Are future improvements in Amp/PreAmps slowing to a crawl?

Dragon_Vibe,

"maybe the Future is already here and we are at the very end of technology breakthrough regards to audio?"

This reminds me of the patent office wanting to shut down decades ago because "everything has already been invented"

Breakthroughs happen every day. As far as designing amplifiers with the express goal of having a particular sound to me is a waste of time. It means the designer has given up on the notion that "perfection" is not attainable. 

Take a look at this viewpoint of distortion...
 http://h-cat.com/images/H_CAT_White_Paper.pdf

Roger

THD analyzers are good for getting you in the ball park. It will at least tell you if you have really bad distortion to deal with. However the problem is that they don't go far enough. IOW they are not capable of detecting small problems that occur at or near the fundamental [image] at the input.

Such problems slip well under the "THD" radar and go unnoticed.

If you examine the full spectrum which includes radio and light frequencies - you will see that only the audio portion of the spectrum requires the medium of air to exist (acoustically). Everything above that can exist in the vacuum of space. Therefore if you design an amplifier to be specifically used for audio then you must include the impact that the medium of air has within the design goals.

"Live" has a playback speed. That constant speed is what is recognized as the "live" indicator or marker to the brain. If you pass the sound as electrical data the does not take into consideration its exact speed - then you have stripped away the "live" marker. The brain defaults to "fake" instantly.

Since audio flows as a sound wave - you have to include the wave as part of the critical data needed by the brain to accept what it hears as "live".

The portion of an audio signal the has the embedded wave information is extremely small and exists right at the fundamental signal. Capturing the velocity of the wave data and using it as the speed governor for the outgoing signal is the trick.The brain is convinced that what it hears is "live" because the "live" marker still exists after the amplifying process is done.

This has already been done and it works perfectly.

Roger

kosst_amojan

"...most people feel that amps with extremely low distortion figures sound analytical and lifeless"

Therefore if you have an amp with no distortion it will sound bad?
The answer is no. Those other amps that have low dist figures sound bad for a plethora of reasons - not because it measures low.

Hear is a simple formula:
Live = no distortion
therefore
no distortion = Live

The live nature of a sound event is exposed by the stable speed at which it is flowing toward you (Mach One). The formula above breaks down as soon as you alter the speed or velocity of the delivery system. The electrical version of the event is the odd man out. You have to force the electrical representation to include the environment of the original venue.
All you want to do is generate an air pattern in your listening room that is an exact copy of the air pattern at the hall. This air pattern is merely a log of 2 things. Instantaneous air pressure or amplitude (Vertical axis) and time (Horizontal axis) . If you alter either on of those properties - you have distortion.

Conventional amplifiers only deal with amplitude. Any work done to reduce its distortion is in the vertical axis and is a poor attempt at keep the timing right.

If on the other hand you maintain constant velocity in the amplifier then you have emulated one of the properties of air - mainly its velocity which is zero. A zero velocity medium guarantees that sound waves traveling through it are not artificially accelerated or de-accelerated. Remember the sound travels at Mach One. The medium is motionless. The sound appears the be live simply because of its stable "playback" speed. 

kosst_amojan

"I'm not sure what you're talking about as "speed" in a wave"

Sound waves travel at Mach One approx 750mph. Thats considerably slower than the speed of electricity / light.

After sound waves get converted into electricity it is the job of the amplifier [chain] to make a larger version of the input signal and pass it on only to be converted back to sound waves by your speakers. When this whole process of converting it one way and then back to the "original" way is examined we can see that the handling of the fragile "data" that is embedded in the signal has been compromised or corrupted. The degree of damage to the signal is not readily noticed because it is so slight. The problem is that the integrity of the payload (embedded image) has been modified by a composite of non-linear events no matter how small. This is like recording with an apple and playing back with an orange.

I wanted to make the electrical handling of the smallest detail the highest priority. This is why I refer to a cloning process because all the data is recovered at a "DNA" level and duplicated or repeated. When 100% of the signal is "cloned" then the copy at the output also contains the marker or cue that it is "live". This marker is the first thing to go in conventional amplifiers.

Aside from the obvious visual cues, a phenomenon at the concert hall tells your brain this is live. Your brain recognizes the medium of air because air does not distort. If you are trying to make your brain accept music fed through (any process other than air) then you better make sure it does not distort otherwise the "live" characteristic is not in the final conversion.

Roger

goffkait

" I'm afraid they've run out of options trying to somehow improve upon the current model. More designers must think outside the box."

That's whats happening for sure. There is only so much tweaking you can do to a given amp or preamp design. You have to go outside the box to get new ideas to try. Unfortunately most designers have given up looking for that "new" circuit that will give them the best results possible.

I saw a video of Robert Harley (TAS) interviewing a panel of the top audio designers at an audio show in 2015. It was very interesting but the last question had a response that honestly I did not expect. He (Harley) asked them "Do you think we will ever be able to reproduce the live experience in music?"

The answer from all members on the panel was the same - "no". This tells me that these designers have hit a brick wall in their designs - sort of like writers block. 

This is why I felt it was necessary years ago to take a different approach. I needed to bypass that brick wall by simply using an entirely different approach. If you step back and take a look at the sound reproduction process and you had no idea where to begin - you have to boil it down to the least number of steps. First you want to capture the live acoustic event as performed in the hall.  Then you want to release or playback the copy into your listening room. If it is a perfect copy (like a clone) it will transfer all of the attributes of the original performance including the belief that the performance is happening now (live)

kosst_amojan

"To date humanity has invented a few variations on 2 basic gain devices and there's 3 ways you can use those gain devices. That gives you a limited number of basic topologies to arrange in some fashion"

This is true and pretty much sums up conventional thinking. This is exactly what I had to break away from. You see those devices and see their limitations. I see the same devices and I see unlimited uses.
What if there was a fourth way to use them? Now that would be outside the box. 

Ever heard of an amplifier with automatic focus?

inna,

" music can't even be recorded as performed let alone reproduced"

It turns out this is not true. You have formed an opinion about the current and past recordings based on what? An admittedly poor reproduction system? If you take the distortion out of the playback electronics you would be shocked at how much captured detail and imaging data is embedded in the recording including cues that trigger the sensation that it is live.

kosst_amojan

  " What if I could power my amp with fairy dust? Oh crap! This is reality and asking "what if..." doesn't make a thing possible. There isn't a 4th way to use them so now what? Just design irrational circuits that do nothing? You see these unlimited uses. Tell me what they are exactly.
And who are you exactly to left handedly insult people as to the quality of what they've heard? I can't stand the kind of pompous arrogance some folks like you wield around here, preaching down to the unwashed masses as if you've beheld divinity and know better than all "

With all due respect - I was not playing the "what if" game. When you say " There isn't a 4th way to use them so now what? " What you are saying is your not aware of a 4th way to use them.

kosst_amojan -
" And who are you exactly to left handedly insult people as to the quality of what they've heard?

When I told inna - "  You have formed an opinion about the current and past recordings based on what? An admittedly poor reproduction system? " I was stating 2 observations. The first sentence is accurate - he has formed an opinion as did most audiophiles. The second sentence was not to imply that he has an inferior playback system but that playback systems (in general) are not up to the task of exposing the degree of resolution that can be recovered from existing recordings. 

kosst_amojan  - 
" I can't stand the kind of pompous arrogance some folks like you wield around here, preaching down to the unwashed masses as if you've beheld divinity and know better than all "

On the contrary - I am humbled by the knowledge I have gained regarding methods of amplifying simply by being persistent and doing all the homework. If you look at the topic of this thread you will see why I decided to chime in with some good news. There are exciting advancements being made in the field of sound reproduction that audiophiles will benefit from.

If you read any of my posts anywhere you will see that I do not speak ill of any brand or designer. How can you have a civil dialog with the audiophile community when you get a post like yours that shows anger and frustration because you are certain that what I'm trying to share as legitimate progress is readily dismissed as nonsense?

Breakthroughs do happen if you stay positive and committed to a specific goal. My goal was to design an amplifier that does not distort. In order to do this I had to dig very deep into the route causes of distortion. I have had to approach this task by thinking outside the box.

When I said the other designers had "writers block" this was a compliment. I was giving them the benefit of the doubt. I could have said "they don't know how to do what I did." which is where we are at today.

You should not become angry at me for making progress that other designers have not been able to duplicate. Somebody has to move the ball forward. I remain humbled by the massive progress I have achieved.

I am using new methods of dealing with circuit weaknesses. I have made it a point to learn how active and passive devices "behave" when power and signals are applied. I have been at this for decades. I have gone through hundreds of circuit configurations to find one that works.

I have developed several new circuits that are able to produce gain from devices that are not meant to have gain. (The 4th way?).

I have found ways to cause a device to inherit a specific property from another device.(The 5th way?)

I have developed a detector circuit capable of extracting the velocity of a signal using current fragmentation. I can take a sample of pure current and cause a fragment (spin-off) to be used independently but with the same purity as the donor current. By converting the gauge of the current to a smaller size - it can be used to reveal far more detail buried in the music signal. (the 6th way?)

I can successfully monitor tiny changes along the time domain axis. This is where all forms of distortion are born. Remember that the significant data embedded in the recording that represents the sonic events is amplitude and time. This is the key to making distortion free amplifiers.
All you need to do is make the output of a circuit provide a larger version of the input with the SAME relationship between amplitude and time.

This gives you a way to recover magnitudes more information.

This is not wishful thinking. I have this technology up and running.

Roger

geoffkait
What are you using for a headphone amp?

" nobody actually building circuits talks in the cryptic code you do"
Thank you - I like to think I am unique. It's probably because they are still building conventional circuits. BTW - what part is cryptic to you?

So without hearing the process I'm talking about - you have totally dismissed it as "snake oil". The one thing I like about these threads is there is a kind of paper trail that we can look back on and say "remember when they didn't believe it was possible to amplify without distortion?"
  
(Lets note the date)

This is why I don't spend much time on these threads because if you introduce any kind of new thinking to solve old problems - people freak out. The ones that complain the most are those with skin in the game. Is this technology somehow a threat to you or your business?

I don't know many people on this thread as far as what they do for a living. I know Ralph who I respect has his tube circuitry but never comes across as anything other than someone who enjoys what he does and is successful at it. I'm pretty sure Ralph does not see me as a threat. My hat is off to any designer that strives for the best. These threads are for the purpose of sharing all things audio. Lets leave out the snake oil please. 

Perhaps someday you will get a chance to hear what I'm talking about.

I won't bother telling you about automatic focus - that doesn't seem go with sound reproduction either but I still use it.

kosst_amojan

Here is the simplified version:
When a recording is made of a performance in the hall and the sound waves strike the microphone diaphragm 2 bits of data are saved.

1: what is the exact air pressure
2: what is the exact moment in time (for that air pressure reading)

This assumes 2 channel optimum mic placement.
This allows you to triangulate the location of every instrument on stage.
That's it - nothing more nothing less. If you do this correctly you have a piece of acoustic history captured.

Can we agree on that?
Is there anyone reading this post that thinks otherwise?

If you can cause your speakers to produce the same air pressure changes along with the exact times - you have yourself an acoustic clone of that event and you will not be able to tell the difference between the live and the "copy".

That is my goal and I am quickly approaching it.

Roger

kosst_amojan

" You can get away with some quite high distortion and still have convincing playback so long as that distortion is low in order and complexity."

Yes this is a good point but under the conditions you just described you cannot pass the cue that it is live. It requires the distortion levels to be taken down by a massive amount so it can approach the distortion figure of air.

I also mentioned that -
 " If you do this correctly you have a piece of acoustic history captured."

I will be making recordings using the same amplifying method. The interest by Hollywood is already there regarding 3D movies and recording with H-CAT mic preamps. It will save the studios mega bucks in post production because they will not have to manually move objects around the theater so the audience has some illusion of placement. (As in Atmos) This process will project an object back into mid air where it came from all by itself.

As far as mics and speakers well - here goes an explanation that I know is going to cause a riot...

What ever shortcomings a microphone has it is introducing a stable flaw. IOW it does not dynamically modulate the location information. The same holds true for speakers (wait for the riot...)

I know all the arguments already - no need to prove me wrong.

"speakers have a lot of distortion - how do you get around that?"
"midrange info riding on the woofer will cause Doppler"
"phase shift in the crossover will screw up the imaging"

All of those things are true but you need to pay attention to the difference between symmetrical and non-symmetrical phase errors.

To all  -
Well its been cosmic folks - time to get back to work.
I was just coming up for air.

Best thing to do is stay tuned to the web site / watch the news / read the magazines and see where this goes.

A shout out to Miguel at Tripoint audio in Florida who just learned that his house survived Irma.

Amen

kosst_amojan

It sounds like you have a great system and I'm sure you are happy with it.
The only thing I can tell you is I had a level of purity that I never thought could get better - then the latest version of the process that has a much tighter auto focus system was put in and wow I ended up with quite a jump in the believably and ghost like imaging. My speakers disappeared completely and seemed to have nothing to do with what I was listening to.

Anyhow good luck to you perhaps you will have a chance down the road to hear H-CAT.

In fact if you don't mind - contact my web site or email me and let me know where you are located.

Roger

kosst_amojan

Please contact [email protected]

Roger

kosst_amojan

I was going to offer to send you a stripped down line stage that would have one input / one output and a manual volume control under no obligation to buy. It would be a fully functional stage with auto-focus.
You can live with it for a week. I make this offer to skeptics who become believers within the first 10 seconds of listening.

This would allow you to hear what is possible.

Roger

I understand - good luck to you and your daughter.

toddverrone,

Saw your system - nice. You made all the cables yourself?  And the DIY resonators?

BTW - Your listening room is way too clean! 

@parrotbee 
" I guess in terms of sound quality - gains will be marginal..."

There is a way to obtain massive improvements in sound quality. At some point there will be a realization that the missing link in analog amplifiers is how much attention is given to the velocity of the [music] signal passing through the amplifier. Unstable velocity creates an image that appears to have sonic "vibration". In much the same way as there have been great strides in discovering ways to limit mechanical vibrations, the same negative vibration effects can come from the electronic circuitry itself.

When this is addressed a radical drop in "interference " is removed from the image. As the electrical vibrations become less and less it begins to move in the direction of more and more "live". This is because "live" has no vibration interference. It is the loss of vibration interference that is recognized by the brain as a live event.
 

"If we all worked on the assumption that what is accepted as true is really true, there would be little hope of advance."

Orville Wright

"I do not think there is any thrill that can go through the human heart like that felt by the inventor as he sees some creation of the brain unfolding to success... such emotions make a man forget food, sleep, friends, love, everything".

(Nikola Tesla)

After decades of research...
Welcome to my world.

Roger

The problem with most DACs is not the chipset on the front-end.
It is the analog stage that follows. The digital data to be converted to analog far out strips the ability of the analog section to pass along the stored image.

" the preamp itself isn't really improving "
This was true up until now.

My point was that unless you attach a distortion-less analog stage to the back end, the full resolution cannot be realized.

I can run a 16 bit dac that will out perform a 24 bit dac simply be using a distortion free (0%) analog stage.

In a DAC the analog stage is the weak link.

I'm assuming that you have used tube output stages and recognized the difference in sound quality.

@parrotbee 
" I guess the loss of distortion has become a single minded pursuit for many amp maker - HAlcro; Neodio; LAvardin - do they not address 'velocity' already?"

This is a good question and the answer is no.

The reason is because you need a method of detecting velocity first before you can correct it - and then you have to do it in real time (without any delay). 

If years ago Ray Dolby was to post a comment the he has found a way to reduce noise in tape recorders - the reaction from this crowd would be "that's impossible - there's no way to prevent the magnetic material from....bla bla bla."

Sorry.

Honestly you know from my previous posts that I do not consider myself as a "know it all" but I do have much more data on the problems with analog amplifiers and ways to make them work without distorting.
Is it difficult to do? Hell yes. Is it impossible? Hell no - I got it to work.

I have tried to share with this audiophile community some of my discoveries regarding sound reproduction that has resolution on a biblical scale. As it turns out I do have a distortion-free amplifying method already and it does produce "live" sound by default. This was the target of my work. It's not magic and I'm no genius. I simply took the time to figure it out. I have had to learn how the brain recognizes a live event from a reproduced event. The accuracy need to satisfy the brain that it is real (live) is a specific property of the sound waves you are listening to. It turns out to be a very simple yet very difficult thing to get right. As I've said until I'm blue in the face - its velocity. The actual speed at which the music signal traverses through an amplifier. It has to match the speed of a sound wave traveling on the outside of the amplifier. The solution is to synchronize the output velocity to the input velocity. It comes in at the speed of sound and it exits at the speed of sound. (same thing as air).

That is exactly what I have done. Period.

When you do that - you have cloaked the electronics and emulated the the most important property of air - its stable velocity. Your brain is satisfied that "this is real and happening now".

The bottom line is going to be the near future. That is when the full method of distortion free sound reproduction will be shown. While it is compatible with current recordings, I am about ready to start making recordings that also have no distortion so as to realize the complete capture and playback with unprecedented results.
I can clone a sound event and repeat it on demand. If the clone is 100% correct - you will not be able to tell the difference. I am not afraid to do the homework even if it took years.

All I can say is watch the news.

Roger

Actually you just pointed out something that I'm sorry I was not more clear about and it might cause confusion. You are describing speed as in vertical (like slew rate) how fast can it switch between power supply rails.

Velocity is not measured vertically - it is speed as seen along the horizontal axis (time domain).
When you drop a pebble into a pond - the rings flowing away do so at a rate that shows expansion. That is the velocity. The height of the expanding waves (intensity) would be in the vertical axis.

This is why I said that the electrical switching (slew) speed is very fast. But the time measurement I'm talking about would be a race from the input to the output.

A drummer hitting a rim shot can cause a near explosion of energy in amplitude (vertical) but it still flows to the audience members at one (horizontal) speed. No matter how large the transit it still arrives at the back of the hall by traveling at the speed of sound (approx 750 mph).

This horizontal rate seen in the amplifier MUST stay fixed at 750 mph for you to think it was traveling through air.

Any non-linear event happening along the vertical axis causes the speed of horizontal path to vary resulting in an acceleration or de-acceleration of the delivery speed. Like wow and flutter in a tape machine.

It is microscopic Doppler and causes the image to go out of focus.
My auto-focus circuitry can produce a countermeasure along the time domain (a time warp) of extremely tiny amounts. As small as one thousandth of a degree of phase shift. These amounts are so small it has to use red shift and blue shift.

The velocity detector itself can pick up less than a nano volt discrepancy in a line level signal. The gain of the detector is massive. This much pressure is used to lock the red and blue shift generators in a dead battle for no motion or movement (reference point). If the music signal begins to drift ahead of the reference (in phase) by any amount (1/1000 of a degree) it is stopped in real time by the appropriate red or blue shift from becoming a harmonic.

It literally cannot distort.

It matches the flow rate and stability of air. The amplifier treats the music signal as an actual wave. It therefore maintains the speed of sound (Mach One) as the exit speed.

The amplifier becomes "invisible" and acts like a hole in the wall.
In fact that is the sensation. Listening through a portal passing the air disturbance pattern of the original venue (in the past) through the portal and into the present without missing a beat and with no distortion.

An actual miracle.

Your brain instantly accepts the experience as live because it recognizes the accuracy of the delivery speed. In the meantime a complete totally stable image of the original event is "phase locked" in front of you.

Your brain can easily pick out a single instrument in the orchestra and filter out the rest with ease. This only happens with live music because the locations are so stable your brain can apply a vector based filter to block anything it is not "paying attention" to.

It is the same thing as being there.
This took me 30 years to figure out.  

Roger

I am not talking specifically about power amps. It applies to all circuitry, phono stages, line stages, and analog back ends in DACs.

This is why if you listen to older recordings made in an all vacuum tube studio using tube mic preamps, they are the most stunning captures.

I have understood the "magic" of tubes for years. I would have been happy to duplicate that "magic" using transistors. Most other designers have defaulted to the second best device - the almighty JFETs , MOSFETs which because of the field effect closely approximates the more linear grid control.

That's why the popular JFET craze is so well accepted. However, they too have time warp issues that need to go away in order to compete with air which is 100% linear and has zero distortion.

The spec for the velocity along the horizontal (time domain) axis = constant

With all due respect I'm still not sure what you are asking.
If you want to know the value of the output velocity it equals the input velocity +/- zero.

It does not add any acceleration or de-acceleration.

The sound of a circuit is the signature of how its velocity is handled.

Are you sitting down?

I can install an auto-focus circuit to control the velocity of a tube circuit and achieve exactly the same sound as a solid state circuit with the same auto-focus. They would both produce the same holographic images.

Roger

Amplifiers that do not control this property are not constant (unstable) and as a result have varying degrees of smear or focus issues based on microscopic time warps.

Ok here is the answer to the age old question "why do tube amps sound better even though they have higher THD?"

Tube amps sound amazing because the velocity variations are minimized. I knew this back in 1969 when I use to design tube amps.
It is why they are still hard to beat.

But - they are still unstable. Removing the variations completely gives you an amp that has no sound of its own. IOW it does not sound like tubes or solid state.

It "sounds" like air.

Let me ask a simple question of anyone reading this post. This might lead to an understanding of what I'm talking about. It also may pull your thinking slightly outside the box.

Do you know what part or area of a circuit determines the signal to noise ratio of the circuit?

Please refrain from wise guy answers. Thank you.

Yes I am not talking about slew rates like volts/us.

The velocity spec is that the output velocity must = the input velocity.
By default the signal entering the circuit has a horizontal (time domain) velocity of Mach One because the music was captured at that speed.
Unless you maintain this "embedded" velocity you are going to inject small amounts of Doppler into the chain producing an out of focus result.

If you record music on a tape machine at 15 ips you must set the playback speed also at 15 ips. If your tape machine during playback is actually 15.05 ips you will have a slight leaning towards Micky Mouse.
If it is playing back at 14.95 ips it is leaning more towards Barry White.
This degree of deviation does not sound like a lot but to the projected image it is devastating.

Even thought the amplifier has no moving parts and does not seem like it can vary the playback speed - but it does.

It alters the velocity the same as a poor capstan servo. It is the equivalent of wow and flutter.

When the music signal encounters a non-linear event - the delivery speed is altered.

A superposed "hologram" will collapse and every instrument in the performance will have the same degree of focus or lack thereof.

Again with all due respect..

"Amps don't cause pitch variation"

This is in fact a false assumption. They can and do cause pitch variation.
This is not something I suspect (as in theory) - it is something I know (as in proof)

I started out years ago with a theory of what was happening in analog amplifiers. Today it is no longer a theory.

The breakthrough has already happened.

I'm really very sorry for coming off as an arrogant "know it all". The facts are on my side. This is one argument that you cannot win.

Its up to me to perhaps provide a better understanding or illustration of what I'm talking about but it is absolutely true that analog amplifiers do vary the pitch.

The reason it has not been known before is because of the amount being so tiny. It is this tiny amount the determines the degree of focus realized in the image.

kosst_amojan,

Thanks for being civil and open minded. I will be happy to fill out your form. You need to give me time to generate the math on a sample of this phenomenon. In fact I was going over some figures that I thought might allow more understanding of the problem found and how my solution is implemented.

BTW - question for you.
" Circuits vary the pitch through intermodulation distortion through summing the fundamental with harmonics."

Where do these harmonics come from exactly?

I'm not trying to be difficult. In fact...

" That has nothing to do with the speed of electromagnetic waves passing through circuits at damn near the speed of light."

He (Kosst) still is referring to the wrong speed.
Electricity travels at pretty much the speed of light. (fast)
Sound travels at around 750 mph (much slower)

It is the speed of sound that has to be included in the amplifying process.
That is why the electrical speed that you can draw current from a power supply (vertically) is the first speed and the rate that the sound wave data enters and leaves the circuit (horizontally along the time domain) is the second.

This is the most difficult aspect of understanding how to separate these 2 speeds that are present in the amp. 

I keep pointing to the one I'm trying to tell you about and conventional thinking latches on to the "speed of light" every time.

When a musical note leaves your speaker it does not make it to your couch at the speed of light. Nor did the original music recorded in the studio travel from the instrument to the mic at the speed of light.

I believe I have the perfect way to illustrate which is which and how the second one encounters damage.

@atmasphere 
" Therefore I can only conclude that this 'effect' is non-existent "

Do you honestly think I'm blowing smoke or selling snake oil?

Don't worry I'm going to convince you that I'm right when I can figure a way to explain it in your own comfort zone. The problem is that the entire research and cure was done by thinking outside the box.

If the understanding of the 2 speeds present is not seen then we can't get past the first significant aspect of how the distortion was even found.

The only "high level performance" is in my amplifiers.

I think at this point I would be better off returning to this thread when I have put together an illustration of the total process. I will have something I can post to my website as well. I need to have something the average person can relate to.

This is stealing from my lab time (and that's a no no)

Thanks for the back and forth.

Roger

@atmasphere 

" This has been true apparently for some years as there is another thread on this site wherein this circuit is discussed, and that thread is several years old."

This can tell you how long I have been working on the same problem.
The difference now is that I have perfected the solution. As far as knowing it will be correct when the customer gets it - that is in the hands of the automatic-focus circuit. (self correcting).

Here is an example of a failure in logic -
(Kosst) " Circuits vary the pitch through intermodulation distortion through summing the fundamental with harmonics."

I then asked "where did the harmonics come from that mix with the fundamental?" I am well aware of analog distortion including IMD.
The problem overlooked here is that in order to have IMD you must have harmonic distortion to mix with the fundamental. IMD requires harmonic distortion to exist. Since my circuitry does not distort - there are no harmonics to work with. IMD is therefore non-existent. Same as harmonics are non-existent.

Harmonic distortion is born when the fundamental signal begins to move up or down the spectrum. It starts out as vanishing low amounts of phase shift near the fundamental frequency.  For example take a  1khz fundamental tone applied to a "standard" amp circuit. It typically has some amount of energy that can be seen at 2khz (as harmonic distortion)

These 2 energies do not co-exist because it is a continuum. They are time sharing their presence. IOW the small amount of energy seen at 2khz is taken from the 1khz energy. The circuit cannot put out 2 frequencies at the same time so it is alternating. 

Here is the kicker...

At that moment in time we have moved the pitch up by a factor of 2.
For that brief time the amplifier is running at TWICE the speed! You are making my point about how an amp can alter the pitch. For at least that brief moment in time you are injecting 1khz at the input and 2khz is leaving the circuit

The pitch has been altered.

If you can observe the movement of the fundamental slowed down (like with a high speed camera) you would see that the fundamental actually passes through every frequency between 1khz and 2khz. 
When it is seen at 2khz - that small portion of the 1khz signal is MISSING. It has to multiplex to be seen at different parts of the spectrum.
This happens so rapidly it can't be monitored by worthless THD analizers. The reason it is at a multiple of 1khz is because of the repetition rate. It repeatedly runs into the same nonlinear event during the 360 degree range of each cycle of the 1000 cycles. Each cycle literally takes a "hit" along an otherwise linear transfer. This "hit" damages the purity of the sine wave and leaves a DENT. Its SHAPE has been altered. If you zoom in on the dent you will see that it looks more like a small piece of another sign wave that is higher in frequency. The dent is small enough to only time share by the percentage of the dent size relative to the total size of the full 360 degree wave. IOW 1% distortion leaves 99% of the fundamental alone.

Here's how my correction works...
 If you zoom in on just the dent you will see that it has a beginning and an end. At the very beginning of the dent it is just starting to deviate from the shape of the input sign wave. This is the point at which the velocity is first becoming unstable. Since the velocity detector has massive gain it can "see" the signal veering off the track early on and applies red shift or blue shift to force the signal to stay on the original path. Since it is only allowed to stay exactly on top of the traced shape of the fundamental - it cannot slide up the spectrum to become a harmonic.
It cannot (PM) phase modulate or (FM) frequency modulate or make side bands.

The mechanism for generating harmonic distortion has been removed and cannot show up anywhere on the spectrum as a temporary burst of energy. As a result the only thing the circuit can amplify is the fundamental. The would be harmonic has been nipped in the bud.
This technique yields an amplifying method the has the same distortion as air (zero).

The beauty is that it works on a music signal the same way since we are cloning the shape of the signal. There are no harmonics or side bands from IMD do the absence of harmonics needed to mix with the fundamental.

About the velocity detector..
This is what I have spent decades trying to design. In order to "zoom" in on the dent which is tiny requires massive gain in a single spot in the circuit. The velocity detectors output is used to drive the auto-focus stage making it self correcting. This correction is only used along the time domain path (not the vertical path like typical feedback) and with this much pressure the circuit has no choice but to behave flawlessly. A NFB loop is not necessary to remove distortion since there isn't any.

As a result the output shape is a clone of the input shape which includes the wave portion of the sound.

Devices needed to create the gain I needed for the detector are not made. So I make my own devices at the factory. During part of my 30 years working on this one concept I was able to learn how to import a specific property from one device into another device and have it act as if it always had this property as its own.

The amplifier can extract critical data from the input signal at a "DNA" level (between nanovolts and picovolts) including the exact sound of an unlimited number of instruments.The actual layout of the original venue is easily decoded and a sharp focus of every sound object is presented in the playback field with unlimited depth.

There is so much more but the net result is I wanted to match the properties of air so the integrity of the sound wave is not damaged.
Even tough it is an electrical amplifier it "feels" like air to the signal and at the point of conversion back to acoustic energy at the speaker it simply allows a continuance of the flowing sound waves at the same speed as it struck the recording microphone - Mach One.

Roger

@atmasphere 
" Hm. This explanation is very different from the one you gave several years ago."

I have made major advancements since then which includes the automatic focus system.

" One problem with your story is that you don't have a measurement means. That's a problem that anyone with an engineering background will point to"

As a matter of fact I have had to develop my own test equipment since there is nothing available on the market that can measure what I'm looking for. I have completed the virtual analyzer as a computer model that easily shows the degree of resolution expected in the hologram. It gives me telemetry on the auto-focus stage that includes its ability to focus down to what depth in the signal. IOW how sharp the image is and how far I can expect to project an object at a distance and still remain within the capture range of the velocity detector. I needed to know how tight the lock is on the fundamental.

If you increase the physical separation between your speakers - it has the same effect as pulling a projector back away from the screen thus causing the image on the screen to become larger. Since gain and distortion are proportional I wanted the auto-focus to have overwhelming control so that no matter how you have your speakers situated it always projects crystal clear acoustic objects..

As the resolution is increased it enables greater depth-resolution. IOW being able to have a fully rendered sharp image of a triangle at the back of the stage that does not "wobble" or smear do to unstable velocity at that distance.

" It is this particular fact that will send up red flags for anyone with a logical mind. "

As far as the red flags - they belong to the "engineer" that is having trouble trying to fit this concept into a conventional template. Of course it doesn't fit because there are parameters involved that don't appear in the text books they are familiar with. Unless they are forced to think outside the box it will never fit.

Eventually the text books will catch up at some point and include newer methods and techniques to achieve results that until now have seemed impossible. 

@onhwy61 
" I love how he stays "in character" throughout the thread. This is high level performance art! "

Anyone who knows me knows that they will always find me in my lab.
This is my life's work and my passion which is why I have succeeded.

It is easy to stay in character when the character is telling the truth.

"The truth shall set you free."

Here are some comments...

http://v2.stereotimes.com/post/hcat-df100-mk-iv-and-high-fidelity-pro-cables-a-followup/

This was coupled with another product under review at the same time.
It also preceded the most recent breakthrough (higher rez) that is yet to be reviewed.

Air is anything but a constant when dealing with the speed of sound. Humidity and air pressure both play a role. Do you have compensation for pressure and humidity?

There is no compensation necessary. Assume 750 mph as normal. Even if the concert hall all of a suddenly became 20 degrees warmer and loaded with humidity the actual speed of sound may go up or down by a small percentage but where ever it ends up - (751 mph?) it is constant.

For it to have an impact on the velocity tracking that I'm talking about it would have to change (as above) between notes in the performance.
And even then it could only screw up the image for that quick second until it remains stable again at the new value. Temp and humidity generally take a long slow time to modify the speed of sound.

OK- so despite my asking several times it appears you did have a measurement system after all! Why didn't you just come out and say so the first time I asked?

This measurement is something that is applied to the circuit design itself - not to the unit on the bench. It tells me the level of resolution that the built unit will have. But it is not something that you can attach as probes to the hardware itself. Once I have the data given by the virtual analyzer it only confirms the current circuit configuration and settings will be repeated for each unit built to that schematic. 

If I were to try modifying the circuit to increase the resolution - it would have to pass the virtual test measurement first before implementing the mod into production.

I designed the computer model to interact with some of the hooks in the spice simulation software. I also use the Tina software (Texas Instruments) simulator.

I actually had to contact TI and notify them of a bug I found in the software that was falsely reporting raw THD measurements. They have since recognized the problem and have updated their software.

Their technical staff suggested I "send them my schematic" and they will see what kind of distortion issues I have. (Ha!) I said no thanks just fix the program.

Let me use a different word to describe it.

What if I said my amp circuits have constant gain?
Did you know that you can vary the pitch by varying the gain?

What’s hurting you right now is you don’t have any means of proving that what you say is true

I’m not being hurt by this. its just like the quote says "A Failure to Communicate"

The concept of two speeds referenced in an amp is just not registering.
Until there is a way I can tell this in a way the an "EE" can understand it - its not going to happen.

There are 2 axis seen in o’scopes
A sound event has data that is displayed on both axis.

1: The event has height or amplitude (you see a transit that might represent a rim shot)

2: It has a time or duration (see on a scope as the horizontal distance from when the transient started to when the transient ends.

If you want to know how long it lasted you would adjust the horizontal sweep rate to "spread out the display so as to get an accurate time or duration of this event.

Hopefully there is no disagreement with this statement..

Take the slew rate for example. It is defined as volts per microsecond.
Volts = height or vertical distance traveled
Microsecond = Time it took to swing that many volts from beginning to end.

I don’t want to confuse you by talking about slew rate - I just want to separate the vertical axis (and what it represents) from the horizontal axis (and what it represents)

When audiophiles remark that "wow this is a fast amp" they are generally talking about the ability of the amp to put out quick transients (tiny bells, upper keys on a piano, triangles etc.)
Clearly they are talking about it being faster than perhaps another amp that can’t "present" the top end as well. The latter may have a worse slew rate or some bandwidth limitation that does not allow the extension of the upper part of the spectrum.

You were closer to understanding what I’m talking about when you refereed to propagation delay or group delay.

If I said that the INPUT signal (source) has a speed (before it even begins to travel through the amp) would that make sense to you?

I believe a have a way to describe this with more clarity.

I agree 100%
I do not have a single device with perfect linearity because it does not (currently) exist.

The harmonics arise from unavoidable variations in gain.

Thank you - this makes my point that you can alter the pitch by varying the gain.The harmonics are the result of sliding the fundamental to a different area of the spectrum.

A change in gain is a change in velocity.
When you stabilize the velocity - you stabilize the gain.
A stable (constant) gain is linear.
A linear amplifier has no distortion.

What I have is a circuit that is 100% linear. (made from non-linear devices).

All that is needed is to use the devices in such a way as to force the output to be linear on a scale that is inconceivable.

A word about my previous comments...
I want to walk back my statements about the text books.
before you get too excited its not for the reason you think.

I’m sure that everything I have done can be found in the text books as individual or separate phenomena. What is not in the books is the composite use of various phenomena to produce a result for which there is no reference or example.

I have developed a way to take advantage of known phenomena involving aspects of how sound waves flow in air and created a circuit that treats the (sound wave) data as if it was in the acoustic environment.

This successfully "feeds" the brain in such a way as to believe these sounds are real (live) and happening in your airspace.

Kosst
 

The problem I've got with the concept of "perfect" or "zero distortion" amps is that it really doesn't matter. How perfect is that amp with a loudspeaker drawing current all out of phase, pumping power back into the amp, and presenting impedances all over the chart? Is it still putting out that perfect signal?

Golly if I had known this I could have given up years ago and saved myself a lot of work.

kosst_amojan,

If designers are discouraged in making improvements or trying to "perfect" a component because another component has its own issues then there would be no progress made anywhere.

Since I have succeeded in cleaning up distortion in amplifiers I guess I'll have to address the speaker issues next. However it might surprise many readers here that the bulk of ALL issues in a given playback system are in fact tied to the electronics. If you had the perfect amplifying method - a pair of Radio Shack book shelf speakers would still give you incredible imaging.

Its the electronics that provides 90% of a systems quality.

Sorry boys your wrong. This is what you believe because you have no reference when it comes to hearing what a distortion free amplifier can do.

I have the evidence of what I speak. I realize it might be hard to imagine that such a breakthrough in sound reproduction can come along but its here.

I have actually cloned the original sound event (concert hall) and I can drop that air pattern into your living room. So help me God.

Stay tuned to the news.

Don't you think that after working on this method for 30 years that it might be something that you would at least give the benefit of the doubt to?

Its not like you went to see for yourself what this could be even if you think it is a hoax or BS. Your looking at this from the wrong end of the telescope. I'm not saying "I think" I can make a cloned copy of an event. I have the working technology already up and running.

You guys seem like a plethora of naysayers and Debbie downers.
Your wasting energy trying to tell me that I cannot do what I just did.
I'm not looking for any special praise or kudos - just to be heard and taken seriously.

The correct answer is "wow this sounds interesting but I would like to hear that for myself"
Instead "Your sounding like a crack pot or nutty professor or delusional"
"You can't do that because...bla bla bla."
"Face it we all live with some degree of distortion in the system"
And then you list 10 things that prevent me from succeeding.
All of which I have overcome.

Every attempt at "3D" without exception has involved some type of parlor trick or "enhancement" circuitry to give the listener the illusion of "surround" sound. 

Zero distortion is 3D.

I'm officially blue in the face so I think this will be my last post.
I want to thank the "other" readers of this thread that did not jump in with reasons I'm going to fail.

Seriously - good luck to those that enjoy the pleasure of listening to music without the hassle of politics or prejudices brought on by brand loyalty or designer gurus.  

Even Fox News thought this was a hoax (at first).

See you in the future.