Compilation of WBF Discussions with Roger Sanders

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#1

Answers as promised; with input from Roger ~

IMAGING:

The direction of the sound you hear from a speaker system is determined by the location of the surface from which the sound emanates. A typical, wide-dispersion speaker radiates its sound out into the room where it bounces off various walls and objects in the room.

You hear these reflections as well as the sound from the speaker. This is especially true of the very strong the reflection off the wall near the outside edges of the speakers. Therefore, when listening to wide dispersion speakers, it is common for the sound and image to appear to be wider than the spacing between the speakers because you are actually hearing the sound coming off the walls beside the speakers.

In a narrow dispersion speaker, there are no room reflections -- at least none that are significant enough to affect the speaker's image. So you hear the sound coming directly from the speakers. Therefore, the width of image will be defined by the space between the speakers. A truly holographic, 3-dimensional image will appear to float in space between the speakers.

If you want a wider image, simply place the speakers more widely apart. Note that narrow dispersion speakers can be placed as widely as you wish without developing that dreaded "hole in the middle" of the sound that you get from wide dispersion speakers.

Because reflected sounds ("room acoustics") degrade the sound coming from the speakers, any speaker that generates an image beyond the outside edges of the speakers will produce a less-than-perfect image. The speaker needs to confine the image to the space between the speakers for best performance. For a discussion of this problem in detail, please read Roger's white paper on the subject at Dispersion White Paper


DIFFERENCES IN SPEAKERS BETWEEN INNERSOUND AND SANDERS:
There are certain fundamental design parameters that have found to work best for all types of speakers. Therefore, all speaker types will have certain features that are similar.

For example, most magnetic speakers consist of a cone-shaped woofer, a dome-shaped tweeter, passive crossovers, and sealed box enclosures. Although most speakers are made in this format, they all have various subtle differences in this basic package of design features that make each manufacturer's offerings somewhat different in sound quality and performance.

Electrostatic hybrid speakers are no different in this regard. They are all vertical line sources, with an electrostatic panel on top of the woofer, so they all appear to be similar. But there are a great many detail differences within this basic layout. Like magnetic speakers, these differences can make a lot of difference in the sound and performance of speakers that appear superficially to be quite similar.

In the case of Innersound speakers compared to Sanders speakers, there are major improvements in the power handling and output of the woofer, the electrostatic panel is now utterly reliable and will play at ear-bleeding levels with multi-thousand watt amplifiers, the speaker is modular in construction (no more truck shipping), the styling is improved, real wood is used, digital electronic crossovers are available, the crossover point is lower, the crossover slopes are steeper, electronic time-alignment is used for the first time, all aspects of the system are user-adjustable, the speakers come with redesigned and more powerful amplifiers, certain internal resonances have been eliminated, woofer cabinet internal design has been improved, lifetime warranties are standard, factory support is vastly improved, and the price is lower. So it is fair to say that the Sanders speakers are quite a lot improved and different from Innersound speakers.


HISTORY:
Here is a very brief review of the key elements of the Innersound/Sanders history: Innersound, Inc. was founded in 1996 by Raj Varma who lived in the U.K. He hired Roger Sanders to design electrostatic speakers for him. Eventually, Mr. Varma expanded Roger's role to run Innersound in the U.S. while Mr. Varma manufactured computers in the U.K.

The key element here is that Roger was an employee of Innersound and never owned or controlled the company. So when Gary Leeds wanted to buy Innersound, the decision to sell was made by Mr. Varma -- Roger had no say in the matter.

When Gary Leeds bought Innersound, Inc. (February 2003), he dissolved the company and started a new one (Innersound LLC). He allowed Roger to become a minority share holder in the new company.

But as majority share holder, Mr. Leeds maintained total control of the company and once again, Roger could not manage the company in the manner he would have liked. This eventually led to Roger leaving Innersound, LLC in July of 2004.

After leaving Innersound, Roger started his own company (Sanders Sound Systems, LLC). Shortly thereafter, Innersound, LLC went out of business.

But before doing so, Mr. Leeds asked Roger to take care of Innersound's customers. Roger agreed to do so and Mr. Leeds transferred critical Innersound parts inventory to Roger so that he could service Innersound's products.

Note that although Roger does provide service for Innersound's products, Innersound's warranty is no longer valid since Innersound is out of business. In any case, enough time has passed that any warranty on Innersound's products has expired. Therefore, Innersound customers have to pay a modest price to have their equipment serviced by Sanders Sound Systems.

In summary, there have been three distinct companies. There was the original Innersound, Inc., which was owned by Raj Varma. Gary Leeds bought Innersound, Inc., dissolved it, and started a new company called Innersound, LLC. He eventually dissolved that as well.

Roger Sanders started Sanders Sound Systems, LLC. Although Roger worked for both of the Innersound companies, his company (Sanders Sound Systems) is not associated with either of the Innersound companies, both of whom no longer exist.

By having control if his own company, Roger can now operate it as he deems best. For example, he can now offer lifetime warranties and risk-free, 30-day, in-home trials. He also sells both factory direct and through selected dealers.

He makes all the engineering decisions based on performance, instead of on the approval of company owners. And he makes improvements in the design of his equipment when he wishes as technology advances.

Unlike the Innersound companies, Roger handles Sanders' company finances so that there is zero debt, there is cash in the bank, inventory in stock, and overhead costs are extremely low. As a result, Sanders Sound Systems can handle difficult economic times easily, successfully, and with greater security than his larger competitors.

While the Innersound companies could not have survived the current economic recession, sales and income for Sanders Sound Systems have been growing during this time. This is proof that the Sanders' business model is successful. The future of Sanders Sound Systems is secure.

 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#2
Why ceramic slots instead of little holes?

Is the slot really a slot longer in one dimension? If so does that invite some flexing of the mylar membrane? If flexing or rippling does this add something to the music?
I asked Roger about this and his first reaction was "good questions". He put some time into responding this morning and here is his reply.

There is no practical difference in the sonic performance of slots and holes. They both work equally well as stators in an electrostatic speaker. But there are some very important pros and cons in the production of parts made with either slots or holes.

There are thousands of holes in a typically-sized stator. When you can punch hundreds of holes at a time using a punch press, the process of producing all these holes is reasonably fast.

But if you have to drill holes using precision CNC machines, the process of drilling thousands of holes takes far too long to be practical. By comparison, machining slots is much faster.

A punch press cannot be used to punch long, thin, slots as the material is severely deformed by the punching process. So slots must be machined.

In summary, if you want to make a stator from a metal sheet, you will use a punch press and make holes in it. If you want to make a stator from other materials that have encapsulated conductors, you must machine it to obtain sufficient precision. You will therefore make slots.

The membrane in an electrostatic speaker is driven uniformly over its entire surface. It operates as a piston, even at high frequencies. This is one of the reasons why there is virtually no distortion from an ESL. As long as the holes or slots are small in comparison to the distance between the diaphragm and the stator, they will form a uniform force field around the diaphragm and there will be no flexing or rippling of the diaphragm's surface.
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#3
I sent Roger your questions and here is his response.

The Model 10c can be operated as a full range ESL. You can adjust its electronic crossover (or eliminate it entirely) to produce full-range operation anytime you wish.

So why do I prefer hybrid operation? Simply because I am not willing to compromise performance. You cannot beat the laws of physics. The fact is that the bass performance of a full range ESL is poor.


A speaker must do several things to perform well. In addition to obvious things like linear frequency response and fast transient response, a high performance speaker must also be able to play loudly enough to reproduce music and all its dynamics at realistically life-like levels. No full range ESL can do so.


High output is a tremendous challenge for an ESL. But technology has advanced to the point where it is now possible to make an ESL play at ear-bleeding levels from the midrange on up. But no full-range ESL can produce deep, powerful bass that can match what you hear from live sound.

Only magnetic speakers have that capability.

There is more to this issue of bass quality than just the amount of deep bass available. Not only is the quantity of electrostatic bass inadequate, but the QUALITY of electrostatic bass is dismal. Let me explain that statement.


One of the truly wonderful things about ESLs is their low Q behavior. By way of review, the engineering term "Q" refers to the "quality" of the sound with respect to control and damping.


In other words, a "low Q" driver exhibits very well-controlled behavior with fast transient response and without any overshoot and ringing. A "high Q"

driver is poorly controlled, has resonances, overshoots and rings after an electrical impulse, and has poor ("smeared") transient response.

It does not stop instantly as it should. Instead it vibrates for many cycles a relatively long time until it finally comes to a complete stop. It "rings" like a bell. This behavior adds extraneous frequencies to the original sound that corrupts it and makes it sound very unnatural.


The massless nature of an ESL means that it can accelerate instantly to follow the musical wave form. Even more importantly, the mass of the air around an ESL totally swamps the incredibly tiny mass of the ESL's diaphragm, so it cannot "ring" and resonate. It simply stops instantly, which gives it great clarity, detail, and transient response.


This is like trying to ring a bell underwater. It won't ring because the mass of the bell is swamped by the mass of the water around it. It simply is no longer free to vibrate and ring.


Therefore an ESL has very low Q behavior. This is one of the reasons it sounds so "tight", "crisp", "quick", and can extract the most subtle details from the sound and reproduce transients flawlessly. Low Q ESLs simply sound more "real" than high Q drivers.


However, this wonderful, low Q behavior of an ESL does not apply in the bass. An ESL is built like a drum, which is an extremely high Q device. An ESL has a stretched, "springy" diaphragm. The diaphragm is relatively large, so couples with a large amount of air in the room.


Air has mass. The relatively large mass of the air to which it is coupled will cause the ESL to resonate like a drum at some low frequency based on the mass and spring rate (tension) of the diaphragm. In ESLs of typical size, this resonance occurs between 50 Hz and 100 Hz. Tap on the edge of any ESL and you will hear this resonance.


This is the fundamental resonance of the ESL. It is the only resonance in an ESL, but it is huge (typically 16 dB) and like all resonances, it is extremely high Q.


Being high Q, frequencies around the fundamental resonance will be very poorly controlled and have a large amount of overshoot and ringing. The sound in this region will therefore sound very different from the low Q sound of the ESL everywhere else in the audio bandwidth. The result of this high Q behavior is that the quality of electrostatic bass is flaccid, poorly controlled, sloppy, and boomy.


Yet another problem with ESL bass is that it is very non-linear. The fundamental resonance causes a large peak in the bass followed by a sharp drop in output below it.


Most full range ESL manufacturers allow this peak to persist in a desperate effort to get their ESL to sound "bassy." But the result is very non-linear

bass response, with no truly deep bass output at all. It becomes a "one
note wonder."

In short, electrostatic bass has inadequate output, poor frequency response with no deep bass frequencies, and its high Q behavior results in poor bass quality. Therefore, I don't consider electrostatic bass to be satisfactory for a high performance speaker.


The solution is to use magnetic drivers for the bass. Of course, bass from magnetic drivers is less-than-perfect too. While magnetic woofers can produce great power and depth that no ESL can hope to match, and they can be made to have excellent frequency response, they also have high Q behavior (although not nearly as high as the Q of the fundamental resonance of an ESL). So what can be done to fix the problems of magnetic drivers so that they can mate well with an ESL?


Much of the problem with magnetic woofers has to do with their enclosures.

The typical closed box (infinite baffle) and vented (bass reflex) enclosures greatly raise the Q of the driver and are largely responsible for the bloated, boomy, and muddy quality we usually hear from most magnetic woofers.

However, it is possible to eliminate these faults with the use of a transmission line loading and other techniques. This is another, long technical discussion that I do not have time to describe in this opus, but anybody who wishes can phone me (303 838 8130) for a detailed analysis and engineering solutions.


The bottom line here is that it is possible to achieve deep, powerful, low Q bass from a dynamic woofer. I do so by using transmission line loading, custom-built low mass and low Q drivers, magnetic damping systems, direct coupling to high damping factor amplifiers, and eliminating passive crossovers with their series inductors. The performance from such a magnetic bass system is far superior to electrostatic bass because it is loud, dynamic, has no resonance, is low Q, and has flat frequency response right down to 20 Hz.


The woofer systems in traditional hybrid speakers have failed to match the low Q behavior of an ESL. Furthermore, their passive crossovers have not had steep enough slopes or low enough crossover points to eliminate the woofer from affecting the critical midrange.


This midrange issue is very important. After all, we would all agree that no matter how good a woofer is, it simply cannot match the magnificent detail and delicacy of an ESL in the midrange. A superb hybrid must eliminate all woofer energy from the midrange. Traditional hybrids have failed to do so and therefore have earned a reputation as being unable to integrate the two different drivers.


But it does not have to be that way. I have resolved this integration problem by having developed a low Q, magnetic woofer system that has better sound quality and frequency response than any electrostatic woofer. I have eliminated the woofer from the midrange by using very low crossover points combined with extremely steep crossover slopes (48 dB/octave). I use electronic crossovers and bi-amp the system. The result is a no-compromise hybrid that easily out-performs any full range ESL by a huge margin.


This makes it possible for you to reproduce Row A concert hall levels and the sound of a grand piano or drum set at live volume in your listening room. All while maintaining the magnificent detail, transient response, and clarity of an ESL. Full dynamic range and spectacular performance is now possible.


So to answer your original question, I use a hybrid because it is the only way to achieve the high output, full dynamic range, flat frequency response, and low Q performance that is essential for a high performance speaker. A full range ESL simply cannot match the performance of my hybrid design.

Because I want the best performance possible, I do not manufacturer or sell full range ESLs anymore.

Using a hybrid is not a cop-out. Using a full range ESL is. It is much easier and cheaper to make a full range ESL than a quality hybrid. But I'm not willing to sacrifice the best performance for the sake of lower cost and ease of manufacturing that full range ESLs offer.


My Ultrastat panels can be ganged to make huge, full range ESLs. I have a few customers who have done so. But the laws of physics cannot be circumvented. The fact remains that ESL bass is of poor quality and simply cannot supply realistic sound reproduction.


I am not willing to deceive my customers to make sales. So I do sell inferior products like full range ESLs.


Turning to your question of dispersion, there is insufficient time to cover this topic in detail as this response is already far too long. If you have not already done so, I suggest you read my White Paper on the subject for a thorough explanation of my logic and reasoning behind my use of narrow dispersion panels. There I explain why I abandoned the wide-dispersion, curved ESL that I invented in favor of narrow dispersion, planar panels.

The link is:
http://sanderssoundsystems.com/technical-white-papers/dispersion-wp


In closing, let me say that once you hear my speakers, you will understand.

You are welcome to use my 30-day, in-home, free trial so that you can hear my speakers in your own home with your own associated electronics and familiar source material. You can then judge my engineering decisions for yourself and report your findings to your friends.

Great listening,

-Roger
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#4
Roger-

Thanks for taking the time to give us insight into your speaker designs and philosophy. I also think it's interesting to see all the people in the industry who contributed to the ML design--and certainly taking estat reliability a few notches up.

I know you can't comment on other manufacturers designs but Roger West of Sound Lab also talks about estats' problems in the lows, namely the "drum head" resonance you referred to and dipole cancellation effects (like what the Maggies run into). But had you considered an approach like Roger took to the resonance problem?
reply from Roger

Hi Myles,

When you refer to Soundlab's approach to the resonance problem, I assume you are referring to Roger West's "distributed resonance" design. By way of review for readers who may not be familiar with this technique, this is where the speaker's diaphragm is broken up into sections of various sizes.
The idea is to produce several different, smaller resonant frequencies instead of one big one.

In theory, this technique should not only reduce the problem of extremely bad ESL bass frequency response, but it could be used to compensate for phase cancellation, which is a major reason that ESLs can't reproduce loud bass.

The different sized diaphragm sections are produced by using diaphragm to stator spacers that are at different distances from each other. Martin Logan also uses this construction method, although Soundlab claims to have a patent on it.

I have experimented with this technique extensively. Unfortunately, it has an unforeseen problem that prevents it from working as expected.

The distributed resonance idea works for isolated, widely-separated, electrostatic panels that can operate as individual drivers. Each will exhibit its own, unique resonance depending on its size and diaphragm tension.

But when you take those same drivers and bring them together to form a single speaker, something weird happens -- the panels now operate as one.
They no longer have individual resonances.

The reason for this is that the panels are immersed in air, which has significant mass. At audio frequencies, the air mass around an ESL takes on the consistency of a gel -- or at lower frequencies, a liquid. The air mass around the speaker then vibrates as a unit (as a bowl of Jell-O would vibrate as one mass), preventing individual resonances from forming.

In other words, distributed resonance does not work. You cannot break up the fundamental resonance of an ESL into multiple discrete ones.

However, the different spacer distances do have an effect on the speaker's fundamental resonance because the diaphragm no longer behaves as though it were at a constant tension. This reduces the magnitude of the fundamental resonance widens its bandwidth. So it does help to improve the massive irregularity of the frequency response, although at the expense of making more of the audio bandwidth adversely affected by the resonance.

Note carefully that even if the technique of distributed resonance worked, the bass performance would still exhibit high Q behavior with its awful effect on bass quality. After all, by definition, all resonances are high Q, and by having more of them over a wider bandwidth, you would simply be introducing the bad, high Q behavior of a resonance over more of the audio bandwidth. This certainly degrades the performance of the speaker.

Finally, distributed resonance would not solve the problem of poor bass frequency response. Instead of having one resonance, you would now have many. So the frequency response would be ragged over a wider bandwidth.

In fairness, the case could be made that the magnitude of the multiple resonances would be smaller than that of a single resonance. Some might find that preferable. But the fact remains that the frequency response that consists of a series of resonances would still be far from linear.

In summary, I find that distributed resonance does not work as claimed.
Furthermore, it degrades the otherwise lovely, low Q, linear frequency response exhibited by ESLs over the rest of their frequency range.

There are only two ways I have found to eliminate the high Q behavior in the bass of an ESL. The first is to avoid driving the ESL near its fundamental resonance. A hybrid does exactly that. But there is a second method that can be used in full range ESLs to make them have low Q behavior over their entire bandwidth.

I have been experimenting with motional feedback. By measuring the motion of the diaphragm and comparing it to the musical signal, the errors caused by overshoot, ringing, and frequency response errors can be identified. An error signal is thus produced that can be fed to the amplifier that is driving the ESL's diaphragm to actively stop the ringing and overshoot.

This works quite well. It not only stops the high Q behavior, but can be used to produce perfectly linear bass frequency response. It will also compensate for phase cancellation. Nice!

But motional feedback has serious problems with implementation. The major one being that a microphone cannot be used to produce the error signal as the delayed sounds from room acoustics get into and confuse the electronic system.

A laser can be used to measure the diaphragm motion. It is immune from room acoustics since it is not measuring air motion, but the motion of the diaphragm directly.

Such a system starts to get very complex, expensive, and impractical for home use. So we have also been experimenting with simpler systems such as using the speaker's own capacitance changes (the capacitance changes as the diaphragm to stator spacing changes) to measure the motion indirectly. This is simpler than using a laser, but preventing the drive signal from influencing the measurement is tricky. More work is in order here.

While motional feedback can solve the frequency response and high Q problems of an electrostatic woofer, it still doesn't address the main issue, which is poor output. The fact remains that a dipole radiator suffers from extreme phase cancellation and radiation resistance losses in the bass that prevent it from producing deep, loud, dynamic bass.

There is no apparent solution to this problem of feeble bass output. Until there is, the dream of a high performance, full-range, crossoverless ESL will remain an elusive goal.

Fortunately, hybrid systems have now reached the point where they perform as well as full range ESLs with regard to detail and clarity. Integration of the two drivers is flawless. And unlike a full range ESL, a hybrid can produce very high output so that the full dynamic range of live music is fully realized. So while I continue to experiment with electrostatic woofers, I don't really see any need for them.

Great listening,
-Roger
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#5
Roger: Thank you for a VERY informative and detailed response. I learned an awful lot from your post!!!

I think one issue is that many feel there is always a tradeoff between the "speed" of a dipolar bass system such as the Maggies and a dynamic speaker system. To my ears, the Maggies have an incredible ability to define say notes on a standup bass as opposed to the dynamics and amount of air moved by a cone driver. That said, what do you think the lower limit is and what factors might influence how low you could extend the frequency response on an electrostatic panel before crossing over to a dynamic driver or say your transmission line approach.
More from Roger
-------------------------------
Hi Myles,

Thank you for your kind compliments on the educational value of my responses. I sincerely try to help audiophiles find the true reasons for what they are hearing. There is far too much hype and voodoo science in this industry, which makes it very confusing for many music lovers. Hopefully I can help with this issue.

Turning to your questions, you bring up several issues that need to be clarified. First, there is no question in my mind that you hear a very different bass quality from Maggies than you do from conventional closed box (infinite baffle) or vented (bass reflex) woofer systems. However, there are several possible causes that could explain the differences you hear. So you need to be very careful drawing conclusions about cause/effect relationships.

There are many differences between Maggies and box speakers besides the fact that one is a dipole and the other is a monopole. While the dipole/monopole issue COULD explain the difference you hear, there are several other differences that could also explain the phenomenon. So you should not assume that the Maggie woofer sounds better simply because it is a dipole.

Specifically, there is a big difference in Q between conventional bass systems and the Maggies. Also there is certainly a big difference in frequency response.

In my tests, I have found that the Q of the typical, conventional woofer system (closed box/vented systems) is much higher than that of a Maggie. This causes a lot of overshoot and ringing in conventional woofer systems that adds a lot of energy to the reproduced sound that is not present in the bass instrument itself. Of course, this dramatically colors the sound.

In other words, conventional woofer systems corrupt the sound significantly. Their overshoot and ringing adds energy that masks the subtle harmonics that give bass instruments their detail and clarity. By comparison, a Maggie's woofer has far lower Q and is therefore "cleaner", which makes it possible for you to hear the detail in the bass instruments much better.

Yet another factor is frequency response. Being a dipole, the bass from a Maggie lacks real depth, just like electrostatic bass. However, unlike an ESL, the Maggie is magnetically driven, so it has a lot of excursion and can produce fairly high output from the mid-bass on up (assuming many hundreds of watts of amplifier power are available). Still, the lowest frequencies will be weak compared to a conventional woofer system.

When you reduce the deep bass, the mid-bass and higher frequencies are more apparent since they are not masked as much by the deeper frequencies. So the mid and upper bass from the Maggies may sound more clear simply because these higher frequencies are not masked by deep bass.

You can easily prove this to yourself by using a DSP (Digital Signal Processor) or graphic equalizer to roll off the bass below 50 Hz on a conventional woofer system. The overall bass sound quality will become "cleaner," although the overall "punch" and depth will be somewhat reduced.

I worked on this problem of high Q behavior in conventional woofer systems for 17 years as it was the main problem in getting a magnetic woofer to integrate well with an ESL. The problem mostly has to do with enclosures. Here is the problem:

Consider an infinite baffle (closed box) enclosure. Its sole purpose is to stop phase cancellation, which it does perfectly by preventing the front wave of the woofer from being cancelled by the rear wave, which is 180 degrees out of phase.

But this closed box introduces several problems. First, think of what happens when the woofer is driven into the enclosure -- it compresses the air inside the enclosure.

Now consider what happens when the woofer needs to come back to its neutral point and stop. The compressed air inside the box acts like a spring that pushes the woofer towards the neutral point. It doesn't stop pushing until the woofer actually reaches the neutral point. This means that the action of the compressed air spring strongly pushes the woofer BEYOND the neutral point.

Of course, the relatively high mass of the driver means that it has a lot of inertia. Since there is very little to counter the kinetic energy of the woofer and bring it to a quick stop, the woofer coasts far past the neutral point until its suspension (and electrical damping by the amplifier) brings it to a stop. The combination of the compressed air spring and inertia results in severe overshoot.

As the woofer reaches the limits of its suspension's excursion, it reverses direction and sails past the neutral point once again, but in the opposite direction. There is some friction from the air and damping from the amplifier that uses up some of this energy, so the distance that the woofer's cone overshoots the neutral point the second time is less than the first time. As a result, the woofer gradually comes to a stop after several cycles of overshoot (known as "ringing").

This high Q behavior introduces a great deal of extra energy to the original sound that should not be there. It can certainly be considered distortion, and subjectively, it adds a "heaviness" and "muddy" quality that degrades the quality of the sound.

But that's not the only problem. Consider the energy that the woofer generates in your room. I think you would agree that a conventional woofer can produce really powerful bass in a room that is hundreds of times larger than the woofer's enclosure.

A woofer generates the same amount of energy on both sides of its cone. The high energy that is enough to rattle the windows and floor of your listening room is magnified hundreds or thousands of times as you cram all the energy into a relatively tiny enclosure. So you can start to appreciate that there is a lot of extremely powerful sound energy inside that tiny woofer cabinet!

What happens to all this energy? Well, since there is really nothing to absorb it, the energy must escape from the enclosure somehow. Some of it is released by bending the flat sides that are present on most woofer enclosures. This is the infamous "box talk" with which you are familiar. It is a major contributor to the relatively high distortion and coloration in conventional woofer systems.

But most of the energy is going to come through the weakest point in the enclosure. That point is the woofer's cone, which is very thin and weak compared to the thick wall of the enclosure. The result is that there is a very large amount of delayed energy coming from the woofer's cone after it has bounced around inside the enclosure for a short time. This "box energy" is loaded with resonances, distortion, and phase error.

A vented enclosure is even worse. It is designed to resonate as certain frequencies. This causes even worse overshoot and ringing. It frequency response is far more non-linear than a closed box. And the "garbage" is different at different frequencies.

All this "garbage" is added to what should otherwise be a very pure reproduction of the original bass instrument. And when you add the garbage, plus all the overshoot and ringing, it is a wonder that conventional woofers are even listenable. Actually, the only reason we tolerate conventional woofers is because our ears are quite insensitive to flaws in bass frequencies. Nevertheless, any thoughtful audiophile recognizes that most bass speaker systems are seriously compromised. They don't sound very realistic at all.

In summary, conventional closed box and vented enclosures are truly awful designs. The only reason they are in common use is because they are cheap and easy to build and relatively small woofer systems can be made with them. But I do not consider them satisfactory for high performance speaker systems as they simply fail to produce "high fidelity" sound.

A transmission line eliminates all these problems. By way of review, a TL (Transmission Line) is essentially a long (typically around 8 feet), tapered pipe that is filled with fibrous material. The woofer is mounted in one end and the other end is open. Now let's look at how this affects the problems of overshoot, ringing, and "garbage" (enclosure energy/distortion/resonance/box-talk).

First, the TL is long enough that phase cancellation cannot occur at audio frequencies because the rear radiation has to travel so far to get around to the front that the phase is shifted well beyond 180 degrees. In fact, we generally design TLs so that the phase is shifted 360 degrees at some deep bass frequency so that the rear radiation from the woofer actually helps boost the front radiation. This really helps support the deep bass frequencies, which are always falling in all woofer designs due to radiation resistance losses. This is one of the reasons that TL's are known for having really powerful deep bass performance and flat frequency response to lower frequencies than conventional enclosure types.


As an aside, since no woofer can produce flat deep bass response due to radiation resistance losses, all woofers need to be equalized below 50 Hz or so. One of the really nice things about digital crossovers is that they offer this feature so you can get truly flat frequency response right down to 20 Hz (assuming adequate woofer excursion capability and many hundreds of watts of amplifier power).

Secondly, consider the compressed air spring problem. Because the TL is open at the far end, the woofer cannot compress the air inside the TL. The air simply escapes through the port rather than being compressed.

Even more importantly is the effect the fibrous material (damping material) has on the air as it passes through the TL. The air runs into the millions of fibers in the TL, which cause a lot of friction.

Therefore, the air cannot pass freely and quickly through a TL. It is slowed down by the friction from the damping material and a lot of energy is needed to push it through the line.

Now what happens when the woofer needs to come back to the neutral point? There is no compressed air spring to push it. Furthermore, the woofer's inertia is used up as its kinetic energy is converted to heat by dragging the air back through the friction of the damping material inside the TL. It is like having a little "shock absorber" attached to the woofer.

As a result, the woofer simply stops at the neutral point. There is no overshoot. Without any overshoot, there can be no ringing.

Now let's look at the rear energy coming off the woofer's cone. For the mid-bass frequencies on up, the air motion is relatively small and virtually all of this motion is reduced to heat by friction with the damping material. There is no possibility for free motion that would cause resonances inside the enclosure that could escape and color the sound. Also, the TL is tapered so will generate an infinite number of infinitely small resonances rather than just a couple of huge ones as is the case in typical, rectangular enclosures. So there is no box talk and no "garbage" coming through the woofer cone.

The deep bass frequencies contain a great deal of power and very long wave lengths. For them, the TL is not long enough to completely convert them to heat. Therefore, a lot of deep bass energy is released from the port at the end of the TL. But remember, it has had its phase shifted so that it comes out in-phase with the front of the woofer, so it does useful work supporting the deep bass frequencies.

Note carefully that this port energy is not produced by any resonance phenomenon. It is clean. It is the original musical signal that has been phase shifted. So it does not produce distortion that degrades the sound.

When a low mass, low Q woofer is used in a TL, and is driven by a high damping-factor amplifier, the problems of overshoot, ringing, and distortion from a conventional cone woofer are eliminated. As a result, a good TL sounds just as clean, pure, and detailed as a dipole. But it has the advantage over a dipole of having really deep bass and being able to play at earth-shaking output levels.

There is one more detail to bring up in this quest for a low Q woofer, and that is the effect that passive crossovers have on amplifier damping. Understand that the Theil-Small parameters consist of Qms, Qes, and Qts. Qms refers to the mass of the driver. It is considered to be the mechanical Q of the speaker and is determined by the mass of the moving parts of the speaker and the behavior of the suspension system.

Generally, for a given suspension, the higher the mass, the more the driver will tend to overshoot and ring. For a given mass, the softer the suspension, the less the driver will overshoot and ring. In other words, a low Qms woofer will have low mass and a very soft suspension. These factors will also reduce its free air resonance point.

Qes refers to the electrical Q. Understand that when you move the voice coil through its permanent magnetic field, an electrical current will be generated in the voice coil. If you place a load across the voice coil, this current will do work. Work requires energy. The energy comes from the motion of the woofer cone.

If you short out the voice coil, the most energy will be required from the woofer cone. If you leave the voice coil as an open circuit, no energy will be required from the woofer cone.

You can easily feel this. Just disconnect a speaker from its amplifier, then push the woofer cone back and forth with your fingers as you alternately open and short across the speaker binding posts. You will clearly feel a resistance to the woofer's motion when you short the voice coil.

This is electrical damping. It becomes greater if the permanent magnet is more powerful.

Qes is highly desirable. It is at its best when the woofer is connected directly to a very low impedance amplifier. The "damping factor" is the ratio between the impedance of the voice coil and the amplifier's output impedance.

For example, if you have an 8 ohm voice coil in your woofer and you connect it to the 8 ohm taps of a tube amplifier, the damping factor will be 1 (one). If that same 8 ohm voice coil is attached to a powerful solid state amplifier whose output impedance is 0.02 ohms (a typical value), then the damping factor would be 400. The lower Q of the solid state amp system causes the bass to sound "tighter" than from most tube amps.

I have simplified the damping factor issue somewhat because I did not mention global feedback. Suffice it to say that global feedback incorporates the speaker load into the feedback loop and significantly increases the damping factor of an amplifier. So a moderate amount of global feedback is very helpful at controlling the woofer's motion.

It is very important to note that the addition of any series resistance between the woofer and the amplifier will tremendously degrade the damping factor. This is where passive crossovers are so evil. They always have one or more inductors in series with the woofer.

An inductor consists of a long, thin piece of wire wound into a coil. It has significant resistance. So it will always degrade the damping factor of the system.

This is one of the major reasons that passive crossovers should not even be considered for high performance speakers. Electronic crossovers are far superior because their amplifiers are directly connected to their drivers without any intervening inductors or resistors that ruin the damping factor.

As an aside, I always get a bit of a chuckle when I hear how adamant some audiophiles are about using huge, low-resistance speaker cables when used with passive crossovers. They think they are helping the situation by using a very low resistance cable, but they simply don't realize that they are connecting it to tens of feet of tiny magnet wire that forms an inductor. This eliminates any beneficial effect the large speaker cable might have offered. They might as well have used a tiny speaker cable since the wire in the inductor is far longer and has more effect than their huge cable.

The Qts is the combination of the mechanical and electrical Q of the driver. As you can see, it will be inaccurate if an enclosure is designed using it without consideration of the effects of a passive crossover.

Finally let me turn to your question about the lower limit of dipole operation and what crossover point to use. There is no simple or direct answer to this. There are many factors that must be considered and none are cast in stone. But I will list a few that will drive your decision.

First, as described in a previous communication, the ESL must not be operated at or near its fundamental resonance or it will exhibit high Q behavior. Exactly how close to the resonance you can go depends on the type of crossover filter used and how steep its slopes are. In my speakers, I find that I can get to within about one octave by using 48 dB/octave slopes in the crossover.

If you use gentler slopes, you must stay further away from the resonance to avoid exciting it. For example, if you use 24 dB/octave slopes, you will need to cross over about 2 octaves above the resonance, etc.

Diaphragm tension and suspension spacing (the ratio of the diaphragm to stator spacer distances compared to the diaphragm to stator spacing) will have a very powerful effect on the fundamental resonance. Lower diaphragm tensions or larger suspension spacing will reduce the frequency of the fundamental resonance (as will altitude, temperature, and humidity).

But lowering the resonance point will also reduce the stability of the diaphragm, reduce the polarizing voltage possible, and reduce the sensitivity of the speaker. So a series of compromises is necessary to reach an optimum among these factors.

The minimum dimension (usually the width) of the ESL will determine the phase cancellation frequencies and therefore how much midrange equalization is required to produce flat frequency response at the crossover point. Each dB of equalization has the effect of reducing the overall output of the speaker by the same amount. So as you reduce the width of the ESL, you quickly find that you run out of available excursion or amplifier power or both because of the added equalization required.

Not all ESLs use midrange equalization. But they must somehow compensate for phase cancellation or they will sound very bright, thin, and anemic. Those that don't use EQ usually increase the amount of radiating area with decreasing frequency. This is a big compromise that is an entirely different subject that I don't have time to discuss in this article. But the problem is the same regardless of whether EQ or variable radiating area is used.

So you will need to trade off a low crossover point for adequate output. This is a critical design compromise where the width of the ESL, the output desired, and the crossover point all have to be juggled to reach a reasonable compromise.

Diaphragm excursion is a critical factor. At some high output level, depending on frequency, the diaphragm will slap the stator, and this produces an absolute limit on output.

The diaphragm to stator spacing determines the maximum excursion possible, which has a profound effect on the maximum potential output of the speaker and its sensitivity. A larger diaphragm to stator spacing can potentially produce higher outputs at lower frequencies. But it reduces sensitivity and requires much higher drive voltages, which are very difficult to produce.

Obviously, if you can get more excursion with high output, you can use a lower crossover point for a given output level. But there are severe limitations on how much excursion you can get from a practical perspective. So you will have to trade off total output against the crossover point.

One nice thing about digital crossovers is that you can store many different crossover arrangements. This makes it possible to use a relatively high crossover point when you want or need very high output levels, and then quickly switch to a much lower crossover point (or even full-range, crossoverless operation) for quiet listening.

As you can see, there are a whole host of factors to consider when deciding what crossover point to use. Therefore, I cannot give you any hard and fast rules without knowing all the various parts of the speaker design that affect the crossover point.

I hope this information has been helpful.

Great listening,
-Roger
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#6
Well while we have your attention... I suppose I prefer electrostatic bass because of its transient response. That is to say I tend to prefer less ringing and slightly decreased output to increased output and more ringing. Many have tried a "servo" to get the woofer to start and stop properly. What role does a servo play?

(Reply from Roger):


A servo is another name for a motional feedback system. The general idea of such systems is to compare the motion of the speaker's cone to the musical input signal, identify the difference between the two, and use the resulting error signal to drive the amplifier to force the cone to move exactly like the musical signal.

These can be made to work in several ways. For magnetic woofers, the most common technique is to add a couple of turns of wire around the voice coil former that is electrically distinct from the voice coil itself.

As the voice coil moves back and forth in its permanent magnet field, this detector coil generates a small current. This current will accurately describe the motion of the voice coil. It therefore can be used to compare to the input signal to see how closely the woofer follows the input signal.

The current from the detector coil is fed into one of the inputs of a differential opamp and the musical signal is fed into the other. The opamp then generates an output signal that reveals any difference between the two.
This is called an error signal.

The error signal then has its phase inverted and is fed into the woofer amplifier's input along with the musical signal. The amplifier will then apply an opposing drive force to the woofer to make it to move only as allowed by the musical signal.

Motional feedback systems can offer a significant improvement in bass response. But they are not quite perfect and have several problems that limit their use.

The main problem is that woofer cones are not rigid. Therefore, the woofer cone does not follow the motion of the voice coil precisely. So even if the voice coil is made to move exactly as the input signal, the cone will not follow it perfectly (although it will be much better than without motional feedback).

The biggest problem in this regard is the phenomenon of cone "break up."
Large woofer cones only operate as pistons at low frequencies. At some higher frequency (usually a little above 100 Hz), the mass of the cone will produce so much inertia that the entire cone will not be able to follow the voice coil. At this point, the cone material will flex as the outer edges of the cone tend to remain stationary while the inner section of the cone move rapidly with the voice coil. The cone is then considered to be operating in "breakup mode."

Of course, the outer edge of the cone doesn't actually come to a complete stop. It tries to follow the inner section of the cone and by failing to do so, it produces distortion. Also, the frequency response is altered as the radiating area of the woofer changes. The flexing cone also produces waves in its surface that cause harmonic distortion.

Woofer driver manufacturers expend a tremendous amount of engineering effort trying to get their cones to break up smoothly, with minimum distortion, and with reasonably smooth frequency response. This is why you will note that woofer cones are not straight-sided and made of stiff material as you would expect if the cone acted like a rigid piston.

Instead, most woofer cones have shallow, curved surfaces that are deliberately made to flex in a controlled and desirable way. They usually are made of soft material like baxtrene, polypropylene, or paper since these materials tend to have excellent damping characteristics and therefore minimize distortion and ringing. They produce "soft" breakup characteristics.

A lot of work has been done in recent years to produce cones of stiffer material like aluminum, titanium, and carbon fiber. Sometimes these are made into very stiff shapes like straight-sided cones with steep sides.

Such woofer cones operate to higher frequencies before experiencing breakup.
But they have rather abrupt break-up characteristics and therefore can sound quite harsh. So they are best used only in the deeper bass regions where they operate as rigid pistons. These ultra stiff cones follow their voice coil motion more accurately and should work better with motional feedback systems than conventional, soft, woofer cones.

But the fact remains that motional feedback only works well for woofers used below their break up frequency. So it can't effectively control the flaws in a conventional woofer in the upper bass and midrange.

Of course, the biggest problem with servo systems is that they are complicated. They must have box of a dedicated electronics and ideally, they should have their own, dedicated woofer amplifier.

Most audiophiles are very resistant to using complex systems. Only a few will even use electronic crossover systems because of the added complexity and the need for multiple amplifiers. So they continue to suffer the poor performance of passive, high-level crossovers. Getting them to use motional feedback systems is a real challenge.

However, in certain applications, the benefits of motional feedback systems can be both successful and audiophiles will accept them. A good example is in large subwoofers, which have their own, internal amplifiers and crossovers, so adding motional feedback is easily accomplished. But for the typical, full-range speaker, including adding motional feedback is not practical from a marketing perspective, so is not used.

I find it very odd that while most audiophiles will spend tens of thousands of dollars in their quest for musical nirvana, but they fail to take advantage of well-established techniques for improving sound. The best example of this is the presence of passive crossovers in high-end speakers -- even in speaker systems that claim to be "reference" or State-of-the-Art. No speaker can be considered high performance if it uses passive crossovers. Motional feedback systems also offer improved performance, but generally are not used.

Anther excellent example are room correction systems. Even if you have a perfect speaker system, it will be degraded by room acoustics. Room correction systems work very well to deal with this problem, but most audiophiles fail to take advantage of them.

-Roger
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#7
Thank you for that informative explanation. I guess the next question was set up by you Roger. What is the difference between active and passive crossovers? Did I hear you say you prefer a brick wall crossover between the woofer and stat panel? At your leisure of course.
OK, put your feet up, boys, here comes Roger's response:


There are many issues involving crossovers, but let me begin with your first question, "What is the difference between active and passive crossovers?" The engineering definition is that "active" electronics (crossovers or any other type of electronics) include amplifiers in their circuits while passive ones do not.

But when audiophiles use the term "active" with respect to crossovers, they usually mean "electronic" crossovers, which are low level and active compared to "passive" crossovers, which are high level and passive. Since this is all rather confusing, allow me to dig into crossover theory and design a bit, with their pros and cons, and it will all soon make sense.

By way of review, there is no speaker type that can reproduce all the frequencies in the audio bandwidth well. So crossovers must be used to split the audio frequencies into bands that can be reproduced well by the various types of drivers (usually woofers, midranges, and tweeters).

This is rather easy to do using passive components like capacitors and inductors. A capacitor inherently will roll off the bass into a tweeter at 6 dB/octave. An inductor inherently will roll off the highs into a woofer at 6 dB/octave. The circuit is very simple as you need only put a capacitor in series with the tweeter and an inductor in series with the woofer.

You may have heard about the number of "poles" in a crossover. A pole is the inherent 6 dB/octave roll off in capacitors and inductors. A single pole crossover would operate at 6 dB/octave. A two pole crossover would be 12 dB/octave, three poles would be 18 dB/octave, etc.

A simple, passive crossover is made of inductors and capacitors that are placed in the speaker cabinet along with the drivers. A single amplifier can be connected to the crossover and the crossover will then energize the various drivers in the speaker cabinet. This is simple, cheap, and easy, so that is what most manufacturers used at the start of the high fidelity industry -- and most continue to use these passive, high-level crossovers to this day.

This type of crossover is passive because its components have no ability to amplify or control the musical signal. It is "high level" because it handles the relatively high power (hundreds of watts), high voltages (up to several hundred volts), and currents (up to a few tens of amps) found at the output of an amplifier.

But passive crossovers have many problems and limitations. The major ones are poor precision, distortion, phase shift, the inability to produce steep crossover slopes, and no buffering. Let me address these in some detail.

It is important to understand that different types of drivers should not reproduce the same frequencies. No two drivers are identical or behave identically, so if multiple drivers are used to reproduce the same frequencies, there will be various distortions and errors introduced as the two drivers interact with each other at the shared frequencies.

This is particularly true when woofers and tweeters are combined. A woofer is extremely limited in its ability to produce high frequencies and will not sound at all the same as a tweeter at those same high frequencies. While operating a woofer at high frequencies only messes up the sound, operating a tweeter at low frequencies will destroy it. So it is very important to confine a woofer to the bass and a tweeter to the highs.

Audiophiles tend to think that the various drivers pretty much stop operating at the crossover point, which technically is defined as that frequency where the signal is 3 dB below the reference or baseline level. But this is not at all true. The output from a driver only starts to roll off at the crossover point. It continues to contribute a substantial amount of energy to the sound far beyond the crossover point.

The output of a driver needs to be at least 48 dB below the reference level before its output is low enough that it can be ignored. So crossover slopes are very important.

A capacitor or inductor inherently rolls off the sound at 6 dB/octave. Therefore, a driver that is driven by a single-pole, passive crossover will continue to operate and produce useful sound output for fully eight octaves above (or below) the crossover point before its output will have diminished by 48 dB.

To put this in proper perspective, let's examine the woofer's contribution to the sound if it has a crossover point of 1 KHz using a crossover with a 6 dB/octave slope. Since an octave is double the fundamental frequency, one octave above 1 KHz will be 2 KHz. The second octave will be at 4 KHz, the third will be at 8 KHz, the fourth octave will be at 16 KHz, the fifth octave will be at 32 KHz, etc.

So you can see in this case, that the woofer's output level will only be reduced by 18 dB at 8 KHz. This means that it will still be generating a lot of output at frequencies that should be produced only by the tweeter. Therefore, a single-pole, 6 dB/octave crossover is virtually useless at keeping the woofer out of the treble region. The woofer's sound will remain a major contributor to the highs and will degrade them significantly.

The reverse is true when you consider the tweeter. If the crossover is at 1 KHz, then a 6 dB/octave crossover will reduce the power to the tweeter by only 18 dB at 125 Hz. Since the energy in music is rising rapidly as you get into the bass frequencies, this 6 dB/octave crossover slope does not adequately protect the tweeter's delicate, lightweight, voice coil from being burned up by excessive power. So 6 dB/octave crossover slopes are simply unsatisfactory.

Loudspeaker manufacturers quickly learned that they needed to use steeper crossover slopes. To make a high level, passive crossover produce a 12 dB/octave slope, you need only add an inductor going to ground to the tweeter circuit and a capacitor going to ground in the woofer circuit.

Now you can see that using our earlier example of a speaker with a 1 KHz crossover point that the woofer's output will be down by 36 dB at 8 KHz and the tweeter's output will be down by 36 dB at 125 Hz. This is a vast improvement over a 6 dB/octave crossover, although there is still far too much overlap of shared frequencies from both drivers for ideal performance and the woofer still operates much too high for great sound.

It would be better to use much steeper crossover slopes. But making steeper slopes with passive inductors and capacitors becomes very difficult because the tolerance of the components is so poor that you can't get their poles to match precisely enough to work correctly. After all, most capacitors have a tolerance of +/- 20%, and it is common for large capacitors like those used in speaker crossovers to have tolerances of +/- 50%. Inductors have only slightly better tolerances.

The precision problem limits most passive crossovers to 12 dB/octave. It is possible to make passive crossovers with steeper slopes, but to do so requires hand selection and individual measurement of each component to select for adequately tight tolerances. This is a time-consuming process that costs a lot of money, so is rarely done.

The simple, common, 12 dB/octave crossover just described contains just 4 parts (2 inductors and 2 capacitors). But it has a major problem. It produces a filter type known as "Butterworth."

This means that its behavior at the crossover point is quite sharp and when both the high pass section (the tweeter) and the low pass section (the woofer) are combined, the power output at the crossover point is doubled. This produces an irregularity in the speaker's frequency response at the crossover point in the form of a bump of 3 dB. Obviously, this has adverse effects on the sound quality of the speaker and is unacceptable.

It is difficult to solve this frequency response problem by using equalization (which would be the best way to do so) because passive crossovers have no amplification with which you can produce equalization. So most manufacturers electrically invert the phase of one of the drivers.

This puts the drivers out of phase by 180 degrees and eliminates the frequency response problem. Of course, putting the drivers out of phase has adverse effects on the sound. But the adverse effects of phase are minimal in comparison to the adverse effects of a major frequency response error. So a compromise is reached where accurate phase is sacrificed for accurate frequency response.

This problem with Butterworth filters is serious and was eventually addressed by Linquitz and Riley when they jointly developed the Linquitz/Riley filter. Crossovers made to the L/R specification have flat frequency response through the crossover point and therefore, the drivers could be left in phase.

But the L/R filter was about twice as complex as a Butterworth filter, so this increased the cost of passive crossovers significantly. Still, manufacturers of quality loudspeakers adopted the L/R filter type in the interest of better performance while manufacturers of cheaper speakers continued to use Butterworth filters and suffered phase anomalies.

There are other filter types that have been developed for crossovers by Bessel and Chebyshev. I won't get into the details of these as they are rarely used and difficult to implement in passive crossovers. But for those who want special characteristics, these special filters can be incorporated into electronic crossovers.

Note that phase anomalies are most obvious when they occur in the critical midrange region. So manufacturers of quality speakers worked hard to push the crossover points out of the midrange.
Since conventional, magnetic, 2-way speaker systems usually must be crossed over at around 2 KHz, they are very bad about having crossover anomalies in the midrange. Also, it is hard to get a wide enough frequency band out of magnetic drivers to make truly high performance, 2-way systems.

So most quality magnetic speaker systems have 3 drivers (woofer, midrange, and tweeter). This allows them to move the woofer crossover frequency down to perhaps 500 Hz and the tweeter's crossover up to around 5 KHz. This really helps eliminate the problems of crossovers in the midrange, although it doesn't completely solve the problems. Having three drivers with relatively shallow crossover slopes means that at most of the midrange frequencies, you will have all three drivers contributing significant sound, which is one of the reasons that the midrange quality of magnetic speaker systems is inferior to the purity of an ESL.

A 3-way system considerably complicates the crossover. You now need 50% more parts. If those parts are used in a L/R filter, you may need twice that amount. So quality speakers end up using rather complex and expensive passive crossovers.

Passive crossovers degrade the sound by inserting parts (capacitors, resistors, and inductors) between the amplifier and the drivers. Particularly troubling are inductors. These consist of a long (many feet) of thin magnet wire wound in the shape of a coil. As a result, an inductor has significant resistance. When you put resistance between an amplifier and its woofer, you ruin the electrical damping that the amplifier could apply to the woofer to help control the woofer and stop overshoot and ringing.

As if the resistance in inductors weren't bad enough, there are often resistors in the signal path too. These are required to give you some ability to adjust the levels of the various drivers to get them to match reasonably well and get acceptable frequency response. But a resistor will ruin the amplifier's damping even worse than an inductor.

There are other problems in passive crossover involving inductors. The main one is hysteresis. Hysteresis is a problem whereby the output signal does not exactly match the input signal because there is non-linearity and phase error when producing magnetic induction in the core of the inductor.

Most inductors use iron cores to make them more smaller and more efficient. As current flows through the coil of wire in an inductor, it forms a magnetic field. This magnetic field is more powerful if it induces magnetism in an iron core.

But iron-core inductors produce a lot of distortion because they have very big hysteresis losses. To minimize this problem, most quality loudspeaker manufacturers eliminate the iron core and use air core inductors.

Due to their inefficiency, air core inductors must be much larger than iron core inductors. Longer wire must be used. More resistance is involved, which worsens amplifier damping, increases costs, and makes the crossover larger. Hysteresis losses are not totally eliminated in an air core inductor, but they are greatly reduced. But on balance, it is fair to say that inductors simply don't behave very well in audio circuits and are best avoided if possible.

Capacitors are also problematical. Large value capacitors are required, so electrolytic types are preferred because of their relatively small size compared to non-polarized caps (like polypropylene, polyester, mica, etc.).

But electrolytic capacitors have their conductors wound inside of them, so they also have a significant amount of inductance, which can alter the desired frequency response of the crossover. Therefore, non-polarized capacitors often are used in the very best speakers, even though these are expensive and take up a very large amount of room.

Passive crossovers cannot be buffered. This means that their behavior and frequency response can be influenced by the external application of inductance, capacitance, and resistance.

Where would these external factors come from? Mainly from speaker cables. Manufacturers of speaker cables know this so deliberately make their cables have various values of inductance, capacitance, and resistance to change the frequency response of speakers.

This is a real crap-shoot because they have no idea of how a particular cable will affect a particular speaker. Of course, the longer the cable, the more inductance, capacitance, and resistance it will have, so the more it will alter the frequency response of passive crossovers.

A speaker is significantly affected by room acoustics. So some cables may compensate for some of the room acoustics in pleasing ways and others may make the room interactions worse. It all depends on the speaker, the cable, the room, and the listener's taste in frequency response.

All these variables are is why there is so little agreement and so much controversy about which speaker cables are "best." Simply put, there is no "best" cable because all the variables involved prevent you from predicting the sound from a system with a particular cable.

I could go on to cover many other problems with passive crossovers. But the main point is simple -- passive crossovers have very serious flaws for which there are no good cures.

Speaker manufacturers know very well that the solution to all these problems are active, low-level, "electronic" crossovers. Such crossovers are actually small preamplifiers that have crossover filters built into them. Think of the "tone controls" that used to be available on preamps and receivers. These could be used to roll off the highs or lows just like crossovers do.

Electronic crossovers operate at low levels ("line level", which is about 1 volt and essentially no power). They operate on the signal from the preamp rather than being fed by an amplifier like passive, high-level crossovers.

The line-level preamp signal is split into the various frequency bands by the electronic crossover and each is then fed to an amplifier that energizes its respective driver directly. There are no inductors, capacitors, or resistors between the amplifier and its driver that would cause distortion, phase shift, or ruin the damping.

Electronic crossovers don't need to use inductors with all their problems as all the frequency filtering can be done with just tiny capacitors and resistors. These capacitors and resistors can be made to very high tolerance with resistors being accurate to better than 1% and capacitors to around 2%. So multi-pole filters can be used to get steeper crossover slopes.

Like any good preamp, electronic crossovers have input and output buffers. So they are immune to the effects of external inductance, capacitance, and resistance.

Complex filter types like Linquitz/Riley are easily and inexpensively incorporated into electronic crossovers. It is also quite easy to make the crossover infinitely adjustable in real time so that the listener can simply tweak the crossover points, slopes, gain, etc. as he wishes to get ideal sound and the flattest frequency response. This is impossible with passive crossovers.

The distortion in an electronic crossover is nearly immeasurable and is vastly lower than in any a passive crossover. The components in electronic crossovers are tiny and inexpensive, so electronic crossovers can be made at lower cost than quality passive crossovers. Since there is no large passive crossover that needs to be housed inside a speaker cabinet, electronic crossovers allow the speaker cabinet to be smaller.

Of course, nothing is perfect and electronic crossovers have their flaws. The main one is complexity and cost from the standpoint of the audiophile. This is because an electronic crossover system must be bi-amplified (for a 2-way system), or tri-amplified (for a 3-way speaker).

So the audiophile needs to buy two or more amplifiers to use with an electronic crossover. This is a major barrier for most audiophiles. But the improved performance is well worth it if you are looking for the best sound quality.

Analog electronic crossovers are still limited in their ability to produce steep crossover slopes. Although the tiny parts involved have much higher precision than the large parts in passive crossovers, they still are not precise enough to produce more than about four filter poles (24 dB/octave slopes). And while 24 dB/octave slopes are vastly better than the 12 dB/octave slopes typically found in passive crossovers, it would be nice to use steeper slopes if there were a good way to do so.

Recently, we have seen the development of digital signal processing. This makes it possible to eliminate the problems and limitation of capacitors, resistors, and inductors completely. The frequency response of the filters can be done entirely in the digital domain by computation. So digital electronic crossovers are not dependent on and limited by the behavior of special electronic parts.

Digital crossovers come with selectible crossover slopes and filter types. It is easy to use 48 dB/octave, Linquitz/Riley filters using digital crossovers, while this is virtually impossible using analog ones. These steep slopes make it possible to completely eliminate the contribution of each driver within 1 octave of the crossover point, thereby reducing the shared bandwidth to a minimum and greatly reducing the stress on the drivers.

Additionally, it is fair to say that all speakers need at least some equalization to produce the best sound. At a minimum this involves increasing the bass output below 50 Hz to compensate for radiation resistance losses that cause all woofers to roll off below 50 Hz.

Digital electronic crossovers include equalization facilities using their built-in, digital signal processor. The amount and type of equalization varies from shelving equalizers (to compensate for speaker limitations) to full-on room correction systems with parametric equalizers and built-in, real time analyzers.

Yet another advantage of digital crossovers is speaker time-alignment. You probably have noticed that some of the best speakers have their various drivers at different planes in their cabinet. This is so that they are different distances from you.

Placing the drivers in different planes is desirable because some drivers are "quicker" and their sound gets to you before others in the same cabinet. Usually the sound from the tweeter arrives before the woofer. If the sound from all drivers does not arrive simultaneously, the phase behavior of the speaker will be adversely affected -- particularly through that region where two or more drivers are reproducing the same frequencies.

Therefore, many manufacturers try to "time align" their speakers by mounting their drivers in different planes. Unfortunately, this is difficult to do with good cosmetic results and often it is impossible to mount them far enough apart to completely correct the problem.

A digital crossover solves this problem by introducing digital time delay into the early-arrival driver. You can connect a microphone to the crossover and it will automatically produce test tones that will allow it to measure the acoustical distance to each driver.

It will then take this information and automatically delay the early driver by the amount required so that all the sounds from all the drivers arrive at your ears simultaneously. The digital crossover will even compensate for temperature differences in the room that alter the speed of sound! The result will be perfect time-alignment even in speakers (most of them) where the drivers are not in the optimum planes.

At this point, you should be gaining an appreciation of why I insist that no speaker can claim to be of truly high performance if it uses passive, high-level crossovers. The performance available from electronic crossovers (particularly digital crossovers) and multi-amping is simply far better. All speakers can be made to perform better using electronic crossovers than when using passive ones.

Now I would like to address a couple of common misconceptions regarding crossovers. The first is phase. Many audiophiles try to avoid steep crossover slopes in the belief that steep slopes cause more phase shift that degrades the sound.

While it is true that the steeper the slope, the more the phase is shifted, this is not what causes the audible phase problems in crossovers. The true cause of the phase errors that can be heard is due to the unbuffered behavior of passive crossovers. Let me explain.

I previously mentioned that because passive crossovers cannot be buffered, their frequency response can be altered by the external application of inductance, capacitance, and resistance. I stated that speaker cables were a major cause of this, which is true.

But I did not mention that passive crossovers will also have their frequency response altered by their drivers -- none of which have perfectly uniform impedance. Since the performance of a passive crossover can only be produced into a specific impedance, the expected frequency response of a passive crossover will not be perfect because the impedance of the drivers varies.

Now anytime that there is a change in frequency response, there will be a corresponding change in phase response. Look at the impedance of any speaker system with a passive crossover and you will see that it looks like a cross section of the Andes mountains.

The phase is therefore similarly altered. It is these ragged phase errors that exist across the music spectrum in passive crossover speakers that cause the adverse phase effects you hear from passive crossovers.

Note carefully that an electronic crossover is not affected by the impedance of the speaker. The speaker is driven by an amplifier, not the crossover. The frequency response of any well-designed amplifier will not be affected by the impedance of its driver. So there will be no ragged phase errors to be heard in an electronic crossover.

But what about the smooth even phase shift caused by the crossover slope? Doesn't that affect the sound too?

Actually, careful testing shows that human hearing is not very sensitive to a smooth, linear, sloping phase shift. So the effects of the phase shift caused by the crossover slope are virtually inaudible. In any case, the smooth phase shift of the crossover slope is vastly less apparent than the phase shift caused by the impedance variations and subsequent phase anomalies in passive crossover systems.
It is also true that human hearing is only really sensitive to phase errors through the midrange frequencies. Phase errors are inaudible at bass frequencies, and quite difficult to detect in the treble. So if you can keep the crossover out of the midrange, and particularly if you can keep it below 500 Hz, the steepness of the crossover slopes becomes a non-issue.

But even if you are convinced that steeper crossover slopes produce audible phase error, you need to look at the big picture. There can be no doubt that using drivers beyond their frequency response capabilities (like using woofers in the midrange and highs) and having a lot of shared frequencies between multiple drivers that are at different distances to you, severely degrades the sound.

This degradation does far worse things to the sound than any smooth phase error can. So it is well worth trading a little smooth phase shift for the much better frequency response, tight control of the sound, and improved sound quality that you can get by using steep crossover slopes.

I am not willing to compromise the sound quality of my speakers. So I simply will not, and do not use passive crossovers. To assure that there is no adverse sound quality from the crossover, I use very low crossover points (172 Hz in the Model 10c and 220 Hz in the Model 11).

No matter how good a woofer is, it simply can't match the spectacular detail and clarity available from a single, massless ESL in the midrange. So I use 48 dB/octave slopes to assure that the woofer only operates in the bass. The ESL reproduces all the frequencies from the upper bass to beyond the treble as a single driver without any crossovers at all.

The ESL must not be operated at or near its fundamental resonance or the speaker will exhibit high Q behavior (overshoot and ringing), which sounds awful. By using very steep slopes, I am able to operate the ESL down to within an octave of this resonance without exciting it.

The equalization facilities of electronic crossovers allow me to compensate for the midrange phase cancellation inherent in all dipole radiators. This is a far better way to correct this frequency response error than by using multiple panels of different sizes and crossovers to operate them at different frequencies as is commonly done in many ESLs.

As a result of the use of electronic crossovers, particularly digital ones, the performance of my hybrid ESL is now better than a full-range, crossoverless ESL. This is because my ESL is operated nearly full range (172 Hz to 34 KHz), and yet it has a superb transmission line woofer that will produce prodigious amounts of deep bass, which a full-range ESL cannot hope to match.

In summary, electronic crossovers are essential if you want outstanding speaker performance. This is especially true of hybrid ESLs.

I hope this information, although limited in scope, has been helpful. If you have further questions, please feel free to contact me.

Great listening,
-Roger
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#8
Bad time to be over fifty and in any high paying profession.
Thank you for all your responses. They have been very informative and challenge some of my preconceived ideas.

I have advocates that the back wave of such a large dipole be killed or attenuated. Is your speaker radiating a back wave? What problems if any does this present for your speaker and dipoles in general.?Should we just leave it alone, buy absorption material or just move it away from wall?
You may attenuate the back wave of a dipole if you wish. But I generally do not consider this necessary or desirable. Let me explain the situation and you can draw your own conclusions.

The problem is that the rear wave from a dipole is just as powerful as the front wave. If this bounces off a nearby wall and comes back through the speaker, the frequency response will be ruined.


The reason for this is that the rear wave of the dipole that is reflected off the back wall will be slightly delayed compared to the front wave. So when the rear wave bounces back through the speaker, it mixes with the front wave and the its energy at each frequency will be either subtracted from or added to the energy from the front wave -- it all depends on the phase angle of the rear wave.


For example, let's assume that the timing (phasing) is such that at 1 KHz the rear wave phase angle is 180 degrees. This means that the rear wave will cancel the front wave and you will hear essentially no output at that frequency.


At 2 KHz, the phase angle will be 360 degrees, so the output at that frequency will be doubled (3 dB greater than the front wave alone). These are the extremes. Other frequencies will be at different phase angles that will either augment or depress the output by varying amounts.


The result of this type of strong reflection will be that the frequency response of the speaker will consist of a series of peaks and dips in the frequency response that resemble the teeth of a comb. So this type of problem is often referred to as a "comb filter." Needless to say, a comb filter sounds perfectly awful and must be avoided.


This problem can only occur if the rear wave bounces off the back wall and then comes back through the speaker where it can mix with the front wave.
This means that the speaker needs to be at least close to parallel to the wall. It won't be a problem if the angle of the speaker to the wall is significantly off parallel as any significant angle will result in the rear wave being bounced AWAY from the speaker. When this is the situation, a comb filter will not occur.

Of course, if the diaphragm is curved, the problem will occur over a much wider speaker angle than if the panel is planar. And note that regardless of the speaker's angle, the further away from the wall, the less likely that the reflected rear wave will hit the speaker on its first bounce. This is why moving the speaker further away from the wall usually helps the situation.


In the case of my speakers, they must be toed inward so that they face the listener. As a result, they are at a rather severe angle to the wall. This bounces the rear wave away from the speaker so that no comb filter will occur. So it is not necessary to place damping behind my speakers to correct the comb filter problem. The angle is sufficient that you can put my speakers directly against the wall and there still will be no comb filter formed.


Placing damping on the wall behind a dipole has a disadvantage. The rear wave will bounce around the room and add high frequency energy to the off-axis sound. So for casual listening when you are not at the sweet spot, dipoles will sound better if you don't absorb their rear wave.


Also, keep in mind that damping materials work best at the higher frequencies. As you get down into the mid-range, they won't absorb all the sound, so you usually will still get a comb filter in the mid-range frequencies, even if you try to absorb the rear wave.


This frequency-dependent behavior of damping materials also means that absorbing the rear wave will result in the rear wave mid-range frequencies escaping into the room while the highs are essentially totally absorbed.

The result will be a relatively mid-range-heavy sound with loss of highs when you casually listen out of the sweet spot.

So I generally do not recommend absorbing the rear radiation of a dipole.

It works better to toe the speaker inward so that it reflects the rear wave away from the speaker to eliminate the comb filter.

There is one exception to this recommendation, and that is if your room is very small. For example, when my speakers are used in recording booths, the rear wave energy bounces around in the tiny room and very quickly reaches the sweet spot, which can adversely affect the sound.


In a large room, this is not a problem because the rear wave energy is so delayed and attenuated by the time it reaches the sweet spot that our ears ignore it. But in a very small room, this is not the case. So I recommend absorbing the rear wave in very small environments.


In summary, for typical listening environments, it is not necessary to absorb the rear wave from a dipole as long as the speaker is not placed near parallel with the rear wall. If it is, then you can probably solve the problem by moving the speaker further out from the wall.


You may absorb the rear wave if you prefer. But this is not effective at mid-range frequencies and tends to make casual listening sound rather dull.

So it is not my preference.

-Roger
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#9
Well thank you for that important piece of information. It proves that common sense will only get you half -way there. And now for some controversy.
Tube vs. Transistor.You recommend solid state amplification for speakers. I can't dismiss solid state out of hand anymore. There are some excellent models out there. I also recognize the short comings of tube amps. At the price I am willing to pay. Hybrid seems the only way to go. Just can't imagine any solid state amp under $30k that I could live with. Ultimately it's the sound that matters. At least theoretically how do you support your choice of solid state over tubes?
Here ya go, another chapter from The Rogernator:

I certainly agree with you that there is a lot of controversy surrounding tubes vs. transistors. But since you asked, I'll explain why I recommend solid state amps for driving my ESLs.

To do so, I must first give you some history and discuss some technical issues in what I hope will be understandable. I have designed and built tube amps going all the way back into the 60's. My most memorable design was a Class A, high voltage, transformerless output, direct-coupled tube amp for driving electrostats. I published the design in "The Audio Amateur" magazine back in 1976. You can still read it on my website at:
http://sanderssoundsystems.com/downloads/TheAudioAmateur0May1976.pdf

I have also used many other tube amps over the years and have also helped design the iTube, which was a conventional tube amp that was optimized to drive ESLs. The point I'm trying to make is that I don't have any bias towards one or the other type of device. I've used, designed, built, and marketed both types. So what I am about to say does not come from any particular partisan point of view. It is simply what I have learned over the last 38 years of research into producing the best sound I could.

I have been in the unusual position (for an audiophile) of having a fully-equipped test bench, including a spectrum analyzer. This has made it possible for me to carefully do both measurements and listening tests to correlate the two and to find out the reasons we hear the things we do. This research has been fascinating and very educational. It has also made it possible for me to develop truly high-performance electronics.

There is no doubt that we all hear differences between tube and transistor amplifiers. The big question is what is causing the differences we hear between them. After all, well-designed examples of both types measure well enough that we should not hear any differences between them. So what gives?

I spent a lot of time looking for the reasons. It was an extremely interesting and entertaining search. I don't have time to explain all the work I did over the years in these studies this message, but will be happy to discuss them over the phone (303 838 8130) if any reader wants to know. I'll just have to summarize here.

To begin, you need to understand how much power is required to play musical peaks cleanly and without clipping an amplifier. It takes a surprising amount.

To see what is going on with an amp when playing music only requires an oscilloscope. These are very fast (the slowest ones will show 20 MHz) and will clearly show amplifier peak clipping when music is playing. A meter is too slow to do so. A 'scope is cheap (you can get them for $100 on eBay all day long). So you don't have to take my word for what I am about to explain. Feel free to get your own 'scope and examine your system's performance.

You simply connect the 'scope across your speaker or amplifier terminals (which are electrically the same), adjust the horizontal sweep as slow as you can while still seeing a horizontal line on the screen. Don't go so slowly that you see a moving dot.

Now play dynamic music at the normally loud levels you enjoy. Adjust the vertical gain on the 'scope so that the trace stays on the screen.

As music plays, you will clearly see if clipping occurs. The trace (which will just be a jumble of squiggly lines) will appear to hit an invisible brick wall. It will appear as though somebody took a pair of scissors and clipped off the top of the trace. That's where the term "clipping" comes from.

If you see clipping at the levels you like to listen, then you are not using a sufficiently powerful amplifier to play your music cleanly. Your system is compromised because your amplifier will have compressed dynamics, sound strained, lose its detail, and have high levels of distortion.

The 'scope will be calibrated so that you will know the voltage at which clipping occurs by observing the grid lines. If you know the voltage and the impedance of your speakers, you can easily calculate the power.

Power is the voltage squared, divided by the impedance. So if the 'scope measures 40 volts at clipping, and you are driving 8 ohm speakers, you know that 200 watts are being produced at clipping -- and this is insufficient power for your particular system because it is clipping.

You will find that conventional, direct-radiator (not horn-loaded), magnetic speaker systems of around 90 dB sensitivity, typically require around 500 watts/channel to avoid clipping. More power is needed in larger rooms or if you like to play your music more loudly than most.

The key point I'm trying to make is that audiophiles usually are using underpowered amplifiers and are therefore listening to clipping amplifiers most of the time. When an amplifier is clipping, it is behaving (and sounding) grossly differently than its measured performance would suggest. This is because we always measure amplifiers when they are operating within their design parameters -- never when clipping. A clipping amp has horrible performance, so attempting to measure it is a waste of time.

In other words, we usually listen to an amplifier when it is clipping and we measure it when it is not. This is why amplifiers sound so different than their measurements would imply. It is not that measurements are wrong, it is simply that we are listening and measuring different conditions.

It is essential to understand that when an amp is clipping, it will sound quite different than when it is not clipping. It is also important to realize that different types of output devices (tubes vs. transistors) clip in very different ways, so sound quite different when they are clipping.

Finally, it is important to realize that an amp does not instantly recover from clipping. It takes several milliseconds for its power supply voltage to recover, for it to recharge its power supply capacitors, and for its internal circuitry to settle down and operate properly again. Therefore, even though an amp may only be clipping on the musical peaks, it will not immediately operate properly at average music levels where it is not clipping.

It should now be obvious why objective measurements don't seem to give much insight into the performance of amplifiers. It is not that objective measurements aren't accurate (they are superb), but simply that we don't usually operate amplifiers within their design parameters. So we aren't listening to them at the power levels where they operate properly and where their measurements are meaningful.

Now let's analyze tube and transistor equipment with regards to clipping, since that is the condition to which we usually listen. There is "hard" and "soft" clipping. If you go back to the oscilloscope investigations, you will see that solid state amps clip "hard" in that there is an absolute, rock-solid, limit to how loudly they will play. As soon as you reach that point, they immediately clip. This point is their power supply rail voltage.

A tube amp clips "softly." This is because tubes produce a cloud of electrons around their cathodes.

This cloud has surplus electrons available so that for sudden current surges (such as musical peaks), a tube can deliver more current (electrons) and voltage for a few milliseconds before they clip. So their clipping threshold is not rigidly fixed as it is in a transistor amp. It varies depending on the dynamics of the music played.

The age of the tube matters a lot in this situation. As a tube ages, its emissions decrease and it cannot develop as many electrons in the cloud. So old tubes will tend to hard clip while new tubes will tend to soft clip.

Transistor amps usually must employ protective circuitry. Tubes do not need any. Protective circuitry will trigger anytime a transistor amp "sees" an excessive or dangerous load. Generally, this means that most transistor amps will trigger their protective circuitry at or about the time of clipping. They will also go into their protection modes at very low power levels if they see difficult loads (like electrostatic speakers).

Protective circuitry works by switching off the power to the output transistors for very brief periods of time. Well-designed protective circuitry will trigger on and off hundreds or even thousands of times per second to limit the power that the output transistors must handle.

Protective circuitry sounds awful. It literally puts gaps in the music, which adds a type of grainy quality to the sound. But more importantly, anytime you flip a switch, whether it is a light switch or an output transistor, you will get a voltage spike. So protective circuitry will replace a smooth musical signal with a chopped up one that has voltage spikes on each side of a gaps in the music. Is it any wonder that transistor amps sound harsh when clipping?

In addition, when a tube amp clips, it produces a lot of lower harmonics in its distortion profile. Low harmonics are relatively benign and don't sound too badly. But distortion is still distortion and these harmonics don't belong there. Also, just because a tube amp makes a lot of lower harmonics, doesn't mean that it doesn't also make higher harmonics, it does. And high harmonics tend to sound dissonant and unpleasant.

This is easily seen on a spectrum analyzer, which shows each harmonic and the percentage of distortion it adds to the sound. It truly is an amazing tool.

Transistor amps tend to produce a lot of high harmonics. This is actually due more to the operation of their protective circuitry and all the spikes it produces. So generally, transistor amps will have more of the unpleasant higher harmonics than do tube amps.

It is important to note that if a transistor amp does not have any protective circuitry, its distortion profile will be much more similar to a tube amp than to a transistor amp with protective circuitry. The effect of protective circuitry is a very critical issue in the sound of solid state amps and should be more widely recognized for the problems it introduces to the sound.

What all this boils down to is that clipping tube amps sound rather soft and smooth. Clipping solid state amps sound harsh and edgy. I think it is safe to say that we would all agree that if you must listen to a clipping amp, a clipping tube amp is more pleasant than a clipping solid state amp.

It should now be apparent from where "tube sound" and where "transistor sound" comes. It comes from the sound of clipping amplifiers, which do indeed sound quite different.

Of course, when clipping, neither amplifier sounds good. They both lose their dynamics, sound "mushy", lose their detail, sound strained, tend to sound harsh (particularly transistor amps), and are somewhat distorted.

Note carefully that human hearing is rather insensitive to transient distortion, so even though both amps will produce several tens of percent distortion when clipping, we generally won't recognize the distortion for what it is, because it is too brief. Instead we will perceive and describe the sound as "harsh", "strained", "fatiguing", "muddy", etc.

To have a truly high-fidelity music system therefore requires very powerful amplifiers. Amplifier power is the single most important factor in choosing an amp. Without adequate power, all amplifiers sound badly. You can pick a clipping amplifier based on it not sounding as badly as another amplifier (tubes usually preferred over transistors), but if you really want clean, dynamic, effortless, and smooth sound, you simply must use adequate amplifier power.

In short, my take on amplifiers is to use a tube amp that clips gracefully if I must listen to a clipping amp. But I'd rather have an amplifier with so much power that it never clips! The sound from powerful amps is dramatically better than underpowered amps, even if they clip nicely.

There are three quality criteria that a good amp must meet. It must have inaudible noise, it must have flat frequency response, and it must have distortion of less than 1%.

Interestingly, tests conclusively show that humans cannot hear distortion of less than 1%. So even though one amp may have 1% distortion and another 0.001% distortion, they will both sound identical to us.

My spectrum analyzer will show distortion down to around one ten thousandth of one percent (0.0001%). It shows amazing differences between properly operating amplifiers. But as long as those distortion levels are below 1%, the amps will not sound any different to us.

It should now be clear that tubes only sound significantly different than transistors when you are listening to clipping amps. If the amps aren't clipping, or if you are using a component that doesn't clip (like a preamp), you won't hear any significant difference between well-designed tube and transistor equipment. So a hybrid amp (tube front end and transistor output stage) that is not clipping will not sound any different than a pure tube or transistor amp. And if it is clipping, it will sound like a transistor amp, not a tube amp, because it is the type of output stage that determines the sound of a clipping amp.

Now with the historical and general information covered, I can now turn directly to your question. So let's examine tube and transistor amplifiers with respect to their performance with ESLs (because I am a manufacturer of ESLs).

Recall that the basic quality performance criteria requires that an amplifier have flat frequency response. This is a huge problem for tube amps due to impedance variations in the load. Let me explain.

One of the laws of physics states that the source impedance must be lower than the load impedance or the load will be starved for current. What this translates to is that the amplifier's output impedance must be lower than the speaker's input impedance or the frequency response will be rolled off in those areas where there is this impedance mismatch.

Tubes are inherently high impedance devices. A large power tube like a 6550 or KT-88 has an output impedance of around 2,000 ohms. By comparison, a large power transistor has an output impedance of less than one ohm.

Tubes cannot drive loudspeakers directly due to their high impedance. To correct this problem, output transformers are used in most tube amps. These transformers have a specific turns ratios that will convert the tube's impedance from several thousand ohms to typically 4, 8, or 16 ohms.

Therefore, if you use the 8 ohm taps on the amplifier's output transformer with an 8 ohm loudspeaker, there should be no impedance mismatch, the frequency response should be linear, and the amp should deliver its maximum power. Unfortunately, this is never the case because loudspeakers do not have a constant impedance across their full frequency bandwidth.

Look at the impedance curve of any conventional loudspeaker and you will see that it varies from slightly below its "nominal" impedance to around 50 ohms. This will cause the frequency response from a tube amp to have errors. This is also another reason why tube amps sound different from transistor amps.

This impedance problem is relatively minor when dealing with conventional, magnetic speakers. But an electrostatic speaker is an entirely different animal. An ESL is a capacitor, not a resistor like a magnetic speaker. The impedance of a capacitor is inversely proportional to frequency. Therefore the impedance of an ESL typically varies from around 150 ohms in the midrange to about 1 ohm at 20 KHz.

A tube amp will be able to drive the high impedance frequency bandwidth (the midrange and lower highs) of an ESL with linear frequency response. However, at higher frequencies, the impedance of the ESL will drop below the impedance of the amplifier and the amp will then roll off the highs to some degree depending on the exact impedance mismatch and the frequencies involved.

This impedance mismatch problem can be minimized with both types of speakers by using a lower impedance tap on the tube amp's output transformer. For example if you use the 4 ohm tap with 8 ohm speakers, you will probably not encounter any impedance mismatch, so the system would then have linear frequency response.

Using the 4 ohm tap with ESLs will help, although it will still not eliminate all the high frequency impedance mismatch because the speaker's high frequency impedance will fall below 4 ohms. But probably only the top octave or two will be affected, which is hard to hear so the roll off may not be noticed subjectively.

But there is a problem when you use a lower impedance tap -- the drive voltage drops. Or to put it another way, the amplifier's output voltage is directly proportional to its output impedance.

Understand that the power available from an amplifier is a function of its output voltage. Ohm's Law is very simple and states that, "One volt will drive one amp through one ohm." With this simple concept, you can calculate virtually anything having to do with electronics as you will soon see.

Voltage is the pressure used to push current through an electrical circuit. Current is the flow of electrons in the circuit -- like water flowing though a hose. Current is measured in amperes, commonly called "amps." Power is measured in watts and is the product of volts times amps.

Resistance is measured in ohms. The term "resistance" is used in DC (direct current) circuits. "Impedance" is the same thing as resistance. But it is used when discussing AC (alternating current) circuits because the impedance often varies with the frequency of the AC.

Since power is the product of volts times amps, you can see that you must get current to flow through the speaker's impedance. This requires volts.

For example, if you have an 8 ohm speaker, how many volts must the amplifier produce to push enough current through the speaker to produce 100 watts? How many amps of current will be flowing through the speaker at 100 watts?

There are simple calculations for determining this. The volts can be calculated by taking the square root of the power times the impedance. So for the example above, the watts are 100, multiplied by 8 ohms, gives you 800. The square root of 800 is 28.2 volts (RMS).

The current can be calculated in several ways, but the most common is take the square root of the power divided by the impedance. So in this case, the current flow would be 3.5 amps.

If you have a 100 watt amplifier, you can see that its output voltage will be limited to about 28 volts. If it could produce more voltage, it could produce more power, so you know that its voltage will be limited to 28 volts or it would have a higher power rating. Of course, all this assumes that the amplifier's power supply and output impedance is such that it can deliver the 3.5 amps needed to produce 100 watts of power.

You can also calculate that if an amp can produce 28 volts into 4 ohms (half the impedance of the above example), that the current would double to 7 amps and the power would double to about 200 watts. Hence you see transistor amps with power ratings listed for both 8 and 4 ohms.

Tube amps are different in that if you reduce the impedance of the transformer from 8 ohm to 4 ohms to match the impedance of the speaker, the output voltage will drop as a function of the turns ratio of the transformer, and so will the power.

The turns ratio is the square root of the primary impedance divided by the square root of the secondary impedance. This always works out such that the voltage will drop to the point where the amplifier will put out the same power at either impedance when driving a matching load.

When driving an ESL, voltage is everything. So when you drop the impedance of the output transformer, you reduce the output that the amplifier can produce from the ESL. In short, you have to trade output for more linear frequency response. This is a huge problem. It's a battle that you just can't win.

Note that OTL tube amps don't solve this problem. They have no transformer, so must relay on putting many output tubes in parallel to lower the impedance. This quickly results in having an absurd number of tubes with all their heat and power requirements. So OTL amps do not get down to very low impedances.

Most get to just around 10 ohms and the best only get a bit lower. As a result, they have severe impedance mismatch issues and are really quite a poor choice for driving ESLs. They also measure really poorly on a spectrum analyzer compared to transformer-coupled tube amps.

By comparison, powerful solid state amps typically have output impedances of around 0.02 ohms. They therefore have no trouble driving any speaker impedance with perfectly linear frequency response.

Solid state amps have high output voltages compared to tube amps. So they will drive ESLs quite loudly before clipping (unless they have protective circuitry that trips them up).

Well designed solid state amps have much lower distortion than tube amps. The best conventional tube amp I've ever measured was a McIntosh 275 with new tubes. It had only 0.3% distortion at an output level of 75 watts/channel (it clipped at 90 w/c). Most tube amps have distortion of somewhat more than 1%, even at levels well below clipping.

My specially designed iTube amp measured only 0.1% distortion in the midrange frequencies and went up to 1% by 20 KHz. It would do so at 150 w/c. But this was a special-built device and is not typical of conventional tube amps.

By comparison, most quality, solid state amps have distortion levels down around 0.002%. This is magnitudes better than tube amps. However, it is also true that humans cannot hear the reduced distortion levels in solid state amps, even though a spectrum analyzer will show dramatic differences between them.

Still, distortion is distortion. Why have any more than you must?

I think you can now see why I prefer very high power, solid state amps without any protective circuitry for driving ESLs. This is because they can drive ESLs with linear frequency response, while tube amps roll off the highs.

Solid state amps are much more powerful than tube amps and can supply vastly higher output voltages. As a result, a good solid state amp can drive my ESLs to ear-bleeding levels without clipping. And remember, it is clipping that produces "tube" or "transistor" sound. If a solid state amp does not clip, it does not sound harsh. It sounds just as clear and soft as a tube amp that is not clipping.

Transistor amps can run cool and efficient. My ESL amp runs only warm, yet can deliver the equivalent of about 1000 watts into an electrostatic speaker. No tube amp can do so, and even a relatively low power tube amp will run very hot and waste a lot of expensive electricity.

Tube amps are expensive compared to good solid state amps of similar power. Tube amps require expensive tube replacements, while a quality solid state amp is a no-maintenance, lifetime item.

Tube amps require biasing. Traditionally this had to be done by the audiophile on at least a monthly basis. This was a hassle and rarely was done, so most tube amps were always running far from their ideal performance levels.

Some tube amps tried to get around this biasing issue by using a "self-biasing" system. But this cut their power by about 30%. Some of today's latest tube amps use servo biasing systems, which are great if they work reliably. Often they don't.

Due to their high internal voltages and high temperatures, tube amps are unreliable. They often fail and have to be returned to the manufacturer for expensive repairs.

In short, tube amps can't drive ESLs linearly, cleanly, without clipping, to high output levels. So why put up with all their problems of heat, cost, maintenance, and unreliability when a properly-designed, solid state amp solves all these problems?

I therefore no longer design, manufacture, or recommend tube amps. I only build very powerful solid state amps that have no "transistor sound" because they do not clip or have any protective circuitry to ruin the sound.

Using my ESL amp on the panels of my speakers, and my Magtech amp on the woofers, results in approximately 1,400 watts of power for the speakers. This makes it possible to reproduce something like a grand piano or drum set at live levels in your listening room without clipping. You can reproduce a full symphony orchestra at Row A concert hall levels.

If you have a particularly large room or play your music at ear-bleeding levels, you can use the monoblock versions of my amps. The ESL amp will deliver more than 2,000 watts to the panels and the Magtech will deliver about 1,800 watts to the woofers. Clipping simply isn't an issue and the speaker can take the power.

This performance simply cannot be obtained using conventional tube equipment. So I really have no choice but to use and recommend excellent solid state amps.

I suspect that much of what I have just said is hard for you to believe or accept since you are obviously a "tube guy." I have no problem with that as clipping tube amps do sound better than clipping transistor amps, so I can appreciate that you prefer tubes.

I would be happy to send you one of my Magtech amps to compare to your tube equipment. I think you would be surprised at how good a high power, low distortion, voltage regulated, low impedance, solid state amp without any protective circuitry sounds -- and it is a reasonable price. I would send you an amp to demo at my expense (free round trip shipping). You can then draw your own conclusions and tell the forum what you discover.

Great listening,
-Roger
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#10
I believe that I heard this iteration of the Sanders at THE Show in 2009, but I am not sure that it is the same as what HP reviewed. However, I will say that what I heard impressed me greatly, as it did for Marty, although he was not quite as excited as I was. I have been a long-term electrostat fan because of the "nature" of their sound, even though I suspect that this "nature" is not necessarily the truest sound out there. Let me make my bias completely clear- I like this sound because it pleases me, not because it is necessarily the truest. Despite the fact that I owned a set of Wisdom M-75 Limited Edition References (hand picked for response), I must admit that I liked the stat personality better.

Given my bias, I found the Sanders to be among the best sounding stats and least obvious hybrids I have heard yet, and therefore liked them a lot. There is little doubt that they are highly directional and thus you must sit in the sweet spot to get the best results. Granted you can change the angles of the speakers or the size of equilateral triangle as HP says to change the image, but this is still a very socially unfriendly speaker due to a very small sweet spot.

Back in the days when I had an A grade system and room, my friends would joke that there should be a jig like one used for brain surgery in the "correct" position at the sweet spot. I even provided pillows to raise the head height of vertically challenged listeners. They were right and when you had your head in "the spot" magical things happened. Listening with my family was not one of them. When audiophiles came over, we took turns listening and when it was not our turn, we stood and observed the chosen one.

Maybe I am getting old, but this longer works for me. I like people more than my sound system. Yeah, I still choose the correct 3-4 seats in a 500-2000 seat theater when I go to a movie or concert because it is an event and usually there are two adjacent seats that qualify, thus allowing me to be happy in my selfishness for two hours. However, I live with my sound system and listen to music, watch TV and movies with it. I want it to disappear and I want ALL of my company to enjoy the experience.

Simple physics and acoustics clearly demonstrates that for a sound to sound real and thus believable (meaning my brain interprets it as real) the sound wave must START in a phase coherent manner, because that is the way sounds occur in the real world. Through triangulation and parallax of the signals obtained at your ears and eyes, your brain determines its position and you are satisfied that the sound is real and located at a specific position. When the sounds bounce off of environmental obstacles, your brain is able to use this information (phase alterations and time delays) to help understand the environment, thus maintaining the live aspects of the sound. The reason why I enjoy listening to my very phase coherent BG speakers on my deck after the sound winds its way through windows and sliders is because it started out initially extremely phase coherent and thus still sounds real as it emanates from my windows and doors, much like the voice of someone talking standing next to my speakers when I am hearing it from my deck. The difference is I believe that they and the music are in my living room and I am hearing it on my deck. Still quite pleasing since as a non-psychotic person, that was all I expected in the first place.

So unless you are lone listener with a system and environment with a vice-like sweet spot, regardless of how good these speakers are you have to determine your priorities versus other aspects of life.
here's a reply from Roger

[FONT=&quot]Thank you for your kind compliments on my speaker designs. I would like to clarify a couple of points regarding directionality and the size of a speaker's sweet spot.[/FONT]

[FONT=&quot]Understand that all stereo speakers, regardless of their dispersion characteristics, have an infinitely small sweet spot. This is that point when you are exactly equidistant from both the left and right speakers. [/FONT]

[FONT=&quot]If you are at different distances from the speakers, their sound will arrive at different times. This puts them out of phase to some degree. A precise image can only be obtained when the sound from the speakers are in-phase where their sound arrives at your ears simultaneously. Therefore it is a physical impossibility for any speaker to have a "wide" sweet spot.[/FONT]

[FONT=&quot]Since you cannot circumvent the laws of physics, if a speaker seems to have a wide sweet spot (where the image appears to be identical over a wide area), then that can only mean that its image is flawed. The usual cause of this is that wide dispersion speakers excite room acoustics. This produces a great deal of delayed and out-of-phase information that mixes with the sound from the speakers and confuses the phase information. [/FONT]

[FONT=&quot]Without accurate phase information, the image is degraded to the point that you simply can't tell when you are in the actual, true, sweet spot. As a result, you get a relatively poor image over a wide area. [/FONT]

[FONT=&quot]In my current speaker designs, I chose to avoid compromise and strive for the most perfect image possible. This required that I eliminate the room acoustics which degrade the imaging of speakers. [/FONT]

[FONT=&quot]To do so, I took advantage of the precision wave and phase coming off the front of a large, planar panel to direct the sound directly to the listener instead of spraying the sound all over the room, which ruins the phasing (and transient response and frequency response). I note with satisfaction that you found that I succeeded in this endeavor as, "magical things happened" when you experienced the sweet spot of my speakers. This is as it should be.[/FONT]

[FONT=&quot]I appreciate that some listeners like to share their listening experiences with others and that you like to have guests sit side by side while listening in your home. It is most regrettable that the laws of physics don't allow all side-by-side listeners to be in the sweet spot with any set of stereo speakers -- only the center listener will be in the sweet spot. [/FONT]

[FONT=&quot]But why should we sacrifice the quality of the image at the sweet spot just because others are listening? Listeners to either side of the sweet spot will never hear an excellent image, so the "chosen one" at the sweet spot might as well hear the best image possible.[/FONT]

[FONT=&quot]But it doesn't have to be that way. It is possible for multiple listeners to all hear superb imaging. The way to do so is for everybody to sit in tandem instead of side-by-side. Then all listeners can be equidistant from the speakers and therefore in the sweet spot. Let me stress again that this is true for [FONT=&quot]all stereo speakers, not just highly directional ones.[/FONT][/FONT]

[FONT=&quot]You may be used to side-by-side seating, but this is not a requirement for social situations. After all, you commonly have passengers in the rear seat of your car where you are in tandem, and you converse easily with them. So why not when listening to music?[/FONT]

[FONT=&quot]If you feel that side-by-side seating is necessary, then listeners outside the sweet spot will still hear excellent sound with the outstanding detail only available from ESLs. It is true that the image will not be ideal -- but it is not ideal when listening to any set of stereo speakers when you are out of the sweet spot. The only difference with my directional speakers is that when you are in the sweet spot, the image is so remarkable that you really can tell when you are there. [/FONT]

[FONT=&quot]I do not feel that the term "head in a vise" is either fair or accurate. Ask any of the hundreds of customers who own my speakers and they will tell you that there is no such sensation or problem. To listen, they simply sit down and relax in their listening chair and listen to music they would with any speaker. There is no sense of needing to locate their head in any sort of precision manner. They just sit down and enjoy the music with the outstanding image quality offered by speakers that eliminate room acoustics.[/FONT]

[FONT=&quot]In summary, if you want really great imaging that is truly holographic and 3-dimensional, then you must listen at a stereo speaker system's sweet spot. For a single listener, this is no problem regardless of a speaker's dispersion characteristics. Multiple listeners must sit in tandem to hear the best imaging. Side-by-side listening will always compromise image quality with any stereo speaker system, but may be an acceptable trade-off for some social situations.[/FONT]

[FONT=&quot]Great listening,[/FONT]
[FONT=&quot]-Roger[/FONT]
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#11
I assume your speakers image well. In assessing the speakers imaging characteristics, what measurements do you rely on? Or do you simply voice it by ear?

Imaging is a geometric phenomenon having to do with the phasing (timing) and shape of the wave front produced by the speaker. This is different than "voicing" which is done by altering the frequency response of the speaker.

Imaging is determined by the physical shape and characteristics of the drivers and is not something that can easily be altered. For example a cone is a cone, and a panel is a panel. There is no practical way to change them. Yet their shape and behavior will largely determine the imaging of the speaker.

While the fundamental phase behavior of the speaker will be determined by the type of driver used, phasing can also be altered by the time alignment of the drivers, the behavior of the crossover, the polarity of the drivers relative to each other, the orientation of the drivers, and the dispersion of the speakers. It is possible to alter the phasing to a limited degree by working with these factors. But the effect will be minimal compared to the inherent imaging produced by the type of driver unless truly dramatic changes are made.

For example, a typical cone or dome driver is a point source, which produces a characteristic image. But you can change this quite dramatically if you use many such drivers and aim them all over the room -- think Bose 901 speakers with one cone facing forward and 8 cones facing backward to bounce sound off the walls behind the speaker.

In this case, most of the sound you hear is that reflected off the walls rather than that coming from the speaker and the image is much different than that coming from a single cone driver. This effect can also be produced by using leaf drivers (MBL) or inverted cone drivers (Ohm Acoustics) that produce 360 degree radiation.

By comparison, "voicing" a speaker involves altering its frequency response so that it is not linear. Generally speaking, most audiophiles will perceive a speaker with perfectly linear frequency response as sounding too bright and lacking in midrange "warmth" and "fullness". It will also not have enough bass to satisfy most listeners. In other words, audiophiles do not like speakers with truly linear frequency response.

As a result, most speaker manufacturers will deliberately alter the frequency response of their speakers to make them sound more pleasing to most listeners. Such "voicing" commonly includes suppressing the highs slightly, increasing the bass substantially, and boosting the midrange modestly.

But voicing a speaker in this way is tricky because loudspeakers interact with the room a great deal. It is the room/speaker interaction that largely determines the frequency response, and because most speakers are sensitive to room placement and shape, it is impossible for a manufacturer to know exactly what his speakers will sound like in any particular room.

For this reason, all good speakers have at least some adjustability so that the owner can tailor them to their room. This means that it is the audiophile/owner who ultimately determines the voicing of their speakers.

Most speakers do not really have enough adjustability to do a really great job of getting the frequency response "right" for a particular room. So most installations will benefit from using a DSP (Digital Signal Processor). By using the RTA (Real Time Analyzer) that is built into all good DSPs, you can see the actual frequency response of your room/speaker system. You can then make adjustments to eliminate problem areas (large peaks and dips) and then tweak the overall frequency response to that which you find most pleasing.

DSPs are extremely powerful devices. As such, they can make major improvements in the sound of your system. But if used poorly, they can truly ruin the sound too. They take practice and patience to use effectively. But it is fair to say that virtually all audio systems can be improved by their use.

As for measurements, the frequency response of speakers are rather difficult to measure accurately in a typical, reverberant, listening room like you find in most homes. Measurements cannot be made using sine waves and SPL meters.

Many audiophiles try to do so by using a test CD and a RadioShack SPL meter. This is not only a complete and utter waste of time, but it is deceiving and will lead you astray because you will think the measurements are accurate when they are not.

The reason that sine wave testing is useless is because the room will interact with the direct sound from the speakers and generate peaks and dips in the response that makes it appear that the speaker has poor frequency response. These peaks and dips will change quite dramatically if you move the microphone just a few inches to a different location. Therefore, the frequency response you will measure is based on microphone position, and does not represent the actual performance of your speaker.

This room interaction problem is the reason that accurate sine wave measurements are always done in anechoic chambers. An anechoic chamber absorbs all the sound from the room so that only the output of the speaker is measured.

But it is possible to measure the frequency response of a speaker in a live room using special techniques. The best is MLS (Maximum Length Sequence) testing. This is also known as pseudo-anechoic testing.

MLS measurements are done by using a quick noise burst (which contains all frequencies) and then using FFT (Fast Fourier Transform) analysis to measure each frequency. A computer is required for this complex measurement technique. But FFT analyzers are now available as computer software for quite reasonable prices that any serious audiophile can use.

To eliminate the room from the measurement, the MLS test is "gated." This means that the system will "listen" to the sound coming from the speaker for only a few milliseconds. Then the microphone is cut off before the room reflections have time to reach it. The result is that the FFT analyzer only hears the sound from the speaker -- not from the room. As a result, it can capture the true frequency response of only the speaker.

But gating has its limitations. It only works for short wave lengths (midrange and highs) which are much shorter than the dimensions of the room. The wave length of bass frequencies are longer than the dimensions of the room. Therefore, you cannot leave the microphone live for a long enough time to "hear" an entire bass frequency without some of the room reflections also getting through. So MLS testing does not work for the bass unless the room is extremely large.

A better way to measure bass is using an RTA. This involves using pink noise and measuring frequency bands (typically 1/3 octave bands). The room is involved in the measurement, but by obtaining many measurement samples and averaging them over time, you can get a reasonably good picture of the bass frequency response. The resolution and detail is low because bands rather than individual frequencies are used, but bass frequencies are non-critical in this regard. So an RTA works quite well for measuring bass.

Because an RTA includes the room in the measurement, it does not tell you what the speaker's bass performance alone is. However, to achieve powerful bass, a room must be involved to support it. So the bass measurement MUST include the room to be meaningful in a typical listening environment.

Think of a marching band. When outside, the band has very poor bass. If you close your eyes and just listen, you will note that the frequency response is similar to what you would hear from a telephone. It's really poor.

But take that same marching band an put it on the stage in a concert hall. Now the bass is spectacular and powerful. That's because the room is confining and supporting the bass energy and making it much louder and impressive.

The same is true of your speakers. Take them outside and they will have very weak bass. But in a room, the bass is really powerful.

In short, you need to measure bass in a room using an RTA. But an RTA doesn't give very detailed information so it doesn't work very well for the midrange and highs. MLS testing is far superior for the midrange and highs.

Also, because MLS testing uses a computer and FFT analysis, it can also display the transient response information. You can see this as impulse response, energy/time graphs, and waterfall graphs. Transient performance is extremely important and a speaker should be evaluated for this performance just as much as its frequency response.

Imaging is a function of our brains that take the phase (timing) and loudness information from two speakers and process it into an image of the original sound. No machine can do this, therefore there is no specific measurement of "imaging."

You can get some idea of how a speaker will image by measuring its polar frequency response because this will give you some idea of its dispersion. Generally wide dispersion will produce relatively poor images because the reflected sound from the room will delay and confuse the phase information. For this reason, narrow dispersion speakers give a much better and more realistic image than wide dispersion speakers. You can get more information on this topic from my White Paper at:
http://sanderssoundsystems.com/technical-white-papers/dispersion-wp

Phase measurements will also give you a clue to the quality of the image. A properly "time aligned" speaker will have the sound from all the drivers arriving at your ears at the same time. With precise phase information, your brain can reconstruct a more accurate image.

Impulse testing is also helpful because it shows the phase behavior between the different drivers. For example, most speakers use passive crossovers and Butterworth filters operating at 12 dB/octave. This causes a 3 dB peak at the crossover point. To eliminate this peak, most speaker manufacturers deliberately put the drivers 180 degrees out of phase.

While this flattens the frequency response, it degrades the phasing and hence the imaging. It would be far better to use different crossover slopes or Linquitz/Riley filters to eliminate the frequency response error. Then the drivers could be wired in-phase. But this costs more money, so most speaker manufacturers sacrifice the phase to improve the frequency response.

But none of these measurements will truly predict the imaging of a speaker. Only your brain can do that. So you will need to listen to the speakers and judge the imaging for yourself.

Always keep in mind that a quality image require a quality recording that is done in true stereo. Few recordings today are made that way. Most are processed to death with artificial reverb, compression, and close mic techniques that are actually done in monaural. As a result, they cannot produce a realistic image -- even if the speakers can.

Older recordings were often made without much if any processing, using pairs of microphones in a natural "hall" environment. So older recordings are often able to produce much more natural imaging than modern recordings. I am convinced that much of the popularity of older vinyl recordings is due to the more natural recording techniques used on them rather than to the fact that they are recorded on vinyl. It is the realism of the recording techniques that make the sound so enjoyable.

With that background, I can now answer your questions in order. Specifically, I pay a great deal of attention to imaging. I deliberately use narrow dispersion speakers to eliminate the room acoustics and I use planar speakers get accurate phasing. I use electronic crossovers and electronic delay to get perfect time alignment of my drivers. I use both MLS and RTA testing to measure frequency response. I measure transient behavior with impulse testing, and produce both waterfall and energy/time graphs.

I "voice" my speakers by giving customers tremendous adjustability in my electronic crossovers so that they can get exactly the sound they want in their particular listening room. Of course, I supply the speaker system fully programmed for what is close to linear frequency response, so the customer need only make two small adjustments to get the sound just right. And it is not absolutely essential that the customer make adjustments. But they certainly can do a lot to make the speaker sound exactly right in their unique room.

-Roger
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#12
OK boys, put your feet up for this one: (angela)

If so, then why do the digital issues of the original recording pale alongside the original vinyl release?
[FONT=&quot]Your question of " . . . why then do the digital issues of the original recording pale alongside the original vinyl release?" is a good one. The answer is much more complex and surprising than you might think and requires a lot of background information before I can answer it. So sorry, but this will be a lengthy discussion. [/FONT]

[FONT=&quot]I agree with your observation that many digital re-issues of the original recording are far inferior to the original vinyl release. Like you, I found this to be very perplexing and troubling. So I embarked on a quest to find the reasons. I think you will find the following story to be most interesting. [/FONT]

[FONT=&quot]In the mid 1960's I had become a serious audiophile and had figured out that better source material was going to be essential if I wanted to get serious improvement in my audio listening experiences. I became so frustrated with the quality of the source material of the day (vinyl LPs and 2-track, open reel tape) that I vowed to start doing my own recordings.[/FONT]

[FONT=&quot]Due to all the restraints on recording imposed by the musician's unions, I found it was really tough to find musicians to record. But after considerable effort, I was finally managed to start using the recording booth at the concert hall of the University I was attending in 1968.[/FONT]

[FONT=&quot]As a poor college student, this was a godsend as I was able to use the finest equipment of the day (Ampex 354 studio recorders, Altec mixers, Neuman and Telefunken condenser mics, etc.) without my having to buy this equipment myself, which I could not have done. I also received excellent training by the staff. But probably the most important advantage was the fact that concerts were always recorded by the University, so I was able to record some truly great performers without union interference.[/FONT]

[FONT=&quot]As part of my recording training, I was also taught to align, maintain, and repair all sorts of recording and electronic equipment -- particularly the studio tape decks, which were always in need of attention. I had access to the University's test laboratory facilities for this purpose. [/FONT]

[FONT=&quot]I soon became one of the top technicians at the University and was able to make all manner of measurements and tests to evaluate and compare equipment to the sound of the recordings I was making. This lucky turn of events put me in the unusual position of being able to do serious, rigorous, scientific testing to figure out the actual cause/effect relationships of what I was hearing. [/FONT]
[FONT=&quot]
What made this so unusual is that I was a concert musician, audiophile, and technician all at the same time. Therefore, I didn't have to just accept someone's opinion about audiophile topics, nor did I have to take the word of engineers who had no knowledge of music or audiophile issues. I could actually do both listening and measurement tests myself to find out the true causes for what I was hearing.[/FONT]

[FONT=&quot]When I graduated from the University, I was able to use my experience there to get permission to do recordings for our local public radio station. I was then able to get past the musician's union problem to be able to record the region's symphony orchestra, opera, and pipe organ. [/FONT]

[FONT=&quot]Because I was doing that, I was also able to get local musical groups to allow me to record them as well. I often could then get them air time on the radio station for which they were very grateful. [/FONT]
[FONT=&quot]
By that time, I was also able to buy my own equipment and start a recording studio as well as do live, on-location recording. So I was very lucky, did a lot of live recordings, and have a wonderful library of music that I have recorded over the last 40 years, much of which is superior to what you can find on the commercial market.[/FONT]

[FONT=&quot]I quickly learned that natural settings in good-sounding concert halls were far superior to the artificially processed recordings that were done in recording studios. Of great importance is that most on-location work was done with a simple 2-microphone setup in true stereo using either the Bleumline, crossed cardioid mics, or widely spaced, omni mic setups. These techniques produced true stereo recordings with natural hall ambience, while studio recordings were done with single microphones that produced inherently dry, monaural sound. [/FONT]

[FONT=&quot]To get the mono sound from a studio to work in a stereo system required panning, artificial reverb, compression, and equalization. Furthermore, studio recording meant that each instrument and performer had to be have their own individual microphone and be recorded in isolation on a single track. The result were multi-channel recordings that then had to be "mixed" to produce the final sound. [/FONT]

[FONT=&quot]This meant that the sound of the final 2-channel mix depended on the judgment of a "recording engineer" to make it sound good. There was no natural or real sound to compare to, so the quality of the recording was totally dependent on the engineer. This was (and remains) a critical flaw in the recording stream. [/FONT]

[FONT=&quot]The close-miked sound from the recording studio was not very realistic, particularly when it was heavily processed and altered to sound "good." I much prefer a simple 2-mic stereo setup in a concert hall.[/FONT]

[FONT=&quot]I continued to do all my own alignment and maintenance work on my analog recording equipment. This was always frustrating because good performance simply was not available. While high-speed, 2-track tape had much better performance than LPs, it was impossible to make recordings good enough that you couldn't tell the difference between the source (usually a live microphone feed) and the recording.[/FONT]

[FONT=&quot]Specifically, even the best recorders were noisy and it was impossible to record a symphony orchestra without hearing tape hiss on quiet passages. The quietest studio recorders had a S/N (Signal to Noise ratio) of around 72 dB. By comparison a symphony orchestra has a dynamic range of about 80 dB. So it physically was not possible to make recordings with a silent background.[/FONT]

[FONT=&quot]The development of noise reduction systems by Dolby and DBX were a help, but they introduced other problems that degraded the sound in exchange for lower noise. The biggest of these was "pumping" or "breathing" noises as their compander circuits opened and closed in response to the music levels.[/FONT]

[FONT=&quot]Linear frequency response in analog recorders was impossible to achieve. It was considered outstanding to get plus/minus 2 dB tolerance from 30 Hz to 15 KHz, which is really quite poor performance.[/FONT]

[FONT=&quot]Distortion was almost a joke. While you could get distortion slightly under 1% at midrange frequencies, the frequency extremes were far worse. In any case, it was normal practice to heavily saturate the tape on loud passages in order to have a quieter background for quiet musical passages. [/FONT]
[FONT=&quot]
While magnetic tape saturates "softly" (like tubes), the distortion at saturation often shoots up to well over 50% on loud passages. Due to the soft nature of the overload, this was sonically tolerable. But the music lacked full dynamic range and had a muddy and confused quality on loud sections.[/FONT]

[FONT=&quot]But the parameter that most annoyed me was the instability of both the frequency and amplitude of the signal. Frequency variations (measured as wow and flutter) rarely were as low as 1%. The slightest mechanical flaw (dirty or worn heads, capstan shafts, or tape guides) would dramatically worsen this. Wow and flutter could easily be heard on critical material like sustained piano tones as a warbling of the sound that was very unnatural.[/FONT]

[FONT=&quot]Amplitude instability made the flutter even worse. You could play back a steady test tone as you recorded it and its amplitude would vary plus/minus 2 dB! [/FONT]

[FONT=&quot]This was highly dependent on the quality of the magnetic coating on the tape. Later advancements in tape technology, particularly the polishing of the tape surface and the use of smaller magnetic particles improved this. But even so, the best reading I ever saw was plus/minus 1 dB at the midrange frequencies (high frequencies were much worse). This was always audible to a critical listener.[/FONT]

[FONT=&quot]Then there was the problem of bias drift. Analog tape is totally dependent on a supersonic bias signal (typically 100 KHz) for achieving low noise, low distortion, and linear frequency response. As the bias oscillator heated up during a recording, the bias current would change. [/FONT]

[FONT=&quot]This would cause a tape deck that I had spent hours "tweaking" to the highest performance possible to change its performance during the recording session. This was truly frustrating as I could never get the best performance from the equipment, even though I was using the finest equipment of the day.[/FONT]

[FONT=&quot]LPs were much worse than tape. I had many pressings of LPs made for customers. The process degraded the sound at every step of the process from using a lathe to produce the lacquer master, through the metal casting, and subsequent pressing with vinyl that was always contaminated with foreign particles that caused the clicks and pops of surface noise. [/FONT]

[FONT=&quot]Note in particular that most of the rumble heard on LPs is actually recorded into the master disk during the lathe cutting process. In most cases, the rumble from the customer's turntable bearing contributes an insignificant amount of rumble compared to the large, noisy bearings in a lathe. Rumble that is cut into the disk cannot be removed by using a super quiet turntable on playback.[/FONT]

[FONT=&quot]So the resulting LP inherently had a large degree of variation from the original master tape. But even if the pressing were perfect and had no errors (impossible), when playing an LP, you have the problem of phono cartridges. [/FONT]

[FONT=&quot]Cartridges are like loudspeakers in that they are transducers and therefore there are large differences in the sound of cartridges. Between the errors introduced during the production of an LP and the variances of phono cartridges, the sound from an LP sounds quite obviously different than the sound on the original master tape -- which sounds significantly different from the live microphone feed.[/FONT]

[FONT=&quot]Although a high-speed, master tape was the best storage medium we had, it still corrupted the sound quite obviously. Everyone could easily hear the difference between the recording and the live microphone feed. In fact, every preamp of the day had a tape monitor loop so you could compare the source to the recording in real time (if you had a 3-head tape deck), and there always where differences you could hear between the two. [/FONT]

[FONT=&quot]In short, analog recording is seriously flawed and never sounds like the source. Something better was badly needed.[/FONT]

[FONT=&quot]By the 1980's, digital recording had been developed that had the potential to solve the technical problems that were insurmountable with analog equipment. Of course, with an entirely new process, there were some teething problems. Initially there were serious problems with insufficient data storage, low-level accuracy of DACs, and a weird problem with 1/3 order harmonic distortion that was eventually eliminated with the introduction of dither. [/FONT]

[FONT=&quot]But the quality of digital recording quickly progressed and by the mid '80s, it was possible to make digital recordings that sounded identical to the live microphone feed. They had perfectly linear frequency response (DC to 20 KHz +/- 0.1 dB), lower distortion than most instruments could measure (less than 0.002% THD), unmeasurable wow and flutter, unmeasurable amplitude instability, and a totally silent background (S/N of better than 92 dB). [/FONT]

[FONT=&quot]Let me take a momentary detour here and comment that most audiophiles today still believe that linear PCM (Pulse Code Modulation) such as used on CDs and that requires a DAC (Digital to Analog Converter) produce digital steps in the wave form. This is simply untrue. [/FONT]

[FONT=&quot]You need look no further than to observe the recorded wave form on an oscilloscope to see that modern DACs work so well that the wave form is absolutely smooth and cannot be distinguished from the source wave form. Even a 20 KHz tone from a CD, which has only two samples at that frequency will be perfectly formed, utterly smooth, and will have distortion of around only a thousandth of a percent. [/FONT]

[FONT=&quot]The purpose of a DAC is to produce smooth wave forms and they do so brilliantly. There are simply no steps in the wave form of a PCM recording. [/FONT]

[FONT=&quot]These audiophiles then further believe that there is some mysterious measurement called "resolution" and that higher sampling rates improve the "resolution" of the sound. This is also nonsense. There is no such measurement as resolution and there are no steps in the wave form of a PCM recording.[/FONT]

[FONT=&quot]So if the sampling rate does not improve "resolution", what does it do? The sampling rate defines the highest frequency that the digital recording system can record and store. The sampling rate in a PCM system must be twice the highest frequency to be recorded. [/FONT]

[FONT=&quot]The CD "Red Book" that specifies the performance of a CD requires a 40 KHz sampling rate. So a CD can record sounds up to 20 KHz -- the limit of human hearing.[/FONT]

[FONT=&quot]"But wait," you'll say, "CD's are sampled at 44.1 KHz, not 40 KHz." True. The additional 4.1 KHz above 40 KHz are used for the anti-aliasing filter. This filter is required to remove any frequencies above 20 KHz, which would confuse the digital converters and cause errors and flaws in the recording. [/FONT]

[FONT=&quot]A sampling rate of 96 KHz will record up to 40 KHz (requires 80 KHz sampling). The extra 16 KHz are used for the anti aliasing filter. The 192 KHz sampling rate will use the first 160 KHz to record up to 80 KHz with the remaining 32 KHz being used for the filter.[/FONT]

[FONT=&quot]So the sampling rate only defines the high frequency limit of the recording. It has nothing to do with "resolution" in PCM recordings.[/FONT]

[FONT=&quot]I have been careful to state repeatedly that I have been talking about PCM recordings. This is because there are other digital recording and playback schemes that DO have steps in them and require different techniques to correct. [/FONT]

[FONT=&quot]For example SACD does not use a DAC. It detects the difference between samples only (delta-sigma processing). Therefore, there are discrete steps in the wave form. [/FONT]

[FONT=&quot]To get adequate smoothing, extremely high sampling rates and storage of massive amounts of information are required. So SACD samples in the MHz region. It then must eliminate the tiny steps that remain (which is noise) by using noise shaping to move the noise into the supersonic region up around 50 KHz. [/FONT]

[FONT=&quot]This system works, but it clearly is inherently inferior to PCM recording. The only advantage of SACD is that no DAC is required. But this is a moot point since SACD has now been abandoned by the industry.[/FONT]

[FONT=&quot]I might also add that to enjoy any true benefits (if any exist) from the SACD medium, the musicians had to be recorded with SACD from the start and all processing must have been done in the SACD domain. But most SACD releases simply copied PCM masters onto SACD for distribution to customers. So even if SACD were perfect, it could do nothing more than present a PCM recording to its listeners. [/FONT]

[FONT=&quot]In this regard, the industry was deceiving its customers. If you pay for an SACD recording, it must be SACD at every step of the recording chain.[/FONT]

[FONT=&quot]Digital steps are also a problem for Class D amplifiers. They too sample at very high frequencies to minimize the size of the digital steps. Their wave forms must be smoothed using a Zobel network.[/FONT]

[FONT=&quot]For the Zobel network to work well, it must be precisely tailored to the load (the speaker) that the amplifier "sees." If the two are not perfectly matched, the frequency response of the amp/speaker combination will not be linear.[/FONT]

[FONT=&quot]This is a huge problem for manufacturers of Class D amplifiers because usually they cannot know the load to which the amplifier will be attached. So they produce "universal" Zobels. These may work well or not with a specific speaker system -- you just don't know until you try it and measure it. [/FONT]

[FONT=&quot]Because Class D amplifiers will not produce linear frequency response with most speakers, I don't consider them to be high fidelity devices. But they do work very well in selected applications. [/FONT]
[FONT=&quot]
For example, powered sub woofers are ideal for Class D amps because the load is known, high frequency response is not required, but high power and cool operation are. So Class D amps are an excellent choice for manufacturers to include in their sub woofers.[/FONT]

[FONT=&quot]Now turning back to digital recording, the "word length" of a digital PCM sample is the number of bits in it. So what do the bits do?[/FONT]

[FONT=&quot]They define the dynamic range and S/N of the recording. In general, you can consider a bit to be 6 dB of S/N. [/FONT]

[FONT=&quot]The CD Red Book specifies 16 bits. Therefore, the S/N of a CD can be as high as 96 dB. [/FONT]

[FONT=&quot]There are some subtle technicalities that I won't get into that alter this slightly. For example the addition of dither (very quiet white noise) will reduce the S/N slightly. I measure an actual S/N on most CD equipment of around 92 dB for these reasons. But using 6 dB per bit is a good rule of thumb.[/FONT]

[FONT=&quot]Today's "hi resolution" [sic] recordings usually are made using the 24/96 (24 bit, 96 KHz sampling), linear PCM specification. This means that the highs will extend to 40 KHz and the theoretical S/N will be 144 dB.[/FONT]

[FONT=&quot]We can't hear above 20 KHz, so doubling the frequency response to 40 KHz serves no useful purpose. And while the digital S/N may be as high as 144 dB, no analog electronics are anywhere near that quiet. [/FONT]

[FONT=&quot]The quietest analog electronics have a S/N at best of 120 dB, and Browning motion of the air molecules around microphone membranes limits them to about 92 dB. So there is nothing to be gained by using 24 bits during playback. [/FONT]

[FONT=&quot]The industry recognizes these facts and that is why the CD remains the highest quality music storage medium available. No human can hear any difference between a properly make Red Book recording and the source. [/FONT]

[FONT=&quot]Many audiophiles doubt this. But I have a standing bet of $10,000 (or any amount of money you are willing to bet) that nobody can hear the difference on a properly controlled test. I've never lost this bet. Contact me anytime for details and arrangements to place your bet and do the test.[/FONT]

[FONT=&quot]This finally brings us to answer your question. If digital recording is so good, why do many old LPs sound much more enjoyable and realistic than their CD counterparts?[/FONT]

[FONT=&quot]To answer this, let me tell you a story about the best recording of Respighi's "Pines of Rome" that I have ever heard. It was recorded in 1959 by Fritz Reiner and the Chicago Symphony by RCA "Red Seal." [/FONT]

[FONT=&quot]The LP had superb dynamics, essentially full frequency range (30 Hz to 15 KHz), great "hall sound", and a high degree of realism. It was pure joy and very exciting to listen to, despite all the obvious faults that were epidemic in LPs of that era (surface noise, distortion, wow and flutter, poor S/N, and general instability). [/FONT]

[FONT=&quot]When CD's became available years later, I couldn't wait for RCA to re-release that recording on CD so that I could eliminate all the faults heard on the LP. RCA finally did so in the late 80's and I couldn't wait to bring the CD home and play it.[/FONT]

[FONT=&quot]Boy, was I disappointed. The CD had essentially no dynamic range, no bass, many of the instruments could barely be heard, and it in general sounded like I was listening through a telephone! [/FONT]

[FONT=&quot]I was furious. I knew that digital recordings could (and should) be superb, since I was making them myself and knew this to be true. So I was determined to find out what was going on at RCA to ruin this recording.[/FONT]

[FONT=&quot]After enduring considerable hassles finding my way through the telephone maze at RCA, I finally got to those responsible for releasing the recording. After hearing my complaint, they explained what had happened this way:[/FONT]

[FONT=&quot]The original master tape recording was NOT made in 2-channel stereo. It was made using a 16 track recorder and multiple microphones -- in stereo -- on each orchestral section (violins had 2 mics, trumpets had 2 mics, etc.) They also placed mics out in the concert hall to record the sound of the hall sound.[/FONT]

[FONT=&quot]They then mixed down the 16 track tape to get a 2-channel stereo recording that could be pressed to produce LPs. The recording engineer who did this work obviously really knew his stuff and did a great job of getting the right balance between the various orchestra sections, blending in the hall sound, and maintaining nearly full dynamic range and frequency response (particularly in the bass). [/FONT]

[FONT=&quot]Although this was a mixdown, he kept it reasonably simple, and did not use compression, equalization, or artificial reverb. The performance was superb and his mix showed it off extremely well.[/FONT]

[FONT=&quot]Twenty five years later, when RCA wanted to re-release the performance on CD, they did not have the mixdown used for the LP. So they had a different engineer do another mix of the original 16 channel tape for the CD. He totally butchered the job. [/FONT]

[FONT=&quot]No matter how good the recording medium, if you put garbage in, you get garbage out. So the awful sound on the CD version of this recording was due to an horrible mix done by an incompetent sound engineer who had probably never been to a live, symphony orchestra concert.[/FONT]

[FONT=&quot]The answer to your question should now be clear. It is usually the later re-processing of an old master tape that is responsible for the poor quality of sound you hear from a CD compared to the LP. It is like comparing apples and bicycles, the recordings simply are not at all the same.[/FONT]

[FONT=&quot]Obviously this problem is not the fault of the digital recording medium, which is actually far better than any analog recording process. You should not assume that the digital medium is the cause of the problem as it clearly is not. [/FONT]

[FONT=&quot]In short, LP recordings are often far better than their CD counterparts because they were mixed in a more natural and realistic manner than what happens in a modern recording or mix. So the LP is much more enjoyable than the CD in spite of all the serious flaws and inaccuracies inherent in the LP medium. [/FONT]

[FONT=&quot]Of course, not all CD recordings are inferior to LPs. A good example is Willie Nelson's album "Stardust." It is available on both formats and was apparently recorded using the same mixdown tapes. The LP is a modern pressing with excellent quality vinyl. As a result, the CD sounds a bit better than the LP because it has none of the technical flaws that are obvious in the LP. But it is obvious that the two are more similar than different in that the actual recording is identical on both. Try them and see for yourself.[/FONT]

[FONT=&quot]This experience should drive home the fact that audiophiles need to be very cautious when making cause/effect judgments about what is heard. Audiophiles far too often make assumptions and assign fault to components or design features without actually knowing that these are the cause of what they hear. [/FONT]

[FONT=&quot]This brings up another major topic -- subjective listening techniques. But this opus is already far too long, so that discussion will have to wait for another day.

-Roger
[/FONT]
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#13
I want to make it clear that my previous post was intended to address the question of why CDs and LPs sound differently. I was not criticizing LPs or those who enjoy them. I have no doubt that you have some great sounding recordings that you like better on LP than on CD.

I note with interest that you state that you also like the CD versions of some music better than what you hear on the LP. You also state that you have heard some great sound from CDs. This confirms my point in my previous post that digital recordings do not sound inherently awful and are capable of reproducing great sound.

The main point of my previous essay was to debunk the myth that because some CDs sound badly, that the digital recording medium is to blame. This simply isn't true and those audiophiles who hold that belief have drawn a false cause/effect relationship.

I went to great pains to point out the fact that a properly done digital recording is so good that it is indistinguishable from the source. Therefore one cannot blame the digital storage medium for the poor sound that is present on some CDs. We must look elsewhere for the cause of the poor sound.

I carefully explained the technical issues involved with both formats and showed that in many (but not all) cases, the difference in sound between an LP and a CD of the same performance is that the mixdown was different. As a result, the recordings were not the same on the CD and the LP and that was the main cause of the differences that were heard.

You are correct that I simplified the situation and didn't cover all the possible reasons for the differences that you will hear between an LP and a CD. My opus was too long. So I will take this opportunity to elaborate on other differences that may contribute to the differences you hear.

Throughout the following discussion, I will be referring to "differences" in sound. This is an objective way of explaining things without making judgments on the quality of the sound. Obviously, if one recording sounds better (or worse) than another, it must by definition sound different.

Once it is determined that a difference is present, then we can then explore the causes for what we hear and make judgments on the quality. But let's keep it simple during this discussion by just listening for differences in sound.

I will also take the time to explain a lot of technical issues regarding LPs because the technology is over a half century old and therefore most audiophiles are not familiar with the issues involved. I apologize in advance if some of this will be review for some of the older readers on this forum.

For purposes of this discussion, I will assume that the recording mixdown (if one is used) will be identical. Obviously, if different mixdowns are used, then that is likely to be the major cause of any difference you will hear between formats. So we must assume identical mixdowns to examine the other reasons for the differences you hear.

The major cause of the differences you hear are frequency response differences. So let's closely examine what can and will alter the frequency response of a recording in both the LP and CD formats.

The industry standard "Red Book" CD specification states that the frequency response of a CD will be from 20 Hz to 20 KHz +/- 0.1 dB. In other words, it will have perfectly linear frequency response across the entire audio bandwidth. Because of the precision and computational nature of digital recording, all properly operating digital equipment easily meets this specification and you may safely assume that the frequency response will be flawless.

By comparison, the "RIAA" (Recording Industry Association of America) LP equalization standard operates in a limited frequency range of 30 Hz in the bass and 15 KHz in the highs. The RIAA specification specifies that the recording and playback will be extensively equalized to compensate for inherent limitations in the LP format.

As an aside, there is a similar equalization process used for analog tape. It is specified by the NAB (National Association of Broadcasters).

Specifically, during the LP disk mastering process, the highs must be significantly boosted and the bass must be heavily attenuated. Upon playback, a mirror image equalization must be applied to depress the highs and boost the bass to re-establish the same frequency response that was present on the master tape.

The purpose of boosting the highs is to reduce noise. Exaggerating the highs on the disk increases the S/N (Signal to Noise ratio) by making the high frequencies of the music substantially louder than the background noise of a stylus dragging on vinyl. When the highs are then depressed back to their original level on playback, the high noise level on the LP will be depressed by a similar amount, thus improving the S/N.

Bass frequencies contain a great deal of energy (imagine the initial impact of a bass drum). This would result in far too much excursion of the stylus on playback and cause it to jump out of the groove. Also, even if the stylus could handle the large excursion, the groove would have to cover a relatively wide area of vinyl to capture this large bass excursion, and this would greatly reduce the amount of playing time of the LP. Since the term "LP" is an abbreviation for "Long Play", it was essential to compress the bass excursion so the that the disk could play for a reasonably long period of time.

Regrettably, attenuating the bass on the LP and then boosting it on playback tremendously increases the low frequency noise. The slightest low frequency motion will be exaggerated and the result will be the well-known problem of "rumble."

I have only supplied a brief description of the RIAA equalization issue. For more complete details, I would refer you to www.wikipedia.org where you can ask for "RIAA Equalization" and get a complete history.

The RIAA equalization specification is difficult to meet with precision because it is spread over three components, all of which must to be quite accurate for the result to have accurately linear frequency response -- and none of them are. The first of these is the preamp that drives the lathe's cutter amplifier, the second is the phono preamp, and the third is the phono cartridge.

Since transistors hadn't been invented yet, both of the preamps had tube electronics, and tubes change their behavior as they age. So the equalization would drift over time.

Furthermore, since digital wasn't yet invented, the equalization had to be done in the analog domain using capacitors and resistors instead of by using computer computations such as is done with digital signal processing. It was therefore critical that the capacitors and resistors had exactly the correct values.

Precision, 1%, metal film resistors were not yet invented, so manufacturers were stuck using carbon composition resistors with a tolerance of around 5%. Even worse were the capacitors. Close tolerance capacitors are hard to produce even today. Back then, most capacitors had a tolerance of 20%.

Furthermore capacitors tend to change their tolerance with changes in temperature. They also change as they age over years. The heat generated in tube equipment caused a significant amount of drift.

So even if the manufacturer of the preamp was very skilled and his equalization components were carefully chosen, the relatively poor tolerances of the parts and the fact that they changed their values with age guaranteed that a significant amount of frequency response error would be present. The problem was worsened by the fact that several components were involved and each had errors that could add up to major flaws in the overall frequency response of the playback chain.

Modern transistor preamplifiers using precision 1% resistors and 2% precision capacitors will have much more accurate equalization than older equipment. But they still aren't accurate like digital signal processing. And many audiophiles prefer to use older tube preamplifiers so do not take advantage of the improved equalization accuracy that is available from modern equipment.

Of course, old LPs were made before transistors and precision components were available, so unless you only listen to recent LPs, the LP itself is likely to have relatively poor RIAA equalization accuracy.

I think you can now better appreciate that the RIAA equalization/mirror imaging process was fraught with error. So the frequency response in the electronics used to produce LPs was far from linear.

But that is only part of the frequency response problem. The phono cartridge is a major player here.
Phono cartridges (like speakers) are transducers and like speakers they all sound quite different. There are big differences in the frequency response of cartridges from different manufacturers. Even apparently identical cartridges from the same manufacturer will not have identical frequency response.

These frequency response differences are caused by variations in the mass of the stylus, with the compliance and age of their suspension system, and with the distances between their coils and magnets.

Yet another problem is that cartridges have a high frequency resonance. This resonance is affected by the resistive and capacitive loading on the cartridge. To make matters worse, it changes as the suspension system of the stylus ages (you should change styli every year for this reason).

To deal with this, quality cartridge manufacturers will specify the ideal input resistance and capacitance of the phono preamp to achieve reasonably linear frequency response from their cartridges. Top of the line phono preamps have facilities for changing their input resistance and capacitance to best meet the cartridge manufacturer's recommendations. But even they will have limited choices and you will not likely get these values perfect. Of course, if your phono preamp doesn't have adjustable input resistance and capacitance, then you can safely assume that your cartridge will have significant errors in its frequency response.

Yet another issue affecting frequency response is the age and wear on an LP's vinyl. Due to the tiny surface area of the stylus that is in contact with the vinyl, the pressure applied per unit area on the vinyl is immense. This produces a great deal of friction as the stylus slides along the groove. The friction causes tremendous local heating of the vinyl.

The heat causes the vinyl to soften and deform. The stylus, although relatively low in mass, has a great deal of inertia at high frequencies. The hot, soft vinyl is therefore stretched and deformed by the stylus as it passes over it. This distorts the groove and when it cools, it looses some of its shape. The result is that the high frequencies on the disk are gradually reduced in amplitude as the disk is played repeatedly. So an older, well-used disk will have lost much of its high frequencies.

Yet another factor that affects the apparent (not measured) frequency response is noise. The hiss that is present on all analog recordings gives the impression of extended high frequencies and "air" around the instruments. This is completely unnatural and inaccurate. But it may produce an effect on the music that subjectively is pleasing, even though the noise itself is not.

As you can see, there are several points where frequency response errors can and will occur in an LP playback system. Therefore, the total frequency response error can be quite high when they are all added together. I have seen over 6 dB of error in some systems, and even the best will show at least a couple of dB of error. Two dB is quite obvious to any skilled listener, and 6 dB will seriously change the sound of the music.

You can now easily understand why an LP playback system simply will not have linear frequency response. The error can vary from minor to dramatic depending on many factors.

A CD has none of these problems and will have perfectly linear frequency response that will not deteriorate over time and multiple playings. The bottom line here is that there will be significant differences in the frequency response of an LP compared to a CD -- even if the mixdown used on both recordings is identical. So you will hear a difference in the sound of the between them.

Our ears are very sensitive to frequency response differences, so you will clearly hear a difference between the two formats due to this factor. Which one you prefer is a matter of personal preference. But you can bet your first born child that the frequency response reproduced by the CD is the more accurate of the two. The inaccurate frequency response of the LP is correctly classed as "euphonic coloration."

Let me put all this in perspective and summarize the facts as follows:

1) Analog recording, particularly when done on LP, is inaccurate and has many flaws. Anybody can easily hear differences between an analog recording and the source.

2) Digital PCM recording is subjectively flawless. No human can hear any difference between a properly produced digital PCM recording and the source.

3) There is some outstanding music and recordings available on both formats. However, modern recordings typically are more heavily processed than older ones and often sound awful. Since most modern recordings are only available on CD, many audiophiles have come to believe that the it is the digital CD medium that sounds bad when in fact, it is superb and it is the recordings that are responsible for the lousy sound quality. There is much really excellent music that is only available on LP because of the simpler and more natural way recordings were made scores of years ago.

4) Wonderful music on an LP sounds great IN SPITE of the flaws of the LP medium -- not because of them.

5) Music on CD may sound awful IN SPITE of the perfection with which digital media stores music. Garbage in gets you garbage out, no matter how good the recording medium. If music sounds badly from a CD, it is not because CDs "sound bad" (they don't), but because the music recorded on them was of poor quality.

6) The differences you hear between an LP and a CD of the "same" recording is usually because the recording is not the same. Probably a different mixdown was used on each. Because you are comparing apples and bicycles, no valid comparison of the two formats can be drawn. You can usually safely assume different mixdowns were used if there is a dramatic difference between the sound of the LP and CD.

7) If the differences in the sound between an LP and CD are minor, then probably the same mixdown was used and you are hearing the frequency response differences between the two caused by the non-linear frequency response inherent in the LP recording/playback process. The CD will have linear frequency response while the LP will not. The frequency response you like best is a matter of personal preference.

In closing, let me make a recommendation that you might find wonderful -- I certainly did. Record your LPs to a quality digital medium so that you can easily listen to them and you won't ruin your precious LPs by playing them over and over.

Because digital PCM recordings are subjectively identical to the source, you will not be able to hear any difference between the music on the LP and the digital recording you made of it. So you will not suffer any loss of musical enjoyment by making and listening to your own digital recordings.

There are other reasons for making digital copies. They include the fact that most albums (LP or CD) usually have only a couple of tunes on them that you really like. So by making digital copies, you can make compilations of the best music of a particular genre off many LPs to which you can listen and enjoy song after song that you really like. And of course, digital media gives you the ability to instantly select various selections, which is far more convenient than trying to queue up LPs.

Additionally a great deal of space can be saved by using digital compilations because the media is so tiny. I have taken the music from thousands of LPs and stored them on just a few flash cards or digital tapes that take up just a few inches of shelf space.

It's really great to be able to come home after work and start listening to my "LPs" on digital format. I can then just sit down, relax, and enjoy the music without having to get up and down constantly while I pick and choose tunes from LPs that must be found, cleaned, played, cleaned again, and put away.

What digital format should you use? The most common is CD-R. But I don't like it or use it. CDs simply don't hold enough music to be useful.

A much better alternative is hard disk recorder or server. Even your computer can make flawless digital copies (use WAV format). But a computer tends to be rather bulky and is not a convenient "component" to put in your audio rack.

Nowadays, the best format is a flash recorder. These are simply awesome as they record on solid state memory flash cards (the same cards used in digital cameras).

Unlike a hard disk recorder, server, or computer, a flash recorder has no moving parts. It therefore is silent and has nothing to wear out. You can use any size flash card and as many as you wish, so storage is unlimited. A large flash card can record hundreds of hours of music. For easy identification of music, you can have a different flash card for each genre of music.

Unlike CD-R, you can copy and erase music from a flash card at will. So you can keep expanding the music of a particular genre on that flash card all you wish as you find more music that you like.

Because a flash recorder is a small computer, the music files on the flash card can be moved, edited, erased, and renamed as you would any other computer file. The music files can also be copied to your computer where you can work with them as you wish (I often make CD-R copies of music from my flash cards on my computer for customers who don't have flash recorders).

Finally, because flash recorders are totally electronic and have no mechanical parts, they are inexpensive. You can get many for less than $1,000 retail. An excellent example is the Tascam SS-R1 that sells for around $500. Marantz also makes many models.

Flash recorders are used by the professional music industry, which is why you probably haven't heard of them. No "audiophile" company makes them. So look for them in the "Pro" market. I sure wish they were available back in the "good old days" when I was hauling 80 pound, analog, tubed, analog open-reel recorders up several flights of stairs to do live recordings in concert halls!

So focus on the music. Enjoy whichever format contains the music you like best.

-Roger
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#14
Another great post.

But could you explain how did you prove that "No human can hear ... " and what you mean exactly by a "properly produced digital PCM recording"? CD, 24bit 192 kHz? Apologies if you explained it in another topic and I missed it. But stated exactly as written without justification it seems to distort the line you were so well presenting.

Disclaimer 2 - I am a subscriber of the Tape Project
Disclaimer 3 - I read all TELDC and even built a small panel in the 90's! Great fun.
[FONT=&quot]You ask two excellent questions. You first asked exactly what I mean by a "properly produced" digital PCM recording and wanted more details. The main issue with making digital (or analog) recordings from analog sources is the recording level. You must optimize the recording level so that the S/N (Signal to Noise ratio) is maximized.[/FONT]

[FONT=&quot]The "signal" is the music in the form of an electrical wave form. You want the signal to be much louder than the noise so that you have a silent background. [/FONT]

[FONT=&quot]You can only make the signal so loud before you reach the limit of the medium to record it without distortion. This upper limit is called "saturation." It is similar to "clipping" an amplifier.[/FONT]

[FONT=&quot]When you reach saturation on magnetic, analog tape, you get a "soft", rather gradual increase in distortion. It is possible to record a little above the saturation level before you start to hear obvious distortion. In other words, analog is somewhat forgiving when you reach saturation and the maximum recording level.[/FONT]

[FONT=&quot]But if you reach saturation in a digital system, you immediately get massive amounts of "hard" distortion. You are actually running out of bits to record on and the result is a harsh, crackling sound that ruins the recording. So when recording on digital media, you must be extremely careful to never exceed the maximum recording level.[/FONT]

[FONT=&quot]All recording media have an inherent noise floor. The noise floor is much lower in a digital system than in an analog one. But if you record at too low a level in either format, the music will be buried in the noise. Imagine what it would sound like when the noise (mainly hiss) is as loud as the average music level. This would be totally unacceptable.[/FONT]

[FONT=&quot]The S/N is the maximum recording level compared to the noise floor. In a top quality, magnetic tape, analog system, the S/N is about 72 dB. In an LP, it is about 40 dB. In a 16 bit, PCM digital system it is about 92 dB. In a 24 bit digital system, it is around 140 dB, although the analog circuits that are required on either side of the digital converters will limit that to around 120 dB at best.[/FONT]

[FONT=&quot]The dynamic range of a symphony orchestra is approximately 80 dB. Since you can't cram 80 dB inside the 72 dB limit of an analog recorder, you can see that it is impossible to have a truly quiet background on an analog recording. [/FONT]

[FONT=&quot]By comparison, the 80 dB dynamic range of a symphony orchestra will easily fit inside the 92 dB S/N of a 16 bit, digital PCM recording, which will the place the noise 12 dB below the quietest sound of a symphony orchestra. This is quiet enough subjectively to have a silent background. [/FONT]

[FONT=&quot]But 92 dB is just barely enough. You have to optimize the recording level to get the dynamic range of the music accurately centered in the S/N "window" to have a silent background while not running into hard distortion on musical peaks.[/FONT]

[FONT=&quot]Because the margin of error is quite small in a 16 bit system, most modern digital mastering recorders use 20 or 24 bits. This gives the operator a lot more leeway with the recording levels. It also makes post-mastering processing easier and less critical. Once all the mixdowns are made, the final product can then be accurately placed on a 16 bit CD without problems.[/FONT]

[FONT=&quot]It is also worth noting that virtually all commercial recordings have had their dynamic range compressed. That means that they have been modified so that the quiet sections are made louder and the loud sections are made quieter. Few commercial recordings today have a dynamic range of even 30 dB. This is one of the reasons that modern recordings lack realistic sound.[/FONT]

[FONT=&quot]The main reason that compression is used is to make recordings more listenable in the relatively noisy environment of an automobile, where most music is listened to today. If the full dynamic range was used, quiet passages of music could not be heard over the noise and loud passages would make it impossible to have a conversation with a passenger or (god forbid) on your cell phone. [/FONT]

[FONT=&quot]This compression dramatically reduces the S/N demands on the medium. This makes it possible to use fewer bits and still have an acceptable S/N. This is one of the techniques used by the MP3 format, which is designed to greatly reduce the amount of data that must be handled. [/FONT]

[FONT=&quot]In short, if your digital recording only involves making copies of music on albums, the S/N will be non-critical and you can easily get the correct recording levels. But if you are recording live musicians, you will have to be extremely careful to get the loudest sections of the music as close to saturation as you can -- without exceeding it.[/FONT]

[FONT=&quot]Recording levels only apply when you are copying analog information to a digital format. If you are coping a digital source digitally, there are no recording levels involved. The copying process is only transferring numbers in the form of bits. This is an "all or none" process, so there is nothing you can or need to do about recording levels. [/FONT]

[FONT=&quot]Now let's turn to a discussion of which type of digital medium you should use. You currently have several options. These fall into two broad categories, which are PCM and MP3. There is also SACD, but since it has been abandoned by the industry and there are no consumer-grade SACD recorders available, I will not discuss it here.[/FONT]

[FONT=&quot]Linear PCM (Pulse Code Modulation) is what is used on a CD. This is currently the highest quality type of digital recording available. [/FONT]

[FONT=&quot]PCM is available in a variety of formats, which are defined by their number of bits and sampling rates. The lower the number of bits and the lower the sampling rate, the less data must be stored. So you will get longer playing times with fewer bits and lower sampling rates.[/FONT]

[FONT=&quot]As I discussed in a previous post, the number of bits defines the S/N (a bit is worth approximately 6 dB of S/N), and the sampling rate defines the highest frequency that can be recorded. The sampling rate must be twice the highest frequency, plus a little more to handle the required anti-aliasing filter.[/FONT]

[FONT=&quot]The most common PCM format is the CD which uses a linear 16/44.1 (16 bit/44.1 KHz sampling) format. It will produce linear frequency response from 20 Hz to 20 KHz with a S/N of 92 dB, absolutely linear frequency response, and virtually unmeasurable distortion -- which is good enough to produce subjectively perfect recordings. [/FONT]

[FONT=&quot]For less critical applications such as for recording voice, FM radio, and yes, LPs, you can use the non-linear, 12/32 PCM specification. This will record up to 15 KHz (which is the limit of FM and LP high frequency response) and can produce a S/N of 70 dB. Since FM and LPs have a S/N of around 40 dB, a 70 dB recording system will be sufficient to not add any noise to the recording. [/FONT]

[FONT=&quot]Therefore 12/32 PCM is good enough that most listeners cannot tell any difference between it and the source when copying commercial source material. But you can hear a difference when recording live music with it. [/FONT]

[FONT=&quot]Actually the S/N of 12/32 is subjectively improved by using noise shaping to move the critical midrange noise frequencies up into the supersonic region. This reduces the apparent noise that a listener hears so that the S/N effectively becomes around 80 dB. [/FONT]

[FONT=&quot]I use the word "subjectively" because test instruments will measure the total noise without regard to its frequency. So instruments will not be tricked by noise shaping and will accurately report the noise that is present. But noise shaping remains a useful technique improving the human listening experience.[/FONT]

[FONT=&quot]Many professional PCM recorders also give you to option of using noise shaping for the higher sampling rates. For example, you can subjectively improve the standard S/N of 16 bit recording above its natural 92 dB limit using noise shaping that is available in some recorders.[/FONT]

[FONT=&quot]Yet another PCM format is 16/48. This increases the high frequency limit to 22 KHz from the CD's 20 KHz. "Super Bit Mapping" (noise shaping) is often used with this format to increase the S/N to around 100 dB. It is commonly used for live recording. [/FONT]

[FONT=&quot]Note that this format is incompatible with CDs. So you are better off using 16/44.1 so that you can copy your music digitally to CD-R if you wish. [/FONT]

[FONT=&quot]You can convert 48 KHz sampling to 44.1, but there is some slight loss of quality when you do. So I recommend that you use 44.1 KHz sampling for compatibility reasons.[/FONT]

[FONT=&quot]The next PCM format is 24/96. This increases the bits so that the digital S/N is 140 dB, which is way beyond what humans can hear or that analog can match. It is overkill to say the least.[/FONT]

[FONT=&quot]The 96 KHz sampling rate is also overkill as the high frequency limit is increased to 40 KHz, which is twice as high as any human can hear. Some audiophiles believe that frequencies above 20 KHz affect the sound they hear in the range of human hearing. There is no scientific evidence to support their beliefs, but even if they know something that science doesn't, the fact remains that no music microphone will record effectively above 20 KHz. [/FONT]

[FONT=&quot]By using 96 KHz sampling, you are wasting fully half the bandwidth of the system and recording nothing but supersonic noise. What is the purpose of that? There is simply nothing to be gained by using such a high sampling rate and recording at this rate requires the storage of much more data.[/FONT]

[FONT=&quot]Then there is the 24/192 PCM format. This increases the high frequency limit to 80 KHz. This is even more wasteful of data and just makes it possible to record even more supersonic noise. It makes no sense at all. [/FONT]

[FONT=&quot]So which PCM format should you use? The CD standard of 16/44.1 will make subjectively perfect recordings where you cannot hear any difference between the source and the recording. That is all that is necessary and will require the minimum amount of data storage in the PCM format. [/FONT]

[FONT=&quot]You can also use the 12/32 format for making copies of analog recordings since even its specifications are far better than analog. This will allow you to cut the amount of data storage in half and double your playing time. [/FONT]

[FONT=&quot]If data storage costs are a concern, then you can seriously consider the 12/32 PCM format. However, you can save far more data space while maintaining higher quality sound by using one of the MP3 formats, which I will discuss later in this article. [/FONT]

[FONT=&quot]The 24/96 PCM specification is becoming more popular. I think this is a marketing ploy rather than offering any real improvement in the sound quality. After all, once you have reached the point where you can't hear any difference between the source and the recording, you have perfection. How do you improve on perfection? You really can't. 24/96 requires vastly more data storage than 16/44.1 and although data storage is relatively cheap nowadays, it certainly isn't free. [/FONT]

[FONT=&quot]The cost of data storage and transmission times brings us to the most popular digital format -- MP3. MP3 uses complex computer algorithms to reduce the amount of data required. The algorithms are highly detailed and there is insufficient time to discuss them in detail in this post. But there are three general things that they do to reduce data storage requirements. [/FONT]

[FONT=&quot]First, they compress the dynamic range. Since commercial recordings are all compressed anyway, an MP3 can reduce its own dynamic range by 2 to 4 times without causing any affect on the recorded sound at all. [/FONT]

[FONT=&quot]The second technique used is that of masking. To explain this technique, imagine a symphony orchestra were all the instruments are playing as loudly as possible. [/FONT]

[FONT=&quot]Now eliminate one violin that is playing quietly amongst all this loud sound from the rest of the orchestra. Will you miss the violin? No you won't, simply because all the loud sound "masks" the sound of the violin so you can't hear it. [/FONT]

[FONT=&quot]MP3 uses this technique to avoid recording some of the low level material during loud passages. Masking also can be used effectively to "hide" background noise. So masking can improve the S/N.[/FONT]

[FONT=&quot]The third technique is reduction of high frequency response. By limiting the recording bandwidth to 15 KHz or less, a lot of data can be saved. Since most of us can't hear above 15 KHz, and analog sources are limited to 15 KHz, this generally does not degrade the subjective quality of the sound.[/FONT]

[FONT=&quot]These and other techniques allow MP3 to record relatively high quality sound while dramatically reducing the data storage requirements. The use of MP3 is why iPods can store thousands of songs and why you can listen to music over the internet. [/FONT]

[FONT=&quot]There are many levels of MP3 recording as defined by its sampling rate. MP3's sampling rate does not define its high frequency response as it does in PCM. It sampling rate defines the number of samples that are processed per second and greatly affects the sound quality. Basically this boils down to the fact that higher sampling improves the sound at the expense of higher data storage requirements.[/FONT]

[FONT=&quot]MP3 sampling rates of 64 KHz and below compromise the quality of the sound. You usually can hear a difference between the source and the recording. The type of music and its inherent amount of compression has a big influence on whether you can hear the difference. But I do not consider 64 KHz MP3 recordings as being "high fidelity."[/FONT]

[FONT=&quot]However, when you get to 128 KHz and higher sampling, it becomes extremely difficult to hear any difference between the source and the recording. In my tests with groups of "golden ear" audiophiles, most could not detect when the source and recording were playing when using 128 KHz MP3.[/FONT]

[FONT=&quot]This is highly dependent on the source material. Typical "pop" recordings with lots of percussive sounds sounded perfect. The most critical material was quiet orchestral works. A sustained piano note was the most taxing test and most listeners could hear a slight difference at 128 KHz sampling.[/FONT]

[FONT=&quot]Some of the better on-line music sources like pandora.com give you the option of selecting MP3 at 192 KHz. This sounds just as good as CD and no listener could hear any difference between it and the source on my tests.[/FONT]

[FONT=&quot]So what format should you use? For audiophile use, I would suggest 16/44.1 PCM because this is the one used for CDs. You can record on any digital equipment (I suggest a flash recorder) that you can use for your own listening. But you can also easily make CD-Rs for sharing with friends or using in your car. The quality will be flawless, so there is really no need to go to 24/96.[/FONT]

[FONT=&quot]But MP3 has big advantages with data storage and future compatibility. You can expect to record about 10 times more material on a given amount of data storage than you can with CD's 16/44.1 PCM format. CDs are likely to be phased out over the next few years and virtually all music will be on-line and in MP3 format. [/FONT]

[FONT=&quot]The great thing about MP3 is that you can use your recordings on an iPod or similar MP3 player. This makes it even better than CD to use in your car or if you use such players during activities or while traveling on airplanes, etc. Since these are compromised listening environments with lots of background noise and relatively poor speakers, there is really nothing to be lost by using MP3 in such situations.[/FONT]

[FONT=&quot]So although I don't consider MP3 to be quiet as good as linear PCM, as long as the MP3 sampling rate is 128 KHz or better, you will not actually hear any degradation of the sound unless you specialize in quiet classical music. In that case, you can move up to 192 Khz sampling.[/FONT]

[FONT=&quot]Flash recorders and computers have built-in MP3 processors. So you can record in either linear PCM (usually limited to 16/44.1 and 16/48) or MP3 in the sampling rate you prefer (usually up to 256 KHz). You won't have this flexibility when using CD-R, although some hard disk recorders offer MP3 processing.[/FONT]

[FONT=&quot]In summary, when I say a "properly produced" digital recording, I am referring to getting the recording levels right using a suitable, high-quality, digital format. If the recording levels are too low, you will hear background noise. If they are too high, you will hear massive amounts of distortion. But if they are just right, you will hear nothing but the music. The recording will then be indistinguishable from the source and you will have a subjectively perfect recording.[/FONT]

[FONT=&quot]This means that you can record your LPs to a good digital medium and the playback from the digital medium will sound absolutely identical to the LP. That is why I say that no human can hear any difference between them. [/FONT]

[FONT=&quot]Your second request to explain how I can prove that "No human can hear . . . " gets to the heart of audio, which is subjective testing techniques. This is a highly contentious issue that will require laying a careful foundation, explaining valid testing techniques in great detail, and providing logical proof at every step. Therefore, discussing testing will be a long article that I will have to attack over the next few days. [/FONT]

[FONT=&quot]So stay in touch. I promise you a fascinating read soon . . .

-Roger

[/FONT]
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#15
Roger-Disgree or agrree you always present things in a common sense manner that translates into real world information the audiophile can act on. It appears you were discussing Stereo imaging-or left to right balance. Could you consider how we would go about measuring front to back imaging and the spatial relationships between musical istruments and voices? Often described as air around the instruments or voices.
gregadd
[FONT=&quot]Hi Greg,[/FONT]

[FONT=&quot]There is no way to directly measure front to back imaging information from speakers. That's because our brains process the information from the speakers into an image, and no machine can do that.[/FONT]

[FONT=&quot]However, we can get some pretty good clues as to how a speaker will image by carefully evaluating its phase behavior. This should be done both with impulse measurements and by carefully examining its design.[/FONT]

[FONT=&quot]Phasing are the time queues we hear in the sound. For example, at a concert we here that the violins are closer to us than the brass because their sound arrives at our ears sooner than that of the brass. A speaker needs to be able to duplicate this.[/FONT]

[FONT=&quot]To do so, a uniform and coherent wave front must be produced by the speaker that is physically vertical and where all the frequencies are launched by the drivers at the same time. Those frequencies then need to arrive at our ears without interference. This is nearly impossible for a speaker to do for many reasons.[/FONT]

[FONT=&quot]First, "cones and dome" speakers do not have planar surfaces. Just where is the wave formed on a cone or dome? Obviously, there is an average surface, but it is not well defined like it is on a truly planar speaker surface.[/FONT]

[FONT=&quot]Keep in mind that the wave length of a 10 KHz tone is about on inch. So just a half inch of distance results in a phase error of fully 180 degrees. So physical accuracy of the moving surface is quite important.[/FONT]

[FONT=&quot]Secondly, the drivers in many speakers are wired out of phase. This is deliberately done many manufacturers to compensate for the problem of the 3 dB peak at the crossover point that is inherent in 2nd order, Butterworth filters, which most manufacturers use. In other words, the phase is sacrificed to improve the frequency response. [/FONT]

[FONT=&quot]Fortunately, some of the better speaker manufacturers have recognized this and they use steeper slopes and Linquitz/Riley filters which do not have the linearity problems of 2nd order Butterworth. They can then wire their drivers in phase. Despite this, I often see inverted phase on some drivers in even very expensive speakers showing that the drivers are still out of phase. [/FONT]

[FONT=&quot]Thirdly, the drivers must be "time aligned." This means that the drivers must be positioned such that their wave fronts are aligned so that the sound from all of them arrives at your ears simultaneously. This can be quite difficult to do from a cosmetic standpoint, and it complicates construction a great deal. So usually only expensive speakers are time-aligned.[/FONT]

[FONT=&quot]Today, it is possible to time-align the drivers using digital delay. This can be used to delay the early arrival driver's information until the later arriving information gets to your listening location. Digital crossovers have this feature. This makes it possible to build a speaker with high-quality cosmetics and still achieve excellent time-alignment.[/FONT]

[FONT=&quot]Fourthly, multiple drivers can only be time-aligned for a specific vertical location in front of them. For example, if the speakers are designed with a woofer at the bottom and a tweeter at the top, they can only be in alignment at one point -- usually when you are sitting. If you are standing, you will be a greater distance from the woofer and closer to the tweeter and they will no longer be properly aligned. So it is best if the drivers are very close together.[/FONT]

[FONT=&quot]Fifthly, shallow crossover slopes cause a lot of overlap between drivers. Since the drivers cannot be perfectly aligned, there will be some phase errors at the crossover region where both drivers are trying to reproduce the same information. Very steep crossover slopes greatly reduce the overlap and therefore the phase anomalies. [/FONT]

[FONT=&quot]Finally there is the issue of dispersion. Speakers that disperse sound widely cause most of the sound to be heard off various surfaces in the room rather than directly from the speakers. The reflections off the room surfaces travel much longer distances than the direct sound from the speaker. [/FONT]

[FONT=&quot]To add insult to injury, there are hundreds or thousands of reflections from the room. These reflections are all different lengths, so the phase from them will be different. The result is that your ears will hear tremendously confused phasing from wide dispersion speakers -- even if the speakers themselves have perfect phasing, which is highly unlikely.[/FONT]

[FONT=&quot]As a result, most speakers have perfectly awful front to back imaging. Listeners often claim that the imaging from such and such a speaker is great, but that only because they have never heard a speaker with truly accurate phasing.[/FONT]

[FONT=&quot]I try to avoid hyping my speaker designs on the forum. But in this case, understanding my design choices will help explain the information I am trying to convey. So allow me to point out that I build my electrostatic speakers out of a perfectly flat, one-piece, planar diaphragm that reproduces all frequencies from 172 Hz up through 32 KHz over its entire surface without crossovers. Therefore it produces a perfectly true wave front across the entire audio bandwidth where phase matters (bass excluded).[/FONT]

[FONT=&quot]Even though the phasing is perfect from this type of speaker design, I go to great lengths to make sure that the phase information from the woofer is accurate as well -- even though bass phase is largely irrelevant to the imaging.[/FONT]

[FONT=&quot]To do so, I mount the woofer high and as up as close to the electrostatic panel as possible. It is then driven by a digital crossover with appropriate time delay to achieve exact time-alignment. [/FONT]

[FONT=&quot]You can even connect a microphone to the crossover and place it exactly at the height of your ears. It will then automatically send out test tones to the various drivers on both channels and it will measure the distance to each driver and the time it takes for its signal to reach the microphone. It will then calculate the correct delay to apply to the early-arriving driver (the ESL) so that sound from both the ESL and the woofer will arrive at your ears at precisely the same time.[/FONT]

[FONT=&quot]The crossover uses extremely steep, 8th order slopes (48 dB/octave) so that there is very little overlap between the drivers. Also, since the overlap and phase error occurs at a very low frequency (172 Hz), there no phase error produced in the critical midrange region.[/FONT]

[FONT=&quot]The speaker is highly directional so that room acoustics from the midrange up are virtually eliminated. Any delayed reflections that occur from the dipole radiation are so attenuated and delayed that they do not affect the phase information at the listening location.[/FONT]

[FONT=&quot]The result is that my speakers produce truly accurate phase and the result are amazing images with a 3-dimensionality, depth, and holographic quality that you simply must hear to believe. The image floats out in the room between the speakers without appearing to be associated with the speakers at all. There is no "hole in the middle", no matter how widely you space the speakers. The image is highly realistic and no conventional speaker comes close to this type of performance. [/FONT]

[FONT=&quot]In summary, there is no way to measure imaging. But if you closely study the phase behavior of the speaker and the way it interacts with the room, you can get a pretty good idea of the imaging you can expect from it. [/FONT]

[FONT=&quot]In closing, let me also point out that the image from the speaker can only be as good as that captured on the recording. Regrettably, most recordings being made today are not done in a fine acoustic environment using stereo microphone techniques. They are done with a single mic in a sterile, nearly anechoic environment, and then artificial reverb is added during the mixdown. Such a recording cannot reproduce realistic imaging, although they can convey a sense of spaciousness that is pleasing. [/FONT]

[FONT=&quot]As I've said before, garbage in gets you garbage out. So don't expect realism from modern recordings. But if you ever get the chance to come to Denver, plan on coming to my facilities to hear some of my stereo master tapes made with just two microphones of symphony orchestra and opera. You'll really feel like you are in the concert hall![/FONT]

[FONT=&quot]-Roger[/FONT]
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#16
Roger, your explanations are wonderful. They paint a very nice end-to-end picture of the topic at hand. So I hope you take this as a small criticism :). Your explanation at the extreme are somewhat inaccurate. For example, when we state a number for compressed audio, it is the data rate, not the sampling rate. The sampling rate of MP3 is 44.1 Khz at data rates >= 128Kbps and doesn't change. The number mentioned is the number of bits per second of audio. Uncompressed audio is 1.4 Mbit/sec. 128 Kbps, is 11 times lower data rate while maintaining the same sampling rate. Note that I said sampling rate, not frequency response. Standard MP3 codecs roll off the high frequencies above 16 Khz at all but the highest data rates in order to keep distortion under control so looking at the sampling rate alone can be misleading. Here is a quick test someone ran:http://www.lincomatic.com/mp3/mp3quality.html

CD:


192 Kbps MP3:


And even 320 Kbps:



MP3 was designed for backward compatibility with MPEG Layer 2 and to be implementable in really low-end hardware. As such, even putting aside the above filtering, it is not able to achieve transparency at any date rate -- your anecdotal observations not withstanding :). To be sure, at 384 Kbps it can sound very good but it still sounds different to good number of audiophiles. AAC or WMA Pro on the other hand, can achieve extremely good quality at those rates perhaps fooling substantial majority of listeners. Still some people though, can tell the difference from uncompressed.
[FONT=&quot]Hi Amir,[/FONT]

[FONT=&quot]Thank you for the additional information. I'm sure your readers appreciate it, as do I. [/FONT]

[FONT=&quot]As I pointed out in my essay, I made no attempt to do a comprehensive analysis of the MP3 format due to limits on my time and the amount of writing involved. After all, a thorough study of MP3 would require a small book. So your additional information is most appreciated.[/FONT]

[FONT=&quot]I just wanted to give readers a little taste of how the MP3 algorithm operates. I then focused on the only thing that users can do to control the quality of MP3's -- and that is the sampling rate. I wanted them to understand that it was essential to use high sampling rates to get high quality sound on MP3 recordings.[/FONT]

[FONT=&quot]My studies and tests of MP3 were not anecdotal. They were carefully designed, controlled, and implemented scientific studies using double-blind, ABX testing and panels of listeners using electrostatic speakers so that the most subtle of details could be heard. My tests were similar to, but much more rigorous than the blind tests done by your on-line source -- and the results were very similar. [/FONT]

[FONT=&quot]Specifically, at sampling rates above 128 KHz, it is really hard to hear any difference between MP3 and linear PCM. This means that for commercial recordings that typically are compressed, heavily processed, and contain a lot of transient material, nobody can hear any difference between them and a CD. Again, I am in agreement with your source that lower MP3 sampling rates introduce audible changes in the sound, so I didn't consider lower sampling rates to be "high fidelity" and I discouraged listeners from recording at those low rates.[/FONT]

[FONT=&quot]The object of my discussion was to give guidelines for those who might want to copy their precious LPs to a digital format for ease of use and protection of their LP collections. Since LPs have rather poor frequency response (lower than that of MP3), very limited S/N, and high noise (all worse than MP3); high-rate, MP3 is a suitable for recording them since MP3 is capable of capturing all the information they have to offer. Therefore MP3 is one option to consider, particularly if the user wants to listen to the music in cars or on mobile devices where MP3 is the standard -- or if he wants to share files over the web.[/FONT]

[FONT=&quot]I continue to prefer linear PCM when I want the best possible recording medium as it offers the best performance. But it certainly has serious disadvantages compared to MP3. These include requiring a lot more data storage, being awkward to use in portable devices, sharing music on CDs is more difficult than MP3 files, and CDs are rapidly becoming obsolete. [/FONT]

[FONT=&quot]Whether we like it or not (I don't), MP3 appears to be the music medium of choice in the future as virtually all music will be internet, mobile phone, and computer based. Bandwidth is a critical issue in all this and there simply will be no place for the rather wasteful use of data storage demanded by linear PCM recording. So we might as well get used to using MP3.[/FONT]

[FONT=&quot]I do most of my listening nowadays off the internet (www.pandora.com and others). All this music is only available in MP3 format, which can be obtained at 192 KHz sampling if you subscribe (lower rates if you don't). I am consistently impressed that I hear no problems in the quality of the sound with this format -- something I can't say for analog recordings. [/FONT]

[FONT=&quot]The good news in all this (such as it is), is that high sampling rate MP3 actually does an excellent job of reproducing music. It isn't quite as good as linear PCM, but it is a lot better than analog. So if we are going to be stuck with an inferior recording medium (relative to linear PCM), it is nice to know that it is good enough that few (if any) listeners can hear the difference between it and the source. And any perceived differences are subtle at worst. So we need not despair over this turn of events.[/FONT]

[FONT=&quot]In any case, it is nice for LP lovers to know that they have at least two good digital media upon which they can record their LPs. For absolutely flawless recording, they can use linear PCM at 16/44.1 or better. They can also use high sampling rate MP3 (at least 128 KHz and higher is better) that will produce essentially flawless copies of their LPs. The choice really comes down to how much money they are willing to spend on data storage and how much convenience they want. [/FONT]

[FONT=&quot]The best choice is to do both. I recorded my LP collection on linear PCM 16/48 and then made MP3 copies of some of it for use on portable devices. In hindsight, I made the mistake of using 16/48 -- I should have used 16/44.1 so that I didn't have to convert the sampling rate when I wanted to make CD copies. Lesson learned.[/FONT]

[FONT=&quot]It is always a good idea to have a back up copy. So for truly treasured LPs, more than one digital copy should be made. Since it is easy to copy the files once they are made in real time for the original LP, this is no problem. Note that one of the copies should be kept elsewhere so that if your home should burn, you won't loose everything. [/FONT]

[FONT=&quot]It would be great if you or others on the forum would take the time to provide more detailed technical information on MP3. It is a big topic about which very little is known by audiophiles. Since it appears to be the medium of the future, we should all learn all we can about it. Again, thanks for your input. [/FONT]

[FONT=&quot]-Roger[/FONT]
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#17
Hopefully, Amir will sign up for that one with his experience at Microsoft.

In the meantime, here's some of Roger's writing on testing:

TESTING PART ONE
(part 2 is next post)

Wouldn't you like to know for sure if that new, ten thousand dollar amplifier you want to buy is really better than your old one? Do different brands of tubes sound different from others? Do multi-thousand dollar interconnects really sound better than ordinary ones? Do high power solid state amps truly sound badly when playing quietly? Does negative feedback make an amp sound worse than one without feedback? Does the class of amplifier operation affect the sound? Do MOSFET amplifiers sound different from those using bipolar transistors? Do cables really sound better when connected in a particular direction?

These are just some of the questions audiophiles want answered. And they need to be answered with certainty before an audiophile drops tens of thousands of dollars on expensive audio equipment.


But there is something very strange about high end audio. Although it is a highly scientific, engineering exercise, most audiophiles base their purchase decisions almost totally on subjective listening tests, anecdotal information, and testimonials from self-proclaimed "experts" instead of from engineering measurements. Therefore, it is hard to know for a fact what components really have high sound quality.


Subjective listening tests can be useful and accurate. But if not done well, their results can be confusing and invalid. Worse yet, poor testing makes it possible for unsuspecting music lovers to be deceived and fail to get the performance they are seeking.


Unfortunately, there are many unscrupulous manufacturers and dealers who take advantage of this situation by making false claims based on "voodoo science" to sell inferior, grossly overpriced, or even worthless products to uninformed and confused audiophiles. I find it amazing that this state of affairs exists for such expensive products.


Audiophiles need to know -- and deserve to know the truth about the performance of the audio components they are considering. Only then can they make intelligent and informed decisions.


This requires accurate test information, which is not readily available. This information can be obtained by objective measurements by instruments and by valid listening tests. Unscrupulous businessmen in the audio industry have somehow managed to convince audiophiles that measurements cannot be trusted. So most audiophiles use listening tests to compare two similar items to evaluate which sounds better.


But most listening tests produce conflicting and unreliable results as proven by all the controversy one hears about the merits of various components. After all, quality testing will clearly and unquestionably reveal the superior product, so there should be no confusion or disputes about it.


For example, the ability of a digital camera to produce detailed images is intrinsically linked to the number of megapixels in its CCD. If one camera has more megapixels than the other, there simply is no question of which one can provide the most detailed image. Therefore, you don't find vidiophiles arguing over this.


The same is true of the technical aspects of audio equipment like frequency response, noise, and distortion. But because the results of most audiophile subjective testing is so variable and uncertain, different listeners come to different conclusions about what they hear. As a result, there is very little agreement about the quality of the performance of audio equipment.


Why is this so? We all hear in a similar way, so what is going on with subjective listening tests that is so confusing?


The purpose of this paper is to investigate testing and answer that question. Actually, the answer is simple, but requires great elaboration of the details to explain the problem and what must be done to correct it.


So let's eliminate the suspense and immediately answer the question of why the typical audiophile listening test produces vague and conflicting results. The answer is that most listening tests contain multiple, uncontrolled variables. Therefore, there is no way to know what is causing the differences in sound that are heard. Allow me to explain this in detail.


This issue of controlling the variables in a test lies at the heart of all testing. Audiophiles need to understand this and control the variables so that they can do accurate listening tests that produce reliable results.


What is a "variable?" A variable is any factor that can affect the result of a test.


An "uncontrolled" variable is the one variable in a test that is allowed to vary because we are trying to evaluate its effect. It is absolutely essential that any and all other variables in a test be "controlled" so that they do not influence the results.


If there is more than one uncontrolled variable in a test, then you will not be able to determine which variable caused the results you heard. Therefore, having multiple uncontrolled variables makes it impossible to draw any cause/effect conclusions from the test. Since most tests are trying to find cause/effect relationships, a test done with multiple, uncontrolled variables can't answer the question, so is simply worthless and invalid.


Let me make this very clear by giving an example of how a typical audiophile listening test is performed and then analyze it for uncontrolled variables. Let's assume that an audiophile is considering buying a new amplifier that costs $10,000 and wants to know if the new amplifier is really better than his current one and worth the large price that is being asked for it. His testing will go something like this:


He may listen to his old amp briefly before listening to the new one, or he may not even bother and just assume he can remember the sound of it from long experience. He will then turn off his system, unplug the cables from his old amp, put the new amp in place, hook up all the cables, turn everything back on, then listen to the new amp for awhile. He will then make a judgment regarding which amp sounded better.


He will usually go one step further and draw some sort of cause/effect relationship as to the CAUSE of why one amp sounded better. Typical examples of such cause/effect relationships might be that one amp had feedback while the other didn't, one was Class A while the other was Class D, one had tubes while the other had transistors, one had the latest boutique capacitors or resistors, while the other one didn't, etc. For the remainder of this article, I will refer to this type of test as "open loop" testing.


Now what would happen if I were to intervene in the above test and change the loudspeakers at the same time that the audiophile switched amplifiers? I think we would all agree that changing the loudspeakers would add another variable and that this make it impossible to determine the cause of the difference in sound that would be heard. We simply would have no way of knowing if the different sound that we heard was caused by the speakers or the amplifier (or both) because there are two uncontrolled variables in the test. Therefore, the test would be invalid and the results could not be used to determine which amplifier sounds better.


Now all rational audiophiles understand this concept of only having one uncontrolled variable in a test. They readily agree that you can only test one thing at a time. They made a sincere attempt to follow this process by only testing one component at a time in their listening tests.


But they unknowingly break the "one variable" rule in their listening tests. Let's look carefully at the above amplifier test and analyze it for uncontrolled variables.


When asked, the audiophile will honestly claim that his test only had one uncontrolled variable, which would be the amplifier. But he would be mistaken. His test actually had five uncontrolled variables. Any of them, or multiples of them could have caused the differences in sound he heard. He needs to control all the variables except for the amplifier under test. So what are the other uncontrolled variables?



1) LEVEL DIFFERENCES. If one amplifier played louder than the other, then it will sound better. Louder music sounds better. That is why we like to listen to our music loudly. Up to a certain limit, louder sounds better to us.


The gain and power of amplifiers varies. Therefore, for a specific volume control setting on the preamp used in the test, different amplifiers will play at slightly different loudness levels.


But the audiophile in the example probably didn't even attempt to set the preamp level at exactly the same level. He probably just turned up the level to where it sounded good. He made no attempt to match the levels at all because he was unaware that this was an uncontrolled variable.


Human hearing is extremely sensitive to loudness. Scientific tests show that we can hear and accurately detect very tiny differences in loudness (1/4 dB is possible). At the same time, we don't recognize obvious differences in the level of music until there are a couple of dB of difference. This is due to the transient and dynamic nature of music, which makes subtle level differences hard to recognize.


Therefore when music is just a little louder, we hear it as "better" rather than as "louder." It is essential that you recognize that two identical components can sound different simply by having one play a little louder than the other. The louder one will sound better to us.


This is a serious problem in listening tests. Consider the amplifier test above and for purposes of demonstration, let's assume that both amplifiers sound exactly the same, but that the new one will play a bit louder because it has slightly more gain. This means that the new amp will sound better than the old one in an open loop test even though the two actually sound identical.


Hearing this, the audiophile will then conclude that the new amp is better and will spend $10,000 to buy it. But in fact, the new amp doesn't intrinsically sound any better and it was the difference in loudness that caused the listener to perceive that it was better.


So the audiophile would have drawn a false conclusion about the amp. This erroneous conclusion cost him $10,000. I think you can see from this example that you absolutely, positively must not have more than one uncontrolled variable in your tests.



2) TIME DELAY. Humans can only remember SUBTLE differences in sound for about two seconds. Oh sure, you can tell the difference between your mother's and your father's voices after many years. But those differences aren't subtle.


Most audiophiles are seeking differences like "air", "clarity", "imaging", "dynamics", "sweetness", "musicality", etc. that are elusive and rather hard to hear and define. They are not obvious. We cannot remember them for more than a few seconds. To be able to really hear subtle differences accurately and reliably requires that you be able to switch between components immediately.


Equally important is that you should make many comparisons as this will greatly improve the reliability of your testing. This is particularly important when dealing with music as different types of music have a big influence on the sensitivity of what you can hear during your testing. You really need to test with many types of music using many comparisons.


Open loop testing only provides a single comparison, which is separated by a relatively long delay while components are changed. This makes it very difficult to determine with certainty if subtle differences in sound are truly present.




3) PSYCHOLOGICAL BIAS. Humans harbor biases. These prejudices influence what we hear. In other words, if you EXPECT one component to sound better than another -- it will.


It doesn't matter what causes your bias. The audiophile in the previous test had a bias towards the new amp, which is why he brought it home for testing. He expected it to sound better than his old amp.


That bias may have been because he expects tubes to sound better (or worse) than transistors, or that it had (or didn't have) feedback, or it was more expensive than his old amp, or that it looked better, or that he read a great review on it, or that is had a particular class of operation, etc. Bias is bias regardless of the cause and it will affect an audiophile's perception of performance. It must be eliminated from the test.


Don't think you are immune from the effects of bias. Even if you try hard to be fair and open-minded in a test, you simply can't will your biases away. You are human. You have biases. Accept that fact and design your testing to eliminate them.



4) CLIPPING. It doesn't matter what features an amplifier has -- if it is clipping, it is performing horribly and any potentially subtle improvements in sound due to a particular feature will be totally swamped by the massive distortion and general misbehavior of an amplifier when clipping. Therefore no test is valid if either amplifier is clipping.


If one amplifier in the above test was clipping, while the other wasn't, then of course the two will sound different from each other. But you are not supposed to be testing a clipping amp and comparing it to one that isn't clipping.


Most audiophiles simply don't recognize when their amps are clipping. This is because the clipping usually only occurs on musical peaks where it is very transient. Transient clipping is not recognized as clipping by most listeners because the average levels are relatively much longer than the peaks. Since the average levels aren't obviously distorted, the listeners think the amp is performing within its design parameters -- even when it is not.


Peak clipping really messes up the performance of the amplifier as its power supply voltages and circuits take several milliseconds to recover from clipping. During that time, the amp is operating far outside its design parameters, has massive distortion, and it will not sound good, even though it doesn't sound grossly distorted to the listener.


Instead of distortion, the listener will describe a clipping amp as sounding "dull" or "lifeless" (due to compressed dynamics), muddy (due to high transient distortion and compressed dynamics), "congested", "harsh", "strained", etc. In other words, the listener will recognize that the amp doesn't sound good, but he won't recognize the cause as clipping. Instead, he will likely assume that the differences in sound he hears is due to some minor feature like feedback, capacitors, type of tubes, bias level, class of operation, etc.) rather than simply lack of power.


But his opinion would be just that -- an assumption that is totally unsupported and unproven by any valid evidence. Most likely his guess would not be the actual cause of the problem. Because different audiophiles will make different assumptions about the causes of the differences they hear, it is easy to see why there is so much confusion and inaccuracy about the performance of components when open loop testing is used.


It is easy to show that most speaker systems require about 500 watts to play musical peaks cleanly. Most audiophiles use amps of around 100 watts. Therefore audiophiles are comparing clipping amps most of the time. This variable must be eliminated if you want to compare amplifiers operating as their designers intended.



5) The last uncontrolled variable is the amplifier. This is the one variable that we want to test. So we do not need to control it.



The above information should make it clear why open loop testing is fraught with error and confusion. It is easy to see why we can easily be tricked by open loop testing, particularly when there is a significant time delay which will allow our bias to strongly influence what we hear. All these uncontrolled variables simply make it impossible to draw valid conclusions from open loop testing, even though we may be doing our best and being totally sincere in our attempt to do quality testing.


But it doesn't have to be that way. It is possible to control all the variables so that subjective listening test results are accurate and useful. Here's how:



1) LEVEL DIFFERENCES are easily eliminated by matching the levels before starting the listening test. This is done by feeding a continuous sound (anything from a sine wave to white or pink noise) into the amps and measuring the output using an ordinary AC volt meter. The input level to the louder of the two amps will need to be attenuated until the levels of the amps are matched as closely as possible (must be matched to within 1/10 dB).


Need a signal generator? You can buy a dedicated one for around $50 on eBay. Or you can download one for free as software for your laptop at this link:

http://www.dr-jordan-design.de/signalgen.htm


You can also use an FM tuner to generate white noise. You can get it as interstation hiss by turning off the muting. An old analog tape deck will generate white noise by playing a blank tape at a high output level.



2) TIME DELAY must be eliminated by using a switch to compare the amps instantly and repeatedly. This is done by placing a double pole, double throw switch or relay in a box that will switch the amplifier outputs. Attenuators can also be placed on the box so you can adjust levels on amplifiers that have no input level controls. Of course, the box will have both input and output connectors for the amplifiers so that they can simply be plugged into the box for testing. For testing line-level components, you will use the same technique, but use the switch to control RCA or XLR connectors.


Need a test box? You can make one or borrow mine. You can reach me at
roger@sanderssoundsystems.com or by phone at 303 838 8130.


3) PSYCHOLOGICAL BIAS must be eliminated by doing the test "blind." This means that listeners must not know which component they are hearing during the test. This will force them to make judgments based solely on the sound they hear and prevent their biases from influencing the results.


Scientists are so concerned about biases that they do double-blind testing. This should be done during audio tests too.


Double blind audio testing means that the person plugging in the equipment (who will know which component is connected to which side of the test box) must not be involved in the listening tests. If he is present during the test, he may give clues to the listeners either deliberately or accidentally about which component is playing.


There is one more thing that must be done to assure that bias is eliminated. There must be an "X" condition during the tests. By this I mean that you can't simply do an "A-B" test where you switch back and forth between components. A straight "A-B" test will make it possible for listeners to cheat and claim they hear a difference each time the switch is thrown, even if there are no differences.


So you need to do an "ABX" test where when the switch is thrown, it sometimes continues to test the same component rather than switching to the other. Of course, if the component is not switched out, the sound will not change, so a listener cannot claim that there is a difference in sound every time the switch is thrown. Listeners are told prior to the test that this will be an ABX test so sometimes there will be no difference and they must be careful and be sure they really hear a difference.



4) CLIPPING can be eliminated by connecting an oscilloscope to the amps and monitoring it during the test. A 'scope is very fast and can accurately follow the musical peaks -- something your volt meter cannot do. Clipping is easy to see as the trace will appear to run into a brick wall. If clipping is seen during initial testing, the listening level must be turned down until it no longer occurs. Then you may proceed with the test.


Need an oscilloscope? You can easily find a good used one on eBay for around $100. If you use a spectrum analyzer (to be discussed shortly), it probably will have an oscilloscope function built into it, so you don't need to buy a separate one.



The variables above apply to amplifiers. There usually are different variables involved for different components. You have to use some thoughtful logic to determine what variables are present and design your test to control them.


For example, preamplifiers and CD players don't clip in normal operation. So you don't need to bother with a 'scope. Cables, interconnects, power conditioners, and power cords don't have any gain. So you don't need to do any level matching. Just use a switch and do the test blind.


Also, consider your comparison references. In the case of an amplifier, you can only compare it to another amplifier because power and gain are required. But when testing a preamp, you don't have to compare it to another preamp. You can compare it to the most perfect reference possible -- a straight, short, piece of wire.


This usually takes the form of a short interconnect, or you can go one better and use a very short piece of wire soldered across the terminals of the test box switch. You need then only set the preamp to unity gain to match the wire and do your testing blind.


There are many variations of the ABX test. A rigorous, scientifically valid ABX test will be done with a panel of listeners to eliminate any hearing faults that might be present with a single listener, and it will always be done double-blind.


But you can cut corners a little bit and still have a valid test. For example, you can do the test single-blind with one listener. What this means of course, is that you will do the listening test by yourself.


You must "blind" yourself. The best way to do this is to have someone else connect the cables to the equipment so you don't know which one is "A" and which one is "B." You can then set levels and proceed to listening tests.


When doing ABX testing with others, it is helpful to give them a little bit of training and let them practice. Tell them that they will only be asked if they can hear any DIFFERENCE between components. Obviously, if one component sounds "better" (or worse) than the other, it must also sound different.


You need not be concerned about making judgments on subjective quality factors initially. Just ask the listeners if they hear any differences of any type and if any exist, you can test that separately later.


Let them know that you will be including an "X" factor in testing where sometimes the switch will not actually change to the other component. Tell them this to eliminate guessing. Point out that the purpose of the test is not to trick them, but to assure accuracy.


Because many comparisons are made, I use a score sheet for listening groups. The sheet has a check box for "different" and "same" that they check after each comparison. I have a master sheet that shows where differences are possible (A-B test) and where they are not (A-X or B-X). I can then score their sheets quickly after the test. I find that listeners are very accurate and that there is usually complete agreement on what is heard.


When testing by yourself, you don't use a score sheet and therefore can only use A-B testing. This type of testing isn't well controlled, but you can usually get a good idea of what to expect. If you need really reliable results, you should back up your personal testing with others using a full ABX test to be certain of the results.


When training a new group of listeners, I deliberately make a small error in setup (usually a level difference of 1 dB on one channel) and have them start listening. The ABX test is extremely revealing and much more sensitive than open loop testing. So even with such a tiny difference, even unskilled listeners quickly become very good at detecting them. Once the listeners are confident in how the test operates, we move on to the actual testing.


During testing, you may use any source equipment and source material you and the listeners like. I let listeners take turns doing the switching. I encourage them to listen for as long as they wish and switch whenever and as often they like while listening to any music they wish. I want them to be sure of what they hear.


Different types of source material make a big difference in how easy it is to hear differences. Generally, it is more difficult to hear differences in highly dynamic, transient music than slow, sustained music.


Actually, music isn't even the best material for hearing some types of differences. Steady-state white noise, pink noise, and MLS test tones are far more revealing of frequency response errors than is music. So I usually include some noise during a part of my testing.


You don't have to use "golden ear" listeners for an ABX test. I encourage the disinterested wives of audiophiles to join in the test. I find that they are just as good or better than their audiophile husbands at hearing differences. They also find the test entertaining. Since I often have visitors at my factory with whom I do these tests, it is nice to keep their spouses occupied as well.


If you find differences, you can then explore their cause by being a bit more creative in controlling the variables. For example, let's say you want to know if negative feedback affects the sound. To do so, you will need to have one amplifier with feedback and one without that you can compare.


But if the amplifiers are different in other ways, such as one being solid state and the other being tubed and the two amplifiers are from different manufacturers with different circuitry, then you will have multiple uncontrolled variables so you won't be able to draw any conclusions about feedback.


To eliminate those variables, you will need is to use identical amplifiers that have switchable or adjustable feedback. Several tube amplifiers have this feature. You would then set one amp for maximum feedback and the other for zero feedback for your testing.


You will find that feedback has a big effect on output levels (feedback reduces the level), so even though you are using identical amps, you will still need to match levels very accurately before starting the test.


You can even do the test with a single stereo amp where you compare one channel with feedback to the other without it -- as long as the feedback is independently adjustable for each channel. You will then do the test in mono, which is perfectly okay as you don't need to listen in stereo to hear differences.


It goes without saying that everything else in the signal path must be left alone during the testing process. Only the components under test can be switched.


Along this line, it is also true that it doesn't matter what flaws the other equipment in the signal chain might have. This is because the signal chain is identical for both test components so any differences that are heard can only be caused by the components under test. Poor quality equipment may make it harder to hear differences, but they won't cause differences.


For example, I have had listeners complain that the attenuators in the switch box may be changing the sound. I point out that even if true, since the switch box has attenuators for both components, both components would be equally affected. So the attenuators would not be a variable and any difference in sound could only be caused by differences in the components under test.


If you only use ABX testing and not pre-screening your equipment using measurements, you will find some components that sound different from each other. You then need to determine the cause of the differences you hear.


Sometimes its easy to determine the cause of the difference you heard, such as when one component is much noisier than the other where you hear hiss when you switch to that component. But sometimes its difficult such as when there is a frequency response difference between the components. How do you determine which one is accurate?


Because ABX testing takes a lot of time and effort, I always subject the equipment to instrument tests first to assure it meets the basic quality criteria (BQC) for high fidelity sound. I find that about 20% of equipment fails to meet BQC on instrument testing.


The BQC are:

1) Inaudible noise levels (generally a S/N of 86 dB or better is required)
2) Inaudible wow and flutter (less than 0.01%)
3) Linear frequency response across the audio bandwidth (20Hz - 20 KHz +/- 0.1 dB).
4) Harmonic distortion of less than 1%

If components fail the BQC, they will sound different on an ABX test. But why bother to go to all the trouble of doing an ABX test on them? After all, you will already know the cause of the differences you will hear because you found it using instrument testing.


Specifically, if a component has a poor S/N, you will hear hiss on an ABX test that will cause the component to sound different from one that has a good S/N and is silent. If the frequency response isn't linear, the sound will be different from one that has linear response -- and the instrument measurement will tell you which one is accurate. If high levels of distortion is present, you will hear that as lack of clarity, muddy sound, a sense of strain, poor imaging, and most of the other subjective comments audiophiles use to describe the sound they hear. If wow and flutter is high on one component, you will not even need to ABX test it to recognize it.


The results of ABX testing usually are quite surprising to most audiophiles. They quickly discover that components that meet the BQC always sound identical to each other. Only if components fail the BQC, will they sound different.


Now I understand that many audiophiles will find that hard to believe. But don't shoot me, I'm just the messenger. If you don't believe that components that meet the BQC sound identical, then you need to do some controlled testing and prove it to yourself.


Amplifiers are particularly surprising in ABX tests. When the test is started, obvious differences usually are heard. But it is also quickly discovered that the 'scope shows the amps to be clipping. We usually have to turn the level way down before peak clipping stops. At quiet levels where there is no clipping, the amps will sound identical -- assuming that they pass the BQC.


At this point, you should know the answer to your question about the statement in my previous post where I said that "No human can hear any difference between a linear PCM recording and the source." When listening tests are well controlled, nobody hears any difference between a quality PCM recording and the source because good digital recordings easily meet the BQC. In fact, they far surpass all the BQC by at least a magnitude.


Because of the problems of analog recording, many preamps have "tape monitors", as do all quality analog tape decks. The "tape monitor" is essentially an A-B test as the source and tape output is switchable, the levels are matched (assuming accurate factory setup of the recording equipment), and the listener doesn't have to know which way you throw the switch, so he is "blind." So you can easily test an analog recording while it is being made. You will always hear a difference because no analog recorder will meet the BQC. You may be able to use a preamp with a tape monitor as an ABX switch box for other line-level components as well.


Digital recording is so good that most preamp's no longer have tape monitors and most digital recorders do not have the ability to playback simultaneously while they record (although some still do). This is because it is reasonable to assume that there will be no difference in sound between the source and the recording in a properly-operating digital recorder, so it is perfectly legitimate to eliminate this costly feature.


But you can still do listening tests on digital recorders if you wish. You will just need to make a digital recording from any source, then compare the two using an ABX switch box. Match levels and then start both at the same time so the source material is synchronized and do the test blind.


If all components that pass the BQC sound identical, then we are lead to the logical conclusion that listening tests aren't needed. Why not just measure the component in question to find your answer?


end of Part One - see next post for Part Two
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#18
TESTING PART TWO (part 1 is previous post)

INSTRUMENT TESTING is a lot easier to do than ABX listening tests. Instruments are far more sensitive than human hearing. As a result, you can learn a great deal more about an electronic component with instruments than by listening. So why aren't audiophiles measuring their equipment?

Mostly this is because of ignorance of modern testing procedures and the mistaken believe that quality test equipment is very expensive. All this has changed with the development of computer-based, FFT (Fast Fourier Transform), spectrum analyzer.

A spectrum analyzer is an amazing tool that will evaluate the BQC quickly, easily, and in incredible detail with simply astonishing sensitivity. You can now buy one for less than $500 as computer software and I've even seen free software for them on the internet.

So just what does a spectrum analyzer do and how does it work? A spectrum analyzer will show you the "spectrum" of frequencies produced when you input a test signal into the device under test.

Conceptually, what it does is quite simple. A perfect component will show the test signal frequency only. No other frequencies will be present. If any other frequencies are present, they are distortion or noise.

The spectrum analyzer will show a graph with frequency on the horizontal axis and magnitude on the vertical axis. If you input say a 1 KHz sine wave, you will see a very large spike on the graph at 1 KHz. You should see nothing else.

Of course, no component is perfect, so you will see many other frequencies above and below 1 KHz. These frequencies will take two forms. One will be harmonically related to the test tone and the other will not.

Those frequencies that are harmonically related are harmonic distortion. You will see harmonic frequencies as multiples of the test tone. So if you use a 1 KHz tone, you will see harmonics at 2 KHz (the 2nd harmonic), 3 KHz (the 3rd harmonic), at 4 KHz (the 4th harmonic), and so on.

Those frequencies that are not harmonically related to the test tone are noise. They are random, present at all frequencies, and hopefully should be at a very very low level.

Some noise frequencies represent problems that need to be addressed. For example if you see a big noise spike at 60 Hz, you will know that you have hum (if your mains is 60 Hz). If you see harmonics of 60 Hz at 120 Hz, 180 Hz, etc., then you will know that you have a ground loop.

If you see significant noise spikes at higher frequencies with a component that has digital control circuitry, you may suspect digital noise is bleeding into the analog circuits. Anytime you see significant noise spikes, something is amiss and you need to find out why and fix it.

Most spectrum analyzers will identify the harmonics and label them with their magnitude relative to the reference test tone, or you can interpret the level from the graph lines. Each harmonic will be defined as a certain number of dB below the reference level or a percentage of it.

The analyzer will combine and compute all the harmonics into THD (Total Harmonic Distortion) as a negative number of dB below the reference tone or percentage of it as you wish. The two are related. For example, if the distortion is 100 dB below the reference level, it will also be 0.001% distortion.

The analyzer will also compute the THD+N (THD + Noise), which will be a little higher than the THD alone. If the THD value is very low, and you are testing at near the saturation point of the equipment, you can consider the THD+N to be the S/N. If the THD is high, you must subtract the THD from the THD+N to isolate the noise and get an accurate S/N.

To demonstrate how to read and interpret a power spectrum that you see on a spectrum analyzer, I have attached two photos of my analyzer showing the performance of one of my Magtech amplifiers. The first graph shows the amplifier's spectrum when there is insufficient bias to completely eliminate crossover distortion.


View attachment 828


The graph labels are hard to read in a small photo, so let me say that the first and largest spike you see on the left is the 1 KHz test tone. It is the only thing that should be present in a perfect component. The frequency spikes to the right of the test tone are the first 20 harmonics and their levels are in small boxes above each harmonic (although it is hard to read the values in the pictures).

I have set the horizontal axis to read linear instead of log so that the harmonics are stretched horizontally for easier identification. Also the frequencies below the test tone are eliminated as they aren't needed for analyzing the distortion. But the graph can instantly be switch to log to show everything any time you wish.

The large blue box at the top of the screen shows the THD, which in this case is about half a percent. The ragged line along the bottom is the noise floor, which sits at about 117 dB below the peak of the reference tone.

The second photo shows the same amplifier with the bias adjusted to eliminate crossover distortion. You can see that all of the harmonic spikes above the 5th harmonic have disappeared into the noise floor -- they simply don't exist. The second harmonic has the greatest magnitude, and even it is 99.6 dB below the reference tone. The 3rd harmonic is -102.4 dB, the 4th and 5th harmonics are about -110 dB. The THD is about a thousandth of a percent.

View attachment 827

The amp is being tested under the worst condition, which is at an output level of just 1 watt. This is tough because crossover distortion is a greater percentage of the total distortion at low power levels than at high power levels and the main purpose of bias is to eliminate crossover distortion.

Also, the S/N will be much better at high power levels as the noise floor is fixed and greater output will tremendously increase the magnitude of the difference between the noise and the signal. Even at this very low power level you see that the noise floor is about 118 dB below the reference signal.

In other words, the spectrum analyzer shows that this amplifier has lower distortion and noise than you will find in most preamps! That is truly spectacular performance. Who says that high power, solid state amps sound bad at low power levels? That just isn't true.

Because the spectrum shows each harmonic, you can evaluate the component for its harmonic structure. For example, tubes are believed to have a greater percentage of low order (2nd, 3rd, and 4th) harmonics than solid state equipment which is widely believed to have a greater percentage of high order, odd harmonics (5th, 7th, 9th, 11th, 13th, etc.).

Low order harmonics sound less objectionable than high order, odd harmonics, so the myth says that tube equipment sounds better than solid state for this reason. I haven't found this to necessarily be true (some tube equipment has lots of high order harmonics and some SS gear has more 2nd and 3rd harmonic structure), but it is a widely held perception among audiophiles.

Refer again to the spectrum of the Magtech shown above. It greatest distortion is the 2nd harmonic and all others are lower -- just like a tube amp is supposed to behave. However, since even the 2nd harmonic is about 100 dB below a 1 watt output level, it is far too quiet for a human to hear, so the point is moot. An amplifier with this little distortion simply sounds utterly transparent and does not alter the quality of the sound with euphonic colorations.

The spectrum analyzer can do much more than provide a frequency spectrum. It can also plot the frequency response of a component in real time. It can measure wow and flutter and you can also see any wow and flutter that might be present by instability in the graph. Most have an oscilloscope function. Most can measure multiple channels simultaneously.

You can see that a spectrum analyzer will tell you an enormous amount about your equipment, is easy to use, and is inexpensive. It is shockingly better than the human ear for measuring equipment. For example, scientific studies show that humans can hear distortion down to only about 2%. That's why I say that the distortion in the BQC must only be less than 1%. But I still prefer equipment with vanishingly low distortion levels.

A spectrum analyzer will show distortion down to around one ten-thousandth of one percent, and it will show all the various harmonics and separate those from the noise, something that is impossible to do by listening tests. So for example, my amplifiers have only a few thousandths of a percent distortion when their bias is properly adjusted. I wouldn't (and simply couldn't) set the bias levels by listening for the distortion. Instead, I get it extremely precisely set by turning up the bias just enough to fix the 5th harmonic to -110 dB.

I think it is quite interesting that human hearing can't hear the difference between the two graphs shown, while there is a obviously a great deal of difference in performance of the two as shown on the spectrum analyzer. This is because what appears to be high distortion in the first graph is only about 1/2 of one percent, which is below the human limit for detecting distortion. Imagine what a clipping amplifier's spectrum looks like when it is producing 50% distortion!

Need a spectrum analyzer? You can "Google" them, find them on eBay, or buy the one I use from its German manufacturer at:
http://www.dr-jordan-design.de

There are many other types of test equipment available. Due to my time constraints, I do not have time to present information on the others. Audiophiles really don't need anything more than a spectrum analyzer, so that is what I presented. If readers of this forum want to present more details, please do. Audiophile education is exactly what is needed.

In short, listening tests, if properly controlled are useful. But they are far less sensitive and precise than instrument testing. They are also much more difficult to do.

This discussion has been about evaluating electronics. Loudspeakers are a different kettle of fish. Science has not yet figured out a way to measure everything we hear in a speaker. A good example of this is the fact that there is no way to measure "imaging" in a loudspeaker since imaging is a process that occurs within our brains that no machine can duplicate.

So speaker listening tests are essential. We still need to control the variables and there are many of them. This is a very complex topic and I do not have time to tackle loudspeaker testing at this time.

But I hope this paper has convinced you that open loop listening tests are unsatisfactory because they can't explain the differences you hear between various components. You must eliminate the multiple, uncontrolled variables that make their results meaningless. You simply must do ABX testing to obtain valid results from listening tests.

A spectrum analyzer will show far more about the performance of electronics than listening tests. These are cheap, easy to use, accurate, reliable, and every audiophile should have one. Considering the tens of thousands of dollars most audiophiles spend on equipment, wouldn't a few hundred dollars for a spectrum analyzer be a good investment?

With it, you can easily determine the quality of the components you are using or want to buy. It offers you total confidence as you will be freed from all the controversy and confusion that is the bane of the high-end industry.

Obviously, there is a lot of detail about the various testing processes that I could not cover in this opus. I only covered the basic concepts involved. So if any reader has questions or needs assistance in his quest to do better testing, he may feel free to contact me for additional information.

I realize that this information is controversial. Readers will have three responses to it:

The majority of readers will find the concepts I presented interesting and entertaining. But they will figure that ABX testing is too much trouble and that operating a spectrum analyzer is beyond them. So they will continue to rely on open loop testing and the testimonials of others who use it. As a result, they will remain frustrated with all the conflicting opinions and assumptions they hear about equipment and will be fair game for unscrupulous manufacturers and salesmen.

A second group of readers will strongly disagree with what has been presented and will vehemently defend open loop testing by saying they can hear differences between components in their listening tests.

Of course they can hear differences -- I never said they couldn't. My point is that open loop testing has multiple variables, which prevents them from drawing valid conclusions about cause/effect relationships from what they hear. As a result, open loop testing cannot be used to reliably evaluate the performance of equipment.

I added that listeners will not hear any difference in the sound of components that meet the BQC and are tested under controlled conditions. Therefore, it makes far more sense to use and trust modern test equipment than to do open loop testing.

I have been testing equipment for 40 years and know this is true. So before any audiophile attacks me about this I will challenge them to the following bet:

I will bet $10,000 (or any amount the challenger is willing to lose) that he cannot hear any differences between electronic components that pass the BQC and are then tested under controlled conditions. I never lose this bet. So before you assault me, you need to stand by your convictions and participate in an actual ABX test. Care to bet?

The third group of readers will recognize that what I have presented is valuable information. They will appreciate it and act on it. They will then have the tools they need to accurately evaluate audio electronics. They will no longer be frustrated by all the conflicting opinions out there, which make it impossible to know the truth about equipment. They will finally be free from the tyranny of the snake oil salesmen.

Great Listening,
Roger
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#19
Congratulations Angela. Would Roger be open to explaining more how the power supply works in MagTech? 100% efficiency sounds too good to be true :).
Hi Amir, Roger wanted to sit down and write this up before he left for CES, so here you go. Time to put your feet up:

[FONT=&quot]THE MAGTECH REGULATED POWER SUPPLY[/FONT]

[FONT=&quot]Many audiophiles have asked if the regulator in the Magtech is truly 100% efficient as claimed. The purpose of this paper is to describe how it works so that you can see that it is, in fact, super efficient. It actually does run cold and truly solves the heat problems of conventional regulators that prevent their use in power amplifiers.

So how does the Magtech regulator work? I'll explain, but readers will need to understand the basics in order to appreciate the problems and solutions involved. Since the technical expertise of readers varies, I will cover the basics. I apologize in advance if some of what I am about to say is review for some readers.

First, what exactly is "efficiency" as applies to a voltage regulator? Efficiency is the amount of energy put into a system compared to the amount of energy that you get out of it. Since energy cannot be destroyed and must be accounted for, any losses in efficiency will be reflected as waste heat somewhere in the system.

Or to put it another way, any heat that is produced by the voltage regulator is a loss in efficiency and results in less power being fed to the electronics than would be the case if the regulator was not present. The exact efficiency percentage can be calculated based on watts in compared to watts out or watts of waste heat produced.

In the Magtech's voltage regulator, you will not find any waste heat. It will pass virtually all of the watts put into it on to the amplifiers.

To see why, it is necessary to understand exactly how a power supply operates. Only then will it be possible to see how the Magtech's voltage regulator works and how it can be so efficient.

The purpose of a power supply is to produce smooth DC (Direct Current) at specific voltages to drive the downstream electronics. A basic, linear power supply consists of three sections, each having different types of output characteristics.

The first section is the power transformer. This converts the mains voltage to the voltage(s) required by the downstream electronics. The output is AC (Alternating Current) in the form of a sine wave.
A sine wave is a smooth wave form without any harmonic structure with alternating positive and negative polarity. There is one positive and one negative wave per mains cycle (60 Hz in North America, 50 Hz in the rest of the world).

The second section is a bridge rectifier. This consists of four diodes. Diodes are electric check valves that flow current in only one direction. These diodes flip the phase (polarity) of alternating waves by 180 degrees so that all the waves are in phase. Therefore the output from the bridge rectifier will be pulsating DC sines at twice the mains frequency.

The third section is capacitance. This usually takes the form of a bank of large, storage capacitors. A capacitor bank acts like a rechargeable battery in that it can store a lot of electrons and release them when needed.

The practical difference between a rechargeable battery and a bank of capacitors is speed. A capacitor bank can be charged and discharged virtually instantly, which is necessary to meet the sudden large current demands of an amplifier.

The main purpose of the storage capacitors is to smooth out the current flow from pulsating DC to continuous DC. The storage capacitors are often called filter capacitors since they "filter out" the DC pulses from reaching the downstream electronics. If there were no filter capacitors, an amplifier would make a very loud hum come from the speakers.

The storage capacitors are also needed to help the power transformer deliver enough peak current to reproduce dynamic peaks that require more current than the transformer can deliver. Think of the current that is required to drive the woofer at that moment when a bass drum is struck . . .

The current required by a Class B amplifier is directly proportional to the energy in the music. So at idle (no music), no current is needed or used. Very loud music will require an equally large amount of current to drive the speakers loudly.

It is this huge difference in current that causes the large voltage changes in the rails (the power supply output voltage) you find in most amplifiers. The difference in the rail voltage between idle and full power in most amplifiers is around 30%. This massive voltage drop causes the distortion, the bias, and the output capability of an amplifier to be modulated by the music.

An electronic circuit's distortion can only be optimized at a specific voltage. Any variation of voltage will result in increased distortion.

Class AB amplifiers are a bit more complicated than Class B amplifiers as they require a constant bias current that requires some power. The bias will be optimized at specific rail voltage. Therefore, the bias will change directly with changes in the rail voltage.

But the biggest issue is that an amplifier's power will fall as its rail voltages fall. So unregulated amplifiers suffer significant performance degradation as the music modulates their power supply voltage.

The rail voltage fluctuations caused by amplifier load are only part of the problem. The mains voltage is not stable either.

The mains voltage will vary depending on the load on the power grid and the load on the house wiring. High load conditions can cause the mains voltage to vary by 10% or more.

For example, compare the electrical load and usage in the middle of the night to early evening on a hot summer day. At night people are sleeping so they are not using electrical equipment and the temperature is cool so air conditioners are not running much.

In the early evening, everybody is home from work, dinner is being cooked, electric washers and clothes dryers are operating, air conditioners are maxed out, people are using power-hungry electronics like big TVs, the lights are on, the water heater is running, etc. So the load on both the grid and home wiring is great.

And when do you listen to your music system? Of course, when power demand is the highest and voltage is the lowest. Murphy is hard at work here.

And there is even more bad news. The amplifier itself can severely tax the capacity of your house wiring to which it is attached. A powerful amplifier can draw all the power that is available from your wall receptacle, which is limited to about 2,400 watts on a 20 amp circuit. This will drop the voltage on that line by several percent -- this is in addition to the losses on the grid and in your home from other power uses.

Furthermore, the mains frequency has a big effect on the output of a power supply. This is because a transformer's power is determined by the current it can deliver in its power pulses multiplied by the frequency of those pulses.

This means that a transformer can deliver about 20% more power when operated on a 60 Hz mains than it can when operated on a 50 Hz mains. Therefore, an amplifier with an unregulated power supply will lose up to 20% of its power supply power when operated on a 50 Hz mains.

All this is further complicated by the fact that the relationship between voltage and power in an amplifier is not linear. Power varies by the square of the voltage.

Power is the product of volts times amps. Ohm's Law says that one volt will drive one amp through one ohm of resistance. If you do the math, you will come to realize that the power of an amplifier is determined by the voltage that it can drive into the loudspeaker (assuming it can also deliver the current required).

The formula for calculating amplifier power is the amplifier's RMS output voltage squared and then divided by the speaker's impedance. As an aside, impedance and resistance are the same thing. Resistance applies to DC circuits while impedance is used for AC circuits. This is because the impedance often varies with frequency in AC circuits but there is no frequency in DC circuits. For calculations, you may use impedance and resistance the same way.

To determine the voltage, the formula is the square root of the product of watts times ohms.

Using these formula, you can see that for an amplifier to drive 100 watts into an 8 ohm speaker, it will have to produce 28.28 volts and deliver about 3.5 amps. Now what happens if we drop the power supply voltage by half? The voltage will then be 14.14 volts and the current will drop to 1.76 amps.

How much power will the amplifier now drive into the speakers? It will be just 25 watts. This is a huge loss.

So you can see that the typical 30% loss of rail voltage in an amplifier results in a very large loss of power -- about 50%. If you add an additional loss of mains voltage due to heavy house wire loading, you will lose another big chunk of power.

When you add all the above factors together, you can see that an amplifier's performance is severely degraded by power supply voltage fluctuations and that eliminating them will produce substantially better amplifier performance in terms of power, distortion, and optimum bias levels. So why don't amplifiers have voltage regulated power supplies?

The problem is that the poor efficiency of conventional voltage regulators results in vast amounts of waste heat. Most amplifiers run very hot and adding large amounts of waste heat to an already hot amplifier is intolerable. It is also expensive in terms of both hardware and electricity usage. So it is very rare indeed to find any amplifier that is fully voltage regulated.

So exactly how does a voltage regulator work and what makes it so wasteful and hot that using it is impractical? The most common type of voltage regulator is called a "down" regulator. This means that it pulls down the power supply's voltage so that it remains stable under the worst case conditions.

For example, all quality preamps are voltage regulated so that their power supply voltages will remain stable all the way down to a mains voltage of around 90 volts (using a 120 volt mains). Only if the mains falls below 90 volts ("brown-out" conditions) will the regulation be insufficient and the power supply voltage will start to fall.

The power supply will be driven by a 120 volt mains most of the time, although it might be up to perhaps 125 on occassion. The difference between 120 and 90 volts is about a 30%.

Let's assume that the preamplifier's electronics operate on 12 volts. The electronic engineer will design the power supply to deliver at least 30% more voltage than that (typically about 18 volts). He will then add a "down" regulator to pull the power supply voltage down to 12 volts, which is about the voltage that the power supply would produce using a 90 volt mains. So for any mains voltage between about 90 and 125, the preamp's power supply voltage will be stable at 12 volts.

The regulator actually works by placing a variable load across the power supply in the form of a power transistor that is shunted across the output. A power transistor can be thought of as a very fast-acting, variable resistor whose resistance can be changed electronically. By monitoring the rail voltage, the electronics can adjust the resistance of the transistor to alter the voltage.

As the mains voltage rises, the electronics will reduce the resistance of the loading transistor, which will draw more power and drop the power supply voltage. As the mains voltage falls, the electronics will increase the resistance of the load transistor, which will reduce the power used by the transistor and allow the voltage to rise.

Of course, the action of the electronics are nearly instantaneous, so there is no significant rise and fall of the rail voltages with changes in the mains. The voltage will remain rock stable to within a tiny fraction of a percent.

A down regulator is very inefficient. This is because it operates by feeding a voltage through a resistance. This causes a voltage drop by converting some of the power supply's current into waste heat.

Remember the above concept because it is extremely important. To repeat -- anytime you apply voltage across a resistance, there will be a voltage drop. The loss of current causing the voltage drop will result in waste heat.

The circuitry in a preamp uses only a tiny fraction of an amp (typically just a few milliamps). So the power involved will only be a fraction of a watt or so.

If you waste 30% of a watt in a voltage regulator, the heat produced and the electricity wasted is insignificant. So nobody cares about the efficiency of down regulators when used in small-signal devices.

But now let's look at power amplifiers. Just how much power do we need to regulate?

The typical Class AB amplifier is about 50% efficient. Why? Because it applies its power supply voltage to its output transistors and these act as variable resistors that control the voltage being applied to the speaker. So once again, we have the issue of producing waste heat because we applied voltage across a resistance.

This means that for every watt that the amplifier feeds to the speaker, a watt will be injected as heat into its heat sinks, and two watts will be drawn from the mains. A powerful amplifier like the Magtech will produce 500 watts per channel into 8 ohms. With both channels operating at full power, 1,000 watts will be fed to the speakers. It also means that about 1,000 watts of waste heat will be fed into the heat sinks, and 2,000 watts will be drawn from the mains.

The Magtech's power supply will produce 2,000 watts continuously, so a regulator must be able to control a minimum of 2,000 watts of power (and more to be conservative). The regulator must be able to regulate at least 30% of the rail voltage in order to eliminate fluctuations in voltage due to the variable music demands. In addition, it must be able to handle more than that to account for voltage variations in the mains and 50 Hz operation.

All together, we are looking at regulating about half of the power supply's voltage. This is a daunting task for a down regulator because it means that under worst-case conditions (maximum mains voltage, 60 Hz mains, and with the amp at idle), the regulator will have to dissipate half the power supply's voltage (and hence half its power) as waste heat.

That means that the regulator would produce 1,000 watts of waste heat. This would turn the amp into a room heater and require truly massive heat sinks. It would waste enormous amounts of electricity, be very large, and the heat would cause failures of parts over time. You should now be developing an appreciation of why amplifier power supplies are not regulated!

Although down regulators are very simple and easy to add to circuits, they are just not practical for use in high power circuits due to their inefficiency. But there are other types of regulators, which are more efficient. These are the "up" regulators.

An up regulator requires two power supplies. These have different voltages where one is set for the worst case voltage and the other is set for the best case voltage (say 120 volts and 90 volts for example).

The two power supplies are connected together by a power transistor whose resistance can be varied to allow more or less of the high voltage power supply to be added to the low voltage one. This allows the high voltage supply to bring the voltage "up" and prevent it from falling based on load or mains voltage. The rail voltage can therefore be kept constant between the two extremes by electronically controlling the coupling transistor.

The big advantage of an up regulator is that it only has to handle a percentage of the total power supply voltage (in this example, 30%) instead of all of it. Therefore the losses and waste heat are only a fraction of those produced by a down regulator. But it requires two power supplies, is more complex, and more expensive than a down regulator.

An up regulator still wastes far too much power and produces too much waste heat. So it is still impractical for use in all but very low-power amplifiers.

The next general type of regulator is not a linear regulator like the types I have been describing. It is the switching regulator.

A switching regulator is rather complex, but I'll simplify its operation for clarity. A switcher fundamentally places a transistor in series with the output from the power supply. This transistor is then switched on and off at a high frequency to feed power to the electronics.

The transistor oscillates at a fixed frequency and its "on" time is varied so that it feeds a percentage of the power supply's current to the electronics based on their need. By feeding a capacitor bank, a switcher can adjust the current flow to produce a stable voltage.

Switching power supplies are very efficient (although not 100%) because their transistors are used in only the on or off state. They are not partially turned on like the transistors in linear supplies where a significant resistance is presented to the voltage that produces waste heat.

However, a transistor does not change state instantly. There is still a small percentage of switching time during which the transistors are changing state and resistance is present. So they still produce some waste heat, although this is relatively small, can be tolerated, and therefore switching power supplies can successfully be used in power amplifiers.

But there are big problems when using switching power supplies in high power applications. The main one is noise -- both electrical and mechanical. When switching high power and voltages at high frequencies, radio frequencies are produced. These emissions can adversely affect associated audio electronics and cause instability, oscillation, noise, and general misbehavior.

Powerful switchers also make mechanical noise because there is physical vibration of the switching transistors due to the high currents involved. Switching power supplies are vastly more complex than a simple, 3-part, linear supply and therefore the reliability of switching supplies can be a problem.

There are also many technical problems when designing switching power supplies that make them quite difficult to make work satisfactorily. I won't get into any more detail about this, but rather simply point out that because of all the problems, it is extremely rare to find a linear amplifier with a switching power supply. They do exist, but are not 100% efficient and are not a practical solution for the voltage regulator problem in power amplifiers.

Of course, I have just outlined the basics at this point. There are many variations on the theme that are beyond the scope of this paper. But you should now have enough information to appreciate the solutions that that follow.

So how can the efficiency problem of high power, regulated power supplies be solved? Well, the answer came from thinking outside the box. Specifically, since the heat is produced by applying a voltage to a resistance, the solution had to come from figuring out some way to eliminate doing so.

There is a way. But it could not be done by regulating the continuous DC from the output of a power supply's capacitors because voltage is always present there. The solution had to be done by figuring out a way of regulating without having voltage present. That sounds crazy and impossible, but it can be done.

What about the output from the rectifiers? This is pulsating DC. While the peak of each pulse is at high voltage and power, the voltage at the end and beginning of each pulse is at -- ZERO!

If the regulating power transistor operated only when the voltage was at zero, then there would be no current present, none would be wasted, and no waste heat would be generated. But how can that regulate the voltage? Here's how:

The Magtech uses two power supplies as you would in an up regulator. I call the low voltage one the "ride" supply. It is exactly like the power supply in a conventional, unregulated amplifier.

The second power supply is the "boost" power supply. It has a higher voltage and current rating than the ride supply and can add massively more power to the ride supply when needed.

The ride supply voltage is set for "easy" operation under optimum conditions, i.e., when the mains voltage is at maximum and the amp is at idle. Under these conditions, only the ride supply drives the amplifier circuitry and the boost supply is just on standby.

Note that for the Magtech amp, this is the "easy" condition when the regulator does nothing. By comparison, this is the toughest condition for a down regulator because it has to drag down the power to the worst case level and dissapate massive amounts of power and heat when doing so.

But in the Magtech, this is the voltage that is desired and that the regulator will maintain. Under these easy conditions, the boost supply is not needed.

When significant power is required, the rail voltages will start to fall. This is detected by the power supply's monitoring circuitry, which then switches on the coupling transistors to connect the boost supply to the ride supply. The additional power provided by the boost supply prevents the rail voltage from falling, thereby regulating it.

The key to efficient operation lies in the way that the coupling transistors are operated. First, they are either fully switched on or fully turned off. This means that they have either infinite resistance or essentially none. This prevents them from putting any resistance in the circuit that would cause them to dissipate heat.

Secondly, digital control circuitry is used to monitor the rectifiers' wave form and cause the transistors to switch states (either on or off) at the exact point where the DC pulses cross the zero voltage point. This is important because even though transistors change states very quickly, they do not do so instantaneously. So there is some resistance during the change of state. This is the same problem that causes switching power supplies to be less than perfectly efficient.

If the transistors changed state while the power supply voltage was applied to them, there would be waste heat generated. By only allowing state changes at the zero voltage points, there is no waste heat.

Now if you are observant and thoughtful, you might comment that this does not sound like very good regulation scheme because the two power supplies are either at maximum voltage or minimum voltage because the regulator operates as an all-or-nothing affair. Your thinking is good, but you are overlooking an important feature in the Magtech's power supply.

The digital control circuitry constantly monitors the pulsating waves from the regulator and the rail voltages. It will then make a decision to turn the coupling transistors on or off at each zero point to add as many or few pulses as required to hold the voltage constant.

Under heavy load, the coupling transistors would remain on (possibly even continually), letting most or all of the pulses through. Under light load, they would only be switched on occasionally to let a few pulses through.

While it is true that the regulator has a maximum resolution of 120 pulses per second, each pulse has to charge up a very large bank of capacitors (80,000 uF). Doing so takes time and much current. Therefore, even though each pulse has a lot of current and energy, it can only make a very small change in the capacitor bank's voltage.

The electricity and voltage in the capacitors is analogous to the water in a swimming pool. You can dump a large, 55 gallon drum of water into the pool (a pulse from the boost power supply), but it won't change the level of the water in the pool (the voltage in the capacitors) very much.

By adding more or less pulses as needed, the regulator can maintain a stable voltage to within 0.2 volts. By comparison, without the regulator, the power supply's voltage would vary by more than 50 volts. Which would you prefer?

You can now see why the Magtech regulator produces no heat and is virtually 100% efficient. Technically, I can't claim that the regulator is absolutely 100% efficient because nothing is perfect and there is a very tiny amount of resistance in everything, including the coupling transistors when they are "on."

But the resistance of the transistors is less than one ohm, so they still do not get even warm when operating. Furthermore, they only operate when the amplifier is working fairly hard, so the regulator isn't even active when the amplifier is at idle or at very low power.

In some ways, the Magtech's power supply is like a switcher in that its transistors are either on or off. But there is no specific oscillation involved as in a switcher. Also, it is relatively simple and operates very little and at low frequencies, so its reliability is outstanding (no failures have ever occurred). And because it never switches under power, there is no noise or radio frequency problems with it.

In short, the Magtech's power supply is unique and solves all the problems of other regulators that have prevented power amplifiers from being regulated -- something they badly need even more than other types of electronics. The Magtech regulator's circuit, and particularly the digital control technology involved is the subject of a patent, which currently is pending.

The Magtech amplifier modules are the same sophisticated ones used in the ESL amp that are capable of very high power, the ability to drive 1/3 ohm loads, can handle the most difficult loads (as presented by electrostatic speakers), and need no protective circuitry that ruins the sound of many solid state amps.

When the ESL amplifier modules are combined with a practical voltage regulator, the result is an amplifier with seemingly unlimited power, virtually unmeasurable distortion, and the ability to drive even the most difficult loudspeakers with ease. The Magtech offers a truly new level of performance in amplifiers.

[/FONT][FONT=&quot][/FONT]
 

Angela

WBF Technical Expert
May 25, 2010
141
0
0
Conifer, Colorado
#20
Congratulations Angela. Would Roger be open to explaining more how the power supply works in MagTech? 100% efficiency sounds too good to be true :).
Hi Amir, Roger wanted to sit down and write this up before he left for CES, so here you go. Time to put your feet up:

[FONT=&quot]THE MAGTECH REGULATED POWER SUPPLY[/FONT]

[FONT=&quot]Many audiophiles have asked if the regulator in the Magtech is truly 100% efficient as claimed. The purpose of this paper is to describe how it works so that you can see that it is, in fact, super efficient. It actually does run cold and truly solves the heat problems of conventional regulators that prevent their use in power amplifiers.

So how does the Magtech regulator work? I'll explain, but readers will need to understand the basics in order to appreciate the problems and solutions involved. Since the technical expertise of readers varies, I will cover the basics. I apologize in advance if some of what I am about to say is review for some readers.

First, what exactly is "efficiency" as applies to a voltage regulator? Efficiency is the amount of energy put into a system compared to the amount of energy that you get out of it. Since energy cannot be destroyed and must be accounted for, any losses in efficiency will be reflected as waste heat somewhere in the system.

Or to put it another way, any heat that is produced by the voltage regulator is a loss in efficiency and results in less power being fed to the electronics than would be the case if the regulator was not present. The exact efficiency percentage can be calculated based on watts in compared to watts out or watts of waste heat produced.

In the Magtech's voltage regulator, you will not find any waste heat. It will pass virtually all of the watts put into it on to the amplifiers.

To see why, it is necessary to understand exactly how a power supply operates. Only then will it be possible to see how the Magtech's voltage regulator works and how it can be so efficient.

The purpose of a power supply is to produce smooth DC (Direct Current) at specific voltages to drive the downstream electronics. A basic, linear power supply consists of three sections, each having different types of output characteristics.

The first section is the power transformer. This converts the mains voltage to the voltage(s) required by the downstream electronics. The output is AC (Alternating Current) in the form of a sine wave.
A sine wave is a smooth wave form without any harmonic structure with alternating positive and negative polarity. There is one positive and one negative wave per mains cycle (60 Hz in North America, 50 Hz in the rest of the world).

The second section is a bridge rectifier. This consists of four diodes. Diodes are electric check valves that flow current in only one direction. These diodes flip the phase (polarity) of alternating waves by 180 degrees so that all the waves are in phase. Therefore the output from the bridge rectifier will be pulsating DC sines at twice the mains frequency.

The third section is capacitance. This usually takes the form of a bank of large, storage capacitors. A capacitor bank acts like a rechargeable battery in that it can store a lot of electrons and release them when needed.

The practical difference between a rechargeable battery and a bank of capacitors is speed. A capacitor bank can be charged and discharged virtually instantly, which is necessary to meet the sudden large current demands of an amplifier.

The main purpose of the storage capacitors is to smooth out the current flow from pulsating DC to continuous DC. The storage capacitors are often called filter capacitors since they "filter out" the DC pulses from reaching the downstream electronics. If there were no filter capacitors, an amplifier would make a very loud hum come from the speakers.

The storage capacitors are also needed to help the power transformer deliver enough peak current to reproduce dynamic peaks that require more current than the transformer can deliver. Think of the current that is required to drive the woofer at that moment when a bass drum is struck . . .

The current required by a Class B amplifier is directly proportional to the energy in the music. So at idle (no music), no current is needed or used. Very loud music will require an equally large amount of current to drive the speakers loudly.

It is this huge difference in current that causes the large voltage changes in the rails (the power supply output voltage) you find in most amplifiers. The difference in the rail voltage between idle and full power in most amplifiers is around 30%. This massive voltage drop causes the distortion, the bias, and the output capability of an amplifier to be modulated by the music.

An electronic circuit's distortion can only be optimized at a specific voltage. Any variation of voltage will result in increased distortion.

Class AB amplifiers are a bit more complicated than Class B amplifiers as they require a constant bias current that requires some power. The bias will be optimized at specific rail voltage. Therefore, the bias will change directly with changes in the rail voltage.

But the biggest issue is that an amplifier's power will fall as its rail voltages fall. So unregulated amplifiers suffer significant performance degradation as the music modulates their power supply voltage.

The rail voltage fluctuations caused by amplifier load are only part of the problem. The mains voltage is not stable either.

The mains voltage will vary depending on the load on the power grid and the load on the house wiring. High load conditions can cause the mains voltage to vary by 10% or more.

For example, compare the electrical load and usage in the middle of the night to early evening on a hot summer day. At night people are sleeping so they are not using electrical equipment and the temperature is cool so air conditioners are not running much.

In the early evening, everybody is home from work, dinner is being cooked, electric washers and clothes dryers are operating, air conditioners are maxed out, people are using power-hungry electronics like big TVs, the lights are on, the water heater is running, etc. So the load on both the grid and home wiring is great.

And when do you listen to your music system? Of course, when power demand is the highest and voltage is the lowest. Murphy is hard at work here.

And there is even more bad news. The amplifier itself can severely tax the capacity of your house wiring to which it is attached. A powerful amplifier can draw all the power that is available from your wall receptacle, which is limited to about 2,400 watts on a 20 amp circuit. This will drop the voltage on that line by several percent -- this is in addition to the losses on the grid and in your home from other power uses.

Furthermore, the mains frequency has a big effect on the output of a power supply. This is because a transformer's power is determined by the current it can deliver in its power pulses multiplied by the frequency of those pulses.

This means that a transformer can deliver about 20% more power when operated on a 60 Hz mains than it can when operated on a 50 Hz mains. Therefore, an amplifier with an unregulated power supply will lose up to 20% of its power supply power when operated on a 50 Hz mains.

All this is further complicated by the fact that the relationship between voltage and power in an amplifier is not linear. Power varies by the square of the voltage.

Power is the product of volts times amps. Ohm's Law says that one volt will drive one amp through one ohm of resistance. If you do the math, you will come to realize that the power of an amplifier is determined by the voltage that it can drive into the loudspeaker (assuming it can also deliver the current required).

The formula for calculating amplifier power is the amplifier's RMS output voltage squared and then divided by the speaker's impedance. As an aside, impedance and resistance are the same thing. Resistance applies to DC circuits while impedance is used for AC circuits. This is because the impedance often varies with frequency in AC circuits but there is no frequency in DC circuits. For calculations, you may use impedance and resistance the same way.

To determine the voltage, the formula is the square root of the product of watts times ohms.

Using these formula, you can see that for an amplifier to drive 100 watts into an 8 ohm speaker, it will have to produce 28.28 volts and deliver about 3.5 amps. Now what happens if we drop the power supply voltage by half? The voltage will then be 14.14 volts and the current will drop to 1.76 amps.

How much power will the amplifier now drive into the speakers? It will be just 25 watts. This is a huge loss.

So you can see that the typical 30% loss of rail voltage in an amplifier results in a very large loss of power -- about 50%. If you add an additional loss of mains voltage due to heavy house wire loading, you will lose another big chunk of power.

When you add all the above factors together, you can see that an amplifier's performance is severely degraded by power supply voltage fluctuations and that eliminating them will produce substantially better amplifier performance in terms of power, distortion, and optimum bias levels. So why don't amplifiers have voltage regulated power supplies?

The problem is that the poor efficiency of conventional voltage regulators results in vast amounts of waste heat. Most amplifiers run very hot and adding large amounts of waste heat to an already hot amplifier is intolerable. It is also expensive in terms of both hardware and electricity usage. So it is very rare indeed to find any amplifier that is fully voltage regulated.

So exactly how does a voltage regulator work and what makes it so wasteful and hot that using it is impractical? The most common type of voltage regulator is called a "down" regulator. This means that it pulls down the power supply's voltage so that it remains stable under the worst case conditions.

For example, all quality preamps are voltage regulated so that their power supply voltages will remain stable all the way down to a mains voltage of around 90 volts (using a 120 volt mains). Only if the mains falls below 90 volts ("brown-out" conditions) will the regulation be insufficient and the power supply voltage will start to fall.

The power supply will be driven by a 120 volt mains most of the time, although it might be up to perhaps 125 on occassion. The difference between 120 and 90 volts is about a 30%.

Let's assume that the preamplifier's electronics operate on 12 volts. The electronic engineer will design the power supply to deliver at least 30% more voltage than that (typically about 18 volts). He will then add a "down" regulator to pull the power supply voltage down to 12 volts, which is about the voltage that the power supply would produce using a 90 volt mains. So for any mains voltage between about 90 and 125, the preamp's power supply voltage will be stable at 12 volts.

The regulator actually works by placing a variable load across the power supply in the form of a power transistor that is shunted across the output. A power transistor can be thought of as a very fast-acting, variable resistor whose resistance can be changed electronically. By monitoring the rail voltage, the electronics can adjust the resistance of the transistor to alter the voltage.

As the mains voltage rises, the electronics will reduce the resistance of the loading transistor, which will draw more power and drop the power supply voltage. As the mains voltage falls, the electronics will increase the resistance of the load transistor, which will reduce the power used by the transistor and allow the voltage to rise.

Of course, the action of the electronics are nearly instantaneous, so there is no significant rise and fall of the rail voltages with changes in the mains. The voltage will remain rock stable to within a tiny fraction of a percent.

A down regulator is very inefficient. This is because it operates by feeding a voltage through a resistance. This causes a voltage drop by converting some of the power supply's current into waste heat.

Remember the above concept because it is extremely important. To repeat -- anytime you apply voltage across a resistance, there will be a voltage drop. The loss of current causing the voltage drop will result in waste heat.

The circuitry in a preamp uses only a tiny fraction of an amp (typically just a few milliamps). So the power involved will only be a fraction of a watt or so.

If you waste 30% of a watt in a voltage regulator, the heat produced and the electricity wasted is insignificant. So nobody cares about the efficiency of down regulators when used in small-signal devices.

But now let's look at power amplifiers. Just how much power do we need to regulate?

The typical Class AB amplifier is about 50% efficient. Why? Because it applies its power supply voltage to its output transistors and these act as variable resistors that control the voltage being applied to the speaker. So once again, we have the issue of producing waste heat because we applied voltage across a resistance.

This means that for every watt that the amplifier feeds to the speaker, a watt will be injected as heat into its heat sinks, and two watts will be drawn from the mains. A powerful amplifier like the Magtech will produce 500 watts per channel into 8 ohms. With both channels operating at full power, 1,000 watts will be fed to the speakers. It also means that about 1,000 watts of waste heat will be fed into the heat sinks, and 2,000 watts will be drawn from the mains.

The Magtech's power supply will produce 2,000 watts continuously, so a regulator must be able to control a minimum of 2,000 watts of power (and more to be conservative). The regulator must be able to regulate at least 30% of the rail voltage in order to eliminate fluctuations in voltage due to the variable music demands. In addition, it must be able to handle more than that to account for voltage variations in the mains and 50 Hz operation.

All together, we are looking at regulating about half of the power supply's voltage. This is a daunting task for a down regulator because it means that under worst-case conditions (maximum mains voltage, 60 Hz mains, and with the amp at idle), the regulator will have to dissipate half the power supply's voltage (and hence half its power) as waste heat.

That means that the regulator would produce 1,000 watts of waste heat. This would turn the amp into a room heater and require truly massive heat sinks. It would waste enormous amounts of electricity, be very large, and the heat would cause failures of parts over time. You should now be developing an appreciation of why amplifier power supplies are not regulated!

Although down regulators are very simple and easy to add to circuits, they are just not practical for use in high power circuits due to their inefficiency. But there are other types of regulators, which are more efficient. These are the "up" regulators.

An up regulator requires two power supplies. These have different voltages where one is set for the worst case voltage and the other is set for the best case voltage (say 120 volts and 90 volts for example).

The two power supplies are connected together by a power transistor whose resistance can be varied to allow more or less of the high voltage power supply to be added to the low voltage one. This allows the high voltage supply to bring the voltage "up" and prevent it from falling based on load or mains voltage. The rail voltage can therefore be kept constant between the two extremes by electronically controlling the coupling transistor.

The big advantage of an up regulator is that it only has to handle a percentage of the total power supply voltage (in this example, 30%) instead of all of it. Therefore the losses and waste heat are only a fraction of those produced by a down regulator. But it requires two power supplies, is more complex, and more expensive than a down regulator.

An up regulator still wastes far too much power and produces too much waste heat. So it is still impractical for use in all but very low-power amplifiers.

The next general type of regulator is not a linear regulator like the types I have been describing. It is the switching regulator.

A switching regulator is rather complex, but I'll simplify its operation for clarity. A switcher fundamentally places a transistor in series with the output from the power supply. This transistor is then switched on and off at a high frequency to feed power to the electronics.

The transistor oscillates at a fixed frequency and its "on" time is varied so that it feeds a percentage of the power supply's current to the electronics based on their need. By feeding a capacitor bank, a switcher can adjust the current flow to produce a stable voltage.

Switching power supplies are very efficient (although not 100%) because their transistors are used in only the on or off state. They are not partially turned on like the transistors in linear supplies where a significant resistance is presented to the voltage that produces waste heat.

However, a transistor does not change state instantly. There is still a small percentage of switching time during which the transistors are changing state and resistance is present. So they still produce some waste heat, although this is relatively small, can be tolerated, and therefore switching power supplies can successfully be used in power amplifiers.

But there are big problems when using switching power supplies in high power applications. The main one is noise -- both electrical and mechanical. When switching high power and voltages at high frequencies, radio frequencies are produced. These emissions can adversely affect associated audio electronics and cause instability, oscillation, noise, and general misbehavior.

Powerful switchers also make mechanical noise because there is physical vibration of the switching transistors due to the high currents involved. Switching power supplies are vastly more complex than a simple, 3-part, linear supply and therefore the reliability of switching supplies can be a problem.

There are also many technical problems when designing switching power supplies that make them quite difficult to make work satisfactorily. I won't get into any more detail about this, but rather simply point out that because of all the problems, it is extremely rare to find a linear amplifier with a switching power supply. They do exist, but are not 100% efficient and are not a practical solution for the voltage regulator problem in power amplifiers.

Of course, I have just outlined the basics at this point. There are many variations on the theme that are beyond the scope of this paper. But you should now have enough information to appreciate the solutions that that follow.

So how can the efficiency problem of high power, regulated power supplies be solved? Well, the answer came from thinking outside the box. Specifically, since the heat is produced by applying a voltage to a resistance, the solution had to come from figuring out some way to eliminate doing so.

There is a way. But it could not be done by regulating the continuous DC from the output of a power supply's capacitors because voltage is always present there. The solution had to be done by figuring out a way of regulating without having voltage present. That sounds crazy and impossible, but it can be done.

What about the output from the rectifiers? This is pulsating DC. While the peak of each pulse is at high voltage and power, the voltage at the end and beginning of each pulse is at -- ZERO!

If the regulating power transistor operated only when the voltage was at zero, then there would be no current present, none would be wasted, and no waste heat would be generated. But how can that regulate the voltage? Here's how:

The Magtech uses two power supplies as you would in an up regulator. I call the low voltage one the "ride" supply. It is exactly like the power supply in a conventional, unregulated amplifier.

The second power supply is the "boost" power supply. It has a higher voltage and current rating than the ride supply and can add massively more power to the ride supply when needed.

The ride supply voltage is set for "easy" operation under optimum conditions, i.e., when the mains voltage is at maximum and the amp is at idle. Under these conditions, only the ride supply drives the amplifier circuitry and the boost supply is just on standby.

Note that for the Magtech amp, this is the "easy" condition when the regulator does nothing. By comparison, this is the toughest condition for a down regulator because it has to drag down the power to the worst case level and dissapate massive amounts of power and heat when doing so.

But in the Magtech, this is the voltage that is desired and that the regulator will maintain. Under these easy conditions, the boost supply is not needed.

When significant power is required, the rail voltages will start to fall. This is detected by the power supply's monitoring circuitry, which then switches on the coupling transistors to connect the boost supply to the ride supply. The additional power provided by the boost supply prevents the rail voltage from falling, thereby regulating it.

The key to efficient operation lies in the way that the coupling transistors are operated. First, they are either fully switched on or fully turned off. This means that they have either infinite resistance or essentially none. This prevents them from putting any resistance in the circuit that would cause them to dissipate heat.

Secondly, digital control circuitry is used to monitor the rectifiers' wave form and cause the transistors to switch states (either on or off) at the exact point where the DC pulses cross the zero voltage point. This is important because even though transistors change states very quickly, they do not do so instantaneously. So there is some resistance during the change of state. This is the same problem that causes switching power supplies to be less than perfectly efficient.

If the transistors changed state while the power supply voltage was applied to them, there would be waste heat generated. By only allowing state changes at the zero voltage points, there is no waste heat.

Now if you are observant and thoughtful, you might comment that this does not sound like very good regulation scheme because the two power supplies are either at maximum voltage or minimum voltage because the regulator operates as an all-or-nothing affair. Your thinking is good, but you are overlooking an important feature in the Magtech's power supply.

The digital control circuitry constantly monitors the pulsating waves from the regulator and the rail voltages. It will then make a decision to turn the coupling transistors on or off at each zero point to add as many or few pulses as required to hold the voltage constant.

Under heavy load, the coupling transistors would remain on (possibly even continually), letting most or all of the pulses through. Under light load, they would only be switched on occasionally to let a few pulses through.

While it is true that the regulator has a maximum resolution of 120 pulses per second, each pulse has to charge up a very large bank of capacitors (80,000 uF). Doing so takes time and much current. Therefore, even though each pulse has a lot of current and energy, it can only make a very small change in the capacitor bank's voltage.

The electricity and voltage in the capacitors is analogous to the water in a swimming pool. You can dump a large, 55 gallon drum of water into the pool (a pulse from the boost power supply), but it won't change the level of the water in the pool (the voltage in the capacitors) very much.

By adding more or less pulses as needed, the regulator can maintain a stable voltage to within 0.2 volts. By comparison, without the regulator, the power supply's voltage would vary by more than 50 volts. Which would you prefer?

You can now see why the Magtech regulator produces no heat and is virtually 100% efficient. Technically, I can't claim that the regulator is absolutely 100% efficient because nothing is perfect and there is a very tiny amount of resistance in everything, including the coupling transistors when they are "on."

But the resistance of the transistors is less than one ohm, so they still do not get even warm when operating. Furthermore, they only operate when the amplifier is working fairly hard, so the regulator isn't even active when the amplifier is at idle or at very low power.

In some ways, the Magtech's power supply is like a switcher in that its transistors are either on or off. But there is no specific oscillation involved as in a switcher. Also, it is relatively simple and operates very little and at low frequencies, so its reliability is outstanding (no failures have ever occurred). And because it never switches under power, there is no noise or radio frequency problems with it.

In short, the Magtech's power supply is unique and solves all the problems of other regulators that have prevented power amplifiers from being regulated -- something they badly need even more than other types of electronics. The Magtech regulator's circuit, and particularly the digital control technology involved is the subject of a patent, which currently is pending.

The Magtech amplifier modules are the same sophisticated ones used in the ESL amp that are capable of very high power, the ability to drive 1/3 ohm loads, can handle the most difficult loads (as presented by electrostatic speakers), and need no protective circuitry that ruins the sound of many solid state amps.

When the ESL amplifier modules are combined with a practical voltage regulator, the result is an amplifier with seemingly unlimited power, virtually unmeasurable distortion, and the ability to drive even the most difficult loudspeakers with ease. The Magtech offers a truly new level of performance in amplifiers.

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