Polymer Audio Research Speaker

On May 12th, I attended the demonstration of the Polymer MKS loudspeaker by Polymer Audio Research at a meeting of the San Diego Music and Audio Guild.  I was glad to be able to attend this meeting if for no other reason than it was the first audio society meeting I have attended in over two decades.  At one time, and for about 9 years, I was the President of the San Diego Audio Society.  That was from about 1982 to 1991, and I haven't attended an audio society meeting since.  Not by choice, though arguably by laziness.  I moved to San Antonio Texas in 1992 and there hasn't been an equivalent organization here that I was aware of.  The only purist audio store in San Antonio is that of Galen Carol, who opens his home to prospective audio shoppers by appointment only.  The local Bjorn's dominates audio and video storefront space at the high end, but is primarily oriented to home theater rather than audiophiles.  There was a purist store called Concert Sound from which I bought a Linn turntable in 1998, and the owner expressed interest in having me operate an audio society here, but the store closed sometime not long after I bought the turntable.  Speaking of laziness, even when I was an audio society President (I think I was chosen or chose myself by default as the original organization created by Ike Eisenson was otherwise crumbling) I did very little, mainly just printing announcements for meetings that were mostly arranged by Bruce of Stereo Unlimited, one of the most well known high end audio stores in San Diego.  Even then, I rarely talked to Bruce directly, but instead had details provided to me by my friend and brother-in-law George Louis.  After I left San Diego, my friend George has retained ownership of the San Diego Audio Society name, but the more functioning successor volunteer organization is the aforementioned Guild.  So for me this was a reunion of sorts.

The location was an incredibly beautiful home at an incredibly beautiful spot on the shoreline at the southern edge of La Jolla, having a commanding view of Pacific Beach to the south.  The meeting was held in the living room which was joined with the kitchen and dining area in a unified space surrounded on two sides with large windows showing the surrounding views.  Virtually all the exterior wall space was covered with large super premium unscreened windows with no curtains or blinds.  I would guess the room to be about 40x25 feet or so, nearly as large as my entire house.

As George and I arrived just before the official starting time, the people from Polymer Audio Research were busy trying to get decent sound by slight adjustments in the speaker position, and removing hum apparently apparently picked up or caused by the speaker wire, which was two large separate conductors for each wire.  By the time the meeting started, the hum problem was mostly fixed, but the sound had not been well optimized.  As Roger and I agreed, they probably should have allowed a whole day for proper setup.

While the overall presentation had some good points, and the speaker itself seems to have many virtues, the sound was far from perfect.  It varied a lot by position in the room.  During the setup phase, somebody said that the best listening position was standing behind the sofa, and that was later repeated.  Well that position did seem to solidify the bass a bit, though the bass was irregular everywhere in the room (surprising for such a large room).  But the behind-sofa position on the center axis was one of the worst locations for a pleasant midrange.  From that position, the midrange was hazy, ill focussed, recessed, and slightly edgy.  The midrange was much better sitting on the sofa itself on the center axis, or even leaning forward a foot or two.  One of the key factors seemed to be the listening height with respect to the tweeter axis.  On or below the tweeter axis, the midrange was best, it quickly became unpleasant above the tweeter axis.  And all the listening positions from the center of the sofa on back were too far back for correct imaging, an proper listening triangle would have been about 4 feet in front of the sofa.

The bass had a different set of irregularities at each position, being slightly more forceful overall behind the sofa, but not really better overall.

My take was both a problematic room and a not-quite-perfected speaker.  I suspected the speaker had crossover and baffle related issues that tend to give it high frequency irregularities above the tweeter axis and edginess.  Notably the baffle seems truncated at the top, diffraction might be improved by extending the box (and the inward curvature) up by another 3-9 inches, more of an egg-shaped top.  About the crossover I don't know what needs to be done.  I suspect physical driver time alignment would help, or anything that improves performance above the tweeter axis without compromising it elsewhere, and perhaps adjusting tweeter level lower.  The people from Polymer audio said the crossover was intended to be very steep, but implemented with few parts.  Perhaps they chose the wrong tradeoffs here.  Crossover design is an art, ultimately like violin making.

While making these guesses, let me say that the room was unremittingly hard, with all ceramic, glass, wood, and plaster surfaces, and that huge number of identical windows.  Such a room might be better designed with each window to be slightly different to eliminate shared resonances, acoustic absorbers at key points.  I see the homeowner has a room correction system, which was not used for this demonstration; Polymer audio brought their own playback system including amplification--Linn amplifiers no less or more.  The amps might have contributed a tiny bit of edginess to the sound.  Further, a key factor was that the vaulted ceiling reached a peak in front of all listening areas.  This reflects all manner of delayed sounds to the listeners, smearing the image.  Vaulting running along the axis of the speakers is *much* better because it lacks this problem.  All manner of damping at the peak is possible (but maybe not visually acceptible) and also, what I did(!),  move the listening position in front of the peak.  That also combines with a flattened listening triangle, subtending about 75 degrees from the listener to the speakers, and a 4 foot listening position to 8 foot high speakers...so the speakers (not the room) dominates--that's what I did to mitigate my transverse vaulted ceiling.  Well the owner of this home could do something like that by ditching the coffee table and putting chair much closer to the speakers.  Actually...his speakers were further back (up against the back wall) and farther apart, subtending a better angle and possibly getting some useful bass augmentation.

Polymer Audio Research made the point over and over of using the best parts available in this speaker, and I don't doubt it.  But always, it's always the design that counts foremost.  Polymer Audio Research is a relatively new company, and perhaps they haven't learned the best way to design yet.  I'm not saying I know the best way either, but I know I've made a lot of mistakes, and it took even longer to realize I made mistakes, so at least one principle is...it takes time and experience.

On the plus side, the speaker was clearly undistorted even in this large room, a testament to the driver and enclosure quality, which IMO is worth the price in that you are getting what you are paying for (you are just not getting the performance it should be capable of, IMO, but this was not a fair test either).  In such a large room, it's nearly a miracle for such a compact speaker, with two small woofers, to fill it as well as it did (pretty well, with noted room-related irregularities, and not sub 30 Hz).  I don't think a speaker like this should be used without subwoofer in such a large room, but a test like this shows it should be able to handle smaller rooms with ease, if you can stand (or better, sit) the way it sounds.

What is an Audiophile?

I like the Wikipedia definition, also found elsewhere, an audiophile is someone who is enthusiastic about sound reproduction.

A friend of mine, George, objected to this.  "That's a definition not written by audiophiles...what do they know?"  He had been trying to make the point that by not bringing a set of personal CD's to a friends house recently for an audiophile listening session, I showed that I was not an audiophile, or not like an audiophile, for certainly any true audiophile would have done so.  This was really a rhetorical argument which was part of another argument criticizing a mutual friend Roger for the way he hosted a previous listening session at Roger's house, starting with ten of Roger's own favorite recordings, rather than jumping into to trying other people's favorite recordings first.  George complained bitterly about having to listen to the whole tracks also.*  The next day George strong statements that of course I--Audio Investigator--am an audiophile, and it would have been ridiculous to say otherwise.

(*Actually, the very first track was interrupted to reverse polarity, at George's request, and likewise many other of the first ten tracks were played in both polarities, with as many as three partial track plays, with George asking many questions to all of us about the perceived differences.  So it was ridiculous that George was complaining...he directed the entire listening session nearly from the start, albeit playing Roger's recordings at first.  But George's urge to control is rarely satisfied. I ignore his demons when I can and love him anyway for his energy and charm.  BTW, he also well knows but usually ignores that I do not believe absolute polarity even makes an audible difference in most playback, let alone being a peculiarly important factor worth endless replays to get right, as it seems to him.  Nor do I believe that most quality players differ in their polarity (no CD players that I have tested differ, including a few specific models he still believes different), that most but not all recordings are wrong, etc., and his whole conspiracy theory about polarity obfuscation.  I've written my specific evidence and arguments before, and will get to that corpus soon when discussing George's most recent negative finding double blind test results submitted in 2012.  As a quick overview, I made some double blind tests for him, and a program designed to create more such tests at will, in 2010.  It took him until 2012 to submit his choices in a same/different test made to his specifications with music tracks he specified.  A significant relation between his choices and the actual polarities was not found.  He did get a slight majority of choices correct, but that would have statistically useful meaning only in a large number of similar tests.  It's also in direct contradiction to his questions, before submitting results, about what I would think if he got all answers correct.  George not only claims to know polarity with his own system and chosen recordings, he can hear it outside a building before entering, on a strange system with strange music.  He is questioning renowned makers of CD players, and recording engineers, about errors he hears with no other knowledge.  The prior from his standpoint should be near certain correctness, making every error count against.  So I think the results should be a big setback for perfect polarity punditry, not an incremental advance.  Admittedly, it doesn't do much for us critics, like me, who had the prior of 'he doesn't hear polarity consistently enough to be useful'.  Of course, George is free to try again with same or corrected paradigm. And I will keep this blog updated with results.  I will have to review old emails for the actual numbers on his official polarity test submitted in 2012.  BTW, I think George may be sometimes be hearing issues in system asymmetry, polarity may cause other effects in highly asymmetric systems.  But he has made certain errors, I am sure, in system reports, for example his saying the Oppo CDP-95 is out-of-polarity with classic CD players like the Sony CDP-507.  I have measured both, easily, as having the same polarity.  He fails to recognize those errors, and only quits arguing for time.  I have tried to argue with him many times that unless he is absolutely sure about his criticism of major players and recordings, he should say nothing.  Best not bear false witness.  He insists he is absolutely sure, despite my disagreements, he finds fault with my test signals, or my use of oscilloscope, and renders that his Cricket polarity tester agrees with him in these cases, though he concedes it is not perfect (I thought I once explained exactly why the Cricket makes the errors that it does).  I pity those equipment producers, recording engineers, and reviewers, who are subjected to this polarity folly.  But it is hardly the worst in the world, and one can hardly expect all audiophiles not to be cranks--in fact, if ever there were a world of cranks, it would be audiophiles..  George is just the clown who does it more perfectly than anyone else, shamelessly, and he claims to do both, but with all the science, there seems (to outsiders but extended family members like me) no time not for advancing the science, so business, apparently, comes first, enjoyment only shared with work.  A recipe for dullness.  Brilliance in defending his ideas, dullness in remembering your response to his repeated question, or from withdrawing from your unease.)

After all, George's argument continued, how can you properly judge an audio system without playing CD's you are familiar with, he continued in an argument that lasted for quite a while.  I conceded that using personal CD's for comparison might be a good idea for doing comparisons, but there is no rule that audiophiles even need do such comparisons at all.  They might simply use equipment deemed by themselves or others to be sufficiently good for enthusiasm, and then enjoy how well such systems reproduce music.  Further, even if audiophiles do comparisons, there is no requirement that they always do so with personal reference recordings.  Audiophiles can and do in many cases make comparisons to the absolute sound, to how similar any given reproduction compares with live music they have heard at some time, or the sounds of particular instruments, etc., and not necessarily limited to the relative qualities of a particular recording.  Or they can simply react as pure subjectivity, how the reproduction made them feel, with no assertion regarding it's accuracy or general tendency to do so.

I like the Wikipedia definition because it gets right to the point and doesn't take sides.

Many audiophiles are constantly taking sides, and the world of audio is full of different schools of thought.  In my opinion most of this is pure waste, the result of endless commercialism among other things, and especially the commercialism regarding tweaks (cables, for example) that rarely make an audible difference IMO.  So each tweak maker, or critic of such tweaks, invents a new school of audio thought to explain and justify their ways, since they may relate in contradictory ways to commonly accepted audio engineering principles.  BTW, one of the best studies on the audibility of absolute polarity published in the Journal of the Audio Engineering Society by renowned engineers concluded that absolute polarity is not generally audible on loudspeakers with complex music, even though it can easily be discerned with certain test signals on headphones.

And it was ironic that George was making this argument at all, given the long history we've had of related arguments.  Way back long ago, maybe 30 years or so,  George made an argument that recording engineers who have pre-knowledge of what a recorded event sounded like at the time have no special knowledge beyond those of a discerning listenter, like himself, in judging a playback system.    After all, George argued, it was not the satisfaction of the recording engineer which was desired, but his  own, from the playback using his own system.  Therefore, only he, George, could be the judge, and he would be the judge using his favorite recordings.

More speaker fine tuning

Some of the response curves showed a bit of peaking above 6khz, that had me concerned that I was too much on-axis with the acoustats.  One needs to be slightly off axis for the best sound, this can be tuned best by ear, but measurement sometimes helps too.  I have felt quite often in the past few months that my close-up position was too much on-axis, having a slightly peaky sound.

So I moved both speakers slightly straighter, by first angling out the super tweeters about a half inch, then moving the speakers to match.

After doing this I also measured the speaker distances to the nose-position microphone.  I was very surprised to find how short the distances are now, about 40 inches to the center of the panels.  But the left side was clearly shorter, I first measured 36 inches but 38 seemed more accurate, whereas the right side was at 40 inches.

So I moved the left speaker back as much as I could, which wasn't much, hardly an inch.  When doing the full system response curves, there was still a slight gap between the impulse in the right and left channels, the left channel still seemed to measure about 0.03 ms faster as if it were about 6mm closer.  BTW, that is a fraction of an inch, about 1/4 inch.

I decided to fudge this one with the Behringer DEQ.  I dialed in short delay of 0.03 in the left panel and  added 0.03 to the existing delay for the left tweeter also, figuring it might be more delayed now.  It seemed that after I set the "unlink" option globally, I could set short delays separately for each channel. I had not figured that out before.

This yielded a measurement in which the impulses for the two channels exactly lined up.  But in listening, it seemed slightly skewed to the right.  So actually I reversed the DEQ setting, dialing in short delay for the right panels only of 0.03 ms, and undoing the delays I had previously added on the left.  That sounded correct.  I can't explain why, perhaps I don't have the microphone positioned correctly or my head is slightly asymmetrical.  In principle I could have moved the panels.  I chose not to do measurements after making this final tweak by ear, since the measurements seemed to steer me wrong on this one.

Another change I made was to reduce the bass levels on both sides.  The bass has been sounding a bit boomy, and the measurements showed  a rise in bass below 80 Hz and the left channel rising above the right in the bass below about 40 Hz, so I took about 1.5 dB off the right sub and 3.0 dB off the left sub.  That seemed to make both the measurements and the sound better, but I subsequently noticed that the Tact measurements are inconsistent, even averaging 40 trials, sometimes the right channel exceeds the level of bass in the left channel all the way down to 20 Hz, and other times left channel has more below 50 Hz.  So some additional relative subwoofer adjusting may be required, about a dB or two, but the current adjustment is an improvement in reducing the boominess if not level matching.

Tact good for tweeter alignment

While I have concluded that the low frequency resolution of the Tact 2.0 measurement program (or more precisely, the display of the measurement program, but it might be the program itself, because of the type of pulse it uses and the number of bins) doesn't work well for subwoofer time alignment.

But it works very well for supertweeter time alignment.  A tone burst (which may be partly digital artifacts) appears in the supertweeter channel.  And using two channels is fine, I was wrong about the delay being auto-adjusted to make the leading edge of the tweeter signal line up with the leading edge of the Acoustat signal.

After the first measurement, it appeared that the supertweeter was lagging by about 0.16 ms.  So I adjusted that exactly (for some reason, I used the "short time delay" menu in the Behringer) in the right channel and got this picture, where white is the tweeter and yellow-orange is the Acoustats:

Yes, for some reason the leading edge of the acoustat signal appears to go down, out-of-polarity.  But I believe the main part of the pulse is what follows, and it goes up.  I still don't understand this.  But looking at the above picture you see that I have lined up the leading downward pulse from the panels with the leading edge of the squiggly burst from the super tweeter, which might go up (depending on which pixel you look at, some of the leading edge of the tweeter signal looks like digital artifact pre-ringing which can be ignored.  So I ignored tiny pixels, but chose the first decent looking line as the leading edge, which I admit is a judgement call.)  There is a bit of ambiguity here as to where the tweeter signal really starts, but we are within a few 0.01 ms here.  When I started, the tweeter burst was half way further down the screen, and that was a mere 0.16 ms difference.  For absolute perfection here, a better measuring device and/or listening may be required.

After doing the above measurement, I realized that the two supertweeters were not correspondingly positioned for the two panels.  I would have to adjust the other one to match this one.  But then it also occurred to me it might be better to push out both supertweeters all the way, so that the front edge of the stands for supertweeters line up with the stands of the Acoustats they are next to.  There is nothing magic about getting the two stands to line up, but it is a more reproducible positioning than most others (except having them line up on the back side) which is helpful for practical reasons.  I often move one or both supertweeters out of the living room for  parties.  If I have them calibrated for any particular position, it helps to make that position an easily reproducible one.

And since I am adjusting the delay anyway, I don't have to time align the positions of the two speakers for the reason people not having digital systems must.  I can choose to optimize the relative position of the panels and supertweeters for other reasons than actual time alignment.  In addition to the reproduciblity issue described in previous paragraph, there are also issues related to dispersion and diffraction.

Basically you don't want the supertweeter firing from behind other speakers, inside a hole as it were, because it's like talking through cupped hands.  If the tweeters are actually slightly forward of the Acoustats, that is helpful in reducing edge diffraction related to the sound projected by the supertweeters.  On the other hand, it could increase edge diffraction related to sound eminated from the Acoustats.  But that doesn't matter as much here for several reasons, the most important being that as a figure 8 speaker the acoustats don't signficantly project sound to the hard right or left, that's a null.  For another, the supertweeters are omnidirectional, which is exactly the opposite, they will product loads of diffraction and other undesirable addition effects when there are nearby boundaries.  Plus, one takes advantage of their omnidirectionality if they are slightly forward of the other speakers, getting more that 180 degrees of free radiating angle directed toward the user rather than in the other direction.

So I moved both the supertweeters out like that, and calibrated both channels like the above for correct time alignment achieved by digital delay.

I listened a bit to radio, KRTU because KPAC was playing opera, and the new setup is wonderful.  Somehow it is both more transparent, more spacious, and more relaxed.  I can also move my had a bit either way without the image seriously distorting, instead the image shifts gradually as I move my head.  This is all to the good, and I think having the supertweeters moved out and time aligned digitally is a big improvement, and a successful day's work.

Using the Tact to adjust crossover time delays

At long last I'm firing up the living room Tact 2.0 program (which runs on a Windows PC) to adjust the time delay that time aligns my Acoustat panels and SVS subwoofers.  I am looking at frequency response too.

Unfortunately for me the Tact program is oriented around room correction (not really measurement) so much that it's unclear to me whether the impulse shown after measuring my system is a full cycle (0, +1, -1, 0) because the tact is simply showing what it recorded, an image of its own test signal, or whether it shows a full cycle because of group delay in my Acoustat speaker, and if the Acoustat had no group delay the Tact would be showing a standard impulse that goes from 0 to +1 and then back to 0.  When I use the Liberty Audio Suite, it plays a signal that looks nothing like a simple pulse, but LAS then mathematically transforms it into a perfect step pulse graph if indeed the unit being tested has perfect pulse response (as most solid state electronics does, or comes close to, while speakers are usually far from it). I think that's called deconvolution, but it might be called convolution (I get the two confused).

Worse, I had a specific idea for adjusting the time delay between the Acoustat panels and the subs.  I reversed the amplifier connections for the acoustats.  Then I mute every speaker driver except the Left channel Acoustat (which is actually playing on the right, because I reversed them) and the right channel sub.  Then when I run Tact in full stereo mode, it plays both drivers for the right channel but as if they were left and right full range speakers.  Then, in principle, I could verify and adjust time delay, by comparing the onset of left and right channels in the impulse picture the Tact produces.

However, because the Tact is so oriented around measurement, it uses some trigger to put both channels into alignment even if they are not.  Or at least that is what it seems.  Whenever I do this measurement, the bass looks like it starts about 6.0 ms (millisecond) later than the panels.  Which is very strange because the speakers are only about 3 feet apart, and I was already applying about 3.4 ms of delay to the panels to compensate.  So the expected error, or difference, should have been less than 1 ms.  But as this measurment appears, there is also a box showing the relative time difference between the channels (which the Tact is compensating for).  And it shows a really big number, like 20 ms.  So once again I don't know what the impulse picture means.  Is it showing the raw measurement before the 20 msec correction it is ultimately applying in the chosen correction number?  Or is it showing the two channels adjusted, as it thinks they should be?  If it is showing them after correction, I can't use this measurement.

Because the Tact seemed to be showing about a 6ms gap with the subs starting later, I increased the time delay on the panels, but for some reason I don't now remember, or possibly just lapse of memory, I increased the delay adjustement (in the Behringer 2496 DEQ) for the Acoustats from 2.47ms to 5.53ms.  I would have made the second 5.47 but the adjustment is course and that exact number is not available.  Anyway, that seemed to have little effect on the graph.

So then I tried bigger adjustments, 9ms, 12ms, 10ms, and in every case it wasn't clear whether it made the bass alignment with the treble impulse better or worse.  Those numbers made no sense at all, the required delay should be between 1 and 4 ms.  But in every case it still looked like 8-12 ms of additional adjustment was needed.

One problem is that the bass may start more slowly, for various reasons, especially the crossover, but also it's limited high frequency response.  It should, I believe, begin moving the instant current is applied, but at first slowly, then building up to a full wave.  The resolution of the display is low enough, and there is also noise, so it is ambiguous where the subwoofer output really begins.

After messing around with the primary "Measurments and Correction" page for hours, the only one that actually allows you to do measurments (and you must have a non-bypass correction number selected, it won't let you do measurments in bypass mode) I finally went over to the dual-domain page, where you can load the previously made measurement, and there it seemed I could get a pretty clear image of the treble impulse, and see that it started around 11.8 ms in the right channel.  (This was with all crossover settings restored to original settings.)  Then I played the bass.  Well, it could have been correct, you could see some rising, maybe, in the bass response at 11.8 ms, where the red line is in the graph below.  But it could also be as fast as 2ms, or as slow as 15 ms, it's hard to be sure, because there is noise and the initial start might be slow, very slow.  True, at about 15 ms it really starts going, it's clearly going at that point, though not as strong as later, there's a cycle of reduced output before the full response builds, it's typical for speakers, especially bass speakers being crossed over, to respond that way.  I have to think that on a higher resolution plot, the very initial slow part, corresponding to the limited high frequency response of the subwoofer, there would be a more clearly visible starting point, and it would be very close to the red line at 11.8 ms, because I was already compensating for the delay with digitial delay in the crossover which should have been accurate to a few inches, which would correspond to about 0.1ms or so, because sound travels 1foot in about 1 ms.  On the other hand, frustratingly, I can't be sure, because it may well be that even with the crossover and all filters in the SVS subwoofer turned off (as usual, except the sub 15 Hz filter I am required to run with only one port filled) there is time delay in the electronics, that's different between the sub amp and the acoustat amp.  I can't explain that much difference in delay happening from analog elctronic processing.  I can explain the 11.8 ms delay as about 6 ms from distance to the microphone, and about 6 ms fixed in the digital processing from the output of the Tact through the Behringer DEQ.  But that should not vary between the treble and the bass.

Anyway, as the graph below shows, the Tact measurement system is useless for setting the delay on the bass, because the starting point of the bass is ambiguous with such limited resolution.  And the graph below was made with crossover turned off, and the bass does have HF response to about 300 Hz, so I think a good measurement system would clearly show the starting point better than this.  Part of the problem here...the Tact has a simple impulse that has limited low frequency information, hence, limited low frequency resolution.  A Maximum Length Sequence system, like liberty audio suite, uses chirps, which have a better spectral distribution than clicks, allowing greater bass resolution.

New System Picture

Tuner Switcheroo

Kenwood L-1000T 2nd from bottom

On the night of Friday February 1st it was becoming clear that I was hearing distortion from the Kenwood L-1000T quite frequently, not just on certain bass heavy programs like Pipe Dreams.  It still does seem that usually the distortion, which has a garbled clipping sound that persists for a few seconds, is usually triggered by audible low bass.  But sometimes it occurrs even when there is no audible bass.  I reason that there might be subsonic bass that is triggering it then, but this is all guessing. Guessing further, I suspect this is being caused by a power supply droop or decoupling capacitor failure.

On February 2nd I did some careful testing to confirm that the distortion is coming from the Kenwood itself, and not my incredibly complicated setup, especially for whole-house listening through Sonos--I actually do most of my listening on kitchen system which has Sonos box. I send L-1000T variable output to a Lavry AD10 which converts to 24/96 digital, then to a Behringer DEQ 2496 for EQ correction, then in analog to Sonos which digitizes again for whole-house distribution.  When I'm listening in the living room, I take the digital output from the Behringer straight into my Tact preamp so no analog re-conversion is necessary.  To bypass much of this, I took variable (and then fixed) output from the Kenwood into the Lavry, then to the Tact, doing the corrective EQ in my DCX 2496 crossover.  So I never did bypass the Lavry, which I believe is working fine.  Analog sources has to be converted to digital somewhere for my living room system to work.  I trust the Lavry, though in a quick online search on Saturday night I was unable to find the input impedance of the Lavry and I had been worried it might be too low, something like 600 ohms, though I believe it is more likely to be 10k ohms or higher.

At first, it seemed like the distortion was not occurring on the Living room system with Berhinger DEQ bypassed.   Strangely, I first noticed the distortion when I was listening to the living room system while in the hallway.  The living room system is so transparent, it actually minimizes certain kinds of distortion.  When I did finally hear the distortion, it seemed briefer and relatively softer.  I sat down and listened to L-1000T in the living room for about an hour, by which time the intermittent distortion was unambiguous.

So I removed the Kenwood from my left hand stack, stashing it in the master bedroom for later investigation and repair.  Meanwhile, my motto is that the symphony must go on, so I brought out spare replacement tuners.  I first hooked up the Kenwood KT-6040, a tuner I thought was one of the best a couple years ago.

Switching from L-1000T, the sonic inferiority of the KT-6040 was all too apparent.  I was so disappointed at first I toyed with bringing back the L-1000T and just ignoring the distortion.  But that distortion gets on your nerves after awhile, you find yourself cringing when you expect it, and the L-1000T needs repair before it gets any worse.

The KT-6040 is equally quiet, but lacks the depth and spatial and harmonic realism the L-1000T has.  It sounds as if the 3D reality of the L-1000T is chopped up into tinfoil panels, one for each musical instrument.  The tinfoil panels are placed in front or back of each other, giving some kind of depth, but nothing like reality.  At the same time there is a cotton-ball like fuzziness to everything, even though there is no lack of highs, maybe even slight excess, but the worst is that the highs seem slightly disconnected from their fundamentals.

I was so disappointed with the KT-6040 sound I brought out the Pioneer F-26.  I'm not sure my F-26 is operating correctly because the "Wide" light never lights up for any of my favorite stations, if at all.  But nevertheless, the F-26 had a far more listenable presentation, a greater sense of coherency and consistency.  The F-26 is also rolled off a tad and perhaps slightly less transparent than the KT-6040, but far more pleasant to listen to, especially in the ultra transparent living room system.

KT-6040 2nd from bottom, F-26 3rd

So I decided on a hybrid hookup, with F-26 hooked through Lavry to play only in the living room, and KT-6040 hooked up through Behringer (using the Beringer A/D conversion also) to play in other rooms through Sonos.  The 6040 sounds OK through Sonos, maybe even better than direct because the highs are slightly tamed.

When I bought the KT-6040, and even the L-1000T I figured I would have a dual tuner setup like this. I had not figured on the L-1000T trouncing my F-26 sonically and therefore being able to play both roles.

Because the remote I have been using with the L-1000T is actually a KT-6040 remote, I was actually able to control more functionality on the KT-6040 than with the L-1000T.  For the first time, I was able to do tuning scans to find new stations.  I spent some time doing that on Saturday night into Sunday morning, checking out some new rock stations.

[Note: the ISOMAX transformer and Classe CP-35 preamp in the left stack are used with my Kurzweil K2661 keyboard.]