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.)
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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.
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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.
