Background Music is Important

Charlie Hansen was not unique is feeling that Background Music is unnecessary and unimportant.

Perhaps, to alphas like him...  Actually many "serious" audiophiles say that.  Another reason the Audiophile hobby is dying.

(To be fair, audiophiles in my local society are Post Judgemental for the most part.  There's little hectoring people over this or that "transgression," as I'd seen earlier in my life.  And though I'd certainly done it and perhaps even more in earlier years anyway, and even then I'd never heard anyone disparaging everything but "serious listening" except in print a few years ago.  It just wasn't a concern, people do what they do, audio "issues" identified by sighted listening were assumed to be always apparent regardless.)

Anyway, to most actual humans, and especially to people who live alone like me--if not all others, background music (or something) is nearly essential to sanity.

Nowadays "or something" has often become some flavor of "news" broadcast.  Those are often stressful, leaving you concerned about everything but connected to nothing, except the next day's broadcast, to feel on top of nothing much more.

I prefer to play music all the time, and most of the time music without words that allows me to just get on with things, but just thrilled slightly by the music.

So I have two icons for my automatic music playlist generator, either music without words (classical and much new age and jazz) and too much tension, and all the rest.

That's the way to do it.  It never seems to work that way on any music streaming service.  Even if you choose "Electronic" you'll eventually get some vocals.  And you'll often get music that's very stressful, like Oldfield (sometimes) or Ligettii.

I just heard about Oldfield for the first time yesterday, and I had to check him out, starting from his classic album "Tubular Bells."

Well starting from Side One was basically impossible as background music.  It's just too stressful.

In that regards, may I recommend myself, an Artist named Charles Peter Peterson, and my one album, Mythic Rocks.  It becomes slightly stressful in parts, but never enough to disqualify from being potential background music.  That's integral to my music "development" process, which is mostly background listening and weeding out the stuff that absolutely doesn't work.

The local Classical station KPAC and Jazz station KLRN generally work as background music too, as does KSYM when playing Musical Starstreams.

Anything Ambient from Eno works, of course.

Recovery, and HDCD again

Sunday night I decided it was time to play some vinyl.  It had been over a month since I'd last played a vinyl record...one was still sitting on the platter waiting for side 2 to be recorded.  Things are often like that here, and this time I have the excuse of the Holidays.

 Cleaning the stylus with my usual device there wasn't the usual predictable noise, but instead just one coincidentally timed full power pop from my system.  I checked all the levels and if anything they were lower than normal.  Then there was silence when I tried to clean the stylus again.

I tried playing records, even just lowering the stylus to the record.  At first I though I had killed my cartridge.  Perhaps I should never have trusted a "sticky" stylus cleaner, especially one in a metal box.

Once in awhile, there WOULD be another pop, typically when "cleaning" the stylus or lowering it to the record.  But then after that there would be silence (and I was amazed at how little stylus chatter there was too).

I tried re-adjusting the tracking force, rebalancing the arm.  Then I tried removing the damping tape from the back of the arm where it contacts the cueing lifter.  Perhaps the tape had swelled so the stylus wasn't actually making contact.  I currently have difficulty seeing very well close up so I wasn't sure.

Anyway, after removing the tape in the back, of course I had to re-balance the arm and set and check the tracking force all over again.

I was in greatest fear that I had damaged the cantilever with my stylus cleaning device.

But finally it dawned on me that it probably wasn't the cartridge.  With the best reading glasses I could find, the stylus looked OK, I just wasn't used to seeing it in the cartridge wrapped up in tape.

Then I suspected the preamp.  I tried playing other inputs and they didn't seem to work either.  Notably the Denon 9000 on the bottom.

The next morning I removed the new pile of equipment beside the rack so I could remove the connections from the preamp so I could replace it with the other XSP-1 in the bedroom which is mostly unused nowadays.  I also had to clear my bench of the Adcom 535 I'd been checking out (update: for awhile, I believed my conclusion that new filter caps were needed was shown to be in error while measuring the other channel.  There seemed to be only some 60 Hz noise in the distortion residual.  But going back to the first channel, I could now see clearly the "distortion" was that same 60 Hz plus a larger amount of 180 Hz plus some actual 2nd and 3rd order distortion.  So it's only one channel that needs the caps...good thing as I only ordered one pair which has now arrived, and I didn't want to spend even more money on this amp).

But, on the bench, I could clearly see that the Emotiva XSP-1 was working fine.  Something else was wrong with my living room system when playing records.

It occurred to me that the Denon could have gone bad too.  Or something else which had been nearly unthinkable...

The Lavry AD10, after 12 years of continuous use, always being left on, had died.  (In future I'm tending not to leave things ON always as I think it shortens their lifespan.)

The Lavry "seemed" to work in that you could set all the settings.  But level lights weren't responding to any input.  Nor was there any signal being output.  I tried switching the clock to Internal and that didn't help.

Currently the Lavry AD10 is the presumed latest failure, and it's on my bench for checking when I get around to it.  Perhaps the problem was actually a cable connection and not the ADC after all.

So now, at least until I get the Lavry fixed (or proven working again) I had to rig up a completely different analog setup.  I have to convert "analog" sources to digital to even hear them because my system depends on digital processing for EQ and crossovers.

Actually something that I had long considered worthy of trying anyway!

The output of the Emotiva is now going straight into the Tascam DA-3000, which is now dedicated to that purpose.  It plays on my audio system when I press "record" (which is actually "record pause") on the Tascam.  So I'm using the Tascam AD converter, rather than the Lavry AD 10 converter, to hear analog sources, and possibly record them.

The Tascam uses later generation analog to digital chips, and is itself quite high end featuring 192kHz maximum PCM and even 5.6mb DSD.  (Tascam is no longer selling this in the USA???)

Though I imagine a real record label would use something even more special, like a Lavry Gold or Weiss AD converter for PCM anyway.

But the Lavry has always seemed perfect, and the peak level holding lights are perfect for finding the best recording level for any vinyl record.

Why do so few devices have peak hold lights like that???  The Tascam surely needs them.

Also the Lavry may use a slightly higher grade of earlier AD chip.  It may have better AD engineering, and it may gain from being a "dedicated" AD converter as opposed to the converter section of a recorder.  The Lavry "Dynamic Range" spec is a few dB's better than the AD-3000 "S/N" spec, but they are following different standards so it's not really clear which is better that way.   The Lavry has even lower distortion specified at 0.0009% as compared to 0.003%.  Either is as good as my preamp.

Roughly speaking, both are at the same level, which you might call something like "prosumer" or "pentultimate".  So they would be worth comparing perhaps.

My initial impression is that the DA-3000 converters, running at 24/96 (which is my standard for "all that's necessary") sound a bit more transparent than the Lavry, but not as smooth and relaxing as the Lavry, and the Lavry is very transparent also (not unlike a good preamp, I've always said).  So it's almost a draw but it still seems I might like the Lavry better, plus it's peak reading meters.

To continue to have a separate digital path for both playing the FM tuner as background music, AND recording special FM programs, I needed to set up my second digital recorder, something I'd wanted to do anyway.  I removed my old Marantz 480 recorder from the living room system when I got the DA-3000 because the DA-3000 was clearly better in every way.  I was hoping to sell the Marantz (I see they're going for as little as $280, so hardly worth bothering).  Then I got the idea I could put the Marantz back up so that each recorder would have a dedicated purpose.  The Marantz would be dedicated to recording FM, and the DA-3000 would be dedicated for recording vinyl.  That way I wouldn't have to be constantly changing the settings (Sample rate, SRC, and input source).

So this was already turning into a two day project, with all the items (and cables!) that needed to be removed from the rack, and possibly repurposed into a new arrangement and connections.

I also needed to replace the Denon 9000, now also presumed dead (no apparent audio output) with my second unit, which (on last testing) had been able to play DVD-Audio discs.  I had been saving the second unit for that purpose, AND because the first unit I got was always the one that I thought might sound best on CD's and HDCD's, even though it was unable to play DVD-Audio discs anymore, after $150 worth of repairs done by a Denon center didn't help (they said I'd start having to replace boards like the power supply board).

So, I'd wanted to exchange these units as well, which will enable me to make another sampling of the DVD-Audio disk Supernatural by Santana, which seems to sound the most magic when played through Denon DVD players, with the DVD-9000 being the ultimate king of the hill-but never recorded before by me (because I usually had my non-DVD-Audio playing Denon 9000 hooked up).

But now, before before the Second DVD-9000 dies, I also wanted to address a long lingering question about HDCD.  It has been claimed by many audio engineers who have studied the matter (but not Pacific Microsonics, which no longer exists, nor Keith Johnson) that no HDCD decoder uses the "transient filter" that is toggled on many HDCD recordings.  It was apparently used when the recording was being made, and one of the reasons the PM One and Two recorders were so good.  But no decoder was ever set up to do that (according to these audio engineers, including the late Charlie Hansen) because it had already been patented by Ed Meitner, then working at Museatex (but not otherwise used, to my knowledge, and that company no longer exists either IIRC, so it appears the patent did nothing but prevent application of this idea where it would have been most useful).

https://www.head-fi.org/threads/hdcd-list.65414

(Ed Meitner appeared previously in my audio experience in the early 1980's.  His amps were being demonstrated at a meeting of the San Diego Audio Society at the same time as the Apogee 1 ohm speakers.  Someone hooked them up without authorization, and the Meitner amps fried.  Someone who might have been Ed Meitner or his rep scolded someone for hooking up something wrong, I never got the details.  We were supposed to have been using the Krells on the 1 ohm Apogees.)

Here is the definitive HydrogenAudio (Objectophiles) table of which discs use which HDCD features.

I retained hope for several years now that this story about HDCD was wrong.  HDCD's just sounded so much magically better than other discs.  I was sure it wasn't just the "peak expansion."  I truly believed in usefulness of the variable Transient Filter.

And they sounded best of all on the Denon DVD-9000 player, which is the only player that has proven to me it does the "peak unlimiting" thing as PMI would have wanted.  It puts out more than just 2 volts, I determined, reaching nearly 3.8v peak on some HDCD's.

Other players seem to implement Peak Unlimiting by lowering the average level, and only having peaks reach the full output level, which for most disc players is nominally 2V.

Back in the day the DVD-9000 was being sold, a friend of mine complained (either about that exact player, or perhaps another) that HDCD was a gimmick where they raised the level twice as high.  Well, he said, I can just raise the level twice as high on my preamp.  It's not fair!

It seemed like ultimately other manufacturers agreed, and that while some earlier HDCD players may have raised the actual output level to implement Peak Expand, others didn't, they lowered the average level to make Peak Expand possible.

So few other HDCD players do the Peak Expansion in the way that the Denon DVD-9000 does.

Anyway, as I became more and more aware of the stories regarding the Transient Filter flag not actually being used, I was still convinced it was important, because otherwise how could the Fresh Aire HDCD discs from Mannheim Steamroller sound so specially good when played through the Denon DVD-9000.

So good I was convinced that resampling through the DVD-9000 was better than playing the original CD bit stream (which needed no other decoding than the Transient Flag anyway).  I was beginning to believe there was just something special about this high priced and limited production player.

The DVD-9000 literature made note of the "new generation" HDCD decoder, along with the new AL 24 Plus digital filters.  IT DOES NOT HAVE A PMI 100 DECODER, as such.  It is the first Denon that supports HDCD that does not actually have the PMI chip.  Denon paid for those things when they were still being made.  The Denon 9000 included DVD-Audio, but NOT SACD.  It was Denon's last chance to display technologies alternative to SACD.  The Denon manual, IIRC, notes licensing to Museatex but doesn't say why.

After just a few months, the DVD-9000 was discontinued, and replaced (for a while) by the 5900, which uses sigma delta chips and supports SACD.  Sony would not apparently license DSD to Denon for the DVD-9000 which used the very different Burr Brown 1704 resistor based chips.  This also may have affected what Denon was able to do for HDCD.

So it could be, I have long wondered, that Charlie Hanson and others who claim the switchable digital filter thing was never implemented were simply unaware of a small number of exceptions, if they exist, which might have included the Denon DVD-9000.

Not knowing people who would know, another way is to try it, as I've long wanted to try.

So I started recording a series of HDCD's that I know use the transient flag, but no others, on the DVD-9000 through the XSP-1 and directly into the DA-3000.  These are the Fresh Aire series on HDCD from Fresh Aire 1 through Fresh Aire 8.  They all use only the Transient Filter flag from HDCD and no others.  (According to Charlie Hanson, these shouldn't be considered HDCD as they don't require HDCD decoding.)

The Fresh Aire recordings are not even included on the standard list of HDCD recordings, but they say HDCD on their boxes and were sold as such (by Audio Advisor at one point).  They light the HDCD light if a player has it.  But they do nothing but transient filter decoding which I reconfirmed with a little ffmpeg script I still have called hdcd-check:

rm junk.flac 2>/dev/null

for name in "$@"

do

    ffmpeg -i "$name" -af hdcd junk.flac

    rm junk.flac 2>/dev/null

done

At the bottom of the output you get lines like this for all the Fresh Aire discs:

[Parsed_hdcd_0 @ 0x7fdd67e02000] HDCD detected: yes, peak_extend: never enabled, max_gain_adj: 0.0 dB, transient_filter: detected, detectable errors: 0

Not knowing the correct level, and also believing the 9000 might benefit from some warm up time, I played several different Fresh Aire discs at increasing levels, until I zeroed in on the +0.5dB level as being correct for these, causing the highest non-clipping level.  (If theses discs had used Peak Unlimiting, it would be something like 2.5dB lower.)  I ultimately ended with my best recording of Fresh Aire 8, which reached a maximum level of -0.114dB with the Emotiva gain at +0.5dB.  I saved this exactly as recorded AND as releveled (by Audacity) to -0.02dB for audible comparison with the original digital, which has peak level of 0dB (Audacity won't Amplify because it is already at 0dB).

Once again, as I have matched the levels ever closer, I found the differences if any increasingly hard to detect.  The non-decoded HDCD played direct from the digital sounds basically the same as the output of the Denon DVD-9000 with "HDCD decoding" being resampled on the DA-3000.

While I was doing this, I still thought how much more wonderful the DVD-9000 decoding sounded, but now that I have both the original and the decoded copy I can say they are basically impossible to tell apart, so far.

Strangely, however, Roon finds a difference and applies different levels of automatic leveling (which I do on an album basis).  It finds my HDCD version to require 0dB level change, but -2.7dB for the original digital.

This is probably an artifact of the re-recording somehow, but I'll have to figure it out before reading conclusions about the audibility.  Unless I can control the actual playback level and ensure all versions are dBa or dBc matched to within 0.1dB, no conclusions can be made.  The peak level matching I've done apparently works differently.  Perhaps it's because my resampling at 96kHz realizes some of the "intersample overs" which are hidden in the original digital.  And also in ways like that the Lavry might be different too.  I'm finding a difference between the phono-to-CD level change using the Lavry and using the DA-3000.  For some reason, the two digitizers show different relative phono to CD peaks.

Bu at this point, I'm also closer to completely believing Charlie Hanson in these regards.  Even though the level matching doesn't seem to match, there is no special magic that the HDCD transient filter decoding seems to add. But I really want to test some of the earlier recordings too.

Charlie Hanson loved HDCD nonetheless because the original encoding converters (PMI One and Two) were so good, and because he believe even though the dynamic range improvement is little better than one bit at most, it's still important.

Like me, Charlie Hanson also hated DSD and MQA.  But unlike me, he was a believer in some flavor of linear phase reconstruction filter (his own) instead of straightforward minimum phase.  His flavor added significant ultrasonic noise, though not as much as NOS or DSD.

I don't think there is much need for amplifiers with no feedback, which was ultimately his motto.

So in spite of his acclaim, I still consider him sort of tweaky, and perhaps not fully to be trusted for that reason, but perhaps sometimes right anyway, and less connected to the big wheels than many.

An insider/outsider or something like that.  So I sort of like him a lot anyway (while thinking he may have been wrong in several key areas).

I thought the idea and design of Pono were not a bad idea, and often considered buying one, but I'd have had to rig it into my system which means basically resampling...  And I'm not interested in headphones.  And they never got that much support from the music industry which of course really killed it.

I would have been interested in a high resolution front end device with AES connections.  Ultimately I got Roon on my Oppo, at least that has coax digital, and through that I get Qobuz for high resolution which it has in some things.

It sounds like Charlie had a very interesting life and career in audio.  And he was a healthy active guy and everything.  But he died from the complications of a bicycle accident in 2006 after suffering from it increasingly for 11 years.

You can't tell any suffering from his online persona.  He seemed to be bright and cheerfully talking about what HDCD does and doesn't do (even that, something barely connected to his business) up to his last days.  I had been reading the HDCD column and was shocked when it was written that he had passed away.

It would be very funny in a way (for me) if the Transient Filter flag actually does nothing.  Because I believed in it so long.

It shows the power of belief.  Belief can make things special and magical, when they are just ordinary or nothing at all.

I'm suspicious this happens a lot to other audiophiles, but not me.  For one thing, I don't really believe in much of anything anyway, anything outside of mainstream audio engineering, in which audio reproduction is fairly well described by its lack of distortion and frequency response errors.

So when it comes to other things, things that Objectophiles wouldn't believe are audible, I have mostly if not entirely disbelief.

This was not at all true for HDCD.  I really believed in it, and that the Transient Filter was a kind of precursor to SACD that Sony wanted killed because it was nearly as effective and yet far simpler.  One filter for transients and one filter the rest of the time, I thought.  That way you could get the perfect transients that were a (usually unheralded) feature of SACD and DSD and NOS type filters, but just when they are needed so as not to be getting a lot of noise and grunge too.

That belief made HDCD's played on HDCD decoding players always specially magic to me.  I believe I wrote about it here.

That's a huge upside perhaps from mere "belief" if it is a false belief, as it now appears.

The downside is that I was afraid to play HDCD material any other way, including on my Roon/Oppo based system which I use to listen to most music now.

HDCD's were a special category that I didn't want to listen to unless properly decoded.

So I spent hours, and more hours, always making sure that in one way or another I could properly decode HDCD, transient filter operation and all.  I was convinced that software converters were inadequate because few could actually model the effect of different reconstruction filters.

And I avoided listening to my HDCD's in any other way.  If the DVD-9000 was not connected as a digital converter (which is often true because the extra Sonos ZP80 I use to send it digital is often removed as being one to many things when I need more room in the rack), I'd avoid listening to any HDCD's.

So while the power of belief can be magical, it can also produce fear, the well known audiophila nervosa.

Orfield Labs

 Orfield Labs owns the quietest anechoic chamber that exists, until supposedly Microsoft built a better one in 2015 but details about that are not public.

I remember comparing anechoic treatments tested at Orfield Labs.  A place to go for that sort of thing.

Also, auditory deprivation testing.

https://www.nytimes.com/2022/11/23/magazine/quiet-chamber-minneapolis.html

Adcom 535 MkII

I picked up this little beauty at an audio swap meet.  The original Adcom 535 was designed by Nelson Pass.  The MkII version added triple darlington outputs (AFAIK a plus) and nice double banana binding post outputs (replacing a proprietary speaker connector).  This is the 535 to own!

But why does one need an Adcom at all?  Well, I wanted to check out the early Pass MOSFET designs.  For that purpose, this is a failure.  This is not a MOSFET amplifier.  Instead, the 535 is one of the last commercial bipolar amps designed by Pass.

This is pretty close to Nelson's ultimate "simplicity" thing.  The Adcom bipolar amps are about as simple as a bipolar amp can get and still be objectively good.  To get still simpler, you need MOSFETS, which came in the next generation of Pass designs for Adcom, the ones with two zeroes.

Now, they do have an IC implemented servo (fine) and also a capacitor input (belt and suspenders?) using a polycarbonate cap (about the best you're going to get at this price point).  A large polypropylene would be better but cost a significant fraction of everything else.  Most amps with capacitors use electrolytics, which are far worse electronically than polycarbonate.

John Curl also designed his first Parasounds with IC servos and got (probably undeserved) flak from Stereophile about it.  David Rich wrote an article about this story.  So Curl redesigned the 2200 using a discrete servo circuit.  Adcom wasn't scolded in this way and kept their original IC servo design which is fine.  (I'll have to check if my HCA-1500A has IC servo.)

I measure 87 Watts into 8 ohms, which I think is about spec.  Clipping is calm.  Response is down 1.65dB at 100 Khz, the 3dB down point is 125kHz (this is traditionally called the "bandwidth") and the 6dB down point is 192kHz.

Power is short of what I need for Acoustats, but possibly sufficient for Revels and my musical instrument speaker.

But what this looks best for is a replacement or backup for my supertweeter amplifier.

For the supertweeters, I'm currently using an ATI 1502, which I'd long planned to replace with another Pass design, the F-5 Class A, but who knows when I'll get around to that.  The ATI is curiously bandwidth shy for a supertweeter amp, rolling off at 60kHz IIRC.  But otherwise it was nicely quiet and had low power consumption which was another plus.  The Adcom seems to have about twice the high frequency cutoff, and I don't need the extra power the ATI has (150W) for the tweeters.  The Adcom also seems lower in power consumption than many amps I have (despite claims of "high bias").

It will be tested on Acoustats however, for the record.

But possibly not without bias adjustment.  At 10V output I measure 0.026% THD+N.  That's perfectly fine, though my much older and higher power Dynaco 410 measures better at 0.014%.

But at lower levels, the distortion on the 525 II explodes.  At 1V output, I measure 0.28% THD+N, more than ten times higher.  At 0.3V output it's 1-2% THD.

[See update below.  It's apparently ripple.]

It's not from noise.  The A weighted noise is a mere 22uV, less than half what the Dynaco has.  The 80kHz filter makes hardly any difference (so I left it out).  Strangely, though, the 400 Hz filter made a significant but not decisive difference.

This indicates crossover notch.  The actual crossover notch recorded on the difference output is a very hashy and asymmetrical distortion..  There is none of that at all in my residual.

If my untouched unit has a bias error, what about others?  This amplifier has a reputation for "hard sound."  Well that's exactly what crossover notch is going to do.

I will be checking the bias to see what's going on.

It might be interesting to see if I can hear the distortion too.  Not sure if I want to bother.

Well, sadly there is no room for improvement by increasing the bias.  For one thing, the bias was exactly at spec already (or slightly high in the right channel at 8.5mV instead of the 7 mV specified level).  Both channels measured almost exactly the same in THD+N.  I tried increasing the bias to 20mV, and it made no difference.  Distortion at 1V output still measured 0.26%, higher than I think what a good amplifier should have.

Furthermore it seems to be an awful hashy looking distortion associated with the low voltage peaks.  I'm still assuming this is some kind of weird crossover notch distortion (but maybe not?).  It is not some innocuous looking 2nd harmonic.  Nor is it some ultrasonic ringing.  Applying the 80kHz filter had almost no effect.  Strangely, applying the 400 Hz filter cut the distortion in half.

I still have considerable concern this is due to my flakey analyzer.  (But it measures 0.003% distortion in bypass.)  I'm thinking of bring the Dynaco back to the bench and re-affirming that it measured amazingly low distortion at 1V and lower.  I might also try different analyzers, better grounding, etc.

I am also seeing similar low level distortion which more clearly looks like crossover notch with the Nikko Alpha III that I have also tested recently.  I believe I deliberately lowered the bias on that amplifier and am planning to check if restoring the bias fixes it.  That was the amplifier that the staff at Audio Dimensions once preferred to all other amplifiers.  It was one of the first commercial MOSFET amplifiers.  But the fact that two amplifiers have shown a weird problem does raise the possibility my test setup is flawed.

The Adcom service manual says that before setting the bias, the amplifier should be heated up by playing 20W for 20 minutes with the cover on.

Starting with the quiescent bias at 8mV, when I ran the amp with cover on for 20 min at 20W  (I actually turned the cover upside down and blocked the sides, because removing the Adcom cover is very difficult, it requires removing all the rear connections first), the bias had at first fallen to 3.66mV as soon as I could measure it.  As the heat sinks cooled off, it rose to 5 mV in 2 minutes, 7 mV in 9 minutes, and more slowly after that finally back to 7.5mV in about 20 minutes.  The Adcom manual says to set the bias when it has "stabilized."  But what is stabilized?  Does that mean as soon as the meter can be read clearly at the highest heatsink temperature?  Or does it mean waiting 20 or more minutes until the heatsinks have cooled down to quiescent level--and in that case why bother to heat them up in the first place?  Or somewhere in between, such as around 9 minutes where's it's barely changing but still slowly falling?

I asked this question on DIYAudio and Nelson Pass himself replied.  He said he designed this amplifier, like most others, according to the 10 second rule.  So the heatsinks heat up to the point where you can keep your hands on them comfortably for 10 seconds (and no more, but he didn't say that in this message).

I was jazzed that Nelson Pass himself replied to my query!

He didn't actually answer my question (and perhaps he didn't know the answer) but what he did give me was a license to experiment!  In fact, any tweak could adjust the bias (it's the big obvious pot on the board) without even using a voltmeter, just hands.  And in fact what they might well do is crank it all the way up because it takes a long time even then just to get warm.

I figured that would probably be OK in an amplifier designed by Nelson Pass and in fact it seemed to be in this case.  I cranked the bias all the way up and at first it measured 19mV at the test points.  Then, with the cover "on" (faked as before) I let it run for 90 minutes (actually, I rechecked a few times to be sure it wasn't getting out of hand).  At 90 minutes the bias voltage had fallen to about 14.5mV.  The heatsinks were now as warm as I could touch with a flat hand for 10 seconds, and no longer.  If you wanted maximum performance this is probably how you would set it.  I might turn it down just a bit for better longevity.   (Anyway, see the update below, I not going to sweat the exact bias setting now...  For now I turned the bias back to 8mV quiescent matching the other channel for now in case I never get back to this project again, as has happened, ouch, so many times before.)

With hours of operation at full on bias (but no signal) in my simulated "cover on" condition, the (unreliable) temp I measured with my IR thermometer was 130F, same temperature as I set my Aragon 8008 BB to.

(I think I adjusted my IR thermometer for heat sinks somehow...at least on the older equipment I look at the numbers seem reasonable and reliable.  Nowadays it works horribly on everything else.  Or maybe it's not actually working well on anything.  I have the feeling that older heatsinks get dusty enough that the high reflectivity of aluminum is not so much a factor anymore.)

In cover off 20 minute testing, 7mV quiescent yielded 93F, 8.5mV yielded 100F.

UPDATE

Finally I am understanding what my Rigol scope is showing me about the distortion output from my ST 1700B analyzer.  The hashy looking "distortion" appears to be related to the power supply at 120 Hz or maybe 180 Hz.  It stays at that same frequency regardless of the frequency regardless of the frequency of the signal the Adcom is amplifying.  So it appears what I thought was "rising low level distortion" is actually power supply ripple.  It apparently does not affect the output much when there is no signal, so I got the very low 22uV noise measurement.  But when the amp starts amplifying a signal, the ripple appears and stays more or less at the same level regardless of output, being swamped by higher outputs.

Meanwhile, the actual distortion, represented by the small wiggles, is actually very small at 1V and below.  This is not a bias problem, it's a power supply problem.  I've ordered two new Vishay 6800 uF capacitors which are rated at 10,000 hours.  These super premium parts appeared to be the best I could get at Mouser to fit the space in the Adcom.  Most filter capacitors are rated only for 3000 hours, which is typical of standard grade electrolytics.  3000 hours was the best I could find now in Rubycon, the brand of the original capacitors in the Adcom, and even they required special order.  (BTW, that means 3000 hours of torture at max temperature and current, so typically it results in 5-10 years of actual use in nice circuits.  10,000 hour capacitors should last 20-30 years in nice circuits.)

Now between the amplifier and just two replacement capacitors I've spent all the money I made selling my Denon DVD player, but I'm getting practice refurbing amplifiers, which is something I have to look forward to for many other amplifiers more important to me now at some point in the future.

After replacing the caps and doing whatever else is needed to fix the apparent power supply problem, I can measure for sure how much the distortion actually changes at low levels as the bias is increased.  Perhaps, in the end, there will be little reason to keep it higher than "7 mV".

Uhoh. DEQ failed

 Paradise had arrived.  With my chairside Behringer 2496 DEQ I could instantly flip from Background to Listening Position optimized EQ's.  I also memorized the typical level so in most cases I didn't even have to re-adjust the level, which was also fairly convenient with the chairside DEQ.

But then work on my house led to power glitches and a restart of the UPS which powers most DEQ's.  (I think I had plugged the chairside EQ straight into house power, which would have been a mistake.)  The chairside DEQ had a sudden death (flashing screen) before I remembered to even take a picture of the new EQ settings.

I created a thread on DIYAudio and quickly figured out there is a faulty C17 capacitor in the power supply, and the "big capacitor" looks iffy so I plan to replace that too.  The biggest issue for me is it's ROHS lead free solder, and I don't have any and have never worked with it before.  My usual methods with leaded solder didn't permit me to remove C17 as I was expecting.  I bought new solder, a new powered solder sucker,  I will need to practice with the lead free solder first so I don't mess up the Behringer.  Oh I forgot I need to get a new tip for my Hakko iron too, it's better to have a different tip for leaded and unleaded.  Some say you should just stick with leaded, crank it up to deal with the lead free solder.

If I mess up that one, I'll have to borrow and fix another PS from another unit maybe, so I can "recover" the magic lost settings.

Meanwhile, I've programmed in my best conservative approximation of what I had before in the midrange and bass DEQ's.

For the Bass, the "No Remote Background" setting has -6dB for 20Hz, and +4.5 for 25 and 32 Hz.  The "No Remote Listening" setting has 0dB for 20Hz and +6.5dB for 25 and 32 Hz.  This is a pretty good approximation of what I had before except there was some additional boosting higher.  Judging by RTA now, that's tricky, so I might wait until I recover the lost settings in my old DEQ.  (According to an entry below, I had boosted about 2dB from 25-80 Hz...but that was layered on top of an existing pattern of boosts and cuts in that range I'd worked out between July and October.)

For the midrange, I had some penultimate settings dialed in already as J3 in the midrange DEQ.  The midrange boost was already permanently dialed in through a PEQ setting, but I had also done some fine tuning of the midbass and lower midrange.  Sometime around then I decided to "move" all boost settings into the chairside EQ, so in the J3 settings these fine tunings are bypassed in the bypass menu.  To restore them into the new No Remote settings, I simply removed the bypassing.  I most likely did some further fine tuning in the chairside EQ after July 3, but it may take some time to figure out what works and I may just wait until the old settings are recovered.  At least the July 3 GEQ settings seem to measure and sound pretty well.  In fact I had several hours of Audio ecstasy on Sunday as my mplay playlist generator running the Albums script had selected Cowboy Junkies, and Rebecca Pigeon albums which I heard as never before.

Sadly mplay was unable to make another Albums playlist because the altTunes folder it uses was already played out and needed to be reset, and it appears modifications in the mplay program in the last few months caused the reset to stop working again.  So I furiously debugged that on Monday morning and got it fixed.  Meanwhile I've been listing to background music using the Music script (which really means Background Music...and it's also albums, just music with no words), which had been my usual until a week or two ago.

I'm creating an all new program tplay to reset the mplay history whenever playlists are not played to the very end.  That has been a big problem for me with picture and movie playlists, not so much music playlists as Roon conveniently halts streaming when the Oppo is turned off.  But it's not a non-problem for audio either, because Roon inconveniently doesn't remember where it was when you go back to a playlist when you've stopped it temporarily to play something else.

I also have been working on splay to shuffle playlists together in various ways, and have added a -new option to mplay to select files only from the last specified number of days.  With these commands, a a playlist of all new files can be shuffled together with a playlist of pre-existing files.

Why 96kHz PCM is "sufficient"

Objectivist audiophiles will point to the lack of double blind tests showing that frequencies above 20kHz are audible except perhaps to the youngest people.  For most of my life, I've had a virtual cutoff just above 16kHz.

There are ironclad digital theories, that basically say PCM is better than intuitionists might think.  A PCM encoded system (with properly designed filters, etc) cannot be distinguished, technically, from a real one with similar bandwidth and through that bandwidth.  So, yes, one can show the various imperfections of PCM using 100Mhz oscilloscopes, but not with comparably bandwidth limited system--they just don't show up within a comparably limited bandwidth.  You don't "hear" the pre-ringing, etc, because it's way out there.

Now, there has long been a particular claim that humans can detect leading edge differences that resemble frequencies higher than 20kHz.  And these claimed detections go to the equivalent of about 40kHz.

There have been some "brainwave" studies that show correlations with frequencies above 20kHz and up to 40kHz.  These are controversial and have not been replicated elsewhere as far as I know.

In any case, it looks like the most we could possibly need is about 40kHz.  And at that point, not may speakers will work either.  I strive to achieve at least some output there with supertweeters.

So it looks like 96kHz, with it's bandwidth of about 48kHz, is more than we need, even according to the most sensitive and potentially false-positive studies, from brainwaves.

"Listening Position" EQ

While doing the disc player tests using Santana's Supernatural much of the listening I did was semi-serious.  I was paying attention, but not sitting in the listening position.

So one good thing that might have come out of these tests (along with showing the Oppo BDP-205 is technically superior and any advantages other players might have had seem to disappear when the level matching is perfected) is that I was reminded that I still haven't solved the bass problem very well.

The basic problem is that the listening position is in a uniquely bad position for bass, and I basically can't change that because my Living Room is a multipurpose room, not a dedicated listening room.  In a dedicated listening room I could put the speakers in the center of the room, and then listen from the back.  That would be the biggest imaginable step for solving the bass problem.  That's what Alan, a local audiophile and retired audio engineer, recommended and did in his own system using Magnepans, until his unfortunate passing a few years back.

Also I could move the listening position more to the back of the room.  Even just a few feet back, not all the way back to the loveseat on the far wall from the speakers, would help.  That's where I used to have the listening position until I discovered the power of close up stereo.  Nothing gives the you-are-there feeling like sitting as close as possible to the stereo speakers for the widest possible stereo image and something like stereo-envelopment.

I tried doing that for a few seconds this week, and whatever improvement there might be in the bass, it simply isn't worth losing the you-are-there feeling for it.  Once you've heard the benefits of close up stereo, you don't want to go back to back-of-the-room listening.

In principle I could cover the entire back wall with bass traps, which is about what it would take to meaningfully change the intense room modes.  But I don't have spare space and even if I did I wouldn't have the money.

My pipe dream that I could cancel room modes using FIR remains a pipe dream, and may stay that way.  You can't dynamically shift phase around because that's the same as shifting time back and forth and is known as wow and flutter.

So I'm back to doing more EQ fiddling because it's the only thing I can do.

The chairside digital eq is one of my bigger improvements this year, I still think.  That's what I used earlier to do a small but very useful tweak to the midrange of my system.  And from there I dialed in some mostly subjective but partly objective bass boosts (and a couple extra cuts too).  There is also a dedicated EQ unit for the subwoofer channels, but that's mainly dedicated to cancelling room modes, and even minor changes seem to mess that function up.

And I've long thought about doing a dedicated chairside bass EQ.  Some have wondered why I didn't just optimize the EQ for the listening spot in the first place.  Actually, I have done that, but also very conservatively, because anything but conservative boosts make the stereo unuseable for background listening, which is what I'm doing 95% of the time.

So on Wednesday I tried some fiddling.  It seemed I could add about 2dB boost between 25Hz and 80Hz on the chairside EQ and it added some nice bass "punch" that I'd been hearing everywhere EXCEPT at the listening position with previous adjustments.

Whenever I added boost above 80 Hz, the sound became dull and I immediately wanted to reverse that change.

Also, I removed the 6dB 20Hz cut.  My whole house has a 20Hz resonance, but who cares when you're at the listening position where it self-cancels?

Right now as I'm writing this, I'm playing Royal Scam in the living room and writing in the kitchen.  The bass is definitely over the top, but tolerable on this album.

I saved the new EQ, which was layered on top of more variable boost EQ I dialed in earlier this year, as the new LISTENING eq setting in the chairside EQ.  I plan to have BACKGROUND and MOVIE eq's as well.  The MOVIE EQ would keep the bass in the entire room from getting too excessive, when I'm showing movies for friends.  The periphery of the room has bass about 10dB louder than the listening position as things were already...so my previous ad hoc but obviously inaccurate approach has been to turn down the subs by 10dB at the Dac.

I've thought about controlling the EQ with midi signals, and then even generating the midi from sensors that detect when I'm sitting in the listening chair and perhaps even when the listening chair has been moved out of the way for movies.

It sounds far out, but not unlike what I did in the last two years with stereo automation.  And, it's still easier than trying to deal with the fundamental room problems acoustically, which would probably require making the living room larger and filling the back with bass traps, at vast expense.

I appreciate how James Bongiorno refused to build a preamp without tone controls, then you can deal with all such issues quickly on an ad-hoc basis.  My ultimate dream chairside control unit would have screens that could emulate all sorts of tone controls, EQ sliders, or other classic control systems for easy ad hoc adjustments.  The tone controls would have a window showing the control types (Baxandall, Bongiorno, Holman, etc) and a control for adjusting the turnover frequencies, with the main control for changing adjustment level with full channel matching).  Old analog preamps often had trouble with channel matching, indeed rendering such controls far less useful or even destructive.  But now it can be done perfectly.

But I think ad hoc adjustment is less needed when the specific cases I've mentioned are dealt with.