New Time Alignment attempt

I tried the new Hantek scope, and tried to do an alignment using my calibrated microphone, held by a vibration isolator and a music stand (it might or might not have been better to have the correct adapter to mount the mike directly to the stand).

I could do a fairly good job aligning the sub and the panel on each side, but problematically I could not set the delay differently for each channel on either the subs or the panels.  I do not understand why, it looks like this adjustment should be possible somehow but it isn't, even on the subs, for which the DEQ is in dual mono mode.  So I had to pick a compromise, which still shows pretty good blending of the panel signal into the bass signal.  This is not entirely unambiguous on the scope at the finest resolution of 0.01ms.

With the super tweeters, I could not get a clear signal off axis or more than 2 feet from the super tweeters.  (On my iPhone using Analyzer from Onix3, I see the supertweeter output clearly from 4kHz to 20kHz...the former greatly attenuated fortunately.)  So I could not do the alignment other than by guesswork.  I could hear some differences using the single positive pulse (dirac, I think it's called) test signal, but I didn't know which was more like correct in the critical range of 6.26ms to 8.5ms relative to 10ms added delay for the panels.  Sometimes I preferred the former but recently I've switched to something like the latter, which I think is more correct. )

I think this was partly because the Hantek scope I has has a fan, which was creating enough low level noise to possibly mask the supertweeter pink noise output.

Previously, my old fanless Tektronix didn't seem stable enough anymore.  But as it turned out, even with the Hantek, I needed to tap off the preamp signal to the trigger input.  I could posslby get the Tektronix to work for this work, or get another fanless scope.

Italian Tuneup Fixed the Krell !!!

[update: sadly, it appears the Italian Tuneups I've been performing don't have long lasting effects.  My current plan is to workout a warmup strategy, that each time gets over an apparent thermal trouble spot that seems to be the crux of the problem, but once the amp gets into a funk it seems non-trivial to keep it going long enough to warm up without shutting down first.  Now I've come close to what originally worked with the first Italian Tuneup, and it doesn't seem to work into the 8 ohm load I figured I could keep connected in parallel with the switchbox, making it a trivial procedure to do the warmup.  So do I try different power levels, different frequencies, different waveforms, ramping up or down, or a different load?  What gives the best "kick" to get started?

So, the glow has faded, but the fact that it worked has indeed opened up a window of understanding and a window of opportunity I'm currently still trying (and tiring) to figure out, so this is still a work in progress on December 12, 2018.]

This may be my discovery of the year, or maybe the decade.  I've been struggling with Krell FPB idiosyncrasies for that long too (though, it has been worth it, see a future article).

It had gotten to rock bottom: the amplifier wouldn't even idle, with shorted inputs, and nothing connected to the outputs (which, with no signal, is fine for this amplifier).   It would just shut down for no apparent reason, in as little as 5 minutes after having been turned on.

The problem had started just one month after it had been serviced for the 3rd time in two years to fix this problem, twice by the Krell factory, which had finally refused to work on it anymore, and then The Service Department, recommended to me by Krell, after which it had worked more perfectly than ever before.  For one month.

All the capacitors have been replaced in these last two years.  Hundreds of capacitors.  And many other parts, many other repairs, and a few upgrades.  I approved every repair and all but one suggested upgrade.  (BTW, it might not be a good idea to perform Italian Tuneup if the amplifier needs other service.)

After I confirmed it was working perfectly after a month of simply listening to music, I brought on the ABX amplifier switcher, to allow me some ad hoc comparisons with the Hafler 9300, designed by the creator of the Acoustats.  I did a few casual nonblind AB comparisons, the Krell seemed fine, and the winner by a tiny margin (if I were actually hearing a difference at all).  But then a week later, when I wasn't doing any AB comparisons, I played a high resolution digital recording, the HRx sampler, at maximum "standard" level (defined in an earlier post).  And curiously it shut down during a softer passage just after a louder one.

Over the next month I performed a few dozen elimination tests, to try to determine if the problem was in the amp or something else, and then which channel of the amp, and what conditions caused the shutdown.  Several times before, it seemed like I had cured the problem, I even played the very same recording that made it shut down the first time since the last service, and it sailed through just fine.  But then it shut down later playing something else.

Now, with the amp only able to idle for a few minutes before shutting down, I thought certain it would need professional repair again.  I'd never be able to sell it for a decent price like this, and I didn't really want to sell it, but I might wait until next year so as not to be putting all my money into this amplifier, just my "surplus."

Or maybe, I'd try to fix it myself.  I was beginning to think about that again.

When I was in my teens (many decades ago), my best friend had a Pontiac GTO.  He loved it for a few years.  When I was riding with him on the freeway one day, we were taking an ad hoc trip from Los Angeles to San Francisco, he floored it for a few seconds.  "Blasting out the carbon," he said, "keeps the engine running smoothly."

I'm not sure if that was the proper technique, but there is something just like that called an "Italian Tune Up" which consists of something like full power operation for some period of seconds or minutes.  And done correctly, this does help remove carbonized deposits from valves and other things.  It's a widely recognized technique, especially applicable to cars that are only driven gingerly on a regular basis.

I decided last week I would try something like this on the Krell.  I waited for the weekend to get it all set up properly.

Into the living room I moved my Sound Technology 1700B, a digital Rigol oscilloscope, and a never-before-used "2000W" 4 ohm audio brake load from China.

The connections from the Krell merely needed to be moved around from the ABX box over to the Sound Technology analyzer, and merged with connections to the load which was newly terminated with Furtech locking bananas.  So it was two sets of locking bananas plugged it to each of the two balanced signal inputs of the analyzer, externally loaded with 4 ohms.

I first ran the apparently (and now proven!) dysfunctional left channel (overheating, anyway, I did not previously know this channel was wrong, but it was clearly running hot, and depending on the experiment the shutdowns seemed to predictably occur at a particular warm up temperature...around 135F).

I ran the channel with a full power signal, just below where visible change to the sine wave seemed to be occuring, which turned out to be around 705 watts into 4 ohms.

I measured distortion as 0.18%, a bit higher that I'd hoped.  Everything seemed fine for 4 minutes, and the left channel was barely heating up more than the right.  Then the shutdown occurred.  The temperature had risen a mere 14F from 78F to 92F.  Like nothing.

Now that test, I had presumed, would exercise the output function maximally, the emitter say (I'm not really sure...in fact I'm not sure this makes any difference, the current through the transistor always occurs in certain proportions, regardless of whether any flows into the load.  This is something I don't know.

But I decided to try a different approach.  After the amp had apparently cooled back down to ambient (I completely shut down the amp at the breaker), which required only a hour or so, I had planned to wait another day, but I was so excited by the prospect, and seeing the amp was now back to ambient, I plunged ahead.

I figured the "worst case" for heating the transistors is the famous 1/3 power output.  Into 4 ohms, the maximum power (before true clipping) is 900+ watts.  So 1/3 power is 300 watts into 4 ohms.

I miscalculated this as being sufficiently close to 30V to just go ahead with 30V, which is full scale on the 30V range of the analyzer.

Anway, that's what I did, I ran at 30V.  And I saw the left channel heat up faster than I've ever seen it heat up before.  More than I imagined it could do.  I was afraid it might shut down at any moment.  But it kept going and going and reached 175F (from ambient 78F) in 15 minutes.  I chose that moment to turn the amplifier off.  I did not want this test to potentially cause problems (though, actually, I've seen it go to 207F without apparently changing anything, and I believe it has a "kickdown" program at 180F which reduces the bias automatically to reduce the temperature).

Well, this was clearly different.  It had clearly blown past the old 135F where shutdowns were occurring.

I let the amp cool down to 160F and decided to just let it run at a maximum sustainable level for hours, partly as a test and partly because this would be another kind of exercise which might help.

I chose the level of 10V or 25W.  At that output level, the temp actually rose a bit from 160F to 170F or so, but I just let it keep running, and it ran for 3 hours without issue.

At both 30V (225W) and 10V (125W) the THD measured 0.07%, which appears to be what other people have measured into 4 ohm loads.  The specification is 0.03%, which appears to be for 8 ohm loads and using the balanced inputs (which I was not doing since my ST has only unbalanced signal output).

I then let the amp idle and cool down idling overnight.  It was still running ok.

The next day I played music at loud and medium levels for 2 hours, and again in the morning for 2 hours, and it still ran OK.

It is, apparently fixed.  And just as important, if the problem comes back, I know what sorts of things I may do to get it going again.

I DO have some ideas as to how to prevent the problem from coming back, as well as what ultimately causes it.  Before this "repair" I wasn't even exactly sure what channel was causing the shutdown.  I suspected the left channel because it runs hotter mostly, and shutdowns seemed to occur at certain temperatures in that channel.  Now, since I only treated the left amplifier channel to make the problem go away, it appears quite conclusively that it was/is the left channel of the amplifier at fault, and nothing else.  And somehow, certain kinds of operation cause it to change in some way that makes it start failing.

If all I do is Italian Tuneups, I don't really need to know the underlying cause.  But my most intuitively satisfying theory is that a certain power transistor on the left side is prone to something like carbonization.  It may have some cracks or grooves which get carbonized under some circumstances, causing an internal short which makes it misbehave.  This carbonization begins to vaporize or disperse when the heatsink temperature is around 135F.  If the amplifier isn't being played hard enough, the computer inside decides this looks like a serious problem and shuts the amplifier down.  WHEN the amplifier is being played hard enough, it quickly blasts through this carbonization before it has time to look like a serious problem to the computer, setting the stage for perfect operation until the carbon builds up again.

I suspect the computer program is being too paranoid, and be more tolerant of slight deviations from ideal transistor behavior to allow the normal decarbonization to occur.  But this suspicion is irrelevant for the forseeable future, because re-programming the computer is nothing I am going to have the time or expertise to do for a long time into the future, if ever, and it is fraught with risks as well.  It is quite likely the "paranoid" computer has saved the amplifier from certain destruction many times, and would be unsafe to operate with much less paranoia.  So meanwhile I simply have to live with the paranoid computer program, perform Italian Tuneups as needed, and try to figure out how to slow down the carbonization (or whatever) build up process so I don't have to do an Italian Tuneup often.  Though, in principle, I could do it ever day before serious listening.

I have many anecdotal experiences that suggest that playing very high frequencies, such as from high resolution recordings, or from FM tuner outputs digitized at 24/96, is an issue.  I noticed this long ago (because of shutdowns occurring while playing FM tuners at modest levels, and even the recent series of shutdowns which started when I played a high res recording), and decided that since I use supertweeters highpassed at 18 or 20kHz, I might as well lowpass the Krell at something like those frequencies.  And in one set of experiments, years ago, I decided the system sounded better this way (with the Krell lowpassed) and even measured better.  Now I never have much confidence in the reliability of my subjective "tests", and later measurements showed no measurable advantage (the Acoustats roll off steeply at 18kHz anyway, so it "doesn't need" to be lowpassed electronically to blend with supertweeters starting at 18kHz), so I had not been doing this recently.   But there is also little point to trying to drive the Acoustats higher than 18kHz because they have a strongly capacitive load which reduces impedance down to less than one ohm at the highest frequencies.  This is problematic for an amplifier which tries to run Class A for everything, and has a bandwidth of 300 kHz.

Another line of resoning with fewer supporting anecdotes is that operating with the ABX tester, and running the Krell with no load while still getting an input signal (the way I have been doing it) so the other amplifier can play the speakers, is not a good thing.  I was told (by Steven at The Service Department) that you don't have to connect speakers to the Krell.  But I did not specifically ask about continuing to run an input signal into it.  With many Class AB and true Class A amplifiers, this would not be an issue.  But it appears the Krell computer (the so called "anticipator") sets the bias level in large part based on the input signal, not just (or at all) how much actual current the amplifier is delivering.  I have seen some evidence that it runs hotter with no load attached than with the speakers attached when there is signal input, as any engineer would expect of a Class A amplifier because if energy isn't being dissipated in the speakers it has to dissipate within the amplifier itself.  Now with a real Class A amplifier this is not a problem, as the amplifier normally idles at the highest bias and has to be engineered for that.  But with a plateau bias amplifier, it depends on the programming of the computer--was it programmed to allow sufficient thermal margin in case the speakers are disconnected?  Now, I've actually done this many times, and the Krell did not seem immediately to be adversely affected.  It might even be, in a sort of Italian Tuneup kind of way, beneficial.  But because it is a highly suspect unknown, with a few supporting anecdotes as well,  I've decided that before doing more ABX or similar blind testing, I'm going to have to modify the tester to switch in a dummy load when the amplifier is not playing the speakers.  I believe this is dooable, there is already a 1k or similar load switched across the unused amplifier by the switcher.  I just have to rewire that 1k load as a connection to a external dummy load.

I'm going to fly with those two "preventative" ideas for now, and try to remember relevant playing details when the next shutdown occurs so I can more and more reduce the causes of the carbonization or whatever is occurring.

UHOH

All was fine on Friday evening, I started off playing music with gusto, quickly cranking up the right side to 145F.

Then, I wanted to do two things.  I wanted to take the historic photo showing the Italian Tuneup setup.  I'd cheat by actually measuring the right channel.

Now, just before, the amp had been heated up.  But I wasn't paying attention to the cooldown.

I got the right channel going, and was just about to measure, when, the Krell shut down.

I was thinking, maybe the right channel needs the Italian tuneup too???  (At the end of the day, there is no clear evidence for this.)  But as it turned out, the now idling left channel was right at that troublesome, for idling, temperature of 135F, where I'd seen the lions share of shutdowns and most of them while idling.

So I switched to measuring to the left channel, pounding it with 225 watts for a 175F warmup, then a slight cool down, then did a lot of interesting measurements.  The distortion remains at 0.07% at most levels at 1kHz, until very high levels.  At 20kHz, 1 watt or less do measure 0.07% as well (all into four ohms btw).  But cranking up to 100w the 20kHz distortion begins to rise.  At 600w, 20kHz distortion has risen to 0.4%.  (Recall, the Pass Labs XA200.8 has 1% distortion into 8 ohms at 20kHz at lower voltages.  I'm testing the Krell with higher power and tougher load.)

So now the left channel had been tuned again, so I switched back to the right channel (a process requiring turning off the amp) and now, to keep things going, I also hooked up the left channel for music playback.

Still, I got to shutdown before I could get a good run at 1/3 power.  Once again the left channel was at the suspicous 135F.

I'm thinking to do this right I'm going to need to run both channels simultaneously.  I have more dummy loads on order.

And maybe the thing is also, no power cycling when hot, and messing around.  Just blast to high temperatures by playing robustly, then keep playing--never pause.  Pausing to mess around messes whatever up.

And so, the waste of time continues...

Back to the Hafler 9300, sounding better and better.  I kick it off at exactly the same level I had the Krell at (with my 0.1dB level matching) and it sounds...wonderful in every way.

But later, once the Krell had cooled down to 85F, I decided to try exercising and measuring the right channel of the Krell.  I could pop off a quick 1/3 power burn and then measurment before the left channel gets back up to 135F, I thought optimisically.

So I ran the right channel at 225w for 15 minutes.  The back right heatsink was 163F while the power regulator heatsink in front was 173F.  Strange.  I proceded to do distortion measurements at different powers and frequencies.

The right channel seemed a bit lower in distortion, 0.06% at 1/3 power at 1khz, and basically 0.06% at other powers.  When I tried 20k, I got half the distortion in the other channel, 0.2%, at high powers.  At lower powers, 1W, distortion at 20kHz was only 0.065%, and at 10W it was barely higher.  So, slightly better than the left channel (despite running cooler) and especially so at 20kHz.

I kept testing for awhile, and strangely the left channel was not pressing upwards to the tricky 135F.  I decided to let the amp idle overnight.

Overnight, something amazing happened.  The problematic left channel is idling identically to the right, at 115F instead of rising to 140F as I had been seeing recently (when it didn't shut down at 135F).  I had only seen that after the last repair by The Service Department when the amp first returned.  Somehow, it seems to have fixed itself through the latest exercise, and even more strange, it seems to have been the Italian Tuneup on the right channel that has fixed the left channel, despite the channels having their own separate computers.

This is still a work in progress, I hope, and not a clean slate as might be hoped, but it continues to seem like the Italian Tuneup is effective medicine that does gets the Krell working better.  For now, I've decided to let the perfect 115F idle in both channels run for 24 hours or more so that it gets remembered.  It's remarkably free of clinking also.

Sunday Nov 18

The Krell has been idling without fail since Friday at 11pm after my last set of full heat stressing and measurments starting from cold (because otherwise I was hitting shutdowns, seemingly still left channel related because the suspicious 135F temperature there, but maybe not).

After that, curiously now, the idling has returned to the wonderful performance first achieved a few months ago when I got the amp back from The Service Department.  That was the first time ever (after 3 previous servicings by Krell factory) that the left channel idled normally at at temp below 150F.  Going way back, even after the first Krell service, the left channel shot up to high temperature quickly, hitting some stop it seemed at 180F, and then bouncing back down, then back up slowly, and so on.  I figured there was a bit of noise or something that kept the bias going up until the kickdown, presumably at 180F as with the earlier KSA "S" models like the KSA-200S, after which plateau 2 or one was enforced, until the temperature cooled enough to re-enable all bias plateaus, then off it went, and so on.  That's the way it was from day one in 2008, predictably making clinks on each kickdown.

Actually, though, Krell in their second servicing fixed the endless rising, but the heat was still too high at idle, nearly 160F.

But isn't this strange...exercising the right channel seems to have fixed the bias level holding in the right.  After 38 hours of idling, the left channel temperature is 121F in the "backmost" (frontmost in my current cooling optimized backwards setup) heatsink panel, 123F in the middle panel (where airflow is slightly reduced due to the sack of digital components upstream), and 117F in the front "power regulator" heatsink.  When the middle panel is warmer in the current setup, things are working good.  When the "back" panel is way hottest...that's what the weirdness used to look like, the back panel weirdly hot and causing the weird thermal cycling.

So it looks like both channels need the Italian tune up, even if it looks like only one is bad, they seem to have some weird connection, or perhaps did this time.  Actually, this narrative ignores that I did do some more testing on the right channel, just not the full warmup thing.  Maybe that's what cleared up the carbon.  But, anyway, I think two channel tuneup and testing is best and plan to do it that way in future.

The right channel panels, turned to have the lesser airflow, are nonetheless all 116F (uniformly for both back panels and regulator panel).  Note that the cooler regulator panel would be cooler too if on the other side.  So this channel is drawing less bias somehow.

My best theory that the funky transistor which sometimes goes weird around 135F otherwise has a slighly different characteristic that the DC feedback of the other balanced half fights against through the servo.  Given such a thing is happening, it's a wonder that distortion still measures about the same, 0.07% instead of the 0.06% of the right channel, and a bigger difference at 20kHz, 0.4% vs 0.2%, all into 4 ohms.

After playing some loud music for 60 minutes, I let the Krell idle for two hours.  It did so without issue, with the left heatsinks returning to 141F.  This suggests there is a bit of problem left, but if generally the Krell is started with some robust playing, to heat to 140F or so, it will run OK after that, and if not, it needs another Italian Tune Up.

Why Keeping the FPB Running is Worth It

For what it's worth, no amplifier I've ever tried in my Acoustat system has sounded better to me, in sighted tests, than my Krell FPB 300.  In blind tests, my results so far align with the literature of blind testing of amplifiers: I have not been able to provably hear the difference between the Krell and my other very fine transistor amplifiers like the Hafler 9300 and the Aragon 8008BB.  Nevertheless, I do not take these blind tests as conclusive either, and I accept that I like the Krell sound best, just not provably.  My audio hobby is emphatically not only about listening to music, it's about learning about many things, including audio amplifiers and how they sound, and even if they can be distinguished sonically at all.  That last bit is actually Audio Science, something I confess I do very little of.  But I like having the equipment for doing so.  And one place to start is with the comparison amplifiers of similar quality, but very different in design and construction, so my top shelf collection, of which my Krell is the crown jewel, the most expensive and exotic--and with lots of stuff to show for that.  It's nice having at least one of those--and I believe mine is still up among the best.

I do want the best sounding amplifier for my system to listen to music anyway, even if it's only that I think it might be the best sounding amplifier, without any proof it actually is.  Then with this "best sounding amplifier" I can compare with other amplifiers and those comparisons become meaningful and relevant.

It has these objective design attributes:

1) Class A operation (through Plateau Biasing anyway), so no crossover notch distortion and less distortion generally.

2) Very low distortion, including into 4 ohms or less, and up to the highest frequencies.  The spec is 0.03% at 8 ohms.  I myself have measured 0.07% at high power levels into 4 ohms, which is excellent and far superior to most power amplifiers.

3) Regulated power supply for the output stage--VERY rare among even high end transistor amplifiers.  Ensuring low dynamic modulation, another issue never measured but easy to understand.

4) Wide bandwidth: 0.5 Hz to 300kHz.

5) DC coupled, so no capacitor sound.

6) High power which doesn't sag, 450 measured watts into 8 ohms, 900 into 4 ohms, even more into 2 ohms.  My electrostatic speakers are very inefficient and have impedance that runs to less than 2 ohms at the highest frequencies.  They are also very revealing of distortion.

7)  It's generally a nice amplifier to use.  It has a nice soft pushbutton to start and the slow start causes no noise or light dimming.  The connections are very nice.  It looks very nice as well as Impressive.

I've often thought, if I had the money and space, it would be nice to have a newer and more highly regarded Pass Labs amplifier, such as an XA 200.8.  Now these monoblocks require more space than I have, and also consume 680W continuously each, for a total of 1340W.  That's about double my average power consumption with the Krell, which is already getting to be a bit much.  So they wouldn't be very practical for me, let alone the $44,000 pricetag.  But what if I had the money, the space, etc, wouldn't it be wonderful?

But it turns out, you can find on many blogs, many people do NOT find the Pass Labs amplifiers to be superior to their favorites, which may be old Krell, Levinson, and Threshold amplifiers.

And there may be objective reasons the Pass Labs amplifiers and other current favorites (including mine, in the lustworthy category) like Threshold and D'Agostino and Levinson might not be as good sounding as my FPB 300, as well:

1) They generally do not have the regulated power supply.  The XS 300 does.  The very top Levinson usually does.  Those may cost way double what even the XA 200.8 costs and consume even more power, just for this pretty obvious feature.

2) They do not have as wide a bandwidth, or as low distortion, especially at higher frequencies.  The XA 200.8 reaches 1% distortion at 20kHz and 8 ohms and conservatively rated power.  The Krell only gets up to 0.1% or so.

Those same considerations apply to most amplifiers ever made by Threshold, Levinson, and even Krell.

3) Output impedance.  SoundStage measured the FPB 300 output impedance as 0.07 ohms.  That compares very favorably with the 0.14 ohm output impedance of a Pass Labs XA 200.8, more favorably with the D'Agostino Momentums at 0.2 ohms, and the D'Agostino Progressions at 0.44 ohms.  These newer amplifier just aren't doing the damping factor like some older ones.

4) Power into 4 ohms, and 2 ohms.  Neither the XA 200.8's nor the D'Agostino amps produce more than 600 watts into 4 ohms.  My FPB easily cranks out 700 watts into 4 ohms still with still very low distortion as I measured, but measurements by others show over 900 watts into 4 ohms with less than 1% distortion.  It was measured by Martin Collums as having over 1500W into 2 ohms (which isn't specified).

Scanning the measurements of amplifiers, only a few stand out as having possibly better measurements for me (with a very inefficient 2 ohm minimum speaker).

1) The Threshold SA12e.  This is the pinnacle of Threshold designs by Nelson Pass and the most desirable (if you can handle the power consumption heat), which has far lower distortion and output impedance than other Threshold models.  It bests my FPB by having output impedance of 0.02 ohms, and possibly even lower distortion.  Nevertheless, I've seen reviews in which FPB 300 and SA12e are subjectively compared with the FPB being declared the winner.

2) The Levinson 33H is right in there also.  Newer Levinsons with hybrid "digital" output stage produce more noise at lower levels like 1 watt.  Lesser Levinsons don't have the 4 ohm power, etc.

3) The Soulution has lower distortion, as does the Benchmark, and the top Halcro, but not enough 4 or 2 ohm power.

4) No doubt many others I've never heard of, but generally less known.

Now, another amplifier which might be still better than the FPB is the Krell KRS-200, said to be a D'Agostino personal favorite.  And it ought to be, given the history, build, original price, and power consumption.  It's a true 200W Class A amplifier with no fiddling around.  There's also the KAS 1 and 2, which are plateau bias amps with regulated power supplies, much fancier and pricier and overbuilt than the FPB amps.  Actually, there may be a multitude of other amplifiers I don't know of, that would have the esseitnal features of regulated power supply, low distortion beyond 500W in 4 and 2 ohms, even at 20kHz and above.  But, of all the ones I read and hear most about most of the time, there aren't many.

All these amplifiers are incredibly more expensive than what I've paid for the FPB 300, even including all my servicing adventures.   I've read negative reviews on Levinson service which no doubt is very costly, Pass Labs gets the highest recommendations for servicing, which they hardly ever seem to need anyway.

Now I don't know, I haven't had those amplifiers in house, and possibly never will.  Even if I did, I still probably couldn't have a firm (and proven) subjective opinion.  But it appears, I have "a contender" and that's what I want.  I want an amp about as good as it gets, but not so much my finances are collapsing.  And I want to be able to compare nearly as good amps of radically different designs, and find if I ultimately can provably hear the difference.  My Hafler is an example of a radically different approach: a very simple design, using MOSFETS and almost all FET circuitry.  My Aragon is a simpler amplifier than the Krell also, but bipolar outputs like the Krell, with massive conventional power supply.  These amplifiers are all objectively excellent, which is a good starting point.

Despite limited power, the Hafler does wonders with high peak power and current, and indefatiguability.  The 9500 has considerably worse measurments, and the 9505 has way more complexity.  I feel the 9300 is the goldilocks of Strickland TransNova designs.  It has the lowest distortion, even lower than the Krell.  But it lacks the regulated power supply, and higher continuous power of the Krell.

Emotiva headroom and distortion measurements (new unit)

Emotiva lists 1.0V as the standard output for the XSP-1, however, the distortion is actually spec'd at 2.0V.  No other voltages are listed.  I was reassured by Emotiva that "6 Volts" is the maximum, but I wasn't entirely clear on whether this was input or output, and I'm concerned about the 3.8V or so from the singled ended output of my DVD-9000 playing HDCD's.

I've measured the maximum voltages now, somewhat approximately because I'm only using "instrumentation loads" like my 10M DVM and 1M oscilloscope.  This is a factor that 10 years ago and before I simply ignored.

Anyway, for single ended I/O the maximum input is very impressive at 9.6V RMS (measured with the gain at -20dB).  Maximum SE output is around 6.8V (gain at +12.0) at onset of obvious clipping, a tad bit of obtuse rounding can be seen above 5.5V (looking on my newest digital oscilloscope).

Driving the output into very hard clipping close to 10V can cause square wave doubling.  This is an undesireable per se, but in most circumstances your home amplifier will already have exploded at 10V sine wave input.  This is not an instability, what happens is that after the top of the sine wave flattens, the center begins to grow negatively proportional to further drive, until this center downward spike reaches the bottom, making the clipping wave, now pretty much square, into a doubled square wave at twice the original frequency.  It shows essentially that multiple internal elements are clipping at not exactly the same time...therefore something like a push pull output.

True pro equipment can handle 10V as a standard level.  That's probably why Emotive lists the standard level as 1V, to make it clear it's not 10V pro gear--which is compromised in S/N at lower consumer levels.

These I will not be pushing the Emotiva limits in any way, though a recent adjustment to the Lavry was to set max level (-8dB) to -17dBU which is 5.4V, which would be a touch too high for single ended output on the Emotiva, but probably gives around 6dB headroom for balanced output (not measured yet, that presents some challenges).

Certainly the input will handle anything I can throw at it input wise.

On my Juli@ card, using RMAA, with balanced I/O last weekend I measured about 0.0005% or 0.0006%, with my residual at 0.0003%, meaning the actual distortion is around 0.0003%, as shown in website graphics (showing better than the claimed spec of 0.0005).  These are measured at the 4V input level of the card, minus about 0.9dB.  I measure this either to 96kHz sampling or 44.1kHz sampling with the same result, showing no super high harmonics, very clean top end.  Basically my residual and the Emotive line up perfectly, save a little more 3rd harmonic (the push-pull output cancelling any 2nd order distortion) and down below -120dB a tiny bit of 7th (generally, stuff below -110dB can be ignored, the Denon DVD-9000 is full of high frequency hash at -115dB but is one of the best sounding CD players I've ever heard).  It was this cleanness that sold me on the Emotiva, among other things, and having the best phono circuit in my collection clinched it as my living room analog preamp, requiring me to get a second for the bedroom I had originally purchased my first unit for several years ago.  The THD levels compare favorably with the BEST current Mark Levinson at $40k, and the Emotiva has the essential-for-me digital volume control if not as nicely implemented as the Levinson.

On my Sound Technology, I get measurements showing that single ended output should be kept below 4V.  (I actually don't use the single ended output for anything except driving the Sonos box at 2V in the bedroom.  The balanced outputs go to the Masterlink in the bedroom, and the Lavry in the Living Room, my two most critical things.)

6V output...4% THD
5V output...0.38% THD
4V output...0.0063% THD
3V output...0.0046% THD (this is about my residual on this analyzer when it's working great, once in a blue moon it can hit 0.0044%, but right after taking these measurements it decided to go beyond 3% resdual, a sign of internal overwarming or something, so I shut it down).

In balanced output, this would probably translate to a maximum low distortion output of 8V, well more than what I need for the Masterlink (4V) or the Lavry (5.4V).  Right now, with cranky Sound Technology analyzer, I can't do the measurement I had hoped to do this afternoon to confirm that.

Bless the Sound Technology, after 3 hours rest it came back on and I was able to record a better residual, which is I think not untypical actually (until after a few hours, it will go up, I adjusted my ST that way so I could do decent measurements without 3 hours warmup, I'm usually just doing a few measurements, though this way, in 3 hours you can't use it optimally).  That is:

Sound Technology Residual: 0.0033% THD

Keep those in mind for the above measurments, the 3V Emotiva output is not exactly the residual, it's 0.0013% higher, but that might depend on the number of minutes of warm up also not that much.  Actually I know from other measurements 2V is at 0.0003%, so the 0.0013% probably says more about the ST and it's state of warmup (remember, it failed just a few minutes later, and I had to let it cool down for 3 hours).

Anyway, with some problematic adapters they may have been contributing, I measured output distortion balanced, once again using the ST as primary source.

At 8V balanced, distortion was a hair above the residual at 0.0038%.

At 10V, distortion had risen to 0.03%.

At 9V, distorition was also around 0.03%

So my prediction about 8V being the highest low distortion output level seems correct, the peak optimal RMS levels are:

4V single ended
8V Balanced

At these output levels, THD has not increased since the lowest output levels, and may even have reached its lowest level.

It goes higher, but distortion starts rising, fwiw, still way below speakers and such a few dB higher.  I'd keep it below 6V single ended and 12V balanced in all cases with great prejudice, though it doesn't really break up till even higher.

I could even crank up the input level on the Lavry, now set to 5.4V, a few dB higher.

The Kitchen HT system sub now re-enabled for stereo

I mainly use my kitchen HT system to play stereo, but I have the two surrounds and sub to add in for actual 5 channel material.  I'm opposed to center channel speakers on various grounds, and I always use the appropriate setup which folds the center as a mono signal into the fromt left and right.

For stereo, I'd always used the "Direct Stereo" output of my Yahama HT-5790.  I measured this, and it clearly bypasses the digital conversions that the various HT modes require.  An alternate is "2 channel stereo" which does do digital processing for 2 channels.  The digital processing is clean except, of course, it introduces noticeable pre and post ringing, the kind that J Peter Moncrieff seems to be telling us is all fine and good.  (And all "objectivist" engineers also, I might add.)

So, using Direct Stereo, I have no "subwoofer" output, because the crossover is implemented in the DSP.

Recently I've been playing FM on the Kitchen Tuner because it attached to my roof level whip antenna which clearly surpasses all my other antennas (merely as a result of height, long I tested inside and it was worse than my indoor antennas).

After I moved on from the McIntosh MR78 with Modafferi Mod, which had a kind of steely, though clearly very high information, sound that was fascinating sometimes but not appealing and actually a bit tiring, I brought out the Sony 730ES (because it was most accessible in my conditioned storage building) and the sound is so much "nicer" somehow, combined with the much better antenna, it's somewhat intoxicating.

But what's always been noticeable was the kind of "miniaturization" done by the Kitchen setup.  This may be partly the almost nearfield location of the speakers, and similar factors.  The speakers themselves are the once Stereophile Class A Revel M20's, which go to 30 something Hz and cross over nicely at 60Hz, as I have always done in the bedroom.

I've always noticed the "heft" of the living room system playing FM, so preferentially even I play the Living Room stereo when I'm in the kitchen, even at the expense of an actual stereo image.  Appropriate somewhat to FM, it was the "concert is in the next room" experience, but like a real concert, instead of the table-top miniature in the kitchen.  Having the subwoofer for regular stereo, now adjusted pretty good, gives things the full size and heft, and even seems to be removing some of the steelyness from the tone--which might have been resulting from mismatch between extended high end and rolled off low end.  It may have been that lack of steelyness which I was confusing for the lack of definition in 2 channel (dsp) mode, much more than the digital resampling and DSP.

At various times I've played the subs, and though this speaker (an SVS 10" from the 2000's which was my very first "real" subwoofer and was a revolutionary upgrade to my bedroom system, replacing the McIntosh M22's I had been using as subs, but that was quickly upgraded to the SVS 39 inch 16 Hz tube). but in the kitchen it had sounded terrible, quickly invoking the "turn it off" reflex, so I guess long ago I had decided Direct Stereo was better and that was that.

But, I wondered, what if I just turned down the subs a little.  I had never really measured the system, or adjusted by measurement, or maybe I'd tried the auto adjust once and hated it, preferring my own manual "by ear" adjustments, notably using the built in level adjust test signals in the receiver, and balancing the different speakers and the subs by ear listening to the test signal.  This was set up before I had an always ready 1/6 octave RTA in my phone.

So I tried adjusting the sub level to get the flattest response using my standard noise source, Stereophile Test Disc 2.

And this required an astonishingly low setting for the subwoofer level.  After several days of readjustment and measurement and listening I've gotten to setting the subs at the low end of the second lowest mark on the level screen, with a 60 Hz crossover and "bass" set to "both".  As I was previously attempting to audibly match the level of the 1kHz centered pink noise signal with the signal in the bass on the Yamaha, I was setting to within the third highest mark.

If I set the "bass" to "swfr" it seems that the low bass disappears in "2 channel mode".  Im suspecting this is a bug in the Yamaha.  That's why I'm using "both," which I assumed would effectively use the bass as fill-in, and since I've set the back speakers to "small" they shouldn't get any bass from the "both".

What I really plan to do is use a behringer DSP to do the fill-in crossover and shaping for the bass, and I currently have a spare DEQ 2496 which I'm clearing the space for now.  Previously, way back in 2009 or so in fact, I bought an extra Behringer of the earlier vintage, DEQ 8020 or something, only 20 bit, and it was much more painful to use having few buttons and unintuitive to me displays, so the "equalizing the kitchen sub" project never got off the ground because I hated the equalizer.  Now I have a spare 2496 which I like (and will have to get another spare if I decide to continue this, or perhaps another miniDSP).

Now what I plan to do run the main speakers straight through, perhaps in Direct Stereo, and capture the extra LR line outputs in the back for the DEQ, which will then perform crossover, eq, level, and perhaps other functions.  However, I'll have to have a switched connection to the sub outs for multichannel inputs, which simply go straight through the receiver

Moncrieff and Linkwitz

J. Peter Moncrieff, editor of International Audio Review, had been quiet for a long time, his website having last been edited in 1999.  I tried to buy back issues in 2013 without success.

But now he's reappeared, with an incredibly long winded (as usual, and I haven't read it all) takedown of MQA, and not just MQA but similar supposedly transient improving techniques in digital audio (which he calls a "modern revisionist digital engineering movement worldwide").

http://www.iar-80.com/page170.html

A lot of what Moncrieff is saying is correct.  Sure, he's making mountains out of molehills, exaggerating to the max, and full of puffery and self-congratulation (which I find entertaining, actually, like listening to PT Barnum, but now if he could only be a little less repetitious it would be less tedious).  But, still, much of it is correct, and I believe MANY wise old fashioned engineers would agree with the thesis that modern short transient digital filters that seek to eliminate pre and post ringing, including MQA, are WRONG at least in principle, pretty much along the lines Moncrieff is saying (though I'm not sure all his thought experiments are precisely correct, they capture the jest in a very accessible way, and his gift at doing that makes me find it worthwhile to read).  The old fashioned digital filters take full advantage of the sampling theorem's promises to capture all the information in the bandwidth window, up to the limit of their technical features (such as the number of times of oversampling...which does benefit from greater than 8 times, just as Moncrieff says, though many old fashioned engineers would say 8 times is plenty good enough) and the new kind not so much--the new kind ARE information lossy, with MQA being the king of the hill in lossiness of anything that claims to be CD quality or better.  If these new digital filters have benefits at all, it's through the euphonic effects Moncrieff describes, not actual higher fidelity.  It's like NTSC color TV's from the 1960's, first they up the color temperature to 9000K to make it look brighter, then they add nonlinear red push to make the skin look natural regardless of that, then they peak the horizontal response to look like there is more sharpness than there actually is, never mind the false edges and other artifacts.  A properly designed TV of the same basic performance would look less bright and less sharp, but would be more accurate and contain more real unobscured information to the serious viewer.  A properly designed digital system may sound duller and less spacious than MQA, but it's true hifi and not euphonia.  Though I'm also of the opinion the difference between MQA and regular PCM would be very hard to hear above the level of chance.  Still, and then perhaps even moreso, why not have true high fidelity?  And especially if you have to pay more to a middleman to have the fake.  Perhaps MQA should be understood more as a watermarking system than a high resolution audio system.  And the same is true of SACD and DSD-64, not that many care anymore.  (I keep my vintage Sony 9000ES, with true 1-bit converter at 10x oversampling, on the grounds that to enjoy the sound the producer intended, you need to use the matching decoder, and many SACD's do indeed sound pretty good for good production reasons and despite the inherent lossiness of the system.  The same is true of HDCD, which I think better than SACD, and has zero information lossiness whether decoded or not, but is dynamically lossy if not decoded.)

On another topic, one great audio engineer passed away recently was Siegfried Linkwitz, the primary inventor and promoter of the Linkwitz Riley crossover now used by many manufacturers and builders (including me, since about 1983),  the designer of many great DIY designs, and creator of a website with a vast amount of incredibly detailed audio  information and analysis. 

https://www.stereophile.com/content/siegfried-linkwitz-rip

http://www.linkwitzlab.com/

This was strangely ironic for me, as about the same time as Linkwitz passing I was discussing crossovers with a friend who dismissed Linkwitz's claim that the group delay (phase shift) caused by properly implemented LR crossovers is not audible, or is at least not audible to him.  My friend described Linkwitz very negatively regarding this claim.  I believe Linkwitz was honest and a careful listener and these differences ARE hard to hear.  Here is Linkwitz' page on the subject:

http://www.linkwitzlab.com/Removed%20pages/x-phs-dist.htm

But I've long considered the idea of transient perfect crossovers to be appealing, so I've launched a new set of projects to try them, using miniDSP processors and FIR digital filters.  There will be a steep learning curve in this.  (Linkwitz has other pages where he details the problems in "phase perfect" approaches.)  Ask me in about a year if I got anything working.

MiniDSP opener

All these distractions from my #1 audio concept: linear phase DSP.

Wow, I can actually run rePhase 1.3.0 on my Mac using the latest Wine from Wine HQ!

Loading in the number of taps, I was reminded of the 48kHz sampling rate limitation of the OpenDRC 2x2 plug in for my OpenDRC-DI.  With my 40kHz super tweeters and all, I wanted to stick with 96kHz, which is what I resample analog sources to.

Well apparently the sampling rate can be boosted to 96kHz using a different "plug in."  I'm beginning to see how this works.  The OpenDRC-DI can accept digital inputs up to 216kHz, but the get asynchronously resampled to whatever the plug-in runs at.  And if the plug in runs at a higher sampling rate, then there are fewer taps.  This IS an issue for the ultimate DSP control, especially at low frequencies where you want a lot of taps to deal with complicated room issues.

This thread discusses the possibility of running the miniSharc 4x8 plug in on the OpenDRC platform to get 96 kHz capability.  The downside is that the taps are reduced to 2048.  Using the standard OpenDRC 2x2 plug in, 6144 taps are available.

It seems to me I could use the 4x8 plug-in on my supertweeters where the 96kHz rate is useful, and then run the 2x2 plug in for the panels and subs.

One nice feature in this regards is that the miniDSP's are going to be putting out the designated sampling rate no matter what the input source.  So, I will only have to do the time alignment between DACs for one sampling rate, instead of separately for each sampling rate.  In other words, I just do the time alignment once, and it's set, regardless of sampling rates and the fact that I'm using a different sampling rate for the super tweeters.

That Finnicky Linn Sondek

I've heard the advice: you are supposed to put your Linn Sondek on a small but rigid and light table...

Well, I'm sorry, my life is not like that, no extra floor space available for dedication to the turntable.

But I thought, I might get away with balancing the Linn by putting some fairly stiff small pieces of paper under the back feet.  I did that a couple weeks ago to a great feeling of satisfaction, seeing as how I'd made the turntable almost perfectly level, instead of the previous off by about 1 degree (or whatever 1 big mark means on my large circular level from KAB), tilted toward the back.

However, in infrequently playing discs since, I'd been noticing the Linn "isolation" was not what it was cracked up to be.

Finally, I put two and two together, and realized that my leveling mod may have increased instability. So I removed those little piles of paper behind the back feet, and presto, the fairly decent Linn mid-bass and up isolation has returned.  I can tap on the underneath the shelf pretty well without visibly upsetting the lp surface.  Previously, anything more that the very smallest touch would set it going.

The Linn guru Mark warned me that to level the table I should "fix" the stand.  He was right about that.  But it's not going to be easy to "fix the stand."  So it can wait until the stand gets moved back after the next remodel (replacing the window behind it has to get done before too long, and the floor underneath it).

Sadly the fixed Linn Sondek feet appear to be an essential part of the whole rig, and they must sit down atop a rigid surface, just as the Linnie advice suggests.  As far as the table being light, that another thing I don't believe and can't imagine accomplishing.

Also I don't believe in the superiority of not clamping, but for casual background listening, unclamped playing is far easier (one must remember to remove the felt washer!!!  a mistake I've made a few times) and works better on Linn than most tables.  It does increase the rhythm and tunefulness (Linn emphasizes the tunefulness, not the rhythm) however it blurs the detail--which is only really important for serious listening.  I find, however, that unclamped playing is slightly harmful for the tonality--which becomes a bit hashier even as it becomes more "tuneful."

The soft springs of the Linn actually protect the bearing when you are applying the clamp.  Only suspended tables do that, usually you are crushing down on the bearing when you apply the clamp, and that could be harmful to many turntable bearings.  On the Linn, the table only goes down a few mm before the platter hits the top of the plinth, which supports it nicely when you are clamping, with only the light springs loading the actual bearing.  I used to think it abominable to press the turntable down to the nice plinth top, but now it seems like it was made just for that.

The best way to clamp is kneeling down so the vinyl is at eye level.  Then, wriggle clamp down slightly for maximum vinyl flatness.  Some, but not excessive, pressure is required, mainly just enough to keep the clamp from slipping off the short Linn spindle.  Too much pressure may cause additional bowing with the edge rising.

Hafler 9300

Now that the Krell FPB 300 shuts down in 10-90 minutes of operation or even idle with shorted xlr's and open outputs, I'm using the Hafler 9300 almost continuously.  The Hafler  a great amplifier, I believe, built with one of the world's greatest designs: the trans-nova.  I'm also lucky to have a minty and perfectly operating copy. I think the Krell amplifier might be a tad cleaner but after going back and rechecking, I've never been sure I could hear a difference, and many times I've mistaken one for the other, using thinking it's the Krell when it's really the Hafler.  This pertains to power as well, I've never had a case where conclusively the Hafler sounded less powerful.  Now I find that recent experiments were marred by volume adjustment creep on the DACs.  The correct adjustment is -7dB for the Krell and -4dB for the Hafler, a -3dB level difference.  I'm reminded of a friend who didn't believe levels could or should be matched.  My matching is within about 0.13dB, the best I can do and lucky it's that good, with the 0.25dB settings on my oldest Stealth DC.

Looking at the 9300 schematic, this is what stands out.

1) It's very simple for a reasonable power amplifier, though perhaps not as simple as I had been thinking, not simplicity uber alles.
2) It'd direct coupled with DC feedback and no servo.  The best when you can do it, and simple.  NO capacitors in the signal path to worry about.  No inductors neither, and no inductors or anything at the output.
3) The schematic is drawn wierdly around the supply capacitors.  There is nothing weird about the supply caps, in fact they are nicely bypassed with 4.75uF caps (probably film).  I had been thinking they formed a surrogate cap coupling, but not in this circuit.
4) What is a bit weird is that the - terminal is driven, and the + terminal is really the signal ground (explaining why the capacitors are drawn as they are...to show the ultimate signal ground).  The amplifier is internally inverting, but this is corrected simply by labeling the terminals in the correct polarity, therefore it appears to the user as a correctly polarized amplifier (just don't connect the grounds...which many amps warn you not to do when it wouldn't have mattered).
5) Essentially standard "the best" layout: input dual jfet diff amps with bipolar current mirror feeding bipolar drivers driving MOSFET output banks.
6) MOSFETs don't need no stinking current limiting, no stinking shutdown, no stinking anything except there are rail fuses, which have never blown.
7)The novel feature is the MOSFETs are in gain mode, making the amplifier inverting, so feedback is taken from their input.  I'm not sure why it is done this way, but I suspect ultimate "peak" power, which is actually the way it seemed on the bench.  Distortion began slowly rising above the rated level (150W at 0.003% distortion) however it seemed it might put out as much as 500 watts peak.  This power availability also translates to more than negligible output at 2 ohms or less.  Basically, there is minimal resistance (I don't see any!) in the output circuit, so it's wide open.
8) Feedback is therefore isolated from swings in power supply, and well as ultimate device linearity, meaning the amplifier doesn't "double down" when the stored power is running down.  That probably a GOOD thing.  It gives the best effort and moves on, rather than getting stuck in the mud.  Power supply isolation comes from the MOSFETs themselves, which are excellently so isolated.
9) Despite that lack of feedback around the output devices themselves, the distortion is incredibly low for a power amplifier.

I only think it would be slightly better with what the Krell has: a regulated power supply.  That where I imagine the differences I probably imagine the amplifier sounds as having comes from.

AND, looking at the 9303/9505 schematic, it's almost entirely different, far more complicated with many more bipolars in the front end and the jfet diff amp buried within all these bipolars.  Also there's a servo which the 9300/9500 don't need, and even...gasp...what looks like a protection cutoff circuit.  Not a jewel like the 9300 IMO--which doesn't need servo or protection circuit.  The Nelson Pass designed Adcom 5500 is somewhere in between, more complicated than the Hafler 9300 but less complicated than the 9303/9505, but, characteristically, uses almost entirely jfets and mosfets with just a bipolar driver stage (like the 9300).  While the Adcom looks like a fine amplifier, the Hafler 9300 is just the cat's meow of simple yet effective designs, I've never seen one done more nicely save the First Watt F5, which is power limited by comparison because there are just limits as to what's possible when leaving out the bipolar drivers.

DVD-9000 and HDCD

Since I now have an Oppo BDP-205 in my living room system, I only use the Denon DVD-9000 to play HDCD.  The DVD-9000 is a special player for HDCD, I have always thought.  Among other things, it seems to uniquely implement the HDCD peak expansion, so that HDCD's can put out 4dB more than regular CD's.  I can say that authoritatively because I have measured it now on my Lavry.  Plus or minus < 2dB.

Further, I have measured it using digital input.  This means I can pipe in SPDIF from my server, as I now do through the "Living Room 2" Sonos Connect, and have the DVD-9000 output HDCD in analog, then resample back to 24/96 digital on my Lavry for playback through my DSP's.

I was worried because there is no HDCD indicator which lights up when you use the DVD-9000 as a DAC.  And the DVD-9000 brochure I have on my computer says only that you can play HDCD discs.  When you use the DVD-9000 as a DAC, it puts the sample rate in big letters on top of where the HDCD indicator would otherwise have been.

I have long thought about doing a similar thing with MQA, though another friend sent me Archimago's takedown of MQA and it looks pretty convincing.  But I think I might add inexpensive MQA decoding capability to my system anyway, though I am intrigued about the "unfolding" thing which might be an option.

Interestingly, the DVD-9000 does not seem to contain a PMI chip, like the PMI 100, to decode HDCD as the Denon DVD-5000 did.  Instead, the Denon has an analog devices chip as "audio decoder" which apparently implements a number of codecs including HDCD and possibly Denon's AL24 now in 32 bit.  Because this processor is a 32 bit chip instead of 16, it may well implement HDCD to the same standard as the unobtanium PMI 200.  But the same thing may be true of virtually all later HDCD players, including the Oppos, that switched to using licensed HDCD implementations on their digital filter chips, rather than PMI chips.

Famously HDCD has no digital unfolding because the core of HDCD is actually changing the digital filter.  What this does is not fully expressable even within a 24 bit stream.  It can exchange bit accuracy for timing accuracy when the program material or producer deems it useful; it's giving them the power to switch the digital filter mode on-the-fly.  Meanwhile, the newer MQA only has one fixed digital filter, and an extremely leaky one to high frequency images which I have usually called aliases.

But when resampling the analog at 24/96, much of that added timing accuracy is captured by the faster sampling rate.

I have also decided to change my "standard" levels so the Lavry is made 2dB less sensitive.  This means I do not have to attenuate my Oppo through the Emotiva XSP-1 before sampling on the Lavry.  It seems to me that keeping things as close to 0dB as possible on the Emotiva is best (though I had previously worried about distortion or even clipping on the XSP-1, I have been told it can produce up to 12V output, though that is not in the specs).

So the new "standard" levels are:

Level setting on Lavry: +8dB (2dB aove 3.88V balanced input)
XSP-1 gain for Oppo BDP-205: +0.5dB
XSP-1 gain for Sony 9000ES: 0dB
XSP-1 gain for HDCD  on Denon DVD-9000: -4dB
XSP-1 gain for phono: 2-6dB
Stealth DC level for DAC for Hafler 9300: -3.75 dB (this matches Krell at -7dB)

Update: this is trickier than you might presuppose, because of sample-overs.
The DVD-Audio of Two Against Nature has shown it necessary to lower the Oppo input level on the Emotiva to -2.5dB if I have this correctly, showing sample-overs of about 3dB therefore.  That may be a decent number for the 9000ES also.  I doubt it would change the Denon DVD-9000 which reserves the highest levels, up to 3.5V, for HDCD extra dynamic range, and I've tested a range of discs for that.

Not Again...the Shutdown saga

I've already started on new, revolutionary (for me) projects: Using a triad of miniDSP processors with AES digital in and out, I'm going to re-implement my crossovers as linear phase, and possibly correct other phase aspects of my system, by using the non-causal filters which can be implemented with FIR.  In principle, the miniDSP's could replace my Behringer DEQ's, but my present plan is to keep both in series, so I display and input more ad hoc "causal" filters (the ordinary kind which can also be implemented with analog electronics) using the Behringer.  Since it's all 2496 AES digital, the only loss from having more gizmos inline is jitter, and as I have previously explained, having more quality components inline hardly affects jitter at all, and it's almost completely unimportant anyway at the levels my system would reach with many more such gizmos.

And that gets to another theme I really want to take up, the fake importance of "simplicity."  What actually matters is quality, simple crap is still crap.  Usually, in order to get higher quality, higher and better aimed effort is required.  This effort can go into sorting the simple crap out, finding the very best pieces.  Or it can go into trying to use each piece in such a way that it's limitations are fixed by other pieces.  Either way is a valid approach to some degree.  Though, in my estimation, at the level audio products are now, some complication, such as multiway speakers, is required to get decent quality.  Others may disagree.  And so what?  But I think, more often than not, simplicity surfaces as "you need simplicity," which is either a nag, or a sales pitch, and both of those are just annoying.  The sales pitch angle of this is obvious.  Sell all your previous crap, and buy this one new (vastly more expensive, of course) piece to replace it all.  It's an old line.  Whence the expression, "don't spend all your money in one place."  But, that is what they are always, and still, trying to get you to do.

And I've started to build a First Watt F5, a Class A amp for my supertweeters (which still seem to add something important...actually by my most recent tuning they are, barely, audible, and this gets me to flat response to 20kHz on my iPhone RTA.  Since I'm using damped cloth dome tweeters which happen to have 40kHz response, there's no added "metallic" quality when I do this.

And since I got it back from repair and upgrades, the Krell has been operating perfectly, better than it ever had before.  It had always had an overheating left channel--it simply went to 180F at nearly any playback level.  That problem is gone, both channels now track within a few degrees, the remaining variation probably caused by airflow.  And the clinks have all but disappeared, especially since I readjusted the air conditioning registers on the opposite side of the room to dump all the air before the center of the room (though avoiding the couch) .

But, a few days ago, I decided to play the Reference Recordings HRx sampler, which has music recorded at 176kHz sampling rate in high rez files which can be read by a player like my new Oppo BDP-205.   I resample the analog output of the Oppo routed through the Emotiva XSP-1 at -2.5dB into the Lavry AD10 set to peak level 3.88V, the Tact is set to 93.3, and the Emotiva Stealth DAC is set to -7dB for the Krell and connected balanced.  (The correct matching level for the Hafler in -3.75, I discovered later, having previously set it to -4.75).

I'm calling the last two levels my current "standard high level."  It's about as loud as you would want to play anything, but not clipping the amp (at least, it's not clipping the Krell).

And, just like back in February, the Krell shut down on track 5, the Satie, which curiously was far lower in level than the previous track.

The next night I played the disk again, starting from the beginning, and starting from a cold amplifier again (in past experiments, that has been critical to reproducing the same failures), and it shut down at the same point in the same song.  This time, however, I had shut down the air conditioning completely so no air was blowing.  I had been thinking I could possibly cure this shutdown problem by moving the amp completely out of the possible air stream, behind the left speaker, where the Hafler and Aragon are now, and bring those amps to the front, where the Krell is.  But, since the exact same shutdown occurred with the AC fully turned off, it doesn't look like that would help.

Since then, I've quit bothering shutting the AC down, since it doesn't seem to make a difference, and instead tried disconnecting the right speaker, then disconnecting the right speaker.

With only the left speaker playing, the shutdown still occurred, but in track 7.

With only the right speaker playing, the shutdown did not occur in the entire album.

Though I haven't totally eliminated other variables, I'd say either the left Acoustat, or the left channel of the Krell is at fault, and the next tests should isolate those possibilities.  In some sense actually I mean that the Krell is at fault, in the sense that other amplifiers work just fine possibly by ignoring to weird load of the Acoustat.  But that's the given that I just have to work around, I don't know why, I like the sound the Krell, the looks, etc.

After playing the Right channel for hours on the right Acoustat, both channels of the Krell had heated up to around 150 (the unused Left channel hotter, actually, driving all the speaker cable capacitance in Class A, and it usually runs slightly hotter anyway), I played both channels, now that they were warmed up, and the shutdown did not occur.

Monday

I've been having an argument with a friend about which tests to do and what they mean.  Without getting into the details, this shows that even totally objective testing (you can hardly get more "objective" than an amplifier shutting down, or not) is extremely complex and difficult.   The kinds of ad hoc subjective tests that audiophiles generally rely on, are worse.

On Sunday, whether this was a optimal test to do or not, I reversed all connections at the amplifier so that the right amplifier channel was getting the left inputs and playing the left speaker, and vice versa.   For all of the subsequent tests, I played the entire disk (FWIW).

Playing the right speaker alone (on the left channel of the amp, but with the right input signal) did not cause shutdown, nor did playing the left speaker alone (on the right channel of the amp), nor did playing both left and right channels on the opposite channels of the amplifier.  In each case I waited 6 hours or more for the amp to cool down to normal "stand-by" temperatures, however this might only take an hour or so, standby temperature seems to be about 10F above ambient because it's running 60W, though it doesn't seem like 60W should run that warm.

But right after doing playing the right speaker (on the left amp channel, with right input) I noticed something.  I had somehow put the Tact preamplifier in Correction 9 by mistake.  I have no idea what Correction 9 does, I mainly use the "corrections" to do measurements since you must select a target correction curve even to do a measurement with the Tact.  I could have been measuring the tweeters, I could have been measuring a wire for calibration purposes.  Anyway, having the Tact in a unknown correction could by itself cause an amplifier shutdown problem.  So, at this point it could be this whole failure episode is nothing but a false alarm.  I'll now have to go back and retry the original test with all channels normal.  I also do not know when Correction 9 had been selected, it could have been selected on Saturday night after I finished amplifier testing (since the amp had warmed up anyway) and started playing other things, and curiously enough found another test that seemed to show the audibility of polarity  (despite my belief it is barely audible, but in this case the MP3 conversion was terrible and possibly non-linear, making polarity changes more audible).  I got very involved with that polarity testing, and it's possible that is when I accidentally selected Correction 9, which is very easy to do by accident using the Tact--the correction selector is more prominent than the volume control and if you are not looking closely, you might be selecting a correction rather than setting the volume.

I proceeded to do the "opposite" test, left speaker on the "right" amp channel, with left input, six hours later (to let the amp completely cool down in standby, however 90 minutes might have been sufficient, I was fooled because it seems the standby temperature of the amp is 10F above ambient).

No shutdown.

The next morning I put both speakers on at once in this configuration, and there was still no shutdown.

Finally, that evening, I put everything back to the starting condition, and turned the air conditioning off because that seems to be a highly variable influence, and one that's not needed for the shutdown to occur.  But, still, now there was no shutdown.

Already I was beginning to think of something I brushed over near the beginning.

After the first three shutdowns (the first, and two tests) was the first time I removed the left speaker banana to disconnect that speaker (because in the previous test, I had removed the other banana).  Right then, I noticed that the knurled knobs holding the thick (2 x 11 gauge) wires into the RCA Gold bananas had come loose.  Somewhat loose...they still basically held onto the wire.  I had long since resolved to change them.  I was planning to replace them the upcoming weekend, but the audio society party, and further amplifier/speaker testing had priority.  But right then, I just re-tightened them, that usually held for at least a few weeks.  Somehow, those big knurled knobs often come loose, while smaller set screws tightened with a tool almost never do so.  I should have known that when I started using these Gold bananas, but caved because they were so easy to use and took very big wires.

Well, if you've been following every line, you will see that there was no failure in any subsequent test, going back to the very first experience.  Whereas, it seemed the original problem was quite reproduceable, it happened 3 times in a row on the same recording at the same level.

So it appears this connector was at fault, and a short intermittant in connecting to the Acoustats seems to make the paranoid amplifier think there is a short.  (I'm thinking of calling it Marvin, the paranoid amplifier.)

On at least the first two shutdowns, it occurred some seconds after playing the Walton track 4 on the Reference Recordings HRx sampler.  Well that Walton is full of very impressive big BOOMs.  Those could possibly shake the speaker (not just from the speaker itself, but the subwoofer right nearby).  That invisible shaking was possibly just enough, a few seconds later, right as the speaker was settling back into it's normal condition, hit a non-conductive spot on the bare copper wire, causing an instantaneous "open."

Thinking about it now, this might even explain some shutdowns last year.  But not the very first, because I was still using the Canare 4S11, which had terminated ends (unless...I had already removed those ends for some reason...those Canare's have been used in several different ways since I bought them).

I was certainly using the Radio Shack bananas from the time I set up the ABX system in January.  However, by that time, I had fixed the "shutdown" problem by making repairs to the speakers.  It was about that time, however, that the overheating problem materialized, and that couldn't happen from the connectors "opening" up...unless very rapidly.

Anyway, lessons learned:

1) Of course, don't use those Radio Shack Gold jacks!  Or re-tighten every week.  But don't count on remembering that, so just don't use them.  This is ESPECIALLY true for the speaker connection, because speakers are known to have significant vibration and movement.  Actually, I don't think I have ever had a problem with these connectors on amplifiers, and I think I may still be using them on the tweeter amp and bedroom amps, without issue, for years.
2) On the speakers, use the best connectors!  I installed Furutech locking bananas on Tuesday.  I had recently removed them from some other connection; I hadn't had enough of them to cover all connections when I set up the ABX in January.
3) When doing tests, don't overlook tiny changes.  If you change ANYTHING not related to the test protocol, because it needed changing, consider that change worthy of an immediate test.
4) Think about other things, about everything.
5) Temporary "opens" in the Krell to Acoustat wiring can cause the Krell to sense short and shut down.  So, if a shutdown occurs, that should be one of the first things to check...the tighness of the speaker connections.  The Manual actually says to check the speaker connections if a shutdown occurs, but I always thought they meant to see if those wires were shorting, not losing contact.
6) It may be good, in some way, that the Krell FPB is so paranoid or "finnicky," because it forces you to fix things and do them the right way.  Fixing things like intermittent connecters improves the sound, also, even for the amps that don't shut down.

But it's hard to feel that way, when Wednesday night, after just playing some Steely Dan, the amp was cooling down, and before I could play something else, had shut down again.

And, 3 times in a row, shutdown, actually as I was attempting to play Buffalo Springfield about 9dB below my standard level.

So, the moral is, Don't Draw Conclusions!  It may not be over yet.

Maybe the Radio Shack Gold jacks are not guilty as charged, but I would not use again on the Acoustats.

I may be mainly using my Hafler.  In fact, when I'm playing Krell, and it shuts down, I should not pout, just switch over to the Hafler as I can do with a press of the "B" button on my ABX.

[The saga continues in a later post.  I've discovered now that doing an Italian Tuneup fixes the Krell, at least temporarily.]

XSP-1

No doubt, the Emotiva XSP-1 is a fabulous preamp.  And in every way except looks competitive with everything on up, including the $30,000 preamps like the Levinson ML52.  It has now, after far too long, taken it's rightful place in my Living Room System.

The Moving Coil preamp must be about as quiet as it gets.  I'm finding it so quiet I can not hear any noise at all, even with my ear to the speaker, except under rare circumstances*.

(*No refrigerator or AC running, in the deepest night, and comparing the sound ear to speaker with the sound immediately to the side of the speaker after playing a chamber concerto at full loudness**, I could finally hear the faintest of pale high woosh.  This has to be consonant with at least 80dB S/N.  Yes this also shows how hum and noise free the rest of my system is.  **It was +1dB gain on the Emotiva .  I listen to as high as about +4dB so far--that's about as high as I need for the loudest I can stand.  0dB gain on the Emotiva seems a higher output level than the XPS-1 I was using before, for which I was always so gain starved I dialed in 4-6dB of digital gain in my Tact, which can't restore the missing information from a too low level digitization. )

So this is far quieter than even the XPS-1, which was far quieter than the (well known to be quiet) dB systems DB1HG, which is an order of magnitude quieter than my old tube preamp with the Jensen transformer I was using in the 80's.  (My tube preamp, which took two years to build, blew away the Audible Illusions 3 that a friend of mine had.  Sadly, I ultimately dumped the old power supply, though not as well built as I would have liked it was still working perfectly.)

With the tube preamp, I could hear the hiss from the listening position after any record ended.

Hiss might make music seem more expansive, increased depth, etc., but it's all fake.  Quiet reveal the real depth and dynamics that are actually in the recording.

So far I can't fault the phono sound.  I think it's better yet than the Emotiva XPS-1, which was better than the dB Systems and so on.  Records are revealed as never before, without anything added or exaggerated.  The first record I played, formerly believed to be worse than mediocre was stunning in fidelity at  (Rick Wakeman's Journey to the Center of the Earth).

It seems fully immune to turning the tensor light on, but not entirely immune to turning it off, a very tiny pop.  With the XPS-1 it was more like an explosion, which might max out the Lavry inputs (set to about 3V).

My biggest concern was this: as a result of running the XPS-1 balanced to my Lavry ADC (balanced is by far the best way to use the XPS-1, btw, according to my measurements) I can no longer switch single ended signals (from other disk players than the balanced Oppo BDP-205) passively, as I was doing before.  Now, I MUST run all the line signals through an amplifying device.  How much would other things loose.

THAT's actually what I spent most of the first weekend investigating, and my ultimately conclusion was high level inputs lose very little, if anything.  Though at first I thought otherwise, I finally figured out why it was sounding strained--it was clipping the input of my Lavry ADC set to 3V for 0dB full scale.  One always has to be careful about the gain settings.  I wasn't paying attention to the fact that the balanced outputs were putting out twice as much as a single ended input of the same level (and, as it turns out, the Emotiva doubles the level of single ended inputs when it outputs them balanced, that is the way it does the conversion, which is what one would expect but I hadn't been fully thinking it through).  I somehow had been thinking the Lavry would automatically adjust for this, so a 2V "pseudobalanced" (through an RCA to XLR adapter, mine was a Cardas) input might be equivalent to a 4V balanced.  That's the way that SE and balanced outputs often work--the balanced has double the voltage because that's what you would get with a equal differential signal.  But, no, from the Lavry's perspective 2V is 2V and 4V is 4V, whether balanced or unbalanced.  SO that is actually adding about 6dB gain to everything, including the phono, compared with no preamp, at the supposed "0dB" level.

But since I have an 0.5dB resolution gain control, I can now easily fine tune that for any input.

I try to adjust all inputs so that the peak reaches the -1dB level, but NOT the 0dB level.  Reaching the 0dB level on the Lavry is likely to indicate clipping.  If the peak never gets past -1dB, you are perfectly safe.

I have determined so far, I am safe with the Emotiva set to -2.5dB for the Oppo BDP-205 and -3dB for the Sony 9000ES.  This is with the Lavry at -10db setting (this makes something like 3.8V the 0dB level).  This puzzled me because with the balanced outputs of the Oppo, it is putting out nearly twice as much differential voltage as the Sony.  But I have to remember, the Oppo is going balanced straight through the Emotiva, wheras the Sony is being converted to balanced by the Emotiva, which is also doubling the output.  I've been meaning to compare these two players on SACD.  I have always believe the 9000ES to have a special place in SACD reproduction (when it works...) because it has the same true DSD converters as the original Sony SCD-1.  Everything after that went to sigma delta, which might actually be better, but isn't truly native DSD anymore.  Well the 9000ES isn't DSDx1, it's actually DSDx10 !  Sony was trying to see what they could accomplish with a purist approach to DSD at first.  So the 9000ES may be magic, but it's also possible, not yet verified by me, that a Good Enough sigma delta will be even better.  So I need to do this test.  From relatively casual listening so far, I can't tell, my FUD is telling me I need to keep the 9000ES.  Of course I am also keeping my Denon DVD-9000 for HDCD.  And the DVD-9000 has a special place for HDCD.  It is one of the VERY few players that implements HDCD as Keith Johnson specified--it almost doubles the peak for the "peak" mode of HDCD.  So that means that HDCD's actually reach almost 4V, which was why I've had my Lavry set to -10dB for some time--that seems to work for most HDCD's on the Denon to the Lavry.  Now, however, with balanced conversion rather than adapting, I'll need to cut the gain level to -5dB or so.  So far I've determined I need about -4dB but lower might be needed.

The 6dB of "bonus" gain provided by the unbalanced-to-balanced conversion would not apply to amplifiers with truly separate balanced and unbalanced inputs that have different sensitivities.  However I suspect most power amps, including my Krell FPB, have the same sensitivity for both balanced and unbalanced inputs, just like my Lavry ADC.

Convenience

It is also extremely convenient for me to have the volume control AND muting switch (another plus) right underneath the turntable.  Previously I had to jump 10 feet away to mute the system on my Tact digital preamp, or I had to flip to an unused input--which was very risky because if there were CD player signal on that input it would be 10dB higher in level than the XPS-1 I was using.

One exception is that it was much easier to set the loading level.  With the XSP-1 the loading switches are on the back.  Of course, on the XSP-1 this is good because ALL the analog circuitry is in the back, and shielded from the power supply AND digital controller and display.

It's also nice to have the push-button selector switches, and to have lighting around all the controls so you can see where they are in the dark.  It would have been nicer to have lighted labels on all the buttons as well, but only McIntosh with their glass panels has that.

One friend of mine is very upset that Emotiva did NOT include a polarity switch.  I think polarity is of marginal importance, but I would have liked to have had a polarity switch also--for convenience.  Most people can change polarity by reversing speaker connections, but it takes a lot of work for many audiophile systems (and mine, with 3 speakers on each channel).  For me, this is not an issue because I can change polarity in my downstream Tact digital preamp, which is easier from the listening position anyway.   If I were into the Polarity Obsession, I would not buy this Preamp on the principle that it doesn't have a polarity control, and such products deserve boycott.  Solid state high end big dogs like Levinson and Krell generally do have polarity switches.

Meanwhile there are many features that are completely unimportant to me, such as the built-in crossover.  Nothing like this provides close to the flexibility of my Behringer DSP's, and my next thing will be the use of 3 miniDSP units which, using digital FIR filters, can correct phase response and produce a linear phase Linkwitz Riley.

On similar grounds, I doubt I'll be much using the tone controls, but I think digitally switched tone controls are generally a good idea.

Having two balanced inputs is excellent.  Many historic balanced preamps only had one balanced input.  All in all, having 4 single ended and 2 balanced inputs is excellent, and I'm finding I do not even need to cascade and additional output selector switch as I was doing before even to include my two tuners in the lineup (I don't need to include the tuners because they are directly encoded in digital by the Sonos Connect units that see their fixed outputs).  Actually I have observed that the variable output of the Pioneer F-26 is sufficient inferior to the fixed output (depending on level setting) that I may prefer it going through Sonos anyway, and I decided to remove the F-26 variable output connection to the analog preamp so that it doesn't load down the fixed output any because there is no buffering in the F-26 output.  So that actually leaves me with one spare unused input!  (That sort of thing never seems to last long.)

Performance

Audiophiles have unfortunately been trained to believe measurements are meaningless.  I don't believe that for a moment.  Although I suspect I could not easily hear distortion differences below 1%, I take it as a given that, generally speaking, distortion is one thing that needs to be eliminated to achieve perfection.  And likewise noise should be eliminated.

And the Emotiva specified distortion levels, and also noise levels, are as good as I have seen anywhere.  My own measurements yield the same or better numbers (which confounded me, as the distortion level I measure is lower than the residual of the Juli@ soundcard I am using for the measurment).  This is especially true for the balanced inputs, where I have measured as low as 0.0003% distortion.

Meanwhile, Levinson specifies 0.03% even for their $30,000 532 preamp, as well as their others.  It appears, however, that the Levinson numbers are extremely conservative for the top of the line 532, but not very conservative for the "inexpensive" $10,000 model.  Stereophile easurements of the 532 preamp show noise and distortion about as low as the Emotiva.  Measurements of other Levinson models show higher levels of noise and distortion than the Emotiva.

The spectrum of distortion is also excellent, being confined generally to only the first two harmonics being visible above the noise floor.

Comparisons

The two chassis behemoth Mark Levinson Model 52 preamp is certainly fabulous.  However, these detailed measurments at Secrets does not show superior distortion performance (it shows virtually identical performance as the XSP-1 at 0.0003% THD, same as the Emotiva specifies for balanced to balanced, and that it pretty much mostly what the Levinson measures in the review at 4V output.   (Levinson specifes the 32 conservatively with the same 0.03% spec as used for their "cheapest" model, at $10,000, which does not exceed the spec quite as vastly.)

Now, the Levinson might be superior in other ways, but the only apparent one is the immense output levels, up to 20V RMS.  The Emotiva is only capable of 12V RMS.  I can't imagine needing as much as 12V.

I'd like the polarity switch, assignable input names, and 0.1dB assignable default levels of all Levinson's since the 90's.  As John Atkinson said, praising the 38, it is an audio reviewer's dream, and aren't all audiophiles reviewers?  I've long lusted for a 380s, not to mention a 32, but I wonder if I'd actually find their performance equal or even inferior to the brand new Emotiva.  I saw my beloved Classe CP35 fail, and before than I needed to have my Aragon 28k repaired.  20 year old gear is pushing it.  And the price and gamble aspects of it.

I have seen inadequate noise performance on the less expensive models and earlier generation models.  For measured performance, probably little before the current Model 52 will compare.  Old classics from Levinson and Krell won't make the grade--as the latest and greatest Levinson is a mere equal.  Spectrai??

The famous Vendetta SCP-1 claimed some S/N in the 90's IIRC.  However, that was at a very high output (taking advantage of the enormous headroom, as compared with the "standard" level for performing such a test.  Stereophile measured 82dB, A weighted.  The spec for the the Emotiva is >39dB, which is <3dB different.  My experience suggests the Emotiva is an equal in noise level, but I have not made a measurement (which...I do have equipment for...).

The Volume Control

One of the things that sold me on the XSP-1 in the first place was the "R2R attenuator."

I envisioned something like the relay switched resistors in the very expensive Krell KRC series of preamps from the 1990's, and the Placette passive and active preamps, as well as brand new hifi attenuators made in China and sold on eBay nowadays for about $129.  I also imagined this kind of attenuation in the top of the line Mark Levinson model 32 Preamplifier and all the preamplifiers Mark Levinson has made ever since.

Mark Levinson actually says "discrete" resistors and says they are switched by "analog switches" not relays.  But what are "analog switches"?  One intrepid DIYAudio'er enlarged the original ML 32 interior photo and matched the volume control "analog switches" to the these CMOS devices--which their manufacturer Vishay calls analog switches.

In 1995 even Rowland Research was praising the virtues of the (now outdated for high end) Crystal Semiconductor (now Cirrus Logic) CS3310.  The CS3310 is rated at typical distortion level of 0.001%.

Both that chip and the more updated Muses 72320 and the 2005-era Cirrus Logic CS3308 and CS3318, internally use precision resistor ladders (R2R).  I suspect the Emotiva uses the CS3318...notably that chip has the same 117dB noise and 0.0003% distortion specs as the entire Emotiva XSP-1 itself.  The preamp itself can avoid being cumulatively worse than it's weakest part because the volume chips are run differential balanced--along with everything else, which increases S/N.

I don't recall that the 90's era Krell KRC preamps with discrete resistors AND relay switching, came close to the noise or distortion of the 532 and XSP-1, despite using relay switching.  (Notably, David Rich was critical of Krell Preamp circuitry in the 1990's.)  It's hard to make discrete circuits as clean as the best chips available now, such as the LM4562 opamps that Emotiva uses (though, the OPA 211 could be even better, no audio gear uses that).  The best chips I just mentioned are the process of three decades of intense competition to build the best linear amp.  Older chips are passe (especially the one sadly used in my Denon DVD-5000's which I was hoping to use as DACs).

Meanwhile, it also appears, that electronic switching in attenuation has been tamed, though it is not perfect, and I'd pay a premium for relay switching given the same quality electronics, but it appears that right now the only way to have it all would be to build my own preamp that does everything right.

Now, that master of the universe Paul McGowan of PS Audio has very little nice to say about any kind of chip volume attenuator in this thread which started out as a damning discussion of LDR (light dependant resistor) volume controls.  A perfect illustration of typical High End Audio's stupid dismissal of the very high level engineering that goes into the best analog chips.  However, later on, McGowan is forced to recant slightly because it is revealed that the well regarded Pass Labs XP-30, designed by another master of the universe, actually uses the Muses 72320 chips.  McGowan then admits that a good enough designer can use them "properly."

And it turns out there is something very special about the Muses 72320.  At face value, the Muses has no better performance than the plain old CS3310.  However, uniquely, the Muses can be used in bypass mode, where only the internal switched resistors are used, and not the internal (and not quite the best) opamp.

None of the Cirrus Logic chips can be used in that bypass mode (I determined by reading the datasheets linked above).  However, the CS3318 has an internal opamp that is almost as good as the best opamps anyway, and it uses it in a clever way that can use gain as well as attenuation, thereby maximizing the performance by reducing the need for actual attenuation.

So, as you can see, little is lost by using one of the best chip volume controls nowadays.  And that is how the XSP-1 can be about as good as it gets.  The best IC preamps are better than mortals can design discrete, because IC's are so well perfected now.  If an IC preamp used the SOTA chip, the OPA 211, it would probably be better than anything that could be made discrete.

[I might add more detail to this review later.]