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?
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?
I'm having a problem with several other comments here as well. One was the speed of the power supply and another was the speed of the amplifier. Long ago I discovered that any power supply has a timing constant and if the amplifier goes lower than that constant IM distortion will rise. So that is a rule I've been careful to follow. But from your comments it sounds as if you are talking about something else.
Also the speed of the amp is another area that is well understood. Are you referring to the speed of the amplifier as propagation delay?
I think both @atmasphere and @roger_paul deserve a lot of credit even if for nothing more than being so passionate in their endeavors. As Albert Einstein once wrote on a blackboard: “Not everything that counts can be counted, and not everything that can be counted counts.”
@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.
Hm. This explanation is very different from the one you gave several years ago.
I'm very open to the idea that there are new vistas to be explored. That's why I have patents. But I think you need to get your story straight.
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; and like me they will find it odd that you've had a circuit to compensate for this 'effect' but you've not developed a means of measurement. Anyone can see that such will lead to QC problems.
So my advice to you is to develop solid repeatable measurements. You already have the circuit (allegedly); so it itself is your means of measurement, as if that was not abundantly clear earlier. It is this particular fact that will send up red flags for anyone with a logical mind.
It wouldn’t be the first time someone automatically called something snake oil before coming to grips with what the idea actually was. Folks sometimes appear programmed to jump on just about anything that deviates a little too much from the standard model. Like ducks on a June bug.
"
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.
Do you honestly think I'm blowing smoke or selling snake oil?
Since you have consistently avoided answering a rather simple question, one that **should** have been very easy, I'm forced to conclude regardless of what I want to believe that the correct answer to the question above is 'yes'.
Here's why: you claim to have a circuit to correct this 'speed' issue. At this point we don't have to know anything technical about it; its mere alleged existence points to a means of quantifying the 'speed' of an amplifier circuit, which it then corrects. 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.
But apparently despite the existence of this circuit, it has not helped in the means of measuring the speed of the amplifier circuit, which it surely must do, otherwise how could it correct; for that matter you would have to have a reference to know how to set the 'speed'. The logical issue here is obvious: obviously the 'circuit' does not exist. Otherwise you would have a specification to look for so that when you shipped a product out the door, you would know it was correct.
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)
@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.
"
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.
Alright- so several times I've asked the same question and met with obfuscation. So I have to assume that you (Roger) have been unable to quantify the effect you are talking about.
Since you claim to have a circuit to compensate for this effect, the means to quantify the effect is apparently at your fingertips and this should be obvious to the casual observer.
Therefore I can only conclude that this 'effect' is non-existent. Otherwise you would have already told me how you quantify it. Kosst-amojan stated the exact same problem in a different way in the post just prior to this.
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."
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.
Your analogy of analog tape does not hold water when applied to amplifiers. Amps don't cause pitch variation. Speed variation in analog recorders does.
From your description, if I had to guess its almost as if you are describing the speed of the amplifier in terms of propagation delay- that the delay from input to output must be at the speed of sound.
But I suspect that interpretation is not correct either...
But at any rate you've not answered the original question- which is: have you got a means to quantify this?
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.
As I understand it Roger, you have a means of detecting this 'velocity'
but what I found peculiar last time I engaged in this topic was that you
had not quantified this velocity as a specification. I'm pretty sure
you're not talking about risetime/slew rate.
The question is, have you quantified this circuit operation with a spec? Or put another way, have you quantified the 'velocity' spec?
At least you confirmed one thing- that you are not talking about risetime/slew rate.
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?
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.
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.
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.
That is the question I asked before and got a long answer that seemed to contain neither yes or no. Have you quantified, do you have a spec for this circuit?
I am utterly fascinated by what you have written. What you say is interesting insofar as comparatively I recall going to a talk by Nordost and Vertex cables when they explained that they looked at sonar technology in designing cables which has not been done before - likewise LAvardin make their designs to get rid of 'memory' in solid state circuits. I look forward to how this develops and if you/colleagues will make such a design available for jo-public.
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.
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).
Although I find much of this amusing, I do agree that speed is important. Our amps don't sound like tube amps partially because they are so fast; there are very few solid state amps that are as fast as our amps. In that way our tube amps differ quite a lot from the vast majority of tube amps.
As I understand it Roger, you have a means of detecting this 'velocity' but what I found peculiar last time I engaged in this topic was that you had not quantified this velocity as a specification. I'm pretty sure you're not talking about risetime/slew rate.
@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.
hi @ roger_paul 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? It is fascinating that single ended amps and valves - particularly OTL designs have such legions of fans and they major on the reproduction of a live sound.
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.
I see two things improving these days, DACs and class D power amps.
Other than incorporating new, more capable DAC chipsets into new preamp models, I agree with others who have posted here, that the preamp itself isn't really improving. In my case, I have never used the DAC in my preamp, I went with an external unit, one based on an FPGA chip, where firmware updates improve the performance of the DAC, like getting a "new" unit every year or so!
I must admit that it's been years since I auditioned a class D power amp, and back in the day, was not impressed with the sound (to me it sounded "brittle"). With all the new class D designs, eventually I'll take the time to update my experience with class D and audition some of the new amps.
"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".
@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.
One of the best designers of pre amplifiers and amplifiers Kara Chaffee designed a couple of tube preamplifiers. The Ultra Verve uses the 6SN7 tube. The key with this preamplifier is to let the tube do the talking. Keep everything so simple, class A single ended. Just get the most out of the tube.
The point. Yes things have slowed and they should slow until there are real advancements in the way music is recreated and reproduced.
I guess in terms of sound quality - gains will be marginal, but the means with which to deliver the results will keep developing - Class D and Class T amps being two such examples. My opinion is that with 'improved' technology logic will dictate amps will become: 1. Smaller and 2. More efficient I think it is arguable that such amps may well have 'room correction' technology built in without the distortion affects of tone controls. I thought that the likes of Tact/Lyngdorf was going to take over the world so to speak but oddly - as a company - they didn't - although digital amps appear to be the way to go for the majority of householders
Think most missed Nelsons real point- {largely solved on a practical level} amplifiers have existed for decades that can run a loudspeaker reliably affordably and efficiently.
Y'know, I can agree with Mr. Pass. "The beauty lies in the differences'. That holds true for any and all of the components that we apply to stroke our personal preferences and situations; your tastes, your listening space, what you listen to, and how... What I like may abhor you, or nod in general agreement.
"There is only one black and white." No.
Like the shades of gray between them, even the extremes can be debated. The validity of that is demonstrated every day in the posts and pages of these forums. There is not, nor will be, a 'one size fits all'.
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