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

willemj

I think the engineering science to design and produce a straight wire with gain reached sonic perfection in the 1980's.

I would agree with this except I would say it reached its sonic "limits" in the 1980's. It never reached perfection. The need to be energy efficient is a noble cause as well however I think that removing distortion found its limit when it was clear that lower THD measurements, while impressive to those buying by specs only, proved to be seemingly unrelated to the actual sound of a system. Tube gear (with admittedly higher THD) still dominated the high end market.  Even today the tube gear still enjoys a comfortable percentage of the high end market. Many SS designers would be happy if they can get their gear to sound like tubes.

kosst

Where exactly are you getting perfect signals?

One of the biggest surprises I've run into is how much information is actually captured in the recordings. The resolution of the image embedded in the source is vastly superior to what was assumed as a limitation. The real problem seems to be with the resolution of the playback system.

If you recall I was offering to send you a preamp - not an amp.The entire playback chain is the amplifying process. The core process I have developed is zero distortion. This circuitry is an analog "block" that you use to make a phono stage, a DAC (output stage), a line stage and the voltage gain stage in the power amp. The output section passes the phase coherent signal to the loudspeakers. There will be limitations when driving speakers that do not adhere to the same coherency do to its design. Obviously you must have at least decent speakers.

Perfect mediums? Which perfect medium exactly?

If you are asking about the medium of air - there is no distortion presented to the sound waves traveling through air. The two times you use air as a conduit starting at the original venue is the air space in  studio or hall and the air space in your listening room. These two segments have no distortion. The electronics between those segments has to approach the same level of purity found in air.

For example just by switching the preamp in an existing system of otherwise decent electronics can give you many more magnitudes of resolution and information compared to a conventional preamp. The same holds true for the phono stage. Without replacing your power amp.

I'm sorry for the confusion but when I talk about amplification I am not always referring to a power amp.There are enough problems before you get to the power amp that already restricts what you can expect to be passed on to the speakers.

what perfect gain devices are you using in your perfect circuits?

Here you are talking about two different things. It is possible to have a perfect circuit without perfect gain devices as long as the result of a unique configuration produces the perfect output. I also manufacture my own devices used in critical areas of the circuit because they don't make a device capable of behaving the way that is required to cause the overall circuit to have a linear output. My target was to detect velocity in the source signal and force the output to be synchronized to the recovered velocity which is clearly embedded in the recording. This gives you a way to match the playback speed with the recorded speed. If you don't do this you end up with an unstable or smeared image. This "out of focus" nature of such an image hides the extremely fine details missing in playback. The more stability - the more resolution. This can be seen by simply reducing mechanical vibrations in the system as well as better overall grounding which also helps to stabilize the image.

What is perfect gain?

Perfect gain is gain that does not change under load. Again I'm not talking specifically about a power amp. The desired holographic image will collapse do to any non linear segment in the entire chain. Perfect gain in the analog world is a fixed numeric value that is used to amplify the input signal at any point in the 360 deg range. If you take a line stage that has 6db of gain or an amplification factor of 2.0 then of course to be linear it would need to stay at 2.0 during the entire dynamic range.

If as the signal approaches the first positive peak and it falls short by a tiny amount (like the amplification factor at that moment was 1.998) then this is a sign that the amplifier has literally slowed down and the signal has a degree of compression that makes the shape of the current signal (fragment) look like a lower frequency. The same holds true for any segment in the 360 degree signal that somehow ended up with a gain of 2.002 - in this case the speed of the amp has gone up. The expected value of the peak has been passed and is seen as an acceleration in velocity. This non linear nature is what causes the playback speed to vary. Harmonic distortion is the direct result of accelerating the fundamental frequency to a higher part of the spectrum.

By detecting the velocity rather than trying to make a crude "comparison" as in classic feedback attempts - you simply hold the velocity constant. Constant velocity is constant gain. Constant gain is linear. Under these circumstances, there is no way for it to cause the speed to vary and as a result it does not have the ability to generate harmonic distortion. Holding the velocity constant is done as a phase correction. The greater the ability to detect velocity gives you better ability to hold it constant. This type of correction is done along the horizontal or time domain axis - not in the vertical axis the way classic feedback works. It also does it in real time preventing any form of distortion from appearing at the output of the circuit. This is what makes this method of amplifying distortion-less.

Detecting velocity is extremely difficult and requires sensitivity of astronomical proportion. This is what took me years to figure out.
This high gain detector drives the automatic focus system 

Advances in the velocity detector over the years has given rise to resolution as seen by the version nomenclature like the X-8, X-9, and so on. It currently stands at X-12 and is the final version because at this level of correction - the detector now gives 100% control to the auto-focus system. No additional correction is necessary.

The detector can discern changes in velocity as small as a few micro-degrees. As a result the output of the circuit can be totally phase locked to the fundamental (input) signal. As long as the lock holds - it produces a clone of the input signal. The lock is good down to (and below) the noise floor. When no signal is present the auto-focus can maintain a lock on the actual Johnson or shot noise seen at the noise floor. Any music signal rising up from the noise floor is already totally locked. This means that the auto-focus can accurately project tiny sound objects in the background at tremendous depths while providing massive detail for anything in the sound stage at any distance. It does this with 100% transparency  starting from a jet black background. Percussion instruments are startling and because the brain is use to listening to live music coming through air - it readily accepts sound waves with similar stability as live. 

The average person listening to a full system that works this way for the first time can't make it passed the first 10 seconds without looking around and saying "what the hell is this?" they cannot wrap their head around what is taking place because it seems both impossible and live.

Although a conventional system can sound spectacular - after listening to a system that does not distort - and returning back to the conventional system - it now sounds distorted. 

Here is a simple test for distortion:
Can you tell if your listening to tubes or solid state?
Do you recognize a 12ax7 or a mosfet?
Does it have a "signature" sound?

If you can recognize any device used anywhere in the system - its distorted. It is leaving a finger print of that device on the music signal.
You cannot hear H-CAT. It has no sound. By re-enacting the disturbance pattern of the original air space in your own listening room you have cloned the sound event. The result is pure music with instruments suspended in mid-air as a ghost like image.

I rest my case.

Roger

stfoth,

Early on in the project (years) you can see the THD drop to low levels by making a specific part of the circuit take over the handling of the input signal.
Once the distortion hits the noise floor - two possibilities exist. If the noise is seen on a spectrum analyzer is -100db  then the distortion is either -100 as well (or less). Listening tests at that point reveal a caliber of resolution that can be associated with those numbers. After that point any further improvements (hearing more resolution/less distortion) is happening below -100db. If you monitor it over time you can see the random nature of the noise will occasionally drop below -100 for a brief time. When this happens it exposes the harmonic measurement which if it was -100 then even in the absence of noise it would still read -100.
Instead it measures -130db. This quick peak indicates that the distortion is still being driven to deeper levels and you can verify by what you hear that you are going in the right direction. After that point the THD analyzer is of no use. Now we take over with the math to determine how far down it wold be. Because of the phase lock kicking in and stopping it from being able to generate harmonics - you now can use the degree of resolution as a gauge going forward. As I continue to raise the sensitivity of the detector by X amount - it translates directly into an increase in resolution. It is obvious when resolution goes up - other instruments that you never heard before are now "visible". Again, continuing to raise the detector output gives you the numbers needed to calculate the degree of lock presented to the fundamental. At this point it no longer can distort at least as seen by harmonic output. Now we are down to how much bandwidth are we limiting the fundamental to vary. As the detector continued to improve I can calculate how much phase shift is now allowable as far as deviation from the fundamental frequency. Since I know that the detector sensitivity can be triggered by as little as a few micro-degrees, It ensures that the smaller the allowable phase shift - the tighter the focus. Remember the detector drives the auto-focus.

The correction uses phase shift countermeasures that are extremely tiny and guarantees the fundamental is now the only thing that can exit the stage. The red shift / blue shift torque is held at a balanced point by a hair trigger which is the velocity detector. All of the correction as I mentioned is done on the horizontal axis. (time domain line). To do this the detector has to be rotated by 90 degrees so that it is literally seen by the music signal as a path that it must travel through to reach the output. (unlike monitoring the voltage or amplitude levels as in the vertical axis).

Trying to use negative feedback driven by amplitude measurements is no way to accurately correct anything. As you may know by the time you get this "sample" of output to use as a countermeasure its too late. That "piece" of music already left the circuit as distortion.

By using 90 degree phase detection capable of seeing a micro-degree of phase shift you have plenty of time to fix the problem in real time. Once this process takes over and we know the sensitivity of the detector we know the max deviation from the true fundamental. At this point it is right in the ballpark of the same phase shift you would expect sound to experience traveling through air. (virtually zero).

At this level of phase purity we have emulated the linear property of air.
The sound of the unit now seems more like a hole in the wall or portal through which sound waves are allowed to pass through unaltered.
The hardware is cloaked.

Hope this helps.