Interview with Roger Sanders

Exclusive interview


Exclusive and in-depth interview with Roger Sanders by Matej Isak. Roger really took his time and went trough the answers in most elaborate way. Intriguing and direct look into the high-end audio society.

1)  Where and how did  all started for Sanders Audio?

I started building audio equipment with my father back in the 1950s.  I was very frustrated with the poor quality of equipment available then and soon realized that I would have to make better equipment myself if I wanted excellent sound.

When I went to the university in the 1960's, I started working in the audio/video repair department where I learned all about the technology involved in audio.  I learned a great deal about audio and test gear by repairing, using, and aligning audio equipment.

I also ran the recording studio at the university's concert hall.  It was there that I realized that the secret to high quality sound is good recordings.  I have continued to record symphony orchestras and local musicians ever since.  I have some truly realistic recordings as a result.

It was at the university where I first encountered electrostatic loudspeakers.  They were astonishingly clear!  They were so obviously superior to magnetic speakers that I knew immediately that I would have to spend my entire life being involved with them.

The reason that electrostatic speakers are so good is that they are the only type of midrange driver that has essentially no moving mass.  Magnetic speakers simply cannot match the performance of electrostatics in the midrange because they are heavy so cannot be accelerated quickly and accurately at treble frequencies.

Electrostatic speakers had serious shortcomings and problems.  I have spent the last 42 years doing the basic research and design work required to overcome their problems.  The result is that the electrostatic speakers I have designed do not have the limitations of other electrostatic speakers.  They can play at extremely loud levels and are very reliable.

My work with electrostatic speakers was widely recognized in audio circles.  I was therefore commissioned to write articles and books on the subject.  Some of my articles can be seen on my website under "Audio Related Articles."  My book on electrostatics is entitled "The Electrostatic Loudspeaker Design Cookbook" and was published in 1994.

In 1996, I was hired by a company in the U.K. to start a company called Innersound.  I designed and manufactured electrostatic loudspeakers for them.

By 2002, I had grown Innersound into a moderately successful company.  An investor therefore bought it from its original owner and hired me to run the new company.  But business conflicts arose so I left Innersound in 2004.  I then started my own company (Sanders Sound Systems) and now manufacture my own electrostatic speakers.

I became a systems engineer out of necessity.  Electrostatic speakers are very hard to drive and very few conventional amplifiers are able to drive them well.  So I was forced to design a special type of amplifier to drive electrostatic speakers.  This is known as the ESL amp.  Interested readers can learn all about it on my website.  See my "Electrostatic Amplifier" white paper.

This and many other interesting and informative white papers can be found on the "Technical White Papers" page of my website.  The direct link is:

The ESL amp was so successful that I received hundreds of requests to develop a better amplifier for driving magnetic speakers.  Such speakers require that their amplifier have a regulated power supply.  No high power amplifiers are regulated.  So I developed a unique, linear, regulated power supply for amplifiers that is 99.7% efficient.  This makes it possible to finally use regulated power supplies in amplifiers.

My Magtech amplifier is the only high power amplifier available that has a regulated power supply.  It therefore has become a very popular amplifier and is now my best selling product.  Since it is the only high-power amplifier on the market with a regulated power supply, no other amplifier can match its performance.  You can read details about this unique design on the technical white papers page of my website.

Electrostatic loudspeakers are capacitors.  So they also make unique demands on cables.  I developed cables specifically designed for them because convention cables simply didn't work well.  To round out my line of products, I also design and manufacture a preamplifier so that customers can get an entire system from me.

2)   What drew you to the high-end industry?

I have never been drawn to the high-end industry.  In fact, I am repelled by it.  It is full of deceit and dishonest manufacturers and businessmen who are only interested in stealing money from gullible audiophiles.  I very much dislike being associated with such unethical behavior.

I believe in being honest with my customers and providing truly superb products at reasonable prices -- something that is largely absent in high-end audio.  Unfortunately, most of my potential customers are involved in high-end audio, therefore I am forced to be involved with it

3)  Would you consider yourself as an audiophile?

No, I am not an audiophile.  I am a concert musician, scientist, and engineer.

4)  Kindly list all of the present products from Sanders Audio?

My products include several electrostatic speaker systems, electrostatic amplifiers (both stereo and monoblocks), magnetic amplifiers (both stereo and monoblocks), preamplifiers, and speaker cables for both electrostatic and magnetic speakers.

5)  When did you established Sanders Audio?


6)  Do you produce everything in house?

Yes.  I learned from running Innersound that out-sourcing is a bad idea.  It is difficult to control quality or production timing when you have to rely on others.  It also costs more to have others make your products compared to doing it yourself.

So at Sanders, we build everything possible in house.  We produce all our own electronics.  We have a complete machine shop for building chassis.  We have special facilities for manufacturing electrostatic panels.  We have a cabinet shop and a paint shop.  We have complete test facilities.  We have a metal finishing shop.

7)  You don't implement any tubes, but only solid state. Can you elaborate?  Have you ever even thought about implementing tubes into your products?

That is incorrect.  I have designed, used, manufactured, and sold tube equipment.

My first tube amplifier design was a unique product that was designed to drive electrostatic speakers directly at high voltage without any audio transformers.  You can read an article on it in The Audio Amateur Magazine back in 1976.  A copy of this article is on my website under "Audio Related Articles."

I also designed and manufactured a conventional tube amp that was customized for driving electrostatic speakers when I ran Innersound.  We manufactured and sold this product for a short while.

However, tube equipment is inferior to solid state equipment and simply does not perform as well.  Please read my "Tubes vs. Transistors" white paper for details.  I refuse to compromise on performance, so I only manufacture solid state equipment today.

8)  Tubes vs Transistors? How you see it?

This is too long a topic to discuss here.  I have written a "Tubes vs. Transistors" white paper where I discuss this issue in great detail.  Please read this article on the Technical White Papers section of my website.

9)   Do you think balanced topology is a must for best sound?

No.  Balanced topology is inferior to unbalanced.  This is because balanced equipment is much more complex so has twice the noise and distortion of unbalanced equipment.  Simplicity and purity is best.

But balanced equipment is able to reject external noise fields (RFI and EMI) better than unbalanced gear.  Therefore balanced equipment is needed for professional applications where high external noise fields are present -- like when making live recordings.

The typical home does not have any problem with external noise fields, so balanced equipment is not needed in the home.  So I always use unbalanced equipment unless I am forced to use balanced equipment due to external noise problems.

Note carefully that there is nothing magic about balanced operation.  One simply uses two phases of the same signal for balanced operation.  Only one phase is used in an unbalanced system.  Therefore the signals in a balanced and unbalanced system are the same.  So there is no actual sound difference between them.  However, because there are two identical signals in a balanced system, the noise and distortion will be doubled.

Audiophiles usually think that balanced operation is better because it plays slightly louder.  Louder sound sounds better to us, so most audiophiles are tricked into believing that balanced sound is better.  But if they matched levels in their testing, they would discover that there is no difference in the sound of a balanced and unbalanced system.

10)  Do you see your speakers as state of the art?

Yes.  There is no speaker made that can match the performance of mine.

This is another long topic.  So let me just summarize by pointing out that to reproduce the midrange frequencies well requires a massless and distortionless driver.  Only electrostatic speakers are massless, so they are the only type that can accurately reproduce all the detail in the midrange.  One simply must use electrostatic speakers for S.O.T.A. sound.

Electrostatic speakers are the only type of speaker that is driven uniformly over their entire surface.  By comparison, magnetic speakers are only driven at one point, so there is always a lot of bending and distortion of their moving surfaces.

The diaphragm in an electrostatic speaker remains flat, thereby producing vastly lower distortion than magnetic speakers.  For these reasons, a S.O.T.A. speaker must use an electrostatic midrange.
However, the laws of physics make it impossible for electrostatic bass to be deep, powerful, loud, or linear.  Therefore the so called "full-range" electrostatic speaker has unacceptable performance because it cannot reproduce dynamic music at live levels.  No full range ESL can be a S.O.T.A. speaker.  It is necessary to use conventional magnetic woofers for good bass performance.

Unfortunately, magnetic woofers have many problems with their enclosures that cause them to produce with a lot of overshoot and ringing that makes it impossible for them to integrate well with a massless electrostatic speaker.  As a result, hybrid speakers have a poor reputation for quality sound and none have been S.O.T.A.

But it doesn't have to be that way.  I spent 17 years solving that problem.  The solution is to use a transmission line enclosure system to virtually eliminate overshoot and ringing in the woofer.  When such a system is combined with a very low crossover point and is bi-amplified so that the amplifier is directly connected to the woofer, it is possible to stop the woofer fast enough to match an ESL.  This is what I use to get excellent integration.

Another critical factor is the crossover.  Conventional passive crossovers have very poor performance.  Active, electronic crossovers are far superior -- especially when digital technology is used.

No speaker can be considered S.O.T.A. if it uses a passive crossover.  All speakers will perform better when driven by active crossovers and individual amplifiers for each of their drivers.  I do not compromise, so I only make speakers using digital crossovers and bi-amp my speakers.

The recent invention of the digital crossover has made electronic crossovers produce spectacular performance.  They are particularly useful because they have much steeper crossover slopes (typically 48 dB/octave) than conventional passive crossovers (typically 12 dB/octave).

This is extremely important because woofers do not simply stop at the crossover point.  That is the point where they just start to roll off.

A woofer must roll off by about 50 dB before it will be inaudible.  A woofer using a passive crossover will be producing sound for about four octaves above its crossover point.  This means that it will operate through the critical midrange.  I am sure you will agree that a heavy woofer cannot reproduce the midrange frequencies well.  So using passive crossovers guarantees poor midrange performance.

Many speakers use midrange drivers to improve the sound of the midrange compared to using a woofer to reproduce the midrange.  While this helps, it does not eliminate the problem of operating the woofer in the midrange.  You simply end up mixing the poor woofer sound with the better midrange driver's sound.  The result is a mixture of sounds that are different and that degrade the sound quality.

By using digital crossovers at 48 dB/octave and crossing the woofer over at just 172 Hz, my speakers eliminate the woofer from the sound by just 350 Hz.  As a result, the entire midrange is only produced by the massless/distortionless electrostatic panel.  The integration between the two drivers is perfect, so my speakers sound like a pure electrostatic speaker -- but with full frequency response and deep, powerful bass.  They then have S.O.T.A. performance.

Digital crossovers also offer many features that are simply impossible to do with analog crossovers.  For example, the speaker can have all its drivers time-aligned using digital delay instead of by mechanical means.  This makes it possible to achieve perfect phasing between the drivers for superb imaging.

Digital crossover also offer equalization for each driver -- and virtually all drivers need equalization. Digital crossover even make it possible to use dynamic equalization, which makes it possible to equalize the woofer for powerful deep bass -- but then automatically reduce the equalization at high SPLs to prevent overdriving the amplifier and damaging the woofer.

Historically, electrostatic speakers cannot play loudly.  Mine can.  It has taken me 42 years to solve this problem.  But my electrostatic speaker are arc proof, so you can play them at live levels using multi-thousand watt amplifiers without any damage or problems.

Therefore they can reproduce dynamic music at live levels, which no other electrostatic can.  The clarity and detail they produce is simply unmatched.  Their phasing and imaging is also unmatched.  For all these reasons and many others, my speakers are simply the best available.

11)  What would you say sets apart Sanders Audio design above other manufacturers?

I offer the very finest performing products available -- but at fair and reasonable prices.  You can spend more money, but you can't find any components that actually perform better than mine.  My products are genuinely unique and like no others.

12)  Do you use pure Class A in your signal chain?  Or is AB the ultimate choice?

It is not possible, desirable, necessary, or practical to use Class A operation in powerful amplifiers.  So I do not use it.  Like other beliefs in the audiophile world, the idea that Class A operation is better is a myth.  Let me explain.

Class A operation was required back in the 1950s because tube amplifiers of that vintage had very high distortion.  This was because negative feedback had not yet been invented and the tubes and transformers of the day had very poor linearity and limited bandwidth.

The purpose of Class A operation is to reduce distortion -- that is all it does.  Using it would reduce the distortion of a typical tube amp from the 1950's from about 7% THD to about 3% THD.

Class A operation is extremely inefficient and produces a lot of heat.  Amplifiers of the 1950's had very low power.  Fifteen watts was typical and the most powerful ones only produced 40 watts.  At such low power levels, the fact that Class A operation wastes half the amplifier's power as heat was insignificant.  So nobody cared if the amplifier was operated in Class A.

The invention of the transistor combined with the development of negative feedback made it possible to reduce distortion levels to around 0.001% -- without using Class A operation.  This is vastly less distortion than any human can hear (humans cannot hear less than 1% distortion), so there is no longer any need to use Class A operation.  Today, an engineer can design a Class AB amplifier that has the same low distortion as a Class A amplifier.

This is fortunate, because it simply isn't possible to use Class A operation in powerful amplifiers due to all the waste heat.   For example, my Magtech amplifier would produce about 1,000 watts of heat at idle if I operated it in Class A.

It isn't possible to cool such an amp using convection air.  I would have to water cool it if I ran it in Class A.  And in any case, I couldn't get enough power from the mains to run it at the high power levels required to reproduce music cleanly.

In today's amplifiers, the distortion from a Class A amplifier is no lower than in a Class AB amplifier.  So there is absolutely no reason to use Class A anymore.  I therefore use Class AB -- as do all other manufacturers of quality amplifiers.

13)  How important is the power supply in your view?

A quality power supply is essential.  A component can only reach its full potential when fed suitable power.  The power supply forms the foundation of a component and must not be compromised.

14)  Classical music or studio recorded music. What do you see as ultimate test for ultra high-end?

The most difficult test of an audio system is a full symphony orchestra.  No other type of music can stress it to the same degree.  So when evaluating the performance of an audio system, I always want to hear full symphony.  Of course, I listen to other music as well, but if you must pick the most revealing and difficult music, a symphony orchestra recording is essential.

15)  What is your opinion of digital music revolution and the birth of MP3?

Digital recording is one of the very best and most important things to ever happen to audio.  It lies right up there with the invention of the transistor.  When I started doing live recording back in the 1960's, all our recordings were severely flawed due the limitations of analog recording equipment.
Analog was so bad that it was standard practice to compare the live microphone feed to the recording to assure that the recorder wasn't performing too badly.  It was always apparent that the two were very different.  It was impossible to make a truly accurate and realistic analog recording.
When digital recording became available, it was finally possible to make perfect recordings.  Nobody can hear any difference between a live microphone feed and a recording of it.

You can still see remnants of the deficiencies of analog recording in that many preamps today still have "tape loops."  Recall that the whole purpose of a tape loop was to compare the incoming sound to the recorded sound by flipping the tape switch.  This was necessary to assure that the analog recorder was working reasonably well (but never perfectly).

Most preamps today have abandoned the tape loop because there is no audible difference between a digital recording and the original sound. There is simply no purpose or need for a tape loop anymore.

I will elaborate further on this shortly, but first let me comment on MP3.  It is incorrect to lump all MP3 recordings together.  MP3 recordings perform very differently depending on their data rates.  It is essential to specify the MP3's data rate before discussing this format.

Low data rates (less than 192 KBPS) degrade the sound while high data rates (192 KBPS and higher) sound flawless.  So you always must specify the data rate of an MP3 recording before you can comment on it.  As long as you use a data rate of 192 KBPS or higher, you cannot hear any difference between an MP3 recording and the original source.

So my opinion of MP3 is that high data rate MP3 is outstanding, while low data rate MP3 is only suitable for speech.  In other words, MP3 is fine as long as you select the appropriate data rate.

Most audiophiles do not have a full understanding of how digital recording works, so they misunderstand the process completely.  So allow me to shed some light on the situation.  Since I do not know your level of technical knowledge, I will start with the basics. I apologize if some of this is review, so please bear with me.

I will be discussing mainly linear PCM (Pulse Code Modulation) recording because this is the format used by CDs. There are other digital systems (like SACD and DSD), but they are not as good as linear PCM so I will not take the time to discuss them at length due to time constraints.

The linear PCM standard used by CDs is specified in detail in the "Red Book" developed by the Sony and Phillips engineers when they invented the CD back in the '80s.  I will be referring to this standard often throughout this discussion.

The Red Book engineers wanted to produce a digital recording system that would reproduce music perfectly -- but use the minimum amount of data so that they could maximize the recording time on a CD.  They wisely did not compromise performance -- but nor did they use more data than what was necessary.

Analog recordings of the day were compromised in many ways.  Specifically, the frequency bandwidth of LPs and FM multiplex broadcasts were limited from 30 Hz to 15 KHz.  The S/N (Signal to Noise ratio) was limited to around 40 dB.  Even the best studio, open-reel tape decks could barely achieve a bandwidth of  20 Hz -  20 KHz with a S/N of 68 dB.   By the late 1970's better tape oxide formulations got the S/N up to 72 dB.  No analog system could capture the full dynamic range of a symphony orchestra.  None had a silent background.

Analog tape decks has loads of flutter (caused by imperfect capstan bearings, capstan shafts that weren't round, and tape scrape flutter (where the tape moves in tiny jerks across the tape head).  LP's were plagued with "wow" due to eccentricities in the disks caused by the center hole not being perfectly concentric with the record grooves.   Both wow and flutter are inaccuracies and variations in the frequency.

Tape decks also suffered from amplitude flutter caused by variations in the tape coating thickness. If you recorded a steady state tone from an audio generator, you would see +/- 2 dB fluctuations in the output from a tape deck.   This flutter is easily be heard on music that has sustained tones.  A good example is slow, sustained piano music.

The frequency response of these analog recording systems was not only limited to just a portion of the human hearing range, but the linearity of the frequency response was quite poor.  It is typical to see frequency response variations greater than plus/minus 3 dB.

As if all this weren't bad enough, the THD (Total Harmonic Distortion) and IMD (InterModulation Distortion) of analog recording systems often exceeded several percent.  As a result, analog recordings always sounded obviously different than the live microphone feed.  Analog tape recorders simply could not capture, store and reproduce music with accuracy.

LPs had substantially worse performance than open reel tape.  So by the time you made a recording on tape and then transferred it to disk, the accumulated errors were very severe.  As a result, the LP playback on an audio system was quite different from the original, live performance.  A vastly better way to record music was needed.

The Sony and Phillips engineers who invented digital recording decided to produce a digital system that would solve all these problems.  This meant that the S/N of the new system must be greater than 86 dB.  This is minimum needed to produce the full dynamic range of a symphony orchestra, which is about 72 dB.  The noise floor needs to be at least 10 dB below that to produce a silent
background.  So they settled on a S/N at a very conservative 96 dB, which was 10 dB better than the minimum required.

The engineers wanted to capture and reproduce the full frequency range of human hearing, so their CD was able to record  20 Hz to 20 KHz.  Actually the nature of a CD is such that it will record right down to DC, which is zero Hz.  But the highs are limited to the extremes of human hearing at 20 KHz.

Only a child can hear 20 KHz.  But the Red Book engineers made no compromises, so pushed the frequency response all the way to  20 KHz even though no adult can hear that high.

They insisted on having extremely linear frequency response.  Plus/minus 3 dB produces very obvious flaws in the reproduced sound, which was simply not acceptable.  So the Red Book frequency response was specified to be better than 0.1% across the entire bandwidth.

The Red Book engineers would not accept any short-term frequency and amplitude variations.  They found the usual analog wow and flutter errors in analog were in excess of 2%.  These flaws ruined the realism of the sound and could not be accepted.  To solve this problem, the engineers used a quartz clock instead of mechanical devices to lock the frequency and amplitude to incredibly low levels -- less than 0.001%!

Finally, the THD and IM distortion of the system had to be reduced to a tiny fraction of a percent.  A typical CD player will have digital distortion well under a thousandth of a percent.  The usual limit on distortion will be the inherent distortion in the analog output buffer amplifier, which will
be far more than that produced by the digital system.

Now let's look at how the system works to solve all the above problems.   You have surely noted the two key specifications on a PCM system.  They are the sampling rate and the word length (number of bits).  Just how do these work?

Most audiophiles completely misunderstand how they work.  For example, they think that the sampling rate defines the "resolution" of the system.  They imagine that the sampling rate defines how many times the wave form is sampled in one second and that the wave form is then reconstructed during playback as discrete points.  They then further imagine that these points are connected by straight lines that form "stair steps" in the digital wave form.

Now based on this view of digital recording, it is completely understandable that these audiophiles would conclude that a digital wave form is missing information compared to the original analog wave form.   They would further assume that a higher sampling rate would provide more detail to the wave form, thereby increasing the "resolution" and accuracy of that wave form.

Their view is total fantasy.  Digital systems simply do not work that way.

In particular, the whole purpose of a DAC (Digital to Analog Converter) is to produce a perfectly smooth, complete, and accurate wave form.   There are absolutely no "stair steps" in a digital wave form.  In fact, a digital recording system produces a far more accurate wave form of the original signal than any analog system can.

If you doubt this, let me point out that if there were "stair steps" in the wave form, the distortion would measure extremely high (higher than 50% THD).  But the distortion in digital signals is vastly lower than any analog system, measuring only a couple thousandths of 1% at worst, while analog recording systems always measure several percent.

In short, a digital system produces a perfect and complete wave form.   There are no steps in it.

So if the sampling rate doesn't determine the "resolution", just what exactly does it do? Before explaining, let me point out that there is no such specification as "resolution" in audio engineering.  This is another audiophile myth.  So instead of defining resolution, the sampling rate defines the highest audio frequency that the system can capture, store, and reproduce.

In a linear PCM system, the sampling rate must be twice the highest frequency of interest.  Since the Red Book engineers wanted to reproduce  20 KHz music, they had to sample the music at twice that -- 40 KHz.

You will probably point out that Red Book CD does not sample at 40 KHz.   It samples at 44.1 KHz.  Why?

Because the digital system must not be fed any frequencies higher than its recording limit as these will confuse the system and produce a lot of distortion.  So all frequencies above  20 KHz must be eliminated by a filter.  This is called an "anti aliasing filter."

The anti-aliasing filter will require some additional bandwidth in which to operate.  By using a digital filter, the Red Book engineers were able to roll off the high frequencies at 96 dB/octave, thereby needing only 4.1 KHz of additional sampling to accommodate it.  So the sampling rate of a CD is 44.1 KHz, which makes it possible to capture the full range of human hearing.

Note that this means that in the worst case (20 KHz), the digital system will only sample the wave form twice.  So if the audiophile belief that this produced "stair steps" in the wave form, then a  KHz sine wave would actually be a square wave!

But it is easy to see that this is not true.  Feed a  KHz sine wave into a digital system like a digital signal processor or digital crossover and observe its output on an oscilloscope.  The digital
components just described will feed the analog signal through an A/D converter to digitize it, then back out through a DAC to convert it back to analog.  So the signal will have gone through a pair of digital converters.  If audiophiles were correct, you would see a square wave at the output of the DAC.  But instead you will see that the output is a perfect sine wave of vanishingly low distortion -- it will not be a square wave.

In short, linear PCM digital does not have any "missing pieces" or "stair steps" in the wave.  The entire purpose of a DAC is to reconstruct the wave completely accurately and with virtually no
distortion.  They do so magnificently.

So where does upsampling fit into this picture? To begin, let me point out the obvious, which is that one cannot add information or replace "missing pieces" of a wave form after the fact.  So upsampling cannot produce accurate musical information where none was originally recorded anymore than one can reconstruct a drop-out on analog tape.

So what is the value of upsampling?  Not much actually.  But remember the extra sampling needed to produce a sliver of bandwidth for the anti-aliasing filter I mentioned earlier?  It was only 4.1 KHz wide and required that a digital "brick wall" filter be used at 96 dB/octave.   This conserved data space on the CD.

Some audiophiles believe that such a steep filter degrades the sound -- even though the filter operates in the supersonic range, which is well above those frequencies that humans can hear.  They believe that a more gradual, analog filter will sound better.  By upsampling the data stream, they can add all the bandwidth they want and by doing so, they can use shallow analog filters.  Usually these operate at 24 dB/octave, but some are as shallow as 6 dB/octave.

Can the effect of analog anti-aliasing filters be heard?  Obviously, anything that does not produce frequencies in the range of human hearing cannot be heard.  But that doesn't keep some delusional audiophiles from believing that they can hear the effects of supersonic filters.  Once again, valid testing would conclusively demonstrate that they cannot.  But that doesn't keep CD player manufacturers from using upsampling if they think that is what audiophiles want.

Now let's look at the word length.  Red Book CDs operate using 16 bits.   Why? What effect does the number of bits have on the sound?

Simply put, the word length defines the S/N of the system.  Each bit is worth 6 dB of S/N.

As mentioned previously, one needs a S/N of at least 86 dB to produce a silent background.  Sixteen bits will produce a S/N of 96 dB, which is about 10 dB better than required.

Actually, in the real world, for many technical reasons including the need for "dither" and the fact that very few analog electronics can produce a S/N of 96 dB, most CD players only produce a S/N of about 92 dB.  But this still produces a silent background and full dynamic range and is far better than any analog recording system.

The Red Book engineers picked the number of bits required to achieve a silent background and record the full dynamic range of all music.  They did not use any more bits than necessary, nor did they include extra bits that would waste data space.  Simply put, 16 bits is the number of bits required to reproduce music perfectly (with a silent background).   I'm sure you would agree that all properly-recorded CDs have silent backgrounds.  You do not hear hiss and noise like you do with analog recordings.

So why would anybody want to use more than 16 bits? What would this gain you? Many audiophiles believe that using 24 bits will produce better recordings.  But when pressed to explain why this would be, they can't tell you.

The truth is that 24 bits will produce a digital S/N of 144 dB.  Note that I said "digital" S/N.  In reality, we can't listen to a digital signal.  We must convert it to analog to play it through a speaker.  There is no analog system that is can produce a 144 dB S/N.  About the quietest analog electronics can be is around 120 dB.

But the quietest microphones are only 92 dB due to Browning Effect.   This is the noise caused by the vibration of air molecules at room temperature striking the diaphragm of a microphone and causing it to make a small amount of noise.  So it is impossible to record music with a S/N greater than 92 dB.

A digital S/N greater than that is of no practical use when playing back music.  After all, a silent background is silent and it is impossible to make silence any quieter.  So adding bits on playback is simply a waste of data space.

Although there is no point in using more than 16 bits on playback, there is a good reason to use more than 16 bits when recording.  Understand that to get the full dynamic range of a 16 bit system, you must accurately place the dynamic range of the music in the 16 bit "window."

If you have the level too low, you won't use the entire 16 bit range and you will hear background noise on quiet passages of music.  If you have the level too high, you will exceed the maximum level defined by the 16 bits and massive distortion will be the result.

Now when playing back music, the recording engineer will always know the levels and it is a simple matter for him to place his recording correctly in a 16 bit window.  But when recording -- especially when doing live recording -- the maximum sound level is not exactly known.  So the recording levels must be conservative as exceeding the maximum digital recording level will result in massive distortion that will ruin the recording.

So for recording, it is best to have some extra headroom.  Therefore, recording studios use 20 or 24 bit recording systems.

The extra headroom provided by more bits also is useful when mixing and post-processing where equalization may be desired.  Boosting the energy at some frequencies using equalization requires more bits and might exceed the maximum digital limit.  Once the recording is made, mixed, and processed, the final product can then be accurately placed in a 16 bit window so that it has a silent background.

So for recording, or 24 bit systems make sense.  But there is absolutely no point is using more than 16 bits for playback.

This brings up the topic of "High Resolution" (Hi-Rez) audio.  Many audiophiles believe that higher sampling rates and more bits increase the "resolution" of the recording.  This is utter nonsense for all the reasons that I have outlined above.  The Red Book CD standard makes perfectly accurate recordings and increasing the sampling rate and word length doesn't not make the recording any more perfect.

The latest fads in sampling rate is to use 96 KHz or even 192 KHz sampling.  96 KHz will record sounds up to 40 KHz (80 KHz captures the 40 KHz sound and the remaining 16 KHz are used for the anti-aliasing filter).  192 KHz sampling will record 80 KHz sounds (160 KHz sampling captures 80 KHz sounds while the remaining 32 KHz are used for the anti-aliasing filter).

Now think about that.  What good does it do to record 40 KHz sounds? No music microphone records above 20 KHz, so the additional  20 KHz available in a 40 KHz recording system simply captures supersonic noise and wastes 50% of the data space.

A 192 KHz sampling system is even worse.  Fully 75% of the bandwidth is used to record supersonic noise and wastes 75% of the data space.

Of course, these higher sampling rates are also combined with longer word lengths (24 bit).  So the wasted data space is much greater than just described as one needs about 30% more data space for the extra bits.

And amazingly, these "Hi-Rez" recordings actually degrade the sound quality! This is because the supersonic noise will produce intermodulation products (beat frequencies) below the frequencies being recorded.  For example, noise frequencies at 37 KHz and 38 KHz will interact together to form intermodulation frequencies at 1 KHz, which is a frequency humans can hear.  So hi-rez recordings will actually produce noise and distortion in the audio bandwidth, which degrades the sound while making no improvement in the sound in other ways.

There is an interesting article on this subject that I think you will enjoy.  Here is a direct link to it:

I hope this discussion has helped you understand how digital recording works and that you now recognize that your previous views were inaccurate, which led you to draw false assumptions.  The fact is that modern, linear PCM digital recording at 16/44.1 sounds flawless and does not produce any steps or gaps in the musical wave form.

DSD and SACD digital formats work differently than PCM encoding.  They actually do produce steps in the musical wave form because they do not use a DAC to smooth it out.  As a result, they are extremely noisy.

To deal with this noise problem, engineers use "noise shaping" to move the noise up above the audio region so we cannot hear it.  This result is that they sound as good as a CD.  They don't measure as well as a CD, but a human cannot hear their flaws.

Because DSD and SACD do not use a DAC to produce a smooth wave form, they must sample at extremely high rates in order to make the "stair steps" in their wave forms small enough to keep distortion at a reasonably low level.  This is why the sampling rate of these formats must be several MHz.  DSD samples at 2.8 MHz and some of the newer DSD formats sample as high as 8 MHz.

Sampling at such high frequencies requires a huge amount of data storage.  Since data storage costs money, it is very unlikely that these formats will gain wide acceptance or that the major recording labels will release music on this format.  It simply makes more sense to use a DAC and PCM encoding to make recordings that are technically better and use less data than DSD.  So do not expect DSD to take over the market any more than the now defunct SACD did.

16)  Where is the fine line between, resolution, transparency and musicality?

There is no "fine line" between resolution, transparency, and musicality.  These are subjective audiophile terms that do not exist in science and engineering.  There is no such measurement as "resolution" or "musicality."  Since these terms are pure mythology, there is no "fine line" between them upon which I can comment.

17)  How do you tune your products or what is your specific goal in creating Sanders Audio in house sound?

I do not "tune" my products.  I simply make products that have the lowest noise, lowest distortion, and most linear frequency response possible.  There is no tuning involved.  One measures them and designs them for the best possible performance.

18)  What is the reference for your when designing any products?

The goal of my products is to reproduce music in the most realistic manner possible.

19)  Is there a need for a high price in high-end?

No.  There is no correlation between price and sound quality -- at least not once you reach a price point where the manufacturer does not have to compromise.   For example, a manufacturer will need to spend at least $1,000 to build a superb amplifier, and several thousand dollars to make an excellent speaker system.  To those prices one must add the cost of advertising and profits for middlemen.  But there simply is no need to spend tens of thousands of dollars on any component to make it sound superb.

20)  Sanders Audio and vinyl? Yes?

Vinyl is a very flawed recording medium.  I have produced many vinyl LPs for customers back when that was all that was available.  I refuse to compromise performance so have no use for vinyl today.  All the recordings I make now are stored on digital media because it can produce a perfect copy of my recordings while no analog system can.

Audiophiles constantly make false assumptions because they fail to do valid testing that will reveal the true cause of what they hear.  They commonly believe that digital recordings sound badly due to the digital media upon which the recordings are stored.  But the true cause of the poor sound they hear is due to the recording itself -- not the digital media, which is essentially perfect.

In other words, garbage in gets you garbage out.  Many digital recordings sound truly awful, despite the fact that the digital media is superb.  But the bad sound is not caused by the digital media -- it is caused by the poor recording stored on it.

Many vinyl recordings were made long ago before modern processing (compressors, equalizers, artificial reverb) was available.  They were recorded in quality acoustical environments and recorded in true stereo (rather than in mono using only one microphone) so they sounded very natural and realistic.  Some older recordings on LPs are really wonderful.  These recordings sound great despite the flaws and limitations of the LP storage medium.

I have thousands of LPs.  I have recorded all the music on them to one of my digital flash recorders so that I can listen to my vinyl music conveniently (LPs are a hassle to find and use).

Since digital recordings are perfect, they sound absolutely identical to the LP.  Therefore I never actually play my LPs anymore.  I listen to the digital copy instead.  This also saves my LPs from wear and tear (each time you play an LP, you damage it).  I keep my LPs in my museum rather than playing them.

I have no problem with audiophiles who enjoy old vinyl recordings.  That's great.  But PLEASE don't try to tell me that vinyl recordings are better than digital ones.  That is simply not true.

I accommodate those audiophiles who like playing LPs by having a phono stage in my preamp.   It includes all the important features like adjustable gain for moving coil or moving magnet cartridges and it has adjustable resistance and capacitance for ideal loading of cartridges.

21)  What were your explorations on of Class D?

I do not consider switch mode (Class D) amplifiers to be high fidelity devices.  This is because they do not have linear frequency response.

Their high frequency response depends on the character of the load (the loudspeaker).  Therefore they must be specifically adjusted to your specific speaker to have linear frequency response.  Since this is not practical, I do not use or recommend switch mode amps for full bandwidth speakers.

However, they usually have a lot of power and are relatively efficient.  So switch mode amps are excellent for driving woofers (which require a lot of power).  Because woofers do not reproduce high frequencies, the poor high frequency response of switch mode amps is not an issue when driving woofers.

For all these reasons, you commonly find switch mode amps used in sub woofers.  In that application, I think they are excellent.  But I will only use linear amps (Class AB or A) amplifiers for full range sound reproduction.

22)  How close can one get with digital reproduction in comparing to analog in your opinion?

As detailed previously, digital recording media is flawless.  By comparison, analog is very poor.  This really is no contest.

23)  What do you think about products coming from East (China)?

The Chinese have access to the same technology as the rest of us.  They can make excellent audio products if they decide they want to.  At this point in time, they are not doing so.  But I expect that to change in the not-to-distant future.

24)  Any plans for more affordable Sanders Audio components?

I already sell my products at the lowest price I can.  I will not compromise performance to build products to a lower price point.  I am only interested in building the best components possible and I make plenty of money doing so.  I do not need to make more money, so will not make cheap, compromised products.

25)  Many say first watt is most important. Is it?

That idea is totally absurd.  Every watt is important.  None is more important than any other.  All must be perfect if you want superb sound.  Today it is very easy to make subjectively perfect electronics.  So this topic is nonsense.

26)  Is there such a thing as too much power?

No.  An amplifier should always have excess power to assure that it never runs out of power and clips.  Any excess power will simply not be used.  There is no harm in having extra power available.

27)  Continuing from previous question. Is Solid State Pure A class an ultimate perfection?

As discussed previously, the audiophile belief that Class A is somehow better than Class AB is absurd.  It was only true a half century ago.  There is no difference in performance between properly designed Class A and AB amplifiers today.

Solid state equipment simply has better performance than tubes.  So SS, Class AB amplifiers are the best option available -- which is why most audiophile amplifiers made today are of that type.

28)  How important are the room acoustics from your experiences?

Room acoustics are critically important.  They are the second most important factor in your audio system (the first is your loudspeakers).

This topic urgently needs to be addressed by all audiophiles, because no matter how good your loudspeakers are, they will interact with your room, which will degrade the sound.  This topic is so important that I have written a White Paper on it.  Please refer to the Technical White Papers page on my website and read the topic called "Acoustics."  It will explain room issues and give practical advice on how to deal with them.

29)  There is quite few DSP and room correction solutions, but they seems to add too much to the music. What are your experiences?

DSP systems are extremely powerful and effective tools.  They can improve all audio systems and every audiophile should use them.

But like any powerful tool, they are often used incorrectly, which can degrade the sound.  When you say that "they seem to add too much to the music", that means that you have not heard a DSP set up properly.  If you did, you would quickly recognize that they improve the sound substantially.

30)  When is it simple enough (design and topology) to say stop, this is where it all ends and amplifier is finished?

Modern, solid state, Class AB, high-power amplifiers do a splendid job.  They have inaudible noise, inaudible distortion, linear frequency response, and sufficient power.  It is fair to say that no practical improvement is needed.  So yes, we can say that amplifiers are now good enough that there is no need for improvement.

However, almost audiophiles are using underpowered amplifiers.  Therefore, their amplifiers are clipping the musical peaks, which makes them perform poorly and degrades the sound.  It is easy to demonstrate that most audiophile systems require at least 500 watts per channel to play dynamic music cleanly at the live levels most audiophiles enjoy.

Using amplifiers of inadequate power simply guarantees poor performance.  I do not understand why audiophiles use under-powered amplifiers.  But as long as they do, then their amplifiers will always sound poorly.  Under these conditions, it is fair to say that better amplifiers are needed.  But what is actually needed is just more power, not new amplifier technology.

31)  What sets Sanders Audio from competition?

Three things.  First, I do not lie to my customers to make a sale.  I tell the truth about equipment and if that means that I lose a sale, I accept that.  Fortunately, I make sufficient money, so I do not need to lie and cheat my customers and can afford to tell them the truth.

Secondly, I refuse to sell "snake oil."  By that, I mean that I won't sell equipment that isn't necessary or that doesn't actually make an improvement in the sound.   I only sell equipment that really makes a difference in sound quality.

Thirdly, I charge very low prices for my equipment.  I only sell components that offer the very best value for the money.  Some audiophiles don't understand that and think that because I don't charge absurdly high prices that my equipment isn't as good as the more expensive stuff.  But my equipment is the best you can buy, regardless of price.  So you can spend more money, but you can't get more performance than what I offer.

32)  Do you ever plan to bring any other components to your line except the amplifiers, preamps and speakers?

Probably not.  I already make everything needed for an audio system except the source component, which today need only be a computer, since the future of music is on-line.  Since there are plenty of computers available that work splendidly, why should I make one?

Also, my business is very successful and I am totally busy trying to keep up with demand.  So I don't need not add products to my line to make a profit.  Frankly, I just don't have any more time to design and make more products.

33)  What would you say, that is the secret of Sanders Audio success?

I sell the finest performing products at reasonable prices.  I also do not use conventional sales, marketing, and distribution.  Instead, I sell factory-direct everywhere in the world and take care of my customers directly from my factory.  This cuts out dealers and middlemen who cost a great deal thereby making it possible for me to reduce my prices dramatically.  Do you know that dealers get 50% of the sale price?  I would therefore have to double my prices if I sold through dealers.

There are far too few audio dealers to serve everyone.  By eliminating dealers, I can serve everybody -- not just those who happen to have a dealer nearby.  As a result, I can sell to everybody in the world and therefore sell a lot of equipment.

34)  How do you see the state of present high-end society?

It is a disaster.  Everything you hear in the high end audiophile world is false.  This is easy to prove if you do measurements and valid listening tests.

The high end audio industry is now focused totally on making money by tricking gullible audiophiles into parting with their money to buy worthless and overpriced equipment.  For example, power conditioners do not do anything that the power supply in any component does not already do.  So they cannot and do not improve the sound.  Yet the high end audio industry has convinced most audiophiles that a power conditioner is necessary to get great sound.  This is a hoax.

Worse yet, while power conditioners do not improve the sound, they can degrade it.  This is because they insert series resistance between the mains and powerful amplifiers.  As a result, a powerful amplifier will have its current restricted, which will cause it to clip at a lower power level than it would if you just plugged it directly into the mains.  So power conditioners should not be used -- yet the audio industry manages to convince audiophiles to buy them.

I could go on with endless examples (like fancy power cords, expensive fuses, outboard DACs, etc.).  All are unnecessary and don't improve the sound.  But the audiophile industry refuses to tell the truth and continues to sell such products.

35)  And what about High-end magazines?

The high-end magazines are accomplices in this fraud.  They should be protecting audiophiles by educating them, revealing the truth, doing valid testing, and informing audiophiles of the deceit in the industry.  But instead, the magazines are prostituting themselves to the money the manufacturers give to them in the form of advertising.

36)  Do you believe in traditional High-end audio magazine reviews?

No.  Subjective magazine reviews are rubbish.  The only reviews that are meaningful are those that contain engineering measurements.  Measurements will reveal everything there is to hear in electronics (but not loudspeakers).  Unfortunately, few magazines measure equipment nowadays.
There is no magic.  There is nothing that a human can hear that an instrument cannot measure.  In fact, instruments like a distortion analyzer can reveal far more about a component than what a human can hear.

For example, humans cannot hear distortion of less than 1%.  But a spectrum analyzer can show distortion and all its harmonic components down to less than 0.0001% (one ten-thousandth of one percent distortion).  That is vastly better than a human can hear.

Note carefully that science has not yet figured out how to measure everything about speakers -- only everything about electronics.  For example, no instrument can measure loudspeaker imaging.  This is because the image is formed in our brains, not directly from the speaker.

Instruments can measure things that affect imaging (like phasing, dispersion, and frequency response), but they cannot directly measure imaging.  So one must still listen to a speaker to fully evaluate it.  But this is not true of electronics, which can be fully evaluated by instruments alone.

37)  Advertising is the king? Sad and true?

Advertising isn't king -- money is.  Usually that involves advertising.  Unfortunately, some form of advertising is required for a company to become known.  After all, you can have the best products in the world, but if nobody knows about them, then what good are they?

The good news is that the internet exists.  The best advertising is word of mouth.  With the invention of the internet and audio forums, satisfied customers can now tell the world.  Most of my sales are generated on-line through forums.  This type of advertising is free.

38)  I like to think, that serious audiophiles are more than intelligent when spending the money. How do you see this?

I disagree.  It is apparent that most audiophiles are extremely gullible and make very poor decisions about spending their money.  They fail to demand proof of the claims made by the snake oil salesmen, so they don't know the truth or whom to believe.  They therefore get tricked into purchasing over priced and/or worthless or poor-performing equipment.  Considering the high cost of high-end equipment, I think this situation is insane.

The audio press should be protecting audiophiles, but since the press does not, audiophiles need to do valid testing so that they can learn the truth.  This can be done with measurements or well-controlled listening tests.

Snake oil salesmen have convinced audiophiles to ignore measurements and instead "believe what they hear."   Audiophiles who buy into this then must do valid listening tests.  But they don't.  Let me explain:

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 really 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.  These need to be answered with certainty before an audiophile spends thousands of dollars on expensive audio equipment.

But there is something very strange about high end audio.  Although sound reproduction 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, misleading, 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 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 generally is not available.  This information can be obtained by objective measurements by instruments and by valid listening tests.  Unfortunately, unscrupulous businessmen in the audio industry have 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 and conflicting opinions 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 fundamentally linked to the number of megapixels in its CCD.  So the specifications and measurements of the number of pixels is accepted as an important factor when choosing a camera.  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 audiophiles have been told that such measurements are not to be trusted.

But the results of most audiophile subjective testing is variable and uncertain.  So 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 have multiple, uncontrolled variables in them.  Therefore, there is no way to know what is causing the differences in sound that is 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 what you heard.   Therefore, having multiple uncontrolled variables makes it impossible to draw any cause/effect relationships and conclusions from the test.  Since our listening tests are trying to find cause/effect relationships, a test done with multiple, uncontrolled variables can't answer the question, so it is useless and invalid.

Let me be very clear about this 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 could not be used to determine which amplifier sounds better.

Now this is not new information.  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 amplifier test previously described 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 to us.  That is why we like to listen to our music loudly.

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

In any case, the amps probably would have had different loudness levels even if the preamp setting was identical.  This is because amplifiers have different gain and power levels.

Note that 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 understand that two identical components will sound different if one simply plays a little louder than the other.  The louder one will sound better to us even if the two actually sound identical.

This is a serious problem in listening tests.  Consider the amplifier test above and for purposes of this discussion, 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.

The audiophile will then draw the conclusion that the new amp is better and will spend $10,000 to buy it.  But in fact, the new amp didn't really 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 new amp sounding better.   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", 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 the amplifiers immediately.

Equally important is that you should make many comparisons between the components 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 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, so it did.  It was especially easy for his bias to influence him due to the time delay involved as he changed cables.

That bias may have been because he expects tubes to sound better (or worse) than transistors, or that the new amp 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 the performance that an audiophile perceives.  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 it.

4) CLIPPING.  Clipping is when an amplifier is being driven beyond its power and voltage abilities.  This produces massive amounts of distortion, compression of the dynamic range, loss of clarity and detail, a sense of strain, harshness, and generally bad performance.

It doesn't matter what good 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.  The amp that is clipping will sound worse than the one that isn't.   But you must not test a clipping amp (that is grossly misbehaving) to one that isn't clipping (and is performing well).  That is not a valid test at all and doesn't tell you how an amp sounds when it is performing properly and within its design parameters.

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, and does not occur at the average power level.   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 amp that is clipping peaks as sounding "dull" (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 simple amplifier 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 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 with far less power.  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 bias to strongly influence what we hear and make it difficult to recognize level differences.  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 determine how the two components sound.

But it doesn't have to be that way.  It is possible to control 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 their 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 to produce a steady test tone?  You can buy a dedicated one for around $50 on eBay.  Or you can download one for free as software for your laptop computer at this link:

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.   If you want to compare preamps, CD players, DACs, or other line-level components, then you need a suitable switch and connectors for them.

Attenuators should be placed on the box so you can adjust levels on amplifiers and line level components.   Of course, the box will have both input and output connectors for the amplifiers and other types of components so that they can simply be plugged into the box for testing.

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

3) PSYCHOLOGICAL BIAS must be eliminated by doing the test blind.  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 makes it possible for listeners 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 this will force listeners to be accurate and only indicate differences in sound when they are indeed present.

This is not a trick.  It is done to assure accuracy and meaningful results.  Listeners should be 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.  Only when no clipping is present can you proceed with the test.

Need an oscilloscope? You can easily find a good used one on eBay for around $100.  There is software you can use on your computer to turn it into a 'scope.

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.

But 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 important to give them a little training.  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.

Because a good test will involve many comparisons, it is helpful to use a score sheet for listening groups.  The sheet has a check box for "different" and "same" that they check after each comparison.  You can then use a master sheet that shows where differences are possible (A-B test) and where they are not (A-X or B-X).  It is then easy to 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 can use a score sheet and you can only do A-B testing.  This type of testing isn't well controlled (although it is vastly better than open loop testing), but you can usually get a good idea of what to components sound like.   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.  They can go back and listen to the same section of music over and over if they wish.

There are no tricks involved.  This is science and 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 on slow, sustained music.  For example, it is harder to hear differences on pop music than on lyrical piano music with long tones.

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 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 fun.  I find that they are just as good or better than their audiophile husbands at hearing differences.

If you find differences, you can then explore their cause by being a bit more creative in designing the test.  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.

Therefore you will need 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 monaural, but that 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.

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 both components have attenuators, they would be affecting both components under test equally.  So they 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, you will find many 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 a significant amount of equipment fails to meet BQC on instrument testing.

The BQC are:

1) Inaudible noise levels (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 (20 Hz - 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 if this is so, 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 need to ABX test it to know that.

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 (and many do), 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 well-controlled listening test and prove it to yourself.

Let me stress a very important point.  Many audiophiles immediately become defensive and think that what I just said is that all audio equipment sounds identical.  NOTHING COULD BE FURTHER FROM THE TRUTH!.  Of course many components sound different from each other.

But they think I said that all components sound the same (which is untrue).  They have heard differences between such components, so immediately disregard the whole idea of ABX testing.  This is a tragedy.

I said that components "that meet the BQC sound identical."  This is true.  They do.  But a great many components do not meet the BQC, so do NOT sound identical.

The point of doing controlled testing is to find out what is causing the differences in sound that is heard.  I am not saying that audiophiles are deaf.  I am trying to help them understand how to do testing that will show the true causes of the differences they hear between components.

Valid testing requires that you apply some basic scientific principles to the task.  Science is not incompatible with the audiophile world.  In fact, it is an essential and very helpful tool in finding out the facts and determining what is causing the differences we all hear between components.

So don't dismiss science.  After all, it is science and engineering that provided you with the components you now enjoy.  There is no magic and magicians don't design audio equipment -- engineers do.

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.  Unless you are testing very powerful amplifiers that can deliver hundreds of watts per channel, you will find that you 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.

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?

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, the mistaken believe that quality test equipment is very expensive, and because audiophiles are often told that measurements are not to be trusted.  All this has changed with the development of the 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 others will not.

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

Those frequencies that are not harmonically related are noise.  They are random, present at all frequencies, and hopefully should be at a 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 get it fixed.

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 (the peak of the test tone).

The analyzer will combine and compute all the harmonics into THD (Total Harmonic Distortion). This will be a negative number of dB below the reference tone or distortion percentage of the test tone.  You can chose which you like to use.

The two are related.  For example, if the distortion is 100 dB below the reference level, it will also be 0.001% distortion.

The spectrum 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, 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, I have attached two photos of my spectrum analyzer showing the performance of one of my Magtech amplifiers.  The first one shows the distortion when there is insufficient bias to completely eliminate crossover distortion.  You see the first 20 harmonics and their levels in small boxes above each harmonic (although it is hard to read the values in the pictures).  The large blue box at the top of the screen shows the THD, which in this case is about a half of one percent.

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 4th harmonic have disappeared into the noise floor -- they simply don't exist.  The second harmonic is the highest, and even it is 99.6 dB below the reference tone.  The 3rd harmonic is -102.4 dB, the 4th harmonic is -110 dB and the remainder are unmeasurable.  The THD is incredibly low at around one thousandth of one percent.

The amp is being tested under the worst condition, which is at 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.

Also, the S/N will be much lower 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.  But even at this very low power level you see that the noise floor is about 118 dB below the reference signal.  This is a very quiet amplifier.

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 is a myth.

Because the spectrum shows each harmonic, you can evaluate the component for its harmonic structure.  For example, many audiophiles have come to believe that tubes 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 (7th, 9th, 11th, etc.).  Low order harmonics sound less objectionable than high order, odd harmonics, so the theory goes that tube equipment should sound better than solid state.

I haven't found this to always be true.  Some tube equipment has lots of high order harmonics and some SS gear has more 2nd and 3rd harmonic structure.  It varies between amplifiers so it is inaccurate to make a blanket statement about this.

Refer again to the spectrum of the Magtech shown above.  Its 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, the distortion is far too quiet for a human to hear, so the point is moot.

The spectrum analyzer can plot the frequency response of a component.   It can measure wow and flutter and you can also see any wow and flutter that might be present by instability in the graph.

So you can see that a spectrum analyzer will tell you an enormous amount about your equipment. It is shockingly better than the human ear.  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 be less than 1% to be inaudible.

But 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.  This is far more sensitive than human hearing.

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 adjusting the 5th harmonic to -110 dB.

I think it is quite interesting that human hearing can't hear the difference between the two spectrum analyzer graphs shown above, while there is a great deal of difference in performance of the amplifier.  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 many tens of percent distortion.

In summary, 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.

I hope this discussion has convinced you that open loop listening tests are useless unless components sound grossly different.  For example, you don't need to do an ABX test to detect differences between a 5 watt, SET tube amplifier and a 500 watt solid state amplifier.

But for the typical audiophile test where subtle differences are the norm, 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.

But a spectrum analyzer will tell all there is to know about the performance of electronics.  These are cheap, easy to use, 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?

Both spectrum analyzers and blind, ABX, listening tests are accurate and will reveal the true causes of any differences you may hear between components.  By using these tests you can determine with accuracy the quality of the components you are using or want to buy.  You will then have total freedom and confidence as you know the truth and can ignore all the controversy and confusion that is the bane of the audiophile industry.

39)  Would you say that high quality is more affordable today or you have to pay premium price for best components and sound?

There is no correlation between price and sound quality once the price has risen to the point where the manufacturer doesn't have to compromise performance for price.  You do not have to pay very high prices to get great sound.  Generally the extremely high priced equipment is not as good as moderately priced gear.

40)  What is the definitive goal of Sanders Audio?

My purpose is to bring the joy of beautiful music to as many people as I can.  To do so, I must make the very best performing equipment and teach audiophiles enough about audio so that they will get and use it.

41)  What do you see the future of audio and where do you see your roll in it?

The future of audio appears to be dismal.  The snake oil salesmen seem to have taken over the industry and they keep coming up with yet more silly ideas and components to sell to unsuspecting audiophiles.  The situation is getting worse and worse.   I do not see any end to it.

My roll is to tell the truth and educate audiophiles by spreading information like I am doing here.  But I am only one voice in the wilderness.  I cannot change the industry.  But I will do my best to help inform as many audiophiles as I can get to listen.

42)  Who are your musical inspirations?

I am a trained, classical musician.  I love the great symphonies and operas.  I perform in various orchestras.  I also like big band jazz and have played in several bands.

43)  Is it possible to achieve the same feeling with high-end audio as in concert? Is this one of the goals of Sanders Audio?

Audio systems cannot exactly reproduce the sound of a live concert.  They can be quite good if the recording is done really well.  But there is simply no substitute for actually experiencing a live concert.  Even better is being one of the musicians in an orchestra.

However, this does not apply to most modern music.  This is because when you go to one of today's modern concerts of pop music, what you are actually listening to is a big PA system -- and public address systems are actually just very large, poor quality, audio systems.  You can get better sound through your home audio system.  So to get a true concert experience, you must listen to an acoustic orchestra.  But if you do, no audio system can match it.

44)  How to remove mystics from audio?

Educate them.

45)  What is the difference between audiophile and music lover?

There are a variety of different types of audiophiles, so you cannot lump them all together into one group.  For example, there are audiophiles who love music.  They are really music lovers who have had to become involved in the audiophile world to figure out how to obtain a good audio system.

Then there are the audio hobbyists who are more interested in the hardware than the music.  I would call them technology geeks rather than music lovers.  They mostly listen to music in order to test their audio equipment.

Then there are the audiophiles who are "audiophools."  These are the highly opinionated, close-minded audiophiles who view sound more like religion.  They take extreme stands on things like tubes or vinyl and will defend their opinion that these devices are the best and everything else is awful.  They consider anybody who doesn't see it their way as an idiot.

46)  What do music lover get when buying Sanders Audio products?

They will get closer to the original performance than with any other equipment.  They will get this level of performance at a fair and reasonable price.

47)  What is your reference when designing and testing new products?

The sound of a live performance is what I am trying to reproduce.  But the reference used to achieve that are engineering measurements and specifications.  For example, I make all my products to have linear frequency response, low distortion, low noise, and perfect phasing.  All this is measurable and very important because a component that meets these criteria will sound superb.

48)  Who would you say typical Sanders Audio customers are?

I have no "typical" customers.  I have many different types of customers.  For example, in addition to audiophiles, I also supply recording studios, scientific laboratories, musicians, universities, music schools, industrial applications, and even children's hospitals who need non-magnetic speakers for their electroencephalographic brain measurement labs.   But the majority of my customers are those who want to get the greatest enjoyment they can from their music.

49)  Any last thoughts for our readers?

Do not believe anything you hear in the audiophile world.  It is all false.

Especially do not believe unproven claims you hear on the internet or from manufacturers and dealers.  While some of these sources may be sincere, most are just repeating the lies they have been told by others.  Demand valid, scientific proof of their claims before parting with your money.

Any subjective listening test that does not use a switch box, blind listening, and matched levels is invalid and worthless.   Never believe the results of uncontrolled listening tests.  Since the audio press does not do valid testing, you will need to do your own testing to learn the truth.

Finally, when choosing audio equipment, focus on the three factors that will actually determine the sound quality of your audio system.  Note that since all modern electronics are essentially perfect, you need not be concerned about them -- as long as you use adequate amplifier power.  What really matters are loudspeakers, rooms, and source material:

1)  Loudspeakers are the most important component in your system.  All are seriously flawed.  You should put most of your money and effort into getting the best ones you can.

2)  Rooms interact with loudspeakers to seriously degrade the sound.  It is essential to deal with this problem using proper positioning, room treatment, and DSP.

3)  Source material is critical.  Garbage in gets you garbage out.  Finding well-recorded music is very difficult.  There is very little of it.  But you must search for it to get excellent sound.

Above all, enjoy your music!

-Roger Sanders

Matej Isak. Mono and Stereo ultra high end audio magazine. All rights reserved. 2006-2013. ..:: None of the original text, pictures, that were taken by me, links or my original files can be re-printed or used in any way without prior permission! ::..