THE QUEST FOR AUDIO NIRVANA

At the urging of many friends and associates over many, many years, I have finally decided to write this series of 3 articles covering a wide range of audio subjects. I intend to bare some truths and debunk many myths that have been around for too long. For sure, these writings will undoubtedly unleash a lot of fervor by smashing some belief systems that have been awry for too long as well as hopefully shedding some light in areas that have been either obscured or just plain ignored. Ever since the beginning of the stereo era, the audio industry has been on a zig-zag course instead of heading for the truth. To be sure, there have been, and probably always will be, far too many charlatans, dilettantes, con artists, and incompetents in our industry which should be based much more on scientific truth and a whole lot less mysticism. Starting here I have prepared 3 articles of which the first is on electronics and the 2nd and 3rd are on recording and playback respectfully. So without further adieu, let the slings and arrows fly. SEARCHING FOR THE HOLY AUDIO GRAIL Audio is the only industry known to man that requires little or no scholarship to be a practitioner. At times it seems that any totally incompetent quack working in their garage can come up with a product and get it to market without any qualifications whatsoever. The only requirement seems to be a belief system. Unfortunately, belief systems dont cut it because thats all they arejust belief systems. They dont have to be right and most of the time they arent. Of late, we have a very polarized academy in the world of audio. On one side are the subjectivists who tend to cast a suspicious eye on scientific doctrine. On the other side are the objectivists who tend to believe that science is the only answer and that the so called golden eared types are living in la-la land. Both of these groups are, to be sure, lacking in comprehension of a great deal of information regarding the bigger picture of audio. I, of course, have always taken the middle road inasmuch as I use scientific procedure to obtain results BUT am always concerned about how the final product performs in its assigned task. In other words, I know that there is correlation between the two even if we have not yet defined all of the parameters. So lets start with some truths. There is no such thing as good distortion. Period. End of argument. All distortion is bad and thats final. Of late, some have had the misguided belief that if we go back to the horse and buggy days of the mid 1920s with 8 watt single ended tube amps with no feedback and higher levels of low order distortion, then all of our sonic prayers will be answered. This is just plain nonsense. Are people to believe that all of the last 75 years of scientific progress is just a bunch of hogwash? I think not. I think the descriptive word better should be replaced with the word different, when discussing the sonic attributes of these products. Does anyone know a single person that was alive in 1925 that can state

that back then these things were high fidelity? Can you imagine the auxiliary equipment that was used then such as electromagnetic speakers and solid iron needles, etc. I could go on, but I wont. Anything and everything that is wrong with an electronic or acoustic device can be classified as distortion. In no particular order these are amplitude distortions, frequency distortions, time distortions, phase distortions, power supply anomalies, flux radiation distortions, etc. One of the things that has yet never been done is to establish any kind of weighting to all of these various kinds of distortions. In addition, it is known that the range of human hearing is in excess of 130dBs. Therefore, to state for example, that an amplifier has distortion low enough that it is below the threshold of hearing is just plain wrong. The human ear is an incredibly sensitive instrument and actually acts like a spectrum analyzer with much more dynamic range. Signals, that are buried in noise for example, can be heard by the ear although we cannot consciously identify them as such. There is one other kind of distortion that may be the most insidious of all and that is convolution distortion. This occurs when you have a string of devices, starting with the microphone, followed by whatever multitude of stages in the recording console, followed by the stylus and or laser, followed by more processors and/or an RIAA stage, followed by a line stage, followed by the power amplifier, and finally ending with the loudspeaker. Every one of these devices adds its own intermodulation products, and then in series gets convoluted all on top of each other at every step of the way down the chain. By the end what you have is an absolute mess. UGH! After all is said and done, this may be the unfortunate final frontier in that if every device in our playback system were perfect, we would still have all of the convolution products that preceded it. At this point some of you may be wondering about basic noise floors and how is it possible that masking is not taking place. The reason is that noise, as long as it is low enough not to be consciously intruding, gets basically rejected by the ear. This is so because as stated earlier, the ear-brain link acts like a real time spectrum analyzer. Since noise is fundamentally random and has no correlation to the desired signals, it is ignored by the ear. However, all of the grunge lying below the noise can have a tremendous effect on what we perceive acoustically. Before we go on, lets talk about some numbers so that we have a basis to work with concerning distortion. Virtually everyone relates to distortion numbers incorrectly. For example, if an amplifier had say 1% THD as read on a distortion meter, is it really one percent? Actually, it isnt. The distortion amounts that are read on a distortion analyzer are in Volts and the ear doesnt hear Volts, it hears power. So, if we convert the 1% to power we get a totally different perspective. Lets say a nominal 200 watt per channel amp is producing 1% THD. As read on the analyzer this would be 40dBs. But 40dBs in power is 1/10,000th or 20 milliwatts. On the other hand, since the ears hearing curve is logarithmic, 40dBs represents a RATIO of only 16 to 1. In other words, for every doubling (or halving) of the loudness it takes 10 TIMES (or 1/10th) the power. 20dBs equals 4 times (or 1/4th) and 30dBs

equals 8 times (or 1/8th) and 40dBs equals 16 times (or 1/16th). Therefore, if you divide 100 by 16 you get, you guessed it, 6.67%. Now thats a long ways from 1%. You can scale this up or down in either direction and the RATIO of the results will be the same. So when you apply these true numbers to the human hearing curve it is easy to see why anomalies and distortion artifacts that are buried in the noise can be detected by the ear. There is no black magic here. The so-called golden eared types actually can hear this stuff. The descriptions and verbiage associated with this situation may be a whole different can of worms. I have prepared a chart that shows the true relativity of all these factors.

THD

Dbs

Distortion

Distortion

Hearing

True

Reading

Power

Watts

Ratio

Distortion

10%

-20

1%

4:1

25%

1%

-40

.01%

20mW

16:1

6.67%

.1%

-60

.0001%

200uW

64:1

1.56%

.01%

-80

.000001%

2uW

256:1

.39%

.001%

-100

.00000001%

20nW

1024:1

.098%

.0001%

-120

10(-12)

200pW

4096:1

.024%

You will immediately notice the staggering difference between what the distortion meter is reading verses the true distortion numbers. The first line shows that the true distortion is actually 2 times greater. The second line shows that the distortion is actually 6.67 times greater. The third line shows

that the true distortion is really 15.6 times greater. The fourth line shows that the true distortion is actually 390 times greater. And the fifth line shows that the true distortion is actually almost 1000 TIMES greater, etc. It should be quite obvious to everyone now as to the fact that humans CAN hear artifacts down in the mud. Quite simply put, it is the RATIO of loudness factors that determine what we perceive acoustically and not the THD numbers off of the distortion analyzer. As an aside to this, I have said for years that phono cartridge separation numbers are a mixed blessing. In other words, 30dBs is just not enough as this only amounts to a ratio of 8:1 loudness. Thats pretty awful when compared to todays CDs for example. However, it is a mixed blessing in that this lower separation actually helps to fill in the middle somewhat. It is however, quite amazing to listen to the difference between two otherwise identical cartridges wherein one has 30dBs of separation and the other has 40dBs. The difference is absolutely obvious. Unfortunately it is no mean feat to manufacture cartridges with such a consistent high level of separation but this should be the number one main concern all other factors equal. At this juncture I would like to digress for a moment into a little musical escapade. There are those out there that believe that harmonic distortion is meaningless. These persons are, of course, idiots. The following shall prove my point. Firstly, some musical tidbits are in order. In natures natural world, the most complex musical structure, no matter how dissonant or polytonal, can only exist within a span of 2 octaves that is, say from C to the second octave C. In other words, 15 natural notes. It doesnt matter how complicated the harmonic structure is. It still can only occur within 2 octaves. Any notes that are outside, either above or below these two octaves, are merely repeats. For example, the third D above our starting C is merely the II note, etc. Any notes below the C are called suspensions. Any notes above are called octaves. Secondly, human hearing is leading tone sensitive. This is an extremely important human recognition ability. It allows us to always hear the melody on top no matter how complex the structure below the melody is. You may ask, why is this so important. It is important because in nature, the musical scale is tempered. Im sure you have heard the expression the tempered scale regarding the tuning of instruments. Whats happening here is that the frequencies of the notes are not mathematically perfect hence tempered. As the musical notes go up in frequency, the pitch is ever so minutely bending sharp. In musical language we call this cents bending. Let me give you a couple of examples. In the early days of manufacturing electronic organs, they had 12 master oscillators which were divided down by flip-flops to produce the succeeding lower octaves. Unfortunately, these divisions were mathematically perfect and not tempered. The effect of this is that you could not have one of these electronic organs work with other musical instruments that were tuned according to the tempered scale. It was an absolute definite clash. In my early career, I was working in a club once when a female vocalist came in and wanted to sing with my duo (organ and drums). She had never sang with an organ before and for a whole set she sang everything SHARP. I

heard this and she heard this and the audience heard this. Not pleasant. Later in my musical career when I switched from organ to piano I had to re-acclimate my voice because I was singing everything FLAT. It was an experience to say the least. It took me a couple of months to get my ear-brain-voice linkage working properly. Today of course, virtually all electronic organs are made with a DSP processor which produces EVERY note exactly according to the tempered scale so that problem doesnt exist anymore. What am I getting at here? That third C in our previous example is the XV note or the fourth harmonic. It is the highest harmonic that if it were distortion would still be tolerated by the ear because it is enharmonic within the two octave natural order. The third harmonic is the 2 nd octave G which is the XII note or the perfect 5th raised an octave. Now lets talk about the next octave, the third. This is what I call the killer octave which shall become painfully obvious as we continue. The fourth C is the 8th harmonic. Now then, the third octave contains all the harmonics between the 4th and the 8th!! That is the 4th, 5th, 6th, 7th, and 8th. All clustered together within one octave. WOW. How many of you readers have ever thought about that or realized that. Ill bet that very few engineers ever thought about this situation. Distortion products created by say our amplifier, are NOT tempered. Therefore, all of those harmonic distortion products, that is the 5th through the 8th, will definitely clash with the natural harmonics created by the real music. And all of this clashing happens within ONE octave namely, the third. There is absolutely no question that the ear can hear this discumbobulation. I believe that this is one of the reasons for the so-called transistor sound. Im sure that the editor doesnt want an engineering course to be presented here. It should therefore, be obvious to everyone that this is a major problem that needs to be addressed. I have always stated throughout my career that I am more concerned with the character of the distortion performance rather than just the numbers. It is incredibly difficult if not virtually impossible to make the distortion products behave in a declining manner as the frequency goes up. This is obvious because we are running out of bandwidth and consequently losing feedback at the same time. Coupled with the fact that under most circumstances of solid state engineering over the last 40 years, we have not, for the most part, yet learned how to treat semiconductor devices in the natural manner of which they are comfortable operating wise. Another equally difficult task is to try to design circuitry that operates in the current mode as opposed to the voltage mode. Initially, this can be done rudimentarily if one also accepts throwing bandwidth and noise right out the window. Fundamentally, tubes do not exhibit the same kind of mechanisms that solid state devices have and therefore do not have the same kinds of problems. In the last few years, some extremely powerful simulation programs have become available that can now allow us to delve deep into the problems of circuit design to the degree that was simply just not possible with normal test equipment. Im talking about being able to go down into the mud so to speak

nominally, to 180dBs of range. The very best analytical test equipment on earth cant, at present, go much below about 120dBs or so. There are some special painstaking methods that can be used but they take forever to get results wherein the same results, and more accurate results, can be achieved with simulation in a matter of minutes. One of the most formidable tools is from Electronics Workbench and is called Multisim. Using these tools has allowed us to devise completely new circuit philosophies that have distortion products that are 1000 times lower than what we have previously achieved as well as having bandwidth much greater than we have had in the past. Of course it will be a few years before most of these concepts become a reality and show up in products, but the writing is on the wall and we can definitely see the future. Finally, as far as correlation is concerned, we dont arrive at answers until we get there. Therefore, all these new revelations that we uncover must be carefully evaluated lest we end up chasing the proverbial ghosts. Next are a few more audio tidbits for you to ponder. So you think that your 200 watt amplifier is really 200 watts, eh! Guess what folks: it isnt. Actually, its far from it. Let me explain. It would only be 200 watts when producing a SINGLE NOTE. Now if you add a second note, it isnt 200 watts anymore. Some examples are in order. Lets say a note is being played that is 2 octaves above concert A or about 1760 Hz. Now, 200 watts is represented by 40 Volts RMS across 8 ohms. Lets assume that the voltage for that note is required to be HALF of the available output or 20 Volts. This 20 Volts is 50 watts. Now lets have another note at say low A at 55 Hz. also at 20 Volts or 50 watts. Guess what! The voltages are directly additive and in fact, the 1760 Hz is in effect modulated by the 55 Hz. Now the average sum of these two signals equals 40 Volts which should be 200 watts but, it isnt. Each frequency is only amounting to 50 watts for a total of 100 watts. Already, our 200 watt amplifier has been reduced to half its size with only TWO NOTES. Now lets add in a 3rd note sufficiently away from the other two notes. We must now divide the available output voltage by a factor of 3 which gives us 13.33 Volts per signal. This results in approximately 22 watts per note. That sure is a long way from 200 watts. Now can you imagine what happens with very complex musical structures? The true available UNDISTORTED output power is but a mere fraction of what the amplifier is really rated at. If you dont believe me just hook up an oscilliscope to the output of your amp and watch the clipping. Those of you with amps that have true peak indicators can surely know what Im talking about. Lastly, lets examine a situation that will become increasingly important as we develop new circuit concepts and ideas. For most of the last many decades in audio, the standard topology for power amplifiers has been the half bridge. In other words, we have a hot output and a ground. As we delve into the depths of trying to improve performance, this topology has a built in automatic brick wall that is virtually impossible to get through. Im talking about the ground. In this type of circuit all of the signal power on alternate half cycles must pass through ground and up through the transformer windings and

through the rectifiers and finally completing the loop through the filter capacitors. It is absolutely impossible to make and guarantee the integrity of the center tap on the transformer to better than 120dBs. What this means is that in this concept the bottom line as far as power supply anomaly distortion will hit this brick wall. The ONLY way that we can get around this problem is by using a full balanced bridge design. The reason is quite simple. In a balanced bridge circuit all of the signals are circulating and NO POWER goes through ground. Ground is merely and only a reference. All of the anomalies previously mentioned actually cancel out provided that the design is done properly. In other words, in our search to improve performance beyond the 120dB range, this is the first major hurdle that must be overcome. There have not been a lot of balanced bridge amps manufactured but it is the only way to go in order to push the envelope forward. One other note of caution is in order. I am ONLY referring to power amps here because we are dealing with just that power. I am NOT advocating balanced circuits for any other audio components and as a matter of fact, trying to utilize balanced lines in home audio is a stupid idea to say the least. There is absolutely NO ADVANTAGE to using balanced lines and as a matter of fact, there are some very serious drawbacks. I will explain this at a later time however, Walt Jung wrote a very good and definitive couple of articles a few years ago in one of the major electronic trade magazines of which the exact one escapes me at the moment. He really nailed the rationale. PART TWO In the first installment, I discussed some aspects of electronics relating to matters audio. In this second part I will divulge a few unknown or a least obscure things relating to the recording process. Firstly, I would like to reiterate a previous statement: Stereo DOES NOT exist in the real natural world. I really want to drive this point home. All sounds in nature are created in MONO. However, as everyone knows, mono has no depth, no dimension, and no presence. Therefore, stereo becomes a necessary evil that at least provides us with some semblance of the foregoing. It has been up to our ear-brain link to fill in all the gaps of missing or out of kilter information. Even though stereo itself is a lousy format that we have been living with for some 45 years, it didnt have to be that way. There has been a mathematical derivation that could have been provided for us right from the beginning had the engineering community been smart about the whole process. Unfortunately, it wasnt to be but, that wont be for long. Anyway, let me define for you what I mean by the words dimension and presence. Dimension is a quality that has 3 derivatives. First is the main aspect of stereo which encompasses the lateral or left to right dimension. The second aspect is the front to back dimension. The third derivative is the one that is missing which is the vertical dimension. For the most part we live on a flat earth wherein alll of our lives and the events in our lives occur in the lateral plane. Also, our ears obviously are on the sides of our head and not on the top of our head and on our chin. Our human

perception is much more acute in the lateral plane than in the vertical plane. For example, try locating an aircraft WITHOUT moving your head. You will find it to be extremely difficult. Virtually everyone looks up into the sky and tries to locate the plane with their eyes and then, once located, its much easier to slightly capture the sound direction. In stereo recording, all of the information is recorded laterally and there are no extra channels to capture the vertical information. There is some savings grace due to psychological aspect of associating high frequencies above low frequencies. That is why virtually all loudspeakers have the tweeters at the top and the woofers at the bottom. If you want to have a good laugh, try turning your speakers upside down to hear what they sound like. You will be very surprised to hear how weird they sound. The second thing I want to explain is presence (my definition). This is the aspect which determines your seating position. When I go to an event, I want to sit right down in frontfirst row. I have several friends however, that cant stand to be that close. They generally sit about 30 or 40 rows back. Now, if you are a recording engineer, where do you put the microphones in order to satisfy both parties? The answer is: YOU CANT. This is a problem of time and not amplitude. You cannot fix this problem with the volume control. Another part of this dimension problem involves the spacing between the microphones. In the old days of wide spaced mikes (called stereophony) the mids and highs were quite acceptable generally however, the low freqs were really a problem. This occurred because with wide spaced mikes the wavelength difference at low freqs caused multiple eigentones and cancellations in the playback environment. Most recordings tended to sound cavernous and boomy at the low end. In order to solve this problem, coincident mike techniques were developed and employed with the capsules virtually on top of each other with angle spacing of between 110 degrees and 135 degrees. This virtually solved the low frequency problem but now introduced problems at the high end. For example, if the capsules are in effect say about 1 inch apart, then that distance represents a wavelength of about 10kHz. When played back with the speakers say 8-10 feet apart, a gigantic hole is produced wherein one must find the exact sweet spot in order to listen. There have been other techniques that have been used to help with these problems such as the MS and crossed figure 8s (a Bert Whyte specialty). But the problem really involves time and distance and not amplitude. A few decades ago when I was faced with these kinds of problems trying to record my Steinway grand, I was indeed frustrated. After pondering this for quite a while, I devised a totally different concept in mike technique. Firstly, I used a MONO center channel that went through a low pass filter with a cutoff at 250 Hz. I then placed a pair of mikes (in stereo) on either side that covered the

frequencies of 250 to 500 Hz. These of course fed two bandpass filters which went to a summer on each channel. The distance on either side of the center mike was exactly 3 feet. I placed then, a 2nd pair of mikes further outside to cover the frequencies of 500 to 1000 Hz. These were placed exactly 1.5 feet outside of the first pair and their outputs went through a corresponding bandpass filter and into the summing circuit. I then placed a 3rd pair of mikes exactly of a foot to the outside of the 2nd pair which covered the frequencies of 1000 to 2000 Hz with the respective output going through bandpass filters to the summing circuit. Finally, I placed a 4th pair of mikes exactly 4.5 inches outside of the 3rd pair which covered the frequencies of 2000 Hz and up wherein their outputs went through high pass filters to the summing network. I would liked to have used another pair in order to make the last band 4000 Hz and up but I didnt have any more mikes available. To say the least, the results were absolutely astonishing. Where I had mush before where everything was very hard to distinguish, now everything was crystal clear and the piano had absolute focus and position. Obviously, this is a very difficult procedure however, I wanted to prove a point. Correct mike location and technique is THE absolute most critical aspect of recording. Im sure that you can comprehend that a 2 channel system can only approximate the exact position of a source of sound in space. If you were to compare this with radio direction finding which uses TRIAGULATION of three receivers to locate a source, its easy to understand that a 2 channel attempt is going to fall short of the mark. Another thing that should be understood is that on an absolute basis, there is actually very little LEFT ONLY and RIGHT ONLY information. Virtually all of the primary acoustic information falls basically in between the loudspeakers. Yes, I can hear the cries of not true coming from all those who believe that the sound stage extends beyond the speakers to the outside. This is of course true but, it is due to the reverberant field and not the stereo composite signal itself. Notice also, that I said very little and not none. As a matter of fact, the most important information, for the most part, is located right in the center or mostly in the central field. Obviously there is no speaker there. This is what I meant by our ear-brain link having to fill in the gap of missing information. A long time ago Nakamichi tried to improve recordings by adding a center fill microphone jack on their cassette decks. Sometimes this helped but most of the time it didnt mainly because the wrong microphones were being used. In order for something like this to have even half a chance, a thorough understanding of the vectoring of microphones is necessary. With this 3 mike scheme simple cardioids just will not work because there isnt nearly enough isolation between the mikes. What is needed here is super or even hyper cardioids and they must be set up very carefully in order to make sure that there is as little leakage between them as possible. Now lets get to the meat of some suggestions and proposals. For the last 45 years or so, we have not had much choice in recording techniques because we were limited to only two channels. Of course I

assume that everyone remembers the absolute disaster that befell us in the 70s. Im speaking of course about quad sound. Anyone with 5 cents worth of brains could have or SHOULD have seen the lunacy of this approach. As it turns out, I and others who tried to scream loudly, were ignored. But unfortunately, after the millions of dollars were wasted on this fools errand, we engineers had the last laugh. Sorry bout that. We are now faced with another idiotic adventure by those who are proposing multichannel recording???? We havent conquered (and never will) 2 channel recording and some people want to use 5 channels? This is absolute nonsense for the following reason. All primary musical information COMES FROM THE FRONT. It doesnt matter what event you go to whether a concert or a club etc., all of the musical information is in front of you and not behind you. So the question becomes: is there a better way? The answer is, you bet. We have staring us in the face an incredible opportunity to have our cake and eat it too with the introduction of DVD audio discs. These discs have about 8 times more capacity than a regular CD. Why not make proper use of this gift horse in the mouth. For example, the 5 channels could be used in the following manner. Three for the left, center, and right and a fourth for a high center, and a fifth for a low center. This would give us that missing vertical dimension. The playback speakers for the high and low center would be placed accordingly and only have to be reasonably acceptable in quality. To go a step further, many sets (say four sets) of mikes could be located at intervals of say about every ten feet with the first set being close miked, etc. Sound wild? Well wouldnt it be nice if you had an adjustment control on your preamp where you could actually dial in your preferred listening position. Incredible concept. Trust me, its doable. All it takes is some industry leaders with GUTS. Finally, a little anecdotal story is in order to prove some of the things Ive been talking about. Many years ago I did a round robin listening experiment with a couple of friends. I had them over to my place wherein I had selected 6 recordings that I considered absolutely spectacular relative to miking, tonal color, etc. We played these recordings and indeed they sounded tremendous on my system. We then went over to the first friends house and my six records sounded absolutely LOUSY. He then pulled out six of his favorite records which indeed sounded glorious on his system. We then went to the third friends house and lo and behold, both my six and my first friends six recordings sounded just terrible at our third friends house. But then he pulled out HIS six favorite records which sounded incredible on his system. We then came back to my place and wouldnt you know it, both of my two friends six recordings (each) sounded like absolute crap on my system whereas my original 6 sounded great. The point of all this is to show all of you just how much we are still in the stone age and just how far we have yet to go. Its going to be a long and bumpy ride but hopefully someday this industry can clean out the cobwebs of egotism and get down to some truthful scientific discipline and come up with credible answers to this very enigmatic audio puzzle.

PART III This final 3rd installment of this series will undoubtedly cause mucho controversy. Im sure that lots of nerves and egos will get rattled however, the truth is inviolate. True scientific enlightenment shouldnt be contaminated by whims and a lot of marketing nonsense. It seems, unfortunately, that our industry has incredible amounts of both. So, without further adieu, let the war begin. I believe that over the years, this entire industry has lost sight of what the goals of audio should be and that is the faithful REproduction of the live musical event. While quality aspects are almost as equally important, the primary function is still to create reality. If you went to a club and the piano had some of the unisons a little out of kilter, does this totally nullify the performance? I have heard Oscar Peterson play on a piano that had a few notes out of kilter but it did not diminish the performance. It was still REAL. The problems of reproducing the live event (or the studio version of the same) are truly monumental. As Ive mentioned in previous writings, there are several pitfalls that make things painfully obvious that we have a long, long way to go. To paraphrase Peter Aczel of the Audio Critic a few issues back, there are many evils standing in the way of true reproduction. The first is the recording itself. What you get is what you get and you cant make it better. You can make it different but not better. All of the information that you are going to have is already there, bad, good or indifferent. The next item is of course, the room and its acoustic properties. The third item is the loudspeaker. And finally, the lousy stereo format, which weve unfortunately been living with for the last 45 years. All of you MUST know that our electronics are the strongest link in this chain and not anywhere as weak as the other items. I tackled electronics in part I and recording in part II. Now its time for loudspeakers and acoustics. Its time to repeat for the umpteenth time the fact the ALL SOUNDS IN NATURE ARE CREATED IN MONO. I hope by this time everyone has had a chance to think about this and begin to comprehend this fact. Therefore and obviously, stereo reproduction is not real and does NOT occur in the real natural world. Just like digital, stereo is a concoction of mankind. Obviously the real question becomes how can we make it better or to be more precise, how can we make this more real. Easier said than done. Enter the dragons There has been for many decades, a general perception (incorrect, I might add) that a perfect loudspeaker would emulate from a point source. Nothing could be further from the truth. The key word here is point. Two points cannot convey an image with a total lateral as well as vertical dimension. Actually, I should say that NO vertical dimension can be conveyed and the lateral dimension is limited by the aperture. I know that a lot of you will want to shoot me for these statements but bear with me. Some of you may remember that many decades ago there were those that advocated a pulsating

sphere as a perfect reproducer. Well, this may be true IF AND ONLY IF one is just trying to create sonic energy. And, it would also be necessary for one to be INSIDE the sphere. After all, many have thought if sound in nature is really spheroid in nature, then why not have a reproducer that creates spherical sound waves. Unfortunately, this just doesnt cut it because sound sources DO NOT radiate acoustic energy absolutely equally in 360 degrees of arc. In addition, we can promulgate all the theories we want but the truth is that there cannot and never will be a perfect reproducer. It is not to be in this universe. The better question would be: how close can we really come? In order to begin to answer that question we must get physical. There are 3 basic mechanisms of propagating sound waves that begin to satisfy the PRIMARY goals of stereo reproduction. These are one, a vertical slot at least 6 feet long and preferably 7 fThe second is a scalar triangular diaphragm with the top aperture at least 6 feet off the floor. And thirdly, concentric radiating rings ala the Quad ESL63 and its newer brethren. The vertical slot approach is very difficult, if not impossible, to truly implement properly. What is required is a slot width that is no wider than half a wavelength at the highest frequency to be reproduced, say 20 kHz. For this frequency it would require a slot width of about of an inch. Obviously, in order for this arrangement to reproduce frequencies down to around 200 Hz, it would have to have the worlds most gigantic magnetic structure behind it and be fed with the power of Niagara Falls. Obviously, there have been many units designed similar to this such as the Magneplanars and quite a few ribbon designs. But the laws of physics and mathematics regarding slots is resolute and cannot be broken or altered. The physics of slot mathematics apply equally to sound as well as light and those who would like to pursue this can find a lot of literature in the science libraries of most universities and technical schools. I am jumping here to the third device, like the Quad, because it exists and has been a very successful approach. All credit should be given to its creator, Peter Walker, as it is a magnificent reproducer. Its only drawbacks are that it is a compromise regarding its radiating field. This lies somewhere between the near field and the far field. If most of you arent aware, a cone radiator will have a fall off of radiated energy that is the inverse square of the distance. On the other hand, a slot radiator will have an energy fall off that is just the inverse as there is no square term. The Quads, due to their multiple radiating rings, fall somewhere in between. Therefore, you cannot get too far away from the Quads and it is absolutely mandatory that they be off the floor. I have found that the most optimum distance is around 6 feet in front of them and boy are they glorious. But like all things, they cannot turn a crummy recording into a peach. Im not going to say much more about them here because to me, they are the best thing that has come along yet even though there are some drawbacks. The third shape factor is as mentioned earlier, a truncated scalar form. To describe this is much

easier than trying to build it. To digress, most people (and virtually all speaker designers) seem to think that flat frequency response, low distortion and wide bandwidth are the primary goals to achieve. Unfortunately, they are all wet. It just aint so. These are SECONDARY importance items. The first order importance items are as follows. Number one: equal speaking time at all frequencies between about 200 Hz and 20kHz. That is, the time response should be optimized so that the leading edge of any frequency waveshape should arrive at the ear at exactly the same time. Easier said than done. Of course, any planar type radiator has a huge advantage in this regard. There is simply no cone loudspeaker system on earth that can equal thisperiod. However, the second first order importance item is the real killer. It is as follows and all speaker designers should pay particularly close attention to this. The radiation angle should also be as equal as possible between the aforementioned frequencies of 200Hz and 20kHz. Trust me when I say that this is no easy feat to accomplish. As a matter of fact, this is singly the most impossible attribute to achieve. So far, the Quad comes as close as any speaker ever has but it still has a quasi-comb filter effect due to the multiple diameters of the radiating rings. But it is still the best that has ever been done in this regard. Now, how could this be made better is the question. A six-foot tall scalar triangular radiating membrane, which is slightly curved at the base, would be ideal. In this, only the top 5 feet would be the radiating element. The width of this membrane would be such that the radiating angle would be approximately 1-1/2 radians or about 80 degrees. Many years ago Dick Sequerra produced a speaker called the Metronome. Although this first device was a bit crude and the proportions were far from correct, it did have some absolutely amazing properties that I have yet to EVER experience in another loudspeaker since (save for a later version of his metronome which was much taller). Assuming that one has an optimally recorded event (not an easy task) there are two distinct things that a loudspeaker should do. Firstly, one should be able to walk right down the middle on the centerline of the speakers while closing ones eyes (therefore letting ones ears do the work) and as one approaches the speakers one should get the experience of walking right smack up to the orchestra. WOW. Folks, I have experienced this and it is enough to send goosebumps right up your spine. This is the kind of thing Im talking about when I state that the radiation angle must be constant. A couple of recordings that absolutely knocked me out with this effect were Ken Kreisels recordings from M&K Sound. Ken is one very sharp recording engineer whom I might add has probably the finest set of ears in the business. He KNOWS what things are supposed to sound like. The second effect that is an acid test is as follows. As you are walking down the centerline of the speakers, there should be a point where you are absolutely dead center between the two speakers. At this point the sound should jump right straight up over your head. This should actually be obvious as there is a point right between the speakers where the lateral plane now becomes vertical. This is the second effect that the Metronomes accomplished. If one could imagine this loudspeaker being another

2 feet taller AND being planar instead of using cones, the sound would be truly the most spectacular ever produced. Ah, but there is a fatal fly in the ointment. Those of you who read my first two installments might recall the situation of my house verses two of my friends houses wherein each of us had our 6 recordings that sounded great in our place but awful in the other two places. So, whats going on here? Simply put, the microphone placements, coupled with the loudspeakers in the playback environment are not mathematically adding up right. The angles (or vectors) are not aligned correctly for the majority of recordings. I thought long and hard about this and came up with a solution, which is not necessarily practical or cost effective. However, it is doable and someday someone might attempt this. It is as follows: design a mechanism that I call SpeakerTrak. What this would be is a platform that would allow the speaker to be moved left or right, up or down, rotated, tilted, and moved forward or backward. Obviously this contraption would have to be extremely solid and produce NO mechanical vibrations of its own. Most importantly, it would mandatory that this be remote controlled. Therefore, one could adjust the speaker from the listening position. Incidentally, the electronics could also have a microprocessor with memory so that once a position is located for a particular recording, the position and settings could be stored in memory so that every time you played that recording, the speakers would automatically align themselves to the correct position. A device like this might cause some consternation in a lot of living rooms however, this is the only scientific avenue that could achieve the correct acoustic results. This so-called platform could also be located on the wall or hung from the ceiling. Imagination prevails here. Do you think that Im nuts for proposing this kind of a solution? To be honest, I have over one thousand recordings and I think about only 30 or 40 of them are really listenable with my present setup. If only I could MOVE the speakers for every recording. Now lets tackle another acoustic problem which involves home theater. As I have mentioned previously, having a TRUE center channel situation is the most desirable thing to achieve. BUT doing it correctly is easier said than done. Firstly, a processor with the correct algebraic solution is mandatory. This will be available shortly. When incorporating this scheme, it is absolutely mandatory that the (necessary for stereo) cross-coupled error signals be eliminated from the opposite channel. You can get a feel for what Im talking about when listening to headphones, as there is obviously NO cross-coupled error signal. The biggest problem is that the overwhelming numbers of center channel speakers that are available are just positively awful. They are designed to physically fit in with the screen (mostly large TV) and as such could not (should not) be considered as hifi reproducers. Enough said about them. What one must do in order to have a decent musical situation is to get the screen as high off the floor as possible so that a decent loudspeaker can be located under the screen and still be tall enough to provide accurate reproduction. You must bear in mind that with this

kind of setup the center channel now becomes the MOST important speaker and should be of the highest possible quality. Also, FORGET ABOUT THE REAR for music reproduction. It is a red herring and I predict that this potential multi-channel recording baloney is going to end up just like Quad sound of the `70s, that is, a debacle. This situation is truly a fools errand wherein, just like in the `70s, marketing idiots seem to be spearheading this nonsense. I was right about the nonsense back then and trust me, Im right about it now. The rear speakers should only be used for the surround in home theater where the effects can be truly spectacular. But in music, THERE IS NO REAR. So there you have it. The very best playback system would consist of the following: left, right, and center speakers with high and low center speakers with a subwoofer. And, there should be provision to adjust the physical properties of at least the three main front speakers. And finally, there should be the ability to dial in your preferred listening position. A tall order, huh? Well, weve had 45 years of chickens all scattering in a myriad of directions and very few roosters. Somebody now needs to step up and scream COCKADOODLEDOO.

James Bongiorno