Amplifier Raised on Brass Feet from Mapleshade

Raising my Aragon 8008 BB on a quad set of custom order brass Mapleshade Threaded Thick Carpet Heavyfeet has lowered the operating temperature of the amplifier considerably, and improved the sound slightly.  The feet allow the amplifier to sit directly above the carpet without a wood platform underneath.  The lack of a makeshift wood platform (a repurposed shelf from an audio rack) has also cleaned up the appearance considerably, created more foot room, and for their part the brass feet look very nice too.

I was very surprised from the large drop in heatsink temperature.  In fact, one of the many excuses for not putting these feet on the amplifier sooner (I actually bought them about two years ago) was that I feared that while the amplifier might sit slightly higher above the visible surface of the carpet than it previously sat on the wood shelf, the (plain pile) texture of the carpet would still impede airflow more than the wood, and the last thing I would want for the already too hot Aragon amplifier (which used to idle at about 133 degrees F at the top of the heatsinks) would be for it to run hotter.  Of course the actual feet that I purchased (which required a custom order because of the stud size of the amplifier) were the carpet-ready variety, with a 3/4 slender spike which pushes through the carpeting.  But I still worried that it wasn't enough.  The total foot size is 2 1/4 inch, leaving at least 1 1/2 inch above the carpeting.

As it turned out, not only were my fears unwarranted, I now regret that I did not put these feet on the amplifier at the very first moment I could to keep it cooler.  The heatsink temperature at idle has now lowered to about 117 degrees F, a 16 degree drop in temperature, which might mean a twofold or greater increase in longevity, or technically MTBF (mean time between failure).  For quite awhile, I even ran the amplifier night and day (mainly because it was more convenient, and I figured that it might sound better, but the truth is the idle bias on this amplifier is so high it probably makes little difference, it pops right up to full idle current within a minute of being turned on, and for longevity reasons it's desirable not to keep the amplifier hot so much).

As to the sound, it seems a bit clearer and cleaner, and sounds slightly more relaxed as well.  No minuses whatever.

Since my listening position is now just a couple of feet back from the speaker plane, the extra foot room is appreciated also, and this makes the new listening position much more guest-friendly for taller people than me, which most people are.

Given that I now have (and have had for about two years) a great infrared thermometer to verify before and after temperatures, putting the feet was about as easy as an audio modification can get.  Disconnect amplifier, turn on side, remove old feet, then add new feet.  The old "feet" were three layers of adhesive felt sliders.  The amplifier came with one set of adhesive feet like this, which were only about 1/4 inch thick.  Once I became aware of how hot the amplifier was running,  after a few more months of inaction, I added two more adhesive felt feet on top (or underneath, actually) the existing feet.  All this time I had the amplifier's feet resting on a solid wood board.  Strangely, i don't recall that adding the extra felt feet made any difference in the operating temperature, though I do remember seeing idle temperatures as high as 136 degrees previously, so I'll give the benefit of the doubt to a 3 degree improvement.  (Given changes in ambient temperature, HVAC operation, and so on, all these measured differences should be taken with a grain of salt anyway.  But I measured the 16 degrees of improvement with the Mapleshade feet on the same afternoon/evening, and I've been seeing the 133 degree idle temperatures for a fairly long time), so while I'm uncertain about how much if any change was made by trebling the felt feet, the big brass feet have made a large and clearly measurable different in amplifier temperature.

Although this was fast, I chose to do some extra stuff that other people might not have bothered with.  Rather than simply assuming all was OK after screwing on the feet, I took the small cover off the top of the Aragon to check the wiring on the right side.  Those feet are right underneath the active guts of one channel.  I could see into the area just a bit using a flashlight, but not well enough, so I removed the cover after I'd set the amplifier back down.  (One problem with this is that you can't screw the feet on and have the covers removed at the same time.)  Toward the back of the amplifier there was no interference with the new screw stud.  But in the front there was a bundle of wires running through.  Most of that bundle simply got pushed up by the stud, leaving just one insulated wire running a bit closer to the nearby power transistor than desirable.

I first put a nylon wire tie onto the screw stud itself.  That meant that no wire was directly contacting the screw.  But then it was clear that no matter how I wrapped the wires, the straggler still turned rather close to a power transistor.  So I then put a second wire tie around the entire bundle of wires, just above where the first screw tie wraps around the screw.  The effect of that is to keep the straggler wire in place with the rest of the bundle, AND to lock the whole bundle in place, just slightly above the screw and separated by nylon and air for thermal insulation.  After examining that for awhile, I put the cover back on.  I didn't remove the larger cover on the right side of the amp because there is no circuitry where those screw holes are, or any power transistors, though the screw studs could have pushed a wire out of the way, it would be otherwise harmless.

As it turned out, only one of the feet allowed the stud to screw nearly all the way in.  On the other feet, the screws would turn in leaving about 3/8" above the foot.  For most of the feet, I screwed the stud in first as far as it would go, then I screwed the foot and screw assembly onto the amplifier.  In every case this made it impossible to feel the screw starting position by turning the screw backwards (as I usually do) so I had to simply screw the screws on clockwise as most people do, and it wasn't easy to get the screws lined up.  For the other foot, I screwed the screw in not as far as it would go, but to the same distance as the others.

Had I really examined the amp first, I might have used the special foot (which allows the screw to go nearly all the way in) just at that location where I was forced to use wire ties, that way I could have avoided the wire ties.  But it was that sort of endless thinking that kept me from doing this otherwise fairly simple but very important upgrade.

New Crossover and EQ for the Living Room System

I spent much time over the 2-day pre-Christmas weekend, and the 2-day Christmas Eve and Christmas Day holiday I also enjoyed, adjusting the Behringer DCX 2496 that serves as a digital crossover, eq, and time alignment device for the living room system.

When I started, I was thinking the sound was a bit thin.  The adjustments I made added solidity to the sound without adding too much bass heaviness, punch without bloat, impact without boom.  It takes skill and patience to do this.

In the process of adjusting, I used several tools.  I used my Kurzweil keyboard, which I temporarily moved in front of the listening position.  I created an instrument (Kurzweil calls any configured sound a "program") that's just a pure sine wave, extending as low as 16 Hz.  I started with the Default Program 199 and just changed "piano" to "sine wave", then I lowered the notes by two octaves.  Being able to play sine wave bass tones makes it easier to hear which ones need to be lowered and which ones needed to be raised.

I also used my B&K oscillator, which has digital readout.  Actually, this little oscillator is not as easy to use as it could be.  I choose a 100Hz range which actually gives me 10Hz to 200 Hz, with 100Hz in the middle.  But adjusting to specific frequencies below 50Hz gets harder and harder, as only tiny changes of the knob can scroll past a bunch of frequencies.  It works well enough to do slow sweeps to pick out problems.  But it might even have been better if I had used the Kurzweil and assigned a slider to do frequency adjustment.

I did not use any level measuring instruments.  I used my own ears, either at the listening position, or in other locations.

Let me run down the changes I made (and not elaborate all the history of what I tried to do, which would take too much time to describe).  I'll just give a few historical observations and/or rationalizations.

1.  I changed the crossover between the subwoofer and the Acoustat panels to 24 dB/octave Linkwitz-Riley at 80 Hz on both sides.  I am using this crossover with no polarity inversion on either side, and it is a nice property of the 24dB/octave linkwitz-riley that it does not require any polarity inversion.

Previously I had been using the crossover in a more ad hoc fashion.  I had been crossing over the bass side with 48dB/octave linkwitz riley, but left the highs with a gentler 24dB/octave slope.  My reasoning was that the subwoofer had best not be producing any mid bass since the panels do it so much better.  Well doing things this way there seemed to be a peak in the 82Hz region.  So I then also separated the crossover frequencies so that the panels were being crossed over at 88 Hz, and the bass around 80 Hz or maybe even lower.  This sort of worked.  But when I changed both sides to LR24, the peak at 82 Hz went away, and generally sound was smoother on both sides it seemed.  My ad hociness was requiring more ad hociness to fix.  Previously I didn't take the crossover settings very seriously, on the grounds that the speaker drivers are introducing so much additional frequency-related variation any crossover setting is really only an approximation.  It seems now that I should have been taking the actual crossover settings more seriously, especially when I was not seriously analyzing what the additional variation was that needed to be corrected anyway.  Given that you don't really know what else needs to be done, the nominal crossover is a good place to start.

2.  I eliminated the frequency contouring I was using for the super tweeters.  I removed the lowpass above 20kHz, which was rolling them back down again on the high side.  Now the super tweeters are crossed over at 15.5kHz, and that is it, they then continue on as high as their response (or more likely, the signal source) allows.   The effect of removing the lowpass was that the output level increased slightly, but I could not hear it as such, I just occasionally see a level indicator bar on the Behringer when previously I would see none.  I tried reducing the levels of the super tweeters, but ended up back where I had set them before, with the left channel only reduced 2db because it seemed to have about 2dB more output.  I confess the level settings make little sense to me and no measurements I've done have clarified the matter at all.  I would think the level settings should be much lower, but making it lower and the super tweeters lose the magic.  BTW, the magic is not at all brightness.  It's actually a kind of smoothness, where grain and grit go away.  As you increase the the level of the super tweeters (which actually have very little output that I can hear directly, as one would expect with crossover at 15.5kHz) the sound just gets smoother and more palpably real.  I just quit increasing the level because turning it up even more seems--to my mind--obscene.  I wish I was better able to do high frequency measurements to determine where the super tweeter level should be set.  But ultimately it comes down to what feels right anyway.

3.  I changed the bass EQ's.  I retained the huge notch at 45 Hz with -11dB.  I tried changing that big notch also, but whenever I flatten that notch, even to a very similar -9dB, the sound goes to boomy fast, without any improvement in bass tunefulness.  But many other bass eq's are changed, and many are new.  One thing that's also new is that I eq'd each channel differently, depending on several factors, both the pre-existing response, and the additional level that could be handled without causing distortion.

Notably, compared to before, I added various degrees of boost at or below 32 Hz.  That's because just just below 34 Hz or so the bass was tending to sound very weak overall, especially compared with the boom in the range 36-48Hz.  I specifically added a boost at 32Hz in both channels, since 32Hz is often found in music, and it needed boosting a lot (probably more than I actually boosted it).  Then I boosted tiny amounts at 27hz, 25Hz, and 20Hz.  The right channel starts to buzz when I boosted it for the lowest two frequencies, so i did that boosting on the left channel only.  Then, corresponding, I left out the 27Hz from the left channel, since it was already boosting 25 hz.  These are little 2db boosts, but they help.  The 20Hz and 32 Hz boosts are fairly narrow around 1/3 octave, but the 25Hz boost is a wider 1/2 octave or so.

I left the 45Hz notch alone in the end, but that was not without trying many other variations, such as moving it to 44Hz and then moving another notch at 57 Hz down to 48 Hz.  No matter what else I did, the 45Hz notch seems to be needed, so I ended up keeping it.  But the 57 Hz cut has been changed depending on the needs of that channel.  On the right side, there is more boominess above the 45 Hz notch, so it now gets notched out at 48 Hz.  On the left side, there is more boominess in the upper 30's, so it gets a notch at 39 Hz.  With the full LR24 crossover on both sides, there was no special boom that needed cutting in the upper 50's, so the 57 Hz cut is gone gone gone.

[I will be adding in here the last adjustments made in this time frame.]

Don't forget the center

Did some serious listening this week.  Listened to Infrared Roses by Grateful Dead.  This is one of my favorite albums.  All instrumental, mainly percussion.  Did Grateful Dead produce anything else like this?  I confess I'm not so attracted to the usual Grateful Dead hit albums.  My taste runs to instrumental recordings (I have Shadowfax in the background right now), and hummable melody is optional.

On living room stereo this was fantastic and real at the same time.  Very 3 dimensional.  Much of it sounds like real stereo miking, but other parts seem dubbed in, particularly the parts where one thread goes swinging back and forth between the speakers.  But at one part I wondered, is this a drum with pan-pot, or more than one different drum across the stage?  Anyway, it was very enjoyable.  There was a feeling like I was listening through giant electrostatic headphones, an incredible sense of transparency.  I can't remember if I'd listened to this seriously on living room stereo before.  I certainly never experienced it like this, the incredible spaciousness.  There was in a few parts a slightly, very slightly, grating quality to the highs.  Is this because my Acoustat panels are old and need to be replaced/refurbed?  I sometimes worry about that.  Or is it because electrostats including my Acoustats are very "revealing"?  It might be useful to fire up the electrostatic headphones one day and investigate further.  Last time this really bothered me and I had time, I replaced the old electrolytic caps in one of the Acoustats with a Solen, and that seemed to make it better.  That was in 2011 I believe.  I do need to do the other side soon.  How about 2014?

It was sounding a bit ping-pong on bedroom stereo.  I wasn't paying close attention at first, but then I noticed how the new adjustable bed was not centered vs the triple width record cabinet in the front center.  Hadn't I done that for the new bed yet, or had it moved by itself?  Anyway, I moved the bed, then the sound became wonderful again.  The center image was restored.  So many times I've heard other people's system with no center image, and my own systems have often fallen that way out of neglect and lack of serious listening.  A good center image is not just critical, it's crucial.

Compared to the living room system, bedroom system sounds smoother in some ways, but lacking in depth.  It was not the mind blowing experience it was in the living room.  But with center image fixed, it was still very satisfying.

Linear devices

My friend TIm has been thinking a lot this year about the design of audio amplifiers.  His first idea (later abandoned) was that MOSFETs are the most linear devices in their most linear region, which is typically the high current region surrounding 1/2 the maximum current.  A Class A MOSFET amplifier biased to that point would have negligible distortion even without feedback.  This optimal linear operation does not just reduce distortion to the 2nd order that some tube circuits do following the square-law operation of their devices, it would virtually eliminate distortion.

With regards to feedback, he follows the results of Peter Baxandall, who showed that either very little feedback and a tremendous amount is optimal--following an inverse goldilocks distribution.  For a typical transistor amp, this would be less than 1dB or greater than 35 dB.  Anywhere in the middle, from 1-35dB, and the feedback isn't actually reducing distortion, it's increasing distortion by trading away some distortion harmonics for others.  Often meter readers (such as myself) may not see this because the added distortion is in very high order harmonics which are cut off by the bandwidth limitations of the amplifier.  But the mere attenuation of this distortion doesn't eliminate its effect, which can be an added graininess.  (Subjective speculation here is my own.)  To really understand what's going on, you have to do the math, as Peter Baxandall did.

The problem with using the larger amount of useful feedback, 35dB or higher, is that you must first build an amplifier with 35dB extra gain.  For a large power amplifier, this is not an easy task.  Far easier to make the amplifier low distortion in other ways, such as using very linear devices, class a operation, and complementary operation (where complementariness is useful--this requires inversely matching PNP and NPN transistors, which even the best only approximate, and the worst are worse than useless).

So you can see this is pretty heavy stuff.

For awhile, Tim was fascinated by the Adcom amplifiers designed by Nelson Pass and others, particularly the 5400, 5500, 5800, and 5802 (the last not being a Pass design).  They use MOSFET outputs, and are biased into the high Class AB+, the 5802 consuming more than 400W at idle, for example, more than many so-called Class A amplifiers.

But by the time he actually got around to figuring the correct bias points, Tim realized these amplifiers don't even come close to the optimal linear area of MOSFETS.  (Actually, some of my questions led him to this disappointing analysis.)  For the devices in these Adcom amplifiers, something like 3-8 amp quiescent bias would be required.  And in contrast, despite the high dissipation in these amplifiers, it's spread over a large number of devices (for power handling reasons), resulting in less than 1amp per device.  The high quiescent power results not from super high bias but from lots of devices with large rail voltages, and relatively high, but not optimally high bias.

Further thinking revealed that it's basically not possible to use MOSFETS effectively at their most linear region in practical power amplifiers.  Either they would have to be especially low output super Class A amplifiers, or you would have to use liquid cooling of some sort with custom devices.  MOSFETS seem to be made for voltage amplifying circuits, not power providing circuits, at least if you are mainly thinking about taking advantage of the most linear operating region.

When NOT in their most linear region, MOSFETS actually have little to offer that's better than bipolar transistors.  The linear region in MOSFETS ends abruptly, and you don't really want an amplifier in those cutoff areas at all, as it may be worse than a bipolar cutoff region.  But that is exactly where all practical MOSFET amplifiers operate much of the time, notably the aforementioned Adcom amplifiers, but even Nelson Pass's most cherished designs like the F5.  His more commercial Pass Labs amplifiers are worse.  For example, the XA30.5 is really a high bias Class AB amp with 150W maximum power output.  It's not even close to the Super Class A that a truly linear MOSFET amplifier would use.

So then he got about to comparing the linearity of MOSFETS with transistors used in some transistor amplifiers, such as the Parasound amplifiers I own (I use a Parasound HCA-1500A in my master bedroom…where I used to use a Parasound HCA-1000A).  He quickly came to like the devices and the designs of these amplifiers very much.  They use very linear output devices in optimal circuits. He particularly liked the HCA-1500A.  (This made me feel good.)  But all the amplifiers in the series (850, 1000, 1200, 1500) were about equally good.  He was not so impressed with my cherished Aragon 8008BB, noting the bipolars used in the front end and driver are not as good as the JFETs and MOSFETS used in corresponding positions in the Parasound amplifiers.  All this came as a big surprise, since previously Tim was anything but a worshipper of John Curl (though he did think Curl was right about a  lot of things back when he designed the famed JC2 preamp for Mark Levinson).

Amp-100 to be sent back

I've felt good about Audiosource equipment I've bought in the past.  Their small speakers (recommended by my uber Audiophile brother-in-law) were sufficiently good to outshine a pair of Polk's which cost and weighed twice as much.  Their Amp One is currently in service as my office amplifier, and seemed like an excellent value for an amp with 100W RMS output.  But there was a problem with the $119 (at Parts Express) Audiosource Amp 100 I noticed from the moment I plugged it in that made me want to send it back.  The chassis makes a quite noticeable hum, which I measured as around 53dBC (approximately 45dBA).  I did mostly dBC measurements because the sound is mostly low frequency, and improperly tossed by the A weighting, and too close to the meter and room residual when measured in dBA.  I could measure as high as 55dBC within 2 inches of the unit, reducing to 53dBC at 6 inches which is a more fair weighting.  The highly tonal nature of the sound makes it more noticeable than the dB ratings would suggest, and it is especially noticeable in the remodeled second bedroom with it's sound blocking wall and door (I would guess ambient noise level below 20dBA).  Actually, I haven't always paid attention to chassis noise, I wouldn't notice it in my office filled with computers and other noise making machines.

While I decided this noise level was unacceptable, and an hoping for better performance from a Parasound Zamp V.3, it wasn't much more than a transformer type Tensor lamp.

On the whole, I don't believe audiophiles have paid enough attention to chassis noise.  They may obsess (as I do) about the difference between 16 bit audio, with a 96dB S/N ratio, and 24 bit audio, which in practice gets to about 120dB S/N ratio.  Meanwhile, chassis hum at 55dB with listening level around 70dB (and often less, this is a room for a lady and they generally keep the spl low), you have a 15dB S/N ratio--very unacceptable.

I wouldn't be surprised if many of my older amplifiers, including my two currently unused Parasound HCA-1000A amplifiers have even more chassis noise than the Amp 100 either.  It does seem like equipment gets noisier as it gets older.  And I'd be absolutely certain my McIntosh MC225 tube amplifier makes more chassis noise.  I'm hoping a brand new Parasound Zamp will be better.  It seems like Parasound made their zone amplifier as small as possible, and in this case small is good for a lot of reasons, including less chassis to vibrate.

Parts Express has a 45 day return option and quickly issued me an RMA.

SACD of Wish You Were Here

I waited and waited for the 30th Anniversary SACD of Wish You Were Here.  It was promised in 2005, but by that time Sony, who was then the owning label of this recording, was deep into a format war (DVD HD vs Blu Ray) and had just dropped it's involvement with SACD like a hot potato.  Sony had ruthlessly promoted SACD at the beginning (refusing even to include DVD-Audio capability in any of its DVD players), then, when it becomes useless to the company's grand strategy to rule the world, it drops it in a flash.  No new SACD players were being introduced by Sony, and no new SACD recordings either.  I'm not certain that the Blu Ray format was technically superior to the other, in fact I thought it was the other way around, but Blu Ray prevailed, largely through Sony's relentless singing of deals with movie studios.  Meanwhile, and despite being declared dead many times, SACD soldiered on mostly through audiophile recordings on small labels.  DVD-Audio, which had been my preference to SACD, simply because I believe in high resolution PCM over 1-bit-like systems, has also continued, the latest knockout for the format being the 40th anniversary release of Lark's Tongue in Aspic (which is fabulous, fabulous!) which appeared a year or two ago.

Meanwhile, the 30th Anniversary SACD of Wish You Were Here was not appearing.  By 2008 or so I had given up looking.

But next thing I knew, I saw it being closed out in a famous online recording store (can't remember whether it was Elusive Disc or Music Direct).  So I immediately ordered a copy.

And let me tell you again, as with the Lark's Tongue in DVD-Audio, the SACD of Wish You Were Here is fabulous, fabulous, maybe even beyond fabulous.

This recording sounds incredibly analog like when played on my Denon 5900 (redigitized by a Lavry AD10 for digital processing in my system).  There seems to be zero grain at all, just pure smooth infinite resolution.  I remember this from my LP's of this recording, but here it is far cleaner as well, so you can really hear the infinity.

I have never heard this utter grainlessness on a regular CD, though my Mobile Fidelity Gold Disk of Meddle comes pretty close, thanks to fidelity to the master tapes (no added compression).  Most of the differences heard with audiophile versions seems to be differences in the mastering as opposed to differences in the formats themselves.  Still there may be ultimate advantages to the advanced resolution formats, and I've come to believe that SACD, despite my earlier misgivings, is one of them.  Despite the questionable nature of what is going on in the highest frequencies, above 10,000 Hz, there is measured to be 20 dB less noise in the midband, and SACD recordings do often seem to have very clear midrange.

Wherever the clear superiority of Wish You Were Here in SACD comes from, I am grateful to have it.

Equalizing the Living Room system with a few digital notches

The need to equalize the living room system became most apparent when I was checking out the sound transfer to the newly remodeled second bedroom I am setting up for a friend.  Despite the soundproof door, there was noticeable bass leakage to the room with the door closed.  But opening the door, then the bass boom was overwhelming.  The excess bass is less noticeable at the listening position, but the bass line still sounded quite blurry there, possibly being affected somehow by the boom in the rest of the room.

I was playing one of my bass test recordings, Bass Ecstacy by Bass Erotica.  I have tuned the bedroom system so that it plays this very well, with bass sweeps audible down below 16Hz, and fairly level from there up.  In fact, I used this recording for the final room tuning in that room, which is why it can play Bass Ecstacy particularly well.

But the recording wasn't sounding as good in the living room.  Still too boomy.  Then I took a look at the digital crossover, a Behringer DCX 2496, thinking about adding some room node suppression, and I noticed I had already added some bass EQ, but I had turned it off.  Ah, yes, I remember doing that some time ago.  I couldn't hear much difference with the EQ turned on, so I decided to leave it off.  I can't remember when I dialed in the EQ a few years before that, some time after I gave up on having the Tact 2.0 RCS adjust the full bandwidth response (only the most recent Tact units permit you to target the bass only, which is the most easily pleasing approach).

The EQ I had set before was an 8dB reduction at 45 Hz with a fairly sharp Q of 4.  That would correspond to a bandwidth about a 1/3 of an octave, a little bit more than 1/3 of an octave actually.  But when I first started playing with the control, I though it was even narrower than that.  Decreasing the Q to 1 made a noticeable difference, but the price for removing the boom was to suck out the bass.

I was determined to solve this problem without doing more measurements (which I've done many times anyway) but simply cut-and-try.  And that actually seemed to work out (refreshingly well).  OK, I did play recorded tones from 16Hz to 160 Hz that I have on my music server.  These tones are supposed to have some warble, but I don't notice it.

Clearly a huge amount of suppression is needed at 45 Hz alright.  The room lights up with bass, even if it's less noticeable at the listening position not far from the center of the room.  But if a Q is used much smaller than 3, the bass cutout affects tones as low as 32 Hz, where there's a nodal suckout.  And the area needing bass suppression is fairly broad, from about 36 Hz to over 70Hz, but with most of the suppression below 50 Hz, with the peak boom at 45 Hz.

All the above mitigates solving the problem with only one notch filter.  I ultimately settled on using two notch filters, one at 45 with a Q of 5.0 and amplitude of -11dB, and another at 57 Hz with a Q of 3.2 and an amplitude of -6.  This cleans up the boomy area pretty well without affecting surrounding areas, particularly at 32Hz which is already weak.  It does create a bit too much loss just around 50Hz, but not too bad.  I dialed in some boost at 30 Hz, 3.5dB of boost with a Q of 5.  Actually, it could use some broader boost below 30 Hz, but hard to dial that in with a parametric.

The result is that Bass Ecstacy is now far more listenable in the Living Room.  It has a coherent and tuneful bass line, not just boom.

I know, some say I should analyze the entire bass with something like RoomEQ Wizard and dial in a few dozen notches.  When that sort of analysis is done with hundreds of notches like the Tact, it's less than impressive to me.  Optimizing the listening position too much doesn't seem good somehow.  It's a lot easier just to use a few notches when tuning by hand, and focus on the really bad room nodes.  I found in adjusting the master bedroom system that combining 1/3 octave graphic EQ (done sparingly) with parametric notches is the most flexible and intuitive, and has given me the most satisfying results.  I have a second Behringer DEQ 2496 to do the graphic as well as the parametric, but it's currently tasked to correcting the European EQ of my Kenwood KT-6040 tuner.

I should remember not to turn off the Bass EQ now.  It sounds much better with it on.

Remodeled Master Bedroom has much better sound

My master bedroom was remodeled in early September, just before my first cataract surgery on September 18.  The carpeting was replaced with the top grade of Armstrong LUXE Vinyl Plank flooring, with the recommended padded underlayment (which is specifically recommended by Armstrong for better sound absorption).  Surprisingly large gaps in the walls just below ceiling level were thickly filled with mud and floated.  (The original builder had covered up gaps with up to two inches of tape with a thin coat of mud.  Eventually, the paper had torn, leaving large gaps that looked like serious cracks but were really just torn paper.)  The gap between floor and drywall was filled with OSI acoustical caulk and then covered with 5 1/2 inch victorian style hardwood baseboards. (Once I heard how those victorian baseboards "sounded" to a knuckle wrap, I knew they were the ones for me.)

Perhaps even more crucially, I removed all the four 2x2 Sonex panels from the left side wall in the room.  Instead, I have moved two additional wood CD racks into that area for sound dispersion instead of absorption.

I didn't (and still haven't) carefully measured and adjusted speaker position.  (I don't want to do this at all until I have time to get it right, the soft flooring is certainly dented below the rounded ends of the speaker stand spikes, and I'm just going to ignore those dents since I like the flooring so much, but I don't want a proliferation of dents from moving the speakers around too many times.)

But despite the careless (relatively) speaker positioning, the sound is fabulous, and I have an ever better center image than I ever have had before!  I chalk it up mainly to the removal of carpeting and Sonex panels, which were excessively damping high frequencies.

I had largely been in the camp that says you want to listen in the deadest room possible.  And I'm still quite suspicious of relying on resonances and reflections for "good" sound.  But I now think it's good not to have too much damping in the mids and highs.

Door re-test with iPhone, shows 4dBA improvement only

Since I didn't measure the door beforehand with my wonderful Galaxy (not Samsung, Galaxy brand SPL meter from 6 years ago, recommended and sold by Home Theater Shack for use with their famous equalization program) microphone, but only the iPhone, I need to trot out truly comparable before-after measurements, with a huge disclaimer.

The disclaimer is that currently I measure background noise level of 39.0 dB (+/- 0.5 db).  So that sets a limit on even how low infinite sound blocking could achieve.  Actually, I now measure dBA under same conditions as before in the room as 42.0dBA, compared with 46.0dBa previously.  In other words , only a 4dB improvement (comparing apples and oranges by using the Galaxy for the after-measurement showed 9.5dB improvement).  But since the ultimate noise floor is 39.0 dB, that then suggests that the unbiased estimate of the true noise level would be about 39dB (summing 3dB higher to 42dB, assuming no correlation), and more like a 6dB improvement.  It sounds to me more like the larger 9.5dB improvement I estimated earlier.  Anyway, it is showing over 30dB of total noise reduction now (well, from the listening position) which is pretty good.  Before the new door but after the wall upgrade it wasn't seeming like much more than 20dB reduction, as indeed it was measuring (much of that coming from the preceding wall improvements; the door is off axis from the hall that leads to the living room, so actually this isn't a test of the door per se.  Instead, it is intended as a realistic relevant test from the listener to the non-listener.  This is what I seek to maximize here.)

Anyway, with the current best measurement showing 

Door still needs sealing

The IsoDoor presents numerous difficulties to the installer because the seals and even the threshold are not initially installed, as they are with most door you buy that are intended to have seals (interior doors don't).  But now that the door is installed without seals (after two days of hard work by a very experienced carpenter about my age) I did some measurements.  The iPhone couldn't be found so I used a read SPL meter this time, the Galaxy recommended by (and purchased through) the Home Theater Shack.  I should have been using a real SPL meter all along.

The iPhone app RTA showed almost identical A weighted and C weighted results.  Well that was wrong, I see now, with the Galaxy the A and C weighted results are quite different, and my guess is that all the iPhone results, even with C weighting supposedly selected, are essentially A weighted possibly by the limitations of the hardware itself.

I say that also because the Galaxy C weighted SPL's don't show any improvement in sound reduction due to the new door over the C weighted SPL's I got with iPhone on the old hollow door.  In fact, there is only 16 dB reduction between living room and bedroom NOW, according to the Galaxy.  And the iPhone C weighted measurement showed 21dB reduction...with the old original lightweight door.  That can't be right (and btw, it doesn't sound that way to my ears either).

Comparing A weighted measurements across the two devices, I'm seeing a useful 9dB reduction from the earlier door.  Exact comparison before and after will have to wait until I find the iPhone.  The new numbers A weighted are 71.3 dB in the living room and 40.9dB in the bedroom, a sound reduction of 30.4 dB.  The previous iPhone A weighted measurements showed 21.5 dB of A weighted reduction with the old door.

It sounds like much more reduction than that.  But when you close the new door, still lacking seals, what you hear is mostly the high frequencies.  The midrange and bass are obliterated, but the extreme highs are hardly changed.  So when I play pink noise it sounds like distant very high frequency hiss in the bedroom.  The sound seems to be leaking around the door (but not under the bottom, which still lacks threshold but does already have the sweep).

Strangely, the extreme highs may even be attenuated less than before.  That's how it sounds.  It seem like the extreme hardness of the door is providing a better path for high frequencies around the door than before.

So seals will have to be added.  But at the same time, the door fits very tight into the jamb and it's unclear if the seals can even be added.  This is very worrisome.

It would have been far far better had Sound Isolation Store pre-mounted the seals.  In fact, other sound blocking doors (typically modified metal doors) come with pre-attached seals.  Such doors are either somewhat more expensive or far less attractive or both.

Bedroom STC

Actually, I don't yet know exactly how to calculate STC, but here are the measurements I made prior to the installation of the new sound isolation door (IsoDoor from Sound Isolation Company).  The doors are being installed as I write this...

The measurement I make is intended to be a relevant measurement simulating a person sleeping on a bed just away from the door.  I measure the SPL in the room one foot inward of the AC outlet (about 18 inches inward of the door), between 3 and 12 inches from the wall, and at elbow height.  I move the microphone in and out to get the lowest measurement (reducing one frequency modes just a bit), and the optimal position varies slightly from test to test.

The source measurement is made at the listening position (either nose position, or right ear position).  Source is Stereophile Test Disc 2, pink noise track, both correlated and uncorrelated.  I set the level control on Tact to 80.0 (about as loud as I dare at 1am) with the Sonos level at maximum.  The IOS app RTA was used running on my iPhone 3G.

C Weighted SPL's

               Listening Postion        Sleeping Position
Corr        67.7                             46.5
UnCorr   66.8                             47.2

While the correlation decreases output at listening position (due to relative lack of bass augmentation) it actually increases it slightly at the listening position.  Correlated is probably the more relevant test, since recorded deep bass is typically monophonic (with central image) and that shows 21.2 dB of sound reduction (this is actually about 10dB more than what I would have expected based on casual listening--there doesn't seem to be much reduction at all).

A Weighted SPL's

               Listening Postion        Sleeping Position
Corr        67.7                             46.2
UnCorr   67.7                             48.2

The A weighted test surprising shows almost exactly the same results in the correlated test, about 21.5dB of sound reduction (the 0.3dB difference well within measurement uncertainty, btw).  The uncorrelated result is 0.1dB worse than with the C weighting.

New System Photo Explained

The previous living room system photo was very outdated.  Taken in mid 2010, it showed the Krell, Acoustats, Elac supertweeters and SVS subwoofers.  The speakers are the same but the Krell is now offline (and to the side, still visible in new photo), having been functionally replaced by the Aragon 8008 BB.  The listening position has been moved up to 3 1/2 feet from speakers, the most important change of all.  Normally facing straight ahead, the listening chair is rotated in the photo to avoid hiding all the equipment, but is otherwise in correct spot.  And now there are additional piles of equipment to the left and right of the two center components.  From left to right and top to bottom, here are the electronic components:

Far far left:
120VAC to 240VAC hammond stepup transformer (for KT-6040)
Radio Shack Remote Extender

Far left:
Jenson Iso-Max...used to isolate ground from Kurzweil K2661 cables and convert to unbalanced
Behringer DEQ 2496...used to equalize output of Kenwood KT-6040 tuner from Europe.
Classe CP 35...used (sometimes) to amplify output of Kurzweil providing 10dB more gain
   Could also be used for transformerless input and conversion to unbalanced
Pioneer F-26...classic analog super tuner, used for living room system only
Kenwood KT-6040...digital remote tuner used whole house through Sonos
Sony CDP-507...original CD only player used for guest CD's

Left:
Aragon 8008 BB...amplifier for Acoustats full range above 80 Hz

Right:
Sonos ZP 80 zone player
Behringer DCX 2496...3 way digital crossover with delays and eq
Tact RCS 2.0...main system preamp, selector+level+eq, room correction not used

Far Right:
dB systems 5 way selector with teflon jacks
Lavry AD 10...digitizer for F-26 and Denon 5900
Denon 5900...universal DVD, DVD-Audio, SACD, HDCD player
Acurus A250...power amp used for super tweeters
Belkin UPS and audiophile power conditioner...used for all except Aragon

RTA not working on Samsung Galaxy S4

Literally, the application named RTA does not work correctly on the Galaxy S4.  It is unable to show an interesting graph of pink noise response from my living room audio system (pictures below, sorry they got turned upside down somehow automatically).  Furthermore, the response shown doesn't vary much with orientation of the phone from the listening position.  Whether I point the front of the phone at the quite directional electrostatic speaker or not makes very little difference.  (That lack of directionality might be good for some things, like measuring actual noise exposure, but bad for measuring specific sounds or system responses.)  In case you are prepared to argue the problem is the ultimate SPL limit, which may well be around 90dBA as it is for other Android phones or so I have read*, I tried measuring at two different system SPL levels, which were around 75dBA and 65dBA, and they showed the same uninteresting picture, simply having an enormous peak around 8kHz and rolling off dramatically above and below that by 10's of dB's.  (Tthe left is 75dB upside down and the right is 65dB turned left.  And, strangely enough, it matters little which way the phone is turned, just as the pictures might suggest, but actually I had the phone laying flat in the same direction for both pictures.)

In contrast, the application named RTA on my iPhone G3 shows a very interesting and plausible frequency response of my living room system, broadly flat through the mid to highs but with a few peaks and dips below 600Hz, and then rolloff at the very lowest and highest frequencies.  I'm sure the rolloffs at the top and bottom are iPhone limitations, I have 16Hz-25kHz nearly flat response, but I think the rest is fairly accurate, or if not, it's simply the result of 4 and a half years of smartphone wear or careless measurement technique resulting in minor reflections.  The picture changes in interesting ways depending on orientation--notably the extreme highs get flatter when the bottom of the phone is pointed right at the speakers.  That's what I mean by an 'interesting' picture, when you make changes, you can see their effect reflected in the graph.

A different Android app running on the Galaxy S4 gives an equally plausible but different picture.  This is from AudioTool, the paid version:

Because AudioTool does show plausible response, I suspect the problem is that the default calibration of the free version of RTA isn't good for the Samsung Galaxy S4, and unfortunately you can't change the calibration on the free version.

(*I don't know why Android phones are limited to 90dB.  Tiny capsule microphones usually have a much higher limit, something like 110-140dB.  The iPhone SPL app can measure up to about 105dB, but that could be because the app is automatically compensating for known microphone compression.)

Types of Listening

1) Annoyed listening (e.g., neighbor's dog barking).

2) Enjoyable Background Music Listening

This is fine, even if audiophiles consider it inferior to the point of condemning it altogether.  One can't always devote full attention to music, in fact, for most of us it's rare to have both the time and inclination together.  So having pleasant background music can be one of the greater joys in life, if done well.  I prefer the absence of talk altogether, but can listen to non-commercial radio also if there is minimal talk.  One of the radio programs I used to love to hate was "Adventures in Good Music" which seemed to be more talk than music.  Whole house music systems like the Sonos I use are helpful, as were auto-reversing tapes, and FM radio.  Internet radio based on lossy compression--I don't like it much.

Background music listening can't provide the intense pleasure that serious listening can.  But because it is something can can be done over a relatively larger proportion of one's time, and I'd say in principle the more the better, it can constitute a large part of audiophile enjoyment.  It should not constitute ALL of the enjoyment, for that would be saying that NO serious listening was being done.  But taking up about 95% of one's total system playing time, and providing about 50% of the total enjoyment, sounds like it would be about right for a working person with many other interests.

3) Serious Listening

You can't be doing much else while doing this.  Drinking a wine or beer, maybe, but even that's pushing it.  But rather than concentration, the goal is relaxation, relaxation into the music, not thinking about much else, letting thoughts go away generally, including thoughts about the qualities of music systems.

4) Comparative Listening

Listening to more than one system variation with the aim of determining differences and establishing relative or absolute quality levels.  This is most definitively "Comparative Listening" when done in one listening session or a planned series of sessons.  But long term comparison is possible also--so basically what determines whether listening is comparative or not is the degree to which thoughts about audible differences arise in the mind at the time.  So while I do little but long term comparison anymore...I'm still guilty of comparative listening, though to a lesser degree than some.

This is what audiophiles are known for.  However, contrary to some self-appointed categorizers, there is no essential need to do this to be 'an audiophile.'  As stated in an earlier post, 'an audiophile' need only be enthusiastic about music reproduction, not necessarily dedicated to the labor of improving audio systems through comparative listening, component substitution, and/or tweaking with accessories or special treatments.  I'd suspect that too much of those latter things, and one is well on the way to audio burnout.

Besides which, comparative listening as it is usually done should not be taken seriously.  At minimum, comparative listening should be done Double Blind, where even the experimenter doesn't know which equipment has been chosen.  And level matched.  Any results other than those from a level matched double blind test should not be taken seriously at all, ignored, or not even thought of if possible.  It is very easy to be wrong, our expecations and pre-conditioning (including, the previous playback) can easily dwarf actual equipment differences, and usually do IMO.  Unfortunately, most audiophiles I have met seem to lack the mental compartmentalization to doubt their own hearing sufficiently, in which case I'd suggest they'd be better off not doing comparative listening at all, because any judgements made are likely wrong, and after awhile a large set of wrong or random judgements may cascade to very wrong audio configuration or (just as likely) audiophilia nervosa, when one can't enjoy listening to anything anymore because of the feeling it needs improving somehow.

The reference is double blind testing (which includes ABX type) in a full trial of at least 30 trials, testing to a 95% confidence interval.  This is a lot of work and very hard to do well.  I have conducted 3 full blind tests in my lifetime on a well regarded audio guru.  Not one of those tests came close to the 95% confidence interval.  The audio guru chose the tests because he felt he could not fail on even one choice.

I make Comparative Listening a special category of listening because...it is fundamentally flawed.  Thinking of the need to make comparative judgement not only takes out much of the inherent pleasure in listening to music, it makes itself impossible.  A correct comparative judgement can only be after the fact of listening.  It need not be long after, it could be mere seconds, and that may be the best for some kinds of tests.  But once comparative listening is the goal, it becomes impossible NOT to think about the ultimate conclusion, and such thinking makes it impossible to feel the inherent pleasure, which is what the comparative judgement should be all about.

Ghostly Heifetz sound made real

Just before breakfast on Monday May 27, 2013 (Memorial Day Observed) I decided to do no work not urgently needed.  So while I would wash underwear (urgently needed) I wouldn't bother mowing the front and back lawns, let alone thinking about any of the other million-and-one chores that were still undone (and mostly still are).

I like to make this 'no work' rule from time to time when I can, a virtual sabbath, since I often find myself dreading the endless-weekend-work even more than my job, and don't have the organization and whatever-it-takes to get all my infinite household chores done during the "week" (M-F) itself, something that some lifestyle coaches say is a very good idea (leave the weekend to socializing and fun).

And I find a virtual sabbath helps focus the mind in a strange way.  If on the other hand, I decide to mow the lawn on a weekend day, that does usually get done, but little else worth speaking about.  What little time there is for relaxation on such a day generally gets spent on the easiest and least spiritual forms of relaxation.  Yes there are lots of things easier than sitting down and seriously listening to a full disc of music--a task so difficult, in fact, that many audiophiles never do it, I am finding.

Having acquired a rule-bending personality, I don't like to make my virtual sabbath a day that must be spent doing uplifting things....like listening to entire discs of music.  That would be a very tough rule, and I'd never follow it.  Rather, I let things happen.  Eventually, given that I never have to something physically demanding, I will eventually get around to the spiritually uplifting things.  After doing everything else, as it were.  And this Memorial Monday, I actually did.  I got around to doing these uplifting things:

1) Watched four 30 minute lectures from The Great Courses on Complexity.

2) Read the preface and first two chapters of J.M. Keynes' masterwork, The General Theory of Employment, Interest, and Money.

3) Listened to the Heifetz Concertos disc from the RCA Living Stereo series on Hybrid SACD.

OK, #3 was the very last thing I got around to.  By 12:30 AM (actually Tuesday Morning if you follow the rules) my eyes were blurring to the point where no more reading or watching would be possible.  At this point, listening to music with eyes closed would be the best thing to do.  And indeed it was!  I enjoyed the entire disc, only falling asleep a few times...

Unfortunately, I didn't listen very critically at first, mainly focussed on my obsession with the center of the image, which requires small left and right movements of the listening chair to correct.  After that, it was a long time before I noticed something about Jascha's violin.  It was sounding ghostly.

Thinking of that had me laughing, and helpfully helped wake me up a bit.  Of course Heifetz sounds ghostly, he's been dead for decades!  But while the thought might have been funny, I clearly wasn't getting the best sound.

The fix was simple.  I turned up the level.  These classic recordings are recorded at a very low average level...allowing incredible dynamic range, and also (I am guessing the engineer's intent) hiding the analog noise level.  Turning the volume all the way up to about -1dB relative to 0 (actually, the Tact showed 92.1 and 0dB is 93.8) made it sound just right.

I am running the output of the Denon 5900 into a Lavry AD10 set to maximum gain.  I think this leaves about 2dB of analog headroom, thus my -2dB was actually -4dB relative to the recoding.  But still, even -4dB is an impressively high volume level for my system.  Most of the time I have the Tact set to something around 80.0, which would be -13.8dB.

After cranking the level, I was thinking how this compared to modern recordings.  Modern recordings would most likely sound cleaner, clearer, more transparent.  This recording has a warm fuzzy glow.  However, the warm glow compliments the music and makes it soaring, never edgy.

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.]