The effects of what you call dynamic distortion are fully understood.
What it does is make the difference between loud and soft sounds, less. This means that soft sounds that were unaudible before can become audible, resulting in a sence of more low level detail. On top of that, it's not a linear thing, it actually changes with level. So loud sounds, like a snare drum, seem to have more impact.
So more impact and more low level detail can be very useful for the listening experience. I fact it is one of the most used tools in the box for sound engineers, they call it compression. Tape, vinyl and some recording studio equipment have audible amounts of it. So do some amps and other audio stuff.
Bill, I'm uncertain exactly what you're communicating here, as a couple of your points seem to appear contradictory. On the one hand, you seem to be suggesting that dynamic distortion reduces the perceived dynamic range, while on the other hand you seem to suggest that it expands dynamic range. Perhaps, you're suggesting the it does both, where the dynamic range seems reduced at lower volume and expanded at higher volume. However, isn't that the same as simply saying that the apparent dynamic range can sound (naturally?) expanded by an amplifier's dynamic distortion behavior?
In any case, it is not apparent by the gear on the market how fully understood, or even recognized, such an subjective effect is. At the very least, it is evident that the majority of amplifier designs on the market do not value the subjective benefit that dynamic distortion behavior can have in making the sonic reproduction of their gear sound closer to real live music.
<snip>
Once I heard a report or study, I cannot say from where I got the info, that people that listen to live music prefer SS, and people that listen to recorded music prefer tubes. Consider it an unsubstantiated subjective claim.
<snip>
Runs rather counter to my personal experience, I have a significant number of friends, both classical and pop musicians who prefer tube based HiFi.
I've been to quite a few live and unamplified concerts, mostly classical, jazz and small chamber ensembles, and I too prefer tube amplification.
My first exposure to tube sound as an adult impressed me with its dynamics and liveness despite the other shortcomings of the vintage gear I was listening to. (Not neutral tonally, and fairly poorly controlled bass) It was enough to get me to abandon solid state.
These days my mostly horn based system is not identifiably tubey sounding despite a lot of tubes in the signal chain + transformers, electronic XO (not tube) and DSP based room EQ ( 😱 )
What Bob Carver did (to show his prowess and make a new product) is what I'd call "the hard way". The easy way is to put a tube in the signal chain. There - tube sound; easier than wrangling the same out of solid state devices. Multitudes of products took the easy way, from Lampizator DACs ("The name itself is a play on the Polish word for vacuum tube, Lampa and the pop culture icon The Terminator") to Jolida JD 301 ("Combines the best of two worlds, a tube pre-amp with a solid state power amplifier"). Such combos, obviously, do provide successful audio entertainment at modest cost.
Regarding controls that change how an amplifier sounds, well I'd think that could be marketable. I'm sure in the realm of audiophiles - similar to computers - there are the "PC" and the "Mac" camps. I'm referring to the old addage of those who like to fool around endllessly with their machine, versus those who prefer that "it just works". Perhaps it's not so much that way anymore with computers, but I'd expect for most folks here we are well entertained getting to mess around endlessly with our systems. A simple control could be part of that entertainment -
Regarding controls that change how an amplifier sounds, well I'd think that could be marketable. I'm sure in the realm of audiophiles - similar to computers - there are the "PC" and the "Mac" camps. I'm referring to the old addage of those who like to fool around endllessly with their machine, versus those who prefer that "it just works". Perhaps it's not so much that way anymore with computers, but I'd expect for most folks here we are well entertained getting to mess around endlessly with our systems. A simple control could be part of that entertainment -
Remember to mount said input cap visibly to the top of the amplifier for maximum effect.
Do this only if it cost as much as your months rent. Then everyone can know how good it sounds.
I worked at one of the several Motorola plants in south Florida for 41 years. In the mid 70's through the 80's I built a lot of amplifiers, both for HiFi and Guitar. All were solid state, after all I worked for a place that made silicon, and gave it to employees just by filling out a sample request form, and "home project" was one of the valid check boxes.
I and a few friends made clones of every SWTPC audio amp made, and a few of us cloned their MC6800 computer too. These were the beginnings of what became the Motorola computer club, and Motorola audio club. There was a ham radio club too but I didn't become part of that one until later.
I had a stereo system in my house that consisted of two old Italian made Voxon receivers biamping two different sets of DIY speakers via a DIY crossover. I liked my system and saw no need to change it......until the lightning strike.....Poof, all was dead including the voice coil in one of the University Labs drivers.
So the president of the Motorola audio club sells me his old home system so he can justify buying a BIGGER one. I make some new speakers, and get my friend's Carver M-400 cube and his Phase Linear 4000 Autocorrelating preamplifier (also a Carver design). To me the sound quality was not as good, but it was loud enough to blow that boom box kid right out of the tree in the front yard next door where he liked to crank that thing......
The 4000 had all sorts of gizmos for "restoring the dynamic range lost in the record making process." There was a peak unlimiter, and a dynamic range expander, along with the autocorrelating filter to reduce noise. At first I thought these things were magic. They brought out sounds that I had never heard before. Careful tweaking could almost erase those scratches in the records. After about a year of ownership, I had learned to hate them. Their sound helped a few poorly recorded or damaged records, but brought an unnatural sound to most.
It did however add another secret weapon in the guitar volume wars that erupted between me and the kid next door. The M-400 had warnings in the manual not to use it for guitar or other sound reinforcement, but I did it often. It would simply shut itself down when I made it mad.
Fast forward to some time in the early 90's. A coworker had seen me driving my first car into work one day and wanted to buy it. It was a rusty 1949 Plymouth that I rarely drove, but I went by his house to let him see it. He had a whole barn full of junk, and asked if I was interested in a trade.
I spotted a nice looking Scott 272 laboratory reference stereo amplifier and the matching tuner and both manuals sitting on the hood of a junk car with stuff piled on top of it. The tuner had broken one of the EL34's, one of the 5U4's and two white ceramic resistors in the amp......I swap the car for the stereo. I replaced the resistors and the tubes, the amp worked. Within a week the 1962 vintage Scott had kicked the Carver stuff into the closet, never to see power again.
That was the spark that I needed to revisit the tube amp building that defined my childhood to early adult years. My first new tube amp, a clone of the Scott. I wound up building 3 or 4 of them most with KT88's before the single ended fever hit.....and the TSE was born.
I gave all the Carver stuff to a collector when I moved north in 2014. it had been stashed in a warehouse and forgotten along with some old Pioneer and Fisher stuff.
I and a few friends made clones of every SWTPC audio amp made, and a few of us cloned their MC6800 computer too. These were the beginnings of what became the Motorola computer club, and Motorola audio club. There was a ham radio club too but I didn't become part of that one until later.
I had a stereo system in my house that consisted of two old Italian made Voxon receivers biamping two different sets of DIY speakers via a DIY crossover. I liked my system and saw no need to change it......until the lightning strike.....Poof, all was dead including the voice coil in one of the University Labs drivers.
So the president of the Motorola audio club sells me his old home system so he can justify buying a BIGGER one. I make some new speakers, and get my friend's Carver M-400 cube and his Phase Linear 4000 Autocorrelating preamplifier (also a Carver design). To me the sound quality was not as good, but it was loud enough to blow that boom box kid right out of the tree in the front yard next door where he liked to crank that thing......
The 4000 had all sorts of gizmos for "restoring the dynamic range lost in the record making process." There was a peak unlimiter, and a dynamic range expander, along with the autocorrelating filter to reduce noise. At first I thought these things were magic. They brought out sounds that I had never heard before. Careful tweaking could almost erase those scratches in the records. After about a year of ownership, I had learned to hate them. Their sound helped a few poorly recorded or damaged records, but brought an unnatural sound to most.
It did however add another secret weapon in the guitar volume wars that erupted between me and the kid next door. The M-400 had warnings in the manual not to use it for guitar or other sound reinforcement, but I did it often. It would simply shut itself down when I made it mad.
Fast forward to some time in the early 90's. A coworker had seen me driving my first car into work one day and wanted to buy it. It was a rusty 1949 Plymouth that I rarely drove, but I went by his house to let him see it. He had a whole barn full of junk, and asked if I was interested in a trade.
I spotted a nice looking Scott 272 laboratory reference stereo amplifier and the matching tuner and both manuals sitting on the hood of a junk car with stuff piled on top of it. The tuner had broken one of the EL34's, one of the 5U4's and two white ceramic resistors in the amp......I swap the car for the stereo. I replaced the resistors and the tubes, the amp worked. Within a week the 1962 vintage Scott had kicked the Carver stuff into the closet, never to see power again.
That was the spark that I needed to revisit the tube amp building that defined my childhood to early adult years. My first new tube amp, a clone of the Scott. I wound up building 3 or 4 of them most with KT88's before the single ended fever hit.....and the TSE was born.
I gave all the Carver stuff to a collector when I moved north in 2014. it had been stashed in a warehouse and forgotten along with some old Pioneer and Fisher stuff.
Like I said, to peruse the Pass DIY designs right here. I own both tube amps and have built several Pass designs. He gets it completely: various levels of H2 and lesser H3, whether positive or negative, appeals to a huge segment of the population.
Dampening Factor has nothing to do with this.
People that have never heard H2 / H3 done correctly are fooling themselves into thinking they’ve heard it all. The phase of the H2 must consistent through the signal chain.
There’s a reason why Pass is so successful commercially and more relevant, with the very critical hobbyist on this forum. The guy listens to both silicon and glass, and spends a lot of time with his group listening to what traits matter to most of us: not measurements but music.
Switch to the Pass forum and post what kind of power and amp type you want to build. Stat what you are looking for sound-wise.
Cheers,
Greg
Dampening Factor has nothing to do with this.
People that have never heard H2 / H3 done correctly are fooling themselves into thinking they’ve heard it all. The phase of the H2 must consistent through the signal chain.
There’s a reason why Pass is so successful commercially and more relevant, with the very critical hobbyist on this forum. The guy listens to both silicon and glass, and spends a lot of time with his group listening to what traits matter to most of us: not measurements but music.
Switch to the Pass forum and post what kind of power and amp type you want to build. Stat what you are looking for sound-wise.
Cheers,
Greg
Nelson Pass' ampcamp amps which we called here locally as the Kampana amp was mistaken for a 300b set, in a blind test, oldtimer tubephiles were tricked into believing that what they were hearing was a 300b set, so even avery simple amplifier can deliver tubelike performance at a fraction of the cost, of course ymmv...
@ggetzoff - "He gets it completely: various levels of H2 and lesser H3, whether positive or negative, appeals to a huge segment of the population."
Your nomenclature refers to 2nd / 3rd order harmonic distortion I assume. Hmmmm... So consistent with SE tube amplification which 1, 2, 3, ... other fellas posting here have been taken by.
Did you per chance mean the phase of the H2 must be consistent across the audible frequency range? Who cares what it does along the signal chain - as long as what comes out at the speaker terminals remains phase stable relative to the input fundamental.
I imagine if it shifts out of phase wrt the fundamental at some frequency, whilst remains in phase at some other frequency, that doesnt sound as good as if it remains in-phase with the fundamental at all frequencies.
A guitarist told me the BBE "aural exciter" for electric guitar swapped 180 phase back and forth across the guitar's frequency range - which tricks your ears into accepting louder SPLs - I guess before you start to cringe. (nothin like being able to play even louder before you start to blow people out of the club!)
I can understand the opposite would be desirable for audio entertainment. Unless you're including house parties where no one is really listening critically ;')
Your nomenclature refers to 2nd / 3rd order harmonic distortion I assume. Hmmmm... So consistent with SE tube amplification which 1, 2, 3, ... other fellas posting here have been taken by.
Did you per chance mean the phase of the H2 must be consistent across the audible frequency range? Who cares what it does along the signal chain - as long as what comes out at the speaker terminals remains phase stable relative to the input fundamental.
I imagine if it shifts out of phase wrt the fundamental at some frequency, whilst remains in phase at some other frequency, that doesnt sound as good as if it remains in-phase with the fundamental at all frequencies.
A guitarist told me the BBE "aural exciter" for electric guitar swapped 180 phase back and forth across the guitar's frequency range - which tricks your ears into accepting louder SPLs - I guess before you start to cringe. (nothin like being able to play even louder before you start to blow people out of the club!)
I can understand the opposite would be desirable for audio entertainment. Unless you're including house parties where no one is really listening critically ;')
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Originally Posted by Bill Coltrane: The thing that bothers me the most in most modern classical recordings is the thousands of edits.
One thing that bugs me about those edits: When they splice 2 sections of a symphony movement together, and one section is with the orchestra tuned up, and the other section is at a slightly different pitch (like just tuned up, versus after 15 minutes of playing). Can't those engineers hear what they did?
ggetzoff posted: "The phase of the H2 must consistent through the signal chain".
Well that makes sense. Phase, yes.
1. A typical H2 distortion makes one alternation appear slightly tall and narrow, and the other alternation appear slightly short and wide. It often looks like that (for example the transfer curve of a single ended output stage can cause that).
2. If you have about the same percent of H2 in two cascading stages (driver and output), the H2 is partially cancelled. That is similar to Push Pull, only the two cascading stages are performing "serial" cancellation. Just the math summation of the transfer functions.
3. Now, look at the nature of H2 of a loudspeaker driver. Hmmm . . . what do you think happens when the speaker is connected one way to the amp that has H2, versus when the speaker is connected the other way (reverse phase)? Well . . . You either get addition of the two H2 factors, or you get partial cancellation of the two H2 factors.
4. Years ago, we took a good loudspeaker, and a good 2A3 amp. We added a DPDT switch to the cord between the amp and the loudspeaker. Then, we applied a 100Hz tone to the amp input. One person would flip the switch back and forth, and the other person would listen to try and hear a difference. Then the persons switched duties. Both listeners heard the different tonality or timbre. How about that?
jjasniew,
I expect that for a loudspeaker driver (Full Range, or Mid-Bass driver), the phase of H2 and fundamental is relatively constant from say 200 Hz to 500 Hz.
And I expect that for a on a single ended 2A3 amp, the phase of H2 and the fundamental is relatively constant from say 200 Hz to 500Hz.
Now apply that to what I said in post # 50.
Want to change the sound of your system that has some H2 in the amp?
Try reversing the connections of the speaker cord(s) to your loudspeaker(s) terminals.
(s) = stereo system
No (s) = mono system
One thing that bugs me about those edits: When they splice 2 sections of a symphony movement together, and one section is with the orchestra tuned up, and the other section is at a slightly different pitch (like just tuned up, versus after 15 minutes of playing). Can't those engineers hear what they did?
ggetzoff posted: "The phase of the H2 must consistent through the signal chain".
Well that makes sense. Phase, yes.
1. A typical H2 distortion makes one alternation appear slightly tall and narrow, and the other alternation appear slightly short and wide. It often looks like that (for example the transfer curve of a single ended output stage can cause that).
2. If you have about the same percent of H2 in two cascading stages (driver and output), the H2 is partially cancelled. That is similar to Push Pull, only the two cascading stages are performing "serial" cancellation. Just the math summation of the transfer functions.
3. Now, look at the nature of H2 of a loudspeaker driver. Hmmm . . . what do you think happens when the speaker is connected one way to the amp that has H2, versus when the speaker is connected the other way (reverse phase)? Well . . . You either get addition of the two H2 factors, or you get partial cancellation of the two H2 factors.
4. Years ago, we took a good loudspeaker, and a good 2A3 amp. We added a DPDT switch to the cord between the amp and the loudspeaker. Then, we applied a 100Hz tone to the amp input. One person would flip the switch back and forth, and the other person would listen to try and hear a difference. Then the persons switched duties. Both listeners heard the different tonality or timbre. How about that?
jjasniew,
I expect that for a loudspeaker driver (Full Range, or Mid-Bass driver), the phase of H2 and fundamental is relatively constant from say 200 Hz to 500 Hz.
And I expect that for a on a single ended 2A3 amp, the phase of H2 and the fundamental is relatively constant from say 200 Hz to 500Hz.
Now apply that to what I said in post # 50.
Want to change the sound of your system that has some H2 in the amp?
Try reversing the connections of the speaker cord(s) to your loudspeaker(s) terminals.
(s) = stereo system
No (s) = mono system
@6A3 - I guess I have no idea what H2 distortion is. Thanks for explaining!
I've heard of the "Wood effect" which has to do with the absolute phase of the recording, but never heard of a speaker cancelling the "H2" of the amp. Thanks.
I've heard of the "Wood effect" which has to do with the absolute phase of the recording, but never heard of a speaker cancelling the "H2" of the amp. Thanks.
I understand your confusion, as I didn't fully explain.Bill, I'm uncertain exactly what you're communicating here, as a couple of your points seem to appear contradictory. On the one hand, you seem to be suggesting that dynamic distortion reduces the perceived dynamic range, while on the other hand you seem to suggest that it expands dynamic range. Perhaps, you're suggesting the it does both, where the dynamic range seems reduced at lower volume and expanded at higher volume. However, isn't that the same as simply saying that the apparent dynamic range can sound (naturally?) expanded by an amplifier's dynamic distortion behavior?
In any case, it is not apparent by the gear on the market how fully understood, or even recognized, such an subjective effect is. At the very least, it is evident that the majority of amplifier designs on the market do not value the subjective benefit that dynamic distortion behavior can have in making the sonic reproduction of their gear sound closer to real live music.
There's a difference between actual signal level and perception. Compression or adding distortion (same thing) changes the envelope of the signal. See Sound Envelopes - Teach Me Audio But gives the perception of more impact in percussion sounds. (and more low level detail) This has to do with how human hearing works. Iow the stimulation of critical bands in our ears. See here;Siemens PLM A sinewave stimulates only 1 band. And it's perceived loudness is solely determined by the amplitude of the signal level. But if you add distortion, more bands are stimulated and the perception of loudness increases while signal level doesn't change. So adding distortion stimulates more critical bands in your ears and that leads to a higher perception of sound loudness, without changing signal level. Does this make things clear?
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"However, isn't that the same as simply saying that the apparent dynamic range can sound expanded by an amplifier's dynamic distortion behavior?"
and
"So adding distortion stimulates more critical bands in your ears and that leads to a higher perception of sound loudness, without changing signal level."
So if an amplifiers distortion increases non-linearly (i.e. it doesnt simply double in amplitude with double signal level) then it will - apparently, via mechanisms of human hearing - expand the dynamic range of the program material being fed into it. Then, assuming that live music naturally has a greater dynamic range than recorded, it follows that an expansion of recorded music's dynamic range brings it closer to a live sound.
So where do you set the "threshold" of the onset of such distortion?
Would it be most beneficial to do this as a processor in the line level feeding a power amplifier whose distortion is mostly invariant wrt output power? That way, this dynamic distortion based apparent dynamic range expansion always happens in the same way at any final listening level. The onset threshold remains fixed, while you're free to listen as loud or quiet as you want. With the obvious caveat that the most realistic sound would be at an appropriate average SPL for the kind of material you're listening to.
I never knew non-linearly increasing distortion with signal level could make an amp or system's reproduction of music sound more life like. Not even saying what kind of distortion, other than that which generates harmonics upon the input material. Simply via first principles of human hearing (and that compression is inherent in all recorded music)
Perhaps this accounts for some "measures so-so, sounds Great" phenomena that we know exists, but is hard to rationalize. I can see how an amp with more distortion could "sound better" - even though it measures not as well as another which sounds "flat" or whatever - when that one should be the better sounding unit.
and
"So adding distortion stimulates more critical bands in your ears and that leads to a higher perception of sound loudness, without changing signal level."
So if an amplifiers distortion increases non-linearly (i.e. it doesnt simply double in amplitude with double signal level) then it will - apparently, via mechanisms of human hearing - expand the dynamic range of the program material being fed into it. Then, assuming that live music naturally has a greater dynamic range than recorded, it follows that an expansion of recorded music's dynamic range brings it closer to a live sound.
So where do you set the "threshold" of the onset of such distortion?
Would it be most beneficial to do this as a processor in the line level feeding a power amplifier whose distortion is mostly invariant wrt output power? That way, this dynamic distortion based apparent dynamic range expansion always happens in the same way at any final listening level. The onset threshold remains fixed, while you're free to listen as loud or quiet as you want. With the obvious caveat that the most realistic sound would be at an appropriate average SPL for the kind of material you're listening to.
I never knew non-linearly increasing distortion with signal level could make an amp or system's reproduction of music sound more life like. Not even saying what kind of distortion, other than that which generates harmonics upon the input material. Simply via first principles of human hearing (and that compression is inherent in all recorded music)
Perhaps this accounts for some "measures so-so, sounds Great" phenomena that we know exists, but is hard to rationalize. I can see how an amp with more distortion could "sound better" - even though it measures not as well as another which sounds "flat" or whatever - when that one should be the better sounding unit.
I understand your confusion, as I didn't fully explain...Does this make things clear?
Yes, that clears it up, thanks. The net effect can be that certain distortion behavior makes the reproduction sound more live/real to human ears. Behavior such as advocated by Victor Lamm. This is what open loop triodes, in particular, tend to do. It also highlights a bifurcation among amplifier engineers. The majority camp of engineers seeks to minimize all amplifier distortions, while the minority camp seeks (knowingly or not) to co-opt the distortion behavior of the human ear.
For the minority camp of engineers, this isn't simply a matter of how to make the ample distortion sound pleasant, sort of hiding it I plain sight since it cannot be hidden out of sight. It's that the distortion behavior is, itself, an important feature and not a bug. 😀 To borrow a software industry joke.
As for the majority camp of audio amplifier engineers, the high proportion of amplifiers which measure superbly (well beyond the detection threshold of of human ears), yet still don't sound any closer to real/live music, seems evidence that this psychoacoustic effect seems to be widely overlooked, or, at least, under appreciated.
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One must not neglect the effect a real speaker have on the amplifiers behaviour. Measuring good with resistive load is one thing, measuring good with a speaker something else. Also the behaviour of the speaker itself is importent, something esa meriläinen has documented.
@ Ken - "Victor Lamm" - did you mean Vladimir Lamm of Lamm industries? (How many Lamms can there be doing high end audio?)
From a Stereophile review of their $13,690 L2; "A somewhat unusual combination of high-voltage tube power supply and superlinear high-voltage MOSFETs allows the L2 to achieve an enormously large output voltage swing while retaining practically constant harmonic content of the signal, with an absolute dominance of the second-order harmonic." Read more at Lamm Industries L2 Reference preamplifier | Stereophile.com
I'll rephrase your preceding statement if I may; "The effect is, that certain distortion behavior - such as an absolute dominance of the second-order harmonic - makes the reproduction sound more live/real to human ears"
Put that in a mix with $13K worth of hardware and well executed design and I assume you really do get "The highs were simply breathtaking, even if somewhat paradoxical in nature: sweet and very pleasingly harmonic, yet remaining clean, extended, soaring, pure, and detailed. Not once did the L2 tip over into the chaffy, harsh, or overly analytical, or make me wince in any way—yet I was sure I was hearing as far up into the audible spectrum as I ever had. Beautiful and inspiring".
Read more at Lamm Industries L2 Reference preamplifier Page 3 | Stereophile.com
As I am a "mere mortal" when it comes to this stuff and will never own a $13K preamp (I got a nice Marantz AV 9000 for sale on ebay for $75 - with HDAM technology...) I could at least focus on the "absolute dominance of the second harmonic" part as a clue to which direction to head in, before my 62 yr old ears fail me completely. I assume I could do that by -
- building or otherwise obtaining a SET stereo amplifier, which many here have stated their love of.
- tuning the existing junk I happen to have on hand to emulate the "absolute dominance of the second harmonic" aspect.
- getting a Lampizator or other brand DAC which runs the decoded audio through a triode tube (used as a signal processing element) to impart the "absolute dominance of the second harmonic".
- getting and subsequently working with one of the DSP DACs, where I assume you can do anything transfer function wise you want to the decoded audio signal - including impart the characteristics of various tubes in various circuit topologies and bias conditions. And FAIK, all that's been done already using DSP.
I figure I've only 5 - 10 years before I'm working with the DSP inside my hearing aids... Better move in a direction while I still can hear something - unaided! Hence my interest.
From a Stereophile review of their $13,690 L2; "A somewhat unusual combination of high-voltage tube power supply and superlinear high-voltage MOSFETs allows the L2 to achieve an enormously large output voltage swing while retaining practically constant harmonic content of the signal, with an absolute dominance of the second-order harmonic." Read more at Lamm Industries L2 Reference preamplifier | Stereophile.com
I'll rephrase your preceding statement if I may; "The effect is, that certain distortion behavior - such as an absolute dominance of the second-order harmonic - makes the reproduction sound more live/real to human ears"
Put that in a mix with $13K worth of hardware and well executed design and I assume you really do get "The highs were simply breathtaking, even if somewhat paradoxical in nature: sweet and very pleasingly harmonic, yet remaining clean, extended, soaring, pure, and detailed. Not once did the L2 tip over into the chaffy, harsh, or overly analytical, or make me wince in any way—yet I was sure I was hearing as far up into the audible spectrum as I ever had. Beautiful and inspiring".
Read more at Lamm Industries L2 Reference preamplifier Page 3 | Stereophile.com
As I am a "mere mortal" when it comes to this stuff and will never own a $13K preamp (I got a nice Marantz AV 9000 for sale on ebay for $75 - with HDAM technology...) I could at least focus on the "absolute dominance of the second harmonic" part as a clue to which direction to head in, before my 62 yr old ears fail me completely. I assume I could do that by -
- building or otherwise obtaining a SET stereo amplifier, which many here have stated their love of.
- tuning the existing junk I happen to have on hand to emulate the "absolute dominance of the second harmonic" aspect.
- getting a Lampizator or other brand DAC which runs the decoded audio through a triode tube (used as a signal processing element) to impart the "absolute dominance of the second harmonic".
- getting and subsequently working with one of the DSP DACs, where I assume you can do anything transfer function wise you want to the decoded audio signal - including impart the characteristics of various tubes in various circuit topologies and bias conditions. And FAIK, all that's been done already using DSP.
I figure I've only 5 - 10 years before I'm working with the DSP inside my hearing aids... Better move in a direction while I still can hear something - unaided! Hence my interest.
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So where do you set the "threshold" of the onset of such distortion?
This is the crux of the matter imo.
I think it depends on the music, listening environment and taste of the listener.
If the music is highly compressed, like most modern pop, adding distortion will not be very beneficial.
If the listening environment is noisy, you need highly compressed sound. Otherwise you can't hear the soft parts.
Best would be a knob that controls the amount of compression/distortion, so the customer can dial in what's preferred.
@ Ken - "Victor Lamm" - did you mean Vladimir Lamm of Lamm industries? (How many Lamms can there be doing high end audio?)
Yes, I meant Vladimir. I inadvertently crossed his first name with that of Victor Khomenko, of Balanced Audio Design.
... and their 100-opamp mixing consoles 😎
Jan
I'm glad it's not 100 tubes.
This is the crux of the matter imo.
I think it depends on the music, listening environment and taste of the listener.
If the music is highly compressed, like most modern pop, adding distortion will not be very beneficial.
If the listening environment is noisy, you need highly compressed sound. Otherwise you can't hear the soft parts.
Best would be a knob that controls the amount of compression/distortion, so the customer can dial in what's preferred.
I do not classify pop as music.
it is industrial sounds.
I do not classify pop as music.
it is industrial sounds.
Some "industrial" music for you...LOL.
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