Interesting design philosophy from SoulNote!

[Retro_Pete]

Note:
It is completely unclear what kind of analyzer we are talking about. Until now, no correlation has been established between sound quality and any parameter of an audio frequency amplifier. Measuring devices such as Audio Precision are used at the end of the assembly line as a"go/no-go" tester, nothing more.

The claim that measurable parameters of an audio amplifier do not correlate with sound quality contradicts well-established principles in audio engineering. Tools such as Audio Precision analyzers are not just "go/no-go" testers but precision instruments used to optimize and fine-tune audio equipment. These analyzers provide valuable insights into key performance metrics like Total Harmonic Distortion (THD), Intermodulation Distortion (IMD), and frequency response -parameters that are scientifically linked to perceived sound quality. Research consistently supports the idea that lower distortion and a flatter frequency response contribute to more accurate and pleasing audio reproduction, as demonstrated in numerous studies and publications from institutions like the Audio Engineering Society (AES) and Journal of the Audio Engineering Society.

https://en.wikipedia.org/wiki/Propagation_delay
Propagation delay

Propagation delay is the time duration taken for a signal to reach its destination. It can relate to networking, electronics or physics.


Note.
In a number of modern high-speed op amps, this parameter is specified as time Propagation Delay and is measured at a gain of 1 or (-1) between the input and output when the signal crosses zero.
As for audio amplifiers, two types of delays can be distinguished here:
-propagation delay from the amplifier input to the load (this parameter determines the vector errors of the amplifier and distortions associated with the rotation of the fronts (slopes) of signals during the attack, decay, and more);
-propagation delay between the input RF filter (if used) and the output inductance (if used). This time determines the NFB response speed and the switching distortion suppression in class AB amplifiers that depends on it.

The propagation delay in op-amps and audio amplifiers, particularly regarding the input-to-load delay and feedback delay, is a well-understood phenomenon in audio engineering. These delays can lead to measurable distortions and reduced signal fidelity if not properly managed, which is why amplifier designs carefully control these parameters to minimize any negative impact on audio quality.
Dynamic measurements such as impulse response, FFT analysis, and group delay testing are essential for identifying issues like switching distortion and group delay, which affect the transient response and overall sound quality of audio amplifiers.
it's very much an issue that is taken care of by design.
To mitigate these problems, manufacturers employ advanced circuit design, fast op-amps, optimized feedback loops, and high-quality power supplies.

 

[Petrov-Al]

https://audiokarma.org/forums/index...hilosophy-from-soulnote.1065569/post-17365176

FiddlerBeeps​

«Appendix: As previously menti oned, rotati ons occur in surface waves of fluids under gravity as a result of sheer forces created by the slopeof the fl uid's surface inducing horizontal fl ow. They do not occurin the slopes of electronic signals.»

Weare talking only about the rotation of the fronts, mainly in the attack and decay of signals (and not only) and responsible for the naturalness of the timbre, for microdynamics. And they depend on the time Propagation Delay of the analog amplifier. This has nothing to do with the waves in the air or on the water.
I hoped that this would be clear from Figure 1 of the appendix.

"InFig. 4, counter clockwise rotating signal fragments are highlighted in blue, clockwise rotating
signal fragments are highlighted in pink." and "Distortion products are nothing more than a lag from the real signal, i.e.signal loss."

«Areyou quite certain you're not depicting phase and memory effects?»

As for the manifestation of memory effects, let's consider this using the example of a model with a different bandwidth at the bottom of the audio range.

 

[Petrov-Al]

Retro_Pete, if everything were as you write, then the problem of measuring distortions correlating with sound quality was solved more than 70 years ago. However, even at that time it was already clear that neither THD nor IMD correlate properly with sound quality. Therefore, since then, everyone has tried to find new testing methods. And quitea few have been invented:
- multi-tone signals;
- a mixture of a rectangular signal with a sine wave (DIM-100);
- 1/3 octave noise (white noise, pink noise or pseudo-noise) is fed to the amplifier input, and at the output it is cut out and the distortion products remaining as a result of cutting are analyzed;
- measuring distortions using the inverse intermodulation method (RIMD);
- this is also the SWDT Hafler method;
- this is also the Sapozhkov-Baksandall distortion measurement method.
And this is far from a complete list of new methods, attempts to find a correlation with sound quality. As a result, many developers have abandoned the use of NFB in their developments despite the excellent parameters obtained with its help.

 

[Retro_Pete]

Retro_Pete, if everything were as you write, then the problem of measuring distortions correlating with sound quality was solved more than 70 years ago. However, even at that time it was already clear that neither THD nor IMD correlate properly with sound quality. Therefore, since then, everyone has tried to find new testing methods. And quitea few have been invented:
- multi-tone signals;
- a mixture of a rectangular signal with a sine wave (DIM-100);
- 1/3 octave noise (white noise, pink noise or pseudo-noise) is fed to the amplifier input, and at the output it is cut out and the distortion products remaining as a result of cutting are analyzed;
- measuring distortions using the inverse intermodulation method (RIMD);
- this is also the SWDT Hafler method;
- this is also the Sapozhkov-Baksandall distortion measurement method.
And this is far from a complete list of new methods, attempts to find a correlation with sound quality. As a result, many developers have abandoned the use of NFB in their developments despite the excellent parameters obtained with its help.

We are not in disagreement here, and I don't dispute that modern measuring techniques provide more precise insights into distortion characteristics than older methods.

However, I do question the relevance of the issues these methods uncover once they fall below certain thresholds. As early as fifty years ago, amplifiers could be designed with distortion and noise levels that were effectively inaudible.

Many amplifiers from fifty years ago already achieved noise and distortion levels below these thresholds in regular use. While the precision of tools like spectrum analyzers has improved over the decades, the ability to design amplifiers with inaudible noise and distortion was already well-established by that time.

Here's a short list of quality transparent amplifiers from the seventies:

Spoiler: List

• Marantz Model 500 (1974)
  • THD: <0.1%
  • SNR: 100 dB
  • Frequency Response: 10 Hz - 20 kHz, ±0.5 dB
• McIntosh MC2300 (1971)
  • THD: <0.25% at full power, <0.1% at typical levels
  • SNR: ~95 dB
  • Frequency Response: 20 Hz - 20 kHz, ±0.1 dB
• Accuphase P-300 (1974)
  • THD: <0.05%
  • SNR: >100 dB
  • Frequency Response: 5 Hz - 100 kHz, +0, -1 dB
• Phase Linear 700B (1974)
  • THD: <0.1%
  • SNR: ~95 dB
  • Frequency Response: 20 Hz - 20 kHz, ±1 dB
• Quad 405 (1975)
  • THD: <0.01%
  • SNR: 90 dB
  • Frequency Response: 10 Hz - 20 kHz, ±0.5 dB
• Sansui BA-5000 (1975)
  • THD: <0.1%
  • SNR: ~105 dB
  • Frequency Response: 5 Hz - 50 kHz, +0, -1 dB
• Yamaha B-1 (1974)
  • THD: <0.005%
  • SNR: ~100 dB
  • Frequency Response: 5 Hz - 100 kHz, +0, -1 dB
• Kenwood L-07M (1979)
  • THD: <0.008%
  • SNR: >110 dB
  • Frequency Response: 1 Hz - 300 kHz, ±1 dB

 

[Petrov-Al]

The following model test demonstrates distortion in each part of the signal: attack, steady state and decay

 

[Petrov-Al]

there is a thread where many models of those years are listed and each owner praises "his mare"
https://audiokarma.org/forums/index.php?threads/the-greatest-amps-circa-1977-80.406531/
But that doesn't mean they're all that good in reality.

I know a thread where colleagues have been looking for tests that would correlate with sound quality for over 10 years, but have not found any. Here's the conclusion:
https://www.diyaudio.com/community/threads/sound-quality-vs-measurements.200865/post-2856456
jan.didden

… it will always be possible engineering wise to build a 'better' amp and I expect that other measurements may be required to proof that one amp is better than another, but I have no specific answer as to which measurements this might be.

 

[Retro_Pete]

there is a thread where many models of those years are listed and each owner praises "his mare"
https://audiokarma.org/forums/index.php?threads/the-greatest-amps-circa-1977-80.406531/
But that doesn't mean they're all that good in reality.

I know a thread where colleagues have been looking for tests that would correlate with sound quality for over 10 years, but have not found any. Here's the conclusion:
https://www.diyaudio.com/community/threads/sound-quality-vs-measurements.200865/post-2856456
jan.didden

… it will always be possible engineering wise to build a 'better' amp and I expect that other measurements may be required to proof that one amp is better than another, but I have no specific answer as to which measurements this might be.

Your premise has a fundamental flaw. Measurements such as Total Harmonic Distortion (THD), Fast Fourier Transform (FFT), noise (N), Intermodulation Distortion (IMD), and multitone tests are objective metrics that evaluate an amplifier's signal amplification accuracy and quality.

When comparing two amplifiers, these measurements can identify which one objectively performs better. In this context, the amplifier with superior metrics would be considered the "better" amplifier.

However, if an amplifier with slightly worse measurements already has noise and distortion levels below the threshold of audibility and maintains a flat frequency response, a blind listening test likely wouldn’t reveal a noticeable difference. This suggests that, for practical listening, the measurable superiority of one amplifier may not lead to a perceptible improvement in sound quality.

Now, you’re suggesting that additional measurements might be needed to “prove” one amplifier's superiority over another. But we already have these measurements, and we know that blind testing is essential if there’s a claim of audible difference. If a blind test does reveal a genuine difference, it will likely correspond to something in the measurements we already use. However, you seem to question whether these measurements are sufficient to explain all perceivable differences -is that right? And do you have any examples of differences that the current suite of measurements can’t account for? You must have some in mind; otherwise, what’s the reason for pursuing this?

 

[Petrov-Al]

Retro_Pete: «Yourpremise has a fundamental flaw. Measurements such as Total HarmonicDistortion (THD), Fast Fourier Transform (FFT), noise (N),Intermodulation Distortion (IMD), and multitone tests are objectivemetrics that evaluate an amplifier's signal amplification accuracyand quality.»​

Experienced music lovers choose amplifiers not by parameters, but bysound quality. Above I have given a model of an amplifier withTHD=0.0013%.
Here is its IMD. Do you think IMD = -100 dB (0.001%)is enough for distortion products to be below the hearingthreshold?
Above I have given a test for distortion in attack anddecay. The level of distortion products reaches 130 mV regardless ofthe signal amplitude. Let's assume that we take -50 dB (1/300 Vout =100 mV) as the level of microdynamics signals. As you can see, thelevel of distortion products, which are not determined by any othermethod, exceeds the useful signal.
I worked on the development ofaudio amplifiers. During my entire career, I have never encounterednoise (N) problems. You yourself have given a number of amplifierswith SN no worse than -95 dB. No multi-tone signal tests wereperformed, and where have you seen such a parameter given in theamplifier's data sheet? A standard set of tests was performed: THD(at different frequencies, including the dependence of THD onfrequency in the range of 20...20000 Hz), IMD, SR, transientcharacteristics on rectangular signals, operation in clipping mode ata frequency of 20 kHz, the nature of distortions at an output voltagefrom zero to clipping THD=10%, maximum output power. And, of course,listening with test disks is mandatory.

 

[Retro_Pete]

Retro_Pete: «Yourpremise has a fundamental flaw. Measurements such as Total HarmonicDistortion (THD), Fast Fourier Transform (FFT), noise (N),Intermodulation Distortion (IMD), and multitone tests are objectivemetrics that evaluate an amplifier's signal amplification accuracyand quality.»​

Experienced music lovers choose amplifiers not by parameters, but bysound quality. Above I have given a model of an amplifier withTHD=0.0013%.
Here is its IMD. Do you think IMD = -100 dB (0.001%)is enough for distortion products to be below the hearingthreshold?

Yes, I believe that an Intermodulation Distortion (IMD) level of -100 dB is below the threshold of human hearing. I have seen no other compelling evidence to counter.

Above I have given a test for distortion in attack anddecay. The level of distortion products reaches 130 mV regardless ofthe signal amplitude. Let's assume that we take -50 dB (1/300 Vout =100 mV) as the level of microdynamics signals. As you can see, thelevel of distortion products, which are not determined by any othermethod, exceeds the useful signal.

Which amplifier was this test conducted on, or was it a simulation? Is the effect audible when listening to music?

I worked on the development ofaudio amplifiers. During my entire career, I have never encounterednoise (N) problems. You yourself have given a number of amplifierswith SN no worse than -95 dB. No multi-tone signal tests wereperformed, and where have you seen such a parameter given in theamplifier's data sheet? A standard set of tests was performed: THD(at different frequencies, including the dependence of THD onfrequency in the range of 20...20000 Hz), IMD, SR, transientcharacteristics on rectangular signals, operation in clipping mode ata frequency of 20 kHz, the nature of distortions at an output voltagefrom zero to clipping THD=10%, maximum output power. And, of course,listening with test disks is mandatory.

Blind tests are essential. Have you conducted any blind tests where a noticeable audible difference was detected, only to later discover that it was caused by IMD, insufficient slew rate, or a low-pass filter limiting the bandwidth and preventing accurate reproduction of a square wave?

 

[Petrov-Al]

Retro_Pete, trust my production experience. First,I designed and optimized the correction of models of future amplifiers in the simulator. Then I designed the printed circuit board and assembled a prototype. I was often surprised that the parameters of real layouts were better than the parameters of the models in the simulator. Before testing, a test was carried out for resistance to reactive load using capacitors up to 2 μF.
Blind testing was carried out with the participation of both developers and listeners from the customer. To conduct such tests, I specially developed a stand on powerful relays that were controlled by a microprocessor using a remote control. Listeners voted for a red or green LED on the stand. No one knew which LED corresponded to the tested amplifiers.
In addition, the company had a testing specialist who stayed at work over night (the company was outside the city) to listen in complete silence to individual parts of the phonograms that not all amplifiers handled well.
p.s.
Which of the good-sounding amplifiers you mentioned has a specification that says it reproduces a square wave accurately? And what do you mean by that more specifically?

 

[Retro_Pete]

Retro_Pete, trust my production experience. First,I designed and optimized the correction of models of futureamplifiers in the simulator. Then I designed the printed circuitboard and assembled a prototype. I was often surprised that theparameters of real layouts were better than the parameters of themodels in the simulator. Before testing, a test was carried out forresistance to reactive load using capacitors up to 2 μF.
Blindtesting was carried out with the participation of both developers andlisteners from the customer. To conduct such tests, I speciallydeveloped a stand on powerful relays that were controlled by amicroprocessor using a remote control. Listeners voted for a red orgreen LED on the stand. No one knew which LED corresponded to thetested amplifiers.
In addition, the company had a testingspecialist who stayed at work overnight (the company was outside thecity) to listen in complete silence to individual parts of thephonograms that not all amplifiers handled well.

I trust your production experience -truly, I do. But that’s in the context of actual production. I imagine none of the prototype amplifiers with audible distortion during music playback ever made it to production, correct?

That said, I’m not referring to prototype amplifiers in development; I'm looking at amplifiers readily available on the market (then and now), where the vast majority don’t exhibit this audible distortion. If they did, I’d genuinely like to see the results of a blind test confirming that. So far, the blind tests I’ve seen comparing well-designed amplifiers have shown little, if any, perceptible difference.
Edit: And if they did, they can be explained by current measurements.

 

[Mart Mccall]

. So far, the blind tests I’ve seen comparing well-designed amplifiers have shown little, if any, perceptible difference.

And where there was a perceived difference, it could be explained by the measurement methods that already exist.

I confess that a lot of this thread content is above my pay grade but I know there's always going to be people looking for a USP to sell us a 'better mousetrap'.

 

[Retro_Pete]

And where there was a perceived difference, it could be explained by the measurement methods that already exist.

Exactly. I forgot that rather important part in the last post.

I confess that a lot of this thread content is above my pay grade but I know there's always going to be people looking for a USP to sell us a 'better mousetrap'.

-It's how the industry of HiFi works. Slowly the times are changing. Slowly..

 

[Petrov-Al]

That's the thing, not everyone can feel the difference. Many are satisfied with the quality of mp3 music, especially with a bit rate of 128. Only trained specialists or truly "golden-eared"people can truly appreciate the quality of sound. And then, some need time to listen to a specific amplifier in different genres, at different power... Not all amplifiers are equally omnivorous in terms of both acoustics and genres of music. Some may show them selves better with one acoustics, others with another. The same is true for genres. Therefore, sometimes people select amplifiers for the acoustics they have.

But we are talking about something completely different: what tests can be used to identify those distortions that are not detected by standard testing methods. And I showed with specific examples how the compensation testing method works.

 

[Retro_Pete]

That's the thing, not everyone can feel the difference. Many are satisfied with the quality of mp3 music, especially with a bit rate of 128. Only trained specialists or truly "golden-eared"people can truly appreciate the quality of sound. And then, some need time to listen to a specific amplifier in different genres, at different power... Not all amplifiers are equally omnivorous in terms of both acoustics and genres of music. Some may show them selves better with one acoustics, others with another. The same is true for genres. Therefore, sometimes people select amplifiers for the acoustics they have.

But we are talking about something completely different: what tests can be used to identify those distortions that are not detected by standard testing methods. And I showed with specific examples how the compensation testing method works.

That’s not really the main point, in my opinion. It’s easy to claim that some people (yourself included, I assume) can hear a difference between well-designed amplifiers. Proving it, however, is another matter entirely -and that evidence has yet to be shown here, or in any of the many other discussions where similar claims have been made by dozens of people. Often, the conversation just stalls, the topic shifts, or personal matters get brought in. Thank you for refraining from that, by the way. -So, back on track..

I understand you’ve presented a test that appears to measure something beyond what standard methods can. My question remains: does it matter? Is it audibly noticeable and repeatable?

If not, then what's really the big deal?

 

[Mart Mccall]

That's the thing, not everyone can feel the difference. Many are satisfied with the quality of mp3 music, especially with a bit rate of 128. Only trained specialists or truly "golden-eared"people can truly appreciate the quality of sound. And then, some need time to listen to a specific amplifier in different genres, at different power... Not all amplifiers are equally omnivorous in terms of both acoustics and genres of music.

Well if that's true why bother at all? Only a handful of people in the world are trained listeners and no-one has 'Golden ears' - some may be able to hear to 20Khz and a little beyond, while still young, but is that really of any benefit in appreciating good sound quality?

An amplifier doesn't know what sort of music it's amplifying - it's not music at all to the amplifier anyway - it's just an electrical voltage varying with time - whether it's Bartok or Bachman Turner Overdrive.

 

[Petrov-Al]

This thread is about Kato's philosophy from SOULNOTE
He is not alone in his ideas about the correct approach to developing audio amplifiers.
Here are excerpts from the Interviewwith Charles Hansen of Ayre Acoustics (2015)
https://www.theabsolutesound.com/articles/charlie-hansen-1956-2017/

CH: A real problemis that there are almost no measurable parameters that correlate with perceived sound quality, or even more importantly, on how well the musical experience is communicated to us. This is true for both digital and analog technologies. Given that, I would say that time-related performance is the most critical aspect of sound reproduction in general, and digital is no exception. There is farmore to this than just using high sampling rates or a certain digital filter. And there is the issue of feedback and how it is used —preferably not at all.

When I look back at everything I’ve done, from loudspeakers to analog and digital electronics, I think that there is an underlying thread related to time — both timing issues and the time domain ingeneral. Theear/brain is far more sensitive to time-related information than any other parameter.

In analog circuitry, feedback loops take the time-delayed signal from the output and send it back to the input in an attempt to correct an error that has already occurred. This creates a form of time distortion that cannot currently be measured, yet is clearly audible. The math (and test equipment) tells us that the correction happens quickly enough, but our ears tell us something quite different.

The products that have used very high levels of feedback (yielding incredible measurements) have not stood the test of time and are no longer made. In contrast, more and more designers are copying Ayre’s zero-feedback approach. We’ve never done anything else and we have over two decades of experience in this area.

Ayre’s approach is to understand exactly what it is about DSD that allowsfor its high level of performance. And it certainly isn’t what we were told it was! It turns out that it is really in the time domain, which is no surprise when we understand our marvelous ear/brain hearing mechanism.

 

[onanysunday]

That's a refreshing, interesting take on things that goes against the prevailing dogma (which is fine) because our knowledge of sound and how we measure it and what we know about it is limited at best and full of holes. Thanks for sharing it. It's very valuable, especially for people that may be challenged by it and instead of listening to the information provided and reading through all of it first, try to find or manufacture ways of fault-finding or refuting it. It's a bit long-winded, but I found alot of what I have personally experienced empirically to be reflected in this design philosophy and why I chimed in. Namely, you can't obtain better sound through chasing specs alone which is the prevailing way we are trained to go about it or find meaning through it. Listening objectively, the mind cannot force the ears to hear what it wants them to. Re-read that last sentence again. Everything I've heard with the best specs I could find correlates to a clinical, detailed and lifeless sound to the point of prediction (almost). This helps to explain why that is the case (among other factors that aren't listed) because again, as far as I can tell, nobody has ever considered or shed light on all the factors that effect sound all at once to give anyone alive a complete picture on what sound is and everything that goes into it that results in how it is perceived. It's just never been done, and if anyone claims otherwise, they are overestimating themselves.

The tests we know of (and use) to both design and troubleshoot equipment are not as all-encompassing as we were lead to believe and snapshot measurements we make including sine waves through static testing and measurements cannot replicate the real-world dynamic listening experience. If the piece of equipment was initially designed to show superior "straight-line performance" so to speak as many/most are by chasing the spec dragon, it's difficult to make it a more dynamic and capable piece of equipment without redesigning it to be that way from the start. Also, I'd add on top of everything (while still being incomplete) is the consideration of the Fletcher-Munson curves and how the mangled, conch shape of the human ear doesn't even receive linear sound linearly and naturally de-emphasizes and over-emphasizes specific frequencies, so despite everything else (for instance) you have to take that into account to get something to "sound natural" and linear to us, but that is only one additional factor among many others still not listed or fleshed out for deeper scrutiny or understanding that would give us the complete picture of sound and what all goes into it. Does that then mean audio manufacturers are purposely creating non-linear response that will then be perceived linearly by the listener without the use of a "loudness" feature or EQ? No, not really; unless they want to get laughed at; although some use that as their design philosophy and is why they sound less sterile. It does not matter whether we know everything, but that we remain open to the possibility of new information coming in that changes long-held beliefs to give us a more complete, accurate picture of reality. Letting the mind lead the path with its assumed knowledge is different than going in without preconceptions and allowing the sound itself (as perceived by human ears) to lead the way and provide the tangible inputs which effect outcomes. We never hear music through test equipment and that is its limitation. The only thing that matters is the sound perceived through our flawed hearing. There are no specific tests relating to many of the most important sound parameters such as soundstage (width/depth), musical separation, or the ability of stereo to fill in the phantom space of a missing center channel, etc. and yet, we can all agree they exist, right? So how can we take each of these and measure them independently from one another? We can't. It's just one example of many other unquantifiables that we are begrudgingly forced to live with, or erroneously claim to have sussed out and defined in some way through repeatable, test procedures.

Also, nobody owes anyone anything here as far as I can tell. Nobody has to meet or exceed someone else's impossible burden of proof. Take what you can, and if it doesn't personally resonate with you after doing your own personal experimentation with it through an open mind and clean ears, move on. Life's too short to argue with or prove something to someone else who will never be satisfied.

 

[Retro_Pete]

This thread is about Kato's philosophy from SOULNOTE
He is not alone in his ideas about the correct approach to developing audio amplifiers.
Here are excerpts from the Interviewwith Charles Hansen of Ayre Acoustics (2015)
https://www.theabsolutesound.com/articles/charlie-hansen-1956-2017/

CH: A real problemis that there are almost no measurable parameters that correlate with perceived sound quality, or even more importantly, on how well the musical experience is communicated to us. This is true for both digital and analog technologies. Given that, I would say that time-related performance is the most critical aspect of sound reproduction in general, and digital is no exception. There is farmore to this than just using high sampling rates or a certain digital filter. And there is the issue of feedback and how it is used —preferably not at all.

When I look back at everything I’ve done, from loudspeakers to analog and digital electronics, I think that there is an underlying thread related to time — both timing issues and the time domain ingeneral. Theear/brain is far more sensitive to time-related information than any other parameter.

In analog circuitry, feedback loops take the time-delayed signal from the output and send it back to the input in an attempt to correct an error that has already occurred. This creates a form of time distortion that cannot currently be measured, yet is clearly audible. The math (and test equipment) tells us that the correction happens quickly enough, but our ears tell us something quite different.

The products that have used very high levels of feedback (yielding incredible measurements) have not stood the test of time and are no longer made. In contrast, more and more designers are copying Ayre’s zero-feedback approach. We’ve never done anything else and we have over two decades of experience in this area.

Ayre’s approach is to understand exactly what it is about DSD that allowsfor its high level of performance. And it certainly isn’t what we were told it was! It turns out that it is really in the time domain, which is no surprise when we understand our marvelous ear/brain hearing mechanism.

That's fine; I understand that it's your opinion, but without evidence to support it, it remains just that -an opinion. Or theory rather. Are you open to answering the questions I raised in post #60, or are we at an impasse because of the absence of clear references and supporting evidence to move the discussion forward?

That's a refreshing, interesting take on things that goes against the prevailing dogma (which is fine) because our knowledge of sound and how we measure it and what we know about it is full of holes. Thanks for sharing it. It's very valuable, especially for people that may be challenged by it and instead of listening to the information provided, try to find or manufacture ways of fault-finding or refuting it. It's a bit long-winded, but I found alot of what I have personally experienced to be reflected in this design philosophy. Namely, you can't obtain better sound through chasing specs alone which is the prevailing way we are trained to go about it or find meaning through it. Everything I've heard with the best specs I could find correlates to a clinical, detailed and lifeless sound to the point of prediction (almost). This helps to explain why that is the case (among other factors that aren't listed) because again, as far as I can tell, nobody has ever considered or shed light on all the factors that effect sound all at once to give anyone alive a complete picture on what sound is and everything that goes into it that results in how it is perceived. It's just never been done.

The tests we know of (and use) to both design and troubleshoot equipment are not as all-encompassing as we were lead to believe and measurements we make using sine waves through static testing and measurements cannot replicate the real-world dynamic listening experience. If the piece of equipment was initially designed to show superior "straight-line performance" so to speak as many/most are by chasing the spec dragon, it's difficult to make it a more dynamic and capable piece of equipment without redesigning it to be that way from the start. Also, I'd add on top of everything (while still being incomplete) is the consideration of the Fletcher-Munson curves and how the mangled, conch shape of the human ear doesn't even receive linear sound linearly and naturally de-emphasizes and over-emphasizes specific frequencies, so despite everything else (for instance) you have to take that into account to get something to "sound natural" but that is only one additional factor among many others still not listed or fleshed out for deeper scrutiny or understanding that would give us the complete picture of sound and what all goes into it. It does not matter whether we know everything, but that we remain open to the possibility of new information coming in that changes long-held beliefs to give us a more complete, accurate picture of reality.

Refreshing, perhaps, but true? I remain doubtful. So far, there is no substantial evidence to suggest that it's anything more than theories, and I’m skeptical we will see such evidence.

Science, on the other hand, has a robust and well-established understanding of how the ear functions. This is why, for instance, transparent DACs were developed in the 1980s and why technologies like MP3s and compressed audio work so effectively -thanks to principles like auditory masking. These innovations, among others, have emerged from extensive scientific research and a deep understanding of human hearing, illustrating that our current knowledge and suite of tests and measurements are based on proven, well-researched facts rather than speculative ideas.

 

[Petrov-Al]

Several years ago, the owner of a DarTzeel amplifier contacted me and asked for recommendations on how to improve the sound. I typed in a model of the voltage amplifier circuit and took a Bode diagram. See the attached file. It turned out that only the voltage amplifier has a time Propagation Delay of 360 ns. I asked a colleague to measure the real delay, in hardware it turned out to be slightly less - 350ns.
I replaced the output stage with a Baxandall pair and took a Bode diagram again, the time Propagation Delay decreased by 4 times, see the attached file.
I passed the circuit to a colleague and he modified the voltage amplifier and measured the delay again, it turned out to be equal to 96 ns.
After the modification, the colleague was very pleased with the improvement in sound quality. This is just one example of many.
p.s.
First, a colleague completed one channel and compared it with the unfinished one.