[bigshot]

If there are harmonics in the audible range they'll be reproduced along with all the other frequencies in the audible range. Super audible frequencies can't be heard by human ears. They add absolutely nothing to the quality of recorded music. They are as useless to humans as teats are to a bull hog. And if super audible frequencies are present in recorded music, they can only hurt, never help. More info on that in the article in my sig, "CD Sound Is All You Need".

 

[TheSonicTruth]

Yes - it can. To be precise - THEY can. Say overtones at 41 and at 43 kHz - the difference in this case being 2 kHz - definitely audible.

And will be missing in recording>equipment that goes only to 20 kHz.

So considering the above, could the presence/absence of that 2kHz differential tone in a given song affect the emotional/subconscious reaction to that song? And secondly, is that why some analog fanatics claim analog superiority to digital?

 

[Zapp_Fan]

So considering the above, could the presence/absence of that 2kHz differential tone in a given song affect the emotional/subconscious reaction to that song? And secondly, is that why some analog fanatics claim analog superiority to digital?

Very doubtful. IMD tones in free air are very different than IMD in recorded audio, because acoustic interferences are 3-dimensional and in all but the worst acoustic spaces, are very chaotic in comparison. IMD in recorded audio can be made very obvious if you do it right (wrong), but in an acoustic space, it happens by interferences of physical waves in physical media that are all interacting with each other at once, so it's rare to hear very distinct under or over-tones that are produced by interferences between two fundamentals, let alone harmonics. At any rate, when they occur, they are of a markedly different character than IMD that occurs in an electrical/digital signal or transducer.

Now, if the 43-41Khz 2khz "undertone" were to occur in the acoustic space where the recording happened, you would probably not hear it even if you were there in person. Firstly because ultrasonic frequencies tend to propagate very differently around a room, secondly because they tend to get mostly absorbed instead of bouncing around the room like lower frequencies, thirdly because the interferences producing those tones would be chaotic, transitory, and localized at very small places around the room. Likewise, if they are there, they will be poorly captured by normal recording equipment.

Using ultrasonic audio to acoustically produce audio is a known thing, it's sometimes used to "beam" sound from one point to another in specific applications. It does happen and can be controlled in real life. But this is using very high amplitude ultrasonic transducers, not guitars or drums or pianos.

However, I think acoustic interferences of ultrasonics produced by real instruments in free air is among the very last things any recording engineer would ever care about, therefore there is nothing to be gained by trying to reproduce them at home.

And if the IMD happens after the recording phase, as Analogsurviver points out, it should be considered a bad thing and not something you want to reproduce at all.

 

[Glmoneydawg]

So considering the above, could the presence/absence of that 2kHz differential tone in a given song affect the emotional/subconscious reaction to that song? And secondly, is that why some analog fanatics claim analog superiority to digital?

I love my records.Some of them even sound better than cd's(mastering issues I'm sure)If any of my cartridges where flat out to 43k i would have check with my Boston Terrier for confirmation.

 

[castleofargh]

I love my records.Some of them even sound better than cd's(mastering issues I'm sure)If any of my cartridges where flat out to 43k i would have check with my Boston Terrier for confirmation.

that's when you notice that our preconceptions influence even our experiments. as I don't care about ultrasounds, I usually don't bother with animals. well except for the Bass(fish), because somehow the name in English confused me when I started this hobby. then I learned that the line thingy they have on the sides detect vibrations and are most effective in the low frequencies up to about 100hz(it's fun cause it's true). so from that time I always controlled the low end of my gears with a Bass int the freezer.

 

[bigshot]

Using ultrasonic audio to acoustically produce audio is a known thing, it's sometimes used to "beam" sound from one point to another in specific applications.

I remember a Gilligan's Island episode where they could hear [/SIZE]radio transmissions using the fillings of his teeth.

 

[pinnahertz]

Well, in "Picking capacitors" slew reate may have been mentioned ( don't care to check it, 3 metres away in the library ) - but VERY briefly. Since it was also a thinly disguised advertisement for what later became known as Wonder Cap, there was no warning saying film caps can have VERY large differences in slew rate. From just a few (single digit ) volts per microsecond to aprox 600 ( at least back at the article publishing, but is not much different today ).

What's clear about this is that you don't understand what slew rate is, and why a capacitor of any reasonable quality cannot limit it. But go ahead, cite a reference about a coupling cap limiting slew rate. I will not be holding my breath.

I cannot have a technical discussion with you when you refuse to be specific. You've been asked for specifics, that means capacitance value, dielectric type, voltage rating, DA, DF, ESR, the whole lot. Nothing! There's no point in continuing this.

Regarding differences made in production; IF that first amp heard at friend's contained "vintage, definitely in limited supply, not to be available again- ever" caps of superiour quality, and the one bought later with whatever best available from the current production, but inferiour to vintage caps - you WOULD mind. That's why manufacturers avoid such situations at all costs - and build with whatever is currentlyproduced and likely to remain available in at least few years ahead. That means an old(er) design may well be, ultimately, superiour sounding for this very reason.

You have not stated clearly why vintage caps are better, only that you don't like the new ones. And, I expressly stated that "sounding better" doesn't count!

With the miniaturization and advent of SMT /SMD, unfortunately MOST of the best capacitors once used in trough hole variety have been dropped. Simply because they can not withstand the temperatures of the SMD/SMT soldering process - further made worse with the introduction of the lead-free solder ( which requires yet even higher temperature ). Only the most THERMICALLY robust varieties of film capacitors remain available in SMD form - and those are NOT the equal in quality of say polystyrene as dialectric material. And, yes, capacitors DO have different sound quality - even IF the electrical measurements, down to the last DA and whatever electrical parameter ( triboelectrics, anyone ? NO mention in "Picking Capacitors" , THE fatal flaw of teflon caps ... ) are identical.

That's your opinion, and we all get that. But you have not supplied any proof at all!
When you say "capacitors DO have different sound quality - even IF the electrical measurements...are identical." what you're saying is that even if there is no difference in the resulting wave form parameters, time, energy, frequency (spectrum), there is an audible difference. And that the cause of that difference cannot be measured either! That's where we part ways. That's impossible, and I challenge you to prove it against all electrical and scientific data in opposition.

Now try to figure this one out ...

I do know how a DC servo works. And did post its use at the input as the ultimate no-no.

No, you don't, because a DC servo isn't applied to the input!

Sooo - if your preamp input is directly coupled to a phono cartridge AND there is a catastrophic failure of preamp ( one of the rails blown, latching to whatever remains, blown input semiconductor, etc, etc ) - would you still insist on direct coupling ( best for sonics, no doubt about that ) - or would you insure yourself from blowing up an expensive cart by using ( best quality you can get ) cap ?

No, you have no idea how a DC servo works or you wouldn't be making the above statement. It's absolutely NOT a risk to the input device!

No, that scenario is not stupid. Because it does SOUND better than having to insert another gain stage between the preamp input and - call it that way - driver of the "power amp". And yes, although not +-50 V rails preamp, it was close : +- 45V rails. But, I agree it is not the normal way of doing things and incompatible with most other equipment.

You are , of course, free to think this way. But I ( or anybody else ) can not convince to the contrary anybody - online. Only on the real demo floor, with real equipment of high enough quality.

But your opinions defy science and electronics in specific. You've fallen back to the old audiophile crutch once again: "You have to hear it on equipment of high enough quality". Anyone who disagrees doesn't have equipment of high enough quality. Do you not see how ridiculous that makes you look?

Yes - because a SINGLE "20-20K" minded component in the entire audio chain, from the microphone to the whatever used as the end transducer back to sound, is enough to throw the spanner into soundstage works. Now - go to any studio - and count 'em ...

You can claim "soundstage" all you like, but it remains undefined, and not clear. We simply cannot use that term in a real scientific discussion because it has no definition.

No, I do not want the whole chain to play up to 100MHz - because I am realist. It is unlikely to happen - ever, but certainly not in my lifetime. However, ANY stage that can be made large bandwidth will contribute to more accurate recording and reproduction. And I have NEVER said that < 20 Hz and > 20 kHz is more important than 20 - 20K, or even more important 100 to 10K band. Only if this core band is OK, would I try to stretch to either of the extremes. As the easiest way to start doing it is in amplifiers, I started with them.

Once again, we're into the absurd. So we have a chain limited to 20-20kHz. And now you claim that one device that is say 10-100kHz improves everything? What's it improving if there's nothing there in the first place? Take this example: a full bandwidth audio system (any definition, even yours) has a 3kHz low pass filter inserted in it, which is clearly audible to everyone. Now, you insert, anywhere, an amplifier with flat response to 100kHz, or heck, 1mHz. With your warped logic, that causes an audible improvement. How? Does it somehow undo the 3kHz LPF? This is simply idiotic.

There are microphones made specially for music that go to 100 kHz.

And they are impractical for most applications. It doesn't matter, nobody's using them. You've again focussed on the rare example, citing it as typical. It's not.

As I type this, I am lidstening to...

Nobody cares what you're listening to. Know why? Because you don't share it! It is, therefore, a lie and a myth.

However, I did not know clarinet has so much output above 20 K - at very least to 40 kHz, during the solo clarinet passage. One learns every day...
And the beat box used by the percussionist goes to 50 kHz on this recording - some at least 5 metres from the mike; clarinet between 2-3 metres, depending on player's position on the stage.

Again, if you're going to talk about energy without talking about it's intensity and level, you're not working with reality. The statements above are meaningless.

I agree there is no such thing as zero tolerance parts -

But that's exactly what you called them! It can't be both!

but the best approximation, as most probably used for the AGI, is some Hewlet Packard RLC meter. OK, " measured to be exactly 123456 ohms, with the tolerance from the absolute correct value as specified by XY measurning instrument" - would that satisfy you ?

No! That's not how things are designed, especially if put into production.

And, yes, the values of resistors in RIAA circuit are printed down to the last ohm - like 23732 ohms and NOT 24 K 1% ( or any other value/tolerance) . Caps are not - because, for those less familiar with film caps, even picking them up with anything more than feather touch can change their value - let alone re-printing them. For this reason, no desoldering and measuring the super precise ( aka close to zero tolerance ) parts from AGI 511 ( or anything else built with even approaching this quality of parts ) is NOT recommended.

Wrong! I've measured literally thousands of capacitors, all types and values. Not one has ever changed value by being picked up! Not one, unless it was defective.

No, I did NOT confuse kHz with MHz - correct as written. However, you are right, AGI 511 overall HF limit is set by its line stage, which is approx 100 kHz.

Then you were dead wrong by stating there were no RF signals below 90mHz. There are many, and some of the strongest you'll ever encounter. I cited exapmles.

You are correct in stating that whatever has been used in a studio has been way off the accuracy of the RIAA precision of AGI. Then again, it is not AGI's fault to try to strive for the precision - and studio folks being satisfied with sloppiness.

The attempt at that kind of precision is pointless when the rest of the system is at +/- 1dB or worse. Actually, far worse. There's simply no point at all.

I do not know which schematics of 511 you did find online. It is NOT a single op-amp - in addition, there are ( bipolar?- would have to check it ) transistors used as feed-forward - that's where 100 MHz+ capability of the circuit is coming from, not from the slow IC op-amp.

Here are the 511 schematics on line. You'll need a free account to download it. If you have other information POST IT!

I agree installation and grounding are paramount in any phono ptreamp. But none I am aware of can reject this much RF garbage as 511. Enough is to see the open box - plenty images online.

I've seen the pictures, it's nothing compared to the construction of audio equipment designed for use in the kind of RF fields encountered at broadcast facilities.

I disagree - there still is nothing comparable up to today.

I cited two examples of equipment from that era, direct competitors, that had comparable performance. The only difference is, they succeeded in the marketplace, and the 511 did not.

The only gripe I have with 511 is the use of that ribbon cable for all the connections - because that adds approx 80 pF of capacitance per each input, unfortunataly sometimes this is too much for some of the best MM phono cartridges - after the cabling from TT to preamp has already contributed at least 100 pF - if not more.

You're obsessing about preamp capacitive load when you should be concerned with the composite frequency response of the cartridge, tone arm, interconnecting cable, and preamp. Nothing will ever come close to your claimed RIAA accuracy of the 511, and it won't be a problem with C loading.

 

[pinnahertz]

Yes - it can. To be precise - THEY can. Say overtones at 41 and at 43 kHz - the difference in this case being 2 kHz - definitely audible.

No! If the system is linear the combination if any two frequencies will not intermodulate and produce other products. It takes at least some nonlinear response to cause intermodulation, and it is the degree of that nonlinearity that produces a degree of intermodulation. Mostly, IMD in todays analog gear is a non-issue in anything other than a dynamics processor.

And will be missing in recording>equipment that goes only to 20 kHz.

No, it won't, if that equipment is nonlinear at the frequencies involved. You'd be surprised how many digital devices intermodulate signals near and above 20kHz. The good ones don't, of course.

 

[Kammerat Rebekka]

I'm not sure you can use science in order to explain anything to people who have already made up their minds about these matters. This thread surely is proof of that. Then again I have yet to see anyone's mind being changed over the internet, regardless of logic and fact dropping. People stick to their guns - especially when they've put a lot of money into the shooters.
I do however highly applaud the vehement attempts at pulling this thread back down to earth with knowledge that seems to stem from folks who've been working with electronics in real life ever since Nam. Maybe it's just me but I tend to trust the professionals over the laymen. Similarly I wouldn't want to call a holistic tunnel cleanser if my toilet decides to snuff it.

A little update with regards to my newly aqcuired headphone: it sounds wonderful out of my little Cowon dap and maybe just a tad better out of my myryad z40 amp but I couldn't say for sure.
The supposed power requirement of this headphone among most audiophiles seems warped imho.

 

[Yuurei]

If there are harmonics in the audible range they'll be reproduced along with all the other frequencies in the audible range. Super audible frequencies can't be heard by human ears.

One of my friends thinks that there are harmonics that are not audible, but they make you feel that music indeed sounds better and he is dead serious about it. Apparently it only happens on high-end equipment. Or very expensive, I'm not sure

 

[Kammerat Rebekka]

I have to admit that I too have used the word soundstage in order to communicate x can to x dude. Nowadays I think it has more to do with a spike in the higher frequencies that generates 'space' in music. Couple that together with something large and very open around your ears and it feels like the stage got much bigger...which it did in effect but not in an unobtanium-like manner.

Edited for blabber.

 

[pinnahertz]

So considering the above, could the presence/absence of that 2kHz differential tone in a given song affect the emotional/subconscious reaction to that song?

Unlikely as there is a lot of actual content around 2kHz that would mask all but the most severe intermod products.

And secondly, is that why some analog fanatics claim analog superiority to digital?

There is some validity to this, but not specifically for intermodulation of ultrasonic frequencies, and not as a means to define analog preference specifically. Rather, there is evidence that frequencies in the top octave, from 10kHz to 20kHz, can, in some devices, intermodulate to create products in the upper mid band. Research into this was conducted by Deane Jensen and Gary Sokolich in their AES paper, "Spectral Contamination Measurement", Nov. 1988. They found they could generate a test signal comprised of many closely-spaced tones (120Hz apart, 10kHz -25kHz was one example), and with careful analysis, detect intermodulation products. However, attributing this condition to band-limited systems is not conclusive, especially today. What is important to note is that the paper makes no correlation between the degree of spectral contamination and audibility. Also, there was no correlation between types of technology (tubes vs SS, analog vs digital). A tube amp tested far worse than a SS amp, an analog tape recorder tested far worse than a digital recorder, etc.

So the answer is, possibly under certain conditions, but more study is needed.

 

[71 dB]

Yes - it can. To be precise - THEY can. Say overtones at 41 and at 43 kHz - the difference in this case being 2 kHz - definitely audible.

And will be missing in recording>equipment that goes only to 20 kHz.

Why not have those 2 kHz frequencies in the original recording? Why put instead 41 and at 43 kHz signals and have your non-linear ( = crappy) reproduction gear generate those 2 kHz signals for you? The human hearing range is 20 Hz - 20 kHz. Put whatever* you want inside that frequency range and have as linear as possible reproduction chain to deliver that without arbitrary uncontrolled distortions.

* You can even have an 192 kHz recording with 41 and 43 kHz frequences, filter ultrasonic 20 kHz content and feed it to the IMD module (hardware or software) and mix all the < 20 kHz distortion products to the lowpass filtered content you put on a CD. Then you have that audible IMD garbage on your 16/44.1 audio format if you want it there for some reason. And it's controlled when payback is done with an audibly transparent system.

 

[gregorio]

[1] Place in series as many "20-20k" minded components as required in a typical studio workflow - and it is a miracle in itself whatt little of soundstage that does come trough - at all.
[1a] However, I did not know clarinet has so much output above 20 K - at very least to 40 kHz, during the solo clarinet passage. [1b] One learns every day...
[2] presto, a constant peak of 19.0 kHz at - 66dB crept in ... and I dread converting DSD to PCM and looking at what has been switched on during the second part of the concert - EVERY damn light in sight dimmed - yuck...!
[2b] Yes - it can. To be precise - THEY can. Say overtones at 41 and at 43 kHz - the difference in this case being 2 kHz - definitely audible. .. [2c] And will be missing in recording>equipment that goes only to 20 kHz.

1. There's very little soundstage above 12kHz and virtually none at all by about 16kHz. Which is why it doesn't really matter that analogue distribution media is so inaccurate above 16kHz and why 44.1kHz digital, which is flat to 20kHz, is more than enough. What's really a "miracle" is that you proclaim yourself a recording engineer with many years of experience but not only haven't you discovered this simple fact, you somehow seem to have convinced yourself of the complete opposite of the actual facts. However, maybe your "miracle" of misunderstanding and undiscovered facts is believable, in light of your next quoted statement ...
1a. You're joking right? If you are an experienced recording engineer, how could you NOT know this? Most acoustic instruments have significant output above 20kHz! All orchestral woodwind instruments (flutes, clarinets, oboes and bassoons) have metal key mechanisms which produce clearly audible noise (when listened to or mic'ed closely) and that mechanical noise has a spectra well into the ultrasonic range. In addition to mechanical noise there is also reed buzzing and breath sounds (particularly in the case of the flute) which extend will into the ultrasonic range. With many string instruments we can also have some mechanical noise but also commonly fret noise (a particular problem with acoustic guitars for example), which again extends well into the ultrasonic range. Even the human voice produces a lot of ultrasonic content, breath sounds and lip smacks for example. How is it possible not to know these basic facts even with minimal student level recording experience, let alone the many years of profession experience you claim? It's a "miracle"!
[1b] Indeed and at your rate, you should have a decent understanding of all the basic/obvious facts in a couple of centuries or so!

2. Why do you have to convert DSD to PCM and "look" to see what has been switched on? Why don't you just play the DSD as is and listen? And, I've never seen a light/dimmer produce just a single peak, they always produce a harmonic series (mostly odd harmonics). Obviously you didn't notice that because you can't hear those harmonics either.
2a. Hang on, what sort of alien are you? You can (presumably) hear roughly the normal human range but you can't hear 19kHz and 41kHz and 43kHz are "definitely audible". AFAIK, there's no creature on this planet which has a hearing response like that, ergo you must be an alien!
2b. If, on the other hand, you are saying that the 2kHz signal has already been produced as a by-product of IMD, then why would equipment capable of recording up to 20kHz not be able to record this 2kHz signal?

So, either you're an alien (with a bizarre hearing response) OR you're taking the ludicrous position that a 20kHz system cannot record a 2kHz signal. I suppose it's possible that you're both an alien AND completely ignorant of recording/audio but my money is firmly on just the latter!!

The first type of NATURALLY occuring IMD is welcome - it is, after all, what we are exposed whenever in air and sound foeld, that is to say always. ...[1] The second is an enemy - and needs to be avoided. [1a] That is why competently performing electronics above 20 kHz are required in the first place.
[2] In case you have not moticed by now, I am trying to balance everything into something that , in the end, sounds better.

1. That will come as a bit of a shock to those who like rock music but OK if you insist, no electric guitars, vintage cabs, compressors or other unnatural occurring IMD.
1a. Hang on. If all that unnatural IMD "needs to be avoided" and recording systems which only go to 20kHz can't record that audible unnatural IMD, then that would be a great reason for using a 20kHz system "in the first place". You are contradicting yourself!

2. How is it even possible for anyone to have noticed that? "In the end" you've not only demonstrated little/no idea of what sound is but actually evidenced that you think horrendously bad sound (a soundstage which jumps all over the place) is a superb example of "good" sound. The ONLY thing that's really noticeable from your posts is that you're trying to "balance" ignorance, fantasy and delusion, and you're achieving that by ignoring the actual facts and making up complete nonsense! What's really surprising is the sheer number of times your false statements have been shown to be made-up nonsense but rather than recognise how foolish that makes you appear, your response is just to make-up another whole load of nonsense. A vicious circle of ever more foolishness, ad infinitum ... eventually passing the point (long ago) of a level of foolishness that's not even credible/imaginable. It's literally "incredible", a true miracle!!!

G

 

[analogsurviver]

What's clear about this is that you don't understand what slew rate is, and why a capacitor of any reasonable quality cannot limit it. But go ahead, cite a reference about a coupling cap limiting slew rate. I will not be holding my breath.

I cannot have a technical discussion with you when you refuse to be specific. You've been asked for specifics, that means capacitance value, dielectric type, voltage rating, DA, DF, ESR, the whole lot. Nothing! There's no point in continuing this.
You have not stated clearly why vintage caps are better, only that you don't like the new ones. And, I expressly stated that "sounding better" doesn't count!
That's your opinion, and we all get that. But you have not supplied any proof at all!
When you say "capacitors DO have different sound quality - even IF the electrical measurements...are identical." what you're saying is that even if there is no difference in the resulting wave form parameters, time, energy, frequency (spectrum), there is an audible difference. And that the cause of that difference cannot be measured either! That's where we part ways. That's impossible, and I challenge you to prove it against all electrical and scientific data in opposition.
No, you don't, because a DC servo isn't applied to the input!
No, you have no idea how a DC servo works or you wouldn't be making the above statement. It's absolutely NOT a risk to the input device!
But your opinions defy science and electronics in specific. You've fallen back to the old audiophile crutch once again: "You have to hear it on equipment of high enough quality". Anyone who disagrees doesn't have equipment of high enough quality. Do you not see how ridiculous that makes you look?
You can claim "soundstage" all you like, but it remains undefined, and not clear. We simply cannot use that term in a real scientific discussion because it has no definition.
Once again, we're into the absurd. So we have a chain limited to 20-20kHz. And now you claim that one device that is say 10-100kHz improves everything? What's it improving if there's nothing there in the first place? Take this example: a full bandwidth audio system (any definition, even yours) has a 3kHz low pass filter inserted in it, which is clearly audible to everyone. Now, you insert, anywhere, an amplifier with flat response to 100kHz, or heck, 1mHz. With your warped logic, that causes an audible improvement. How? Does it somehow undo the 3kHz LPF? This is simply idiotic.
And they are impractical for most applications. It doesn't matter, nobody's using them. You've again focussed on the rare example, citing it as typical. It's not.
Nobody cares what you're listening to. Know why? Because you don't share it! It is, therefore, a lie and a myth.
Again, if you're going to talk about energy without talking about it's intensity and level, you're not working with reality. The statements above are meaningless.
But that's exactly what you called them! It can't be both!
No! That's not how things are designed, especially if put into production.
Wrong! I've measured literally thousands of capacitors, all types and values. Not one has ever changed value by being picked up! Not one, unless it was defective.
Then you were dead wrong by stating there were no RF signals below 90mHz. There are many, and some of the strongest you'll ever encounter. I cited exapmles.
The attempt at that kind of precision is pointless when the rest of the system is at +/- 1dB or worse. Actually, far worse. There's simply no point at all.
Here are the 511 schematics on line. You'll need a free account to download it. If you have other information POST IT!
I've seen the pictures, it's nothing compared to the construction of audio equipment designed for use in the kind of RF fields encountered at broadcast facilities.
I cited two examples of equipment from that era, direct competitors, that had comparable performance. The only difference is, they succeeded in the marketplace, and the 511 did not.

You're obsessing about preamp capacitive load when you should be concerned with the composite frequency response of the cartridge, tone arm, interconnecting cable, and preamp. Nothing will ever come close to your claimed RIAA accuracy of the 511, and it won't be a problem with C loading.

I will return to answer the rest later - but only the most critical answer right now.

Damn... are you living on this planet ?

If you EVER - repeat - EVER - measured a truly decent MM phono cartridge, you would have known exactly WHY I am "obsessing" about the preamp capacitive load. Cartridge does not give a damn where the capacitive load does come from - all it can see is the capacitance COMBINED - from each and every part of the signal path, from cartridge connecting clips to the actual input electrical active element on the circuit board ( or hard wired in better tube equipment ).

Now, do tell me WHY do ALL manufacturers of truly decent high ( normal ) output MM carts ( sensitivity around 1mV/cm/sec ) dabble with capacitive load ( usually specified as 100-200 pF ) ? To ease off this nasty hook the manufacturers of preamps, perhaps ? Themselves producing preamps with more input C than healthy for their own carts ?

If you ever did work with any of those, you would have known that they react badly with anything above 100 pF or so - throwing away practically everything fought so hard and gained at such a great co$t as compared to their mechanically much more robust/much less costly siblings of the very same principle of operation - because of those damn additional 100 or so pF in the phono preamp ? If it is only an additional capacitor across the input and can be removed, OK - if that capacitor is imbedded into the circuit as in cable used by AGI, you can not do anything about it. I can beat the otherwise superb 511 for application necessitating as low capacitance input as possible for top flight high output MM cartridges with this : https://www.phonopreamps.com/tc750pp.html All it takes is desoldering the 220pF caps at the input ... - and a decent cabling/grounding from TT to preamp. The resulting response WILL be better - incomparably so - despite TCC-750 having far inferiour RIAA tolerance compared to AGI. TCC-750 can be upgraded for both quality and tolerance of the parts, as well as power supply ... and then it is hard to beat, even with carts that do require higher capacitive loading, which can be always easily added with a Y RCA connector and required capacitance (or resistance) soldered into a RCA male plug - no additional soldering required.

It is easy to add additional capacitance, if required - as well lowering the resistive load, both by parallel connection of either cap and/or resistor. The other way around is impossible. So, it is always sound to have the resistive load higher than nominal 47 kohm ( at which VERY FEW real world cardridges perform at their best anyway) and as low capacitance as possible - you can always add both caps and resistors in parallel to the input, as required by the cartridge in question. There were/are commercially available preamps done just like that - because MM cartridge loading is more important than RIAA accuracy - the error(s) in loading have FAR greater effect than say half a dB error in RIAA response AT THE EXTREMEs - but not in the midrange and around the poles https://en.wikipedia.org/wiki/RIAA_equalization, where errors as low as 0.1dB can be audible.