07-17-14: Sounds_real_audiotilting 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. '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?? |
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? |
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 |
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 |
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.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 itas the impedance dropsit 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 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:
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). |
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, 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? |
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? |
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?? |
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 |
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. |
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??? |
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 |
I think a call to the US Distributor may be in order. Lew... I agree that the DEQX is likely NOT plug and play. From what I picked up from the DEQX web site, one can pay extra for a remote professional set up. To me ... that is just part of the cost. Honestly, I am a frustrated scientist. This stuff is very interesting to me. Problem is my IQ isn't high enough to get the math and science. My math skills are just a little north of the "Jethro Bodeine double-knot head cyphering" level. Here's a guess ... I surmise that if the DEQX's hype is fairly stated, it may do more for my rig than stepping up to $25+ Magicos S speakers. There is nothing wrong with the drivers in my Paradigm S8s. We're talking about a low distortion beryllium tweeter, an aluminum/cobalt mid and polypropylene/mineral infused woofers. I have a sub to augment bass roll-off. The basics are all there. I'll report back. Cheers. P.S. I have a better chance of sneaking the DEQX into my house than new speakers. That factor alone weighs heavily in favor of the DEQX. :) Al -- you are always the voice of reason. What are your thoughts? Could this be transformational or is that un likely? |
There is also some chatter on DEQX on computeraudiophile.com. Actually, I've been looking into it as well, along the lines of my posts above (waay above). The DEQX HDP-4, the most expensive unit, has good DAC inside, 3-way crossovers, and digital volume control. And room/driver correction. What I researched was for use instead of my DAC and preamp, so getting the signal from my computer server and running 3 amps per side to drive speaker drivers directly, avoiding passive crossovers. I found a guy in Texas with very nice and expensive system, such as YG speakers, say he replaced a $30k DAC with it, so the DAC section must be good. He's using it in the same fashion I'm interested, so can't speak to the ADC section - but that has been clarified by Psag above. He's only caveat was the unit only allowed for one subwoofer out and he's using two in mono but needed different time delays on each to address a room mode so he uses a Xilica unit for that. BTW, he also uses Dirac room correction software on his server. That piece I don't fully understand why as the HDP-4 does room correction too... The HDP-4 is very interesting to me. Where I start wondering if it is the best path is for people like me who only use a computer as server. In that case you could use a Lynx Hilo plus Acourate software to achieve the same, but for $3k instead of 5, or use an exaSound e28 plus Acourate for $4k and have 8 channels available (hi/mid/bass for L&R, plus 2 subs) and be able to time/phase align all of them. But I realize the majority of users here aren't running servers as their main source. Yet the HDP-4 can take 2 analogue inputs and several digital inputs, so you could still connect a phono section and a CD transport. BTW, Psag, how good is the HDP-4 volume control? When bypassing your preamp, what changes do you notice? |
Short of outboard active crossovers, I'm starting to get the sense that mechanical (e.g., sloped baffles) and electrical (i.e., 1st order X-overs),....cool, I like that! :-) I believe that the user community should demand more time-coherent speakers from the various manufs. Many people pay a pretty penny (incl you if you go the Magico-S route) for their resp. speakers & I really doubt that anybody wants to listen to added (speaker) distortion after having paid so much.... let us know what you discover about DEQX - like Ngjockey indicated, I doubt that DEQX is simply plug-n-play. I think that the user will have to know something about the physics behind the usage scenario (freq response, phase response, x-over slopes, phase coherency at x-over freq, amplitude of freq response at higher freq, etc) to bring out the best in DEQX. This is my guess. Meanwhile we will await word from you on this subject. Might want to start a new thread. Of course, do not forget to search the Audiogon & Audioasylum archives for existing chatter on DEQX - might give you a jump-start. Thanks. |
Al, You are welcome. The U.S. distributor, based in Colorado, knows the DEQX unit inside and out, so service is absolutely not an issue. Bifwynne, the input impedance is 30-40 K and the output impedance is about 100 ohm on each signal line. My line stage is also ARC (Reference 10). I have paired subwoofers in my system that receive signal from the DEQX. The only output from the ARC is directly to the DEQX. The DEQX handles everything after that. Every once in a while I take the DEXQ out of the system to reassure myself that its truly transparent when in bypass mode, most recently last week. To my ears, despite the extra A/D and D/A conversion (analog source) it is transparent. |
Thanks Al ... caught that stat after I read your post. Didn't do the math, but I think that the DEQX would likely present an ok input impedance for my linestage even with the impedance buffer (330K ohms). Something north of 30K ohms if my "Jethro Bodeine double knot head cyphering" is right. Still very fuzzy about this whole phase coherence conundrum. And even if its real, whether inserting the DEQX device in my signal path will hurt more than it helps. The real problem is that there are so few B&M stores around, especially those that carry the gear in which I am interested, its hard to do serious listening and make rational decisions. Maybe an audio show?? I hate this hobby. |
Holm Acoustics also offers a similar product but it's more expensive. Without the preamp feature, there's miniDSP, Ground Sound modules and what I consider the best of the 'pro' units, Xilica, which is also used on the top-of-the-line Legacy speakers. Almost forgot the McIntosh MEN220. You still gotta know what you're doin', so they're really no easier to use than designing a passive system. |
Psag ... so you say the device is inserted in between the pre amp/linestage and amp?? Well here's a little quandary I may have. My ARC REf 5 SE linestage has to see a combined impedance of not less than 20K ohms. As currently configured, the Ref 5 sees 300K ohms off Main 1 - imput imp. of amp and 337K ohms off Main 2 - imput imp. of custom made subwoofer/impedance buffer/channel summing gizmo. The combined impedance is about 157K ohms. Well north of 20K ohms. Any idea what the DEQX input impedance is. As you can see .. kinda important. Thnx |
Yes, the DEQX units do seem very intriguing. And their pricing does seem very fair, as Psag indicated, especially given that they can be had with both preamp and DAC functionality. I believe that the Trinnov Amethyst offers somewhat comparable functionality, but is way more expensive. And I believe that Trinnov's somewhat older ST2 model is significantly more expensive as well. Lyngdorf offers a room correction processor including preamp and DAC, but it appears that its focus is just on correcting room-related frequency response issues. Also, its analog input impedance is only 10K, which would rule it out for use with some tube-based sources. One question I would have about the DEQX products is if the company is set up to make repairs in the USA, given that they are based in Australia. But I'd have to say that it's definitely something I'll be considering when I next feel motivated to make a major change to my system, although that won't be particularly soon. One reason it might be especially applicable in my case being that my room treatment options are very limited, since the system is in my living room. Psag, thanks for calling DEQX to our attention. Best regards, -- Al |
To answer Al's rephrased question or Bruces's origional, I'll give the example of the speakers I'm working with lately. The midwoofer is an inexpensive 5" that has a fairly nasty and fairly normal breakup around 5KHz and rolls off rapidly after that. The design I'm using it in is basically a flat baffle MTM (+++) with a low crossover point of about 1500 Hz. BSC is handled separately, but we won't get into that because that gets complex. The low pass is a first order (electrical) with an inductor. By itself, no crossover, the woofer shows 30 degrees of phase shift by 1500 Hz even though it is nowhere close to rolling off. With the crossover, it's 120 degrees at the same point and nearly 180 degrees by 3000 Hz. It's the combined acoustic slope that matters and that's measured in Hz and dB. Phase is along for the ride. Things get a bit more complicated. To attenuate the cone breakup 5K, which would still be audible, I added a "tweeked" Zobel. By that, I mean I oversized the cap and undersized the resistor so that it falls somewhere between a filter and impedance compensation. It also comes in handy to get phase dialed in. Tried but couldn't get a notch filter to work well in this case. Got it about 20 dB down. As some of you might have guessed, for a tweeter to cross that low, it has to be particularly rugged and there's only a few I know that capable, particularly with only a second order high pass. Didn't want the crossover that low but that's where the combination wanted to be. It's already 6 dB down by the crossover point, which gets summed back, when drivers are in phase. The tweeter, with crossover, has begun rolling off from around 5000 Hz. Another "trick" was utilized to round the knee. By the tweeter's Fs (resonant frequency) it's down 20 dB. The tweeter's phase shift from 1500 Hz to 20K, before any baffle diffraction and with crossover, is only 60 degrees. Essentially, little to no phase shift without crossover. If you're still paying attention, you might think something's wrong with my math. Shouldn't second order shift 180 degrees? For a high pass, the phase shift is caused by capacitance, not inductance. In order to get the driver's phase aligned, I needed to invert the polarity of the tweeter. By the way, this sims out to a 45 dB reverse null at 2m, so I think it's pretty much on target. Nicer is that it's consistent over a wide vertical and horizontal range and listening distances. Gently sloped plateau on the impedance phase to +30 (inductive) degrees maximum, which by most standards, is quite good. That's a very simple example, even for a two-way. You should see what happens with real woofers. Remember the old spinning plates act, where a guy balanced plates on poles and ran around to keep them going while he added more plates? Now, tie the poles together with strings and springs and rods and hinges and that's speakers. |
Why does so much discussion center around how "perfect" 6db per octave is? There is 90 phase shift at the crossover point and we can add on electrical and mechanical impedance changes in the drivers across their frequency range. So can't we fully expect some pretty significant deviations from "perfect phase" in the performance of any speaker, even with the first order crossover? |
Bifwynne, I'd say that the cost is substantial but fair. The vendor is located in Colorado, and he came to my home in Arizona for the calibration and installation. I don't know if there is a trial program. The DEQX corrects the room acoustics, and it also corrects the speaker, as outlined in my previous post. The unit resides between my conventional preamp and the amps, but it also has full preamp capabilities if needed. A less expensive option deletes the preamp functions. |
"So the question then becomes: Doesn't the presence of that inductive component of the driver impedance (especially in the case of the tweeter) cause a deviation from first order 6 db/octave behavior? And if so, to a degree that may audibly compromise phase and time coherence? And if so, is that or can that be compensated for in other aspects of the speaker's design?" ABSOLUTELY, I had stated earlier that a simple cap seldom produces a 6db slope for that exact reason, no speaker that I've seen shows purely resistive(other than a ribbon). In the crossover, we can only lower impedance of a driver with compensation circuitry. A simple pad in series on a tweeter will raise impedance, but also cause you to need a new crossover. All this is the difference between electrical vs acoustical crossovers, cause & effect. 6, 12, 18 or 24 db per octave crossovers on paper, often end up being larger slopes because of the natural inductance, impedance and capacitance of a driver that must be taken into consideration in the design itself. In my experience without impedance compensation work, a driver with 6 db filters are still TYPICALLY 10 to 20 degrees out of phase... Any speaker that is with + or - 15 degrees of phase for its useable response curves in my mind IS phase coherent. |
07-08-14: BifwynneBruce (Bifwynne) raises a good question, to which I suspect there is a good answer, but I don't know precisely what that answer may be :-) But I'll reformulate what I interpret to be the question, and perhaps one of the others who are participating can address it. Consider a simple two-way speaker having a first order crossover consisting of a capacitor in series with the tweeter, and an inductor in series with the woofer. For each driver that will result in well behaved 6 db/octave rolloff characteristics, which will result in time and phase coherence if other aspects of the design are also supportive, **IF** the impedances of the woofer and tweeter are purely resistive. However I believe Bruce has been alluding to the fact that the impedances of the drivers are not purely resistive. And it would be more accurate (if still somewhat oversimplified) to electrically model them as consisting of a resistor and an inductor in series. So the question then becomes: Doesn't the presence of that inductive component of the driver impedance (especially in the case of the tweeter) cause a deviation from first order 6 db/octave behavior? And if so, to a degree that may audibly compromise phase and time coherence? And if so, is that or can that be compensated for in other aspects of the speaker's design? As I said, I don't know the answers, but those strike me as good questions. BTW, Tim (Timlub), thanks for providing the link to my post about impedance phase angle. Best regards, -- Al |
Psag ... just took a quick peek at the DEQX web site. Very interesting. Problem is that it's not cheap and where is the vendor located? What recourse if it doesn't work well. Also, it obviously entails inserting an artifact into the signal path, presumably between source components (e.g., CDP, DAC and phone pre) and linestage. Oftentimes, not the best thing to do. How does it work if one has an integrated amp with built-in phono section?? Sure wish I could try the device on approval. Cheers, BIF |
Hi Psag, What you are quoting makes absolute sense... If you pull the crossovers and make everything perfectly phase and time aligned along with perfect frequency response, then the only difference is sound between speakers is the materials themselves...ie, how does the box sound, what does a Kevlar cone vs a paper or aluminum cone sound like etc.... So you are hearing first hand, (by correction) how important a flat response along with phase and time alignment can be. |
Bombaywalla, I use the DEQX, and I can tell you its tranformative. Because it 'corrects' the drivers, it has a way of making different speakers sound more similar, which would no doubt be disturbing to some potential users. Also, it makes some recordings sound somewhat different than we are used to hearing them, which is something that also takes some getting used to. |
Thanks for this info Psag. The DSP software definitely considers having phase & time-coherence as an important aspect of its signal processing so as to have cohesive sound. This should tell us something about the importance of time-coherence in speaker design :-) Looks like DSP might be the way in the future... |
Regarding digital signal processing as it relates to these issues, this from the DEQX website: "In addition to frequency-response errors DEQXs biggest strength is restoring phase and time-domain coherence by delaying faster-arriving frequencies until slower-arriving frequencies catch up for a coherent Impulse-response. DEQX even corrects timing delays in frequency groups within the drivers themselves rather than just time-aligning one driver to the next." |
Bombaywalla, sorry for the confusion. I'm referring to a driver's electrical, not mechanical, attributes. Rather than go off on a tangent, if Al catches these last few posts, he might be able to untangle what I'm trying to say. In the meantime, I'll just assume that the only relevant driver attributes that affect phase coherency are the mechanical points Roy discussed in his White Papers. |
07-07-14: BifwynneI am totally confused here Bifwynne!! I don't understand your question - what do you mean by "the impact of the speaker's electrical characteristics on phase coherence"?? I've pointed you to Roy's article that talked about the impact of the electrical x-over on time-coherency. I've pointed you to Roy's article that talked about the impact of the driver construction on time-coherency. The x-over is the electrical part. the driver construction is the mechanical part. These 2 articles should have covered the info you were looking for..... |
Ngjockey ... I concur with your take on the S5 stats and build quality. As far as the X-over is concerned, not sure what you mean by unobtrusive, but I'd be willing to bet we're talking about 2nd and 3rd order X-overs here. Fyi, I traded e mails with Magico tech folks about phase coherency. They freely admitted that the S5 is not phase coherent, but that attribute was a trade off in order to achieve other design objectives. Gotta give these pups a listen! |
Those S5 plots are actually quite smooth. A sign of well matched, quality drivers, a good cabinet and unobtrusive crossovers. That phase dip in the bass is fairly standard and hard to avoid. It's from the woofer, not the xover. Other than the low impedance, it would be a relatively easy load. Of course, there's always ways to make those plots look better but it might not sound any better. |
Thanks again Bombaywalla. I caught the article. It doesn't speak to the impact of the speaker's electrical characteristics on phase coherence. Maybe it's just a non-issue. @Tim ... based on the various posts in this thread, I gather there are not very many conventional speaker brands that are time and phase coherent. Vandersteen, Thiel and GMA come to mind. As to your point about flat impedance and phase angle plots, take a look at the stats on the Magico S5 here: http://www.soundstagenetwork.com/index.php?option=com_content&view=article&id=1043:nrc-measurements-magico-s5-loudspeakers&catid=77:loudspeaker-measurements&Itemid=153 Not saying the S5 is an easy speaker to drive because its impedance plot ranges for the most part between 3 and 4 ohms and its phase angle goes negative in the low bass, but overall pretty flat plots. I think my ARC Ref 150 could drive it ok off its 4 ohm taps, especially since my amp has a pretty muscular power supply - 1040 joules. OTOH, I would not try to drive the S5s with a low power SET amp. :) I find the phase coherence issue to be extremely interesting. It's frustrating because without doing critical listening, it's hard to get one's "ears" around the issue. Thanks Psag for bringing this important issue to our attention. Just not sure what to do with it. :) BIF |
I have never heard Roy's speakers, but I would love to, I've never seen any speaker that was plus or minus just a few degrees throughout its frequency range... It would probably be easier to read a book than try to get all the info from a forum, so much skipped over, so many half truths... very difficult... We have only really discussed baffle step compensation... I have tried several times to electrically time align a woofer rather that the tweeter. It can be done, but throws so many other things out of wack that I've never really been successful. One thing that I would say... it is possible to get a great sounding speaker without very good alignment, but most speakers that "just aren't right" do not have good alignment characteristics.... Any speaker that has very good alignment characteristics, you will find very listenable, it may not be your cup of tea, but it will do most things well. At least that is my experience. Tim |
07-07-14: BifwynneBifwynne, here is some more material for you to read that will address the phase coherency question you had yesterday (7/6/14): http://greenmountainaudio.com/speaker-time-phase-coherence/ this is an article that Roy J wrote for Audio Ideas Guide back in 1997. A small cut & paste from this article: "The causes of phase distortion Time delay is the natural consequence of making something vibrate, whether it's electric fields or material objects. In speakers, only three things can cause time delays: ◾The moving elements (the drivers -- woofers, midranges, tweeters); ◾Their distances through the air to the listener; and ◾The crossover circuit. Let's go over the cause of motion-based time delays first. Different drivers (round, square, flat) have an inherent amount of phase shift, related only to each one's natural resonant frequency. One analogy is a weight hanging from a spring. If you move the other end of the spring up and down very slowly, the spring does not stretch and the weight follows your motion exactly. The phase shift between your applied force and the weight's motion is zero. The moving system is in a 'minimum-phase' mode. If you move more rapidly, the spring starts to stretch and contract -- and the weight no longer follows your driving force. It moves with a different phase. ......" there's a lot more to read but I believe that you should read the section "The Causes of Phase Distortion" to answer your question.... |
Hi Timlub, Ok, no problem. I believe his real question is "do all frequencies move at the same speed"yes, all freq travel at the same speed. But the answer to his question is still "No, they do not arrive at the listener's ear at the same time". I tried to explain that in my post - looks like you missed it? I'll cut & paste here again for your convenience: " the acoustical center of the 18KHz driver would be in front of the acoustical center of the 30Hz driver. Due to this, the 18KHz signal would get a head-start & would reach your ear 1st." if you do nothing to compensate for the fact that the acoustical centers of the 2 drivers are different, the highs arrive earlier. hope that this clarifies. |
