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
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".
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.
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.
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.
I disbelieve in audible differences among well designed (non tweak) digital players and converters. All the various tweaks either do nothing or make them worse, I think.
However, that being said, I've long felt I heard differences, and in at least one case I want to know more.
Back in 2008 when I first started playing DVD-Audio discs on my first high end living room system since 1992, I knew the digital from DVD-Audio discs was limited by design to 16bit and 48kHz at the digital audio outputs of all players.
But I figured I could capture the full 24 bit resolution at the analog outputs by sampling it to 24 bit digital with a pretty nice A to D converter (the Lavry AD10, which I bought in 2009, replacing the theoretically similar but poorly regarded A to D converter inside my Tact RCS 2.0 preamp). Since I had started doing my crossovers, EQ, and everything in digital, I needed to sample all analog sources including vinyl records and FM radio to digital, so why not High Resolution DVD-Audio discs as well?
You could make various arguments against this. You might perhaps say that the noise level is increased by this resampling. It might even be increased to worse than 16-bit levels, around 98dB. But in fact my players, my line level preamp, and my converter are all rated at 117dB signal to noise ratio. So if each stage reduced that by 3dB, you'd still have 108dB, which is more than the maximum from 16 bits.
And what fascinated me since first thinking about it, is that in some fundamental sense the resolution is being preserved, even as a little noise is added.
Nowadays a lot of Audio/Videophiles would find this hard to believe. Virtually any form of copying (if it's even possible) makes a noticeable difference in video. Even just de-compressing and re-compression into the usual MP4 or other compressed video formats. But video is far different from audio. For one thing...the lossy encoded formats inherently lose real resolution on each re-enconding. You can easily understand this when you see that many different inputs could cause the same output--which is the very essence of lossy compression. And there are other reasons relating to the nature of scan lines and things like that.
Analog interphase copying or transmission of audio can be damned good. And it's weirdly reversed from video compression. Instead of many inputs potentially causing one output...now we have one input potentially causing infinite outputs (just being offset by different tiny amounts of time, noise, or other analog "errors").
So, I started doing this just to get the full resolution of DVD-Audio, or at least as much as my 24/96 Lavry AD10 can capture. (I continue to believe that 24/96 is as good as any human needs, and that there's no need for more "time coherent" digital systems like DSD which does get reasonably good when you get to high enough sampling rates like 128x, which is nice, but it's just a waste.)
But then I noticed that even CD's sounded better this way. In fact, it was beginning to seem like all CD's sounded better being played by the Denon 5900 and resampled into 24/96 by the Lavry sounded better than straight digital from the Denon 5900 or any other player.
And resampled in this way, you can also hear the differences among CD players, etc. (If there are any.) And so then I started collecting a bunch of different disc players that seemed like they were optimal for different kinds of media. Like the Pioneer PD-75 for CD's, the Sony DVDP-9000ES for SACD's (said to have essentially the same true 1-bit converters as the SCD-1), and the Denon DVD-9000 for HDCD's.
Now, I got called out on this practice by someone who pointed out that I could indeed capture DVD-Audio digital from the HDMI with a de-embedder. And he was right, I could do that with nearly all DVD-Audio discs.
I did do some tests and decided then that indeed the direct digital was better than re-sampled from the analog outputs.
And while I still kept my battery of different players initially, over time they got displaced by other bits of equipment I wanted to consolidate in my audio rack. Only the Oppo BDP-205 and Denon DVD-9000 remain. Also the laser in the Sony DVDP-9000ES died and needs to be replaced again (something that curiously happens to that player about every 3 months in heavy use, or longer if lower use, I think Sony modified the SACD copy protection requirements after that and the earlier SACD players had much bigger laser assemblies that weren't bothered as much by the heat or something).
So now I was planning to sell the PD-75 but decided not to, somehow the digital played through it (and other players???) seems to have more passion, character, or something, than even the straight digital being streamed by the Oppo into coax.
And all along, I've noticed that nothing at all sounds like the Santana Supernatural DVD-Audio played on the Denon 5900 into analog and resampled into 24/96. The straight digital over HDMI lacks bass, sounds dry, and has far less passion.
I noticed that the Denon 3910 seems to sound much like the Denon 5900 and possible...even better than the Oppo BDP-205.
All this is complicated by level differences. It remains possible (and in fact objectivist audiophiles would predict) that once I assiduously match the levels, everything that has at least 16 bit resolution should sound identical.
I usually haven't done deliberate level matching (only new listening) at all. When I've done it, it's been pretty ad hoc. If I want to systematically compare a bunch of players and transmission methods, I ought to make this systemaic.
And so I begin by laying out the parameters for this. One can NEVER resample the analog back into digital and retain exactly the same peak level. One must ALWAYS allow a little extra headroom in the AD converter. The way I usually do this is by discovering the level (in 0.5dB increments) on the Emotiva preamp that lites the second to the last red light on the Lavry, but not the last light, so that clipping has never occurred. This is obviously disk dependant, so there's a certain amount of guess and try that must go on.
What I ought to do is play Supernatural on all the players that can play it, and find the maximum level they can play it without clipping.
Then after that, there has to be an additional adjustment to raise the level further to match the direct digital. If a digital recording is being made (as I intend to do for resamplings of Supernatural on every machine which will play it) the final adjustment can be done in Audacity. To to make the comparisons fair, every recording should be made at the just-below-clipping level. (I've never bothered with that nicety before.)
In my previous experience, I've heard maddening differences between the Denon 5900 and the Integra RDV-1. I figured the RDV-1 would be better but in fact only the 5900 had the special magic that made Santana sound wonderful.
Now you might say, if I'm going to be comparing the players after having been recorded to 24/96 digital and then being streamed all over...if streaming is "losing something" then it's still going to be "losing something" if I'm using it to compare renditions that were not streamed.
You might think that, but I've felt I've heard the differences between the players best in this exact scenario, where they've been pre-recorded and re-streamed on demand. It's great for comparing phono systems too.
I don't actually think the problem is that streaming is "losing something." I think certain players especially, like the Denon 5900, and perhaps other Denon players, may be "adding something." What they might be adding which is actually beneficial is something like upsampling. The player is converting to the "infinite resolution" of analog as best it can...and some players do this more nicely than others. The Denon players have AL24 which I've always suspected as being superior to what is used by others. Or at least it sounds sweeter, smother, more spacious, and slightly more bassy than other players in a particular way that just happens to enhance Santana Supernatural, and perhaps other things.
It may be nicer to have all digital upsampled to 24/96 in ways other than is done electronically by the ASRC's of the MiniDSP OpenDRC's that I use now. (That's where this job gets done now in my system.)
"Nicer" might even be not as electrically and mathematically accurate, as well. It could involve some kind of bandpass filtering, for example, that just happens to remove out of band garbage.
Supernatural played on the Oppo BDP-205 with all balanced connections requires a +1dB gain in the Emotiva for -1dB peak indication on the Lavry. A limited amount of listening while I was making the recording suggested that the HDMI digital and the resampled analog sound identical on the Oppo. The digital, limited to 16 bit and 48 kHz, sounds notably inferior. On the Denon 3910, the HDMI sounds identical to the Oppo HDMI. But the resampled analog sounds different, much as I described before, slightly more spacious, less grainy, and comfortingly powerful bass. It requires the same +1dB gain using an unbalanced connection to the Emotiva. All consistent with the idea that the magic is in the Denon AL24 digital filter, even though in this case it does no upconversion per se since the source is already 24 bit and 88 kHz. The blue AL24 light is lighting up letting you know that Denon is "improving" the sound somehow.
A confounding factor in a lot of ad hoc background listening impressions is the troublesome way there is slight bass cancellation at the listening position, and bass augmentation most everywhere else. Chairside bass boost control might be nice, perhaps automatically enabled and disabled by sitting in the listening chair.
I keep trying to think how one might fix this completely with FIR. I was thinking of modeling the bass source at the listening position. But this is not invertible, it seems to me. What might be is this: modeling a bass reduction in the far corner(s) of the room.
That might be changed by a frequency dependent phase shift similar to parametric EQ, with a notch phase shift at the mode frequency that excites the far corner, such that it cancels there but not at the listening position. Then, the parametric EQ that notches the mode out completely could be removed, restoring the full bass impact at the listening position.
****
Update: I've made full transcriptions of Supernatural on both Oppo BDP-95 and Denon 3910. I'm not at all sure I even hear a difference, or that it consistently favors the Denon.
Notably, when I crank up the level to +6dB on my kitchen preamp, playing in Audacity on my Mac, I can hear hum and a tad of noise in between tracks on the Denon 3910 transcription. At normal levels I don't clearly hear it at all, and I doubt it makes much difference. My SOP estimate is about -80dB from peak level. The Denon is sitting on my coffee table and I needed to run a 6 foot audio cable to reach it. I pulled out a genuine late series Radio Shack interconnect for this purpose, and it crosses power cables and such at close to 90 degree angles as the best I could do. The Denon is plugged into a different outlet strip as the Oppo (but the same ultimate power conditioner, the Panamax 1500). With more effort I could probably improve the hum issue here in obvious ways (but at considerable effort, like using the rack slot normally occupied by the Denon 9000 which is hard to move). I will try to better for my 5900 transcription. I could NOT make a Denon 9000 transcription because the 9000 I currently have in the rack is the one that only plays CD's and HDCD's, but which seemed to have the most magic sound of all. I'd need to replace it with the second Denon 9000 I have, and I'm not even sure if that will still play DVD-Audio's either.
Meanwhile the Oppo has a balanced connection to the Emotiva, which has a balanced connection to my AD converter, all with short cables. Cranking up to maximum on the Oppo transcription I hear nothing but silence.
So I know in at least one technical way, the Oppo transcription is better. But sometimes the Denon transcription sounds smoother and invites being cranked up in a way that the Oppo transcription does not.
It might be that "crank it up" impulse, when fulfilled, leads to the "magic" recollected moments.
When I think I hear this sort of difference in the Denon transcription, it seems mostly like a "simplification." With the Oppo, I can clearly hear how overdubbing is used extensively in this album. The more "detailed" rendition it provides exposes all the trickery. Meanwhile the Denon seems to bring things together, better expressing the musical intent without peeking behind the curtain. In particular, the Denon emphasizes the musical swells more than technical minutae.
But they are so close I'm very doubtful I could reliably hear a difference in double blind tests, without considerable "training," and perhaps even then. And also quite often the Oppo rendition is nothing short of spectacular, and I have little doubt it's "technically" better.
(A friend always looks to polarity differences as the culprit when things sound "too detailed." It was easy to check this in Audacity. It's visible from the initial waveforms the transcriptions have identical polarity. I never knew I could do such comparisons so easily. But I have no idea if both are "correct" or both are "incorrect," only that it probably doesn't matter much.)
Which is one thing I hate about my modified (for total transparency) QSC ABX box that I currently use only for amplifier comparison testing. It gives no feedback until you are finished all the trial rounds. I think everyone needs a training setup, where you get immediate feedback, and perhaps no score is even tabulated. I need to get up to speed on more recent ABX programs and apps.
For the purposes of ABX tests with the Dynaco 410 last week, I added a line level switch which could shut off the signal to the Dynaco. That enabled me to check my guesses right away without messing with the QSC tabulation. I identified the amplifier (Hafler 9300 vs Dynaco) correctly 4 out of 9 times.
I'm glad to have made these transcriptions for future listening and analysis. But I'm doubtful it's worth keeping the Denon 3910 simply to "improve" the sound of discs, if it even does that. I want to get comparable transcriptions from Denon 5900 and Denon 9000 as well, among others. I have long planned to keep the 5900 because of how magic it sometimes seemed in the past, but it hasn't been hooked up to my living room system in over 5 years now.
I ultimately decided to "amplify" every transcription almost to maximum on Audacity, leaving a mere 0.01dB of unused headroom to be sure no clipping is occurring. And that way they can be compared directly without further level matching. And likely they can be compared with a digital files version as well, if I ever get one (as I have been thinking to do).
It might also be interesting to do CD and SACD transcriptions from various players as well. But I'm going to wait until particular discs excite me enough to try.
To be clear, I think both the Oppo and Denon transcriptions are magic sounding on the Supernatural album, but in slightly different ways that could be only sighted listening bias.
It's possible the RDV-1 transcription I once made which sounded relatively dull had some issue that didn't really represent the best of the RDV-1. In particular it's very easy to mistakenly play the "Surround" tracks which the disc defaults to (regardless of player settings) instead of the dedicated Stereo tracks which sound far better.* I want to check that out too. The RDV-1 should have among the best sounding outputs, and I was using it for several years as my "DAC."
(*I put the disc in, and it soon starts playing the multichannel version, which plays very softly, about 12dB down, because on the stereo outputs I'm getting a mix-down. I have to select top menu on the remote to select the stereo tracks. Then, maddeningly, if the disc automatically repeats, as it tends to do on many players, the repeat again defaults to multichannel version. This sort of thing helped kill DVD-Audio. From the beginning SACD played according to pre-selected mode, but not DVD-Audio.)
*****
Now I discover to my dismay (and forgetfulness) that Roon is also adjusting playback levels. As I had only adjusted the levels of the middle section of the album to match (at 0.01dB peak), and since I have album-wise leveling selected in Roon, that might explain why the 205 was getting 5.1dB of reduction, whereas the Denon only 4.9dB. So I began tossing out the other album sections into my home folder and asking Roon to re-scan. That, most strangely of all, made matters most worst, with 4.3dB reduction for the Denon and 4.9dB now for the Oppo. So I put all the other tunes back into the Denon folder, hoping to bring it back to 4.9dB as well, and now it stubbornly insists on attenuating by 4.8dB, so a mismatch I've managed to reduce from 0.2dB to 0.1dB by radical measures. As the difference gets smaller, they seem more alike.
****
I continued to play around with moving the first and last sections out of the 205 and 3910 transcriptions of Supernatural until finally I got Roon to apply the same level matching to both album transcriptions. It seemed weirdly history dependent, but finally I got the job done by attenuating the first track section to -0.6dB peak for the 3910 version, and leaving this section entirely out of the 205 version. If it were up to me, I'd leave this track out of both transcriptions and that was my plan for awhile I don't like to play it at all. But the first track seems to drive the level matching more than any of the other tracks, and somehow Roon estimates the 205 sourced tracks as louder by about 0.5dB, even when I've adjusted their peaks to the exact same value. So fiddling with the first track let me get the level matching correct for the other tracks I really care about playing, and those tracks have peaks set to -0.01 dB for consistency.
This alone suggests there is something weirdly different between the 205 and the 3910 sourced versions, even as they are now sounding virtually identical after level matching and tricking Roon to play album sections 2 and 3 at the exact same attenuations for testing by leaving a specially doctored track one in only one of them.
(Some might wonder why I still put up with Roon with all it's weirdness. But I like Roon for a large number of reasons, particularly my ability to create playlists for it. And for automatically generated playlists of full albums, there is nothing better than the album-wise level matching option Roon provides. This makes it possible to enjoy one album after another. When there is no level matching one quickly becomes very fed up with automatic play. So level matching is something I need, I only perhaps wish I could switch it off on an individual album basis, when some albums are actually just "tests" of different transcriptions like I've been doing here. For test purposes in other words. But meanwhile I can see why Roon makes it hard to change this. It's generally the secret sauce that makes everything sound good. And audiophiles like me are prone to leave things in test conditions by accident and then foul everything up for awhile. Such as leaving speakers out-of-phase when one was testing that. I've done this sort of thing numerous times, often taking amazingly long to figure out anything was wrong. That's in large part to my mostly-background music listening. I only rarely sit in the "hot" seat, sometimes not for weeks when I've been especially busy. I consider it a feature, not a bug, that I can pretty well enjoy my living room stereo anywhere in the house. And often it's even somewhat cool the bass is exaggerated in most such locations. I only I wish I could revert some of that excess to the listening position.)
I had put my precious PD-75 up for sale. But listening tests immediately remind me that, like several other things in my inventory of stuff, I do not want to sell this CD player. It's magic, and I need to understand that magic.
I do not understand why it should sound so good. It could be in large part, it's cheating with about 1dB louder output than most CD players... That goes a long way btw to explaining the apparently more passionate and intense reproduction this player produces. But I'm not sure it's the only explanation. Anyway, it's a cool player to have. It was the very last thing to get bounced out of the living room system (which continues to have the Oppo BDP-205 and Denon DVD-9000, the latter for HDCD).
When I first read about Legato Link, I hated the idea. If it were only that easy (as described in the first slick Pioneer brochure). Also I sort of remembered some review claiming the high frequency distortion was high, as if a lot of aliases were leaking through. Like No Oversampling.
Well now I think it's just a proprietary name for Pioneer's first original high end digital reconstruction filter. Virtually all CD players have this (some early players and modern tweak players use exclusive analog filtering...or NO filtering) in one shape or anothers. Pioneer's seems to be especially good, that's all. Does that matter say if your ultimate system runs at 24/96 digital? Yes, the reconstruction up to 24 bits might well matter, this is somewhat a grey area in double blind research, though results are usually negative, many individual studies of new systems (eg MQA) have found otherwise.
As do their early 90's 1 bit chips, the highest end version of which only exist in this player and the even more over-the-top constructed PD-95, which was barely available in US in the non-transport-only version. So for people in the US, the PD-75 was it, at least in the Pioneer camp. I think Pioneer had Sony beat hands down sonically, and seemingly by magic, around the time of the PD-75 in spite of Sony's ever improving distortion specs. Then, strangely, Pioneer stopped making their high end one bit chips and moved on the fairly pedestrian level of R2R type DACs, the PCM64, which didn't really get good until around the PCM1702.
Was there some kind of patent dispute? Did Pioneer find that their high end one bit technology didn't scale well into consumer players? BTW, this is why I believe that Sony's ultimate 1-bit technology, used in the first 3 SACD players (SCD-1, SCD-777ES, and DVDP9000--and good luck keeping the latter working with annual laser changes) suddenly vanished, replaced by a total embrace of multibit sigma delta converter chips. My theory is the ultimate 1-bit technologies, while good in vastly overbuilt machines, simply didn't work well enough for cheap machines, and there was no good reason to keep making them. Also the off-the-shelf sigma delta chips were far cheaper.
So anyway this is quite collectible and would be hard to replace as prices increase. Pioneers 1-bit which just sounded better (according to many tweak audiophiles anyway) than anything Sony could come up with until SACD. The even more overbuilt PD-91 has the PCM64 converters, possibly a step down and the PD-95 with the 1-bit converters is in the solid platinum category not far from Esoteric UX-1, if you can find it.
And does it measure bad, with rising noise or distortion at high frequencies. From what I've seen so far the Pioneer measures about as good as can be, and with astonishing linearity and dynamic range. Distortion is right at the theoretical limit or close enough not to matter.
The disc clamping mechanism is super cool also. An all metal mechanism clamps the cd from the top down onto a metal rotating platter. If you are at all concerned about disc vibration, which it seems hardly anyone is anymore, this is almost the ultimate.CD mechanism, equalled only by the higher PD's 91 and 95, and the most high end Esoteric mechanisms, VRDX, like UX-1 which I'd long lusted for but have little justification for. the vast expense
Audiophile tweaks spent a fortune on tricks to supposedly improve CD playback. One very popular strategy was mats. My friend and brother-in-law George sold (and still sells, I believe) a magnetic damper which you put on top of CD's to reduce CD vibrations. I was skeptical that all CD players were engineered well enough for such additional weight and thickness, and in fact it does seem like it doesn't work at all with some players. And it's really just a tweak, not a fully engineered solution.
What's done in the Pioneer Stable Platter mechanisms isn't a trick. Holding the CD flat and free of vibration is on no account a bad idea, so why not do the job with a fully engineered solution as in the Pioneer in the first place?
Hardly any but the most high end players have bothered in some way to reduce disc vibration. Apparently cheap mechanisms can handle the vibrations well enough by other means, mostly optical and electronic. But those are compensatory, why not just do the job right in the first place? This is the ultimate "source matters" audiophile religion. It's mostly BS of course.
I remember being fascinated by the PD-75 the moment I first saw one at an audio show. George hated it. Of course, dampers didn't work with it, you had to use the pioneer-provided thin damper.
So that could have been the problem for Pioneer. The solution engineered to address tweak concerns didn't sell well enough in the tweak market filled with sellers of more tweaky tweaks.
But they sold well enough to keep the old Pioneer in business long past this era, until the 2009 dvr collapse engineered by others because Millenium Copyright Act and market rejection of HD DVD. Until then at least it was Pioneer who made some of coolest disc systems of each era which the PD-75 still exudes in a relevant way.
I was relieved to read that the PD- 75 also did well on rejection of defects on a famous test (which I have in my inventory never used). I had a problem disc that PD-75 wouldn't track which the Oppo did, so however good the PD-75 is the Oppo is even better at handling defects, as you'd expect given it's a top model made 25 years later.
The PD-75 however is notably quieter in operation, and I like that. You feel like it's pulling your CD into a bank vault.
My PD-75 is terrible at handling CD-R's and I wonder what kind of over-optimization causes that. However it's fine at handling hybrids, and its possible in some cases I might prefer the CD layer to the SACD layer (I've gotten that feeling listening to some discs, like Music for Organ, Brass, and Typany sounds which so much more exciting than I recall on various SACD players). How better to test that with a great sounding CD player rather than something which tries to do everything? (Well, both, actually, I guess I need).
But how many extra microscopes do I need in my collection? Most of the others can go, I think now.
One advantage of the PD-75 that's pretty obvious, at least compared with most other disc players the mechanism is very quiet. But that's rarely a problem among the better disc players nowadays unless they are playing DVD's or SACD, which spin at much higher speeds, then such spinning IS often audible, and undesirable.
That wouldn't give it any advantage over streaming, however sometimes it's mentally easier to put on a disc.
In the 41 years I've owned it, the Dynaco Stereo 410 amplifier has been perfectly reliable. I had it refurbed once in 2001 because I was thinking it caused hum, but in fact the hum was being caused by my analog crossover, as I discovered afterwards. The audio shop in Austin replaced all the small electrolytic capacitors on the two amplifier circuit boards. They did not change the very large 22,000 uF Sangamo Computer Grade capacitors saying that it was not necessary nor cost effective. I used the amplifier for only about 2 years after the refurb.
I bought this amplifier in 1982 or so for use with subwoofers. I used two McIntosh ML2's for subwoofers, powered by this amp. For this use I did not need the fan, so I disconnected it, and instead drilled holes in the back of the case for ventilation. This works ok so long as you don't need much power. I have reconnected the fan for test and sale purposes. It is fairly quiet at the low speed.
I also covered the top of the chassis with duct tape. This does somewhat reduce vibration either with or without the fan running, but hockey tape would have been better.
The amplifier switch is broken and the amplifier is always on when plugged in. It should be switched with an external 15A power strip. Inside the amp there is a line fuse and thermal cutouts mounted near the output transistors. There are also rail fuses. There are externally accessible speaker fuses but they have been bypassed and the fuse caps are missing. There is a sensor that kicks the fan into high speed if it gets too hot.
Other than the broken power switch, the amplifier works like new.
At full power, 200W into 8 ohms (40V RMS):
Rated at 0.25% distortion at any frequency 20-20kHz
Right channel THD + N
1000 Hz 0.014%
20 Hz 0.014%
20 kHz 0.35%
Left Channel THD + N
1000 Hz 0.03%
20 Hz 0.024%
20 kHz 0.3%
S/N, A weighted
Relative to Rated Power
L: 119.5 dB
R: 110.5 dB
Relative to 1 watt
L: 97.0dB
R: 88.0 dB
S/N relative to rated power is so good it's hard to believe. In case I miscalculated, the actual measurements were 120uV for the right channel and 42uV for the left channel, read from my A weighted Meguro meter.
Overall this is much better than I had expected for an amplifier of this epoch and age. This was a stripped down version of the Dynaco 400 designed by James Bongiorno, but without the Dynaguard and larger heatsinks (fan was optional on the 400 and mandatory on the 410). It shows the mark of his excellent circuit design. It was very conservatively specified at 0.25% distortion when the actual midband distortion (where it counts most) is below 0.03%, about ten times less. The S/N is comparable to some of the best amplifiers ever made.
Comparing to my reference amp from the 1990's, 20 years later, the Hafler 9300 designed with MOSFETS by Jim Strickland (who also designed the Acoustats) using his patented Trans Nova circuit, what I consider to be one of the best amplifiers and amplifier designs of all time (and too often ignored), the Dynaco 410 appears to be sonically identical, except for the chassis noise of the fan and transformer. (If you eliminate the fan, there is still notable transformer noise louder than most good amplifiers, which the fan tends to cover up.)
With two days of blind testing, the accuracy of my guessing in double blind tests between the Dynaco 410 and my reference amplifier stays around 50%. It seems I can't actually tell them apart.
Thus, the Dynaco 410 passes, if you can hide or live with the fan noise. In my kitchen I might not notice.
Here's an interesting interview of James Bongiorno. I think he speaks pretty well for himself...
https://www.tnt-audio.com/intervis/bongiorno_e.html
I have my own stories to tell about and related to Bongiorno.
I greatly respect James Bongiorno as one of the best and most original audio amplifier designers of the 1970's to his death, in good company with many others like that including Sid Smith, Nelson Pass, John Iverson, Dan D'Agostino, Bob Carver, and Matti Otala. For a fleeting moment or two, Bongiorno may even have been the very top. But sadly his associations and companies never lasted long.
I had the amazing luck to have met him in person 4 times over 30+ years as well as having had some chats on the TunerInfo group in the mid 2000's. He often complained about ego in audio design, but I can't think of any who quite reached his level in that regards. Usually somewhat prickly, he tended to disassemble my honest but limited respect as insincere and unworthy.
Two of my best high school friends worked at Great American Sound (GAS) after graduating high school. They worked at GAS with founder James Bongiorno and also in the post-James epoch. Though Chatsworth was just a few miles from my primary and secondary schooling home, I moved farther and farther away after graduating from high school, but I often or at least sometimes came to visit my friends.
One of these friends, who we always called Bro, was my idea of a geek's geek. Already in high school he'd built his own Black Box for tapping into telephone lines (but not a Blue Box, he was quick to point out). Then he built an amplifier copied from SAE's latest IIIC amplifier.* It was in a big open box where you could see all the parts including huge computer grade electrolytic capacitors. He didn't try to make things look "finished." Instead it seemed he revelled in exposing bent metal and wires, wires everywhere. He used a heavily modified AR turntable, ripped open in the back (tin snips perhaps) to mount an SME arm. Sometime later he used a GAS Sleeping Beauty cartridge, and a preamp based on the Thaedra (but in a small box open on the top and sides so you could see all the tangled wiring). We'd hang out in his bedroom with other friends and listen to stereo, frequently swapping positions for the "hot" seat. He had giant home made speakers including 18 inch woofers. Was anyone else in our high school at this level of audio coolness?
I always remembered it as an SAE copy. However, on one visit another friend once said, "It's actually a Dynakit but he gets a rise by telling people it's an SAE." Just checking now, I see the Dynaco 400 kit was introduced in 1972 and the SAE IIIC having similar design but improved specifications was introduced in 1973. We started hanging out listening to Bro's "SAE" in 1973. Bro insisted he changed it to be just like the SAE:
"So what did you do that makes it like an SAE?"
"I took out the Dynaguard."
"Leave it to Bro to take something out, and call it a feature."
"It's better that way."
"Why? Why couldn't you just turn the Dynaguard off?," I asked.
"Because then I took out the switch and relay too, so it's really direct coupled."
Then Bro showed the speaker wire soldered directly to the wire connecting the output transistors.
"That makes it much better," Bro said.
Bro's "SAE" did indeed always make a noticeable thump when he turned it on. When the thump wasn't there, you knew something was wrong. But many years later, my Dynaco 410 which is basically what Bro had, a Dynaco 400 without Dynaguard, it makes no thump at all. I think the tech in Austin that did the last refurb did a very fine job of adjusting the DC balance.
In a few years Bro had upgraded to an Ampzilla in similarly open chassis with no fan. All 3 of those amplifiers: the Dynaco 400, the SAE Mk XXXc, and the Ampzilla were designed by James Bongiorno, in the course of a few short years, to the accolades of many, including reviews by Bascom King in Audio Magazine who described the Ampzilla as the dawning of a new era in amplifiers.
After starting to work for GAS, Bro always had a TO-3 transistor on his 73 VW Beetle dashboard and liked to paraphrase from a Steely Dan song saying "a transistor and a large sum of money to spend." To which another friend said "You only wish you had the second part."
Then the first time I met Bongiorno was at one of the first GAS factories, in Van Nuys, California. I had been asking Bro if he could get a stepped attenuator for me. I really needed something like that for the experiments in auditory perception I was doing for my senior thesis in college. I was using headphones to present stimuli to test subjects, and normal stereo volume controls weren't tracking well enough to make them precise. Bro invited me over to the GAS factory to see what it was like. He also promised he'd make a small amplifier for me based on GAS Grandson from one of the boards in rework. I said that would be nice but wasn't as necessary as the attenuator. I had been using my Marantz 2270 as an amplifier for the experiments, and I believed it was good enough. (Bro said it wasn't high fidelity. I said "What?" He said "Maybe you wouldn't hear the difference anyway?" "What?" "See what I mean?" Bro wasn't just popping off, we had in fact done a listening test comparing his amp to a real SAE to my Marantz receiver, in Andy Hefley's apartment in Tarzana. Bro claimed to hear a lot of differences. The only difference I was sure of was that the bigger component amplifiers had better woofer control.)
But it would be nice to keep my 2270 in my dorm room, I admitted.
As I arrived, Bro announced my name as Charles. Bongiorno asked me, as "Charley," what I liked to listen to most, and I honestly said I liked FM radio. He then made an inside joke among his staff by asking, "But does FM radio like you?"
I didn't get the joke at all. I figured he was making fun of me because serious audiophiles didn't take FM stereo very seriously (but I had for a long time, and lusted over tuners such as the Marantz 20b).
Bro later explained to me that I wasn't being put down because of liking tuners. Bongiorno had been talking about making a Charlie tuner, named after the figure in StarKist advertisements who says he has good taste, and his friend tells him "But Charlie, StarKist doesn't want tuna with Good Taste, StarKist wants tuna that Tastes Good."
After that Bro came over to the Psychology lab at Pomona College and attempted to get the little grandson amplifier board working. But it was not going well. I made what I thought was a friendly comment but Bro took it badly. He said "Don't disparage me" and then he refused to do any more work on the amp. Fine, I said, it was your idea and I didn't really need it anyway, I just wanted a stepped attenuator.
Some time later, in 1978 when I worked at Audio Dimensions, the parent company of Audio Directions in San Diego, the store carried GAS products. However they were not particularly admired by the owner and staff because we always had something else we liked better, including electronics made by Spatial, Mike Moffat, Threshold, PS Audio, and so on and on. The owner Ike Eisenson was big into modified tube equipment, and technicians like me would do the modifications onsite following his Tu-Be manual. But Ike's well known penchant for modifying every kind of tube equipment meant we didn't care famous tube lines like Audio Research and Conrad Johnson. They, rightfully, didn't trust us. The owner believed his modifications were better than those anyway. Instead we had carried transistorized amps from all the then smaller companies who didn't have onerous contracts and other "requirements." GAS was just another one among many of these companies.
In fact, and though I had never seen him there, James Bongiorno already had a colorful nickname at Audio Directions,. We called him Bongo Jim. I asked "That's because he's a musician, right?" Some senior tech set me straight right away, "And because he constantly beats his own drums." Another senior tech walking down the hall said "And he's more than a bit bonkers from from listening to them too." Ro Pennell, the business manager, popped out of her office and said, "Now get back to work. And quit badmouthing our idiot suppliers."
Meanwhile, Ike gave his top level of respect for Richard Freyer and Damien Martin of Spectral Audio. Ike, who wasn't afraid to tout his Mensa membership, pre-announced their visit to the store staff by saying they were the smartest people he knew. We never carried the Spectral line however, it was too expensive, and they probably did have onerous requirements. In the interview above, Bongiorno disparages Spectral and their 3 Mhz bandwidth as nuts.
Bongiorno had gone to great lengths to eliminate capacitors and transformers from the signal path. I'm not sure if he was the first, but was at least one of the pioneers the use of servo loops in power amplifiers which not only cured any minor DC offset but also eliminated the need for even a coupling capacitor at the input or anywhere else in the signal path. So amplifiers like the Ampzilla and Grandson are Direct Coupled.
On the other hand, Spectral believed that servo loops were not the answer. Spectral did not believe that input coupling capacitors were pure evil, only that input coupling caps had to be high quality caps like polypropylene. They insisted that servo loops introduced their own set of sonic issues which could not be so easily fixed. And properly considered, things like servo loops are in the signal path themselves, so in fact capacitors are not being eliminated by their use.
This is an interesting argument. But I do not believe that servo loops are a significant audio issue when designed and implemented properly. OTOH, nor are good coupling capacitors. All this debate comes to little.
But as it turns out, nearly all of the amplifiers that I have today are direct coupled. This is relatively easy to do in a MOSFET amplifier, like the Hafler 9300, which is fully direct coupled. In an amplifier with Bipolar Transistors as Bongiorno insisted were required for High Fidelity, servo loops are pretty much required for direct coupling, and they are used in all Parasound amplifiers (I have 3 of them IIRC, one in the bedroom now) and Aragon amplifiers (like my 8008BB).
Bongiorno even took his belief in direct coupling to point of making a direct coupled preamplifier, the Thaedra, with servo loops to eliminate any need for capacitors. That is very rare, but hardly anyone builds preamplifiers out of discrete transistors anymore anyway, and I wouldn't be surprised if opamp chips have their own built-in servo loops.
While basically ignoring all the GAS equipment, the Audio Directions staff fell in love with the Nikko Alpha III mosfet power amp. Afterhours we'd just sit in the front room and listen to it. MOSFETs are just like tubes, Ike would explain. Because of MOSFETs, it was a very simple amplifier. It would be over 10 years before MOSFET amps became big in audio, with the ultimate audio genius Nelson Pass himself switching from Bipolars (used in his Threshold amps) to MOSFETs (used in his Adcom, First Watt, and Pass Labs amplifiers). Meanwhile, in the interview above, Bongiorno continues decades later to say MOSFETs are unsuitable for high fidelity amplifiers. BTW my current favorite amp, the James Strickland designed Hafler 9300, is also a MOSFET amplifier. As far as I can tell, Bongiorno never explained why MOSFETs are unsuitable for audio. Sadly I forgot to ask him that when I had the chance on our last encounter.
Once a GAS Grandson amp came into the back room where I worked at Audio Dimensions. Somehow distracted, I first plugged it in, then unplugged, then plugged it in again. That apparently killed it. No one bothered to do anything with it and it just sat there for months like an omen.
I first visited the new GAS factory in Chatsworth to buy a turntable. I hated the Connoisseur turntable I had bought at Audio Directions (the only thing there I could afford). I called it the Con-noisier because I could hear the motor running from across the room. So I bought a Micro Seiki turntable at a special GAS sale, a DD35. I hated it too because the thin wood plinth was so resonant. It was about 12 years before I finally got a turntable I pretty much liked, the Sony PS-X800. That Sony is wonderful until it needs repair, which few techs are willing to mess with. That's how I ended up with a Linn Sondek LP-12 which has been kept in good repair by the same guy who sold it to me, Mark. I now believe that by this time, Bongiorno had already split from GAS.
Then I bought a second hand Son of Ampzilla. I had always believed this was the GAS amp to get because...no fans! Also no problematic servos like the Grandson. (BTW I now have many very robust amplifiers with servos, the Parasound amps and the Aragon. They are indestructible. Also the Krell FPB, which is somewhat finicky, but not so much because of the servos but because the servos are so good they keep everything looking good until the breaking point.) However before long I was plagued by RF noise from this GAS amp. I took the Son apart only to discover the side panels were literally plugged in with giant connectors. That did not look as reliable to me as normal wiring and solder. Though I knew a bit about repairing and modifying tube amps by this time, the Son of Ampzilla was utterly incomprehensible to me.
So I contacted Bro about getting a schematic. He offered me a chance to get it straight from the "horse's mouth."
I drove up from my home in San Diego and of all the amazing experiences I could have possibly had, I visited the GAS factory while Andy Hefley, apparently the new engineer after Bongiorno left, was doing the final design (or re-design?) of the feedback and stabilization networks for the long rumored Godzilla amp. Different parts were clipped in and the square wave response checked. It was not unlike speaker design I've done myself (and seen others, like the late Albert Von Schweikert, do). A memorable example of "Cut and Try" engineering that's more common than many would admit.
I was standing here, wondering when to pop my question about an old amplifier I bought used, while the very future was being re-invented. I was in total awe, as if watching the first open heart surgery. I'd had sessions like that already with senior tweakers at Audio Dimensions. But this seemed more intense and special. Andy, a long time friend of Bro's was clearly seemed in charge, and Bro was helping, and there were others watching and commenting.
*** Update, if I remember correctly, Bongiorno himself showed up to this meeting. However I only remembered this for certain after writing this post. This would make 4 times I met or saw Bongiorno in person. IIRC, at this re-design meeting, there was mumblings about "him." Then finally Bongiorno showed up, and never took his trenchcoat off. He was not particularly helpful to the current situation. He sort of approved the (re-)design Andy was making by not vetoing it. Then, as quickly as he arrived, he vanished, before I had a chance to ask him about the schematic for my Son of Ampzilla. Soon Andy and Bro were loading the Godzilla into Andy's van. I was pissed about missing my opportunity with "the man."
From there, Bro and I hurried off Andy's beachside home near Ventura, California. Andy was replacing the two Ampzilla II's he had running his two pairs of Magneplanar Tympani IV's with the one new Godzilla. I asked whether that was a good idea.* No problem at all, said Andy. One Godzilla has even less impedance than two Ampzilla's. He demonstrated the capability of the Godzilla by welding two copper wires together. As Bro was getting the act together I said "don't bother" but he insisted no problem, and indeed the welding occurred.
(*While making this comment, I also had in mind the other question, about whether it sounded good to a normal person on normal speakers. But perhaps I hadn't considered that the "normal" person who bought the latest 400W amplifier was planning to listen much like this.)
But Andy and Bro played way too loud for me to even enjoy.* I could barely stand it. I was only 24 years old and it felt like my ears were going to explode. After a short while I begged to go outside and smoke a tobacco cigarette. (I managed to quit smoking tobacco in 1991.)
(*So apparently there is a market for amplifiers with fans, among those who listen so loud it wouldn't make a difference.)
In combination with the other drugs, and the long day, and everything, that was it for me. I begged to lie down and Andy showed me to his guest bedroom. I promised myself I'd get up after 5 minutes and beg Andy to turn the level down and give it another listen at a tolerable level, if that were possible. But I was exhausted and had no desire to get up and face the music. Impossibly, Bro and Andy kept playing louder and louder for another half hour or so. Suddenly there was deafening silence, punctuated only by Bro yelling "Fuck!" multiple times. I ran out to see if everyone OK. Andy had gone off to get something. Yeah, Bro said, pointing to some burned out part, "It wasn't supposed to do this." I went back to bed.
I was awakened barely on time to be herded out in the morning. Godzilla was going back for another redesign.
Finally as we were paying the bills at a restaurant where we were having breakfast, I finally popped the question about getting the Son of Ampzilla schematic. (I thought Bro had asked Andy up front, the whole point of the visit I thought.) Andy said he'd be happy to get it next time. But it was the last time I ever bothered to ask about seeing him. I later did buy the set of GAS schematics from a guy on eBay around 2002, but I haven't looked at my Son of Ampzilla since the 80's.
Bongiorno always blamed "poor marketing" for the failure of GAS after he left. However I've always had a different idea based on my own personal experience of two GAS products. I do not believe they were the most reliable. I don't attribute this to poor engineering on the part of Bongiorno himself. We he compares himself (as a "real engineer" compared to Morris Kessler) I think what Bongiorno is talking about is that he designed by the book and also tested things to failure. Most audio products cannot take such abuse, but Bongiorno came not from the likes of audiophiles (whom he often detested) but musicians, who tend to drive amplifiers into oblivion at a sizeable fraction of full power. THIS is why so many GAS and Sumo amps have fans (which I detest) when most other audio products don't. They are designed to be run in problem test cases like 1/3 continuous power. I'd presume you can run a GAS amplifier with fans at 1/3 power all day long, whereas most other audio amps will shut down before one hour under such abuse. I wonder if Bongiorno had anything to do with the FTC audio amplifier power labeling requiring a 1/3 power "preconditioning" for an hour. Most amplifiers will shut down before the hour is finished.
I think the problem stemmed from assembly, not circuit design. I think those very wave soldering machines that Andy was bragging about were the downfall of GAS. Wave soldering is a tricky business to technicians educated on the old fashioned kind of soldering. You may notice now that there are not many GAS amplifiers for sale and many have been repaired many times by now. The Dynaco 410 I have was only refurbed once, and I just did it because I thought it would make it better (it made no difference). If indeed there was a reliability problem at GAS, it might also have stemmed from the cocky attitude of the staff, including Bongiorno himself and my two friends who worked there.
But it's possible that "marketing" was indeed an issue. And it likely was an issue long before Bongiorno left, in large part because of James's own personality. You can see that he didn't get on with audio companies very long, even when he was the head honcho himself.
While Bongiorno is universally praised by audio magazines now as "a genius," that was not necessarily true when he was making the GAS and Sumo amplifiers. Bongiorno was highly praised by his friend (and fellow engineer) Bascom King at Audio Magazine, and to a lesser degree at J Gordon Holt's Stereophile (J Gordon Holt actually preferred the Morris Kessler designed amplifier SAE made before hiring Bongiorno), but he often got a cold shoulder from The Absolute Sound. The sound of one of the GAS amps was described as "Hard as Nails." If you read those early issues of The Absolute Sound, you might find Alan, my other friend who worked at GAS, responding very derisively to these reviews. The reviews probably deserved that, but it didn't help either. Alan was in "marketing" at GAS.
I would not at all be surprised if there were differences between Bongiorno and Harry Pearson that went far beyond audio as such, political. And these would not at all be ameliorated by Bongiorno's tendency to disparage subjectivist audiophiles and reviewers out loud, though perhaps unbeknownst to himself being actually one of them himself, in not founding his requirements in double blind listening tests. But he considered himself an engineer above all the rest except his small set of key friends like Sid Smith and Dick Sequerra. Bongiorno was shooting for objective accuracy, which was after all what is required, into any kind of impedance, because as David Rich says, most loudspeakers are designed by tweeks and have near-impossible load angles.
Sadly when it comes right down to it, GAS products were not the products that audiophiles of the time most wanted. An amplifier of about 100-150W most would be suitable for many, and GAS never built an amplifier in that range. Either the Ampzilla, which was too big, ugly, and noisy because of its fan, or the Son and Grandson which were too small. The Ampzilla may have been a beautiful and inspiring electronic design, with direct coupling, servos, and Complementary Symmetry, but it didn't look like high priced audio gear. It still looked like a kit made from surplus parts, which was famously how the project got started--but it was a kit was never actually delivered, after having been advertised, at some point Bongiorno decided it was too complex for hobbyists to build, which was correct, and he converted kit deposits to deposits on the pre-assembled amplifier, which most accepted.
What might have been a more profitable hit would have been a recasting of the 200W Ampzilla from it's cheap kit-like appearance up to a massive Mark Levinson style chassis, and with no fan!, and hopefully less inflated price than Mark Levinson. Sumo products were generally a step forwards and away from the stereotyped styling of GAS products, but Bongiorno still couldn't get away from those darned fans in his electronically wonderous statement pieces like the Sumo Model Nine.
Why bother to have an amplifier with 130dB dynamic range if the listening room itself can't get 40dB below the level of 1 watt due to fan noise? And generally I want to have my power amplifier in between the two speakers in plain view, polluting the very important phantom center image if it makes any noise at all. (Done properly, nothing is Better than a phantom center image. A speaker there always sounds fake to me... Though I've never heard Bongiorno's trinaural system.) I can't understand why someone as obviously brilliant at amplifier design as Bongiorno couldn't get this basic fact over decades. Musicians are generally more tolerant as they play even louder and already know what the notes are.
Amplifiers like fans, but most amplifier fans don't like amplifiers with fans.
I think the basic problem was, Bongiorno didn't listen to anyone else, and it's not clear he spent much time listening to recorded music either, by his own admission. He was too busy. He knew how amplifiers should be designed, and that good engineering was building an amplifier better in every way every time. Brilliant engineering was being original in finding better solutions every time as well. He was all that. He had no concept of want, need or good enough--which is actually what all good amplifiers are, and not perfect.
This all leads up to the second I met or at least saw Bongiorno in person. That was in the fall of 1981, when the San Diego Audio Society was hosting both Bongiorno (then at Sumo) and Nakamichi (who had just introduced their offset correcting turntables). It was one of the most amazing audio events of my life and I only wish I could remember the details better.
The meeting was held at Audio Directions, had expanded greatly since I left the company in 1979. Ike himself had established the San Diego Audio Society but by 1981 it had become independent, and Ike had left the company, divorced his wife, moved to the east coast, and was no longer working in audio. The store was being run by his former mother-in-law, Ro Pennell, who had put up all the money for Audio Directions and Audio Dimensions in the first place.
In the much enlarged main room, there were more audiophiles than I had ever seen then (and after) in a local audio society meeting. It was packed with GAS fanboys and the curious as well as local audiophile regulars. My friend and I arrived a bit late but found a place in the middle of the packed room.
Bongiorno talked about balanced amplifier design, why it was so much better than unbalanced, in its interactions with both input and output, and also hyping up the specific ideas in the Sumo "Nine" Class A amplifier which he then loved the most, including the 4 quadrant feedback (also touted in his much later Spread Spectrum Ampzilla 2000 and Son of Ampzilla 2000). Back then I really bought into all Bongiorno's hype about how wonderful and important this all was (though over time I've become much more cynical about such things...unbalanced amplifiers can be just fine, and EVERYTHING has trade-offs, for one example I think the unbalanced Hafler 9300 vastly superior to the balanced Hafler 9500--which has nearly 10x more THD+N, possibly from noise, which is one bane of balanced circuits...OTOH I'd love to get my Krell FPB 300 working again with a change to medium bias Class AB, it has incredibly good and fundamentally balanced circuitry and that IS a plus when everything else is right.).
Bongiorno then as always claimed credit for the Complementary Symmetry circuit used in audio power amplifiers. Actually Complementary Symmetry was pushed by the semiconductor manufacturers themselves, after they had made the first reliable NPN/PNP complements. The first Complementary Symmetry audio power amplifier was not the Ampzilla nor the SAE Mk XXXIb but probably the JBL SA600 made in 1966, several years prior to any Bongiorno amplifier design I know of. And like everything else, complementary symmetry is a trade-off. It turns out the silicon PNP and NPN complements are not at all perfect complements. You might do just as well with Quasi-Complementary as used in many famous Marantz amplifiers (including the Marantz Model 15 Bongiorno himself praised as Sid Smith's best design) and amplifiers still being made today. Many designers have made a point of using Quasi Complementary as an inherently superior design. Ultimately here is no "magic" formula which will transport you to a different world, in fact good solid state amplifiers that have less than about 0.1% distortion sound exactly the same, I myself learned in the late 1980's (and reproved to myself in 2018 and again and again). But I digress.
Bongiorno also talked about tuners, why ceramic filters were inferior to the tuned circuits used in Marantz 10 and 20 (which he had some experience with during his short time at Marantz) and which were going to be used in the Sumo Charlie. He didn't think Kenwood was at all on the right track with their Pulse Code detectors, but other "statement" Kenwood tuners were pretty good. Bongiorno specifically praised the newly introduced Kenwood L-02T but not the earlier L-01T which had used a pulse counter design. (He was right, of course.)
(As far as the anti-ceramic filter thing, I believed in that for awhile but ceramic filter based tuners can be incredibly good. Many of my favorite tuners are some kind of mixture or entirely ceramic. I believe my all time favorite L-1000T is a kind of mixture as earlier statement Kenwoods had been. The phase errors of whatever introduced by ceramic filters barely show up in the wash of system phase errors we aren't much sensitive to, anyway. And there's no evidence ceramic filters are especially microphonic. I think the Sony 730ES is all ceramic filter and it's fine sonically. Over time, the "stays aligned" thing becomes more and more important, and ceramic filters are stable in that way, but no more stable than other low impedance circuits. It's the high impedance circuits like in all tube tuners that are fundamentally unstable over time.)
When asked by my friend what speakers he used (something Bongiorno had never discussed), Bongiorno conceded he had had his fill of speakers that didn't account for room acoustics and poorly recorded music, and he himself didn't bother with recorded audio at home anymore. He had just bought himself a grand piano and was playing that instead. I took this as dodging the question. My friend took it as proof that Bongiorno was useless as he didn't use amplifiers on speakers himself, so what did he really know?
While it may have been really cool, the Sumo Nine just didn't have enough power for most audiophiles then, despite the fairly high price (for the time). And the fan killed it for me.
I wanted to go and ask Bongiorno some questions but my friend reminded me about the Nakamichi turntables, which were unusual and should have been groundbreaking, so I went to than presentation next.
Finally the last time I met Bongiorno in the flesh was at T.H.E. Show in Las Vegas in 2009, which was happening at the same time as the January CES show. By this time I'd actually spent a few evenings sparring with Bongiorno online on the FM Tuner List, which had been a serious preoccupation of mine from about 2002-2005. Not that I disagreed with what anything he said in particular, but Bongiorno never responded very well to "questions." He had pat answers to just about everything and stuck with them.
Bongo Jim was also into his own set of things to the exclusion of all others. He raved about how the Marantz 10B and 20 and Sequerra Model One were the best tuners ever made, with his own Charlie following in their footsteps. Well, he had (briefly) worked at Marantz when the 10B was being made and the 20 was being designed, and was an associate of Sid Smith and Dick Sequerra. So this was all like his family. And he was committed to being their street fighter.
Most tuner aficionados don't have very much respect for Bongiorno's Charlie tuner*, and only limited respect for the others despite their ongoing cult-ish followings. They are understood as tuners that might sound "good" (the famous/infamous Marantz Champagne sound) on strong local stations, but are not very good at picking out distant stations or when there is interference. A short example list of tuners that tuner experts would prefer include the Kenwood L-02T, the Sansui TU-X1, the Pioneer F-26, and anything made by Accuphase. Meanwhile Marantz made absolutely unbelievable claims about the performance of tuners like the Marantz 10b, such as 150dB alternate channel selectivity. That's simply impossible, and it would be impossible to measure too. In fact the Marantz 10B is not a highly selective tuner, just somewhat better than the average tube tuner. The tube tuners best at handling urban reception problems might well be the Scott tuners.
(*This is further complicated by the usual story. Bongiorno departed from Sumo at about the same time as the Charlie Tuner had been introduced. Only the first 1000 or so Charlie's got the final alignment by Bongiorno himself, which is just as tricky as with a Marantz 10B, the "official" secrets of which have only ever been revealed to a handful of insiders like Mike in San Diego. Those are the ones with rack handles. Bongiorno claimed that all the rest were never aligned correctly by the Sumo factory. He had given them instructions which they refused to follow. And that was the sort of reason why he left Sumo also. But according to some who have tested both, there isn't actually much difference between the handled and unhandled Charlie's. They are pretty good sounding on local stations and that's it. Many of the Marantz 10B inspired tuners suffer from things like "birdies" on problematic stations, and the Charlie is no exception.)
One of the first articles to pick apart the cult following of the Marantz 10B was the test of tuners by The Absolute Sound in issue number 6 in the mid 1970's, which didn't actually formally test a 10B but compiled a list of anecdotes about them. (Note: this tuner review hyped the Sequerra tuner way up, but was entirely based on audio quality with home transmissions, not real world reception problems, thereby overlooking the limitations of the varistors to be as sensitive as an air capacitor as in old dial tuners. But meanwhile it was the first magazine column to even mention the Marantz 10B cult, only then to immediately pick it apart. I personally have one of each kind of Charlie, with the rack handled version serial number below 500, a collector's item, but I haven't listened to that one much because sadly it had tobacco smell. The unhandled version is "ok" but only "ok.")
The general dislike of early Marantz tuner hype continued on the Tuner discussion list in the 2000's, with the exception of Bongiorno and a San Diego technician named Mike (about whom I have too many personal stories to recount here) and a few others. I myself think the Marants 20B is an incredibly good sounding tuner on clean local signals and worth the refurbing I need to do to mine on that basis. It may well be the best "engineered" of the lot...including even the Sequerra. The 20B stuck to the basic 10B design but basically did it even better with transistors. The ultimate flaw in the Sequerra tuner is the fundamental inferiority of varactors to tuned capacitors.
Well I really wanted to know exactly why Pulse Count detectors were inferior, as Bongiorno and others had long insisted. I had bought a 600T and several people I knew considered the KT-917 as "the best tuner ever." (I regretted selling the one I had in 1989-1991). So I asked Bongiorno what kind of measurement could be done to show the inferiority of Pulse Count Detectors. Bongiorno was not an audio subjectivist (or at least he didn't think he was) but an engineer. So he should have the answer. This time, Bongiorno gave a most memorable answer while trying to duck me as quickly as possible.
"The problem is not in the measurements, it's in the theory."
I've talked about that idea before in this blog, and it's become part of my central audio philosophy. It's really hard to unpack things by measurements sometimes (though I still believe it should be possible somehow). And it's why I adjust my phase corrections "to the model and not the measurements," though, the model itself being informed by the measurements. Neither I nor Bongiorno write off measurements entirely, as many subjectivist audiophiles do. But you can't always tell the important things from simple measurements.
If I remember correctly, Bongiorno also said something like "you'd need an output of 100v to see the difference"). Then it's a question of resolution, and indeed resolution errors can be buried in the noise, as they probably are in tuners. Then, do they matter? Well they represent different things, correlated vs uncorrelated noise. I think in principle it matters, just as Bongiorno said, but maybe not if the resolution is good enough, as with 24 bit digital. Kenwood pulse count tuners are doing something with way less than 24 bit bit resolution. But it's not clear how MUCH this matters when you're only talking about 80dB S/N anyway. It doesn't matter as much as many other things, perhaps.
For the past 2 years, since I removed my most beloved Kenwood L-1000T for chassis modification (vent holes to be done by a machine shop, because the L-1000T runs too hot) I've been using the Kenwood 600T as the kitchen tuner. The 600T is Kenwood's very first tuner with a Pulse Count Detector. It was their top of the line model in the same year the KT-8300 was introduced, the Pulse Count detector gave it about 6dB higher S/N, and was a big marketing thing (never needs alignment). This was just after many of the original Kenwood tuner designers split and formed Accuphase which started making very nice tuners but with conventional detectors. It looks as if some of the senior engineers had been unhappy with the Pulse Count Detector project but Kenwood management insisted on going ahead with it anyway.
What I like most about the 600T is the signal strength meter, which unlike any others (except KT-917) is calibrated in linear increments of 10dB: 10dB, 20dB, 30dB, and so on up to 80dB. The sound is decent, PROVIDED you turn the MPX filter on. Otherwise it has the slightly too bright "classic Kenwood" sound that most "silver box" Kenwoods, including the KT-7500 and highly praised KT-8300 Kenwoods had regardless of whether they used Kenwood's original detector or the Pulse Count detector. Only with the L-02T and L-1000T (and others of that general design) was Kenwood able to get away from the too-bright "Kenwood" sound. The PLL "linear" detector was best. (Note: I am not talking about PLL Multiplex here, as most tuners since 1978 or so have had that, as it's built into nearly all the MPX chips everyone has used since then. A PLL Detector is something very special that only very few high end tuners have).
The fact that the alignment hasn't changed as much as it might have otherwise is also a big plus for the 600T. That's probably why the KT-8300 now sounds notably less nice than the 600T. Mind you, the 600T is a very complex beast with intermediate frequencies! There are in fact a lot of alignment adjustments. But they don't matter as much with a pulse count detector, whose performance is barely affected if everything else is slightly off.
So I've been listening to a Pulse Count Detector tuner for the past two years, and it's "ok." True, I like the L-1000T with it's PLL detector and analog-multiplier MPX (the ultimate best analog technologies for these tasks) best of all, but the 600T is ok. I think I like the 600T about as good as most of my tuners except the very best sounding ones, the L-1000T and the Sony 730ES and Pioneer F-26 and possibly the Marantz 20b (not listened to in awhile). I had intended to use the Pioneer 9500 instead of the Kenwood but it's been too hard to get out of storage. I suspect it's like the F-26 but slightly noisier. Meanwhile I think I may prefer the 600T to the Onkyo 9090 mkII, despite the "inferior" pulse count detector of the 600T. I prefer the 600T with MPX filter engaged to the KT-8300...but I've never tested KT-8300 with it's MPX filter engaged, come to think of it. It may well be that's the trick for getting all of the "Silver Box" Kenwoods to sound good. It wouldn't be surprising if almost everything comes down to the high frequency response--does it tilt up or down?
Highly knowledgeable (and perhaps less egoistic) engineer David Rich positively concurred in the 90's with Bongiorno's assessment that the Kenwood's pulse count detectors were inferior, and that the L-02T PLL Detector was far better (and said on paper the L-1000T had the most promising of all designs except for the digital tuning which is inherently inferior than the analog of the L-02T, he never had one to test though). Then Accuphase in the later 2000's started making much more sophisticated pulse count detectors that did indeed have the required resolution, perhaps.
This came to mind as I'm re-testing my Dynaco 410 for sale. I think I once said something about Bongiorno having designed part of the 410, and Bongiorno himself dressed me down on the FM tuner list. I can't remember whether he wanted credit for all of the 410, or none of the credit. It's basically the same audio circuit as the Dynaco 400, which Bongiorno always takes 100% of the credit for (however, AFAIK Erno Borbely designed the Dynaguard circuit, and somebody else was involved on the 400 too IIRC), but the 410 was released long after Bongiorno left Dynaco. By the time the 410 was released Bongiorno had worked at SAE for several years, then started the GAS company and moved from Van Nuys to the Chatsworth location as GAS had, for a fleeting moment, become a big name. But the Dynaco 410 was Bongiorno's audio amplifier circuit without Borbeley's Dynaguard circuit, so it seemed to me to be even more like a Bongiorno design than the 400. In all his later work Bongiorno stayed away from things like Dynaguard. He wanted maximum linearity.
One thing about the 410 is that it's a very solid piece of metal and weighs a ton. I recall GAS amps being basically made out of bent sheet aluminum. Even the heat sinks. Never cast aluminum. Bongiorno liked the efficiency of that and didn't believe in wasting any money on the casing. However perhaps it would have been better to have more dimensionally stable casing. SAE were a bit more solid too. And Dynaco tended to use rock solid steel. That solidity seems to translate into reliability as well. My 410 has been perfectly reliable for me for 40 years. For some reason, though, I never took it seriously. It never occurred to me until writing this post that this is essentially Bro's amp, the one he wowed us with for several years (until he upgraded to the Ampzilla). It's the Dynaco 400 without the Dynaguard. And for most of the time I owned it, I ran it without the fan without any problems.
I bought this amp with one and only one purpose in mind. To be the poweramp for my subwoofer. I think I used it for a huge "slot-loaded" home made subwoofer I bought at the Bongiorno audio meeting parking lot after the meeting was over.. A year later I bought a pair ML2's to use as a pair of subwoofers. They had some burned out other drivers, but that didn't matter and I was crossing them over with a 24db/octave crossover made by dB systems, which was the second thing I bought at the parking lot sale.
And in 2003 or so I got a Pioneer D 23 four way crossover, sometimes labeled "Series 20" for one of Pioneer's short lived super premium lines (mine is not so labeled, but is exactly the same thing). This replaced the dB systems and gave me total flexibility (and with stepped controls) to set crossover between the subs and my Rogers LS 3/5 A's. But soon I had added a different tweeter (after I burned out the one in the Rogers), the Dynaudio D21AF (one of the best tweeters of all time, I think, and I'm still using them now as my supertweeters) and was then using 3 "ways" of the 4 way D23 crossover.
But without much thanks, the Dynaco powered the ML2 "subs" from the day I got them, and through modifications, until I retired them for a "real" subwoofer, the SVS 16-46, which gives me bass down to 13 Hz in the bedroom, around 2005 or so.
Before then I'd gotten concerned about low level noises. This was in my bedroom after all (where I'd had my main stereo since moving to this house in 1991, since I shared it with my mother) and it wasn't until 2008 that the living room stereo became "high end." Until then it had been mainly used for TV sound at monthly movie parties.
I thought the Dynaco 410 might be a contributing factor so I had it refurbed in 2001. Despite my long time dedication to measuring instruments, I had no way of electrically measuring hum in line level signals. It's tricky and most DVM's don't do a good job with low level signals. I just knew there was hum at the output of the power amp. Refurbing the amp made no noticeable difference IIRC. It still has amazingly low hum and noise. It turned out the hum was being caused by my beloved D23, which itself needs a refurb. It's been sitting in my laboratory skandia shelving since then. I switched to using digital crossovers, first the Behringer DCX 2496. I now have a Meguro Noise Meter which can measure the lowest line level noises you can find in audio equipment with the required "A" rating.
Anyway, as soon as I started using a "real" subwoofer with a built-in plate amplifier, I no longer needed the Dynaco 410, and I always felt my other amplifiers were "better." I had been for some time using a Nikko Alpha III, just like we loved at Audio Directions. (That had been my stand in since selling all my tube equipment in 1991 before moving to Texas). But I decided I didn't totally like the sound of that either, and replaced it with a Parasound HCA-1000A which served as my bedroom amplifier until recently being replaced by a Parasound HCA-1500A which is even better. Dynaco 410 has nothing on either, except somewhat more power than the HCA-1000A.
When the Living Room System got serious in 2008, the first amplifier there was the Krell FPB 300, which went through various repairs. I had originally intended to get something cheaper, an Aragon 8008BB, but it seemed I missed my chance and then had a huge desire to get the Krell. Later, as the Krell needed servicing, an Aragon 8008 BB (and perhaps the very same one) came on the market, so I bought it. The Dynaco 410 has nothing on either of those amplifiers. The Aragon has the same capacity power supply capacitors, except 2 per channel, and somewhat larger transformer capacity. The bias is essentially user adjustable, you can crank it up to high bias Class AB+ like I have done and it then has less than half the distortion of the Dynaco. But at low bias settings, maybe around the official ones Klipsch posted, it is possible the Dynaco has lower distortion. I find it works better with about twice as much bias, and still stays between 120-130F at idle and never gets to 140F in hard use at the heatsinks, which I think is acceptable. It turns out you don't have to think about this too hard, they more bias the distortion, with the heat sinks being the limiting factor. And the Aragon 8008 BB has more heat sink than the Dynaco. I suspect technically the Aragon isn't as well designed as the Dynaco, but it has the advantage of better parts, and is more flexible meaning you can do more tweaking to it w/o completely messing it up. I bought it with aftermarket feet than made it very low, and this pushed the bias up because of self-reinforcing heat effects, to 140F. I put much bigger feet on it hoping to cool it down, but got more than I bargained for as the temperature fell to 100F and in this bias starved condition it had significantly higher distortion. I fixed that by cranking up the bias controls, back up to 120F or higher. I recall meeting one of the founders of Mondial, and he had put something on top of the amplifier to show it was possible AND to make it sound better (because it ran with hotter bias).
Now I use the Hafler 9300, which has essentially the same typical distortion level as the Krell, 0.002%, and with total reliability and simplicity and far less energy cost. I was unable to find a measurement other than power in which the Krell was notably superior. Even in damping factor sometimes the little Hafler edged out the Krell. Both of these amps have about an order of magnitude lower distortion than the Dynaco, and far wider bandwidth out to 300 kHz, which I do think is a good idea (but maybe not 3 Mhz). Things have improved a bit since the Dynaco 400 was introduced in 1972. Engineers know more, and it's not as much a challenge to design great amplifiers anymore either, because the transistors are much better, both bipolars and MOSFETs.
Bongiorno kept his subjectivist faith in every circuit detail being important, and nottesting or proving this faith with double blind testing. So he believed that linearity (and balance, if possible) had to be preserved every step of the way. The truth is, only monotonicity and not linearity is absolutely required in a feedback amplifier. That's why Class AB amplifiers even work, though in principle you could align the two sides for greater linearity than Class A, in reality that would probably take bias levels you wouldn't want to work with anyway. Absolute perfection is not required in every step, but instead good enough. The raved about TIM and SID type distortions in the end turned out to be the same old THD, though there are specific potential problems with intermediate stage clipping or slew limiting that a small number of high feedback amplifiers may have suffered from. So the no-holds-barred ultimate slew rate unlimited design by Matti Otala, the Citation XX, is beautiful but way more engineering than necessary to get decent sound. This fundamental knowledge has now been incorporated in every amp since 1980. Inexpensive Parasound amplifiers have boasted 50 V/us slew rate from the beginning of the company, and up from there.
Actually if you look at the GAS specs, they've been pretty commonplace since the 1980's., like 0.05% distortion (my Hafler 9300 is rated at 0.02%, and typically achieves 0.002%...not unlike CD players). Bongiorno success with Complementary Symmetry attracted many to the idea, but comparable performance was also attained by many other means, including the plain old quasi-complementary Bongiorno broke away from. GAS amps have notably low slew rates by more recent standards because Bongiorno wasn't trying to wow people with such specs. Slew rate is technically meaningless if bandwidth limits are reached first, and GAS amps also tended to have bandwidths less than 100 kHz, less than typical high end bandwidths since then of 300kHz and up. Bongiorno makes fun of Spectral amps with their 3 Mhz bandwidth. That means GAS amps didn't "need" super high slew rate to prevent TIM and/or SID. They weren't going to have it anyway because of the 100kHz bandwidth limit. And Bongiorno used proper termination networks, much like John Iverson, instead of being able to claim "no inductors in the signal path" as with Parasound. Either approach works fine, although Bongiorno and Iverson actually gain stability bragging rights by having extra stuff in the output circuit, which they then also lose because otherwise their amps are so more "bleeding edge" (and ultimate part degradations force frequent refurbs more frequently than other amps).
Now, out around 2000, Bongiorno was back at the top of the heap again for awhile, and perhaps even still, with the Ampzilla 2000, a much more advanced design than before (but still building on the ideas of the Sumo Model 9 that I heard the Bongiorno tout in 1981).
https://www.thefreelibrary.com/Son+of+Ampzilla+2000+Circuit+and+Measurement+Review.-a0133864374
However regarding the amplifier, I like the 100 watt output, the 2kW transformer, and it even looks like the Son of Ampzilla 2000 might not have a fan. Here in it's second incarnation, it even has very nice looking cast aluminum heatsinks on the sides! The price is about right for an amp in the High End category, but not for poor folk as I've become.
David Rich speaks to the Cut and Try amplifier stabilization and loudspeaker optimization techniques I thought I saw being used by Andy Hefley and Albert Von Schweikert. He does not like them, and points out that they won't work with complex balanced amplifiers like the Ampzilla 2000. Anything done to stabilize one servo or feedback loop may well destabilize the others. He believes that amplifier feedback loops, and especially servos, should be designed first on paper and then best verified with a computer simulation. He knows that Bongiorno used to use the pencil and paper method exclusively but started using computer simulation with these 2000 amplifiers. It requires an understanding of Control theory and it's subset Compensation theory specifically for feedback loops. Those that Rich calles Tweak audio designers do not understand such things very well must rely on Cut and Try and sometimes to their downfall. BTW, I'm a fully unqualified Tweak audio designer myself.
In the end, David Rich concludes the Ampzilla 2000 is the most advanced audio amplifier available (at the time). The Son of Ampzilla 2000 is a bit less, sacrificing some of the performance. But he also doesn't think anyone needs such a perfected amplifier.
Reading the description of the Ampzilla 2000, I can't help thinking that it sounds a lot like my old Krell FBP 300, which is also a fully balanced amplifier following the 4 criteria Rich mentions. (Hence the "Full Balanced" in the name.) And like the Ampzilla 2000, the FPB amplifiers are also two stage and have regulated power supplies for everything. And they have the differential servos, plus computer monitoring. I'm not sure if the second amplifier stage is differential, however. Honestly I think even Rich tends to over hype Bongiorno's originality. The downfall of FPB is that the top Class A plateaus should not have been used. FPB amplifiers can burn out their own transistors in an intermittent no-load condition (and some very problematic speakers like Acoustats) because the heatsinks cannot possibly absorb the maximum output of the power transformer (3kW in the case of the FPB 300) and remain below destructive temperatures. And Motorola quit making the transistors with nothing else of the required capability available in the same TO-3 case. I feel a bit guilty about this because I met Dan D'Agostino at a different audio society meeting in the mid 1980's. He had just produced a Class A amplifier with a fan, and I told him I wanted a fanless Class A amplifier that changed the bias level to stay in Class A all the time. Now I see it was a bridge too far. High bias Class AB is just fine.
I've also had the feeling ever since that D'Agostino was a gentleman who listened to his customers, like many of the best audio equipment makers, rather than talking down at them. But we all make mistakes.
One thing I personally don't like about many of Bongiorno's top amplifiers was the use of fans. An actual audiophile would appreciate the silence of having no fan, or fans only switched on only in extreme conditions. Of the GAS products, the only one I liked the looks of was the Son of Ampzilla, which had no fan either. But both the Son and the later Sumo Model 9 are a bit underpowered for speakers like my Acoustat 2+2's. The Model 9 has a fan too which I remember as being too loud for me.
I wish I knew more about the kinds of speakers Bongiorno used when and if he ever used speakers personally. (He admits to designing speakers for Rectilinear, though he hated the company's owner.) I would have placed Bongiorno in the Dick Burwen camp as a believer in loudspeaker and room correction via electronic EQ. And even in 1981 I remember thinking of that and that he probably didn't want to discuss such things in a room full of audio tweaks. He might have mentioned EQ in those talks but didn't dwell on it. Famously he refused to ever design a preamp without tone controls, which had become the standard audiophile thing by the late 1970's.
In his own words Bongiorno often seems to dismiss his audiophile admirers and customers as idiots. He was a musician and an engineer, not an audiophile as such. To him, designing audio amplifiers was a kind of mathematics in which he revelled, but without much touching the ground. And he wasn't that great as an audio consultant, partner or businessman, nor necessarily as an audio scientist either. He frequently become terribly monotonous because of endless and ridiculous self-promotion. He possibly might have had more stable work relations with psychiatric medication, like lithium (though at what cost to his creativity?). However he left us with a few (often slightly rough) gems and applied and pioneered if not invented many good ideas in audio electronics which have been useful and inspiring to many. Few can claim as much.
