Home Sweet Home

My almost annual trip to California resulted in visiting only one currently active audiophile, the very industrious Roger who always has endless audio construction projects and modification experiments in process.  Well this time I thought his sound was far better than in a long time, and possibly due to his use of actual commercial amplifiers, Manleys, which looked sufficiently muscular and I'm sure were undeniably hifi.  Previously he was into SET's with a home made SET having an enormous tube.  SETs can have serious rolloff, and I think his might have.  However, Roger attributed the most recent improvements to damping, specifically the metallic 3M damping material he is now applying in over inch thick layers to potentiometers.  (This looks like great stuff BTW and I think I will start trying it myself in applications where hockey tape wouldn't work.)  He's tossed his transformer volume control (very complicated and expensive) because, in the end, he was unable to find a way to properly damp it (all those transformers!).  Still present were a few of his amazing huge clamps, (everything must be clamped rigid, Roger believes) but fewer clamps than before, or so it seemed, maybe I wasn't looking hard enough.

His system IMO has generally lacked high frequency bandwidth as it's first offense.  He's somewhat apologetic about that, claiming great hearing loss.  However, this time it was much more extended than previously I thought, despite his still taping off the electret tweeters on his Zu speakers.  It actually sounded tolerable bandwidth wise, also with new subwoofer.  (Perhaps the bass even makes the highs sound better.)  It even had one positive quality, solidity, which my system relatively lacks.  A lot of speakers based around cone speakers in small rooms have solidity, though it's often too much in-your-face solidity, Roger's was about just right, in that one parameter.

However, come home, and the improvement otherwise in my system is staggering.  I don't have a thin midrangy system like ESL-57's.  With my gigantic subs, electronic EQ, and huge Acoustat 2+2's, I still rate well on solidity, though it's highly dependent on the actual crossover, EQ, delay, and other adjustments.

And then soon my system became even better.  I was then so inspired to do some more oscillator sweeps, and 1/6 octave phone analyzer readings on the Stereophile Pink Noise track.

I clearly did show some loss at 100Hz in one channel.  Strangely there was NO notch at 100 Hz, (though I had set a notch to 104 at one time, the size was 0.0dB) just notches at 72.6 and 130 which I changed to 72.6 and 140, which somehow made a larger difference at 100Hz than you might expect, not flat but not terribly sucked out anymore.  This improved the "solidity" significantly.  Note the subwoofer is crossed over at 100 Hz, but notches as high as 140 make a significant difference to the sound.

I also fine tuned the Linkwitz range notches to 2729 Hz (1/3 octave, -1.5dB) and 5200 Hz (1/2 octave, -3.5dB).  These not only make for a smooth dip in the 2-3Khz range as various people have recommended, it flattens some issues with the Acoustats themselves, which actually seem to have a slight peak right where it hurts the ear the most around 6kHz and also around 2.7Khz  Previously I used a single notch around 3.5kHz, but that left lumps at the new notch frequencies.  I'm not sure when I zeroed out the sub 1000 Hz notches in the midrange.  When I got the 2+2's at first those EQ's, which I had tuned using the 1+1's,  seemed OK, but later it seemed they didn't quite fit where notches might be useful, and they weren't as much needed.  I didn't photograph the midrange EQ in November when I did my first official 2+2 sweeps and adjustments.  I think I might have zeroed out the sub 1000 Hz notches then, and/or added two notches above 2 khz, but they weren't exactly the same as now, or I added different sub 1000 Hz notches.  Anyway, in the interests of restoring solidity, and perhaps more, I've removed all sub 1000 Hz notches for now from the midrange, and have once or twice tweaked two notches above 2kHz (there could possibly usefully be a third around 8-10 kHz, just to flatten/smooth the response, but that "artistry" will require a lot more development).

The sound even more improved, and more solid, I had accidentally moved the listening chair too far back during packing but lined up to the original tape which puts the ear about 6 inches beyond the peak of the roof, works great.  I started going through my favorite "special" discs, DVD-Audios and SACD, and then just kept up playing CD's mostly, my new years resolution being to play more music, and now I'm pretty much keeping it always playing, most recently from discs since they are easy to collect from my racks (not so easy to put back).  Each different method of selecting music (each different App, or physical media storage system, etc) makes you choose different music--a sadly uninvestigated phenomena.

I've also recently sweep tuned the kitchen subwoofer for greatly improved sound.  It turns out the room exaggerated 41.8Hz, where I now have a -15dB notch 1/2 octave wide.  With this change, FM no longer makes one sick near the sink, for example, because of throbbing bass.  I may now be able to switch back from the remarkably fine Sansui TU-D99X, thin but possibly their actual finest model which introduced the Walsh-type FM decoder they patented, to my still best tuner of all time, the Kenwood L-1000T.  The latter had become intolerable because of the bass exaggeration.

With these minor (though, possibly the most important) changes out of the way, I then moved on to Tweak-O-Mania!

Just before my trip I had been in crunch mode getting my house "ready" for the trip in various ways.  The biggest project of the year was making the old Kimball organ redundant by sampling it, and then getting it taken by Salvation Army.  But projects continued right up to the last minute, including fixing tile in the small bathroom, hoping to make my house perhaps just enough more liveable for someone to move in (which, wouldn't be expected, but there is always a remote possibility, and those possibilities are important if for nothing else than one's own self esteem--I promised to get rid of the Kimball 6 years ago for one thing, and I'd been promising to fix the tile for several years).

So anyway, I don't really mean to complain so much, but because of all the other stuff, highly visible audio project were being deliberately postponed.  If I barely have enough time for all the other stuff, which indeed I barely did, just barely completing everything I'd planned to do, I wouldn't have made if I had dilly dallied with audio projects of questionable importance.

BUT NOW I COULD DO THEM!!!

So finally I had time to fix the big speaker wire messes, including the ways that both supertweeters were hooked up with big extra coils of wire and stacks of connectors that would be hard to explain to another audiophile (but could be explained, if I only had the time, and a patient enough listener).

The supertweeters are now (20 days after returning home) hooked up pretty much correctly, with the wires from the amplifier running to "connector blocks" where they are merged with the thinner gauge wire that connects to the roll-off capacitors and both supertweeters.  The connector blocks are actually repurposed Furutech tightening bananas, which accept a huge gauge of input bare wire.  I strip and twist a generous portion of all the wires together and shove them all into the bare wire inlet of the Furutech and then tighten the two screws as much as possible.  They most connect to the two of 11 gauge wires for each polarity of the Canare 4S11 that runs up to the connector block, together with the 16 gauge wire stranded that runs to the capacitors and speaker terminals.  They do this easily.  This is one idea that I've sort of adopted from my friend George, who uses a special kind of copper ring for joining pairs of speaker wires by a similar method.  He claims his copper ring is the best, blah, blah.  But the same principle can be used with most anything that clamps onto wire, I decided, and one can hardly beat the finely gold plated copper used in the Furutech connectors.

Actually, I fear this clamping method isn't as perfect as George believes.  If there is any room for oxygen, the copper will corrode, and corroded bare copper is a semiconductor.*  The bare stranded wires are not being clamped tightly enough to eliminate copper corrosion.  It's better to have silver plated copper in the first place (George strongly believes that silver or silver plating causes harsh sound, but that's because his reference is the softened sound he expects despite his transducers), then clamped strongly or soldered in some way that doesn't permit air oxidation.  But as far as resistance, etc., the fairly large mass of wire now clamped with two screws in the Furutech connector is good enough for the supertweeters, I have decided.  And it neatens up the wires, gets blocks of connectors off the floor, and it makes the wires as short as possible, with a new twist I'm also employing in the midrange.  I use lower inductance and resistance wire (the Canare 4S11) for the longer runs, and the slightly higher inductance and resistance wire (my silver plated teflon coated milspec copper stranded wire) for the shorter side.  It just naturally happened I had already set things up this way for the super tweeters, but I had a coil of the teflon wire on one side connected to a block of connectors because I had never had time to get around to trimming the wire connections to fit (ever since I had to move the supertweeters inboard of the Acoustats after upgrading to the Acoustat 2+2's).

(*George used to use copper stranded speaker wire that had turned sea green because of corrosion.  He said it's the best just like that, determined by his listening to it.  I'm not sure what kind of wire he was using when he quit playing his stereo 3 years ago, but it's certainly not silver plated.  He generally likes to space wires as much as possible, which creates a huge amount of inductance.  This is all NOT high fidelity to me, but it works for him.  He says that what I do, and things I often say, don't make sense.)

In redoing the supertweeter speaker/amplifier wiring, I discovered that one supertweeter was formerly out of polarity, or at least so it seemed.  Everything is now visibly correct from the speakers to the amplifiers.

I also had a very ugly block of connectors for the right Acoustat, where connected a short piece of 4S11 to a short piece of 16G "audiophile" wire (using polypropylene inner sleeves protecting the bare copper with the corrosive and poor dielectric contact with the vinyl covering) that actually connected to the acoustats.  This ugly block of connectors sitting on the floor consisted of a pair of Furutech locking bananas, a second pair they were screwed into, and a third RadioShack Gold connector merely to keep the two pairs of pairs of Furutech connectors properly spaced.

This all became necessary when the original short piece of custom terminated 4S11 was no longer long enough, either because I have moved the 1+1's out from the wall more (which I started doing in spring of 2019) or because I switched to the 2+2's when then had to be moved as far as possible to the sides, with the supertweeters now on the inside because I can't fit them on the outside and still have a passageway through the room.  So I've had this ugly block of connectors on the floor for about 6 months or more and it's bugged me every day.

My first approach, however, didn't seem fully satisfactory.  I simply soldered 5 inch pieces of the teflon solid wire to the ends of the Canare 4S11.  This solder joint was made in a fairly convenient but questionable way.  I simply wrapped a loop of the silver coated solid wire around the stranded copper, squeezed it down as tightly as possible, and soldered with eutectic solder.  After the pressure of the iron was removed, the wire always slinked a bit, making me worry about cold solder.  I worked on this through numerous attempts for two hours, and finally decided it was good enough, the joint looked smooth all around, just not quite as shiny as I would have liked.  I then was unable to find my "rescue tape" and all my shrink wrap was too small, so I had to cover up the joints with vinyl electrical tape.

That got rid of the connector blocks, but it looked ugly to audiophile eyes and thinking.  It would be much better to use a single piece of wire, preferably the 10G solid core silver plated teflon coated hand twisted wire I use for most everything else now.  But I would only want to do this if I had a locking banana of the "straight" kind, and not the "bent" kind like all of my Furutechs (which I just seem to have run out of again, I buy them in bunches now).

Well, it turned out that in fixing one side of the supertweeter wiring, I freed up a short piece of wire that had exactly the right kind of "straight" locking banana I needed, AND it had locking screws.  (I have some other pairs, of terminated 4S11 with straight locking bananas, but the 4S11 wires are crimped into the bananas, not screwed.)

These straight locking bananas were just what I needed to take one existing short piece of teflon twisted pair wiring and put it to use for the right Acoustat.  Crawling around the floor and especially making the connections behind the highly modified QSC ABX box I use for amplifier switching still took almost two hours of work, but finally the right Acoustat was wired nicely by audiophile standards, with no electrical tape or ugly solder joints.

At first, however, it sounded harsh.  I had the wire almost 18 inches off the ground, going up and over the Acoustat interfaces and the stack of digital equipment down to the ABX box connectors.  It occurred to me this "high" run could be picking up distant interference, or perhaps vibration.

I lowered the wire by bending it (solid core wire must be bent to each new shape) down closer but still about 7 inches above the floor mostly.  I also "damped" the piece of wire by hanging a Velco strap tied to a loop of phone wire serving as a dumb weight, in the middle of the wire run.

At this point it sounded great, and the soundfield seemed more unified than ever before, or at least a long time.

I then remembered that by switching from 4S11 to 10G twisted pair I was approximately doubling the inductance and 50% increasing the resistance.  This would make this shorter run of wire from the ABX box more similar to the twice-as-long wire (now 4S11) running to the other speaker.  So, for the first time in a long time, perhaps, I have basically equal speaker wire inductance on both sides, and the resistance is fairly close also.

I've never been one of those people to axiomatically cut all speaker wires for stereo channels to the same length.  I know this has virtues, but it then leaves you with useless coils of wire, which may better pick up noise and have leakage characteristics.  And with all the wires I have, it gets maddening very quickly.  So instead I terminate each wire to the more-or-less exact length needed (usually with some extra margin, but over time, I've been finding margins not useful anyway, and I just cut things pretty close to the lengths needed).

So just 20 days back home, I've made major improvements some delayed perhaps by years, and I'm enjoying music more than ever.

The rattle in the right Acoustat has not been noticeable, even though I'm consistently playing at the "realistic" level (1.0V input max to the Hafler which is just below it's rated sensitivity for full power).

The Plot Thickens

I found I could eliminate all the rattles in one speaker playing Grouse by lowering the level to -14dB (0dB is approximately amplifier rated output and speaker maximum undistorted level).

Grouse We want to be loved is a veritable audio torture test, as it turns out, with extremely high peak levels, probably loaded with inter-sample-overs too (where the output level goes above 0dB between samples after oversampling...I've noted other post-2000 albums that do that a lot...this can add up to 4dB to the actual output level, I've seen so far) too.  Even though it doesn't sound like the bass or anything is hitting the stops, on this album, it actually is.

I thought of it as an amazing, and pretty good recorded album.  But I tried listening to it at the level of -14dB where it seems no speaker anomalies occur.

At this lower level, the album goes from interesting to uniquely magical.  The transparency of some recorded bits is mind blowing.  There's a bit (perhaps overplayed) that sounds like an old tube amplifier going into high frequency oscillation.  This has an almost impossible you are there quality, but only played at -14dB.  I get the sense, momentarily, that I'l at John Iverson's house listening to his Corona plasma speaker, except that he'd be playing 20dB louder.

One also hears quite a bit of clever panning and other fakery.  It's great fun, even more fun than at -6dB where I had been naively listening the first time.  That seemed like about all I could stand, the album seems far louder than the playback level, even for an experimental rock-or-whatever album.

But thinking about the improvement in sound information at lower levels...surely that's not my perception at work.  Normally we hear more and more at higher levels.  Listening to the album at -14dB (which was very late at night, perhaps 2am, and very quiet) for the first few minutes, I was nearly kicking myself to kick it up a bit.  I argued with myself that I was goind to do this for an entire song (and then, the entire album) as an experiment, which I didn't want to shortchange myself over.

What must be going on is that the speakers themselves lose it more and more at higher levels.  Especially with torture albums which are slapping them around to the maximum extent.  There may be some kind of internal vibration, which gets more and more intense.  Finally, because of some crack or whatever in the internal supports on one speaker, causing a distinctly audible buzz.

But even before that buzzing occurs, there is already sufficient internal vibration to obscure information, leading to greatest effective transparency in the speaker-ear system at some lower potential level.

I had thought that something like this was going on before, but never had such a clear indication.  The Acoustats are like 2 speakers, one with Quad ESL like transparency at Quad ESL like levels.  And then also a head banging speaker which will play much louder, but loses that magical transparency at higher levels.

And it's why some people have gone to extreme measures with floor and ceiling braces to try to fix the Acoustat internal vibration issues.

(There is also an effect in human perception, where we lose information at higher levels.  But that would only be at way higher levels than I was getting to at -6dB.)

In some sense, the audible buzz is a warning indicator.  Turn the level down for better sound.  But I do intend to fix it anyway.

Ohoh. Rattle in the 2+2's

In the past few days, I readjusted the bass EQ using my method (sweep with oscillator and adjust parametric EQ functions to fix all the bulges).  Last weekend the right sub wasn't even working (not sure how long) but after fiddling with it (I suspect loose something related to AC power) it started working again and I'm now keeping it in the "ON" rather than "AUTO" mode which might keep it warm enough to keep from dropping out again.  The faulty speaker is near the doorway were it may get more humidity.  Before I replaced the plate amplifier about a year ago, I had always kept this unit turned on.

I did the EQ adjustments (notch filters) basically from scratch, staring with the biggest bulge (either 40Hz or 45Hz depending on side) and working on down.  I can't remember the last time.  Anyway, it worked out fabulously, better than ever before, very flat bass response and no 100 Hz bulge (which has been one of the toughest to deal with).

So, on Friday I was enjoying some of the most transparent sound I've ever heard, if you can call it that playing electronic music from William Orbit and Grouse.  But then, playing Grouse (which itself is full of "deliberate" rattles and distortion) it became clear there is a rattle in one speaker.  I reversed channels and it was still there.  I tried squeezing the speaker (which seemed to help at first), slight shaking, etc., nothing helped.  I turned out lights and did not see any sparks or smell anything.

I isolated a 15 second piece of music which curiously doesn't sound loud but has a very high peak to average ratio and seems to include a bunch of tones which harmonically beat somehow and stimulates the problem consistently.  Playing that over and over, it seems I can sometimes get the rattle to go away, but then it comes back again.  Sometimes playing at the loudest possible level actually suppresses it longer--but then it starts again.

I'm pretty sure it's "mechanical," a broken stator wire support or something like that.  It probably wouldn't be hard to fix, once I get the stock off of the speaker, which I've never done before, mostly fearing the part of putting the sock back on--which is near impossible to do as well as the factory did it (no wrinkles, etc).

I'm not going to bother fixing this until mid January at the earliest.  It's another "opportunity" to do something I've never gotten around to doing before.

Acoutstat 2+2

I haven't written this up yet?  The Acoustat 2+2's that I set up in July have of course made the biggest difference in 11 years, since I got the Acoustat 1+1's.  Speaker upgrades are big.

I had for long doubted the 2+2's were even better than 1+1's, for the reason that the 1+1's are more like a line source, or so it seems at first glance (more about that later).  So, I reasoned, the 1+1's are the perfect speaker for my small space, because I don't really need more dynamic range (not really, as it turned out) just the thinner profile which works better being closer to.

So, even the likes of Mr Acoustat, a late longtime fan on DIYAudio (and no connection to the original company) preferred the 1+1's, for his mostly smallish rooms.

However, as time went on, I was beginning to think there was a certain loudness I really could not (or was never tempted to ) exceed, because the 1+1's were sounding more closed in, so I always kept the levels somewhat below the 1+1 maximum output levels, so I became more interested in what the 2+2's might actually sound like, or even thought of getting a high output level 2nd set of speakers just to hear "louder" when I really wanted to.

It was about that time, 3 years ago or so, that a friend acquired a pair of 2+2's.  I heard them in his much larger room, and I was blown away.  I felt there was nothing lacking at all, they were simply better in every way.

It was exactly that same pair that came up for sale, with an interface refurb and mod (because the owner feared he might have damaged them).  They still seemed OK if not as glorious as I had remembered them, so I bought them.

It became clear after a day that one speaker had far less treble, and I couldn't even adjust the difference out using the level controls, nor did the "air" mod help.

Fearing the worst, I went through a process of system substitution, and narrowed the fault to one interface unit, and then to the HF transformer.

Well, as it happened, I had a spare NOS Medallion HF transformer to replace the HF transformer.  The rest of the transformers in these 2+2's are NOT Medallions, as I had hoped.  But actually, this makes LESS difference with the "C" mod that has been performed on them.  The C mod reduces HF transformer saturation, so the enhanced anti-saturation construction of the Medallion transformers is less critical.  These units ALSO got the "air" mod and switch added, but the "air" mod seems to bypass all the attenuation, possibly putting more voltage on the HF transformers, and frankly I don't like the way it sounds anyway.  The C mod design, and with the variable attenuator knob type interface, seems like the best interface design to me, and can be dialed in to perfection (though I have only moved the controls a hair off center) with ease.

I had almost finished re-connecting the new transformer, I was on the last lead when I noticed the original had not been wired correctly.  The - HF and - LF leads were reversed.  Well, that for sure was the source of the problem!  But having only one more solder connection left to finish the job, and compared with hours of work (possibly breaking something) removing the new transformer and re-installing the old transformer.  So I decided to leave the new transformer in place, and now with the correct wiring too.

This restored the amazing sound I had originally heard years before.  It was simply better in every way.  Even the "line source" issue is not as I had imagined.  And, it doesn't seem that having one Medallion HF transformer significantly unbalanced the system either.

Many things are at work here.  First, having twice the membrane area means that for any SPL level, there is half as much panel movement, therefore half as much distortion from that mechanism.  And there's less saturation in the transformer as well, and half the level from the Amplifier is required, meaning less distortion also.  So, no matter how loudly you play, having twice as much panel area gets you half as much distortion, and this seems apparent in the effortlessness of the speakers as well as the greater information.  That was the key thing I noticed from the very beginning (and in fact before I fixed the transformer: more information.

It's much like that 70's photo of the listener being blown away, as if the speakers were something like a hurricane.  But in this case, there no physical force or movement of one's body or hair, it's a change in one's mind: there's more there there somehow, at every level, it's more real and satisfying.

Greater information at all levels is the biggest improvement, but there are others.  Perhaps the second is that I have finally adjusted the highs correctly.  With the 1+1's, with their newer style box, one can't just "turn a knob" to change the HF level.  It's not that hard if you have the interfaces unscrewed, as I always did.  But it was hard enough, that I never even bothered to try.  I just left the HF adjustment alone, figuring it was "correct."  But I have no idea if the previous owner adjusted them.  It didn't look like he had, but also it seemed my resistance values were significantly lower than TheAcoustatAnswerMan at DIYAudio said was the factory default.  When he said that, I didn't believe it, because my boxes looked so untampered with.  But now, realizing how the 2+2's sound at about their central position, which appears to be right (for them) but midway in the HF attenuator, I realize I was playing the 1+1's too hot.

So this improvement is actually something I could have had with the 1+1's, I just never tried it.  BTW, the 1+1'\s require a different attenuator setting than the  2+2's, precisely because there is also a LF transformer setting change which gives the LF transformers less "gain."  So, by design, the 2+2's are set to a "central" attenuator setting, whereas the 1+1's are set much higher, according to the factor spec I have heard, just not as high as mine were set, which was about as hot as they get.

Now, I had compensated for the too-hotness of the 1+1's in a peculiar way, which negated the problem for serious listening (actually, NEGATED it too much, I always thought, but it was hard to do otherwise without creating a super tiny sweet spot, head in a vise) but meanwhile meant, for casual listening every where else, I simply got the too hot highs in even greater measure.  And it turns out a lot of the listening I do is casual listening, even from other rooms, because I like the way that works out as opposed to having background music piped in to every room--because then it's too much in your face, it's actually, as I'm quite honest with myself in admitting, a form of information reduction because quite often that's better for background listening.

The method was this: I was listening far off of the central axis of the speakers.  On the speaker axis, the sound is way too hot and even slightly brittle, and very much head in a vice.  Of axis a little, it gets a little better, but the head in a vice got better and worse.  Finally, between 18-28 degrees off axis, there's a range where the high attenuation seemed about right, but all the venetian blind effect was gone, all of it.  So it seemed the best positioning was about 23 degrees off axis, giving full relief on either side from that pesky vice.  That position was also, a bit too much attenuated at the listening position, but it was OK, especially with my super tweeters crossed in at 20kHz but contributing lower.  It was often unnerving, however, to have the highs bright (and sometimes uplifiting because of that) and then sit down and have the highs greatly collapse, even if actually being somewhat more natural but from the other side.

Well, now, I have none of those issues.  The 2+2's are just fine head on, and may be optimal at a more useful angle of 12 degrees or so off of the speaker axis.  The brightness with the level control centered is just about perfect at the listening sweet spot and everywhere else.  This is big, very big, for my casual lifestyle.

So part of this was my previous ill-adjustment, and the other part was something I'd never even imagined before.  The 2+2's are two vertical columns of electrostatic elements, and those two vertical columns are in fact themselves angled slightly from one another and from the central speaker axis.  That slight angling has many useful effects.

For one thing, when one is right on the 2+2 speaker axis, neither element is facing you directly.  As before, I believe these Acoustat electrostatic drivers seem to have a slight issue exactly head on, by themselves, but here they have no issue on the speaker axis because then you are off axis by small amounts (5-10 degrees I figure) which is right where the Acoustat panels my themselves sound best.  But then, as you move off the center of the 2+2's, you get more on-axis with one panel, and less on-axis with the other panel.  The blending effect of that eliminates the worse beamy qualities at every angle, at least in the range of plus and minus 25 degrees.  So now,  I can pick whatever angle works best for other reasons, and not be constrained by a horrible venetian blind effect right at or nearly at the center of the speaker.

Other speakers do similar things.  The top Martin Logan models have for many decades used a horizontally curved membrane.  In my opinion, that introduces many technical issues.  It was amazing that they got it to work at all, but the earliest models weren't very good sounding IMO.  Now I think the latest ones may be fine, but it's still complicated thing to do, as the mebrane must stretch and relax far more as it goes in and out.

The Sound Labs use lots of little speakers arranged in a cylinder.  That has advantages, but disadvantages also.  To get the same amount of radiating area, you need a substantially bigger speaker, and it's far more expensive to make.

Given such possibilities, the Acoustat way of simply angling the two columns a speaker is amazingly elegant and works amazingly well, IMO.  Perhaps, overall, it's the best approach.  It can't be done for more than two columns of panels, and work equally well.  Two vertical columns is the optimal number, as each can be only a few degrees off the central axis.

And that applies to the third improvement.  It could even be that the two angled columns actually produce a narrower effective "line source."  It does require a slightly greater distance from the speaker for full coherency, perhaps 4-5 feet (the 1+1's are coherent at 3-4 feet).  But at that slightly greater distance, the imaging is even better, along with more depth, etc.  The greater information also contributes to this, so the quality of the line source itself may not be much better, but it is far from being worse.  The 2+2's routinely have the spacious depth which I called layered depth that I first heard when I started using the Krell FPB 300 on the 1+1's eleven years ago.

I've continued keeping the minimum (in the back inside) distance from wall at 3 feet, actually 39 inches on the inside to the front wall and 47 inches from the outside to the front wall.  The factory spec of 3 feet in my experience is an absolute minimum you must exceed.  The distances I've listed are the greatest my living room can sustain and still be good for other purposes, as I discovered at my last party.   To make the current speaker arrangement work at all, I had to put the supertweeters on the inside, which is not the best location for them, but works well enough, and allows the 2+2's to be as far apart as possible, which is helpful in my relatively narrow room, and allows for the slightly greater distance from the speaker to the sweet spot, which is now just behind the room center (as I had discovered this year, works perfectly fine with the EQ I had been using, and even better when fully dialed in).  It took a month to find this new slightly further back sweet spot and angle the speakers just right, and then some more time to reset the midrange EQ's to notch out the slightly different LF resonances and the make the usual 2-8kHz slight softening for best sound.

And any amplifier has to work half as hard into the punishing 2-ohm minimum load, which is another big win.  Now I no longer need a 1200W into 2 ohms amp like the FPB 300 (which has now been retired, after one too many failures after professional repairs, when I have time perhaps I'll learn to fix it myself).  The Hafler 9300 sounds equally fabulous without any strain at any useable level.  The Hafler is somewhat power supply limited (compared with the Krell) but with the 2+2's it's become a non-issue.

So, it's a win-win-win-win!!!!  And now I have the incredible 1+1's to use as my Laboratory speaker (though that took clearing out all the junk, which could not have been done in months had I not already been retired).

Turntable 101

A friend asks about buying his first turntable.  I've decided to spell it out here as a permanent contribution to audio buyers.  (Can a person "live" without a turntable, and just get by with digital music.  Certainly, but I'm not going to address those questions here.  I have several turntables myself and would not give up playing vinyl for anything.)

Many people have first heard records played on an all-in-one player, possibly their own.  With very rare exceptions, such as systems made by B&O, these are not very good.  To get something really good, you generally have to buy a few separate parts, though a specialty store may be able to sell you everything at once and perform all the assembly required.  The parts required are:

1) Phono cartridge.  The best cartridges are generally either moving magnet or moving coil.  Cheaper systems use ceramic cartridges which are not very good.  There are also a few exotics like strain gauge that are barely worth mentioning except to audio fanatics (and yes, I have a few myself, in my collection of exotic stuff).

Moving-magnet cartridges are generally cheaper (though some still get very expensive) and more popular.  They generally have a warm sound that is pleasant, but sometimes seems lacking in detail.  Moving-coil cartridges are more expensive (but the least expensive ones are not that much more costly than above-average moving-magnets and still highly respected) and generally only used by the most dedicated audiophiles.  At best, they can have a neutral sound that is highly detailed.  Critics may say they have too much highs.  I myself used moving-magnet cartridges for the first 10 years of my own record playing practice, then switched to moving coils, which I like better.  One of the least expensive but still highly respected moving coil cartridges is the Denon 103 ($300 brand new).  Dynavector also makes highly respected low cost moving coils, which often have high output.  I have used Dynavector's most expensive (but lowest output) cartridges since the 1980's (now $1300, my first was a donation).  There are hundreds of other brands, and technical variations too complicated to describe in a short article.  I can't really advise on moving magnets, but Grado is a well known brand, and makes cartridges available (refurbished) for as little as $50.  Technically Grado calls their design moving iron which has some advantages over a conventional moving magnet.  Rega, Shure, Stanton, and ADC are a few of the vast number of respected brands.

As with most everything in audio, some phono cartridges, especially moving coils, get astronomically expensive ($20,000 and up).  But there are reasonable alternatives as I have suggested which work quite well down to $50, and often the ones that are not-quite-astronomically expensive are available used at much lower prices, and may still work as good as new.  Buying used may be part of your audio strategy (it has nearly always been for me, and though I made many mistakes, but I could not have scaled the audio heights any other way).  The stylus on a phono cartridge can wear out, however the best used properly will last 2000 hours of usage or more.  I have never noticed wear on any of my cartridges.  The cartridge internal damping has failed (after 30 years) but the stylus did not wear out, at least in part, because I really haven't played records THAT much in the past 30 years--though I'm playing them now more than in the 1980's.

2) Tonearm

The tonearm is the device which supports the phono cartridge above the record surface.

Most often, the tonearm is part of the turntable system, but many of the best units are equipped to mount other tonearms, and most often the best tonearms are sold separately, and mounted to the best turntables, which often don't come with tonearms.

One of my systems, a Linn Sondek LP12 Valhalla with Ittok tonearm, came as a package with the Ittok tonearm also sold by Linn (made in the 1980's by some Japanese contract manufacturer).  Theoretically, I could mount many other tonearms onto the Linn armboard, but I haven't bothered.  Instead, I have defied Linn guidelines and traditions by modifying my Ittok arm, coating it with Hockey Tape to provide internal damping.  Other famous tonearm manufacturers, like SME,  use sophisticated damping systems.  I would love to have a recent SME tonearm, which are quite expensive (up to $10k) but what I can afford to do is modify my Ittok, which is still a highly respected arm (but also, following Linn's approach, completely and absolutely undamped, which I don't think is a good idea).

I did buy a separate tonearm for one of my turntables, a Jelco, but I haven't assembled it yet, and I've used separate arms with mixed results in the past.  I would say it's the way to go, but I haven't fully gotten there myself.

Quite common on older Rega turntables, and others, are the Rega RB250 and RB300 tonearms.  They are quite good, but not as good as the best.  One of my friends has a modified RB250.

Some tonearms are linear tracking.  This is not a bad idea, but may suffer in the implementation, and they tend not to be as reliable as simpler pivoted arms.  I once loved my best linear tracker but haven't repaired it since it's second breakdown.


3) Turntable

As stated, the Turntable usually includes the Tonearm, described above.  The rest of the turntable is the spinning platter suspended by some kind of plinth which sits atop a rack, table, or shelf.

Inexpensive but still excellent turntables have been sold by Rega and Music Hall for decades, and now they have a number of competitors such as Project.  Famous old but still available brands include Linn, Denon, Pioneer, and Thorens.  Linns can get very costly when equipped with all the options, which are only recommended for Linn fanatics.

For quit awhile, there had been a bias towards Belt Drives, including the Linn and Rega and many others.  However, it now appears that Direct Drive can be made to work just as well, as can the earlier Idler Wheel turntables.  Generally speaking, 50's-70's idler wheel turntables such as consumer Garrard, Dual, and Benjamin Miracord suffered only from a slight rumble, which can be cured using much heavier bases, and then they become perfectly competitive with recent belt drive systems, and some people now prefer them.  Brand new idler wheel turntables of inestimable quality are now available for $30K and up.  Some audiophiles feel now that idler wheels, at one time cast off as junk, are the ultimate best design, but this is one of a large number of audio cults.  I myself purchased an idler wheel Lenco for future modification but haven't gotten around to setting it up with new custom base and arm.  My own feeling is there is some truth towards the reverence toward idler drive, but I don't yet have a working one myself, being satisfied with either direct drive or belt drive for many years now, and there never has been anything wrong with a top quality direct drive.  What this means is you have many choices among the fashionable and one-time unfashionable in the used market.  In the new market, belt drive is the most common at lower prices, and there are a few direct drives such as Pioneer tables for just a bit more.

Ancient idler wheel turntables can sometimes be had as cheaply as free, but may require some basic mechanical work, if you are up to that.  Garrards and Lencos are especially highly regarded.  Fully refurbished and upgraded professional Garrards may sell for astronomical prices.

Your safest used turntable is a simple belt drive, such as the cheapest Rega models.  There's almost nothing to go wrong with it, and what there is can be replaced cheaply.  I've never owned one myself, however, and I've often gotten myself into trouble with more complicated used turntables, and so often wished I had.

4) Phono Preamplifier

Every record playing system requires a Phono Preamplfier, but most often it is a circuit in some other piece of equipment, such as a Preamplifier or Integrated Amplifier or Receiver.  Quite often nowadays, however, Preamplifiers don't necessarily include the Phone Preamplifier, you must buy one separately.  Even if your Receiver already has a phono preamplifier, a separate one might be better.  Most moving coil cartridges require extra amplification which most Receivers and many Preamps won't provide, you have to check for that feature.

Separate phono preamplifiers can cost as little as $29, but those aren't very good.  Starting as little as $100, however, they can start getting quite good.  Prices go all the way up to astronomical, for which the units may be better or not.

I think the small Emotiva Phono Preamplifier XPS-1 for $209 is quite good, and relatively cheap in this category, but the full sized Emotiva XSP-1 Preamplifier (including Phono) for $1249 is even better.  In the past I've used tube preamplifiers, but except for extremely expensive hybrid models, they are too noisy for moving coil cartridges.  Though I myself have designed and built tube audio preamplifiers, I do not feel tubes have any special magic in audio anymore, but they do have a cult following, and there are people who like the way particular units sound.

The Emotiva preamps use IC op amps.  I do not believe this is a problem because they use some of the best opamps, which are now as good as anything, I believe.  But many audiophiles believe tube or discrete transistor circuits are better, and so there is a market for these units from reasonably priced used models to astronomically priced new ones.



5) Preamplifier

The Preamplifier might be a separate unit, but it is often part of another unit such as an Integrated Amplifier or Receiver.  This is the unit that most often has a volume control and selector switch.  At one time it needed to have amplification, but often that isn't needed anymore, making the name "Preamplifier" a misnomer.  I use the Emotiva XSP-1, which includes the phono preamplifier.

6) Power Amplifier

The Power Amplifier is the part of the system that provides actual power to the speakers.  It is sometimes part of a Receiver.  My first audio systems were based around a receiver, a now vintage Marantz 2270.  After using that unit in the 1970's, I moved on to "separates," a separate preamplifier and power amplifier, and I've owned over a dozen power amplifiers since then.  There is endless audiophile obsession over different power amplifiers, and yet, in most cases, very little actual difference among them, except when you may need extreme amounts of power.

Hafler was a classic brand of power amplifier of top quality for not very much money.  I've owned far more expensive amplifiers myself, and yet I now use a Hafler because it is good sounding and reliable.  Parasound, Emotiva, and ATI also make reliable yet relatively inexpensive amplifiers, and there are many other such brands.  Expensive amplifiers made by Pass Laboratories and Mark Levinson are fine and reliable, but you probably don't need them.

7) Speakers (or Headphones)

This is where it gets most interesting, because it can hardly be summarized at all.  There are many different speaker and headphone technologies, sizes, implementations, and prices, and they all have their adherents.

If you are just starting out, you are almost certain to buy box speakers which have two or more drivers in a box.  These can be very good, and different variations exist up to astronomical prices.

But there are also planar speakers which many feel are even better.  Planar speakers can be ribbon, planar magnetic, or electrostatic.  I myself switched to electrostatic speakers 10 years ago, and I feel they are the ultimate best technology.  But I could be wrong; many people feel otherwise.

Likewise, headphones can be cone, planar magnetic, or electrostatic, and I prefer the electrostatic ones, which are generally the most expensive too.

8) Assembly

Generally, you will buy a turntable with arm and cartridge already set up by a seller, especially if it is a store.  Otherwise, assembly can be a tricky and complicated business, but one with which most record loving audiophiles ultimately become quite familiar, especially if they have a separate tonearm and turntable.

Hooking the rest of the stuff up is basic audiophile work and shouldn't be that hard.

Required Total Bandwidth of Audio System

The correct answer is, at least two times the highest you can plainly hear.

Why two times?  Actually, the highest you can plainly hear is dependent on two things, the power levels involved (and we don't want to go there) and...the bandwidth of the test system.

As the bandwidth of the test system improves, so does your apparent bandwidth.

What's important is not that your bandwidth is "met," but that any further improvement in your apparent bandwidth is not achieved by turther improvements in the audio system.  Thats...as good is it needs to be.

This happens around twice the desired bandwidth, because when you have two separate sytems...the audio system and your hearing system...to achieve any particular total bandwidth over both those entities, requires each half to have twice that bandwidth.

Although my hearing goes to about 16kHz, an appropriate audio system total bandwidth is 40kHz to allow for the nominal human limit of 20kHz.

And since audio amplifiers are only one part of the system which must achieve 40khz bandwidth, it's appropriate for audio amplifiers to have at least 100khz bandwidth or maybe higher depending on rest of system.

Marantz AV8805 surround processor revealed

I've decided it's barely worth considering a surround sound processor which does not have the ability to control the target curve for each speaker.  In the Marantz line, the Audyssey App is supported by the AV7704, AV7705, and AV8805 units, and not earlier models such as the AV8802a.

My favorite audio tester, Amir at Audio Science Review, measured a Marantz AV8805 and did not like what he found in this $4500 unit:

At 4 Volts RMS at the XLR main outputs:

1) 91.2dB SINAD at 1kHz (0.0027% THD)
2) Shocker: THD rises dramatically at 5kHz and above, above 1% just above 10kHz

Also he also thought the jitter spectrum looked relatively busy compared to his favorite DAC, but not so much as to be an audible issue.

Now, in my estimation, Amir did something a bit unfair to Marantz, but often addressed by commenters.  Amir didn't bother to do any measurements at the official AV8805 Rated Output Level, which is 2.4 volts.  He said that was simply a crutch that Marantz used to get better measurements.  4V is the defacto industry standard level for XLR for consumers, and Marantz should be expected to perform well at that level like any top product.

Furthermore, in long discussion which compared the specs and measurements of other Marantz processors done by others, it appears that the 91.2dB SINAD is consistent with measurements done by others at different levels.

For example, Secrets of Home Theater found the AV8802(a) to have less distortion at 2.0V, and more distortion at 5V, which was still below the apparent maximum output.  Amir reiterated many times that he turned the level "down" to 4 volts, though he never indicated what he turned it down from.

This other review of the AV8805 didn't do a single measurement, mainly talking about features and giving subjective reactions to various discs.

After much discussion, it appears that the Marantz AV8802, AV8802A, and AV8805 have the same output circuit and cound be considered comparable.  The specs are identical, and in fact the specs are also identical for the AV7705 and AV7704.  Therefore, conveniently, a review of the audio raw performance of one of these units is equivalent to a review of another.  Therefore, despite many people complaining about Amir only doing one measurement (not true) and calling it a day (apparently true, he shipped the unit out to make room to walk around in his laboratory), we can fill in the gaps by looking at measurements done by others.

Though it's not great, I don't think the 0.0027% distortion is that big a deal either.  Sure, my Emotiva Stealth DC-1 (another product Amir does not like) measured 0.0003% THD in my testing, which is about 10x (20dB) better, and Amir's favorite dacs (I now use the cheapest one he recommended for measurement purposes) do even better, but few recordings are going to be made that well and no speakers would even come close.  Many highly regarded audio products have no better than 0.1% THD, which is often suggested as a limit to the audibility of nearly all distortion harmonics.

I think, and what unfortunately to this moment has not gotten enough discussion is the #2 Shocker.

Amir did some further investigation of this.  First, he ran a 10kHz signal and found a ginormous distortion peak approaching 1% at 34kHz.  Since this is not an even multiple of 10khz, this got him thinking it might be an aliasing artifact.  So, he runs a white noise signal and then discovers only 3dB rolloff at 22.05kHz, where (according to Shannon et all) there should be infinite attenuation from a reconstruction filter.  Acceptible levels of attenuation were only ultimately achieved near 40kHz.

Amir looked in vain for any discussion of digital filtering in the manual, or ways to select different digital filtering options.  Failing to find such, he pronounced the AV8805 flat out broken.

Such slow reconstruction filters are not at all unusual on audiophile gear nowadays.  There's quite a strong following for no filters at all (NOS).

Now the only piece that leaves me wondering about this, even after reading every post in a 24 page discussion (which was quite interesting in itself) is whether the high levels of HF distortion would essentially disappear if Amir had stuck to the Rated Output level of 2.4 volts.

Amir simply believes that such an output level is not worth bothering with, 4 volts is the defacto standard, he tests everything balanced at 4V, and if a unit can't do 4V well that's simply another way that it's broken.  Commenters suggest that 2V is more than adequate for many amplifiers and challenged him to suggest one amplifier where this would not be true.

Here's one aspect which wasn't discussed.  We can't simply tell the maximum input voltage an amplifier can use from its rated input, sensitivity, and/or rated power.  Most power amplifiers have considerable headroom.  So even if it only takes 1.5V input for such-and-such an amplifier to reach rated power, it may be able to handle input peaks up to nearly 3V without clipping.  This is a good reason for preamplifiers AND surround processors to have at least 4V output, as Amir suggests.

Amir does note that the level of the 10kHz distortion products including aliases does go down at lower levels.  And therefore, he admits, the audibility of this problem probably isn't that huge either.  But it represents unacceptible performance for a flagship product.  I agree.

But I'd also just like to know, if the 2.4v output level fixes things.  It seems OK in the AFAIK identical audio circuitry of the AV8802a, as tested by Secrets.  They measured 0.002% THD at 10kHz with 2 volt output with 44.1kHz sampling, and lower levels with higher sampling rates.  That suggests the Marantz would easily meet it's specified 0.008% THD from 20-20khz at the rated output level which Amir refused to test.

Digging a bit deeper still, Secrets also tested 10khz up to 5V with 24/96 data, with distortion at 0.0036%.  Unlike Amir, Secrets did lots of different tests with 10kHz and with pairs of frequencies.  Curiously, however, they did NOT report a test for 10kHz with 5V at 44.1kHz, which might speak to the high levels of distortion Amir saw at 4V output with 44.1kHz signal and attributed to aliasing.  At the higher sampling rates, distortion reduces at 2V also.

Looking at these high frequency and IM tests, the profusion of products does indeed look like it may fit an aliasing theory.  It still bothers me how it seems fine at 2 volts (and possible 2.4 volts) but then distortion products including aliasing products just take off.  It seems to me that aliasing products should be there all along and increase linearly.

It was a similar output level limitation that led to my finally ridding myself of my two Integra Research RDC-7's today.  When I bought these a few years ago I was hoping they'd make excellent DACs, which I could even use in my laboratory.  My first naive measurements at 4V showed such massive distortion I figured my first RDC-7 was broken.  It was only later while testing my second unit I discovered that distortion looked basically OK (just OK, perhaps about 0.005%) until 2V balanced or 1V unbalanced.  Then, distortion products began rising linearly from the noise floor.  At first I thought it must be clipping.  But clipping distortion products would rise much more quickly.  It almost seem like some kind of soft clipping was being applied.

All these high end surround processors are THX licensed.  This means they pass proprietary tests by THX.  We don't know exactly what these tests are, and what they required, because they are not public information.

I wonder if THX virtually demands a kind of soft clipping be introduced in the pre/pro above 1v unbalanced and 2v balanced, in order that amplifiers or speakers themselves not be the clipping elements.

Anyway, the way the shocking issue appears in all the information presented does not suggest that soft clipping alone is the source.  The products that appear in high frequency and IM tests look a lot like aliases in some cases.  However, the rapid rise when levels exceed 2.4v suggests that aliases alone are not responsible.  Then it appears the issue may be the combination of two things, the very slow reconstruction filter AND a form of soft clipping that starts just above 2.4V.

Knowing all this, I've somewhat lost interest in the AV8805, which no longer looks like a reference product and isn't worth the high price. (If you want reference, apparently the Trinnov is the unit to get, at 4 times higher price.)  However, the AV7705 may be OK for the features, and knowing that the limitations aren't likely ever to intrude on the auditory experience.

Amir correctly blames the lack of competition in the pre/pro world to the introduction of HDMI.  Apparently only 2 companies make the require HDMI chips, and they aren't interested in supporting small time manufacturers.

I've never liked the idea of squeezing my audio signal through HDMI, and now it appears (increasing distortion, jitter, etc) that avoiding this would have been good.  Given the current market, however, it's unavoidable.

Much of this impasse has also been caused by the myth of perfection.  Everyone wants "perfect" digital inputs and outputs to things, as these are considered "lossless."  One thing that's lost but barely remembered is openness.  Analog inputs are fully open, transparently, and not subject to being locked down by greedy bastards.

If instead of demanding digital I/O, we had stuck with analog I/O and just made it better, we would not be trapped by greedy chipmakers and other proprietary schemes.

I strongly believe that digital conversion is virtually transparent, especially at 24/96, so there would be no harm in sending analog audio signals from CD player to preamp, which could convert them to digital for processing, and back to analog for amplification.

This is the kind of thing I have done for over a decade now for stereo.  In some cases, curiously, I find the resampled analog to be BETTER than digital.  In not one single case have I found it to be noticeably worse.  Sometimes I just use the direct digital because it's supposed to be better, but I've never actually heard that.

But if reference high end products can get away with 91dB SINAD (and far worse at high frequencies), the -120dB added noise from resampling is just not a big deal, and if we still had analog interfaces there could be endless competition, even from people like me.  (I recently found a "open source" surround processor project.  It hasn't gone anywhere in 5 years, probably due to the HDMI issues.)

I had been thinking the Integra Research RDC-7's would give me this kind of analog processing.  For quite some time I was blocked by the fact that they use a snake multichannel input connector, and I never got around to testing the multichannel path because I didn't want to risk getting the wrong kind of snake for it.  Well, it turns out I could have seen the issue from looking at the schematic, which reveals that the multichannel inputs are not digitized, they are simply passed directly to the output with volume adjustments.  This is the same limitation as on my 2005 Yamaha HTR-5790 receiver.  So the glorious Integra Research would not move me one inch forwards, and possibly a few backwards.

I understand that there were some earlier processors by Yamaha and others that did digitize multichannel inputs, at least up to 5.

But to get with the program, and play with all the new surround toys, I'm still tempted to get the AV7705 at a decent price.