Audiocorder and other stuff

For decades, I used timer recording systems of various types to record FM radio programs, just as people use VCR and their modern equivalent HDR to record TV programs.

My Nakamichi 600 had a timer start mode which causes the unit to start recording when power is applied.

My Nakamichi RX-505 has the same feature, and I have used that as recently as a few months ago to record FM radio programs.

I even had a Sony all-in-one music center (a small box) which had the feature, which prior to 2006 I used to record daily radio programs for playback in my car.  It had been fairly hard to find that unit.  However, after 2006, the march of "progress" eliminated the cassette player in my latest car.

This was a huge step backwards for my personal connectivity.

While there were supposed to be digital alternatives, I found none so convenient.  For example, in order to use podcasts, I needed to connect my iPod to my computer every day and fumble around with iTunes.  For various reasons, this often didn't work, and it was ALWAYS more of a hassle than simply popping a cassette into my all-in-one recorder had been, and finally I gave up.  If the internet radio didn't specifically make a daily "podcast" you couldn't just record a particular program at a particular time.  It seemed to me there is a conspiracy against timer recording.

Perhaps there are timer recorders for digital radio now, I haven't looked.  More recently I use smartphone apps to listen to specific programs I like....however this doesn't help if no one has created a suitable app for the program you want to listen to (and there are several programs like that for me).

Anyway, here is Audiocorder, which lets your Mac operate like a timer recorder for audio.

Then, using an extra program they suggest, you can get it to record internet radio also.

Now, stepping somewhat back from High Fidelity, another interesting recordable thing out there is Pirate Radio.  Some of these are on FM because virtually all listeners already have the required FM radios. But most pirate broadcasters are probably sitting in the less intrusive/detectable/complainable regions of HF (say, 6.7Mhz).  I can pick that up with my new General Coverage Receiver (at least, as soon as I get the antenna set up).   While FCC often shuts down pirate radio stations in the FM band (they say, they only do this if someone complains, which is probably accurate) I doubt they have much time for the non-amateur parts of the HF bands and many pirate stations there are legendary.

Here's one list.

And here's another.

HF Underground is showing a bunch of loggings from today.

And of course there are fully "legal" HF broadcast "shortwave" shows also, but books are filled with info on that (including one book I just bought).

I suspect few radio amateurs would risk their own hard won licenses running a pirate radio station.  But I've known hams tell me right away that pirate radio is among the best stuff to listen to on HF.

Help, Part 6

Around 1970 Saul Marantz was interviewed, and I remember hearing it on a Los Angeles radio station.

Marantz said he went a different way than others of his generation, and that was what led him to perfectionist audio equipment.  By following his own path, he discovered new things, created a successful company, and opened up a new arena for others.

(I recall the introduction said Marantz "invented" high fidelity--it appears to have been invented many times, and still nobody knows what it is.)

Anyway, for what it's worth, that idea of following your own path seemed right to me then and ever since.

Just because "all audiophiles" are supposed to do this or that, I not only feel fine doing something differently, I feel it is good.  Everyone doing the same things would mean little new discovery.

Help, Part 5

To EQ or not to EQ

The argument I have often heard advanced against deliberate equalization is that it introduces errors in the phase response, or the time response.

What is not understood, is that when deviations from uniform (aka flat) response occur, they have already introduced errors in the phase response and the time response, and that using EQ to correct these deviations actually restores all three at the same time: the frequency response, the phase response, and the time response (and this is because they are all, essentially, aspects of the same thing...the signal itself).

Now, this is generally true, which means in some cases it may not be true.  But in specifically what many people believe is the most troublesome case, room EQ (to uniform response), famous TAS audio reviewer (and Professor of Mathematics) Robert Green claims that room response deviations from linear reflections, which most are, are minimum phase, and hence are corrected by the inverse deviation.

So far from being horrible, correct EQ is beneficial to the phase and time.

Now this is not to exaggerate the importance of phase response.  Large volumes of research show it is not audible to surprisingly large rotations.  Anyway.  In all cases like this, existing research doesn't prove it won't be audible at this or that time.  But, if you want to prove you can consistently hear it, it's probably going to be hard enough that you wouldn't bother.  So, if it so hard to hear, it can't be too horrible, can it?  Not to say the challenge of getting perfect time response, hopefully w/o sacrificing too much else, and appreciating it as one can, can't be a fun endeavor on it's own.  But existing knowledge doesn't suggest it's worth sacrificing many other things, notably the frequency response, for.  In most cases.

Then there is the matter of what you might call, deliberate non-uniform response.  Now, those indeed will be adding time and polarity deviations, along with the frequency response variations.  But this is the natural way these things occur, and are generally accounted for in only describing the frequency response (not the "horror" of deliberate EQ in particular).  So if you were, say, achieving the same result with some tweaky infidelity, exactly the same and not added distortion or whatever, your tweaky response correct would have the exact same effect on the phase and time as a deliberate EQ system--preferable a digital EQ but some high end analog EQ's are truly high fidelity also (such as the Cello Pallete).

Now, before the days of digital EQ, analog EQ's had lots of practical issues, including but not limited to distortion causing devices.  It was rare to find a really good one; before the Palette, I'm not sure there was.  Now, with digital input and digital output, a cheap digital one can be as perfect as the 24/96 audio it puts out (in the case of the Behringer 2496) when you use full digital I/O.  The distortion is many zeros below the decimal point, the noise level is better than -144dB.  Jitter does not appear to be any higher in the output than in the input...it does not appear to add to jitter (nor does my whole chain of processors and converters)--and once again not to exaggerate the importance of jitter, but just saying.  And of course, precisely the filter you wanted, with no pole out of place.

Designers of loudspeakers have long understood this, and the importance of compensating for driver impedance and response in the crossover is done.  This is just another form of deliberate EQ, often with somewhat precision-limited parts, but often very carefully considered designs ("engineered"). And correcting the driver response corrects the phase as well.  However, most crossovers do introduce group delay and some time dispersion.  Virtually all scientists, engineers, and manufacturers do apply more crossover than the minimum: just lowpass on the tweeter, the 1930's standard.

The primary issue involved here, IME, is loudness uncompensated listening comparisons.  Without compensating for the difference in efficiency, a more minimal crossover will play louder, and hence have a large tendency to sound better.  If drive power can be increased without issue, the less efficient corrected response, done correctly, would be preferred.  Not that there aren't limits as to how far this can be done without diminishing dynamic range.

Since the scientists, engineers, and manufacturers generally do level adjusted comparisons (though perhaps within a desired range of efficiency) they find the more complex crossover to sound better.

Subjectivist gurus do not do level matched comparisons, so the more minimal crossover sounds better.

The same issue, typically combined with sighted expectation bias and other biases, explains a lot of tweako beliefs, from power conditioners (maybe not a bad idea, but not with the massive benefits claimed), expensive audiophile cables (which if they do anything at all to the sound is probably destructive to simple well made cables), lack-of-feedback, single ended, and NOS.

In every case, if there are actual changes, people may find they prefer them, for highlighting this or that, but perhaps also what people find they prefer in casual testing might not over the long term be appreciated as well.

I think it's very risky to try to achieve fidelity through infidelty, other than correctly determined and deliberately applied EQ and possibly out-of-band filtering.

But, it's your roll.

Now, why does loudness, even trifling amounts, make things sound better?

Because it increases our auditory S/N, and opens up a far greater amount of information.

Now one thing to be clear is we never fully absorb the information of even the most limited presentation.  So why do we need more information?

Because it gives our auditory system a greater sample to choose from; it opens up patterns that previously were hidden by noise (most often...household ambient noise).

Now, there are limits to this that vary a lot, and playing so loud as to cause compression (not to mention clipping) may be undesirable (especially, compression in your own hearing system).  And as I posted before, actually avoiding current or voltage limiting may be far harder than people recognize.

Which is another thing.  Unless you have very high efficiency speakers, or even so, you may need far more power than you realize.  In practice, little spikes of peak power are required up to alarming amounts.

Sanders' estimate was 500W for typical dynamic loudspeakers.

That may be a bit high for typical dynamic speakers.  I would think 100W for kiddie speakers, 200W for serious speakers, and 300W for inefficient high end speakers.

But there I am, giving advice again.

I see John Curl describes his friend Dick Sequerra's Metronome speakers as "kiddie speakers."  They're small, with small drivers, though possible do require more than 200W amplifier.

Anyway, they take clean impulse response to an exteme.  The crazy looking tweeter and weird woofer do actually deliver very clean impulse response, hence the name Metronome.  However, the frequency response is way out of whack, and finding the right "location" may be problematic too.

This is the example of sacrificing much to get the perfect impulse response, an ultimate tweako goal (since the serious scientists, engineers, etc, don't consider it THAT important, though never discounted entirely, and there is some room for doubt).

They do have amazingly well focused imaging, though not all people like that either.  But many people who try them for awhile find they dislike some aspect of this approach.  I'm sure I couldn't stand it at all, though it might be interesting to have a pair for an imaginary magic show.

So that's what they are.  Stage magic equipment.  A good description also of high end tweako stuff which is less well engineered.

Help, Part 4

My friend Tim has intensely studied many transistor and tube characteristic curves for decades now.  He's a very technically and mathematically astute.  He's also done some test building over the years, but mostly his beliefs come from studying the curves and the mathematics.

He says now that push pull BJT* amplifiers work best in Class AB, not Class A.

Of course this flies in the face of everything I've ever read, heard, or thought about, and my most prized audio machine is a Class A amplifier, because I've long been a fan of Class A amplifiers and my belief that they are best for electrostatic speakers.

He agrees it flies in the face of standard and conventional thinking, but insists that nevertheless it's true.  His argument is that the "cutoff" distortion on one side precisely works with the changes on the other side to produce the lowest distortion at some particular AB bias.  Class A doesn't get this benefit, and produces higher distortion.

He's been wrong before, so he could easily be wrong this time.  My suspicion is that it either misses other factors, and/or

But, if he's right, and in many cases even if he's wrong, it's quite possible that Nelson Pass's Threshold amplifiers (which were BJT, he switched permanently to MOSFET after leaving Threshold) were sub-optimal.  The attempt to eliminate cutoff, Tim affirms, would produce higher distortion than the correct Class AB, and would therefore produce higher distortion than necessary, and at higher operational costs.

That's certainly true of some other of Nelson's designs, such as many devices in the single-ended Aleph era, and many of the First Watt designs.

Is this a big deal?  That through charisma or whatever Nelson sold higher distortion amplifiers than necessary?

Firstly, I suspect all the Threshold amplifiers all had distortion sufficiently low, and with sufficient reliability, and relative high power and current into most loads, to be considered fully High Fidelity Amplifiers, such as there is no respectable evidence that people can hear differences between.  So the "loss" to the users of Threshold amplifiers is nil, by the audio objectivists very own standards.

They've been loved, held their value, etc, etc.

So, that's fun, entertainment, and playing music.

Secondarily, when the actuality doesn't really matter, the only thing that does matter is the story.  The story says lower distortion, greater PRAT, or whatever.  The story makes you feel good.  The increased distortion is irrelevant, in most cases, even in many extreme cases where people just seem to groove on the distortion.  Minimum distortion I would generally look for, myself, as much as I can afford.  But if other people like other stories, other presentations ever, that's their thing, at least while it lasts.  I think, in the end, as little as possible distortion, combined with the greatest flexibility (the equipment is supposed to let us Play music, not have Music imposed on us)

I believe many of the beliefs that modern distortion reducing methods create bad sound are myths, having perhaps vanishingly small examples, and none where proper engineering has been done.

But there are actually many different circuits, it's hard to say we have an "optimal" circuit as such, and if we don't have an optimal circuit, then we don't have an optimal class, and so on.  Furthermore some things may create more or less distortion than other things under some circumstances, and have the reverse be true under others.

In fact, as is well known, people love the higher distortion circuits like SET's and transformers for various reasons.  And since this is about fun, not Accurate Reproduction, that's fine, for them, as long as they continue to do so.  And much as I might think it would happen, many people are sticking with one of many higher distortion than necessary paths.

We don't hear distortion very well anyway.  It's the frequency response under load we hear mostly.  People are finding or not the EQ they want.  Some people fear EQ.

Many like me fear this is an expensive, and even more than that, difficult and ineffective way to get very much EQ.  But if someone thinks it works for them, it works for them, fine.

I have actually experienced a "solid state" "dryness" (in a malfunctioning Citation 11) which couldn't be eliminated with (it's very own) EQ.  That happens when you have higher noise and IM than necessary.  You can't EQ that away.  But EQ works quite well for most other things.  When we have digital sources, we can have distortion and noise free EQ, which provides exactly the alteration needed or desired.

EQ lets me easily and precisely play with how things are played, and that's become very important to me.

(Tim variously prefers and not prefers MOSFET amplifiers in Class A.  He has most often said MOSFET amplifiers require very high bias to be in their most linear region, and for practical reasons (would it require liquid nitrogen cooling?), nobody ever does that.  But he may be softening on the bias levels needed to make MOSFET preferable to properly biased Class AB+ BJT's.  I think there may be other factors and ways of looking at the situation.)

Help, Part 3

I have a very simple answer to the question "What is Audio" and I believe my simple answer quickly refutes those stubborn audio objectivists like Peter Aczel and Floyd Toole who say that "Audio is Engineering, Music is Art."

Fun.

Audio is not just music reproduction  Audio is fun, and fun is the ultimate principle that it's all about!

And so of course it is true.  Audio is a hobby that audiophiles passionately follow, and just like all other hobbies, it's ultimately about having fun.

Then I suppose it could be argued that professional audio is different, professional audio is the engineering which serves the creation and reproduction of music.

But even there, and especially for Audio, if your job isn't fun, as many say, you should get another job!  (With the slight disclaimer nowadays, "if you can.")

I'm glad we've gotten that straightened out.

The creation of fun is not something that can be engineered, though the relevant engineering should applied where it needs to be, or it's not going to be fun very long, if at all.

But indeed it does help that we have Entertainers like Nelson Pass*, one of the more fun people in Audio.  He's a very good engineer also, it's easy to see.

(*Pass described himself as Entertainer in a recent Stereophile Interview.)

Avoiding clipping requires high power

Roger Sanders of Sanders Sound Systems (a leading manufacturer of electrostatic speakers today) has a white paper on amplifiers.  He says transistor amplifiers are better for electrostatic speakers because the low impedance outputs can drive the capacitive load, which he characterizes as 1 ohm at 20kHz, with more extended frequency response.  And, because it's easier to get high power in transistorized amplifiers.  He claims most audiophiles have undersized amplifiers, and when they hear amplifiers as sounding different that is because they clip differently.  But the solution is not to find the amplifier that clips better so much as finding the amplifier that has enough power to never clip.

Interesting, and pretty much in line with my thinking.

I don't think, however, I voltage clip either my "200W" (actually something like 500W into 4 ohms at clipping, as published in Audio review) Aragon, or my "300W" (actually something more like 900W into 4 ohms at clipping, as measured in Stereophile review) Krell very frequently.  In fact I'm pretty aware of clipping points because I know the max output of my DAC (which drives the power amp), the digital attenuation I'm using at a particular time, and the digital output level.  (However, there's some wiggle room between what the nominal output is and peak output...so I'm not totally sure how this works out...actually scoping the output for clipping like Sanders suggests is not a bad idea and I've often thought about doing it, just never gotten around to it.)  I believe I generally do not clip either of these amplifiers based on the levels of signals they are receiving, however I do not know.  However, I come far closer to clipping than most might imagine.  At the max levels I listen to, I'm right under those levels by just a db or two.  And that's with 900W into 4 ohms on tap.

He does mention also current limiting.  Current limiting and clipping are somewhat different.  I might be getting more current limiting than I know about, though it's doubtful there is very much since both amps are capable of more than 60A peak currents (IIRC the Krell is around 100A).

Sanders claims that average listeners with 90dB efficiency dynamic speakers in typical listening rooms need as much as 500W on tap to avoid clipping.

That's way more than most people think.  I'd guess more like 200W but I haven't examined this by doing the measurements Sanders has.

Meanwhile I find (in sighted testing only) the Krell to sound better.  Less current limiting perhaps??? I believe that the Krell has flatter output at 20kHz into the Acoustats than the Aragon.  I've seen this in acoustic measurements but never measured it exactly.  It may be because the Aragon is a low feedback amplifer with a damping factor of only 450 below 1kHz, whereas the Krell is a high feedback amplifier with damping factor of 900 or higher and extends to higher frequencies (though, this is not a Krell specification and I can't remember where I saw this measured).  The Krell has zero loop feedback, but the output stage has it's own local feedback (drivers and outputs), as well as more bigger transistors, and power bandwidth to 300kHz.  The Aragon is a more traditional design with loop feedback, but less than 2dB of loop feedback, and power bandwidth of 60kHz, but still lots of transistors for low impedance despite low feedback.

A friend of mine who originally didn't think much of the Aragon has been studying the circuit in great detail, and decided it's actually very good.  The low feedback fits perfectly with this design, and is probably close to the Baxandall recommendations (either very low feedback, or high feedback, but not middle feedback).

He also points out that the Quad ESL-63's have a very amplifier friendly resistive load, not like the capacitive load Sanders describes.  This was part of the ESL-63 delay line design.  However, that is unique to Quad ESL's as far as I know.

Help, part 2

[An ongoing exploration of the audiophile scene.]

Nelson Pass concedes the audio objectivist debate in the objectivist's own terms.  "The amplification problem is solved," he says in September 2017 Stereophile Interview.  But then he precedes in terms very much as in this blogsite I have argued against the hard form of audio objectivism.  He says nobody needs the amplifiers he has produced, or even the circuit designs he has patented.  People can get along fine with other amplifiers and designs.  He describes his job, instead, as an entertainer.  Some people are entertained by his work and his works, and he works to serve them.  I am one of those.

That's pretty much what I've said against Aczel and Toole, who claim exactly that audio is mere engineering which serves art.  Audio itself is art, which partly serves music, but also party serves other things.

Now, for sure mass manufactured products are engineered products.  But there is plenty of art in a manufacturing concern.  And if run well, the product is art, this is well known to all now.

A lot of audio, though, is more personal, much more personal than opening a box and plugging one thing in.

I haven't scratch built anything major since 1982, when I built a tube type preamp with separate power supply.  It sounded great, and bested an Audible Illusions around 1986.  But I lost the massive power supply on my move to Texas, and haven't used it since, as well largely preferring solid state equipment.

I might tinker, such as the modifications I made to the Aragon tape selector circuit which I use in my living room analog racks.  Selections are now unloaded and totally isolated, except for ground.

And there is the ever evolving programming of my DSP's.  I strongly prefer total manual control to automagic EQ.  But that itself is another decision.

Every decision is a form of participation.  Even going to an audio specialist, who plays two things, and asks which you like better...that decision is a substantial participation.

Though, if all you do is make a purchase decision, and have specialists take care of everything else...then it's not very much DIY.  But that's pretty rare among the proletariat.

Most audiophiles are very involved, it's at least a part time preoccupation, not even counting the seriously listening to music part.  I make no pretense that I am not a gearhead, not interested in the vast details...or that I don't collect too much, the latter perhaps being the greater crime.

Some seem to become manufacturers of major equipment on some scale.  Sadly one of my best audiophile friends got massive hearing loss during the metalwork involved in the production of his own Leach amplifier.  And then he hated the sound of the amplifier.

By far most people are surprised by the degree that we imagine, rather than perceive, reality.

People may be scientists, designers, and even engineers, and not get this.

I get it.  I can freely "hear" something to be better.  Hearing is never the same twice, and it's only partly related to the information at our ears anyway.  So, a typical sighted audiophile test, which I myself have done a lot (but generally avoid, I prefer to do no upfront auditory testing, "long term" testing only, and that has not been the source of any of my numerous errors) means nothing.

I won't say it's impossible that there is this unknown X factor, which makes amplifiers sound different, not related to the things we know like frequency response into speaker load and IM and absence of voltage or current limiting.  Possibly there is some "simplicity" "charm" or whatever.  But very dedicated testers have done serious tests for 50 years, a nothing has been confirmed.  All other claims have been debunked.

A friend--the one who took his Sony 3200F to audiophile meetings to show up the big boys, was convince he had the trick.  Just listening for himself, and with some luck, he had found the answers.  Get a 3200F!  He couldn't explain why it sounded better.  His mantra to this day remains that measurements don't matter.  (It seemed strange to me, though, that he says measurements don't matter, and yet by some chance he chose a good measuring amplifier.  Seemed unfair somehow.)

Most people in most of these gathering seemed to agree his amplifier sounded best.  By his account, all of them who seriously listened believed the 3200F better.  Of course virtually all of these tests were sighted, none were level matched (he had an argument against that...it cannot be done because there is more than one frequency to match), so as a proof that the amplifier problem has not been solved, they are inadequate in all ways.

I wouldn't think that I or anyone could have such luck, so much better than everyone else, that I could ever stumble upon The One Answer like that.  Instead, I think his entire enterprise was a repudiation of amplifier differences, rather than the proof of one person's excellent luck and taste.  The amplifier problem was solved in 1968, if not before.  The Sony had modern specifications, such as 0.1% THD and virtually flat frequency response.  Far from proving that measurements don't matter, his experience is more evidence that once measurements get good enough, people can easily be swayed one way or the other, merely by one's presence, or tendency to adjust the volume slightly higher for one's preferred amplifier.  People including oneself.

It is known that we are NOT very good detectors of harmonic distortion, and in particular of the kind most audio circuits produce.  It may take 1% or more to hear it on music.  In short term tests, meanwhile, we are far more sensitive to small differences in frequency response, 0.1dB is roughly equivalent to 0.1%.  In fact, in large part, we may be hearing the harmonic distortion by the effect it has on frequency production.  Bass distortion is often heard as louder bass.  But, the same nonlinearities which produce THD also produce IM, which we are reasonable sensitive too, it's just harder to define a single measurement for.

Anyway, most modern amplifiers don't vary much in frequency response into most loads.  You have to go to tubes, generally, to get something significantly different from a $300 receiver.  Or something very culty.

[Electrostatic speakers, and sometimes speakers with very complex crossovers, can show significant differences with many amplifiers.  That is why the David Clark Amplifier Challenge specifically excluded electrostatic speakers, and not because they often shut down amplifiers...that would merely invalidate a particular trial.]

While to be absolutely careful about it and avoid false positives, matching frequency response, for virtually all solid state amplifiers, into voice coil speakers, is probably not necessary to make it impossible to pass either ABX or prove a consistent preference.  Matching just at 1kHz is sufficient in most cases, and it was often done that way in many ABX demonstrations.  It generally comes close enough for most people, in a double blind test, to find what they thought was an easy difference into a difference very hard to hear at all, so hard one has to be careful instead of wild about making judgements based on fleeting thoughts.

I'm a proponent of the idea that blind AB Preference Testing is better than ABX, and similar statistics can be applied to determine if the preference has been statistically verified, or could be chance.

One of the problems, however, is that many audiophiles will see the raw numbers as some kind of confirmation of their earlier views, despite having virtually no statistical significance.  And the second is (as I found in the very first double blind test I performed in 1983) the subject may argue that their preference changed...in such a way as to make nearly all of their "identifications" consistent.

So ABX is a kind of way of forcing the situation to prevent these kinds of false arguments.  But if you don't fear these false arguments, preference testing is faster and requires only a relaxed judgement, the same as for actual listening.

Most of what people hear sighted is based on "thinking" of some kind.  People have to think of the words to describe what they are hearing.  And this thinking more freely ranges into the negative depending on endless other considerations (technically know as biases) when things are free to be criticized.  Like those complicated amplifers from Connecticut, for people opposed to complexity and what you might call the Eastern outlook, which I recall from my youth as the starched shirts, the greater formality, than I was used to in California.

BTW, a new Threshold has arisen, with new headquarters in Texas, not far from where I live.

This reminds me of how I myself was prone to criticize the tin ears of audio engineers, when I was a college student, just hearing the ways of subjectivist audio.  I picked up my first copy of Stereophile in 1975 and quickly became an avid reader of that and The Absolute Sound.

Anti-establishment, having arisen in the wake of the Vietnam War, touched everything.  It had become the natural outlook on everything.  So of course those engineers were wrong about feedback. No Feedback is better.  You can hear it yourself, when the equipment doesn't blow up when the feedback is removed...

Usually a lot louder too...

Some people, all of my audiophile friends, and all the audiophiles I know socially, are subjectivist in the sense of still having this outlook, that the audio engineering directorate doesn't get it, you have to go offstream to the likes of John Curl (a strong anti-ABX subjectivist who personally knows we still haven't solved the amplifier problem, which might surprise some because his products have been objectively excellent as well, so a perfect example of a good engineer who is also a strong subjectivist) to get your best amplification (most of the amplifiers I own today are Curl amplifiers, which keep working and are never bad sounding, not because Curl has special lock on what makes amplifiers sound good).

But I no longer see it as just the Engineering directorate, it's the more credible group of audiophiles as well, those who have seriously considered the issue, done well designed blind tests on themselves, and/or understand the nature and consequences of perception.  And audio scientists and so on.

I should perhaps disclose also that I myself R an engineer now too, as a member of two engineering societies involved in Audio.

Without double blind (and level matched) testing your findings are worthless, and without statistical verification they are just one useless data point.  Most audio tests are at the worthless level.

Worthless tests are fine, as an amusement.  But it goes wrong, when people come to feel inadequate.  Then it's audio nervosa.

The creed of audiophiles should be much like that of radio amateurs.  An audiophile should try to bring greater pleasure into the lives of other audiophiles.

That's tricky.  I don't think it means, necessarily, that we should lie.  Mostly less verbosity, and occasional avoidance of telling the whole truth, as perceived perhaps, and there's great justification in that.

But I think generally it doesn't involve black vs white comparisons.  It doesn't involve forcing others into one's own agendas, perspectives, or even practices.  It involves listening to others and trying to appreciate things from their perspective, what they are reaching for, and the ingenuity of their solutions so far.

Anyway people learn best when they want to learn, when they are at the point of sincerely wanting to know something enough that they ask.  Before that point, advice is merely poor salesmanship or street fighting.