The Guyz Behind Mondial

I put my Aragon 8008BB back online on Sunday, after a week of A/B/X testing with a Parasound HCA-1500A and a Carver TFM-24.  I've got an RMA to return my Krell FPB 300 to Krell for repair.  I miss the sweet sounds of my Krell FPB 300 and want it back online soon.  But that's no excuse for downtime.  The good audiophile must make do as well as possible with the best that is on hand.  The show must go on.  I have no patience for audiophile nervosa.

Just last week I discovered an obituary for Paul Rosenberg, the Mondial co-founder who along with Tony Federici were responsible for the great Aragon and Acurus components of the 1990's. This was also my discovery of the audio journalist Ken Kessler.  Paul and Tony sound familiar to me I think I once met them.

Ken affirms by telling the story of the big Mondial amplifiers having been designed by Dan D'Agostino.  In an alternative accounting I find more believable, the circuit and circuit board design are credited to Mike Kusiak, whose initials are one the board.  Robbii Wessen did the industrial design.

Coming from the pen of Ken Kessler, who appears to have actually had some good times with Paul and Tony, who probably heard it from them, this may be the true story, or at least "true enough" in the sense that D'Agostino might not deny it--there may have been collaboration involved.  I still find it hard to believe he got deeply into the details, I would more imagine that Dan may have proposed the basic circuit (on a cocktail napkin or something like that) and Mike and others did the layout and fine tuning.

The Aragon is a nicely made amplifier.  However, from a cold start it sounds more than a little strident.  It appears to need about 60 minutes of warm up to sound best, and even then it may be strongly on the Yang side (like the Guyz themselves?), not unlike a lot of high end audio products.  My actual Krell, however, sounds sweeter.

I have a strange memory which may not be entirely true.  I recall someone like Tony demonstrating an Aragon amp at a meeting of the San Diego Audio Society, possibly following a Krell demonstration, sometime in the mid 1980's.

Tony heard my thoughts on the Krell and said to me, "I have an amp just like what you want."  He said.  "There's no fan, it's not as expensive, has lots of Class A power, and sounds warm like the Krell."  So he led me and a few others into a smaller room, where he played some music on the amp.

It wasn't bad but was sounding a bit too bright.  Tony agreed and wasn't happy.  Then he noticed the tiptoes the store owner had put under the amp.  "I told them not to do that," he said.  "I remember Brian bragging he could put anything on top of those."  "But not my amp," he said.  "My amp needs to get warm, and it only gets warm down on the low feet.  That gives it the warm sound," he added.  "Here, I'll just put this towel on top of the amp.  That'll help it warm up."  We listened to it later and agreed it sounded more warm and better.   But I was bothered afterwards about Tony just leaving the object on top of the amp.  "Oh, you can't hurt it," he said.  "It would shut off it it got too hot.  It's got that notch too", he grinned.  I would not recommend doing this, btw.  The temperature regulation circuit might fail to act in time, especially in an ancient unit.  Nor do I strongly remember any particular person saying exactly this, but it was something that still left me a bit worried, especially using tube amps whose misuse can be very dangerous.  Now I hear others recall Paul and Tony as care free, so it seems consistent that Tony might be pretty laid back about warming up his amp.  And it now does seem they're sensitive to airflow like this, due to relatively weak bias regulation, at least in my unit (which was an early Connecticut unit, and also features the very robust original transistors).

I wasn't that tempted.  I used tube amps for midrange and bass mostly through the 1980s, which I had obtained for very low prices, and the high cost of even value amps like the Aragon seemed a bit out of my league, though by that time I could have afforded one, even with my barely affordable house payments.  At the same time, my Citation II tube amp, which sounded wonderful, often had a frying sound, and I wanted desperately to get rid of it and get something else.  But I couldn't get myself to do that either.  Finally when I was out of a job and selling my house I sold the Citation II and basically have never much looked back at tube amps.  Maybe just because I got sick of them?  I confess now I have regained a bit of interest in tube and other amplifiers recently, coincident with running various different backup amps.  Even if I can't prove I even hear a difference, as in one session of ABX testing.

When I got finally got an Aragon in 2010 as a backup for the Krell, it had very low feet, not original, actually just 1/4 inch stick on rubberized felt pads.  I ran it on top of a flat board (NOT the carpet!).  Still, it ran much hotter, 180W at idle.  When I put the Aragon on the brass carpet spikes from Mapleshade, this elevated the amp about 1.5 inches above the carpet.  The ventilation was vastly improved, and the amp temperature went down from 135F to 115F.  But part of that change was from the bias dropping.  The idle was now 120W, which BTW is close to specs.

The bias circuit is a simple transistor, no IC.  I think it has a function that runs the amp hotter when it is cold, then slows down as the amp heats up, reaching an equilibrium of sorts.  But perhaps on the tall feet it is not getting hot enough to get going.

"I told them not to put the amp on tiptoes," Tony said in my extended memory.  "This amp, mine, is biased to sound just right on soft feet.  That makes it sound more relaxed, just like a real Krell."  It does turn out that my Aragon was early production, made in Connecticut.  This memory makes me wonder whether my copy is the same one I encountered before.

After listening to the Krell for months, and then testing the Parasound and Carver amps on my Acoustats, the Aragon sounded beyond just bright, it sounded strident.  I found one passage about 10 minutes into Pathetique on RCA Living Stereo Hybrid SACD that was screechy.

I was determined to find out why the amplifier sounded different, and badly so at that.  So I first measured the Carver amp which I will soon be returning to its owner, and then my Aragon.  The Carver looked pretty good.  The Aragon looked horrible in the right channel, precisely where I had heard the screechiness.   It was showing 0.18% THD at 1kHz and 0.84% IM.  The left channel looked ok at 0.012% THD.  These were measured, in fact I did all my Aragon measurements with the actual Acoustat speaker loads.  I did the Carver measurements earlier using an 8 ohm dummy load.  The Carver measured 0.0029% THD.  I used RMAA to do the measurements with a Juli@ card on an old PC, the measurements were very contaminated in the case of the Aragon by ground loops because it has a grounded plug, the Carver only has an attached two wire cord.  I spent hours unsuccessfully trying to fix the ground loop, first by re-routing the AC connections to a single power strip, and then by bypassing the Juli@ card itself with an external optical input DAC from Schiit.  Nothing worked, the very complicated setup using the DAC only made things worse because there is still a ground loop formed by the outputs of the amp to the inputs of the Juli@.

I was about to cut out some new felt feet and do that heat raising trick by lowering the bottom of the amplifier a fraction of an inch from a flat board to block ventilation somewhat but not completely.  But then I thought, why not tackle this bias thing head on, by adjusting the bias?  Better to run higher bias WITH better ventilation also, for the best of both.  In fact, the improved ventilation made possible by the taller feet means I can run even higher bias more safely than a lower bias with deliberately blocked ventilation.

I had avoided doing this before because measuring the bias on the left channel is all but impossible, though very easy for the right channel which is on the side of the chassis.  But since the bias problem appeared to be only in the right channel, this would be a piece of cake (well, as bias adjusting goes...it's a long multi-hour process).

So I went ahead.  It turns out it is very easy to adjust the bias on the Aragon 8008BB, even on the left channel.  There is only one potentiometer on the entire amplifier circuit board, right in the center.  It's a multiturn sealed pot which couldn't be nicer to work with.  It took several attempts to turn the screw enough even to make any difference, then I discovered I could turn it around several times to achieve the desired result.

You measure the voltage in mV across the bias resistor for every transistor.  Aragon specs put these at 12mV for the inner channel, and 8mV for the outer channel.  At first this bugged me a lot, but now I understand...you set the bias so both channels reach the same temperature.  That's actually very important when you have a stereo amplifier.  When the channels are thermally balanced they will help keep each other at the correct point.  When they are not thermally balanced there will be a thermal bias oscillation where one channel gets hotter then cools itself down, followed by the other doing the same thing.  The larger the thermal imbalance between the two channels, the greater the thermal oscillation.

Only after much turning of the bias pot on the right channel did I get around to hooking up the now quite complicated measurement apparatus, which also took very precise location of the spike-footed amplifier to reach all the necessary cables, especially the speaker load through the Canare Four Cross speaker wire.  I aimed to achieve a bias like I had before of about 0.26mV, with a heatsink temperature which would now be lower than the 135F the amp had idled at with the original low feet.

The measurements came back excellent!  The THD distortion dropped from 0.18% to 0.0048% and the IM dropped from 0.84% to 0.036%.  Once again I think the real numbers might be even better if I could remove the ground loop problem.

Now however I noticed that the left channel wasn't getting warm enough any more, and sure enough its distortion numbers had gone way up.  So I had to readjust the left channel up to get the distortion down again there, though it seemed that once I got the left channel THD down to 0.86% simply adding more bias wasnt help much.

So I removed the cover of the right side wondering how hard it would actually be to change the bias, if not measure it.  It turned out to be remarkably easy to change the bias by turning the sealed multiturn pot in the center of the board.  At this point, I didn't need to check bias, I could measure the distortion.

Finally I idled the unit for hours, and decided several times to back down when the temperature in one channel or the other exceeded 125F, or if they appeared to differ.  I tried to adjust both channels to the same long term operating temperature.

After a 3 hour idle test, temps ranged 120-130 in the right channel, and 120-127 in the left channel, with low temps in front and highest temps in the middle.

I decided that was close enough, perhaps hotter than desired...but I need the Class A for good sound.

Which is indeed enormously improved.  This is still a Yang sounding amplifier, but very clean now.  Not as warm as the Krell but a lot closer.

DSD in blind testing

Here is the abstract of the report on blind tests done on DSD in 2004.  DSD was apparently compared with PCM 176/24, and participants were unable to hear a difference.

Of course this cannot in principle satisfy anyone who takes the "listen for yourself" advice seriously.  That advice is an nothing more than an invitation to be conned by slight of ear.  If somebody bothering to make a "superior" sounding system cannot PROVE it sounds better to at least some people if not everyone, they ought to just quit.  And the only way it can be proven is through repeatable blind test results.

This is especially true if we are not just talking about one person's private sound reproduction system, but a scheme to be generally to be used in recording and distributing music.

Squeeze Me Together

I've been enjoying high rez music through my Logitech Transporter SE, recently purchased as NEW! on eBay.  Logitech released these SE units, lacking the famous selector knob with force-feedback, in 2011 at a somewhat reduced price from the original model from 2009.  Many complained that Logitech was selling off the spare parts that might be required to repair the original Transporter model, which had been ingloriously discontinued.  But few complained about missing the selector knob.  Most users used it a few times, but then settled in using either the remote control or the Squeezebox web interface or both.

Squeezebox had been an independent company, for some reason then bought out by Logitech who didn't seem to carry the concept very far.

Actually, IMO, the concept doesn't need to be carried very far.  I'd be happy (at least for a start) with a simple Ethernet to SPDIF device.  I imagine such a thing could be built to high standards and still sell for less than $100.

One thing, you do actually need to decide which files the SPDIF is going to be outputting.  So, essentially, you need some kind of "player" (which selects file, plays it, allows for pause or stop).  However, as I understand it, this functionality exists in the Squeezebox software itself, now Open Squeeze.

Now at least I've got the bill filled, plus lots of extras, with a real Transporter.  It turns out some of those extras are at least temporarily useful.  For some reason my Tact processor isn't accepting 88.2kHz inputs, even over coax or AES/EBU from the Transporter.  It does accept 96kHz inputs.  And, it turns out that 88.2kHz is quite a popular high rez format.  So as I do with all my various disc players, I take the (extremely high validated quality) analog output from the Transporter, and resample it to 96kHz with my Lavry AD10.  And, as that always does, contrary to audiophile purism, it sounds marvelous.

Sadly, at this time I don't run balanced all the way to the Lavry…none of my selector devices permit mixing single ended and balanced inputs.  I now intend a large rebuild of the Aragon 28k to turn it into a mostly-passive selector device which would take both balanced and unbalanced inputs, and send a single unbuffered balanced output to the Lavry, with polarity reverse and other features.

However, in that longer term, I should also fix the Tact, and/or exchange the bedroom and living room Tact devices (because I think 88.2kHz still works in the bedroom unit) so I can have direct digital (though I suspect it would be Very Hard to tell the difference).

Anyway, though I don't need it now, here's a discussion of how to replace the Transporter using a Raspberry PI.

You need:

1) Raspberry Pi 2, Model B
2) Hifiberry DAC+ or Hifiberry Digi+ (spdif output!)
3) 5v, 2.5A power supply micro-USB
4) Case
5) 1gb microSD card
6) Picoreplayer (software)

Apparently Hifiberry and Raspberry snap together.

In the week since I ordered the Transporter, apparently Google and Amazon have figured out what I want better than I could.  Now I'm seeing tons of "network players" that might work nearly as well or perhaps even better than the old Transporter.  I have some fondness for the actual Transporter however because (1) I read about it in Stereophile and was incredible impressed by top notch measurements in 2009, (2) that high performance has also been validated in measurements and usage by Archimago, and (3) it uses an allegedly open source Squeezebox software.  The two downsides are that resolution is limited to 96/24 (well that's the same as my system anyway) and that it doesn't support DSD or DSD128 (which would be handy to have to play-into my system through analog reconversion).  I had been thinking about getting an extra gadget, a Pono, to support DSD but it would require the new preamp to use the best-quality balanced outputs of the Pono.  Also the Pono would let me experience first-hand the moving-average type reconstruction filter Ayre has developed.  And that brings to mind a 4th limitation of the Transporter: it doesn't have optional digital filters.  (Now I can imagine a more "ultimate" network player as being one that supports HQPlayer.  However, most machines capable of supporting HQPlayer are going to be troublesome "real computers" that I'd rather not have to deal with.  HQPlayer is part of the emerging player standard of Computer+USBDAC which I am diametrically opposed to.  Somewhere I think the creator of HQPlayer might describe a DIY network player+DAC.)

Anyway, here are some of the alternative players:

Pioneer Elite N-30 (listed for $250!)
Teac NT-506 (does DSD128 and may have variable filters, high quality clocks and clock input!)
Arcam AirDAC

Great American Sound

One of my best friends worked at G.A.S. for awhile, he introduced me to James Bongiorno in 1975 (I also met him later a few times) and Andy Hefley (who became one of the principals after James left).  All very interesting experiences.  I never bought a G.A.S. product new, I was a student at the time and they seemed too expensive, but later I did buy a second hand Son which developed an RFI problem I never fixed.  Having seen my own GAS unit have troubles, and other troubled units, I often wonder whether reliability was the actual issue which brought them down, not one of many others which have been alleged.  But the split between the founder and the remainder certainly didn't help either.  At the time of my last visit to the GAS company in Chatsworth, only a few miles from where I had attended all my primary and secondary schools, James had already left the company and GAS had installed Wave Soldering machines all over (this led to their debt problem I suspect).  Wave soldering was a very new thing at the time, and can itself lead to unreliable construction if not done correctly--which may not be easy.  OTOH, I suspect they went to Wave Soldering after the regular soldering wasn't going very well either.  Andy was doing the final feedback loop adjustments on his prototype Godzilla, which that evening we played at his beach house...

Given the charismatic James, it's no wonder every G.A.S. product was iconic and worth learning about even if his associations or companies didn't last long.  The first amplifier, Ampzilla, has been said to be one of the most copied transistor amplifier designs.  James seemed equally proud of the Thedra preamplifier, which near the time of his death he said there had never been anything like it before or since and he'd never use anything else (he did also create a new generation Thedra, much different, during the 2000's).

Here's a copy of the Gassette, a flyer from GAS describing the Ampzilla and the Thaedra.

So what's special about this?  Many things have been widely copied, others not.

1) Separate MC and MM phono stages (no transformers!)

2) MC stage has servo loop (uniquely so as far as I know)

3) Line (tone) amp also has servo loop

4) DC coupling enabled by servo loops (no coupling caps!, except in phono)

5) Tone Controls with far-out hinge points (160Hz and 4kHz)

6) Tone and Volume controls use stepped attenuators using sealed metal film construction.

7) Low filter is Bessel type with 5 positions.

8) Very low impedance line outputs

Like many before his time, James insisted that preamps should have tone controls.  His final redesign of the Thaedra had the most flexible controls he had ever done (though still not quite what the Cello Pallete offered).

That's very interesting I think now.  Personally, no gear that I have owned has had easily accessible tone controls (I'm not counting fancy DSP) since 1979, about the time JB left GAS.  Before that time, I had been using a Marantz 2270 receiver which did have tone controls which I never used, and it especially irked me that the "stepped" controls were not stepped at all, they were using ordinary pots with a mechanism which made it seem like the controls were stepped.  Indeed when set to "flat" the Marantz 2270 controls were not exactly flat, as you could easily see on a square wave.  And there was no tone defeat switch.  I couldn't wait to ditch the Marantz once I discovered this.  Actually this kicked off a year of experimentation as I tried different IC op amps to replace the line amp in the Marantz.

If only I had bought a Thedra preamp, it might have been all different.

Amplifiers ARE different

Krell FPB 300 has been taken offline.  Amplifier a friend says may be the best I have, the Parasound HCA-1500A, essentially the brilliant Curl amplifier design that has lived on through upgrades in the 3500, JC-1, and beyond.  Curl knows what he is doing, we agree.

It doesn't take long for me to hear the difference.  It is brighter, more forward.  The image comes slightly in front of the plane of the speaker.  And deep images are just as deep, kind of beyond forever in the background.  The Krell has a more subtle depth, what I call "layered," which seems more real.

About the brightness too, one might consider it "more open," or even "more transparent."  But I feel the Krell is equally transparent, just more subtle.  The Parasound, to me now, seems artificially lightened, as if someone turned on the stage lighting.

I'm thinking this comes down to You Can't Fool Mother Nature.  A low bias Class AB amplifier is not going to sound like a Class A amplifier with enormous reserves.  The Class AB no matter how well designed is ultimately going to give away the cheat.  (And it could be that my 20 year old Parasound needs a bit of a tune-up, to reduce the crossover notch.)

And the regulated power helps too.

I need the higher power too.  By my meters and calculations I have already put the Parasound beyond its ratings and probably it's limits, discovering the huge musical crash that starts "Her Majesty" on Abbey Road.  The Parasound clips at a lower input voltage which my DAC can exceed, but it also has more gain so I may have been playing louder than usual too.  So without adding more gain to the Krell, the actual limits are about the same, but I can't push the Krell into clipping or near it, and I can with the Parasound, adding an extra barely noticable strain (you can't tell much in the Her Majesty crash).

But this could be all wrong.  I'm not setting up my ABX test rig to test Parasound among other smaller amps until I get the Krell back from repair.  If I feel safe about this, I may ABX the Krell then also.  Hooking the Krell up has always seemed too dangerous though it has always been in the hoped future.

This is threatening the universe of magic to collapse for sure.  But I know how to understand such things.  My suspicion already is that I won't be able to tell differences.  But that won't mean I'll concede nothing matters.  I'll just know I've found a real subjective/objective anomaly, a doorway to greater discovery.

My long term plan has always been to have ABX available at all times.  Preamps IMO should have ABX capabilities built-in, and that's what mine will when I get around to building it.  For starters all inputs should have adjustable gain using the lowest noise amplification if needed, OPA211's.

I'd been playing the marvelous Saint Saens on Reference Recordings on SACD on my 9000ES.  After a month offline, I hooked up the Denon DVD-9000 and stuck the disc into that machine to see if its CD layer was HDCD.  Indeed it was, and I played that.  Did not sound anywhere near as good as the SACD.  Grainier, lumpier.  I'd never heard that same grainyness on the Krell.

So the Parasound is making SACD sound relatively better, and R2R sound relatively worse.  I'm also suspicious of the CD layer on that SACD.

Vintage, and why Magazines cannot be Truthful about It

Well, Stereophile is "trying."  Art Dudley is a vintage equipment expert who talks about specific equipment a lot (like Quad ESL 56's) but for October 2016 writes about vintage in general.  And it's mostly a bunch of nonsense, as is what I'm going to say.

I guess you could say I'm a vintage guy also.  I have systems filled mostly with equipment from eBay and Audiogon.

This is NOT because I have any particular love of a era-specific vintage sound (though I am interested in certain things, such as the sound of 1-bit converters doing DSD which I think is "the real thing" not obscured by fancy sigma delta DAC operations, as described here:

I suspect many delta-sigma converters with a DSD input might not be switching all of the switches from 0000 to 1111 at the same time, which would emulate a Sony-type, late-Eighties single-bit converter. It might sound a little weird, but a simpler method would be to internally translate DSD into short word-length PCM patterns, taking advantage of the dynamic range of the 5 to 6 switches (partial analog, as it were).

No I buy pre-owned equipment for several obvious reasons:

1) Affordibility!  The Krell Amp I paid $2700 for originally (actually the tab is about $4000 so far, including the first repair in 2010) cost $9000 when new in 1998.  But forget that, the most obvious brand new replacement now costs $36,000 (the Momentum amp in Stereo)!  There's been enormous aspiration inflation in the super high end.

I'm now faced with needing the second Krell repair.  I'm partly hopeful that they didn't fix it completely the first time (I think they didn't bother much with the left channel last time but did great job on the right channel) and once they've rebuilt both it should be more reliable longer.  Often this happens...once something is finally repaired completely--often taking multiple attempts--it then goes for a long time without needing more.  But you have to be persistent and willing to pay.  Looking at the alternatives...I will need to pay.

Sure there are other $6000, say, brand new amps which would fill the bill also.  But they wouldn't have the ingredients of the FPB design in the slightest.  No Class A operation to hundreds of watts.  No zero feedback output (or so I thought until today...now it appears there *is* feedback around the output drivers...so this means that the Krell has nothing on Pass Labs or the Class A Levinson designs...which I will may someday consider).  No servo loop DC coupling.  No full power regulation (Pass's XS 300 has that...for $85,000).  I'm not even sure D'Agostino's Momentum offerings have all these features*, and maybe I could live without some of them, but I don't know, I also think these features are part of a very special magic I need to drive my Acoustat speakers.  (*I haven't seen that the Momentum amps use something like plateau biasing, but if he says they are Class A they must.  Pass, of course gives you more like old fashioned Class A with his XA amps...with the bulk to match the wattage!)

2) Selection.  A lot of great stuff isn't made any more.  Or if it is, it's just way out.  Such as, say, direct drive turntables.  Some believe this is a better approach.  (Actually, I merely think it is interesting and might be better in some ways.)  OK, I could buy a VPI direct at $30k, or a Continuum at $90k, or a Rockport at $150k.  Gone are the days you could get TOTL Pioneer for $2500, or a decent Sony for $500.  But many of these classics are still on eBay, sometimes working sometimes not, for about those same prices, and some of them can be fixed.

3) Buying experience.  I do know and understand the high end audio buying experience.  I've bought my first high end item, a pair of Infinity Electrostatic Headphones, at Mel Shilling's Music and Sound in 1975.  And that was after years of visiting audio salons in Woodland Hills, CA (where there were 3 major high end audio stores when I was in high school).  I've been to high end salons countless times, though less recently.  I even worked in a famous store, Audio Directions, for a year (as a technician, but of course everyone in a store is also a salesperson).

And...I hate it.  You have the limited selection.  You have the factory mega high prices.  You have to make a decision in a limited time, and you can't really tell anything, even when you have very patient and unsnobby salesperson (I will concede audio salespeople today are generally far nicer to me today than in 1974).

I will have to concede that in it's totality Im not totally fond of the vintage buying experience either.  First there's the enourmous fear (will it actually arrive?  will it be rusty junk?  will it smell like moldy tobacco?).  Then, sometimes some of those fears come true.  Then, even when they don't come true, you have only have a few carefree years before needing expensive or even unobtanium repairs.

But the high is looking at the stuff in endless pictures, reading about it in endless blogs, arguing about it, showing it off, taking your own pictures, and, sometimes, actually using it.

The real advantage to the vintage BUYING experience is that you can take your time, and nowadays you can do it anywhere--because you can do most of it online.  To be honest, I don't miss the brick-and-mortar salons that much.  Even with nice salespeople, they are still salespeople.  I'm sorry, but I prefer to think about things, and especially   make buying decisions...Alone!  And back to #1, the essentially unlimited choices in vintage.  BTW I've found a new aggregator HiFi Shark who looks in all the online sales forum, so now almost anything can be found somewhere.

(Now for some people, Vintage is friends, connections, local places.  Those are the lucky people who live in such places, or perhaps just the well connected and informed in ways that I am not.  But for them even more than me, Vintage is Everywhere!  Everything!  And not just in stores.)

I guess the bottom line is I'm going to be sticking with the way I do things.  I do sometimes buy new equipment when it seems to meet my needs best.  I promise to stretch a little to do so whenever reasonable.  It can be better to pay now rather than later.  But generally I will be making up my own mind, primarily with my own head, ears, and heart.

Meanwhile, the magazines are supported by advertising primarily by manufacturers but also dealers.  So of course they are going to paint a picture where every day, in every way, things get better and better.  In high end audio, this doesn't mean they really get better, it mainly means they get more expensive.  There is no scientific proof that audio amplifiers have gotten significantly better since Frank McIntosh sold his first amplifiers.  That was when amplifiers were finally made to be good enough, and good enough in 1946 is still good enough now.  There is no scientific evidence that in a blind test you'd be able to hear the improvements since 1946.  In a blind test you might well prefer the 1946 model.  This is not to say there isn't a kind of technical progress in many ways.  Things have gotten a lot better in a lot of ways.  But, by and large, these differences don't matter, except to those who believe they want them.

Here I follow the audio objectivists.  While subjectivist audiophiles can paint one of many narratives (which have changed and folded and recombined etc etc) following the progression through technologies of amplifiers, turntables, etc.  The worst subjectivist audiophiles can be bullies just as bad as the worst objectivists.  And the bully subjectivists are usually worse, IME.

The fact is that these narratives are so wrong they even go circular, as has now happened with turntables.  What was old and abandoned long ago, is now The Thing all over again.

This just shows all these narratives are mostly crap.  Audio is more like women's fashion than a march of progress forwards (though has happened also, mostly in objectivist ways).  Audio has not only been good enough to be greatly enjoyable for over 100 years, but in many way certain things cannot be objectively proven to be better than they were decades ago.  Amplifiers are in fact the key example: almost all differences which people think they hear in modern amplifiers can't be shown to be there in blind testing.  I don't follow the objectivists completely (nor do they themselves except in rhetorical fantasy) that this means we conclude there are NO differences.  I would take it as meaning there may be little-or-no differences.  Far less important differences, if any, than people imagine.

Now in transmission systems, we've gone in both directions, with MP3 and Cassette Tape being convenient but unmistakeably low fidelity.  With 16/44 uncompressed digital audio we get to stuff that even in 1983 is so transparent most people would have to strain to hear further improvement (in blind testing...not their imaginations) but some can do it...as has been shown by recent DBT with different digital filtering technologies (or at least there has been one demonstration of such...by Meridian selling MQA...perhaps we should see if there is replication...).  I believe it is true and proven that digital filters can make an audible difference in digital, though it is less than most who believe they hear differences generally imagine.  One of the cornerstones of HDCD, for example, is that the digital filter can be varied by the producer to take on one of several shapes of different utility.  And now MQA is clearly a similar kind of thing but far more advanced.  And the DSD's are a different kind of digital that remove the digital reconstruction filters, but introduce their own formula of noise shaping as a substitute, which categorically you could describe--as yet another kind of filter.  So that's the fine edges of the digital stuff.  Will there be fine edges on the fine edges?

When people like Cookie Marenco say that if you can't hear the difference between a copy of a CD in the original...I'd pay pretty good money to see someone do that in DBT.  I've run very careful DBT on subjectophiles and seen nothing that would surprise an objectivist.  If what Cookie says is not pure nonsense, it would be very very hard I believe.  Marenco runs a production company that makes recordings in DSD, and someone I'd put with the crazies in subjective audioland, though not perhaps as a successful audio recording entrepreneur--which is what counts as they say.  There are very smart DSD pushers like Hiro and Miska on Computer Audio and I'd admit I've learned from them, but also Archimago who I consider far more fair minded.  DSD64 is a waste it appears.  DSD128 looks like a decent performing high rez format, though is it really necessary compared with well done space conserving 24/96 done by the likes of Ayre QA-9?  And I will say flatly that I personally believe that it will always be possible to find a way to make 24/96 sound as good as the best of anything (in equal number of channels, etc), possibly with a few more tweaks we haven't thought of yet.  But if a producer chooses to use DSD128, well whatever, I'll analog convert it to 24/96 (when and if I ever get suitable equipment) which I believe can always be made sufficiently transparent for human audition.  There's probably a reason why Sony hasn't, for example, published DBT showing the superiority of DSD, 1-bit, or any of their digital technologies as far as I know.  It can't easily be proven, and basically hasn't (I reported on high end German DBT which failed to show difference between comparable DSD and PCM--hope I find more about that).  Demon Sony Digital seems anti-consumer in endless ways, such as the inability to do any downstream digital processing.  Are we to throw out our digital EQ (and for me, crossover) just for some fantasy that might be so inaudibly different as never to be established by DBT?  The whole thing is much like a Super Audio extension of DRM, DRM which doesn't allow you to modify anything, even to adjust the presentation size to fit your room.  If I'm wrong, then at best DSD64 was a premature broken format, with even DSD128 available before the penetration of suitably capable technologies which may one day give us DSD128-compatible DSP---and or a case of requiring endless upgrades to finally realize the alleged potential--not to mention costly equipment).

So, given all the lies, crap, goodness from the beginning, and background progress not particularly aided by the subjectivist occupation of Audio, there is absolutely no reason you cannot do your own thing, just like me.  Though of course I believe you might do a little better after reading me.

Handling the DSD noise

I was noticing that the Krell FPB 300 was clinking a lot, yet I was only playing at a fairly soft level, playing the RCA Living Stereo hybrid SACD of Wallenstein Symphony of the Air.  I think that these old analog recordings often contained HF noise that was ignored, and in DSD encoding there is no low pass filtering.  And I think this range of early manufactured SACD's and/or this specific one is notorious for HF noises as described in the pages of The Absolute Sound.  HF noises that can potentially be destructive to amplifiers, etc.

So I thought, what if I low pass the Krell at 20kHz?  This might not be a good idea in general, but since I am adding in the supertweeters at 20kHz, it might be the right thing to do anyway.

So I tried it, and even ignoring the great reduction in amplifier clinking I thought it sounded better too.  More relaxed sound.  BTW I noted in a wonderful performance of the San Antonio Symphony on Friday that live music has a particular relaxed sound that is often more like LP's than CD's.

I fine tuned the adjustment and the pink noise is flatter than ever, and notably dipped at 20kHz when I turn the supertweeters off (it was a bit like that before also).  So that the high pass and low pass match I now have them both set to 12dB/octave, which would increase the supertweeter because it used to be set to 24dB/octave (LR) with two LC filters.  But I didn't delete the second LC filter, I moved it down to 9kHz because the first filter seemed to run out of attenuation (maybe or maybe not, I'm not sure the meaning of the -15dB setting when using LC, does that limit each filter to -15dB attenuation?).  The supertweeter also has a notch 4-6k to remove any metallic sound.  AND it has it's own built-in filter which I've never really accounted for, I think it's around 10kHz.  But you see my filters have already attenuated it greatly by then (and perhaps my second filter is redundant).

On the panel DEQ, the HC (12dB/octave presumed for HC, this is a cut!) seemed to cut less highs (by the graph) than H6, which would be 6dB/octave.  That's true, the shallow filter would have to start
earlier and cut more prior to 20kHz than the 12dB octave.  One more reason to prefer HC to H6, though one might argue a 6dB octave cut would be less sonically harmful.  In a quick audition I couldn't hear the effect of either filter, though it was my impression from extended listening that applying HC/20kHz to the panels sounded more relaxed.  Actually it's hard to know the true acoustic cut because the speakers themselves roll off as steeply around 18-20kHz.  Meanwhile the amplifier is pushing harder and harder due to the capacitive load.  Certainly doesn't make sense when a supertweeter is covering the range with wider dispersion and response to 35kHz.

I had been planning to just play this disc in CD mode to keep my amplifier from blowing up, but it sounds so much better in SACD, and now I can play it at +3dB on the Krell (since the 9000ES output is a bit low) at it sounds fabulous.

I'm wondering how much time the amplifier will run before the next repair.  And how it will go when it goes next time.  It is clinking more.  That could just be loose panels, but when I sturdied the amplifier current draw, the clinking is generally after-the-fact but often does follow high current use.

The way the Krell works is that more bias current is demanded based on a comparator, so linearity is always maintained.  However, when the amplifier ages, more current is demanded more often until fixed limits apply.  So you get sharper bursts of high energy, which leads to more clinking.

I'm keeping the 18 year old Krell plugged straight into the wall with the factory power cord which is quite robust.  That way I'm operating by the required standard for safety listing.  No more monkeying around with switches and current testers.  However it goes now, it should be safe.  It could go for a another decade or a minute, nobody knows.