the WMTMW use of double mids, not double woofers, because of those shorter wavelengths vs. the size of our skulls. === We get reflections off any hard surface-- it matters little that a Thiel's mid surface might be flat or corrugated around its coax tweeter. This is because any 1" tweeter, without a several-inch deep horn around it, is omnidirectional below 5kHz. That means it pushes waves between ~1kHz and ~5kHz across the face of the cabinet, since they cannot escape to the rear. So those pressures escape to the front as they move across the face of the cabinet. Hence, reflections. === Putting the measuring mic for DEQX up close to a speaker is pointless (except for fixing up a subwoofer), as what the mic would then be hearing is coming from drivers at much different path-length-differences to the mic compared to the path-lengths to an ear ten feet away. We all know how walking up to a speaker changes everything we hear. Perhaps they are suggesting this for fixing one driver at a time. That has problems too, because any driver's tone balance is different at ten feet away vs. ten inches away. === Horn speakers can be made time coherent, but our best technology leads to that speaker being at least a four-way if not a five-way design, to stay far enough away from horn cutoff points on the low-end of each driver, and the high-frequency breakups which come from running a large mid high into the upper voice range, and a compression driver with a 4-inch diaphragm into the high treble. Also, with 4 to 5 horns stacked up, their vertical height would make for very strong changes as one stood up or even just sat higher. The nicest sound I ever achieved on horns was to use the lowest order of electronic crossover possible (12dB/octave, 'second-order') on a three-way horn system. The tweeter horn was moved far back on top of the mid's horn, and mid horn `way back on top of the woofer's folded horn, to equalize the driver-to-ear distances for people twenty+ feet away. This describes a system I put together for Taj Mahal. I had to add a small amount of EQ to smooth the mids, boost the ultra-highs, and for flat output to 40Hz. Of course I had to reverse the polarity on the mid horn because 12dB/oct. crossovers need that to avoid cancellations at the crossover points. Since everyone was 20 to 70 feet away from either the left or right speaker (mixed to mono), everyone heard a smooth blend from a speaker whether seated of standing. Sure there was phase shift from those speakers, but it was far less severe than any higher-order crossovers would have been. I received very many compliments on the ease and clarity of the sound. === I hope everyone sees my answers are lengthy because I include WHY something is audible or will measure a certain way, so you finally get a proper technical perspective on the VARIABLES that must be considered, and also HOW they must be considered. Magazines and reviews leave out all these variables-- make of that what you will. Best, Roy |
Hi Roy, I don't mean to beat a dead horse, but I think that in a W M T M W array, it would be very rare indeed for any of the drivers to be equi-distant to floor and ceiling, and those surfaces would typically be different enough to reflect some frequencies differently as well.
I would have guessed that in an attempt at a staggered driver time coherent horn design that the horns themselves would get in the way of each other with the surrounding horns causing early reflections. Wouldn't a deep throat coincidental multi-driver have more early reflections than a flatter design? |
Sounds Real,
For an eyeball estimate, the acoustic center is approximately where the voice coil former meets a cone or dome- the glue joint. But this is true only in the upper-middle range of any driver, whether tweeter, mid or woofer, where each one's frequency response is still flat.
To measure it (within +/- 1/8th inch at best for a tweeter, much more for a woofer), one sends an impulse, a click, to a driver having no crossover.
On a `scope, one examines when that click arrived compared to when the `scope's sweep was triggered by the click electrically.
Now, what we are looking for as markers will not be the top of those two spikes that click generated. We are looking for when each spike just begins to turn upwards from 0.0 at its bottom-- when each just begins to rise up. That is a very difficult transition to judge, which leads to inaccuracy.
Regardless, that time-difference times the speed of sound is your distance from the mic to the acoustic center in a driver's upper-middle range. Compare that to the tape measure distance and you often get close to the eyeball estimate I mentioned above. Of course, the test mic will be expensive, not a $200 special, for those cheaper ones have their own phase shifts in the audible range. Figure $1000 for a proper mic, plus a $1000 wideband mic preamp. Even so, the results will still be rather inaccurate. I was able to find ways around this, fortunately.
===
Unsound,
It is not that the two woofers are equidistant from their surfaces but the fact that we have two (four in stereo) woofers rather near two surfaces with you living in between.
Do have a look at that new drawing I posted to get an idea where "the bass source really is", which is my red dot in that drawing. Imagine what standing waves would then occur in between a red dot on the ceiling and one on the floor when the measuring mic/your ear is placed somewhere in between. Double trouble has been my experience.
No doubt about the outer horn-surfaces making reflections. But those reflections would go mostly upwards, and we can apply wool felt or acoustic foam to minimize most of them. I still think the biggest problem to be getting far enough away from the speakers so stand up/sit down differences would not drive us crazy- a large living room, say 30 x 40 feet is probably enough.
I would like to hear a Klipschorn corner horn triamped with time-delays applied to its mid and tweeter, since the woofer is so far back inside (~4 milliseconds) and the tweeter is so far in front of the mid driver (~2ms). Again, one would be stuck with using second-order crossovers on the drivers with the mid driver in inverted polarity.
Best,
Roy
|
Roy. thanks again. ^ Wouldn't using 2nd order cross-overs compromise the whole effort? |
Certainly, but a lot of other problems would be fixed. |
Roy. thanks again. ^ Wouldn't using 2nd order cross-overs compromise the whole effort? |
I just remembered the name of the old poster that started a time coherent speaker company; Karl Shuemann. To be fair to all here's a list of those companies that have made time coherent speakers that I'm aware; of starting with Karl's (which fortuitously falls into alphabetical order):
Audiomachina
Dunlavy
Greenmountain
Meadowlark
Quad
Thiel
Vandersteen
|
^oops!
I forgot the original Ohms with genuine Walsh drivers
and perhaps their progeny from HHR-ExoticSpeakers, German Physiks and Huff might qualify too. |
Unsound, are all of the companies listed in your last post still making time coherent speakers. Of course, Vandersteen, Thiel and GMA are obvious. Not sure about the others.
Btw, been in contact with DEQX. Will try and arrange a home demo. If I can pull it off, I'll report back. |
Bifwynne, Dunlavy and Meadowlark are gone. I'm not sure if the newer Quads qualify. Some Thiels are, for the time being...the future looks glum. I have doubts that the newer Ohms qualify. I'd hazard a guess that the HHR ExoticSpeakers do, but I haven't seen verification. I would say the same for the German Physiks and similar Huffs, at least for the models that use their DDD drivers exclusively, I have doubts that their full range models qualify. FWIW, I have confidence that the others still do. |
Bifwynne, if you're in contact with DEQX, if you wouldn't mind, perhaps you might ask them why their system approach includes analog conversion rather than keeping the whole stream in the digital domain? |
Bifwynne, unfortunately all of the companies stated in Unsound's post do NOT still make time-coherent speakers. (John) Dunlavy quit making his speakers long time back & I believe that he, unfortunately, is not with us anymore (correct me if I'm wrong but that's what I remember reading in one thread). And, Meadowlark also does not make speakers anymore. Unfortunately there was something very unsavoury that went down w/ Pat McGinty (owner/designer of Meadowlark), his company & the location where he was making speakers. Audiomachina also does not make time-coherent speakers anymore. I was reading an old Meadowlark Osprey speaker review on enjoythemusic.com The Meadowlark Mantra
Three design principles underlie every Meadowlark loudspeaker: time coherence, first-order crossovers and transmission line bass............
McGinty argues articulately on his web site that time coherence is essential to long-term musical satisfaction and avoidance of listener fatigue.....
http://www.enjoythemusic.com/magazine/equipment/0204/meadowlarkosprey.htm link provided if anyone is interested in the full review. |
Bombaywalla, unless this:
http://audiomachina.com/philosophy/
is out of date, I think they still are. |
Bifwynne, Unsound, found this website of Meadowlark/Pat McGinty showing off all the Meadowlark speakers. I loved the looks of the Meadowlarks due to their superb wood finishes - always such a pleasure to see! http://www.patmcginty.com/index.html |
Bombaywalla, you are correct we lost two great proponents of time coherent speakers and both gracious gentlemen as well, John Dunlavy and Jim Thiel. R.I.P.
|
Hey guys I found this wonderful, very simple, animated essay from Pat McGinty (of Meadowlark) explaining time-coherence (in which he believed sincerely). Please read this (in the beginning he goes off the deep end relating the birth of the stars to time-coherence but please bear with him - he's trying to make a point: http://www.patmcginty.com/Dbench2.htm |
You are right Unsound, Audiomachina's website seems to be up-to-date (I saw one post dated June 9, 2014) so it seems that they are alive & well producing time-coherent speakers. Thanx for the correction. And, BTW, you guys have to read these "Pearls" (of speaker deisgn) from the owner/designer of Audiomachina speakers - esp. for determined naysayers. These notes echo practically 100% what Roy Johnson has been trying to educate us with since 2002 when that time-coherence thread started by RBischoff appeared on Audiogon. Just like Roy's texts, this essay on speaker design is very well written (& I can see time & again the points that Roy has tried to drive home into us re. 1st order x-overs & time-coherence) - a superb & easy read: http://audiomachina.com/pearls/ |
Duh, I forgot to add Duntech to the list. |
Unsound, not sure I understand your question. You ask why the DEQX "system approach includes analog conversion rather than keeping the whole stream in the digital domain?" Perhaps ... it is because the device is inserted in between the source components and the power amp. Seems that at some point the device has to go digital to analogue in order to drive the amp.
Am I misunderstanding your question?
I hope to hear back from DEQX this week. I've got to nail this time coherence issue one way or the other.
Btw, I been shopping around for insulated 10 gauge solid core copper wire. I intend to make my own speaker cables. May cost me $15 bucks ... gasp! |
^Please forgive me, I was confusing DEQX with Holm Acoustics. |
Looked up quadrature and unless you're masochisticly inclined to imaginary vectors, this might be easier...
http://sound.westhost.com/ptd.htm |
Ngjockey,
I have looked at this site for very many years. The Soundwest site has enough errors to mislead someone relying upon it for 'basic information' and a bit of the math.
Specifically:
In its Section one, the author does not understand a tweeter is still not time coherent when its wires are flipped over to invert its polarity (paragraph 3). He goes on to mis-represent the amount and degree of cancellations between mid and tweeter when the tweeter is not in the right position (below Fig. 5). What he presents instead is a graph showing TWO IDENTICAL, PERFECT, FULL-RANGE DRIVERS interfering, not a graph of one mid crossing over to one tweeter.
In its Section two, the information in the paragraph below Fig. 10, about phase shift and its audibility on square waves, is just plain wrong (even stating we can't hear it, then giving real examples of how we can hear it).
In Section four, on the audibility of phase distortion, not only is he wrong about its audibility, but he goes on to present an argument based on sound coming from live instruments.
He does not get it that we want to PRESERVE whatever phase relationships exist in the music, no matter where we sat, no matter where the recording microphones were placed. Can you spot the big flaw in his argument based on hearing live music? I have seen this exact bad-logic presented on many other forums as the main reason not to bother with making speakers time coherent.
In his Conclusions, he claims the room acoustics and bad recordings will hide much of what should be gained from making the speakers time coherent. To me, that makes it obvious he's never lived with time-coherent speakers for any length of time.
He mentions how a little pair of speakers in his workshop will reproduce a square wave at one frequency if he holds the mic in just the right place. I can see he does not recognize those speakers likely still have a phase shift of 360 degrees at some frequency, and how that will make a CONTINUOUS square-wave signal still appear square.
He does not remember that 360 degrees of shift at some frequency means the previous square-cycle is then projecting/delaying some of its frequency-components INTO THE NEXT CYCLE, and so on. He should have been examining only the first half of the very first square-wave cycle-- its first up-and-down only, to figure out what a speaker is doing.
===
His are the answers I find quite common on the web, but not in most of the professionally-reviewed papers published by the AES. Their important papers on speaker design can be purchased by anyone as their three Audio Anthologies books. There are still errors in too many of those, but one must know calculus and physics quite well to find them.
I think the general public should not take a writer's claims about audio design for granted, unless the writer also presents the scientific concepts and logic behind those concepts, and WHY those have to be correct. Which is what I've endeavored to do.
Best regards,
Roy |
In light of Roy's feedback to the sound.westhost.com material, here is (what seems) a much better website to read up on quadrature signals. The article deals with complex signals but it is not complicated - the graphics make it much easier to understand. if you don't want to read the article, scroll to Fig 10 directly & you will see why 2 signals in quadrature (i.e. separated by 90 degrees of phase) add up to a constant i.e. adding 2 signals in quadrature does not give you another signal; rather it gives you a scalar/just a number. As Roy was saying earlier on - this can happen ONLY with a 1st order cross-over where the phase difference between tweeter-mid, mid-woofer is 90 degrees & when these signals add up at the listener's ear they appear as tho' there was no additional delay thru the x-over. http://www.dsprelated.com/showarticle/192.php |
That's how I was presented this subject. A good link, thank you. Made me flash back to all the horrible homework involved. And then, as the math of physics became ever more advanced during grad school, one wound up using this math daily...
Best,
Roy |
Here are my answers to important questions posed earlier, and some clarifications. To the OP: Psag, you originally asked if a sloped baffle is important. Speaker designs that avoid this are instead using the phase shifts of their crossovers to make sure there are no cancellations/suckouts in frequency response. That is about all their designers look for/measure during the design phase, since they do not make any measurements in the time domain. I think those designers would have an easier time developing their high-order crossovers if their drivers were first stepped back from each other, as on a sloped baffle, and they got rid of the sonic reflections off their front surfaces. === Bifwynne, at the beginning, you asked "perhaps someone could explain in layman's terms what causes speaker to operate out of phase. Does it have something to do with the use of caps and chokes in the x-over? Or perhaps the attribute of a dynamic speaker creating its own back EMF by reason of the voice coil moving in a magnetic field?? Incidentally, do all these electrical dynamics operating in tandem cause the electrical phase shifting that gives most amps a headache? " Let us begin with the phase definition. If a speaker's woofer and tweeter were out of phase more than 'a bit', they would show a dip or even a complete suckout in frequency response, at or near their crossover point, with the microphone placed where your ear would be. As we see from Stereophile's tests, most speakers do not have this issue. So all of those must be "in phase", "phase coherent", "phase linear", or "phased aligned" As I explained earlier, that does not mean they are time-coherent speakers. As a reminder, the opposite IS true: time-coherent speakers are always phase coherent. What makes the phase go weird? -- In the speaker cabinet, it is from the drivers' locations/no stepped baffle, and having too many drivers per frequency range. -- Any crossover circuit's inductors and capacitors delay the signal or advance it, respectively. Resistors do neither. A simple first-order crossover circuit has an inductor going to the woofer, and a capacitor on the way to its tweeter. At their crossover point AND ALL other frequencies, the time-delay created by the woofer's inductor is precisely offset by the time-advance created by the tweeter's capacitor. This is not possible with higher-order crossovers, because the values of their more-numerous inductors and capacitors cannot offset each other. -- The back-emf from any driver is also a contributor to time-delay in its lower-range, whether woofer or tweeter. Thank you for pointing this out. I should have mentioned this earlier. That back-emf situation is altered by the type and size of the cabinet behind a woofer, and the size of any rear-chamber on a tweeter, and from ferrofluid in its magnet gap. -- Any cone or dome breakups change the arrival-time as we go up the scale, but mostly we would hear ringing, sibilance, maybe 'dirt' being added to the music. Regardless, the best cones will not show a loud ringing at some frequency (as with most metal cones available in 2014) nor have a ragged frequency response in their upper ranges. -- And yes, all these phase shifts will talk back to the amp. However, the crossover circuit's design is the primary cause of large swings in a speaker's impedance curve, above 100Hz. Those variations are 'electrical phase shifts' only. These swings in impedance do not reflect the acoustic phase at one's ear- no direct correlation. The amp gets a headache because large swings in impedance means its output voltage (the pressure it puts on its electrons) is no longer sync'd up with WHEN those electrons are allowed to move by the crossover parts (inductors and capacitors). When the values of those caps and inductors do not offset each other, the result is exactly like pushing a child on a swing at the WRONG time. === Bifwynne, on the first page, you speculated on the effects of mics, of recording and mastering, processing, playback, etc. Each of those areas has unique problems, which do not sound like phase shift from a speaker. Each process produces a time delay in the highs and sometimes the lows, but only a speaker can put phase shifts (plural) across the main tone range. Also, whatever that signal is, I see no reason for home- and studio-speaker designs to distort it more. On that same page you asked "How are small speaker manufacturers able to design speakers without the benefit of the R&D budget, engineers, and testing facilities that some of the larger manufacturers have at their disposal?" For me, it's been knowledge, education, and longer, much wider experience. My talent seems to have been expressed as an ability to make the cognitive leap between seemingly unrelated factors, which then made one more link to hearing vs. measurement. All of this has led to me not needing an anechoic chamber (I can always go outdoors for that). I also found the fancy digital test gear gave misleading and often incorrect numbers, compared to analog test gear. When a designer does not really understand the fundamental physics of how and why drivers move and respond as they do, nor how crossovers delay the signals, then their only recourse TO IMPRESS their board of directors, is the anechoic chamber/digital route, for that is what the AES and any university would also advise those board members responsible for hiring 'a great designer'. Such a designer then blames the sonic differences between his and other speakers as 'we all hear differently'. His board of directors and all reviewers and editors gladly go along with that bullcrap. We all certainly listen for different things. But here we have found, that as a speaker is made more and more time-coherent, everyone AGREES on the sounds heard in each and every tome range. They all hear 'the bass' in the same way, etc. === Ohlala, on page one, the possibilities of off-axis cancellations you mention turn out to be non-issues on music, especially when the cabinet is not large, and has little sonic reflections from its surface. === Timlub, on page one, your speaker design is only phase coherent at its crossover point, not time coherent, as you may know. Your electrical crossover slopes work well because they are combining with the phase shifts of your particular woofer and tweeter, which I am sure you suspect. Thank you for sharing your experiences! Appreciated. === Bifwynne, you ask too many (good) questions! On page one you ask, "here the ultimate Q. How can one tell whether a speaker is time and phase coherent? Critical listening? Reviewer comments? Bench test? If critical listening is that important, the real challenge for us is, as many have written, that it is not easy to meaningfully audition speakers. So what's a person to do? I'll ask again, how important is time and phase coherence? FWIW, ... really more as an FYI, ... Paradigm's web site states that its 'speakers have phase coherent crossovers designed so that the summed output of the drivers is completely and accurately rejoined.' Is that hype? It is true at all frequencies?" On my website, I have suggestions on how to audition speakers. I know these work. They are simple, taking only time and effort. The time-coherence part of the audition sounds like clarity and depth, and when time-coherent speakers are designed with the best parts, the musicality is greatly improved. With the very best, you find yourself never, ever thinking about 'the sound of the bass' or 'the highs'. Instead, you subconsciously always focus on the music and how it is being played, and its emotional and physical connection to you. When a speaker is time-INcoherent, the music is fragmented, leaving you to hear only 'the details' and 'the soundstage' or 'the air', or 'the impact'. Right now, I see only Green Mountain Audio, certain models from Thiel, and Vandersteeen as making time-coherent speakers. The Audio Machina company is part-way there. With any others claiming time-coherence, I've seen no proof on their websites, or in Stereophile tests. === Ivan_nosnibor, I appreciated your thoughts, thanks. However, the time delays in your digital crossover circuits are fixed time delays for each driver, when the real problem is the amount of time delays are different at each frequency. You remark on hearing perhaps the highs 'imaging closer to you' on non-time-coherenet speakers, with the mids 'not projecting as far into the room', and so on. I have found instead it is about the lack of depth in the highs, caused by the smearing of a late-arriving mid, and so on down the musical scale. WHEN the highs arrive is not WHEN you hear the image, but only a portion of that image. One example is hearing the esses and tees of the singer's voice arrive from the tweeter's location above the mid, not from the mid driver's location, where the main part of her voice comes from, listening with eyes closed. That is one sound of a tweeter arriving too soon. It can also sound like the band is leaning forward, for want of a better word. It can sound like the rhythm section is behind the beat (as they would be in those speakers). === Almarg, Your described a square wave as "the summation of an infinite number of sine waves, one being at its 'fundamental frequency"' (the frequency with which its pulses repeat), plus others at every odd multiple of that frequency (i.e., the 3rd, 5th, 7th, etc. harmonics). The amplitude of each harmonic decreasing as its order (i.e., its frequency) increases." This is all true, but only of an ongoing series of square waves. The analysis is somewhat different when we examine just the first up-cycle, without even the first down-cycle following it. Just an FYI, seemingly never mentioned on the internet nor in textbooks. === Mofimadness, the Loudspeaker Design Cookbook is generally excellent, but all previous issues got the concepts of phase time-coherence somewhat wrong. It has been awhile since I looked over a copy, so I can't remember where the problems showed up. I advise to take its advice with a modicum of salt. === Bfwynne, the Revel 2 and Magico have oodles of phase shift, mostly from their crossovers. What you are seeing in the Stereophile tests is just as John Atkinson says- the mid and woofer take longer for their sounds to arrive. What is not readily apparent is how the phase (time delay) is changing at EVERY frequency. Otherwise, one could fix the Magico and Revel 'problems' by moving their tweeters back, etc. Actually, Almarg gave you a very excellent answer. === Usermanual, you ask about us proving we are time-coherent. 1) This would not change our sales. 2) It cannot be done in a singular graph or 'scope image useful to a layman, by anyone including us. This is not a case of sour grapes- please read my letter to sixmoons regarding the issues with measurements. Note some of my graphs do not line up correctly with my text on their website. In the 1994 Stereophile test on our Diamante three-way, remember JA always measures at 50 inches, right in front of a speaker's tweeter. That makes ANYONE'S mid and woofer too far away, relative to the tweeter. JA then moved his mic straight down, to get farther from our tweeter, closer to the mid and woofer, looking for our claim of time coherence. You see our step response get sharper, more compact. But our frequency response goes to heck because he is now going VERY far off-axis of both mid and tweeter. Again, this test was done in 1994. In the intervening twenty years, every aspect of our sound, and of any measured performance, has improved. Above all, trust your ears more than measurements and reviewers. My letter to sixmoons shows why this has to be so. === This covers page one, I think. Perhaps page two will be much, much shorter. Best, roy |
Thanks Roy. I think I have the instinct, but not the brains for the math and physics. B'li neder (not a vow), I will teach myself the math and physics when I retire as Caeser's tax collector.
So ... I'm all set up. Nice electronics, good music ... listening to some hi-rez redbook CDs right now, ... good looking wife and ok speakers. Why not just go for the DEQX and call it a day?
Btw, just anecdotally and IMO, I think my Paradigm S8s (v3) are made of decent kit: beryllium tweets, aluminum-cobalt alloy mids, good woofies, tweets and mids are ferro-fluid cooled and damped, super neodymium magnets in the tweets and mids, 20,000 gauss magnetic flux density in the tweets and 15K gauss magnetic flux density in mids and woofies. What am I missing except time coherence?
Why not DEQX? |
Why not DEQX?
Bifwynne
Bifwynne,Roy has already answered your question some time back on this page 4. Did you miss reading it?? Here's a cut & paste from his 7/16/14 post: Bifwynne,
DEQX seems fine in theory, and certainly makes a positive difference. For me, it has serious limitations because it cannot measure exactly what needs to be corrected. This leads to results that depend on the music being played and sometimes a limitation in one's seating position.
In particular, DEQX cannot see the immediate reflections from the cabinet surface surrounding the tweeter. It cannot correct properly for anything happening below middle C because of floor-bounce effects on the microphone are not the same as they are to our ears on music.
There are other issues, but to me, those are the two largest ones. I find that a much higher level of coherence is achievable passively. here's more info from Roy on DEQX in his 7/17/14 psot to the whole group This varying time-delay is what DEQX-type components are trying to correct, and what regular digital crossover circuits never attempt to correct (offering only fixed time delays, such as one millisecond). To correct the varying time delay, a heck of a computer is required, hence the high cost of DEQX type of gear.
Measurement issues and limitations still confuse DEQX type of gear, for two reasons- we cannot (yet) program that computer to how we actually hear on music, and that a measurement microphone cannot resolve the (countless) reflections off the front of a cabinet. If I had spent money on a DEQX, I would first place an "F-11" pure wool felt all around the tweeter, and then run the calibration routine.
Best,
Roy more info on why DEQX has limitations from Roy's 7/19/14 post: Putting the measuring mic for DEQX up close to a speaker is pointless (except for fixing up a subwoofer), as what the mic would then be hearing is coming from drivers at much different path-length-differences to the mic compared to the path-lengths to an ear ten feet away. We all know how walking up to a speaker changes everything we hear. Perhaps they are suggesting this for fixing one driver at a time. That has problems too, because any driver's tone balance is different at ten feet away vs. ten inches away. Bifwynne, this is plenty of info for you to understand why DEQX has limitations & is not a panacea for time-INcoherent speakers. Don't you think? |
Roy, Thanks for your fantastic contribution here. We can only hope for more really knowledgeable folks like you to take the time and educate us hobbyists. On the 1st or 2nd page I posted about a way I was intending to get at this. I have, at least for this purpose, the advantage of having only an optimized computer as audio source. My plan is to use Acourate software on the server, a multichannel DAC, and independent amps connected directly to each driver, without passive crossovers. Acourate allows the use of a variety of digital crossovers, and allows for time alignment of the drivers. BUT it is limited to a single time delay between any pair of drivers, much like the limitations you describe for DEQX (which I previously considered too, but a needlessly expensive option if the only source is a computer). Clearly this will not solve 100% of the problem - something I learnt from you. But what's your very educated guess: will it solve maybe 80% of the problem vs a non-time-aligned 3-way speaker? BTW, would love to get your thoughts about this XO white paper by Dr Uli Bruggemann, the guy behind Acourate. As of now I'm using B&W 804S. Obviously not time-aligned. Probably not even phase-coherent. So the setup described above would first be used with these speakers. And eventually I'm thinking of building my own speakers using top-notch drivers, the Loudspeaker Cookbook as guide. I'm a mechanical engineer and handy building stuff. Assuming I do a good job selecting drivers and building the cabinets...sounds like I'll end up with very good speakers in terms of bang for buck...what do you think? |
Bombaywalla, not trying to be a troll here. I restated my DEQX point because I already have speakers. However, as Lewinskih01 kinda alluded to above, the device may not be perfect ... for all the reasons Roy mentioned, but it may get me to a much better place.
Trying to arrange for a DEQX audition.
Kudos to Roy for his well written posts and dedication to our hobby. |
