Living Room System

Saturday, May 21, 2016

MQA heard and liked

I heard MQA and it sounded good to me.  The audio society sponsored a demonstration with alternating cuts of standard PCM (the first was actually an 88.2kHz 24 bit) vs MQA.  The first person in the audience to describe it used the exact word I was thinking: "natural" for the sound of MQA.  And the presenter echoed that.

I hear clearer attacks and decays, and ambient echoes following notes.  Though the presenter said he heard more highs, I wouldn't necessarily agree.  It was not obviously different in any sort of macroscopic way such as big dynamics or frequency response.  It was in the tiniest details related to focus and I think ultimately timing that things improved.

This confirmed my suspicion that MQA is a worthwhile format, and I will fit it in when convenient.

Of course neither you nor I should believe the results of a sighted test with 3 comparisons.  Done this way the testing was both invalid and insufficient, or it would have been insufficient if there had been validity, but since it was a fully sighted test there was no validity period, the effects could be entirely based on selective memory and preconceptions.

I heard...exactly what I hoped to hear, actually.  I've long understood MQA as cleaning up the timing details.  The culmination of the "apodizing" filter ideas pioneered by Bob Stuart.  Intially he was able to clean up the pre-ringing, but making the post-ringing worse (or maybe I have that backward?).  Such special filters have become commonly available in last decade or so.  The idea of MQA is that if original encoding is accounted for, the pre and post errors can be eliminated.  One can re-create the closest possible replica in time to the original, closer than single ended high resolution systems with apodizing filters, because you can simultaneously account for the phase errors in both encoding and decoding.

That's the idea, anyway.  It sounds good.  It sounds even better than the ideas which went into HDCD, which I've also been a fan of (not without some mixed feelings, but the best HDCD's are better than SACD in every way IMO, whereas CD is only better than SACD in some ways...and yes that's what I meant to say...).  In fact, this is the best sounding idea since...I don't know,  I can't say digital recording itself since that has had issues...perhaps it's the best idea since magnetic recording, though Doug Sax showed that even bypassing that could be helpful.

It's of course just fantasy that my barely tutored ideas about what little I know about digital technologies has any bearing on how they would actually sound.

But anyway, Archimago has done some excellent research on MQA, he's not at all a fan of it, but in one of his tests, pecularliary involving the DSP impulse correction part of MQA which could, in theory, be applied without MQA, he quickly passed his first Double Blind Test (DBT...he knows how to do DBT and does it correctly and sufficiently).

Well Meridian themselves published (not as a journal article, but an "update") in JAES a brief description of their own positive DBT results.  This article was favorably discussed at websites like What's Best Forum, but trashed at sites like Hydrogen audio.  The Hydrogen Audio critique was this: the "control" side of the test used old reconstruction filter technology, whereas the test included apodizing filter.  Well then the test could prove nothing about MQA itself, only the use of apodizing filters, which was become almost standard among the technically literate nowadays.

Wait, that's similar to Archimago's tests.  Not at all actually, but in both cases they were testing a subset of the techniques used in MQA vs old-fashioned-digital.

Anyway it does appear, contrary to the strict interpretation of Meyer-Moran (an AES published test in the 2000's) which some apply (essentially, all digital conversion is transparent, up to 10 layers thick, regardless of digital technology) that some digital is different than others.  This is blasphemy in some objectivist quarters...but can we say that the numbers are now coming showing some variations in digital above 44.1/16/1983 can be audibly different?  And these are involving sets of technologies used in MQA.

OK, this sounds like more than just my fantasy now.

But I have 3 questions:

1) How technically successful is MQA.  How much more accuracy does it provide to impulse response?  How much resolution is lost to make room for the packed information?

2) Is MQA the best possible impulse cleaning technology, and if not, will it prematurely lock us into an inferior standard?

3) Can MQA be used in conjunction with room equalization DSP?

#3 is crucial for me.  Actually I don't know how deeply I can get into MQA without abandoning my use of DSP for crossovers, time alignment, level adjustment, and room EQ.  And I simply won't do that.

For sure I cannot simply tack on an MQA DAC at the end of my system.  First the encoded signal cannot be DSP'd like ordinary PCM.  And once it has been DSP'd, the MQA information would be corrupted.

Quite possibly I can do what I currently do for SACD and HDCD.  I take the analog output of a device which plays SACD or HDCD and re-encode it to PCM digital at 24/96.  (My DSP is limited to 24/96.)  Little resolution is lost, of course the "magic" of DSD is lost, but as I've previously argued, the magic of DSD is hype--there's really nothing there even at best than in high resolution PCM.  A bit of the magic of HDCD might be lost too (the final reconstruction filtering selection...I get that in the intermediate conversion to analog but not the final conversion to analog).

It seems like this can in principle preserve the same part of MQA that would be preserved by people not using MQA DAC's for MQA but instead "MQA converters."  In fact, I could use an "MQA converter" as well, something that takes MQA and converts it to digital at 24/88 (or 24/96, but in this case I think 24/88 would clearly be better).

Working in this way, MQA cannot possibly deliver more than the format it is converted to.  I remember that MQA used end-to-end would in principle provide better impulse response that 24/192 or even much higher, but if I'm converting MQA to 24/96 I'm getting the impulse response which my converter can provide for it, nothing more.

This still makes MQA an excellent (or at least as well as it actually works) way to stream high resolution quality with much lower bandwidth.  Also a way to store that higher resolution quality on my harddrive taking fewer bits.  It's just not giving me something better than high resolution quality I already have for high resolution content.

If there is something more than this that MQA can do, if you actually have MQA end-to-end, but you don't get without having MQA end-to-end, then MQA could be limited by being a closed proprietary system.  It would be better if we had the pieces of MQA available to us and could plug them in appropriate places ourselves.  So I could route the "magic" around my DSP and plug it back into the final DAC (which would have to be a special MQA DAC permitting PCM and bypass "magic" inputs).



Sunday, May 15, 2016

Choosing new UPS for the Living Room System

The Belkin AVR1500 UPS in the living room started beeping.  Fairly quickly I shut it off (along with the living room system, all except the Krell and Subs go through the Belkin).  When the "low battery" beeping starts, it means the batteries aren't holding any charge, and the charging circuit is running continuously.  This might (though it is not supposed to) produce a little hydrogen and I considered such UPS hydrogen generation a possible cause of my illness in 2014 after I returned from a week vacation.  When I got back home from vacation, the kitchen UPS fan was running continuously.  I had the beeping turned off (since I didn't want to be bothered with brownout and outage beeps).  It took 6 days before I figured out why the fan was running: it was because that the battery charger and fan run continously when the battery won't charge anymore, and during that time I felt queasy enough to report it to a doctor.  I got an ultrasound which said my gall bladder had "sludge" and the specialist and my PP recommended removal.  But I replaced the UPS batteries and I haven't had any similar abdominal problem since, and no UPS problem either until those batteries from 2014 failed in 16 months, and I decertified the UPS itself and found that the backup UPS in storage was inoperative also.  So with the latest failure, all 3 Belkin UPS units I bought in 2010 have failed or become suspect and I obviously recommend everyone Stay Away from these long discontinued Belkin UPS units, as perhaps I should have done, though I'd gotten a total of 11 years usage from them now, for about the same cost as a single decent equivalent, and they seem to be fine as AC power conditioners as such, which I wrote articles on in 2010, if I could only disconnect their faulty UPS part, which I may investigate doing.)

This is hardly proof that the UPS caused my illness (tainted food had been my first guess and still is, travel stress my second guess) but I do strongly believe that it is best to avoid hydrogen, therefore avoid overcharging UPS batteries, just in case.  Since that incident I've always had the "beepers" turned on my UPS's so I can't ignore low battery warnings.  And so when the living room UPS started beeping last week, I fairly quickly shut it off--but now I know it must actually be unplugged also.

On the following Saturday I removed the battery box (I also finally marked the new right Acoustat position with masking tape because I had to move the speaker out of the way).   It had been hours since I actually unplugged the Belkin, and days since I shut it off, but the battery box was still warm, I estimated 110 degrees.  Both sets of batteries seemed to stick together a little, that seemed maybe only because of the adhesive holding the labels on, but the batteries themselves had a slightly warped look as if the plastic had melted a bit.

A little thinking about this led me to the conclusion, I won't be getting replacement batteries.  I won't be using this Belkin UPS anymore.  It could be unsafe, especially if you somehow missed the low battery warnings.  Though I might have prevented the warped batteries had I unplugged the unit at the first beeping, I don't want to take chances.  I think it could be dangerous, and I've known from day 1 that Belkin had quit making UPS's which was why I bought 3 at closeout prices for less than the price of one, and this suggested their UPS's might have serious issues which led to Belkin getting out of the UPS market.

But what to get now?  I do like the reliability and concept of the BrickWall UPS.  It uses no sacrificial and noisy MOV's, it uses a system of chokes and non-sacrificial capacitors to filter the power in such a way that surges can't happen.  It has a very long warranty and I believe the company says in over twenty years they've only replaced a few.  It's safe, works, and lasts a long time.

I figured this kind of system would also clean the noise on the AC power.  Just now I tested that, with the TV turned off in the bedroom, the AC power still sounded fairly noisy on my Audioquest Noise Sniffer.  It actually sounded quieter on the Cyberpower Metal Box Sinewave Interactive UPS in the kitchen.  I was very disappointed.

I though about my options.  I could use the PS Audio 1500W Power Plant Supreme that I already have.  I took that off the shelf and it's AC power was quietest of all.  A model of perfection.  But I don't entirely trust these units.  A few years ago it ended up blowing the fuse on two Parasound amplifiers after the PPS got into a tug of war with the APC UPS on the same circuit.  So under difficult circumstances the PPS can output dangerous AC that can at least blow fuses, if not worse.  This is sort of like the Belkin in that PS Audio made these "Premier" units in China, but for some reason after doing that decided to move their production back to the USA.  So there's reason to be suspicious of these units.

I think I'd trust a brand new PurePower UPS, but boy are they expensive!  The lowest power unit still in their model lineup is the 1500 and they run $3250.  If I'm going to plug everything in my system into it, as everyone recommends, I'd need the 3000 and they run $5850.

Cyberpower makes a online UPS like the PurePower but it has a very noisy continuous fan.  Likewise for the online UPS's made by APC and Tripp Lite.

Given all the expensive and unobtainium equipment in the Living Room, good protection is needed.  I think the UPS protection is somewhat better than the Brickwall protection because of the continuity of power (doesn't go on and off repeatedly) and brownout protection.  And it's annoying when power outages happen and all the Sonos units don't come back up correctly, for example.  The Behringer units don't always come back up correctly.  And often power outages have multiple jerks on restart.  I think repeated starting and stopping can lead to failures even on purely electronic units.

So I had decided to get another Metal Box Sinewave Cyberpower UPS (PR1500LCD).  It seems to have quieter power (Cyberpower does advertise EMI and RFI filters, FWIW, but I would have never expected it to be quieter than the BrickWall, though my current tests may not be definitive on that) and I think better protection overall.  It may not last as long though (and there will be replacement batteries to get every few years, hopefully more like 5 than the 3-2 I was getting with the Belkin).  In the kichen system, I've never noticed the fan to come on, and it's powering a lot more stuff.

As I was writing some of this article, and thinking how good the system sounded, powered by the new Cyberpower.

After a few more days of looking around online, I decided to get a UPS actually designed for audio equipment, the Panamax MB1500.  This looks to be identical to the Furman F1500 (Furman and Panamax are the same company now) and typically sells for the same price as the Furman ($1299) but I managed to find one, brand new, for $811, from a well regarded online store Newegg.  Every other online vender wanted the list price, $1299, or even higher (!!!).  Now I see that Newegg is NOT an authorized dealer for Panamax (Panamax has very few Authorized online dealers, hardly any of them are big names), and Panamax won't honor either the product or attached-equipment warranties because of that.  Oh, well, if I had known that I might or might not have made the same decision.  Except that I simply would not buy this for $1299, which seems to me like a rip-off when I can get the Cyberpower PR1500LCD for only $389 from an authorized Cyberpower dealer, and it's very similar as far as being a line-interactive UPS with sine wave generated power and some EMI/RFI filtering.  I am just hoping that being a UPS designed for audio purposes like the Panamax or Furman the EMI/RFI filtering will be a little better.  But without hard facts, or knowing if it would even make a difference, I wouldn't spend $890 more, but I was willing to spend $422 more.  For $890 more, I'd like the kind of filtering that used to come with the Monster HTPS 7000.

I had been a little inspired by this review of the Furman F1500.  Unlike my old Belkin, this looks like a very serious UPS (I suspect Panamax/Furman gets an OEM from the likes of Cyberpower as the features seem very much like Cyberpower features).  And as a "conditioner" they show the effect of the EMI/RFI filtering, at least at audio frequencies.  The noise (or is it distortion?) is highly and visibly reduced on the spectrum graph above 3kHz, though strangely it's slightly increased below 3Khz by a very small amount.  (This is typical, from what I've read elsewhere, and some even say the tiny increase at lower frequencies makes the improvement at higher frequencies moot…I'm going to guess and hope this is not so and that that the mixed improvement is still an improvement overall.)

Why not just do the "try it and see" thing?  I don't trust subjective tests other than true DBT with statistical analysis, that's much more hard work than audiophiles are used to, and I can guess in advance from my previous experience testing things that objectophiles say don't make a difference that I would be almost certain to get a negative result in such tests.  Any single sighted test, or even a year of sighted usage, actually proves nothing and can only create superstitions (which many audiophiles are know are cursed by).  But still even if something can't be proved to be better isn't proof that it can't be.  So I'm doing this on a "it might be better" basis based on objective considerations (though I realize the particular objective considerations I'm using, as describe here, are themselves subjective) and basically "it might be better" is worth something but not a lot.

Here is a review of the Panamax MB 1500.  Note it looks absolutely identical to the Furman F1500.  But in this review they also show pictures of the inside.  Yes this does look like a much more serious UPS than the Belkin.  And it does have fairly serious looking AC filter parts as you can see behind the outlets in the back.*  These look just a little less impressive than the very impressive loooking AC filter parts in the Belkin I have now (which would have been great if it also had a good UPS instead of the flaky and potentially dangerous UPS it does have).  But the filters look a lot more than just hype, and likely more impressive than the filters in a typical computer UPS like the Cyberpower PR1500LCD.  Now I don't know what difference it would make, but the typical prepackaged "AC line filter" parts you can buy off the shelf from Mouser or Newark such as the Corcom RFI filters don't have much effect below 100kHz, and interestingly the Cyberpower specs list 150kHz as the lowest frequency.  I'd guess the filter parts in the Cyberpower are similar to the Corcom parts, single part filters, though probably not as expensive.

*I could imagine a standalone power conditioner with parts like the Panamax/Furman in a cheap box for $200, or a nice box and outlets like the ones it has for $500, or at an audiophile price of $2000 in an even nicer box.

So it is very much looking like when doing AC line filtering, you can do a lot for cheap at the highest frequencies, it's the lower frequencies where it starts costing real money.  For comparison, look at the incredibly complex filtering stuff you used to be able to get with the likes of the Monster Power HTPS 7000.  (If you can't see the pictures in that blog, look here.)

The entire Monster 7000 HTPS chassis is filled with cool looking parts, big yellow box film capacitors that look like Wimas (probably aren't actual Wimas) and heavy wound chokes, and two large isolation transformers (which Richard Marsh himself says were very expensive and high performance isolation transformers for the low current outlets).  This is not a UPS at all, just a "power conditioner" which really means that it filters noise and distortion from the AC 60Hz, and yet it fills a very large box with parts.  Anyway, I don't know the actual performance of this unit, just that a fairly serious designer (Marsh) did the best that he could with the funds available, but it probably shows the minimum amount of "stuff" you would actually need to filter out the noise and distortion lower than the 3kHz cut off of the Furman and Panamax UPS's.   You can see that just having a few parts behind the outlets just isn't going to do it, you need to fill a whole box with filter parts if you want to filter the lower frequencies well.

I believe Monster called the HTPS 7000 "stage 5" or maybe "stage 7" filtering, and you can see that each set of outlets does have about 5 or 7 major "parts" that the AC power must flow through.  By this standard of rough measurement, the Panamax and Furman UPS's are something like "Stage 3", just like my old Monster 3000 power strips (which all died after less than 5 years btw, but it probably was not the filter parts which died but instead the stupid MOV's and the voltage/current display).

Now I'd spend the extra $$$$$ for a good filter if I really knew it was important, but I don't.  I tend toward believing the objectophile view that well designed equipment, like pretty much everything I already have, was already designed to handle AC power as it actually exists and doesn't need any more "conditioning."  Further, my entire living room system runs off a 20A dedicated isolated ground circuit.  It is often said to be a better thing to get a dedicated circuit than to buy a fancy AC conditioner, and that's my starting point also.  I also have the power company transformer for the houses adjacent to me in my own back yard with about 50 foot buried lines capable of over 600 amps.  I know of no specific problems with my AC power.  It consistently measures 120V within a volt or two, and about 2% unfiltered THD on the PS Audio Power Plant Premier.  That's about as good as utility power gets.  Long long ago I had ground loops in my bedroom system, which was my first "audiophile" system in this house.  But those were not related to the power as such but to how the old cable TV system was grounded.  I worked around those by removing all ground loops from my system and isolating everything related to TV from things related to audio.  That was a great learning experience which I think all audiophiles should have.  Don't fix hum with band aid but by actually removing ground loops and equipment with faulty power supplies (which were the second problem I discovered, many 20 YO units have faulty power supplies which produce audible hum even when said equipment is disconnected from everything else).

Much later my electrician discovered that my house grounding was broken, and I had him fix that right then and there with a new heavy duty copper grounding rod near the AC panel.  The broken house ground probably had something to do with the cable TV ground loops, actually, the cable was grounded to the plumbing (and later the panel) which should have been grounded to the panel which should have had a working ground rod, instead I could measure 120V between the ungrounded panel and the water-pipe grounded CATV wire.

IF I really knew this conditioning thing was hugely important, I'd spring for the hugely expensive PurePower 2000 or something like that.  Or at least I might.  Or not, I mean that's actually more than I spent on my minty used Krell power amplifier including the first factory repair.  There's also a particular problem with large regenerators.  You apparently MUST plug EVERYTHING in your system into it.  That's because while the regenerator gives you pristine clean power at its output, it compounds the amount of grunge at the wall outlet itself.  So with one of those, for the best results, you must plug in the power amplifier to it also.  I'm disinclined to do that with power amplifiers generally, and it would also mean that the PurePower 2000 wouldn't be sufficient, and I'd need to get the 3000 or still larger!

Speaking of regenerators based on On-Line UPS's, you can get those far cheaper than the ones made by PurePower.  You can get a nice 1500VA online UPS from Cyberpower for about $560, or similarly from Tripp Lite or APC.  The problem with online UPS's made for computer server rooms is that they consistently have noisy fans.  I've read numerous blogs where people change out the typically small fans inside such units for 5" or larger fans that more quietly run at low speeds.  I have some experience with that sort of modification, I did something like that on my Amiga 2000 computer about 23 years ago.  But it's tricky and very time consuming (I spent months modifying the Amiga 2000) and I just don't have the time for that style of equipment redesign anymore, and especially for something I'm suspicious is not very important anyway.

If the line AC noise and distortion at lower frequencies is really all that important, I might opt for the basically passive approach of filters (like that Monster 7000, it's been discontinued but you can get used ones cheap on eBay) or use an isolation transformer.   The isolation transformer thing doesn't look bad, if you shut your system off when not in use and don't really need the UPS part.  Typical isolation transformers tie the output neutral to ground, I've read that should be cut for use in an audio system, and the neutral left "floating".

WRT the Monster, however, I'm suspicious of their long term reliability.  I bought three very fancy Monster 3000 power strips (about 10 pounds of solid metal) that looked indestructable around 2004.  By 2010 they had all failed after 4-5 years usage.  Perhaps the 7000 being a more expensive and serious product would do be better.  But the fact that Monster has discontinued the 7000 and many of the higher end models doesn't look good.  There are many online stories about failed Monster Power products too, and it seems like many of the upper tier models have been discontinued and not replaced with anything similar.  In the story I linked above, one 7000 owner asked Monster if his failed unit could be repaired and Monster said no, it had sacrificed itself to save his equipment.  Well it would be nice if it could save equipment and NOT sacrifice itself in the process, as it appears the BrickWall filters can do, for example, and Panamax/Furman seems to use a similar series suppression technique in their UPS's.  Of all the non-Monster surge strips and supressors I've bought over the years, only about 1 other unit has lost its surge capability.  All the others, regardless of age, still show the "surge working" indicator light.  With all it's nice looking filter parts, it might have been nice if Monster could have made the 7000 repairable, for example by having a pluggable MOV or avalanche diode section.

Anyway I like the idea of UPS and active voltage monitoring to protect my expensive equipment from all the possible kinds of power events, and I like not having to manually reset stuff after a power failure, and the living room hosts my two Tuner +Sonos nodes that I use elsewhere in the house at any time of day, so having an actual UPS is an actual and not an imaginary benefit, though perhaps not a huge one.  I've used a UPS/conditioner in the living room for 6 years now and I'm not sure I want to go back to a non-UPS solution (though, unlike the Kitchen system, which also has DVR's and security cameras, it's not an absolute necessity).  It's likely nothing else could be proved to sound better, but convenience is convenience.

(Although it IS inconvenient to have to replace the batteries every 3 years or so, and that has given me some pause already.)

I would like to see an online UPS which absorbs, rather than creates, further noise and distortion back at the wall outlet.  With such a UPS I could plug my line level equipment into it, and the power amplifier as always straight into the wall, and have it be all plusses.

In fact, the effect of any given piece of equipment on the wall power should be an important design consideration.  Devices should not only be themselves immune to powerline noise and distortion, they should tend to absorb rather than create them on the power line.

This is similar to my room equalization concept, where modes present anywhere must be absorbed, if not completely at least partly.  It's not all about me the one optimal listening position.

I believe generally speaking an incandescent light bulb is said to have such a property: cleaning up the powerline it is plugged into.  That's not what we should use, they simply waste power, but power supplies should be designed to be like that.

Powerline noise and distortion is a fun thing to think about, and I think for some a ready excuse for perceptual variability, but not really the hugest factor for most people I think, and especially me (because of my high quality utility power, dedicated circuit, and well designed equipment).

It seems, actually, so long as I wanted the UPS features (and...I could possibly give them up for this system) there aren't many choices.  Either mega expensive PurePower, or Furman/Panamax, or some server grade unit with quiet fan...my research last time made me pick the Cyberpower PR1500LCD for the kitchen and I'm very happy with it.  Not only is it perfectly quiet, it seems to have low output noise.  It's highly programmable just from the detachable LCD screen (with same functions available in computer app but I haven't tried that).

Now I haven't researched APC much.  My old server grade APC was noisy, provided noisy power, and pulled the bedroom circuit to it's knees just to provide "stable power" when there was hardly any actual problem.  Since then APC has made a series of UPS's where they at least suggest Audio/Video usage: the J, H, and S series but I can't find any but J anymore (I recall the S as the sinewave series and quite expensive like $1500).  The J is stepped not sinewave and amazingly cheap.  Strangely it looks physically like the Panamax/Furman.  Perhaps the same factory makes both models, the Panamax/Furman being the sinewave upper end model which might have been the APC S if APC sold the S anymore which they don't.

Now about why a line filter, smoother, etc might be useful...

It's always said "well designed equipment doesn't need one" which is of course begging the question.

But why would, in principle, it be advantageous.  In non-audiophiledoublespeak.  I'm not going to say it's because the power is everything, everthing else is just modulated power.  That's audiophiledoublespeak.

Well power like crap could cause some sort of peculiar problems, like power supply motorboating.  But I'll leave those aside for the moment, and just focus on the obvious.  Any piece of equipment can only reject the line noise to a particular degree, what you might call the AC Noise Rejection.  This is sort of reflected in the Hum and Noise specification, but some of the Noise, and even some of the hum, may be generated subsequent to the power supply.  AC Noise Rejection is the specification we actually want.  And we can imagine for much equipment you might have an A weighted figure not much different from Hum and Noise, say -90dB.  This is caused by electromagnetic chassis coupling and by failure of the power supply circuitry to eliminate all the noise.  Even sophisticated regulated power supplies do not have infinite line noise rejection.  But for "well designed" equipment it's in the same low range as the audio material itself.

But this hides the fact that the actual noise generated can be very spotty in it's spectrum.  And furthermore not all noise is equally bothersome.  And the AC Noise Rejection itself can have irregular frequency response.  So if a burst of particularly annoying noise occurs at a particular high spot in the frequency response curve of the AC Noise Rejection, it can, in principle, poke through to perceptible levels, even if -90dB of say pink noise would not be a problem.

In a real life complex system, the vulnerabilities to line noise are increased.

So it's reasonable to believe, at reasonable expense, pre-filtering ac line noise is a good idea.  Etc.

I have yet to be convinced of the need for crystals, quantum field generators, and the like.

The objectophile objection is written succinctly by Ethan Winer.

Ethan says it's all just electricity, which is easily measured.  "All competent audio gear rejects the normal amount of noise on the AC line."  You can measure the noise at the output of your power amplifier, with and without conditioner in place.  If you have a good amplifier, there will be no difference, which means the power conditioner may be cleaning the AC power to the power amplifier, but it makes no audible difference.

Well that's exactly the sort of thing I created this blog for.  I did at one time make such a measurement, and I decided that the conditioned power reduced noise measurably.  By now I completely discount that measurement, I believe now it must have been error.

It's actually not easy to do such measurements.  I would agree with Ethan that a single number noise measurement is likely not to show any difference.  But the best way would be to produce a full spectrum.  I think there is likely to be a visible difference in the noise spectrum somewhere below -100dB from full signal level.  Does that make a difference?

I suspect it would not make the kind of consistent difference that would be required for DBT.

But it might make audible differences in two kinds of ways:

1) Just enough difference at some moment of music to change the momentary perception, though not necessarily in a repeatable way.

2) Difference at some moment when the power is unusually noisy.











Wednesday, May 4, 2016

The Playlist is The Music

The Playlist is The Music, in so many ways.

Even just picking one song from one's collection, or deciding to listen to an FM radio station, or an Internet rebroadcasted radio station, or an internet originated radio station, or a station within a paid service (and which and which), or a particular song on a particular service, has long been a hard thing for me to do, and not getting much easier.

These things don't just come to me, as I imagine they might for a different sort of music listener, maybe even most.  Perhaps just people have something in mind as they walk into their audio room, or flip on their phones (I so rarely do the latter it barely counts).

Though to be honest, the biggest thing stopping me from always having music playing, in my post-23* audio decadence years is just to decide to listen to music, or even have music on (which is 95% of how it starts for me, as background music, though some considerable fractions of such become serious listening as the system is sounding so good it pulls me in nowadays, rather than seeming slightly astringent).

(*In my first few days of personal audio, as I quickly progressed from an Admiral radio alarm clock, tube based, to a beautiful and clean Fisher FM 80 I picked up for $20, to a tube based reel tape recorder, to a Dual 1209 with a handful of records.  Certainly before I got my records it was easyist, I simply turned on the radio (last station) or pressed play on the tape recorder.  But I don't remember any difficulty choosing music until at least 23 of age, about the time I started working for a high end audio store.  Perhaps listening became too serious or something.)

So a playlist makes this burden a lot easier.  Or at least you then only have to choose on thing: which playlist, and then all the rest of the music is selected.

I had this hope with computer network based audio systems I could simply let it choose music at random.  I started on that project in 2005 when I bought my first set of Sonos modules (I now have 6 active zones).  It has never proved to be satisfactory, despite endless work.

Towards this end, I have always organized my music in 3 different folders.  Sonos conveniently allows you to select an entire folder.  And then you can choose "shuffle."  I had the main folder with no-talk classical music, that I can listen to as background (I don't like talk in the background, it's distracting) or serious.  And then everything else, rock, pop, opera, are in a different folder.  Then a third folder of stuff I own but don't really want to listen to much, because it's too loud, or jarring, or whatever.

I did this folder shuffle play for several years.  But ultimately it became tiring.  Even as my collection expanded the sense of hearing the same songs over and over didn't go away.  I now figure that listening to a collection of albums as a collection of songs doesn't work.  If anything, you need to be able to listen to random albums within some category.  I haven't figured out how to do that in Sonos.

Sonos album selection isn't as nice as on some other systems either.  It gives you a huge scrollable list, either with scroll bar or alphabetic shortcuts or both.  Well that works if you know what you want.  But if you're trying to decide, a full page display of album displays works nicest for browsing, and I haven't seen Sonos do that.

As I never listened to pop radio much, I've never picked up on the hot music such a listener might find they have to have.  So how even does (or did) one decide what to get?  Possibly listening with friends, then extrapolating, that's about what I did long ago.  Now I do that again with the audio society, though not so much extrapolating.

Mostly for quite some time, including now, I get my original recording purchasing ideas reading audiophile magazines, Stereophile and The Absolute Sound.  Nowadays I'm cutting out pages with interesting reviews.  I don't actually save the magazines anymore, though I did once (and still have all of those).

I think it would be wonderful to have playlist compiled from such reviewing magazines, or reviewing sites, famous reviewers, etc.  Here's Prince's party playlist, this should be on every subscription streaming service.

I'd also like to be able to share playlists with my friends, say for example listen to what they listen to or play their playlists.  My network at work has had such a feature.  I'd like to have that kind of sharing with my audio society friends.

Streaming services need to have "stations" corresponding to sub genres.  A station is an endless playlist.  Pandora has been the best at letting me create stations according to my wishes.

Speaker Cable Resistance vs Inductance

Update:  I've found the Audioholics page where they actually measure (not just simulate or calculate) the high frequency response of several different cables.  As a result of this, I'm no longer going to argue that thinner zip cord has lower inductance or better high frequency response than thicker zip cord (as I did on the previous version of this page, based on an incorrect calculation).

This information is far better than what I previously had at hand, and I'm revising my conclusions.  (When you get new data which leads to different conclusions, what do you do?)

Their results indicate that:

The inductance of standard zip cords does cause measureable high frequency loss.  All zip type cables have significant loss due to inductance at 20kHz.  I was correct about these ideas.  However, the loss *is* sufficiently small, smaller than simple-minded calculations I was using previously, that I wouldn't much worry about it in 10 foot lengths and for gauges as thick as 10 gauge.  At 50 foot lengths it does become more important and I'd suggest 4-cross cables (as I used to always suggest for thicker gauges).   As standard 16 gauge wire was not tested, I can't directly compare, but it is looking like the high frequency loss in 16 gauge zip cord isn't better than 10 gauge and I'm getting a sense now that inductance actually decreases slightly as you go to thicker zip gauges, though I haven't yet found a useful table that compares multiple gauges of identical zip design.  (Previously I had the incorrect idea that inductance doubled up as you went to thicker gauges, and that was clearly wrong.)

What's going on?  The low frequency inductance doesn't increase as much as I had previously believed for the larger gauges, and peculiarly the skin effect works to reduce, rather than increase, inductance per se, and this may be a more powerful effect for larger gauge cables.

At ten foot lengths, the actual measured losses with old generation zip type Monster cable in 10 gauge is less than 0.1dB at 20kHz, and perhaps more pertinent the the increase in 20kHz loss beyond insertion (DC) loss is a tiny 0.02dB.  Even doubling these losses because of my 2 ohm load at 20kHz doesn't make them very important (the Audioholics tests used a 4 ohm load).  I believe standard 10 gauge zip cord would measure similarly to the old generation Monster.

What does look very bad, as I had believed before, is the cultist idea of physically separating the conductors.  Any significant degree of doing that causes large high frequency losses from the increased inductance.

Here's another interesting set of measurements, especially see Figure 4, Series Impedance showing the impedance curve of many different cable.  The zip cords follow a familiar pattern.  The heaviest zip have way lowest impedance at DC and low frequencies, but start curving upward in impedance earlier.  The "Fulton" cable tested I know is a very heavy zipcord, like 1 gauge, but I can't exactly remember, that shows these tendencies clearly compared with the 18 gauge and 24 gauge zip.  Of all the zip cord types, the Fulton has the lowest at every frequency despite starting to curve up earlier, mainly because it's curving up earlier from a much lower starting point.

Thinking about this does make me wonder again.  In a certain sense absolute impedance matters.  But change of impedance matters too, perhaps more in some cases.  I actually do think it's wrong when, as with the Fulton cable, the point at which the cable itself has many times it's DC resistance is still a fairly low frequency, and rising up from there, so at 100kHz and above it's just bairly better than 24 gauge zip.  Is this wrong?   I'm having trouble thinking about it...a huge change in cable impedance actually having little effect--precisely because it's so small in the first place, and essentially unimportant.  Do I care if the output of my speaker is 80dB or 80.1dB?  Not much, I can usually turn up the volume more if I want to.  But if the highs roll off by more than 0.1dB, I'd be concerned.  I'd like to see less than 0.3dB rolloff in any case.

With even 12 gauge zip, it clearly doesn't matter, there's a migh higher hinge point and not much change until 10kHz.

This may be the best way to think about it:  The cable impedance variation is important only as it relates to the speaker load impedance, not with regard to the cable impedance itself.   In the limiting case, if the cable had 0 ohms DC resistance, even the tiniest rise in impedance due to inductance would be, relative to 0, infinitely large change.  But that doesn't matter, what matters is change that approaches the load impedance as it approaches the load impedance.

Given a particular load impedance, say 4 ohms, what matters is an cable impedance which would cause a 0.1dB change.  A 0.1dB change is about a 1% change (1.158% to be more precise).  That means the cable impedance has to approach 1% of the load, or 0.04 ohms, to be significant.  Changes in the cable impedance below 0.04 ohms are unimportant, even if they represent a large proportion of change such as 1e-10 to 1e-3, which is a millionfold change, doesn't matter because it is still well below 1% of the load.

Now a thinner cable has more DC resistance, so it changes the baseline for assessing this change, but this doesn't have much if any effect.  Suppose we had a cable whose DC resistance caused a 0.1dB loss for DC.  It requires the same exact 0.1dB additional loss at high frequencies to be important.  So that hasn't changed, but what additional impedance is required?

In our 4 ohm example, a DC loss of 0.1dB requires 0.04 ohms of resistance.  How much additional impedance is required to get us to 0.2dB of total loss?  It's going to be slightly more than 0.04 ohms but not much.  About 1% more, or something like 0.0404 ohms.

Now looking the impedance curves of different gauges of zip cord, they seem to approach being the same at frequencies above 20kHz, with the larger gauges always having the lower impedance.  My sense then is that the larger gauges are always going to be better, having more extended highs, even though they have less "baseline" margin as described in previous paragraph.  The baseline shift doesn't buy as much as the added inductance losses of the thinner cables.  (I intuit this but I'm unable to prove it yet, and it might be close or slightly reversed at intermediate frequency ranges.)













 


Sunday, May 1, 2016

Major Major Upgrades

Response Measured May 1, Sub below 100 Hz nearly flat
Many things going on in my life, too little blogging about the audio updates, but there have been many, and the state now is....the improvement is staggering...and this isn't BS.

Actually the biggest change isn't even in the audio system per se, but with my new central air conditioning system, which has variable speed compressor and variable speed fan.  The fan runs constantly at an inaudibly low speed while temperature and humidity are nearly perfectly controlled.  (Not to mention, when the old unit kicked in--which was constantly while running the Krell, and intermittently otherwise, it made a loud buzz clearly audible inside.  I have never heard the new outdoor compressor inside, nor the fan on low speed, which has all that's been necessary so far.  And I could not run the fan continuously during cooling months...it would lose all humidity control.  I like the constant ventilation, which includes AprilAire 2200 filtering.)

It gets high marks from friends who don't like blasting air conditioning, and I do indeed check the temperature and humidity in various rooms a lot, and this is amazingly good.

One aspect of that deserves further mention, and that is the relocating of the thermostat away from the Living Room, where it was positively interacting with the Krell Amplifier.  A few minutes after turning on the Krell, the AC would start running...and never stop.  Air would bounce directly on the amplifier and back to the thermostat on opposite side of the room, keeping the AC in constant operation.  Even if I set the thermostat a few degrees higher, the temperature at the thermostat would quickly catch up.  Meanwhile, the rest of the house was being chilled colder and colder.  The Krell and Thermostat were having a one on one, even though it's actually on the opposite side of the room, but that's where the AC air is directed on high speed fan, which the old unit mostly used when it was cooling.

Now the thermostat is in the master bedroom, and I can run the Krell all day without causing the AC system to run much more than otherwise, and you never notice it turning on or off anyway because the fan just keeps on running at inaudibly low speed.  The bedroom stays at a perfect 78 (the setting) while the living room gets a few degrees warmer (not bad, and remember there is continuous ventilation which is keeping things from going to the other extreme)  instead of arctic blast next to a toasty furnace.

So now, I use the Krell FPB 300 all the time, and that's another huge upgrade right there.  Previously I had to use the Aragon 8008 mostly, and I didn't get around to hooking up the Krell often.  It's easy to turn the Krell on and off with the soft touch metal button on the front, just as easy as any kind of remote control if not moreso, so who needs remote controls?  (Actually, I do have the Krell infrared.)  I do turn the Krell off when I'm not playing music.  It's a huge power waster otherwise, and could cause excessive wear on the output transistors.  It runs up to 180 degrees on the heatsinks by design, and mine has some problem in the left channel which tends to keep slowly ramping up the bias until high temperature causes it back down to level one or two.  So it will ultimately be running up to 180 degrees (F) even if just idling all day (measured the other day in the 155-160F range).  The rebuilt right channel doesn't have that problem, it idles closer to 145 degrees.  The left channel will ultimately need to be rebuilt.  My plan it is to run it until it actually breaks.  (This is basically the way it came back from Krell service in 2011, with some bias instability in the left channel.)  But no reason to use up the remaining life in the left channel in idling.

I understand now how the Krell operates, and my Kill-A-Watt was sometimes overloading (it blinks and ultimately shuts off as watts go above 1400), and I did indeed worry about the previous setup which included Insteon controls for both amplifiers, the Kill-A-Watt, and lots of jumper cords and so on, all at least 14 gauge.  I've seen how high current causes something almost like fire to shoot out from sockets of all kinds, expecially it seems extension cords.  Best not to have them for audio reasons too.  But especially with the Aragon over in the barely accessible corner with it's heavy rocker switch.  The Krell soft metal touch switch is the greatest thing.

So now, the Krell plugs straight into the wall socket (the thickest gauge all brass Pass & Seymour, in a dedicated insulated ground circuit using 10 gauge wire) using only its Krell power cord, exactly as Krell instructs to do.  That is certainly the safest, and maybe even the best of all things to do.  Another serious upgrade compared to to how things were at the last audio party--Amp straight into wall instead of switches and meters and extension cords.

Both subs now have single power cords the entire distance.  The right sub now has an audiophile power cord, 14 gauge with multiple conductors and Cardas copper.  It was cheap enough and I could get exactly the right length, so I could toss the stupid 8 foot 16 gauge extension cord.  Another Upgrade!

Since I only have two outlets (one duplex outlet) on the dedicated audio circuit, I must use a Y adapter somewhere, now right at the outlet itself, with both subs and the Belkin UPS power conditioner plugged into it.  That's not really an upgrade just a minor reorganization to allow the Krell to be plugged straight into the wall instead of the power conditioner.  This is the heaviest duty 15A Y adapter you can get.  It's much safer right at the wall socket then it was before (before it was sitting on a pile of books on the left sub--don't do this).

The speakers have been angled, measured, reangled, repeatedly.  My angle idea is pretty wide angle, about 15 degrees toed in being parallel to the walls, which means about 15 degrees towed away from the listener.  One advantage of this width is that there is very little change in sound a few degrees in or out.  It is slightly rolled off at the top, but I've decided that sounds OK for now.  Much work was done on this since the party.  The effect is a very relaxed listening position, yet very good imaging, yet spaciousness, everything.  It almost seems like there is some magic effect in making a small angle from the wall, which is precisely what Acoustat recommended.  As if the relationship to the wall matters for reflecting the backwave too.  For whatever reason this angling seems best, and I'm not sure how many upgrades to count it as.  Back last year I was listening the Acoustats just off axis, which was generally unpleasant, but I accepted it sometimes and got used to it others.

(15 degrees isn't the exact measurement, I've forgotten the details when I was measuring such things, and they may have changed since then anyway.  Remeasurement is needed soon.  Even the tape is off right now because of what I'll describe next.)

The last change was to move the speakers in closer together.  The wider speaker speaker angle had combined with the distance between the speakers to make the sound phasey sometimes.  Moving the speakers back closer together just a few inches tightened up the image completely, side to side, nothing phasey about it now.  Meanwhile the image is usually completely in front of the listener, from  one speaker to the other, indicating correct focus.  I could only move the speakers a few inches closer without a larger rearrangement, and that's what I did.  I might even move the speakers closer if I could, just to see what that would do, but I can't.  Anyway, this seems to be a good point.  Once again, this is a super huge upgrade, going from a phaseyness which had been getting worse as everything else got better to tight focus.

The bass EQ is organized around a completely different ideal.  I no longer accept a non-flat "room curve."  Flat seems to work best, matching the electrostatic panels nicely.  The bass is far more extended, but actually has a bit less boom than the Acoustats run wide open.  I've tuned the bass by running Genrad oscillator over and over and over, and tweaking individual Paremetric Equalization functions (PEQ's) in the Behringer DCX 2496 crossover.  The bass level was lowered there and also at the digital controls of the SVS PB13 with the new Sledge amplifier.  I think I used to have that at -3dB and now it's at -9dB.

In addition to the measurement, I've included my observations both at the listening position and around the house at places where room modes seem bothersome.  I tamped down all these modes whether these made much difference at the listening position or not, even accepting a slight loss in measured flatness at the listening position in order to apply cuts at the locations where the huge room modes were apparent.

Just tamping down these out-of-room resonances by 3dB seems to eliminate all the out of room issues.  I haven't even noticed the room modes anymore, anywhere (though I imagine I will eventually) it's such an improvement.

And there's been endless retuning by ear, but most of the EQ adjustments come from finely turning the oscillator back and forth and now using the least amount of EQ, preferably the highest Q, to damp out model type problems without adversely affecting nearby frequencies.  I've been much more systematic about canceling out problems without creating new ones.

The totally new bass ideal and achievement might even be the largest improvement of all.  Everything is different now.  I can organ music at 0dB and still now have the speakers and/or walls coming apart.  The bass is just always there, and always seems right, often surprising with it's impact, but never too full, and not just at the listening position.

Actually it seemed (though it could be a mistake) that at high levels the Acoustats had a buzz around 85 Hz.  So I moved the LR 24 crossover up to 100 Hz.  BTW I no longer cheat with an LR48 rolling off the subs and an LR24 on the panels.  Both crossovers are set to LR24 at 100Hz.  This doesn't seem to help so much near the crossover but across the range an octave lower and higher.  This also made a few bass modes around 80 Hz in the room more apparent, but they are usefully cancelled out through the DCX for the subs (while generally I have avoided non-crossover EQ for the panels).  Another huge upgrade, or several!

Along with reducing the bass a lot, so there is zero rise over the midrange level, I reduced the highs so that (at one point...for some reason not in the measurement shown above in the highs because of technical issue) there was no rise in the 20kHz bar.  At one point I was showing a 12dB rise at 20kHz, the highs only slightly drooping before that.  That 12dB rise was entirely due to the supertweeters, which I then (and now) cross over at 20kHz.  I somehow convinced myself that the measurements were wrong, this added magic something to the sound, making (as I said above) sound real everywhere in the room, and in every room.

But then one day in the past few months I did somehow hear the supertweeters more clearly.  And they were adding a nasty sound.  So I kept turning them down, and I basically got to flatness and they sounded OK.  (I couldn't actually go lower than flatness, though it doesn't look that way in the above graph, I think I had the phone turned around).

So, much better supertweeter level, sounds cleaner and measures flat, a big upgrade there.

I've decided I like the sonic sugaring of a Gundry dip.  Linkwitz has defended this on grounds related to the sensitivity of the ear at different angles, and the stereo configuration boosts the apparent highs around 6kHz, which is also a region the ear finds offensive in excess (metallic, etc).  As soon as I tried it, using a very low Q PEQ around 6kHz with 3dB loss I decided it sounded correct.  I haven't tuned this as much as I might, but it seems to work, and I'm counting it as another important upgrade.  The Acoustats seemed to measure too high in this region especially on axis anyway.  It's possible I should revisit the speaker angling because of this change, or examine the tradeoff between the too.  I only started playing with the dip after I had decided on the current angling.  This is almost the only non-crossover PEQ used for the Acoustats.  It is dialed in through a Behringer DEQ 2496.

I discovered that my Sonos system is applying digital gain to the living room.  I have reset the volume level (using Fixed doesn't help, that makes it fixed to 10dB digital gain) to -10, which cancels it out.  So things that were sometimes sounding distorted no longer do.  Upgrade!

I rediscovered two of my old LP-to-digital organ transfers.  I hadn't much listened to the Robert Vickery recording because the huge deep bass got out of control.  But now I listen to it at 0dB and higher.

Another old transfer which was one of my favorites, but strangely I never listened to it much, Magnum Opus Volume One with organist Welch.  It turned out one channel was 6dB lower.  I thought I had made a fixed transfer, but that was not what got ripped to my harddrive, or maybe I hadn't.  It took a whole evening to figure out how to fix this with any of my digital editors.  I couldn't get SOX to do it at all.  Nor Audacity.  None of these programs seemed to have a way of changing the gain in only one channel.  Finally I figured out how to get the job done in Wave Editor.  I put each channel in a separate layer by turning off a different channel in each layer.  Then I could change the gain for one layer.

So now with all the upgrades I can listen to these organ recordings at digital 0dB and higher (which is approaching 90dB in my low sensitivity system, about 96dB is my max, the Tact has about 6dB of digital gain).

The third bass note in Spanish Harlem now, for the first time ever, sounds right.  I think that was relating to resonance around 80 Hz.





Tuesday, April 5, 2016

Another two failures, one fixed by reset

Back in February I found that the Behringer DCX 2496 digital crossover had failed.  It wasn't turning on at all.  I was very busy at the time, so I decided to go two weeks before setting up a spare unit I had on hand.  Meanwhile I could listen to the Acoustat 1+1's full range.

I didn't like the full range sound.  Very lacking in bass.

But then as I was setting up the spare DCX, I discovered that the DEQ 2496 I was using for the Acoustat crossover (80Hz to 20kHz) wasn't working properly.  Regardless of settings, even if I had all parametric EQ's turned off so it couldn't possibly be crossing over, it was rolling off the bass around 300 Hz.

So this means my assessment of the "full range" of the Acoustats was possibly in error.

However I was able to fix the DEQ simply by re-loading the "default" settings and then reprogramming it.  It hadn't actually died, it had gotten confused.

Programming the DEQ for a Linkwitz-Riley 24dB/octave filter is easy once you've figured it out.  Basically one must use the "LC" parametric setting at the crossover frequency, and then dial the cutoff to -15dB.  But that -15dB is not really correct, it really keeps rolling off as far down as I've been able to measure at 12dB/octave.  Putting two LC's together gives 24dB/octave rolloff, with -6dB at the specified frequency, exactly as Linkwitz-Riley requires.  The L12 and H12 parametric settings don't work well for this.






Sonos Digital Clipping



One of the more important discoveries I've made recently is that the Sonos Connect in the living room can clip it's digital output when the Sonos zone volume control is set to maximum.  The highest volume level that be set without any danger of digital clipping is about -10dB (10dB down from maximum).  When I say clipping I mean the top portion of waveforms is being chopped off, and it can sound very badly distorted.

This is a huge bummer in my opinion, and I believe it is a change from the way Sonos was when I first started using it in 2005.  Back then I recall that when you set the volume control to maximum, you got "0dB" gain, so that recorded music which just barely hits the top digital level, also known as 0dB, came through unaltered.  Back then I also determined that the volume control operated in 24 bit mode, so that you could reduce the volume level as much as 48 dB (8 bits) without losing information.  Now you have to be careful on the high side so that your music doesn't get clipped.

The bummer is even huger because most Sonos interfaces, such as on my Mac and on my iPhone, don't give any repeatable way of setting the level control.  There are no markings and no numbers, it's a fully stupified interface in which you can slide the bar up and down, but with no clear indication of where you are unless your eye is sharp enough to see pixels, and you can't even be sure you can trust pixels.  I really hate stupified interfaces like this.  You know the digital system knows exactly where the level is, but it isn't telling you, just giving you a crude indication like the crudest of crude old potentiometer controls.




Only the original but now sadly discontinued Sonos Controller CR100 had markings on the volume control display.  There were 5 major markings including top and bottom, and between each one there were 6 notches.  I believe this corresponded to 12dB between each major marking and 2dB per notch.




Fortunately I still have a few CR100's.  It appears I can completely eliminate the possibility of digital clipping by setting the volume level 5 notches down (as shown in the picture above) for -10dB.  So it appears that sometime in the past 10 years Sonos has added 10dB of "digital gain" which can produce higher levels than in the recording itself.

Digital gain is often useful, especially when you have ad hoc recordings or use line inputs.  I myself use the line inputs of the Sonos system a lot--and it was the reason I chose the Sonos system over competitors in the first place.  I can have a tuner or two in one room where I get good FM reception, and a turntable in another room, and a tape deck in another room, and still use them all in my whole house audio system.  I love being able to route analog signals to different rooms, sometimes several rooms playing the same source, sometimes different sources.

But digital gain isn't a good idea generally for professionally produced recordings.  Especially on popular music, often the recorded music comes close to or even hits the top level, 0dB.  So if you add digital gain to a signal which is already hitting 0dB, the maximum digital level, you are going to get serious clipping.

And the Sonos digital gain isn't just applied to line inputs, it is applied to the commercial recordings, mostly from CD's and some from downloads, on my hard drive.

When researching this issue I did discover that Sonos now observes the Sound Check option in iTunes.  This could apply digital gain adjustment.  I'd never known about Sound Check and had never used it, I checked to make sure I hadn't set the Sound Check option in iTunes preferences, and I hadn't.



Setting the volume level to "fixed" doesn't help either.  If I set the Living Room system to fixed, it still adds the 10dB of digital gain just as if I had the volume control to maximum, and I will still get serious digital clipping on commercial recordings.

This may be true for all of my 8 Sonos zoneplayers, but I haven't actually tested them yet.  I need to connect a device with digital level readout, such as a Behringer DEQ 2496, and then play test tones at -20dB.  If this plays at -20dB or less, there is no digital gain and therefore no danger of digital clipping.  I was able to do this easily in the Living Room because it already has a DEQ in the signal path.  I see I still have a "spare" (not yet set up for it's intended role) DEQ I can use to run these tests on different zoneplayers.

I'm thinking that it's possible I forced the Living Room zoneplayer into a "gain mode" accidentally by pressing on "volume up" too many times.  I seem to vaguely recall getting some warning that I was setting level too high once.  But there appears to be no option or setting which permits me to undo the unwanted digital gain, and searching online hasn't revealed anything yet either.  I should contact Sonos about this, but that's yet another thing to do and I've been very busy.

The living room system is now so finely adjusted that I can actually play at about 0dB or very close to it on many recordings.