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The Royal Device WHITE PAPERS |
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since 3-3- 2004 |
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THE UNIVERSAL EQUILIBRIUM and THE PERFECT BALANCE
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For decades, we have been taught that an amplifier and loudspeaker is an audio system, but an amplifier can be used in several applications other than with loudspeakers. This is only one application out of thousands. The fact that we use an amplifier (the only thing that we have in the today technology) to drive a loudspeaker is not because they were born one to the other. Normally it is the contrary and so they do not match as perfectly as they should. In the HI-END HI-FI world, to encourage customers to spend a lot of money, new products have been invented for which incredible claims are made, which are considered to be real truths, but from which it is impossible to get a real sound. If we consider integrated systems in any discipline, we will easily find that there is a always a perfect balance (equilibrium) between the individual components and the entire system, whereby everything is running properly. I have never seen a car using a wheel of a truck, or the wing of a Boeing 747 used to build a Concorde. If you look to High-End Hi-Fi world this is the rule instead, but everything in this universe MUST have its own PERFECT balance. Let's take, as an example, speaker cables. When we make the Perfect ZERO OHM cable we (think) will may have made the best electric coupling ever possible (transferring maximum power in the related bandwidth), but due to a very very big electronic simplification (that everyone does) we forget the problem about "compression of sound" and "temporal distortion" that exponentially comes out more you couple the amp to the speaker electrically speaking. Also here there is a PERFECT BALANCE that MUST be found on each system instead, that normally it is not "THE ZERO OHM CABLE" or the cable that is 55 times bigger than the amp connector output. Normally making things simple we understand ..... but make them too simple we normally will get in trouble" A Hi-Fi system is like a car Asking for a speaker cable is like to ask which gear or which transmission line I have to buy for my car. Which car? Well, I have not chosen it yet, but I would like to have a spare one just in case this will brake down. The hi-fi system is like a car, the only difference is that we would like to make the car by ourselves instead of leaving it up to a complete engineering team. I can understand the reasons.
SOME ISSUES on HI-EFFICIENCY approaches and driver choices. 1 - Horns sounds as horns, this is the first reason why I have not used them as midrange. My system is a full range without crossover integrated with a wooden treble horn with a .75" soft dome tweeter, not a compression driver (for the treble above 8KHz) and the back-loaded horn subwoofer for the low end (starting from 10 Hz to 40 Hz). The horns in these two ranges of frequencies do not sound like horns. 2 - There is a very important reason to use Hi-Efficiency: to be able to use amplifiers without feedback. 3 - The first requirements to achieve a good system and start listening to music instead of distortion is: 3.1 - No crossovers in the loudspeakers 3.2 - No feedback on amplifiers There are several problems with feedback to be understood. All cheap amplifiers use feedback, and there are no amplifiers above 30 -40 Watts without feedback. This is a very important point. There is no solid state amplifier that can be designed totally without feedback. Bi-wiring your loudspeakers is another way to improve the sound quality, depending on the manufacturer's design. Bi-wiring reduces sound compression. There are several physical reasons for this.
TO THE TOP There are several rules of physics which dictate that a crossover kills sound. If you would like to design a good speaker you should thus avoid the use of a crossover and choose a woofer which is correctly integrated with the treble. The acoustical crossover point must be out of the midrange, at least above 5KHz, and the higher it is, the better a speaker will sound. Any acoustic crossover in the 500Hz to 4000Hz band will be clearly audible and will result in annoying sound. Just some points: A 6 dB/slope on a woofer requires a coil in series. The coil introduces delay. In the time domain a complex signal can be divided into his fundamental plus all its harmonics that are in phase with the fundamental (harmonics have different amplitude, this give the timbre of the sound). The coil in series to the woofer will feed the woofer first with the fundamental, later on with the second harmonic, then the third and so on. The woofer will reproduce them as they arrive. What does the tweeter have to do to restore the initial phase relationships? It has to do exactly the opposite. But this is practically impossible. Even if you put a single capacitor in series to the tweeter (that is the minimum to get a Hi-filter) that only in theory does exactly the opposite than the coil in series to the woofer, this will never happen. The reasons are: 1 - The coil & the capacitors are calculated on the crossover point and with the filters calculated onto a resistance, not to a driver that sends back electromotive energy out of phase with the other drivers and to the amp output. 2 - The mathematical calculations work only on sine wave signals and not on transients contained in music. This is the biggest problem, because it puts the two drivers out of phase with each other. 3 - The tweeter is 10 to 100 times higher in velocity in doing this. So you will never have this reproduction in phase to the woofer, even if you put them mechanically in line. This is a second out of phase problem. The final result for a speaker made with a good woofer without filter and a tweeter with its capacitor (to protect the tweeter from harmful frequencies) is much better than any speaker with a complete crossover. The only thing is that you have to choose a good woofer and work on it to cross it mechanically to the tweeter. The biggest problem in loudspeaker design is the difference in driver velocity. It is not important what the amp does or what it is able to do, but what it is arriving to our ears in terms of phase response between the different drivers used in a loudspeaker. Compression on the tweeter (horn loading), reduces the velocity of the coil, dropping it more nearby woofer velocity levels. Of course there are choices also on the woofer for velocity response. The use of paper cones and foam surround normally helps a lot. Woofer diameter is very important for velocity problems. The larger the woofer diameter, the worse its velocity response. The smaller the woofer diameter the faster is the velocity response, but the higher will be the woofer resonant frequency, reducing bass response. The choice of a 8.25" woofer on the Laura & Diana is the only way to make really a "quasi" full-range speaker that has good bass response and very high bandwidth extension. Considering very good "custom" drivers 4 - A 6" driver cannot go below 60 - 55 Hz, even if it can reach over 10-12 KHz. 5 - A 10" driver can go below 35 Hz, but will drop roughly at 3 - 4 KHz 6 - A 8.25" very well done can reach up to 8-10 KHz and have a resonance frequency of 37-40 Hz. Of course it is better to have 10Hz more below (the difference between the 6" and the 8.25"), than have 2 more KHz on the high. The reason in very simple. With only 50 Hz of resonance, I need to reinforce the low end by means of a SUB or with a more complicated back-horn loading. Crossing over the tweeter at 8KHz or 10KHz will not change sound quality, because we are into the very very high harmonic range of the audio spectrum. Choosing a 10" driver to get more bass, will encounter the biggest problem in Hi-Fi: with or without a crossover, the acoustic crossover between the two drivers will be audible as "temporal distortion" on the music signal (emissions out of phase) as explained above.
TO THE TOP AMP TO SPEAKER INTERFACE PROBLEMS QUESTIONS on the above - Crossover and AMP feedback. All this sounds like a good argument for doing the crossover electronically and having a separate power amplifier for each speaker drive unit. Integrating the power amp with the speaker also has the added benefit of not needing long lengths of expensive speaker cable. This is the great neglected area of Hi-Fi, probably because speakers and amplifiers tend to be made by different companies. The amplifier/speaker interface is one of the most messy, uncontrolled areas in the whole signal chain - with so many variables at work there's no wonder a small industry thrives making fancy speaker cables. As for the evils of feedback - well that really is a minefield, and there's not time or space to go into at this time. I will say that no feedback does seem to be an attractive proposition but that feedback has been given a bad name by decades of poorly designed electronics. ANSWERS 1 - Any Hi-Fi Amp does not feed a resistor load. (I have been involved into electronics for decades and my first job (for about 4 years) was designing regulators for DC motors up to 100 HP or more. All these MUST be with feedback) 2 - DC motors do not reproduce music 3 - A woofer is a DC motor and not a resistor. It becomes an AC motor when you move its cone backwards and forwards Out of the R, L, C, equivalent circuit we must add a voltage generator counterclockwise toward the amp. The woofer produces energy. This (energy) generator causes big problems because the voltage sent back to the amp is not only reversed (a first type of out of phase) but also sent back to the amp later on, when the woofer starts to move. 4 - A woofer moves very very slowly in respect to the signal fed by the amp. Points 3 and 4 are the ones that create all the problems, not from the R, L, or C of the equivalent circuit point of view of the woofer. I have used my experience in all the electronic fields I have worked in to try to understand and this is what came out. 5 - It is not right to say that the output resistance of an amp is ZERO. This is only theory. Even if you apply a lot of feedback the output resistance has a real value, it could be low but still a value higher than zero. 6 - This means that the energy sent back to the amp by the woofer is not really all short circuited by the low impedance of the amp output. 7 - The feedback of the amp will try to correct the situation sending back to its amp input the modified feedback signal. Here is where the mess is made. Here is where the sound starts to change, introducing any type of distortion (we can say a lot of things on this topic) Some take the position that often the feedback of power amps is blamed for poor sound when in fact its the amp/speaker interface where a lot of the errors occur. This is correct, but we have to feed speakers, not the dummy loads of resistors. 8 - Amplifiers without feedback do not have any problems in this regard, while I found feedback to be one of the worst causes of degraded sound quality. The signal back to amp arrives several times later on and added to its input completely out of phase and so on. 8.1 - Amplifiers with feedback are perfect if they drive loads without energy sent back. As an example, several Hi-Fi magazines published very low distortion measurements with a capacitor or a coil on the amp output. A woofer is ten time worse because it moves back and forth, coils and capacitors do not. 9 - From this point the first thing I have to say is that the slower the amplifier response, the better it is to be coupled to a woofer (not to a resistor). 9.1 - The two worlds are too different (amp & speakers). We have always been taught that the amp must be the fastest possible. You cannot do this with motors. The regulator (the amp in this case) must have the same inertia of the motor - the speaker in this case). If it is so everything starts to work properly in terms of sound quality. 10 - Damping factor rise up applying feedback to the amp. But the damping factor is not so important as everybody says. I will say what I think later on. An often-observed very important point about cables: The amplifier/speaker interface is one of the most messy, uncontrolled areas in the whole signal chain - with so many variables at work there's no wonder a small industry thrives making fancy speaker cables. Now I say what is my thinking (this came out after several type of experiments and from suggestions given to me by other engineers). I have just tried to think about "something" (the cable in this case) NOT AS AN INTERCONNECTION. Then I have started to understand. The cable is a media that MUST do 2 things and not only 1 as normally thought. 11 - CONNECT two completely different worlds - amplifier and loudspeaker 12 - DIVIDE two completely different world. - amplifier and loudspeaker (This is not a stupidity.) 13 - The higher the resistance of the cable, the less the amp will have problems. GOOD. This point reduce distortion. 14 - The higher the resistance of the cable, the higher will be the power loss. BAD if is too much. This point does not introduce distortion. 15 - The higher the resistance of the cable, the higher is the serial inductance (normally) and this will cut high frequencies. BAD if it is too much. This point does not introduce distortion. 16 - The higher the resistance of the cable, the less will be the capability for the amp to force the woofer to follow the amp signal (damping). This is both GOOD and BAD. 17 - Lower is the resistance of the cable, worst it will be the sound because the amp will have problems. VERY-VERY BAD This point introduce distortion with feedback amp. These problems are avoided without feedback. This can be a start. Reducing the problems where distortion is generated. POINT 16. damping factor can be increased in several other ways without using feedback. Tubes: 16.1 - Output transformer over-dimensioned (few are able to make tube transformers) 16.2 - Type of tubes 16.3 - Current choice on the anode 16.4 - Other factor In a tube amplifier the transformer is a "saint", because it helps in dividing the two worlds. A good transformer has a serial impedance of .4 to .6 ohms in a tube amp without feedback. That is already more than enough for damping. Woofer damping can be increased by using big magnets on woofers. This gives 2 good points and one bad. GOOD Increases efficiency and Increases damping BAD Easy to have break up and more difficult to have quite flat frequency response. Also cabinet choice (horn back loading, bass reflex etc. changes the type of damping) AMP The harmonic distortion introduced by a GOOD amp with no feedback can be very low (.1 %) if you use amps of 15 Watts/channel and use it at 1-2 watts. But here you will need hi-efficiency. Remember that high efficiency is NOT equal to HORNS, but HIGH EFFICIENCY = HIGH EFFICIENCY. I won't go further on, because here you understand that now the approach to make a stereo system becomes a real philosophy in choosing the correct type approach itself then the choice of speaker then the type of amp and then cables, without talking about what is on the market. The cable, at the end, is the medium that has to be the right compromise for the amplifier and speaker and must be fine tuned considering all the technical parameters of the two. Now from this we can understand the reason why there are so many cables (maybe nobody knows) but if you make a simple calculation as to how many combinations of amp+cables+speakers there are, you will see that it is impractical to listen to them all. But the worst thing is that nobody knows what to suggest to you. Who does it, does not know the problem, or maybe he does know perfectly your amp needs and your speaker needs. My thinking: 1 - The slower is the amp better everything will sound. 2 - The higher the efficiency of the speakers the less power will be needed and less distortion will be introduced 3 - The higher the efficiency the less will be the distortion of the woofer (cone moves less), distortion will decrease in a quadratic way. The above simple rules after all the tests made and the suggestions of people in the field for me have become a sort of LEGENDA. Many times I do not understand the reason why of something, but that does not mean that it is not true or real. I thank everybody who continue in helping me to understand what I don't know of what I am looking for.
TO THE TOP Bi-wiring is needed to reduce woofer energy sent back to the tweeter (this energy produces a lot of sound compression - normally very high if mono-wiring is used). Cables must be chosen very well in this regard. If you bi-wire, then the energy of the woofer is reduced by the fact that this energy to reach the tweeter it first has to go back to the amplifier output (that is NEARLY a short circuit) then go towards the tweeter again. The going back and forward (THRU THE CABLES) plus the very low impedance of the amplifier output ( that nearly shorts this energy) reduces a lot the ENERGY then the sound compression (not all of it but reduces a lot or so), this depends on several other factors. If you do bi-amp as you suggest is something not even imaginable. The difference is incredible. Then you can (in case of no crossover speakers or crossover speakers) do as following: passive filter ( or no filter) Amp -------- woofer preamp ---- passive filter Amp -------- Tweeter (amp power needed is 1/10 of woofer amp power) then you eliminate crossover on the speakers and connect drivers directly. In this case the filter at the amps input MUST be review, because are function of the amplifier impedance inputs and so on. Bi-amp is the maximum. It is where you can start feel the real music. If you do bi-wiring you have to leave the crossover on the speakers If you do bi-amp then you have to eliminate the crossover. Each driver is then fed by itself
TO THE TOP
1 - Figure 1 - Normal
connection
Let's take a signal from the amp to the speaker in the positive wave ( the same applies to the negative wave). 1. Let's take a signal from the amp to the speaker in the positive wave ( the same applies to the negative wave). The current flow (IL) generates an electric flux (H vector) that produces an electromagnetic field (EMF). This EMF. Is produced in both wires and interfere one to the other producing distortion. Even if it is really hard to believe, the current in both wires is not exactly the same (this is just a simplification). If we consider as an example “signal cables” we have at each end some current dispersion for the “plus node” towards the internal circuitry that normally have dispersion towards the power supply ground. A signal cable connecting a CDP to a Preamp or to an Amp. The current flow entering the Amp input (plus) will not completely exit from the “minus” connection, because some, will be delivered to ground thru the dispersion of the cabinet chassis and the other various ground connections. Transformer coupling devices has less problems in this case, in respect than direct coupling, but needs to be perfectly loaded. We have to remember that introducing a transformer can alter the “frequency response” and many transformers (one after the other) into the audio system chain could maybe not the right solution for good sound. As said, everything has to be seen into a perfect balance. This is just to give you an idea of the reasons where sometimes we have a complete system that lack of highs or lows or has too much treble or whatever you like, but looking at it seems to be “perfectly balanced”…. The audio system should be measured completely and not one piece after the other. Returning to cables: 1 - There is a voltage drop between the “+” of the amp and the “+” of the speaker due to the resistance of the cable. 2 - The cable has a serial inductance that will delay the signal proportionally to frequency This delay changes in any point of the cable in respect the opposite point of the other cable. 3 - There is a capacitance distributed long the cable between the two cables that reduces frequency response. The H vector on the positive wire (H IL+) will not be exactly the opposite of the H vector of the negative wire (H IL-). The two EMF will interfere one to the other because there is a difference between the two and this must be avoided. The difference between these two EMF is something that we don't want. This difference will interfere with the current signal itself on the other wire. This is only a simple explanation and does not want to be exhaustive because the problem is much more complex than above explained. 2 - Figure 2 - Shielded cable connected both ends. The difference here in respect to Fig. 1 is that there is a second current flowing between the shield and the NEGATIVE WIRE itself due to differences between the shield and the negative wire in terms of resistance, capacitance and inductance. The shield in this case only shield towards external currents (EMF generated by other wires nearby). Here the things become worst than fig 1. The current flow into the shield MUST be avoided. The shield itself becomes a electromagnetic generator towards other wires. Plus in case of signal cables you will have HUM. Figure 3 - Shielded cable connected only at one end
Here you won't have the problems of figure 2 but only it will avoid interference towards others external conductors (EMF generated by other external wires nearby). To have a good HI-END CABLE we have to reduce serial inductance and parallel capacity leaving the right series resistance that depends on speaker impedance, WF dumping and type of amp output stage. Type of Wire, type of Shield, type of insulator, type of diameter, type of twisting of wires and of internal wires (there are thousands of combinations) will give the final result. What we use for our Royal Device cables is not even one of the above solution and our speaker cables are finely tuned for our speakers toward a Tube NO Feedback transformer output amplifier S.E. using good materials without increasing costs to unaffordable levels. We do not make cables just to sell cables, even if we can design cables even for other type of speakers. The speaker cables are not the least, speaker cable MUST be designed and designed depending on the followings: 1 - type of amp - feedback or not - tubes or solid state - transformer or direct connection 2 - type of speaker - vented box, closed or back-horn loaded - horn frontal loading or direct radiation - number of speakers involved in the total system - mono or bi-wiring - multi amp or mono amp system Who sells you speaker cables without asking anything of the above and without knowing how to approach the problem, this is just a “smart guy” I would say. I am not saying that you won't hear a difference, but surely you won't know if this is really the better "real sound" could come out from that particular system. Now, if you just go and buy the costlier speaker cable on the market and connect it to your system without considering the above, well you will just throw your money out of the window. Everybody does normally, but everybody is free to do what he recognize as the best for its money. You still have to consider the fact that even if you find a lot of funny cables on the market (that cost half of a Cadillac per meter) this is not proved that this will sweet your system especially if these are made of “silver “ or “gold” with “proved type of twisting” and other, but if you open the small box in between (some cables have), than you will easily find that there is a resistor (or else) to “re-balance” everything. Strange isn’t it?
TO THE TOP Questions: Maybe the problems can be solved using a current drive amplifier instead of using a voltage drive amp. Answers: My opinion is that it is not a matter of current or voltage drive. It is like talking of crossover with series components or parallel components. The very big problem in feeding a speaker is the completely different nature of the two components: the speaker in respect to the amp. The speaker cable here plays the same role as a jury that try to clear down two men fighting each other for something and both of them have completely opposite idea for the solution. The best solution does not exist talking about these terms, because each one has its own positive and negative points. The best choice is to try to choose the speaker you like in terms of efficiency and sound quality and then try to find the amp that at his best have the characteristics to drive such a kind of speakers (this is the more difficult job. Selection MUST NOT BE dictated by the rule: "this is the best amp ever made"), then a very good fine tuning can be to set the right cable that MUST NOT be a very good SHORT CIRCUIT. Normally better you couple the two (electrically speaking) worst is the sound you get. If you would like to enter more deeply this subject you can read my opinion at "thoughts" in http://www.royaldevice.com/thoughts.htm . Here you will also find the problems related to amp and speaker matching and the problems created by a very good electrical matching in terms of sound compression and temporal distortion. Moreover there is no way to control the current in a speaker /this is what a current drive amp does) without the use of a feed-back.
TO THE TOP Changing the power cord doesn't always make a difference. It depends on the system and the power supply design of the systems itself. Sometimes the differences are not there or even the sound gets worse. I am not saying that changing the power cord you cannot ear the difference, But changing it you can have either the following situations: 1 - better sound quality, 2 - same sound quality 3 - worst sound quality. It is exactly the same as on speaker cables. If you just use a very good conductor (a zero ohm cable) on certain systems you just get troubles (sound quality decreases a lot due to feedback, the amp gets problems) and not a better quality.
TO THE TOP Here's a scientific explanation for the reported sound change after burn in of amps and cables.
An amplifier, or any other piece of audio
electronics, could quite conceivably sound different after a burn in period
because the various measurable characteristics of the electrolytic capacitors
will change as the electrolyte is "formed". A similar process happens for all
components that makes an audio system.
TO THE TOP Do STANDING WAVES CHANGE, changing the type of amplifier? Standing waves are everywhere in any room, this is due to the impossibility for the air to exit the room when air pressure starts to be produced by a speaker. In each situation, it is not only a matter of standing waves, because you changed the amplifier, and starting from here you had a different response, the room has remained the same. In any case you cannot remove standing waves by means of an equalizer not even with some absorbent matter on the walls nor using tube traps (these just reduce some). These normally work above 300 Hz at least. There is a completely different approach to remove standing waves. The only way to remove standing waves below 200 Hz is to change the shape of the room differently for all the walls, normally impossible for any audiophile, that would encounter problems with its furniture and so on. If you want to have a look in how you have to change walls and ceiling you can go and see everything on THE RD AUDIO ROOM Here you will see some ofl the techniques applied to practically remove standing waves in the environment of the listening position. You will see that a big intervention has been made for the back wall where there has been built a wall with a thickness of 90 cm (3 ft) of fiberglass wool with different shapes and height to make rid of the back-wall reflections (for bass response I mean) starting from here you can cancel standing waves intervening on the remaining walls and ceiling. Please see all the pictures of the ceiling and walls o the web previously mentioned. Now apart from the above problem that was already present on the room (every room has standing waves), there is the possibility to compute them. I have made a PC program able to compute all the standing waves and their distribution in the room below 200 Hz starting from the room dimensions. My opinion is that the problem must be solved at its root searching it looking to the differences in respect to the previous conditions (before changing the amplifier). My opinion is the followings: 1 - The new amplifier could have a lower damping factor and maybe also a reduced negative feedback (these two parameters normally are coupled one to the other). This means that if cables has remained the same, the capacity of the new amplifier to damp woofer backward electromotive energy is less. This also means as consequence that bass response will be much higher. At this stage we cannot say if bass response will be better or worst. In a normal approach If the speaker is chosen correctly with cables and amplifiers the response is normally much better, but to enter this approach it needs to review many wrong hi-fi approaches known up to now. 2 - Due to the above maybe the speaker must be better de-coupled by means of some anti-resonating systems. 3 - After that you can try a new placement for the speakers. Normally this is the first thing to do. My approach now is to make it as the last step, because, my actual approach is to understand what is happened when one changes the amplifier. 4 - If you would like to put the speakers on the ceiling instead than on the floor I would I suggest the following: 4.1 - More the woofer stays nearby the ceiling, more bass response you will get. Normally it is 3 db more because you load the woofer in a Pi-grec/2 steradians instead of Pi-grec. The woofer loading is proportional to the frequency and can be calculated, but this is not the case (this is only the theory but very important for bass response) 4.2 - Keep the speaker at least the maximum lower you can from the ceiling. I don't remember if your speaker have the bass reflex hole on the front. If it is so, you are lucky, this reduces the resonances at the resonance speaker frequency and in its environments. 4.3 - If you can keep away the speakers also from the side walls and if you can from back wall. All this will reduce a lot bass response. Remember that the woofer loading is due to the reflection of the back wall, the side walls, and the floor. In your case the ceiling. Every wall or ceiling reinforce of about 3 db all the frequencies up to 200 Hz (wavelength is roughly 1.7 meters (5.6 feet) at this frequency). The reinforcement is different for each wall and depends on the distance of the speaker from them. The worst case is when you have a particular frequency that arrives to the listening point reinforced by all the walls and ceiling. TO THE TOP Feed Back and Temporal Distortion: What is this? Many of you have tried NO FEED BACK (F.B.) amps and found vividness, expression, rhythm and dynamics over other F.B. amps and preamps. The same is for NO crossover speakers even if these have sometimes “coloration” instead. This issue is related new and came out in the last few years. It seem that science still does not know the reason why and all the measuring stuff seems to say exactly the contrary. Measures says that F.B. amp have less distortion. Well, this could be considerably true if we just rely onto the type of distortions known or discovered up to now by science, but it is a matter of fact that any time my daughter plays a CD onto its completely F.B. system, I can ear a type of distortion that is not “Harmonic distortion”, either is not “inter-modulation” but something else even if her bedroom door is completely closed. Maybe no-one up to now has considered the fact that some other type of distortions are present into our audio systems, but I am a fellow not so easy to convince so, even after having gone thru all my Radiotron Designer Handbook and designed a lot of F.B. and no F.B. stuff and crossover and no crossover speakers, I have not found what I was searching for. So I have started to just make up my mind in a different way. Now, here it follows: 1 – Harmonic distortion. All F.B. amps have reduced amount of this. Moreover you have to let the amp clipping to get some…who is using amps above the rated power? No one. 2 – Inter-modulation distortion. Almost all the good “state of the art” F.B. amps have none or at least negligible of this even onto capacitor or coil dummy loads. Another thing to say is that in the Hi-Fi world all the manufacturers just look to its product and no one would like to enter the “system” theories, too much time to do it. Moreover, normally no one has a fixed objective point of view and reference, just because no one know effectively what is written onto a CD or LP. This just because to know this you have to amplify and make a “complete audio system” come up to work. I have always been involved into hi-fi, but worked as telecommunication “system engineer” for tenth of years then I have started to ask myself: “What is happening here?” If we still are hearing annoying stuff, there is a small percentage of possibility that we are entering the problem in the wrong way and I know there is people that do not even understand there is distortion here, just because get used to it. May we be able to face the problem in another way? Now, here is how I did it and what I have found. The problem is not just related to a fixed product, but to a “system”. Every manufacturer measure its product , but does not measure a complete system in terms of finding out different type of unknown distortion. I mean, we cannot find or solve the problem if we just look always at the same type of distortions of the amp or of the speaker or of the preamp itself or else by the way. We have just to connect at least two things together to get that annoying material stuff……so we have to make up our minds of what can produce that stuff. Now. The problem, as I understand, rely onto a SYSTEM and not onto a PRODUCT taken by itself and measured as we have done up to now. Ok then , let’s start. Let’s see what happens when we connect two things together. The biggest problem rely onto the amp to speaker interface, but the same is considerably present also into a CDP by itself even if the preamp input does not have a so “difficult load” to consider. What I personally found is the following: 1 – F. B. tryes to set up a correct way to control what the amplifier is doing. F.B. is needed to simplify amp design. It is much easier to make amps with F.B. I don’t know if there is a way to make a Solid State amp completely without F.B. At least there is some on local amp stages, this just to reduce S.S. amp factor too high. Tubes do not have this problem. 2 – F.B. reduces noise, reduces gain and increases bandwidth and reduces DISTORTION too. Harmonic distortion, but we said that we are not using the amp above the rated power and we can measure this just connecting an oscilloscope or reducing the volume level.. Someone would say that maybe the THD comes out because the speaker has a too low impedance. NO it is not, let's just connect a speaker that is made only by a wide band Woofer and that's all. Non crossover, nothing else, only two wires and a Woofer that reproduce up to 9 or 10 KHz. We all know the impedance of the Woofer as it is. If our amp is not even able to feed this, well in this case how we can think it will be able to feed a speaker using complex crossovers? but 3 – F.B. destroy sound. This is almost recognized by the way. We are not saying F.B. is not good for amps purposes. I have been designing D.C. and A.C. motor power supplies for years also, and this cannot be done at all without F.B. I am saying this is not good for sound reproduction. That’s all. We can afford many principle to understand this, but it is simply easier to consider the followings in the time domain other than enter into complicated maths concepts that at the end only 10 people into this world maybe able to understand starting from the beginning up to the end: Amp to speaker interface (amp with F.B.) 1 - At the amp input comes a signal to be amplified (time=T0). Let’s start with a half sine wave of a certain frequency (we consider it as a burst) 2 – Amplifier make his job and amplifies this signal that arrives at amp output after a certain time (T1) 3 – The signal is fed to the speaker thru the speaker cable 4 – The signal arrives to the speaker after a certain period of time (T2) 5 – speaker (normally the WF most) starts to produce air pressure (T3) because it starts to move 6 – Speaker in this case produce energy, a lot of energy (T3). How much is this? Well, if we consider low efficiency speakers that normally have 0.1 % efficiency, we cannot say that all the power we feed to the speaker goes into heat, if so, they would burn out very quickly. No, the fact is that a lot of this power fed by the amp returns back to the amp itself. Only a very small percentage of this power is translated into air pressure, and a small amount is converted into heat. Now, the remaining power goes back to the amp. Here is the first big problem. The Energy goes back in opposite phase and WITH A TIME DELAY. Woofer is an AC motor when it moves and produce energy. The same energy the amp has fed to it (reduced by the heat product and the efficiency conversion) But WF moves very slowly in respect to amp velocity and has a mechanical inertia that cannot be disregarded. 7 - The amp output gets then, an OUT OF PHASE and DELAYED in time SIGNAL back (T4). Take care. The amp output impedance is not ZERO. If it was so, there would not be any problem with F.B. (but we know there are), because this power coming from the speaker would then be ALL short circuited. But this is not unfortunately, and who thinks that introducing more F.B. will reduce amp output impedance (to short circuit EMF coming form the speaker) just forget that more F.B. will return more signal to amp input, so even if you reduce the amount of EMF on amp output, more of this will return to amp input due to the increase of F.B. loop capacity… so…. Note: EMF: WF Electro Motive Force What happens with F.B. amps: It happens that the amp output in the meanwhile is changed in respect to the amp output at point 2 for two reasons: 1 – amp input is feeding a "new signal" (or a new portion of this) (T4) 2 – At time T4, at amp output there is the sum of this "new signal" with the one coming from the speaker that has been produced by the input signal at time T1. But now we are at T4. The result is that amp, by means of the F.B. loop will say to its input that something is wrong there (on its output). He does not know the reason why. It just notice the fact that this is happening (T4). 3 – the input, using the returned signal coming from the output (by means of the F.B. loop), will just do its job (T5). There is a time delay also into the F.B. loop. (Normally this produce Inter-modulation distortion in the amp using it by itself with a resistor dummy load) Which is the job? The job is to correct the amp signal to restore the initial situation. BUT……. THE TIME IS DIFFERENT. The signal entering the amp input at this moment is different, it is another one. At T5 there is another signal entering, while the amp would like to restore the situation of a signal started at T0 and then changed later on. How may I change this situation? How can I be able to go back in time? Well, we should enter the Einstein relativity theory to do this, but in this continuum space/temporal reality we have to consider speed of light as a limit. I know we can exit this, but our amps and speaker rely onto this reality. So let’s stay here at the moment. So, if we stay here, this is not possible then. If we correct the input signal, we will correct a signal that is NOT the one that amp had amplified in time T0. But F.B. amps do not know this, so, cleverly they will do it. But we were the guy that have told them to do it. So why now we are not satisfied of such a job? The first error here, as I can understand, it is that F.B. is relatively well working into “continuous signals”, (sine waves), if we exit this context, we find that overshoots comes out from F.B. amps (frequency domain), and a certain inertia is also present into amps too. But amps are not perfect and fast enough to avoid this problem. The strange thing is that more fast gets the amp, worst it will come the sound if the amp has F.B. I am not saying that the amp is not good if it is very fast, but I am saying that is not good to feed speakers. That’s all. Maybe they are good for other purposes as inverter for A.C motors or heating your house anyway this is a good application, maybe not or so much worth for your music. If you look to the following diagram you will see what happens in the time domain. The graphs do not want to be exhaustive of course, but they need to just give the right idea. Signal is DISTORTED in TIME, this is the reason why I have called this TYPE OF distortion as: “TEMPORAL DISTORTION” There are several types of “temporal distortions” introduced by an audio system chain. 1 – Electrical temporal distortion 2 – Acoustical temporal distortion All of them come in when a F.B. is present, just because there is a time lag in between the signal fed and the signal returned by the next stage and when a crossover is present. Talking about crossover it is exactly the same, just because the signal fed IS NOT a “continuous signal” as a sine wave, but a “TRANSIENT SIGNAL” as music does, but even on continuous signal, crossover destroy sound, because not feeding dummy load but speakers. Where temporal distortion is produced: Electrical temporal distortion is introduced in the following points: 1 – Amp to Speaker interface 2 – Preamp to amp interface 3 – CDP to Preamp interface 4 – WF to TW connections 5 – TW to WF connection 6 – Amp to crossover connection 7 – Crossover to driver connection 8 – A/D converters interface 9 – D/A converter interface 10 – In any digital or analog remix Acoustical temporal distortion is introduced in the following points: 11 – Differences between driver velocities 12- Differences between driver mechanical phase in respect to ear position 13 – Differences between speaker positioning. More it will come on this issue. There is plenty to say. Keep in touch and see the below graphs only for understanding purposes. FEED BACK EFFECTS ON SOUND QUALITY Here follows the block diagram of a FEED BACK amplifier and its signal treatment:
This is what really happens - the final result is the last graph Let's consider a small portion of the music signal as shown on the following graph, and let us analyse what happens on the GA point amplified into the small circle. Any portion of the music signal could be taken in consideration, the result of the F.B. analisys does not change.
Now we get the Ga signal and we analise it into a new graph by itself
NO FEED BACK EFFECTS ON SOUND QUALITY Here follows the block diagram of a NO feed back amplifier and its signal treatment:
Lets take in consideration the following graph to understand briefly how things are going.
Please note the difference between signal SR and SR1. This last is much better. Take care that on graphs the things are exaggerated just for drawing clearness purposes. Both signals SR and SR1 could be better depending on the AMP drive capacity. A higher damping factor by the amp would just worst the case. Practically, to simply explain how the “temporal distortion” is introduced, we can say that a F.B. amp intervenes with a certain delay in correcting a situation already happened, provoked by the speaker. Here is where the problem relays. The result is a lot of distortion introduced onto a new signal at amp input. If we do not have F.B. this does not happen, or let’s say that the thing is quite relevant only on the last amp power stage that is recognizing an anomalous situation, but does not try to correct it. This is an important point. In any case, to avoid problems at this stage it is better to correctly choose the type of tube and relevant circuitry, the current flowing into the tube and the output transformer. Another important point using no F.B. amps, is the choice of the type of speaker loading (horn, reflex, etc.), this changes things again. A good approach will significantly reduce any type of distortion introduced by the use of no F.B. amps connected to a speaker. Maybe this type of distortion hasn’t been recognized yet or measured by anybody. The reason is very simple: The “temporal distortion” is not measurable, but it is audible: why? Just because the “temporal distortion” introduced when connecting an amp to a speaker is measurable only as the result of an increased amount of harmonic distortion and inter-modulation distortion. This is the reason we do not see it. It is already into the THD and TIM, but the difference is the total amount and the cause that generates it. Now, the thing is that we normally see the final result and try to look for the causes. But if we measure, we just do it dividing the two worlds and here is the mess. If we measure then we can measure: 1 – the THD/TIM introduced by the amp itself 2 – the THD/TIM introduced by the speaker itself but looking to the final result, we do not think that there is also a: 3 – THD and TIM introduced by the connection between the two!! This is the problem. So at the end, we do not understand the reason why we hear such annoying stuff..if we have not measured it! Is that so difficult to understand? Just to make an example and to give an idea: 1 – if we measure a F.B. amp on sine wave by itself we can find a 0.001 THD 2 – if we measure the speaker by itself on sine wave at a certain level and frequency we find a THD of let’s say 0.5 % Let’s say more or less the same is for TIM. When we start to feed “transient signals” (music: here is the problem) we can measure (if we could do it) let’s say 5 or 10 % of total distortion… or much more. Where does this come from? It comes from the connection of the two things together and from the fact that we are feeding transient signals and not sine waves. We can enter this topic in the future, there is a scientific/mathematical rule for this. The result at the end is: A lot more of distortion than the one measured. So, just adding all the items, we find that the biggest amount of distortion is generated by the two above facts (connections and transient signals). In the case of Amp to speaker connection, the distortion is produced by the big out of phase signals between the two, this is the reason I have called this type of distortion “temporal distortion”, because it is generated by different time functioning. If we consider that all audio chains have F.B. starting from the source to the end (CDP has F.B. also into the digital filters they use and this does not solve the problem, but just swap it into a different domain), we can start to understand how much distortion we are used to. This distortion is as present as the air we breath.
Roberto Delle
Curti Roberto Delle Curti, Aliante president, is born in Novembre 1954 and is an electronic engineer. He has been involved in all electronic fields since 1972 either for analog and digital applications, working in all field as automotive, telecommunications, radio links, medical, computers and audio, as “designer” and “system engineer” either in Hw and SW applications. He built the biggest audio room of the world with the biggest horn subwoofer of the world applying the ALL NO F.B. and NO CROSSOVER starting form the intern side of its CDP to the Speaker, relying onto speaker efficiency instead of electric power kilowatts F.B. amps. He is a musician too, playing guitar and piano, and normally makes comparison only towards “live play” and not toward one audio system to another one. The "real sound" is there to listen at…. http://www.royaldevice.com
By Constantine Soo
From a family
rich in musical training, Roberto developed a strong affiliation with
music and
its stereophonic reproduction, using electronics such as the Marantz 1030,
1060, 1120, 1200 & 250, Acoustic Research speakers such as AR 2AX, 3A, 11,
10 Pl, and last not least, the LST which he still owns.
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