ASRC - Asynchronous Sample Rate Conversion [updated 7/19/17]

Nowadays I have trouble talking about asynchronous interfaces for SPDIF and AES because everyone else is only thinking in terms of USB interfaces.

USB interfaces are a whole other ball of wax I don't want to think about, along with I2S, as I have detailed in an earlier post, because they inhibit the chaining of digital devices and generally lock you in to proprietary software.  That kills the fun, and right now I can't use those interfaces anyway because my style of digital audio relies upon the chaining of digital audio devices, such as the Behringer 2496 DEQ units I use as crossovers and room correction, and the Tact RCS 2.0 I use for volume, balance, and polarity adjustment (though it is capable of much more).  USB and I2S assume you are connecting a computer to a DAC, with nothing in between.  In principle you could chain USB and I2S devices....but you'd have to write all new operating systems for every step of the chain.


Anyway, SPDIF/AES interfaces can also be implemented in synchronous and asynchronous versions, the asynchronous version generally involving ASRC -- Asynchronous Sample Rate Conversion.

ASRC has always seemed very fishy to me, because it essentially relies upon endless interpolation.  The data presented to the ultimate digital converter is not the data you started with.

No interpolation process can be perfect, and especially when in this case it involves guesswork about incomplete data.  Oversampling is entirely different because it is a kind of feedforward instead of feedback.

In some sense, synchronous conversion is perfect because it does not touch the contents of the data.  It only introduces tiny amounts of timing variation, which could be nearly eliminated by using buffers or other stabilization measures.

Here is a pretty good discussion of ASRC at What's Best Forum.

One digital receiver chip which implements ASRC is the Cirrus Logic CS8420.  While bit perfect transmission introduces no THD or dynamic range limitation whatever, the CS8420 is specified as having 128dB dynamic range and -117dB THD at 1kHz.  That's pretty good, and better than virtually all DAC's, but it is not perfect.  Further it seems to me the imperfection is increased by jittery inputs.

Benchmark DACs may have been among the first to include ASRC--they made a big point about it--but I'm suspicious that earlier ones also had this going back to Mark Levinson and possibly even Wadia (though Wadia seems never to have used the term ASRC).  High end digital pioneers implemented ASRC in their own FPGA's.

When I asked Kingwa about whether the Master 7 Singularity used synchronous or asynchronous receivers for AES and SPDIF and he replied Asynchronous.  However I'm not 100% sure he understood that I was not asking about USB.

In this little discussion, an industry expert says that ASRC is essentially PLL done in the digital domain.  That is exactly the idea that has occurred to me in the past month.  But once again, I must add that 'the digital domain' means that it alters your audio bits, and such alteration can never be done perfectly, and the imperfections relate to stuff like noise and ground loops in the signal that would otherwise cause only tiny alterations in timing.

On the other hand...Emotiva shows impressive improvement using ASRC.  The Emotiva Stealth DC-1 (a inexpensive sigma delta DAC including AES input and balanced outputs which I use for the subs) allows you to turn ASRC on or off.  I have generally turned it off, but maybe I shouldn't?

Emotiva has been using the AD1896 ASRC - same a Benchmark DAC 1.  (Not sure if that's true of Stealth DC-1 though.)

Amirm at WBF isn't impressed by the Jtest results shown by Emotiva.  He says ASRC needs a test to expose its weaknesses the same way Jtest exposes jitter caused by AES/SPDIF.

(I can't believe that Amir and many other posters I've enjoyed reading at WBF are now banned.  I made a rambling post to an ages old thread over there a few days ago, so apparently I'm not banned yet.  You cannot say that Amir is an objectivist flamethrower, like, say, Ethan Winer--though I liked him also.  At one point, I remember a great debate between Amir and Winer but that was at AVS.  So I wonder what's up over there.  It's not just objectophiles who have been tossed apparently.  Other forums were lots of interesting people are now banned include SBAF, over there it seems like over half of all posts are by now banned posters.  But SBAF always sounded tricky.)

Update 7/19/17

Here's one of the best discussions I've found about PLL vs ASRC digital input receivers (aka DIR).


It turns out that my new favorite DAC, the Denon DVD-5000, which was considered State of the Art in 1998 (a mere 19 years ago) uses the very well known CS8414 dir by Crystal Semiconductor, which is a standard digital PLL type of receiver (not ASRC.  The 8414 was a follow on to the 8412 which many still regard as the best of it's type.  The following CS8416 is generally considred not as good.  But all of these chips have a self-jitter specification of 200ps (not wonderful).  That specification tells us very little, however, about how incoming jitter is suppressed.

Within the ASRC type receivers, there is actual considerable jitter suppression done before the ASRC.  The ASRC is said only to encode a tiny portion of the incoming jitter, the "residual" jitter, into the data.  Which, as proponents of ASRC say, is precisely what the DAC chips themselves do.

(So, I ask, why don't they leave the DAC chips to do that, as they previously had, and simply use the "superior" jitter suppression to do what it can do without using any explicit ASRC at all?  I mean this as a serious question, however I would not be surprised if the ASRC chips handle this residual jitter better than the DAC's do.  But where are the numbers?  Where are the measurements?)

Generally the ASRC chips have much better self jitter numbers, like 50pS.  But the newest synchronous receiver chip, the DIR 9001, also has a self jitter spec of 50pS.  This tells us nothing about incoming jitter suppression, and the self jitter specs may have been differently determined, but it is possible that DIR 9001 is better than CS8414.  Or not.  Some people still seek the old Crystal receivers.

Both ASRC and PLL type receivers require some kind of buffer, or the pre-ASRC jitter rejection could not possibly work.  There is some argument at the above link as to how large this buffer needs to be, with some saying it can be quite small.  I think I can see why that might be possible in some cases, but I'm not sure if it can accomodate all cases.

This Hobby is Getting out of Hand

Reading the story of a guy who started loving Magnepan 3.6s and Martin Logans, and then discovered Apogee's, now his favorite of all.  And just a few years into his Apogee kick, he has a whole "forest" of Apogee's, and remarks... "Man, this hobby is getting out of hand."

Back at the AcoustatAnswerMan thread at DIYAudio, poster Kouiky writes incredibly dense, condescending and essentially content free prose relying on marketing buzzwords and inuendo about how Acoustat lovers are fooling themselves into believing Acoustats are the best speakers ever.  Sound Labs are the real deal he claims, but while criticizing others for fantasy he also hardly backs his claims up with any evidence, just rationales.  Blowhard!

This kicks off a certain amount of somewhat less content free arguments (though, from what I've read so far, no one except for me is really explaining why a 'flat' line source speaker like the 1+1 is so wonderful, then I get to my comments on page 106 where I don't do such a good job either).  The bottom line is that flat gives you choices, curved does not, if the speaker is curved what's the point of angling it?

Of course the big thing is price, I got my 1+1's for $600 plus $300 shipping, and a pair of Sound Lab Ultimate 1's go for $45,000, that's 50 times more money and approaching the appraised value of my home a few years ago.  (Since then, to be fair, I've spent another $1000 or so on Acoustat replacement parts, and my home is now appraised at $70,000.)

[Oh, there are cheaper models?  How can I again live with a small speaker shouting up at me from the floor?  I've gotten used to tall towers making waves throughout my little cottage.]

I'm perfectly open to believing the Sound Labs might be better based on reputation and the fact the company is still a going concern continuing to do R&D.  Though I should mention that several of the very experienced audiophiles I know think the Sound Labs aren't very good and the good ole Acoustats are probably better.  But they could be very wrong--by all accounts Sound Labs have gotten better and better and what I consider the best Acoustat product, the 1+1*, hasn't been made in a long time and even then was only created by combining panels that had been designed in the 1960's and 1970's.  I myself haven't heard Sound Labs at least since 1979 when I saw them at a store and I'm not sure I even heard them then, but they looked nice.  It's ultimately true that everything is a compromise, and anyone might like this or that better for unpredictable reasons.  So behind the curtain, some chose the Acoustats, and others would choose the Soundlabs.  Even technically there is some issue with using lots and lots of small electrostats in a big curved frame, as compared to big panels.  Little panels are indeed more subject to things like comb filtering.  Now, I am not saying Sound Labs hasn't solved this problem well enough for most people, and quite possibly me too.  But there might be some who like the other approach.

(*I can't say I've systematically heard all the others.  I only heard 2's and 3's once, and that's it.  But many people who have heard them all, such as Mr. Acoustat and E-Stat, have also settled on the 1+1 as the best model as well as something essentially unique in audio: a 1-way narrow and flat and tall line-source speaker.)

I used to believe I had to have 1+1's instead of anything similar (even Magnepans) for several reasons, one being I have such a small room the speakers need to be narrow (and also to provide viewing area for the TV for my parties).  Now I hope to have projection screen TV system soon, so then I could have significantly wider speakers, but it might still be hard to fit in the biggest Soundlabs, even if I had the money with no other pressing needs.

Poster E-Stat (who can also be seen at other blogs) seconds the notion that Sound Labs are better, but emphasizes the incredible value of the Acoustats...he still has 1+1's in his garage system.

E-Stat (possibly in another website I saw today) tells a microscopically short story of being at HP's house: three incredibly transparent systems.  Will we ever see pictures?  Apparently not that kind of transparent.

One point that is made in the Sound Lab vs Acoustat argument is how the Acoustat has quite an enormous resonance in the bass.  But it turns out that virtually all dipolar speakers do this going back the Peter Walker's Quad ESL 57's.  This is how they overcome the natural bass cancellation that a dipole would otherwise see.  WRT Sound Lab, in the 1980's John Atkinson remarked about the very large bass resonance they had.  Kouiky snorts that since then things have changed, Sound Lab now uses 'distributed resonance' (which he does not define).

The alternative to having a bass resonance is an electronic correction--which is precisely what Linkwitz does with his dipole speakers.  Along with Linkwitz most expensive designs like Orion, you must use a very complex electronic equalizer/crossover and multiple amplifiers.

I've always thought of Linkwitz as the greatest, but when I last heard a friends pair of Linkwitz latest top shelf design...the one that replaced Orion...I thought it was outstanding for a dynamic speaker system, and possibly slightly better than my 1+1's in immediate transparency, dynamics, and good soundingness (though I have equalized my Acoustats to sound equally good by adding in the 'Linkwitz dip')...but not enough so for me to bother with it and concurrently lose the Acoustat advantage in utility in my current nearly near-field 'electrostatic headphones' listening configuration.  I wasn't blown away, though possibly a better configured Linkwitz system would do so (better room, better source--it was a chromecast, and better electronics).  If I were to go to a better electrostatic system--assuming there is one--I'd want to be blown away by it, not just a small step up but a transformation.  And I think a tall line source makes a better 'whole house' system for background music and parties.

One very well heeled audiophile I knew in 1980 who could buy anything he wanted thought nothing had better bass than Acoustats, so he used Model 3's and 4's together as the substitute woofer for his Hill Plasmatronics.  He also had Magnepans but didn't think they're bass was nearly as good.

Anyway, some of my testing in the past couple weeks has been running the Acoustats full range (to see if that would make the amplifier shut down quicker--couldn't tell) and I was shocked, shocked at how good the bass sounded.  I've been equalizing and measuring and re-equalizing over and over since 2009 and still not attained the amazing bass clarity of the Acoustats in my room down to at least 40 Hz with no room equalization at all!  My thinking has run along the lines of snapping up two pairs of the now easily attainable 2+2's to use as subwoofers which would take no more floorspace than my two monsterous SVS PB13 Plus Ultras.  But the SVS are also useful tables in my small room, and I think the sub+electrostat matching thing is a great challenge.

In that other blog somewhere I read that yes, the Acoustats and most other dipoles have large bass resonances.  However, they don't excite the room in more than one direction, thereby avoiding all the room modes associated with the two other directions, which is in most cases FAR more important than the bass resonances in the speaker itself.  I concur and this is the essential part.  This is when the questions about HP's listening room came up because E-Stat said he was there hearing the Sound Labs and said the room was fairly small, but others said the room was supposed to be 25 feet long which eliminates much of the room mode problems.

BTW, here is the page which shows a pretty good guess about Roy Esposito's Air Mod.

From the description of what it does I'm not sure I'm interested, but I'm thinking some similar kind of re-working could deal with the problem I experience (the 100 Hz boom and the 120-200 Hz suckout) and if I could deal with it there it would be better than electronic, because it's a huge waste or impossible to fix a suckout with EQ.

Anyway, back to content free reviewing, the review of the Acoustat 2+2's in 1986 in The Absolute Sound by William T. Semple comes close to that (though, the Manufacturer's reply wasn't any better).  And much of what Semple does say is way out wrong, or just twisted to make it sound worse than it is, such as this, which strangely appears among the list of the speaker's strengths:

"JN wrote that the Acoustat was 'relatively neutral', a sort of backhanded compliment, which is, in fact, equivalent to calling the speaker colored."

Funny I don't recall seeing Mr. Semple reporting in any other issues of The Absolute Sound, and he writes a lot like HP.

But there are a few bits of truth.  The Acoustats are not seemingly as transparent as some speakers somehow, and they don't have the most extended high end.  Andy Szavo says if you find inadequate highs you can adjust the control.  But sadly the HF control doesn't increase the extension of the highs, it merely increases the volume of the highs (update: in the C mod, it may actually increase the extension a tiny bit, but it is already at diminishing returns, and making the adjustment requires messing inside the interface box not just turning a knob...I have never done it myself)  and I can report to you that the volume of highs from the Acoustat 1+1 is not only sufficient, it is slightly excessive, especially on the beam.  Only about 10 degrees off the beam is the quantity of highs not in excess, and that (circa 10-15 degrees off the beam) is precisely where to sit, because most of the alleged problems, such as head-in-a-vice beaminess, go away there.  At 10-15 degrees off axis, you can move around a bit without changing the balance.  And then you also have the opportunity to add a omnidirectional ribbon tweeter, as I have, which is pure magic, and adds back every quality the Acoustats were supposedly missing.

But I think the Acoustats are nevertheless sufficiently transparent to be moreso that 95% of all speakers, including about 95% of high priced (over $10,000) speakers.  Only a select few, such as Quad 988's, and Maggies that have ribbons, are more transparent.  Cones and Horns can forget it with very few exceptions (such as Yg).

And the way to deal with the tiny lack of high extension, is to add a super tweeter, which is not a curse it's an opportunity.  Of course dealers and manufacturers want you to believe your $2400 speakers are all you will ever need.  But if you are a real enthusiast, you will never believe anything like that about anything, and will add some kind of super tweeter, be it ribbon, plasma, or a specially made electrostatic.  As I usually do (right now I'm running the Acoustats w/o supertweeter because everything is a mess because of testing, and I can't actually say I'm missing much).

The slightly imprecise imaging of the 1+1's is actually The Absolute Imaging you will find in concert halls.  The pinpoint imaging of speakers with tiny tweeters is not at all like the real thing.

It is so unusual to have a full range speaker that radiates full range from most of the front.  When you walk up to it, it doesn't get 'closer' like speakers with tiny tweeters.  That makes you think it's less transparent, has less highs, etc.  But again, let me tell you, when you walk forward a few feet in a concert hall nothing much changes either.  So once again the Acoustat presentation is actually surprisingly like the real thing.

I have got to dig up the actual TAS review of the 1+1's, which Semple reports is in the Summer 85 issue.  Thanks him for that bit of info.  Semple makes that sound like a much more positive review, though they had not apparently heard the "C" mod.  Then Semple goes out of his way to actually put down that earlier review, comparing it to something that Julian Hirsch might write.  (That sounds a lot like Harry's writing to me.)

It wouldn't surprise me if Strickland had not kissed HP's butt.  They had a fundamentally different view of where audio was going and should go, with Strickland more sympathetic to the relatively (here we go again) affordable, and HP siding more with the no limits side.  And Strickland more (but not entirely) on the audio objectivist side, along with his friend David Hafler (who was not getting very much attention in the pages of TAS at all).

Problem was, however, electrostatic dipole speakers are not intuitive to most people, and they are much more dependent on proper setup (as in, you get nothing until you get there) than most speakers (which can be fairly good even without the intensive setup).  So Strickland needs high end audio dealers, but the dealers are siding with the more and more easily profitable 'no limits' audio, the kind that lets then get by selling 5 $50,000 speakrs instead of 125 $2000 speakers.  The rest of audio retailing sinks to the Best Buy model which demands easy price-based sales with no support needed.

HP's destruction of Vendetta Research was soon to follow.

Now though I've never seriously considered buying the $50,000 speakers, you can blame me for part of this.  Despite what I'm saying above, I've always enjoyed reading HP and the other high end magazines.  I've been a long time subscriber.  It doesn't bother me so much I'm reading about the unaffordible.  It's like reading a gossip column.  And eventually, things I never imagined having, things such as the Krell FPB 300, became available to me.

But then you could say for me the magazines weren't serving the dealer trade--as I imagine their primary purpose, and I remember when TAS and Stereophile blew past their original concepts and started publishing advertising.  At first and for some time, it was only Dealer advertising.

No, for me the magazines are serving as a gateway to the endless amount of re-cycled audio.  So they really are the 'racing forms.'

Things I do differently (as of June 27)

I'm back to the Dac 19 until I have time to retest the Master 7, or set up a 4th Behringer DEQ 2494 to allow extra delay in the bass line--and also, I would have 4 screens running, one with a spectrum of each frequency range, and one with the level monitor--see, I love these things, except they could be better, such as polarity for one and 0.1db adjustment and delay to 1000msec.

But it's not the same as before I returned from vacation (and soon acquired and hassled with the Master 7, now in hiatus) because:

1) I have subs on a different circuit.

2) I play FM tuners via Lavry AD10 digitization, rather than Sonos digitization.  I could not do this before, failed years ago and over and over, because of the subs competing for power to deal with low frequency noise with the Krell, causing Krell to shut down.  Old tuners have lots of LF noise.

3) I've gotten all Oppo connections, even 88kHz samping (which I had though was simply broken in the Tact for years, except the Tact and Behringer both show the digital looking normal) mainly by tightening the RG6 connections.  This permits me to play discs of all kinds and use the digital outputs.  The Oppo is a better universal disc player for many reasons, including that it handles DVD-Audio better than anything, and does Blu Ray Audio, and to have the digital output from it online is a top level breakthrough.

4) I have corrected the absolute polarity, which had been reversed since using the Dac 19 mostly--because it inverts, though I remember in the very beginning I had figured out that "in" on the setting button meant "in" correct polarity, which is the reverse of my intuitive expectation, then I promptly forgot that for the next couple years and was playing out of polarity.  I now have a simple polarity test.

5) I corrected the relative polarity on the subs.  As of yesterday I am using a polarity reverser connected to short XLR cable to the sub.  That way it doesn't require a tricky preamp adjustment in one channel followed by physically reversing all the other channels, a setup bound to lead to future mistakes.  This, of course, has easily profound consequences.  I have probably down adjusted the bass a bit because it was too much.

6) On re-examination, Sonos outputs do not need to be at -8dB to be 0dB "bit perfect" digital outputs. I had either misdetermined that some time ago, or some update changed or fix it.  Now I have all Sonos digital outputs set to Fixed and there is no clipping as I once noticed.  So now I am getting the full dynamic range through Sonos again, and it's up there with how CD's sounded before.

7) The Acoustats survived a tough ordeal (and contrary to my expectations for awhile) and are now fused at 3A, which should make them even safer.  I have determined the transformers needed for my broken spare interface and re-order.  When fixed, this interface will be modded to match it's pair.

8) Solid core 14ga silver plated teflon coated wire, essentially equal lengths in both channels (a first for me), now secured to Acoustat posts with Furutech bananas.  The wire softens the edge but doesn't lose resolution, compared to Canare 4S11 I was using, very slight difference if any.  The locking bananas are essential and better than previous ones I was using.

9) I remember certain things about using the Krell.  After using breaker to shut off (always turn off at front panel first) wait for complete cool down.  Or, if you fail to do that, and it starts shutting down, do that and remove from power for awhile.  Otherwise, it seems (don't trust me) to be safe to turn off from the front panel when warm, make some change, and then turn on again, without any waiting beyond what it takes to make the change.

10) Because it sounds so much better now for all of the above reasons, I am enjoying actual digitial inputs now.  I haven't used re-sampling from my digital players for awhile.  For playing CD's I can use the Integra Research RDV-1 and take it's digital output (it has the Apogee clock which is probably better than Denon DVD-9000).

11) I now have a fully working DVD-9000 online, it plays DVD-Audio and everything.  My old DVD-9000 has become a spare DAC for now.

12) The super tweeters are off for now.  They need to have their time alignment fixed, and that depends on the disposition of the Master 7.  I now suspect they were 300 ms delayed, which is terrible.

13) I have all room remote control of the L-1000T thanks to a new repeater, and I've reprogrammed the kitchen remote for that.

14) Batteries were replaced in living room remotes, two needed serious cleaning including the DVP-9000ES remote.

(And all this is not counting the Master 7, which seemed to work pretty well until the day it was taken offline, being by far my best sounding DAC, and was part of lots of listening and testing as well as frustration.)

That's the scorecard of actual and mostly huge and objective improvements, just in the past 6 weeks.

What has driven a lot of this in the past 3 weeks is random shutdowns of my FPB 300.  Each time I figure out some fix which seems to help get it past something, and it's back.  While this has in some sense helped lead to some improvements, it's been very frustrating and not helpful to the rest of my life.

Sunday I ran the Dac 19, which had never caused problems before in 2 years, and it was performing flawlessly, until midway through the movie Steve Jobs the amplifier died again.  And wouldn't restart.

I changed dacs to the Denon DVD-9000, and set up a dual DEQ system for the subs to delay them about 330msec to account for that much delay in the DVD-9000.  It played for awhile, then shut down again (playing bass heavy Santana Supernatural, but crossed over at 100 Hz).

So now it appears shutdowns occur for all 3 dacs, so the best explanation remains that there is an intermittant problem in the amplifier or the speakers.

Now, shut down a third time Sunday 10 minutes to midnight playing Supernatural again through the Oppo.  This time (not always before though) it shut down while apparently the DVD-9000 was muting--I heard the relays click and a moment later the Krell relays clicked and it shut down.

Tests I finally did last week showed that either speaker connected to the right channel caused shutdown, but neither speaker connected to the left speaker does that.  All that time I was using the apparently correctly functioning at the time Audio GD Master 7 Singularity.  I should have done those tests the first day.  By the time I had done them, I had already desoldered the HF transformer connection for testing, then I had to solder it back (because, overlooking the amplifier, I immediately and wrongly jumped to the conclusion that the speaker had been gone bad.

On Sunday night I duplicated the shutdowns using the Dac 19 Anniversary that I bought two years ago and has never been a problem, AND on the Denon DVD-9000 that I had been using for a year and concluded was both my best CD and HDCD player (but being semi-broken, it won't play DVD).  You can't say the output of the DVD-9000 wasn't correctly engineered--it was Denon's statement piece.

Since then extensive speaker play of the left channel only on very demanding music has caused no shutdowns at all, many hours so far, whereas on both channels (and right only) shutdowns were occurring in about 5 minutes except at low levels they might go longer.  (My 'rebooting' fix did seem to give me a longer playing time for awhile, but days later the problem came back.)

This problem has got to be a failure in the right channel of the amplifier.  I will be using something else until it gets another factory repair.

One thing: if no failure occurs in a brief test (maybe few hours) you can't tell if you have really fixed something, even if last time it happened in 5 minutes.  That's what intermittent is like

So The Thing which means something is failures.  When failures occur, you can be sure something bad was in the chain then, which you can then narrow down by elimination, replacing with other things.  But once again, if failure doesn't occur in limited testing you can't be sure.  But if you find something consistently in the failure path, with before and after changed, it is most likely the culprit.

Master 7 thumping and then channel dies

On my 70 foot coax connection to the Tact, and AES after that, the Audio GD Master 7 Singularity sometimes makes thumps up to -20dB, but more typically only up to -60dB, when tracks change, when the Oppo is turned off, etc.  I suspected this what happens when the digital signal clock from the source gets off the beat in some way and the PLL servo follows for a while before deciding there is no PLL lock and then it cuts off entirely.  Before cutting off entirely, the thump occurs.

DVD-Audio discs are the worst for this kind of thing.  HRx discs seemed less affected.

I played a great HRx disc over and over, and zero thumping at all.  The spectrum of the input to the DAC (shown on one DEQ) was matching the spectrum of the output of the DAC (shown on another DEQ).

Then, the next day, it appeared on the DEQ spectrum display that over 20 hours there had been a -20dB thump from the Master 7 going down to very low frequencies, while the Oppo had been turned off.  I turned on the Oppo and now it seemed the Master 7 was not picking up 24/96 originating from the Oppo at all.  I cycled through inputs on the Tact, and sometimes it would pick it up, and sometimes not.  This was the first clear sign of a serious problem from the Master 7 since I had hooked it up using AES.

Checking it out on my Fluke 8060 meter, I noticed that the left channel, still working, did make "thumps" up to 0.2V when I was changing sources on the Tact, though usually only 40mV.  And when those thumps occur, they only slowly, over seconds, return to 0V.  The DC servo is set to a very low frequency.  For practical reasons, I'd prefer a faster return to 0V, otherwise DC is being pumped out into my speakers.

But the left channel was no longer working at all, no longer putting out signal, though it was still thumping.  Perhaps it got 'locked up' somehow.

I've taken it offline.  I need something extremely reliable.  Will retest later.  Now I'm thinking the whole episode since Master 7 arrived has been enormously frustrating.  Now I'm again beginning to worry it was behind other problems like the shutdowns, though I actually don't have any clear evidence, other than the thumping, and now the apparently dead channel, that it had been doing anything wrong after I connected the DAC itself using AES.  I'm don't know what I'm going to do with the Master 7, can I get it fixed, can I get as much as half of my money out of it???

Hooked up my Denon DVD-9000 as DAC.  It is very sweet sounding, but I fear not as magically transparent as the Master 7.  I suspect the Denon could be much better with a better analog output stage and balanced outputs.

Back when I ordered Master 7, I was thinking of Kingwa as a sort of latter day James Bongiorno, a charismatic and uncompromising designer making legendary products.  But Kingwa is apparently doing much better, as it seemed James left company after company and finally his two signature companies GAS and Sumo failed, whereas Audio GD has continued for more than a decade now.  So maybe Kingwa got it right.

There was a downside I know of.  GAS equipment was not the most reliable.  I 'failed' a Grandson amp once merely by power cycling it too fast, it was killed by its own servo loop.  Later I had endless RFI issues with my own Son of Ampzilla, which I was never able to fix (I still have it in storage many decades later).  Seeing the way it was built with lots of plug-in boards made me wonder how robust it was.

Perhaps what was missing was better production engineering.  Perhaps Bongiorno had taken on and/or delegated too many things to people who were not qualified.  There is something to be said about a company with more than one good engineer, good enough not to always be a yes man.

Now I'm wondering if Audio GD doesn't indeed follow suit.

A friend of mine, reading about the Master 7 Singularity, thought it might better to use the standard SPDIF receiver chips, custom receiver programming is work for a team of engineers, checking each other out, beta testing, and so on.  I told him custom digital receivers were nothing new, Levinson and others were doing that long ago, the Levinson 360S also uses a custom programmed receiver.  Recently I've been wishing I had bought one of those instead, I'm thinking it wouldn't have issues like I've seen, and it was fully tested and looked fantastic in test by John Atkinson.

Update: I connected the Denon DVD-9000, which never had a hitch reading the signal through all my digital stuff from the Kitchen Oppo.

But it has a large latency, over 300 ms, that I can't correct for with my DEQ.  I'll need another DEQ, that still seems to be the best option.  That will give me more parametric filters too...

I learned to like the sound of the 9000 after some changes.  But the latency makes it useless.  I heard the latency right off, but I didn't measure it until much later, after coming to think for awhile it wasn't there anymore--but on speaker pop it was like two separate sounds, bouncing back and forth.  But for some of the day before learning to like the 9000 I had been listening to the DAC 19 again.

I found the SpeakerPop sound to be a good way of setting the alignment by ear.  I think I can do that within 0.2 ms or so, though I actually set it to 3.79ms, a little lower than before.  I need to measure it of course but that requires either the Tact setup or REW or a scope.  I tried a new scope app and it wasn't much use for lack of triggering.

But there was a problem doing the 88 kHz with the Dac 19 again.  But I then found I could fix it by re-tightening the F connector that connects to the cross attic RG6.  Perhaps that was why the Master 7 hadn't been picking up the Oppo signal well, though that didn't explain the channel dropout: I'm thinking that was caused by me flipping through the Tact selector a few times, making those little thumps I described before (and if they stayed little, I could live with that, and it was seeming that normal DVD-Audio track changes and that sort of thing were only -60dB or so LF thumps.  But when I got home, possibly some noise on the line had caused the -20dB thump clearly visible on the peak hold spectrum, and then the Oppo connection wasn't working on the Master 7 reliably, possibly explaining that.

BTW both the Dac 19 anniversary and the Denon are completely thump free.  On the Denon there seem to be endless relays clicking for this and that.  But there is never a thump.  The Dac 19 has no relays but is still thump free, even when I was messing with the F connector aformentioned getting 88kHz to play, it was a live line going in and out until I finally got it tighened correctly but no thumping--a very difficult case (the very case that had probably caused the -20dB thump on the Master 7).


I'm now hoping Master 7 "fixes itself" like the Krell did simply by being powered off long enough.  We'll see.  Otherwise, I'll have to see if I can get warranty repair before trying to sell it.

At it's best, the Denon DVD-9000 is nearly there, and it clearly has more resolution than the Dac 19, but more closed sound which is actually quite nice and sweet and helps you enjoy the music.  I'd be playing it right now, overall I consider it better than the DAC 19, except the midrange delay muddies up everything, and even 300 + 3.79 ms adjustment isn't quite enough to fix it, and that's more important than anything.

I realize now the Denon DVD-5000 on the super tweeters probably has a similar delay, meaning that I would be better simply delaying the bass line.

But if I can get the Master 7 going again, and weird problems don't come back, I could put the Dac 19 on the tweeters, which had been the plan (though, when the Oppo line wasn't working, I was thinking of putting the Dac 19 into my analog resampling group).

I've been listening to straight through digital, and I think because of finally getting the polarity correct, after who know how long, loving it more than ever.  I even made it through the 2017 Grammy collection just now.

Klipsch vs Altec

Just surfing around, I found this very interesting discussion of Klipsch vs Altec, two major (and very long lived) manufacturers who have made well regarded horn based speakers.

In a 1965 test, the Altecs are said to have originally sounded colder than Klipsch, but got better with certain improvements in the 1970's, and now some posters prefer them (notably, the Model 19).

Klipsch has also gotten better, it seems the K55 midrange driver sounded bad above it's range and was not crossed over steeply enough, a later revision added steeper crossover which improved sound considerably.  Many Klipsch owners had been using Altec midranges to get away from the K55.  Similarly a crossover revision added steeper crossover for the K77 tweeter, which didn't sound good low in its range.

Rebooting the Krell

Every owner of a FPB 300, FPB 600, or subsequent Krell amplifier should know this.  I don't see it in my owners manual, so either you have to figure it out (as I have done...though I think I read about it somewhere but can't find the original discussion now) or, contact Krell (using form on website) with a problem, and they may tell you to try this first.

Manually flipping the breaker at the back of the amp, and/or unplugging from the wall for awhile, is one way which can fix certain kinds of problems, notably the problem I had been having for the last 3 weeks in which I blamed virtually every other component first (I should have thought of testing the Krell back during the first panicky week, but since it just came back from service, since I know it can output well in excess of the 5 amps required to blow Acoustat fuses, and I didn't want to believe the Krell was at fault, perhaps only input/output cables, AC power, the speakers, etc).  The amplifier was shutting down after less and less time, first time it shut down after several hours, then almost an hour, and finally it took a mere 5 minutes to get to shutdown.  (It was not blowing the breaker in back, that is serious stuff and you must contact Krell then.  My FPB 300 was shutting down with the power light still illuminated, and no other lights, and the manual says that means either inadequate AC power, or shorted speaker or cables.)

Finally, just as I was about to contact Krell, since it had finally seemed I had proven the amplifier was at fault because the problem stayed in the same amplifier channel even after I changed both the inputs and outputs to the ones normally used by the other channel, using a now dedicated 20a line, it occurred to me I should try the unplugging trick.  So I did, on Wednesday night, 3 weeks after I started having problems.

Most recently it seemed that using the right channel of the Krell would cause a shutdown within 5 minutes.  But after unplugging the Krell for a few hours, I played 90 minutes full tilt without a problem.

This is pretty good evidence this makes a difference, perhaps less that I have finally fixed the problem, but I really hope so of course.

I think what happens is that the Krell while still AC powered saves information about the bias voltages required to reach different plateau levels.  This enables the plateau thing to work as advertised.  But if the Krell is subjected to a catastrophic situation, one in which fuses blow, for example, it can store bad info, and this bad info leads to future misbehavior.

Now, this could simply be a bug in the original plateau bias system (which is managed by a chip I've seen on the schematic, I haven't looked closely but I usually call this "the Krell computer").  Maybe later amps fixed this "bug."  But it's also a kind of feature.  If a user is doing bad stuff, such as running against AC power limits as I think I may have been sometimes doing (until I moved the subs to a different circuit this month) the company wants you to contact them so they can advise you to do things differently.

Also, take this as a correction or addition to the stream of posts I was making this month while trying to track down this problem.  After I had given up trying to tweak my way around the shutdowns, I "determined" (incompletely and falsely, apparently) that the Acoustat speakers were at fault.  I had seem the Acoustat MK-121-C interface fail before--that happened to me in 2009.  That time, instead of the amplifier shutting down without any sign of distress, it blew fuses.  It blew fuses more and more often, and the problem quickly went away when I replaced that interface unit with a New Old Stock replacement interface unit I purchased on eBay for $75 more than I had paid for the pair of speakers with two interfaces.  It seemed like a bit of a ripoff then (it wasn't at all, such things are unobtanium and only a distant dream now, I was simply amazing lucky, as if the heaven of Acoustat lovers wanted to keep me in the fold) but I thought it was well worth it, now that I had fallen in love with Acoustats.

Let me add, though there has been some great fear at times, mostly this month has been lots and lots of fun....trying stuff, seeing what different things do, and being forced to play more music because I had to do more tests.  (Sometimes, quite often actually, I don't play music because I'm feeling down and I don't want to take the trouble to figure out what to play--that's the hardest part for me.  But when I've got to figure something out, I have GOT to play something.)

My biggest fear actually was that I'd be stuck with a somewhat malfunctioning new $2500 dac which the factory wouldn't fix (because they aren't responsible for complex SPDIF setups or something) and couldn't unload either because it was customized and when I told any potential buyer of the problem I was having (that is always my policy btw, full disclosure) they'd move on too.

So when it finally appeared that the DAC was not the source of the problem (because problem went away when I disconnected the right speaker) I was inclined to immediately go with the new idea that the speaker was at fault, even though I did not check to see whether it was the right speaker OR the right amplifier channel that was doing this.  I should have tried the right speaker on the left channel.  An obvious thing to do to a thinking audiophile, but I wasn't thinking too clearly.  Also, I strongly didn't want to blame the amp because though it has been fully repairable by the factory so far, it's also costly and huge amount of effort to pack and ship the amplifier back.

I finally did do the required test this week, and found the problem always remained in the right amplifier channel even driven by the left DAC channel and playing the left speaker.  For a day, I had been planning my message to Krell.  But when writing a friend about the problem, the reboot idea came back to me.  I did try that last year before contacting Krell, but it didn't help last year, the amplifier truly needed the Capacitor Service it got, it was 18 years old after all, and came back operating better than it ever had before in my 9 year experience (leading me to believe now it was actually a bit off when I bought it).

This time, rebooting fixed the Krell, I hope, and problems will disappear for long enough to consider it a perfect repair.

Update: Further investigation has revealed the Krell does indeed have an 8Mhz microprocessor in each channel to control the plateau bias, monitor DC offset and other bad situations, and prevent using the amplifier in different world region than intended.  This is discussed at DIYAudio.

The microcontroller in FPB 300/600 is a 68HC711E9.  The generic form of this now discontinued microcontroller has 512 bytes of ram, 512 bytes of EEPROM, and 12kbytes of EPROM, along with 8 channels of 8 bit A/D conversion.  The specific 711E9 version seems to more like 10 channels of A/D conversion, or so it seems from the schematic.   They connect to various sensors and circuits in the amplifier.  Other than this microcontroller, in some ways the FPB is simpler than other amplifiers, because it does not have specific protection circuits of various kinds like typical amplifiers do--where often most of the complexity is.  The microcontroller handles all the protection issues, and can shut down the amplifier with various combinations of lights lit.  I wonder if this microcontroller isn't a descendent of the Motorola 68000 series originally used in Mac and Amiga and the earliest Sun computers.  The first Amigas also ran at  8 Mhz.

I don't know this as a fact, but it has seemed like that 512 bytes (or more) of ram inside the microcontroller could get corrupted by certain situations, requiring the power cycling I have described.

One thing I might have done to corrupt it was this.  I was changing cables and for some reason I had to get underneath the power cord.  So I flipped the back breaker and temporarily removed and replaced the power cord, then flipped the breaker back on.  This would not have been a problem EXCEPT for one thing: the amplifier was already hot when I did this, and I went back to full power operation immediately after restarting.  If the amplifier does some sort of calibration when AC power is started (as I am guessing) if you reset the breaker when the amplifier is warm it might get the bias calibration wrong.  Krell advises using the soft touch power button to go into standby instead, but of course you can't do that if you are removing the power cord.

From now on, if for any reason I remove power or flip the breaker on the back, I will wait until the amp cools before restarting.  If this is the actual cause of the shutdown issues I was seeing, it's a real hoot because I had so many other ideas.

Now I remember a similar situation occurred just before I put the amplifer in my air conditioned storage building in 2011 for 2.5 years.  That time I was trying to reduce "clinking" by moving the amplifier to the left side of the room.  So I flipped breaker, unplugged and replugged in the new location a couple minutes later.  Then I turned on amplifier while still hot, and it shut down within a few minutes.  Fearing a major problem, I put the amp away.

When I took the amp out of storage I had the nerve to try it again.  And then, except for the right channel always getting much hotter (a problem fixed during capacitor service in February 2017) it ran fine for almost a year.

Peterson Information Theory

(Named after me, of course.)

In Peterson Information Theory, the composition of a composite signal is the important thing, not what can, under some circumstances, be reliably detected later.   In other words, the information is what goes in.  A fairly intuitive notion, actually, until about 1900, when the importance of the observer became paramount in many fields.

So this is similar to how many subjectophiles believe that analog has infinite resolution, whereas digital has finite resolution.  However, that intuitive believe, in the full context of PIT becomes very qualified.

For one thing, at worst, properly implemented digital has incredibly fine resolution, about 100 times finer the the apparent quantization, for various reasons, which in the case of 24 bit digital is pretty mind boggling and probably adequate to be considered "infinite."

For another analog itself is limited at many levels, even if you discount "noise" as PIT tends to do.  Noise does not "obscure" simply because PIT doesn't involve itself with "obcuring," there is a kind of belief that in the infinitude of listenings, all the obscuring will be seen around in one way or another, freak unrepeatable occurances perhaps but important to the evolution of feeling.

One obvious level of limitation of the supposedly unlimited resolution is the quantum level, though that is quite far away.  But any reduction in voltage, say, does actually reduce information in that way, and it turns out, in others.

Many aspects of circuits we deal with actually do not involve fully continous phenomenon.  For example, both magentic tape and analog recordings do not slice at quantum levels, even at best they slice at molecular levels.  So now we are up quite aways from "infinity" or even the 10e-33 of quantum levels.  Infinite resolution there never is.

But I'm not claiming infinity, just a different way of scoring.

And so it matter that the DAC actually attempts to do 24 bits, or just fakes it.  No question about it, true PCM chips like 1704 "attempt" to do 24 bits in that there are resistors that get switched in and out to do that.  The attempt falls far short (perhaps 20 bits or so actually accuracy) because thermal effects and others cause everything not to be as perfectly calibrated as when the manufacturing was done.  But in PIT, that doesn't matter as much as the information not being there, for that lack of 24 bit performance is caused by an "obscuring," not by a lack of doing.

But all kinds of Sigma Delta systems never ever try to "do" high accuracy.  They fake it by doing low accuracy lots of times with feedback so they ultimately achieve something like high accuracy.  But that is what PIT views as not actually doing it, but faking it.

When things are so faked, they lack their attachment to reality, they become unreal in one way or another.  So Sigma Delta reproduction is cleaner than clean, cleaner than the real thing itself, because it is built on repeated self cleaning, rather than just doing things right, or as best as possible, in the first place, with maybe a little crud added here or there in the process.

When we demand that each layer of a system be truly crud free, we cannot achieve what we set out to do, the crud creeps back in some other way.  Best to create processes which create only very little crud in the first place.  So we have balanced circuits, cascodes, Class A operation, differential, etc.  Those are all low crud.  Feedback is only a way of taking errors in one domain and shifting them to another domain.  It may be needed in some cases, but best be low crud in the first place.

Anyway, that's my unscientific irrationality spelled out, and why I like no feedback (or non-global at least) and true PCM such as R2R type DACs.

That's why I wanted a Master 7 Singularity, not because I wanted to experiment with NOS or I2S.

The Master 7 has a highly sophisticated non-feedback direct coupled semiconductor circuitry, combined with dual differntial 1704's--a configuration which really helps pull out more of the true 24 bit "doings" going on in there.

Mind you, op amps are so small, feedback around them is probably ok, not unlike local degeneration. The only op amp that should be used is the best, OPA 211.

Tubes are mostly used as coloring agents, not to make things more transparent but to add tint or bloom.  Tubes can in some cases be done at the state of the art, but that is exceedingly rare.  I know of some almost unique tubes and designs up there.  Allen Wright's balanced RTP3D preamp looks pretty good, that's the least obscure, then there are some ultra rare high current microwave tubes, rarely done (but never to state of the art, which would include full balanced) an extremely tricky to work with,  and balanced OPA 211 with passive EQ between amps (no product that I know of does that).

Oversampling is not bad because that's not feedback, at most it's feed forward, or more technically, I see it as the ultimate end-procedure of the digital process, which helps enable the full potential dynamic range.  It's not endlessly overloading tail catching feedback.

Strangely 1 bit may be sort of ok (at DSD128 rates) because it is actually doing what it is doing, one bit is a perfect 1 bit.  Any multibits until full multibit is only an approximation, which must endlessly dither to serve the present physical world, and therefore strangulate the past with the present, rather than letting the past flow freely in.

*****

Thinking about the above, I wonder if I haven't become something like a Schitt salesman.  They are also in the anti-delta-sigma camp.  I don't think ANY true audio objectivists I know (about) would consider this anything but the usual subjectivist crap.  To a fully died Objectophile, all that matters is the (measured) distortion and noise, and getting there the cheapest way possible, which is delta sigma.  Delta sigma dacs became popular when they could approach and then ultimately exceed the performance (S/N, dynamic range, linearity, and distortion) of the R2R chips.  And that became the big commercial end of the Burr Brown PCM 1704, though it appears they were still making a few as late as 2010 or so still being under intense demand by high end companies.

There is a growing list of High End audio companies that have embraced the pro-PCM and anti-Delta-Sigma ideas like I have presented, including:

Schiit (their statement DACs are all PCM and they strongly defend this)

Audio GD (they had their feet in both waters, but are seem to have more fully embraced PCM solutions now, if nothing else to own the low cost cult market)

MSB (they make multichip PCM converters from high to super high prices)

CH Precision (with their megabuck products, you can have it either way)

At the end of the day, I won't disagree with the idea that speakers and room acoustics are more important.  If there is a there here, it probably isn't that big.

But somehow I've fallen into the trap, and I don't necessarily feel good that it has distracted me from the more important speaker adjustment and EQ work.

But it has also been a learning, it's a little bug to make you keep listening, and I've had lots of fun exploring my edge of the world.

At the same time, there is nothing really Objectively wrong about the PCM 1704 chip, it has adequate measured performance, anything that seems to have Even better performance (like the latest ESS chips for example) isn't really necessary.

It seems to me that once you attain "adequate" objective performance, you ought to be free to explore your own angles without too much sneering.

Many people feel that they ought to be able to explore their own angles no matter what, and measurements don't matter, etc.  This is where I do not go myself, but if other people want to go that way, it's their hobby.

The zero jitter synchronous DAC

Just to show a fully synchronous (as I think I like) DAC can be made to have zero jitter, along with the usual full information preservation of fully synchronous, consider the "memory dac."

In the worst case, it could simply store entire songs to memory, then play them out.  That's the proof that this is possible.  However an open line that just stays running digital audio for hours...you wouldn't want to wait until the next day to play your audio would you?  So, while it's possible to to have a simple spdif DAC with an absolutely perfect (or at least from the stanpoint of the 'better' DAC) clock, at minimum you have to change the clock speed in the dac to match the source over any reasonably long amount of time.

But it wouldn't be hard not to have it that hard.  Or just the obvious thing to do is keep a sort of clock quality whenever a digital SPDIF/AES line is locked up, a notion of how much the clock has moved around, so how much needs to be allowed for.

The quality analysis is used in figuring the size of a buffer needed to keep up with changes.  Then there may be some lag when any playback is started, because of allowances for small irregularities.

Now in this scheme, you can't have a perfectly on-time clock, you need to set the ultimate clock based on the analysis of the line as well.

And in worst cases, you may need to make slow changes to the clock rate to keep things going.  But that's very small changes, and very slow changes.

OK, so maybe not exactly zero jitter.  But it can be zero above some very low frequencies where it is even more inaudible than other frequencies.  (Remember, digital jitter being important in audio is just hype, it isn't really, though the hardware obsessed audiophile just wants to make everything perfect just because, I draw the line at two way digital interfaces because that's a loss of flexibility for me.)

Now this quality thing could be remembered, perhaps until you give some kind of reset, or when the cables are changed.  So after the first time out it has pretty good knowledge of the source clock.

AES/SPDIF Forever!

I am deeply philosophically (as well as somewhat economically) opposed to the use of 2-way interfaces, including USB and I2S, as a means of audio interconnection.

First, it must be said that AES/SPDIF are perfectly fine.  Under the normally specified limits (is it 30M for coax, 200M for AES ?) these can be shown to produce typically 200 pS jitter (my complex system does that), and that is 50 times below demonstrated audible limits in pure sine waves, and 100 times demonstrated audible limits in best case music on headphones only (famous AES publication).

In my system a typical input goes through 4 digital interconnections, 3-4 of them AES and the other SPDIF.  I could add a few more AES layers, and it would still be 200 pS.

I banished Toslink only a couple years ago (and I had actual glass Toslink cables) because Toslink transmission is measureably worse, above 500 pS, but not because it actually makes a difference.  I tried an unbalanced coax into my dac, which I was afraid might cause ground loops, but it didn't anyway, so I went wired all the way, wired is demonstrably better, and no sacrifice in this case.  Complex systems like mine (which uses some prosumer equipment) generally work best with AES, thought, technically, there's a small loss in bandwidth that comes with the signal isolation, but much less so than toslink.  But coax seemed fine with the Audio GD Dac 19 (I had coax muting issues with the Master 7 Singularity, though I haven't retested since enabling the PLL), and the Onkyo and Denon DACs I have used.

Anyway, this transmission jitter, produced by a dumb detector, is nothing in the final analysis.  Any decent receiving system using PLL or Asynchronous reception can in principal reduce effective jitter to zero, so whatever jitter ultimately appears in the output could have been avoided (personally I prefer PLL which I think of as good enough, and it uniquely preserves the original data, which somehow I think is important, I  will admit though I'm not an expert on what Asynchronous can do).  Most jitter that we have seen has been highly avoidable with AES and SPDIF.  And down below 200 pS, it wouldn't matter anyway if the DAC manufacturer did nothing about it.  But certainly high frequency jitter can be reduced to zero, the only frequency variation being the comparatively slow changes of the source clock, which may have to be accomodated, but that very low frequency (much less than 1 Hz)  jitter is even less audible, within the typical bounds.

[There is acutally a matter of trust here.  We trust that the source of the transmission has a very stable clock, since in the end that is what is driving things, the other end has to follow in the long run, and it's best not to be snapped around.  In the beginning CD players and the like may have been among the most expensive things in an audio system, and have the best clocks.  Nowadays, it's typically to have inexpensive front ends, and have all the expense in the DAC system, so the DAC manufacturer would like to have it their way in the clocking...I think that may be possible with with USB and I2S...but personally I'm fine with trusting the front end, I always attend to that, in my AES/SPDIF practice.  Or I trust another one of them...  Rolling sources is my fun, you see.  You who use cheap sources typically use USB or I2S now anyway, so keep on with your mega expensive dac clocks and so on, and never mind me, though I suspect that any non-super-cheap device is going to be OK, have long term clock drift at essentially undetectable levels anyway, so good for those who don't feel the need to bother with the fancy back end anyway.]

I2S itself is a two wire interface when computer systems have been going one wire.  I2S was designed for doing communication within one box.  It requires setting up a master-slave relationship, which in an interconnect system really requires software negotiation.  Now you're talking proprietary software of all kinds, drivers, stuff you will endless have to buy and to mess with.  And USB has already proven itself to be that way manifoldly.

AES/SPDIF has been plug and play since the beginning, it is an open system and easy to understand, and philosophically there's something I like about the free flowing nature of it.  Easily divided, endlessly rerouted, distributed, switched, split,  (not combined).  That's the kind of flexibility anyone who is messing with a complex system wants.

But as always, the industry would prefer to sell top to bottom systems, wipe out all that complexity, and the drive to systems like USB and especially I2S exactly does that.  Getting to any kind of distribution beyond 1 to 1 with I2S...isn't that pretty difficult?

In my view, that's back to being the victim of industry, rather than the beneficiary of it.

AES/SPDIF is good enough!  Let's stick with that!

More Rational Audiophile Hobby

I strongly agree with Archimago's comments here.

As per my comment there, I will try to avoid using the term The High End.  However, I think it's reasonable to use a term like 'serious' audio.  Even a $39 streamer can be 'serious' audio, it's not a matter of price, it's a matter of being oriented to technical quality vs low price and marketing sheen.

I very much like that Archimago thinks digital at 24/96 may be worth a bit more than 44.1/16, but beyond that is a fools errand.  That's were I stand.

Also that he thinks DSD128 is possibly worthwhile, but DSD256 and beyond becomes a waste of bits (as an end product, I might add).

A lot of what I do is solidly in the "rational" audio camp, such as my embrace of DSP for speaker and room correction, and even sonic euphanization.  Culty subjectivists won't step near these things, and I get criticized for "over-complexity" when I'm simply doing the rational things which must be done.

But why oh why do I bother with a $2500 DAC?  Does that show me to be strongly irrational?  Why do I bother with collecting particular spinning disc players (that used the best R2R chips)?

Let me first say this is a new thing for me.  I didn't even use any component level DACs until about 3 years ago.  I'd been an audiophile for 44 years before without having anything like that.  Most recently, I had simply used the DACs built in to the Behringer DCX 2496 crossover I was using, which used fine chips but had analog circuitry not at the highest level of performance, and not at all suited to the gain structure of my system, putting out 10V RMS at 0dB.

So buying my first component DAC was a means to respectable technical performance, since now 0dB would correspond to 2V, giving me lower noise and distortion always.

It's true that the particular DAC I chose would be more difficult to specifically justify (the Audio GD Dac 19 anniversary) as many similar products had equal measured performance.  I could have gotten something of similar objective performance for a bit less (though, not much).

But it strongly appealed to a certain iconoclastic side I must have.  I've taken on the R2R or 'true PCM' religion, somehow, perhaps for no good reason, just because it sounds good to my thinking.

I'm not saying everyone should do this, but I have chosen to.  BTW, the ideas also sound good to many audiophiles I know.  I think it's ok for me to nurture a few ideas out of the mainstream.

One think I won't do, however, is step too far beyond things that could be objectively seen as having audibly inferior performance.  I won't use high noise or distortion products, like SET amplifiers, or NOS DACs.

So when I figured the ultimate requirement for the gain structure in my system, and I was still able to get a DAC with AES input, balanced outputs, DC coupling with servo, and (my little iconoclasm) dual differential 1704's, I decided to go for it.

And I still like it very much, as I hope to for decades to come, because I indeed think it is good enough just as it is.

I  dislike being scolded by people who say money shouldn't be spent on such things, people like the commenter who calls himself Blog in the posting linked above.

As I've pointed out before, looking closely at what such people actually do, THEY have their little irrationalities also, and some of them even have far more expensive audio systems than I do, despite the scolding being dished out to others.

Love, Pain, and the Finicky Audio Gods (final update, Sunday June 18)

I fell in love from the first moment, opening the clean and nicely packed box.  The sound exceeded my best expectations: clean, organic, potent, now even FM (which I digitize for DSP) sounded high definition.  Digital files from my Sonos system now sounded as good as the best high definition discs played on my Denon DVD 9000.  Wow was this a good choice, I felt about the Audio GD Master 7 Singularity for the first 5 days.

As they say, it's not love unless it sometimes hurts.  The hurt began on the 6th day, when I was setting up sound for the living room TV for a party.   Sadly the Tact requires me to cycle through the 5 main digital inputs rather than just press the one I want (note: it is always best to provide a button for each input, rather than having to cycle through).  As I hit #2, which connected to my HDMI Audio stripper (which strips the audio bitstream either directly or converted to SPDIF, they both work the same for me because the Oppo BDP 95 which provides the video is set ONLY to output SPDIF) the most horrid snapping sound emerged.  I quickly pressed the Digital Input button again to select the next input.  Of course in addition to being scary it was embarrassing.

It was a chamber of horrors, each inactive or disconnected unit connected to the Tact delivered a very loud, potentially system destroying noise.  And when I started I had only 2 out of 5 inputs active, and the Oppo source (fed through about 65 feet of RG6 cable from the kitchen) was not fully started yet.  And sometimes the lights on the Tact aren't very clear, so it's not clear which input has actually been selected (sometimes it looks like several inputs have been selected, because the Oppos is low to the floor, and may need some internal repairs).  After a couple cycles through I made it back to the safe Sonos digital input which I had been playing for 5 days, but decided in future it might be best to switch to an unused input on the Master 7 itself.  I tried that and that worked perfectly, selecting any unused input on the Master 7 itself gave me perfect muting immediately.  But to get there, I had to crawl on my knees around the Acoustat where there is very little room.  But I soon mastered that task.

Now during this party, I played the movie sound from the Oppo, it might have been 48k or 44.1k.  There was no problem at all.

Also still at this time, I had the Master 7 hooked up via the coax input number 3, using the same AES to Coax converter as I had always needed to use before with the Audio GD Dac 19 Anniversary, which I had purchased a couple years before.  (Actually, I was first using the Dac 19 with Toslink, because the Behringer DEQ has Toslink and AES outputs only, and no coax, and I thought the optical isolation from all my noisy digital stuff might be nice.  But later I found the noisy digital stuff was not a problem at all for the digital coax, and the Toslink had such poor definition I could not read the jitter on my digital audio tester, whereas the coax had effectively none, I could connect earlier in the chain and get the exact same jitter number.)

I had not once ever had a muting issue with the Dac 19 anniversary.  If I selected an unused input in the Tact when I was using the Dac 19, it simply muted without a sound.  I had heard people warn me that with an Audio GD Dac I might have muting problems, but I had none at all with the Dac 19.
Other Dacs I have running might clink as the kHz light goes out, but no dacs I've had in the past also have done anything but always mute perfectly, no matter how complex the input chain.  Because of that muting, however, in the living room system, if I erroneously selected a bitstream output, I would simply get nothing (and not the terrible noise you get on some systems).  So I was completely shocked and unprepared for the strange noises I got at the party, but there wasn't any more technical delay than usual.

The movie party (3 videos) went OK, but not without blowing one speaker fuse, and hurridly finding one in my "laboratory."  I was grateful I could find one in just a few minutes, I hadn't needed a speaker fuse in a long time, I had fancy $49 (discounted to $29) audiophile speaker fuses too.  I was afraid I was going to have to use another of those, but then I found my package of 5A fuses from Radio Shack.  I looked online to confirm the Acoustat needed Slow Blo.  Problem #2 solved.  BTW, the speaker fuse had blown sometime as the Oppo went into stop as the movie was ending.

After the party I found that by removing rarely used inputs and simply capping them with shorting plugs I could stop the noises from unused inputs.  Problem 1 solved!

But then I also finally got around to actually installing the AES cable directly from the midrange Behringer DEQ to the Master 7, bypassing the need for an AES to coax converter.

Now this was different, I thought I'd try an unshorted input again, and now, no noises!  Somehow changing to the AES input of the Master 7 had eliminated all the lack-of-muting noises I was getting with the SPDIF cable.

So, shorting plugs no longer needed!  I could plug the unpowered units back in again too.  Problem 1 solved even better!

However, the Oppo still appeared to be a problem.  After I had played all the party videos on the Oppo, at the very end, relays clicked as the drive stopped, and the Krell amplifier shut down, for the first time since it had come back from Capacitor Service at Krell in February.  I dared not try the Oppo again.  I could probably live without a direct SPDIF connection between the Kitchen Oppo and the Living Room system, I thought then.

Something didn't seem right, however, with the sound after the party.  Now I didn't seem to be getting the new warm and organic sound of the DAC.  It sounded more edgy.

Then I remembered that just before the party, I changed the polarity menu option in the Tact to Normal.  I believed that had been part of an incomplete test during the first few days.  So maybe THAT was part of the magic.

I changed the polarity option to Inverted, and now I was getting the good sound again, it seemed.  (I wrote about this in an earlier post).  To confirm my suspicions, I downloaded a polarity test called Speaker Pop, and found that, indeed, the polarity of my system was correct only if I selected the Inverted option on the Tact.

I did not believe or want to believe that my new jewel, the Master 7, was inverting the polarity.  I figured this was just a long standing problem with the Acoustat speakers themselves.  I had previously gotten ambiguous information from the Tact's own impulse test program as to the polarity of the Acoustats themselves.

So a day later I removed the Denon DVD 9000 from my rack, and turned it into a DAC by turning the front panel INPUT knob to Coax.  I ran the exact same signal (from Sonos to Tact to Behringer DEQ) but also through a commercial AES to coax converter (no, this is not going to invert the signal polarity, which is a property of the data not the carrier) into the Denon, and its output showed the expected correct polarity, that is whatever I had selected by the Tact polarity menu.  The unbalanced outputs of the Master 7 showed the reverse.  But what about the unbalanced outputs???

I found I had no XLR adapter that would let me connect XLR to my scope.  (I ordered one from Markertek but didn't have it until later.)  But right then I figured out a cheat.  I could simply touch the positive probe to the positive output on the Krell.  The Krell is certainly not inverting, and simply reflects the polarity at its input.  I found that the polarity on the Master 7 XLR outputs matched that on its single ended outputs in being the reverse of the Denon and the reverse of whatever I had selected in the Tact.

I also discovered something else: I had sometime a few months ago put the subwoofers in different RELATIVE polarity.  And the subwoofers own PHASE control didn't seem to help.  I figured out a way to fix that.

By now I was really really liking the sound.  I was playing the Crystal Cables sampler from SACD which I had analog resampled to 16 bit digital.  But not playing very loudly, one track caused the Krell to shut down again.  I was simply playing this from the Sonos Connect digital output routed through 9 feet of RG6 cable to my Tact.

Hoping that I could keep my new DAC and work around this problem, I made made new Teflon twisted pair speaker cable to replace the super low inductance but relatively high capacitance Canare 4S11 speaker cables.  The Krell still shut down.  So I then also replaced the Baldur XLR cables that connected the Master 7 to the Krell with Belden 1800F cables.  THAT fixed the shutdown and I was even able to boost the playback level 12dB and play the track over and over.  I thought all the problems were now solved.

I had been thinking, however, that the Oppo sounded better when my system had inverted polarity.  I didn't know whether it was safe to test the Oppo again.  But since I had switched to AES input, muting hadn't seemed like a problem, so I made a recording of the SpeakerPop sound, and put it in the Oppo.  After the track had already started playing, I changed the source selection on the Tact so the Oppo was now selected.  I started at a very low level and really had to crank it up.

No, sadly, my second big polarity guess turned out to be wrong.  The Oppo had correct polarity.  I tried the disc in all my other players, and they all had correct polarity also.  I had been thinking the Integra Research RDV-1 had incorrect polarity just like the Oppo.  Nope.  That makes one good guess out of three so it really looks more like I can't identify polarity than otherwise.  The Sony DVP-9000ES would not play the CDR I had burned on my Lacie, and I'm not sure if it generally has the ability to play burned CD's.

In testing the Oppo, I also wanted to see if the polarity selection in the audio menu made any difference.  It was somewhere doing this, messing with the audio menu in the Oppo, that the Krell shut down again, for the first time in several days.  I had been hoping I had fixed the problem by running an AES cable to the Oppo, but apparently not.

During one of the polarity tests on the living room disc players, the Krell also shut down.  And later, when I was playing the Pioneer F-26 as digitized by the Lavry (for better fidelity than when I usually digitize it with Sonos) the Krell shut down.

These later shut downs were beginning to surprise me.  They didn't seem to be related to any faulty muting on the DAC, as previous ones had been.  But now I thought of something else.

I now figured that the reason for the shut downs was specifically that the Krell was trying to upbias and there is insufficient AC power.  This is one of two possibilities when the Krell automatically shuts down but shows only the "power" light illuminated.  The other possibility is that the speaker or cable has shorted.  I had pretty much ruled out the second possibility long before, the speakers were still working after all (thankfully!).

Now it didn't occur to me at first there was anything I could do about this.  I have a single 20A dedicated line for my entire system, but not for just the Krell (as Krell recommends).  But then I figured out how to do it.  I could power the subwoofers on a different circuit than the rest of my audio system.  Inspired, I set about doing it, and in just a few minutes I had moved cords around so that both subs were plugged into the regular living room circuit and not the 20A audio circuit.  This would leave only the low power components on the same circuit as the Krell.  I didn't want to move THEM to the other circuit because that circuit is incredibly noisy as it also powers the front yard photocell operated lights, and they put tons of noise on the line.  But the subs have digital power supplies which aren't affected by that.

This made a huge difference right away.  Now I could play jazz and indie rock on FM through the L-1000T through the Lavry instead of Sonos, and run it that way all night long.  And I did that just to be sure I could do it.  Problems solved!  Or so I thought.

It still bothered me a lot that I couldn't play the Oppo.  I was thinking, maybe if I activated the PLL jumper inside the Master 7, it would keep the Master 7 from responding in a very bad way to sampling frequency changes and muting from the Oppo.

Just to be sure this was a good thing to do, I emailed Audio GD, and I also emailed a friend.  Neither response really answered my question, however Kingwa said that the default is NO PLL (which I was somewhat uncertain about) and the digital receiver is ALWAYS ASYNCHRONOUS.  Those two bits of information were very useful.  But Kingwa also said I should not be running SPDIF longer than 10 meters.   I replied that I had been and still do run lines 50 feet (actually, 70 feet) with other DACs with no problems at all, and other people I know run SPDIF lines as long as 120 feet.

My friend had only used I2S and he always runs the DAC through his preamp first.

I'll express my strongly negative feelings about I2S in a future post.  Meanwhile, I simply went ahead and changed the jumpers for PLL and tried the Oppo again.

This was different or so it seemed.  With the polarity correct, and PLL enabled, the sound was superb, better than I remembered it going back to the days when I first had the Oppo and Denon 5900 plugged straight into the Lavry into the living room.  The glare and haze had disappeared, leaving the sound a bit darker, but far more real sounding, and with punchy bass.  I was playing the Santana Supernatural DVD-Audio, sending 24/96 across my 70 feet of cables into the living room, and it was great.

Problem solved!  I cranked it up and started grooving.  A half hour went by.  Relays inside the Oppo were clicking on every track, making other DACs mute, but there was not a sound from or through the Master 7.  It was now muting and changing sample rates and everything over a 70 foot cable at 24/96 without a hint of problem.

But then, 40 minutes later, the Krell shut down again.

OK, so maybe I can't use the 70 foot spdif cable anymore.  I started thinking of other options for my monthly party, such as running the Oppo audio through my team of Sonos zone players in every room.  I wouldn't be able to get high resolution, but it would be OK for movie sound.

Feeling somewhat down, I decided simply to listen to a cassette I had recorded in the bedroom from an FM local broadcast the previous week.  Piped into Sonos in the bedroom and back out in the living room in glorious 44.1/16 digital piped through my system.  Pure sonos connections had NEVER been a problem with the Master 7, in 2.5 weeks to this point, so  I could just relax.

But this time, after about one hour, the Krell shut down.

Now it was beginning to dawn on me.  Something had changed since the Party, and it was not just the fact that I had switched to AES cable after the party.  (Actually, in another set of breathless tests I performed earlier, I had switched back to Coax hoping that, despite the muting issue, perhaps it handled other things better.  It did not seem to, and I got shutdowns using Coax just like using AES, and the sound wasn't any better either.  But correcting the polarity made a big difference to the sound.)  It was because, at the party, the loud noises had caused some permanent change to the speaker.  Most likely the HF transformer inside the Acoustat interface had an intermittent short.  That's how these things start.  You get a tiny short and then a fuse blows.  Then, the fuse blows more and more often as the short gets bigger.  Except, in my case, the Krell FPB is so clever it can figure out when the speaker is shorting, and shut down before the fuse has a chance to blow.

THAT was what had been happening all along, ever since the night of the party, I now concluded.  I have at least one bad transformer in my Acoustats that needs fixing.

So to test that, I first ran the left speaker playing FM through Sonos, just as I had done the night before with a cassette tape of an FM broadcast.  I played louder and louder on the left channel, but after 90 minutes the Krell had not shut down.  Back to the right channel, I played FM through Sonos again through the right channel, and in 20 minutes the Krell had shut down.

So I have proven that there is a problem with the right speaker now.  THAT was the only speaker to blow it's fuse at the party, and the issue probably started right then.

So that could explain everything.  Perhaps now using the AES input I will have no more issues with the Master 7.  But it still seemed that PLL helped me get the Oppo to play much longer and sound better, so I'm going to keep PLL enabled.  Likewise for the other changes I made, replacing the Baldur XLR interconnects with Belden 1800F, and putting the subs on the other circuit.  Those changes were all for the best anyways.

I remember now that I had long ago had the same trouble playing the F-26 and L-1000T tuners as digitized by the Lavry AD10 instead of Sonos.   The few times before I had tried to do that, the Krell had shut down.  I figured maybe the tuners had too much low frequency noise resulting from the FM tuners, I just couldn't do that.  But now I think the problem was that I was putting both the Krell and the subwoofers on the same circuit, and the Krell was trying to get more current on some peak and competing with the subs for that power.

So while it's possible that all the shutdowns I saw resulted from a growing transformer short in one speaker, it also possible, and it was part of the reason this took so long to figure out, there there were actually more than one cause, and probably two, for the shutdowns.  One being the transformer short, and the other being inadequate AC power.  At the end of the day, though removing the Baldur cable seemed to help playing Taguita Militar, that might have really represented the "inadequate power" scenario because (1) it happened consistently over and over at the same point in the track, and (2) in addition to extreme highs, that recording also has deep powerful bass, in alternation, creating the situation where the Krell and the subwoofers were competing for AC power.

I haven't actually proven the left speaker does not have a problem, it just didn't show up in 90 minutes of pretty loud (not extreme) testing.  I will continue testing the left speaker and go back to using the Audio GD DAC using the safer AES input with PLL enabled (belt and suspenders) but meanwhile record the right channel output using a Masterlink at 24/96 so I can see problems up to 40kHz.

The interface which blew was the original interface that came with the speakers.  It's brother was already replaced in 2010 with a spare new old stock unit in museum quality which happened to appear on eBay exactly when I needed it.  The unit which has not yet proven to be defective on the left channel was that new old stock unit which I have now used for 7 years.  It has a much later serial number.  So there is hope that it survived the cruelties of the past few weeks.

Ironically just a couple months ago, the leading Acoustat interface refurbisher posted to The Audio Circuit that people should not use 5 amp slow blow fuses anymore.  That recommendation was even a bit risky in 1980, but in 2017 many power amplifiers have enough voltage and current to blow a transformer before they blow the 5 amp fuse.  Actually, Dr. Strickland originally specified 3 amp slow blow, as that had a decent chance of protecting the speaker in most cases.  But there were so many complaints of unnecessary fuse blowing, he gave in to pressure and changed the specification to 5 amp, which would stop most, but far from all catastrophic events.  That just barely passed muster with the 200 watt amplifiers of 1980, but not the 500W amplifiers of today (my FPB 300 actually has 600 specified watts at 4 ohms, and more like 900W at measured clipping, and over 1500W into 2 ohms, mind you, it's so smart it stops the moment it detects a short, making no foul noise in the amplifier or speaker, but it doesn't necessarily stop full power signals into a not yet short).

I went to what was described (and seemed) like the last Radio Shack store in San Antonio to get a collection of fuses, I'm not actually sure which is best, 3a slow blow and 5 amp fast blow, however they didn't have 5 so I got 3 and 4 amp in fast blow.  But I'm using 3a for now.

I played the Left speaker by itself for hours Friday night, playing FM mostly through analog Sonos, not wanting to extend the experiment too far.  But by Saturday I was testing the Dac 19 through system connections to the kitchen Oppo, including 70 feet of Belden RG6 for coax SPDIF.  It was working fine.  I had hooked up one of my DEQ's as a monitor meter (having level and spectrum peak hold, etc etc) and saw nothing wrong,  Santana Supernatural DVD-Audio through the Oppo.

After several days testing to prove the fault with one speaker, and so far only one speaker, I hooked up the Master 7 again.  Now I didn't power the amp yet, I just used my DEQ monitor to check out all the inputs on my Tact.  Everythng was working flawlessly, and no spurious crap, EXCEPT using the Kanex HDMI de-embedder to strip SPDIF from a standard HDMI signal (in this case, from the Oppo BDP-95 set to only output PCM up to 96kHz).  Using the de-embedder in both modes gave a signal with periodic low frequency bursts up to -10dB apparently going down to DC.  Even though off, the Krell seemed to be making noises like it was detecting DC at intervals.  Somehow the likely high jitter of the HDMI signal (which itself comes from a CAT6a house network) is overloading the jitter removing capacity of the Master 7, and the result is bursts of low frequency which would not be good for directly feeding DAC into a DC coupled amplifier.

Well anyway the HDMI de-embedder has always been a troublesome piece.  Quite often I have found it not to work on either position.  But I have never noticed DC pulses.  Still, these -10dB pulses were nothing like the high pitch max level shreak I heard the night I first selected the de-embedder to play the Oppo output at my party.  So from destructive shriek to quasi normal operation with bursts of DC is some progress anyway.  But I'm simply not using the de-embedder now for sure (except testing purposes as it clearly makes an interesting test).

But testing further I also found something to work which had never worked before.  Playing Reference Recordings 88 kHz data discs on the Oppo!  I had run into a brick wall with these discs, they can be downloaded to my kitchen mac, but the only network available from there now (I've de-certified the part optical connection to living room) is Sonos, which doesn't do high rez.  So I had to laboriously convert each file to 44.1kHz, which I did for one disc, but I have others that have been sitting there, and I've never enjoyed the original high rez.

Bnt now I can, now that these 88kHz discs play perfectly on the kitchen Oppo, over 70 feet of RG6 wire (actually in several segments, one going through the attic), into the Tact where it gets level adjusted, output through the Tact AES output to the DEQ, and from there by AES to the Master 7, now it all worked perfectly and sounded wonderful, much as the Santana Supernatural DVD-Audio sounded better on the Kitchen Oppo digital output than ever before and ever through any other of several means I have to play it (but, the Integra Research RDV is extremely troublesome on this disk, it goes through various broken frames of garbage and takes persistance to play the DVD-Audio, so much so, I've given up.  I'm getting another Denon DVD-9000 which is said to have full DVD capability, hopefully full DVD-Audio capability, and I now think it sounds much better thank than the RDV at the analog output anyway.  Speaking of which, before I use any new analong input now, I think I'll certify it as OK by using the Behringer as a monitor as I just did.  (I used the supertweeter DEQ on top as monitor, turning off everything related to the super tweeters and disconnecting them from the DEQ, then changing it to simply monitor the analog input connected to the XLR output of the new unused channel of the Master 7, I've got glorious mono now that one of my Aoustats needs repair and my spare interface in storage was never repaired either.  But I believe I will have my other Acoustat back online before long, better than ever.)

Don't let 'em tell you it can't be done.  For me, it seems AES input does the trick, even if there's an intermediate length of SPDIF back to the source behind that AES.  And I'm not really sure if PLL is helping, but it seems to be working for nearly everything, and I think it prevents stuff from going badly awry,  and sounded better to me.  Now I'm back to the rock solid digital connections I had before, mostly, with one more important one and one less unimportant one.

So now I just need to get one of my Acoustats fixed.  But that's not a problem, as they say, it's an opportunity.

A lady friend of mine knew the explanation right off.  You get expensive new toy, the old toys get jealous, they want money spent on them.

More AC Power, Please

I was just feeling warm with the incredibly ware, rich, natural, grainless, presentation with all the latest changes, including polarity fixes.  Then, last Thursday, the Krell FPB 300 shut down with only 1 light.  The manual says that means shorted speakers or cables, or insufficient AC power.

I run my entire stereo on a 20A dedicated circuit with 10G wire, but not specifically the Krell.  I see now that Krell recommends running the 400, which mine in power already effectively is (when they fix everything, they set it to the 400W program, the 400 and 300 they've always been the same anyway, except for Cast input and other input board changes.

The ultimate peak power doesn't change, but the ultimate bias levels permit 400W class A at least for a very short while (it can't do that for long, the temp will rise above the cutoff and go back to Plateau 2).  This is a lot of power.

Anyway, I also was somewhat worried about the new Master 7 Singularity DAC.  But after much trouble, I finally figured out how to test the XLR offset DC voltage, and there was none (though both + and - were 0.5v above the ground pin, which should be OK).   There might be other issues, but before finding my wonderful new dac unusable I should investigate less expensive possibilities.

I also (finally!) wound a set of new PTFE speaker cables from solid core teflon coated silver plated copper milspec wire I bought on ebay.  I used that the replace the Canare 4S11 speaker cables I had custom made by BlueJeans.  The Canare is low inductance and high capacitance.  This maximizes any peaks, and can set unstable amplifiers oscillating.  The Krell should not be unstable at all, but the draw of the cable is almost like a short at super ultrasonic frequencies, perhaps that is setting the Krell to anticipate the maximum bias level, even though I was listening at -15dB.

Whatever, I decided I also didn't like the screw on terminations, just the slightest looseness could cause arcing.  So I decided I was moving on, but not to the old junk but to the new project I hadn't gotten around to for a year.

So I put on the wires, ran the same song at what I thought was the same level (I had briefly turned it down, I couln't remember exactly what it was originally).  And, halfway through the song, the Krell shut down again.

OK, so now I tried replacing the Nordost Baldur 0.6 meter input balanced wires.  Now, no shutdowns, no shutdowns at louder levels, I raised levels all the way up to an earsplitting -4dB on this song (which has very impulsive flutes and metallic sounds, drums, etc, an audiophile recording which originated on SACD which I digitized to 44.1 to play on my Sonos system.  It's "Taquita Militar" from a Crystal Cables sampler.  It's a very tough high frequency test on an extended bandwidth system.

Anyway, so dumping the $500 audiophile cables (which by conventional standards should be OK, just slightly higher than normal capacitance, and extremely low inductance, and supposedly vanishingly now propagation delay) have fixed the problem.

(BTW, Baldur was discontinued by Nordost several years ago.  Perhaps a lot of people had trouble with such a fast interconnect.)

But I still think, if I had more AC power, perhaps the Krell wouldn't have shut down but actually rose to the challenge of the highest bias level.  Even free of the Baldur cable (though, I think I'll be trying some ohter things.  I've ordered Belden 1800F cut to length by Blue Jeans, and I'm looking into some other PTFE cables cut to spec).

Mind you, I don't really know if the problem is inadequate AC power.  But a friend of mine thinks I have so much equipment it shouldn't all be on once circuit, even 20A, and that is correct.  I should have at least 2 and maybe 4 circuits to be fully audio grade (each sub on it's own dedicated circuit, as well as the krell and the conditioner).  And with balanced connections to the Krell it won't hurt it to be on a different circuit, and the Subs already have balanced connections to their DAC (an Emotiva).