Criterii de masurare THD+N

[Vizitator]

Propun ca baza de discutii un nivel de 0,776v ca referinta pt 0 dB (0,776v/600 ohm/1mW) si o formula de calcul a coeficientului de distorsiuni armonice: K= Rad (Uh2*+Uh3*+...Uhn*)/Uf1Uhsau K=Rad (suma patratica a tensiunilor armonicelor)/tensiunea frecventei fundamentaleNu conteaza daca citirea se face pe o impendanta oarecare din moment ce exista un raport de tensiuni care teoretic trebuie sa fie constante.Daca sinteti de acord cu aceste lucruri haideti sa le discutam cu calm...

[adrian_pic]

The THD is defined by the following formula:

where terms 2..N are the power levels of the harmonics and term 1 is the power level of the fundamental (the pure tone).

 

si ca sa nu se faca confuzii

 

The THD+N is defined by the following formula:

where term n is the noise power level

[hpavictor]

Cat de importante sunt distorsiunile la amplificatoare PP sau la cele SE ?

[Vizitator prutus]

Cat de importante sunt distorsiunile la amplificatoare PP sau la cele SE ?

Pai nu e bine sa fie cat mai neimportante?La tuburi pp au dinamica, se are finete, oricum ai lua-o sunt bune toate si cu si fara distorsiuni THDPana prin anii 70 amplificatoarele la care producatorii se oboseau sa le masoare pe alea thd, ca de altele nu prea stiau, lucrandu-se preponderent pe tuburi, erau de ordinul x%.1% era parfum.Cand au aparut tranzistoarele cu germaniu nu a fost prea mare problema pt ca distorsiunile de crossover erau mici si problemele erau mai mult de stabilitate si fiabilitate precum si de zgomot termic care se suprapunea peste raportul semnal zgomot al radioului, distorsiunile fiind ultima grija cand singura sursa de auditii era radioul cu maxim 55 db raport semnal zgomot....Cand a venit siliciul peste noi, un tumult de impresii a dat cu noi de pamant: crossover la greu, thd-ul de 0.5% suna mai prost decat 2% pe tuburi, solutii primitive si dinamica de doi lei...restul durere, suspine si cel mai important aparat de masura pe care ti-l puteai lua era un glob de cristal in care puteai vedea viitorul: reintoarcerea la amplificatoare cu tuburi sau dimpotriva amplificatoare cu 300 de tranzistori si chestii mici negre cu multe picioare , hibride, pick-up-uri care suna mai bine decat cd-payer-ul care inca nu se inventase, dar mai scumpe decat un Rolls Royce si... un MAR muscat!Cui ii mai pasa de distorsiuni?Am un prieten care a terminat electronica acu vreo 13 ani si acum castiga bani in prostie incat ar putea cumpara in fiecare luna cate un sistem cu 0.0000x% "disensiuni de-ale cum le ziceti voi" si cand ii vorbeam despre vintage s/n si alte chestii mi-o taie:Adica tu vrei sa spui ca poti face un amplificator care suna mai bine decat telefonul meu Nokia?Ia asculta aici un Youtube si mai ia un snitel ca se raceste!Cam atat mai conteaza thd-ul...

[Untold]

Io zic, parerea mea, ca depinde de educatia acustico-muzical-artistica a urechii, ca sa ma exprim asa.Crezi ca n-ar fi nicio diferenta intre urechea unuia care bate cu pickamer-ul toata ziua si urechea lu' Andre Rieu sau Yehudi Menuhin, sa zicem?Si nu numai de ureche e vorba, ci si de organizarea interna a neuronului.Cat despre snitele, doar daca-s cu soia, ca e post...

[Vizitator prutus]

menuhin si celibidache nu ascultau muzica la amplificator ci direct in sala de repetitii ori concerte, mai bine zis nu o ascutau ci o faceau...

[Vizitator]

Din câte înţeleg în acest subiect din această secţiune "se cere" Criterii de măsurare THD+N practic nu trebuie să interpretăm/scriem/compunem partituri...cred ,ne interesează probabil acurateţea rezultată în urma procesului de amplificare a unui semnal muzical(din punct de vedere thd şi zgomot, uneori acest zgomot poate "înghiţi/îneca" o parte din semnalul iniţial util).

[Untold]

menuhin si celibidache.....nu o ascultau ci o faceau...

Chiar si pentru asta iti cam trebuie ureche... Acuma sa nu-mi zici de Mozart (sau Beethoven?) care a compus muzica, surd fiind....

[hpavictor]

Va propun sa analizam un text gasit pe net ( din pacate in limba engleza ) care trateaza majoritatea termenilor folositi in tehnica audio :

 

" Specifications And How To Understand Them

 

There is much misunderstanding regarding specifications, in both directions – in understanding what they mean, and consequently in the ability/inability to interpret their real meaning and implications.

 

There are basically two schools of thought in the audio world today. The so called objectivists claim that specifications are of great importance and that unless something can be measured, it is either irrelevant, or even nonexistent. The so-called subjectivists claim that specifications are all but meaningless and useless, that it’s all down to the human ear.

 

The reality is: both are important. An amplifier which measures well can very possibly sound just dreadful, and that another amp, which by comparison doesn’t measure nearly as well as the first, can sound far better than the first one. But an amplifier which measured terrible probably are sounded less than terrible.

 

Over the years, probably all of us heard it said many times that tube amps sound better, despite the fact that they often measure rather poorly, and can easily be shown to generate distortion levels surpassing 2-3% THD only, not to even mention intermodulation. In the early seventies, it has been shown that the human ear likes even harmonic distortion, but simply hates odd harmonic distortion. Tube audio usually (though not always) tends to distort even order harmonics (2nd, 4th, 6th, etc), which explains why it is liked by most listeners. However, I do not accept the argument who cares if it sounds nice if it makes you feel good. I believe we should be after the truth, as faithful and true to life reproduction as possible, not coloring or letting color just because it may sound nice. This means we need to take into account both the measurements and the audition results. That is easy to say, much harder to do. Specification sheets tell to us a good part of the story if are detailed enough, but can also be misleading. This article will try to outline some possibilities in the text below on preamplifier’s/power amplifier’s specifications.

 

Power output, continuous

 

A continuous power output is a voltage delivered either at some point, like 1kHz, or anywhere from 20 Hz to 20 kHz. You can calculate quite easily what voltage your amp delivers at the output using a simple formula:

 

 

 

Example: we want to know what a 50W/8 ohms amp delivers:

 

 

 

This is useful when looking at a schematic and trying to figure out how much headroom you have over the nominal power before clipping (though you’d need to know much more as well - you need to subtract 0.7V per every transistor, you need to know how much will the supply lines sag (drop) when fully loaded, etc).

 

To obtain the true continuous voltage, simply divide the above result by the square root of 2, or, if you want to work it out straight away, in the above formula omit the power in watts times 2 - forget the 2. Either way, you end up with 20 Vrms.

 

Ideally, a power amplifier should deliver this voltage into a load. Remembering Ohm’s law, this would mean that every time the load is halved (4 instead of 8 ohms), the current must double, and since voltage is constant and the current doubled, the watt specifications must also double (since according to Ohm’s law, watts are the product of voltage times current).

 

You can calculate the required current using the following formula:

 

 

 

If we consider those things, we will be able to see through many twists which are handed on a daily basis by audio manufacturers.

 

Examples: a well known tube audio manufacturer declares his power amplifier delivering 15W into 8 ohms, and 17W into 4 ohms. At first glance, this would appear to be all right, more power into lower impedances, just as it should be. But implementing Ohm’s law, we see that the above really means his power output drops significantly. To hold up, it would have to double into half the impedance, i.e. go up to 30W, yet it delivers only 17W. How big a drop is this? 15W/8 ohms is really 10.95V, while 17W/4 ohms is just 8.25V - hence, we have a net voltage drop of -25%.

 

With A/V receivers, we have a somewhat different game. You have to look carefully at the power specifications, because several very large Japanese manufacturers have started declaring their output powers into 6 rather than the customary 8 ohms. Let’s look at this case numerically. A receiver is rated at 90W/6 ohms. That works out to 23.24V continuous. Square this and divide by 8, and you have an 8 ohm power rating of 67.5W. Well, 67.5 watts doesn’t look half as good as 90W! Oddly enough, this is actually not too bad. 67.5W/8 ohms is 4.11 A peak, while 90W/6 ohms is 5.48 A peak. Since no speaker is a pure resistive load, this tells you the amp will deliver the goods with nominal 8 ohm speakers which in real life are more like 6 ohm speakers, and there are quite a few of those around.

 

The other three aspects we should look for are at what frequencies, at what distortion levels, and the number of channels used at the same time.

 

At what frequencies simply asks whether the power rating was made at the traditional 1 kHz point, or was it swept across the audible spectrum. It is quite normal for power amps to have power drops at the extremes, usually more so at the bass extreme (where most of the energy requirements are, and where output stage and power supply weaknesses will show up first). Therefore, if the measurement was made at 1 kHz, you can bet there will be less power on tap at 20 Hz, but how much less, you are left to guess and gamble on with your money. However, if clearly stated “20Hz-20kHz”, then this means the power will not drop below the stated level, though it will still be more in the midrange than in the extreme bass. Obviously, this is a much more fair and telling specification than the first one.

 

The other factor is at what distortion levels is the said power produced. Transistors in general tend to distort more as more current is required from them, and in real life, this means that the distortion will increase with the decreasing of the load impedance. 4 ohm power ratings will tend to be at greater distortion levels than 8 ohm power ratings if the manufacturer wishes to stress the power output, or, if he wishes to stress low distortion, the watt difference between 8 and 4 ohms will tend to decrease while the distortion levels will remain quite small. This is why power amp manufacturers like to use parallel/series pairs. Since power outputs are given values, using more transistors to share the work among them also tends to reduce distortion, noise and increase available impulse power outputs into lower value loads. The downside is increased cost and the need to match output devices rather closely, again increasing the price.

 

The last aspect is how many channels are simultaneously driven. In their effort to appear better than they are, some manufacturers, notably those from the UK, give specifications for only one channel being driven at just one frequency, typically 1 kHz. We would call this practice less than honest, because it’s downright misleading. You may well discover that an amplifier rated in this manner at say 105W/8 ohms, drops down to say 90W/8 ohms with both channels driven, and down to say 75W/8 ohms with both channels driven 20Hz-20kHz. This happens because if only one channel is driven, the power supply designed for two channels caters for only one, and is thus under far less strain than it would normally be.

 

Power output, impulse

 

At a glance, this would appear to be a redundant specification, one of those half mad advertising ideas designed to make something look good. However, it’s is not so, this is a very handy specification which tells you quite a lot.

 

To spite everybody, German DIN 45500 standards did specify one version of this something like 35 years ago, the so-called music power. In general, this is simply short term power an amplifier can deliver without clipping and at a set maximum distortion level, like 3%. The logic is as simple as it is flawless - it’s much better to have higher distortion, but be able to let the amplifier reproduce something than to have it clip, or current limit.

 

The only caveat to this is defining what exactly is “short term power”. IEC standards, for example, define this as 20 milliseconds or less. Old DIN 45500 standards were more stringent - they called this pre-clipping power, 1-in-4 cycle (meaning three cycles up to nominal, one cycle absolute preclipping maximum). This is better simply because it is repetitive, and hence includes power supply recovery time as well, which if not solid, will limit second and all other subsequent bursts.

 

The current IHF standard, concocted by Edward J. Foster while he chaired IHF, is also good and compares almost exactly to the old DIN standards, though his logic was somewhat different. Whatever, it’s a solid specification and approximates real life better than usual. It becomes especially useful when it lists dynamic power into 2 ohms, as well as the customary 8 and 4 ohms. This will give you a good idea of how the amplifier will behave when it runs into some nasty load, which in real life could still be a great sounding speaker

 

Some manufacturers do provide specifications for 8, 4 and 2 ohms, some don’t. It doesn’t take much knowledge to realize that those who don’t probably have current limiting of some sort built in which acts faster than with those who do give the specification. However, this is almost always due to one, or both in combination, of two factors - relatively weak output power stages and/or weak power supplies. While you are not likely to hit such serious loudness levels unless you are a headbanger, you would still have to live with the knowledge that your amp has been the object of significant savings, despite the price you paid for it.

 

Example: a well known Japanese firm sells integrated amps rated at 2x95/140W into 8/4 ohms using two pairs of 100W/8A devices per channel, while another famous, this time US manufacturer, sells his integrated amp delivering nominally 2x85/130W into 8/4 ohms using two pairs of 150W/15A devices per channel. It should come as no surprise to you that the first doesn’t even rate his amp for 2 ohms, while the second specifies his with 2x270W into 2 ohms as IHF burst power.

 

There is a third group of rather odd specifications, coming from Scandinavia. Some manufacturers there do declare their burst power outputs, and not just into 2 ohms, but into 1 ohm, and even 0.5 ohms. Most impressive you say, but look again - their impulses last all of 0.5 milliseconds, or just 1/40 of what IEC requires. 0.5 milliseconds, or 500 microseconds, is such a short time that this specification is basically meaningless, and is designed to mislead the buyer, making the unit appear much more powerful than it really is. Their opinion is: that was given not as an indication of power, but as an indication of stability, saying that the amplifier was stable down into 0.5 ohms! I would just love to see their stability with impulses of the nominal 20 millisec duration, and one will get you ten they would either oscillate, or a fuse would blow somewhere (unless the protection switched it off first, or something).

 

My personal view is that even 20 milliseconds is far too short, that this should be something like 100 or 200 milliseconds (I’m not the industry and am hence not shackled by economic considerations in my constructions).

 

All in all, make sure you carefully read the impulse power ratings if given at all, make sure they are meaningful and if you want anything of a load tolerant amplifier, make sure 2 ohms ratings are also supplied.

 

THD

 

Every sound we hear consists of the base frequency, say 1 kHz, and its harmonics. Harmonics are multiplied basic frequencies. If multiplied by even numbers, we call them even harmonics (in this case, 2, 4, 6, 8, etc kHz), and if multiplied by odd numbers, we call them odd harmonics (in this case, 3, 5, 7, 9 etc kHz). At the risk of oversimplifying, basic sounds define the volume, and harmonics give the “color” to the sound, they supply the additional information our ear interprets as everything else but the sheer volume.

 

As stated before, the human ear is especially sensitive to even harmonics which it likes, and odd harmonics which it doesn’t like, in both cases if distorted. Distortion here means changing the volume, relationship and form of the harmonics; for example, if the amp distorts the second harmonic, but leaves the third more or less undistorted, subjectively, we will probably find the sound very pleasing, if a little odd. However, if the second harmonic is relatively undistorted, and the third is rather distorted (relative to the second), we will probably find the sound to be unpleasant, harsh, etc. Remember, things are not quite so simple, but in general, this is the way it goes.

 

In the early seventies, in a series of texts published in IEEE, professor Matti Otala, the Finnish amplifier guru, proposed that the best amplifier will distort only a little, and then equally across the harmonic spectrum. The small overall distortion is clear enough - ideally, an amplifier should not distort at all, so we obviously want amplifiers to distort as little as possible. However, his second thought is of equal importance - if it must distort (as it must, since we have no perfect semiconductors), then you should tweak it so it distorts equally. The point is, it should not change the structure and relationship of harmonics. And herein lies the problem, as this is not at all easy to do.

 

Traditionally, amplifiers were run at great amplification factors, which caused them to produce open loop (without any feedback) responses up to only several kHz. Then much feedback was applied to reduce distortion, and to extend the frequency response. However, any signal entering the amp which was above its open loop point would cause distortion, which was both harmonic, but much worse, also of intermodulatory nature (called Transient Intermodulation Distortion, or TIM - see next group). The way to go is to reduce the amplifier’s open loop gain factor, which will also reduce its inherent open loop distortion and make its response wider to start with, then use moderate to low feedback not to correct what’s wrong, but to iron out further what is already good.

 

This is, as usual, easier said than done - but it can be, and is, done. In fact, there are amps out there which use no overall (output back to input) feedback at all. My very personal view is that amps with zero overall feedback never sound very good; feedback as such is a very useful thing, provided you don’t misuse it to cover up your design incompetence. But this is a general truth, isn’t it? You can take the best of anything and misuse it, just as you can take the humblest of things and put it to extraordinarily good use.

 

My design philosophy is: make it good and wideband to start with, then apply moderate to small overall feedback just to iron it out, to improve it that little extra bit.

 

If you check the amplifiers schematics which can be foud on the market, can recognize their feedback is 20-26 dB for modest returns, but for example the H/K’s 6550 integrated amplifier give a wide bandwidth with just 14 dB (5:1) of overall feedback and for that reason it simply sounds better than all others.

 

This should have carried the message that unfortunately, we don’t know HOW low THD specs were achieved. Obviously, on two extremes, we have a downright rape of the electronics by massive feedback, and on the other side zero overall feedback. And it’s in between that what you should look for lies - unfortunately, you have no way of being sure.

 

There are a few pointers, however. When you see THD specs like say 0.09%, down to 0.05%, you should make a note of that model. When you see something like 0.01% or less, you should avoid such models if they are cheap. Next, still on the same subject, look at the power specifictaions - if impulse power is quoted into 2 ohms, it’s reasonably safe (though not guaranteed!) that amplifier is using moderate to low negative feedback. You see, most of the distortion in any power amplifier section comes from pre-drivers (if present), drivers and output transistors. Now, you can push that output transistor pair just so much, so if you have large power outputs into 2 ohms, you can safely assume there is more than one pair of transistors at work. But if so, then into easier loads, the work is divided between the pair, and hence, distortion tends to be smaller, since each transistor is kept more within its linear operation range. And if so, then you don’t need much feedback to keep your house clean.

 

Back to H/K 6550 example. Its rating is: 50W into 8 ohms at 0.09% THD, 70W into 4 ohms at 0.3% THD. This told me two things, which I checked and subsequently verified. One, they are using just one output pair (2SC3281/2SA1302), because the difference between 8 and 4 ohms is not. And two, the rising THD into lower loads PROBABLY means moderate feedback (and yes, they use only 14 dB of overall feedback, which is just 5:1).

 

Now about the H/K 670. Its rating is 80W into 8 ohms with 0.09% THD, 120W into 4 ohms at 0.3%THD. But its impulse power into 2 ohms is 310W. Using the same logic, this is another low feedback design, but this time round, with at least two pairs of output devices - no way they could squeeze that much 2 ohm power form just one pair without frying it. A look inside and at the schematics confirmed this - two pairs of Toshiba 130W output devices (2SA1962/2SC5242), but because there are two pairs sharing the load (even if it is more work here due to greater power rating), overall feedback is even lower at just 10 dB (only 3:1).

 

I repeat, this is a very loose logic, but it does work up to some point, and despite its limitations, it does allow you to deduce much from a standard specification sheet. To sum up - when you see low, low THD values, there is a good chance this spec was achieved by increasing overall feedback, and that is not good. Stay away from that, whatever it is.

 

When you are offered a zero overall feedback amplifier, don’t rush out to buy it. Most such designs are rather load sensitive, because overall feedback also reduces an amplifier’s output impedance, thus improving the damping factor. Also, the term “zero overall feedback” has been much misused lately.

 

IM (Intermodulation) Distortion

 

This is a specific form of distortion in which two or more tones by adding and subtracting produce new tones, not found in the original signal. For example, let’s say we have a 3 and 4 kHz fundamental tones which the amp must reproduce simultaneously. We would say it intermodulates if in the resulting output signal, beside the fundamental 3 and 4 kHz tones, we found (4-3) 1kHz, (3+4) 7 kHz, (3-1) 2 kHz, (7-1) 6 kHz, etc tones. As you can see, the two basic tones produced their difference and sum, but then these started intermodulating, and so it goes.

 

This may look appalling, but in real life, while this does happen, further variations tend to be of smaller overall value and decay. But make no mistake, EVERY amplifier ever made intermodulates, the questions being how, and how much, not if.

 

If the level is below 0.1% worst case, we’re home and dry.

 

In general principle, this form of distortion is also connected to overall negative feedback, but also very much to the overall design of the amp and especially its overall margins. You want a number below 0.1%, but again, if you come up against a really wild specification, like say 0.008% at rated power, either somebody is raping the amplifier with negative feedback, or is using paralleled output devices, and many of them - in which case it will be expensive.

 

TIM, TID

 

This applies to Transient Intermodulation Distortion. Basically, this is much like steady state intermodulation distortion, the difference being that this form of intermodulation occurs with transients only, and cannot be seen or measured using classic steady state measurements. In general terms, it occurs due to inadequate amplifier speed, when a fast and sharp transient simply throws it off balance. It is also caused by the time delay (lag) between the input stage and overall negative feedback, when the necessarily delayed correction signal is too late to correct the problem, and goes on “correcting” a problem which is no longer there.

 

Obviously, this is in direct connection with open loop bandwidth (i.e. amplifier response without any overall feedback) and overall feedback. If the incoming signal is below the open loop bandwidth limit, there will be no TIM, since the amplifier has a bandwidth large enough to pass the signal without having to wait for the feedback. The sure-fire method is to design an amplifier with a wide open loop feedback, then install an input filter which has its turnover (cutoff) frequency below that point. This way, no incoming signal of reasonable amplitude will ever be able to upset the amp, which will always have an open loop response wider than the widest incoming signal, and hence, there will be zero TIM.

 

H/K uses this general principle, but of course, it’s hardly the only company doing that. I have seen some Japanese amplifiers using the same principle, however, they were of the more expensive kind.

 

To sum up - look for a TIM rating, though you’re not at all likely to find it, most don’t give it. If they do, it’s safe to assume that’s a decently built unit probably worth the money.

 

SID

 

SID stands for Slew Induced Distortion. The rate of exchange (or “speed”) of the incoming signal may be greater than the capabilities of that unit. If that happens, the preamp/amp will slew limit, i.e. artificially slow down a fast signal, causing a form of compression - or worse.

 

You may think this is here for historic purposes only, but it’s not. While this problem is indeed greatly reduced today, it is far from gone.

 

In the early eighties, the name of the game was slew rate. Companies, and especially Japanese companies, were deadlocked over who will have a better slew rate. They eventually invented very novel methods of measuring slew rates, Sansui being the undisputed leader in this field. Their AU-919 models boasted slew rates of 350V/uS which any designer will tell you is awesome - and in those days, they didn’t have any of the high speed output devices we have today.

 

The trick was that a “new” logic was introduced. Somebody somewhere in the Far East decided that it was really all about input stage speed, and if that didn’t overload, all was well, because any overload downstream was not as bad. So they produced input stages with tremendous speeds, but I’m sorry to say, that didn’t improve the sound of their products. In fact, they staid about the same, if not taking a downward turn.

 

I think you can deduce this yourself. If I need a speed of say 10 V/uS, and my input stage has a speed of 350 V/uS, then obviously I don’t have a problem. But what about the rest of my amp? What good is a high speed stage coupled to a slow, slow output stage? No good at all, of course, because the end result will still be slew limiting, only it will be shifted to the output stage. Therefore, we can conclude that in order not to have and SID, we need a preamp/amp faster than the fastest input signal.

 

Fine, so just how fast is that? How fast is fast? Let’s do some math. Let’s first see just how much speed we really need. If we say the audio bandwidth is up to 20 kHz, for safety’s sake, let’s assume we want a 40 kHz bandwidth at the very least.

 

Now, let’s assume we want 100W into 8 ohms. That works out to 40 V peak. This means our maximum required slew rate is all of 14.17 V/uS - and that’s assuming twice the audible bandwidth. And that’s for a 100W/8 ohms power amplifier. For a preamp, required to deliver say 2V in peaks (assuming power amp input sensitivity for full rated power is 1.5 V), this works out to 0.5 V/uS. Now, there is nothing wrong with having more to much more speed than you will need - it’s nice to have much to spare. But what I am saying is that crazy specifications such as those given by say Spectral should be taken with reserve - great reserve. Working backwards, it turns out they have some power devices the likes of which nobody has ever even heard of, much less seen.

 

You can actually calculate the slew rate, though I must warn you at once that there may be hidden aspects inside the amplifier which will offset your calculation. All you need to know is the preamplifier/amplifier’s power bandwidth, or its response at full rated output, and its rated output. First calculate its output voltage as per formulas given above.

 

When you have that, use the next simplified (rough) formula:

 

Voltage x 0.141 x 0.628 x Power bandwidth limit taking 10 kHz as one unit.

 

For example, say we have a 100W/8 ohms amp with a power bandwidth of say 150 kHz. Since 100W/8 ohms is 40V peak, we have:

 

40 x 0.141= 5.64x0.628= 3.542 x 15 = 53 V/uS.

 

Remember, this is no precise method, just a rough formula to give you some idea of what you can expect.

 

Memory distortion

 

This is a relatively newly discovered or formulated phenomenon. It has been researched by Lavardine of France and Lundahl of Sweden, but others are also working on it.

 

In greatly simplified terms, this is basically a thermal effect, but a hidden one. When a transistor has a fast transient passing through it, in passing it will work much above its normal levels, and will consequently heat up. However, since it is a fast transient, this heat will not have time to spread around, and will stay localized to the semiconductor substrate, while the outer casing will give little, if any indication that there was some heating up. Yet, because the substrate did heat up, its operating characteristics will change and will stay changed until it cools down, and since cooling down is a gradual process, its operating characteristics will also gradually slide as it cools down. All of which means the preamplifier/amplifier will be in a sliding, changing, pulsating mode.

 

There are methods to overcome this problem, but they have not been quantified yet, and hence there is no specification figure anyone can give you. I included it here just in case you do see some reference to it in the literature. For more on this, visit http://peufeu.free.fr/audio/, a site put up by Pierre-Frédéric Caillaud.

 

Frequency response

 

Quite simply, this is the frequency range some unit will respond within. It is customarily given at its cutoff -3 dB points, though many will also say what it’s like in the 20 Hz - 20 kHz region. This is, in fact, the preamplifier/amplifier’s basic operating range.

 

In case of preamps, it is usually given at nominal output levels, typically 1 V (since this is most often what is required to drive a power amplifier to full rated power, though some require more). In case of power amps, it is usually quoted for 1W/8 ohms.

 

Sometimes, you will find as a small signal bandwidth (as opposed to power bandwidth, which will then be called large signal bandwidth).

 

Obviously, when running anything at a fraction of its nominal power, it will display better results than when driven at full blast. Also, this could be greatly misleading - for example, some people design their product for 1+ MHz bandwidths (Swiss company Goldmund hitting 3 MHz), but then intentionally limit the input signal to say 100 kHz. This does help in reducing or even eliminating TIM, but if measure input to output (not always the case!), you will be informed that the response is just 100 kHz at -3 dB.

 

Modern practice has it that the frequency response should be no less than 150 kHz at -3dB point because of the new wideband sources, such as DVD-A and SACD. 300 kHz is not at all unheard of even in the economy sector (e.g. Sony). A wide bandwidth means low phase shift in the audible sector, but as mentioned above, it’s possible to have misleading results.

 

What is the reasonable upper limit? Ask that, and you will get as many answers as you ask designers of audio. I know of no definitive answer, so I can only offer my own view. For preamps, anything over 500 kHz is basically nonsense, because with a bandwidth of 500 kHz, your phase shift in the 20 Hz-20kHz range will be less than 0.3 degrees, or ten times below the arbitrary hearing threshold. Your frequency deviation at 20 kHz will be less than 0.08 dB, which is way below hearing levels. For power amps, about the same holds true, though in their case, I would put in an input filter limiting the incoming signal bandwidth to about 150 kHz or so. Therefore, frequency response as such is of rather limited information value, but should not be overlooked.

 

Power bandwidth

 

This is a far more meaningful specification than frequency response. That’s because it tells us how a unit will behave under full nominal drive, not a fraction of its drive, and it doesn’t take a genius to realize at a fraction of the drive any unit will have better specs than at full drive, when everything is stretched to the limit. As noted under the SID section, you can work out the rough values of power amplifiers on basis of this parameter and its known output power.

 

Note however, that the power bandwidth is under great influence of the load impedance. It will be best for 8 ohms, but will deteriorate for 4 ohms and lower loads. By how much is impossible to tell, because it depends on any specific design. Yet it will deteriorate, something you need to remember if you go for say 4 ohm speakers. You can expect about 30% reduction of the power bandwidth with 4 ohms in comparison with 8 ohm loads. This is a very rough estimate, remember. God alone, if even He, knows what will happen with 2 ohm loads. I can’t help being a little cynical here - odd that those optimists who rate their amps into 0.5 ohms loads never mentioned their power bandwidth even into 2 ohms, let alone 0.5 ohms.

 

Damping factor

 

Damping factor shows the difference between an amplifier’s output impedance and the nominal load. For example, if the amplifier has an output impedance of say 0.1 ohm, and the speaker has a nominal impedance of 8 ohms, then the damping factor will be (8:0.1) 80:1.

 

How should it be measured and declared? Ideally, 20 Hz to 20 kHz, but few give it that way. Most simply state a figure, some say at what point, but that’s about it. The reason is simple - it usually starts off as a low value at 20 Hz, but starts to rise about 1-2 kHz. This happens because the feedback starts to be reduced in value, but also because of the usual output inductor and load compensation circuits. However, high quality designs, with large open loop bandwidths, demonstrate much smaller deviations and therefore, it’s mostly such designs which will give a full bandwidth figure. To the best of my knowledge, in the economy sector, only Yamaha quotes this value on a 20Hz-20kHz basis, and it’s usually a very good figure, too. The reason for this is some special circuitry used inside, but for you, it’s good news all around.

 

Why is damping factor important? Because it tells us two important things. To be able to implement a good transfer function (transferring the signal from the amp, via cables to the speaker), we need to have a worst case difference of about 1:10, i.e. the load impedance needs to be at least ten times greater than the source impedance. If that is so, deviations due to load variations will be on the small side. Unfortunately, in case of audio, we have additional variables we are hard pressed to control, namely the cable. As a part of the signal path, it has its own inherent combination of impedance, inductance and capacitance, all of which WILL influence the signal fidelity, rest assured. How, and to what extent, is anybody’s guess, and depends on any specific situation and components. Also, cable runs are of importance - it’s hardly the same thing if your cable is 2, or if it’s 6 meters long. That cable appears as a series impedance with the amplifier, so if it happens to have a highish impedance, and is long to boot, you may have a greatly deteriorated damping factor, resulting in a loose sound, possibly with overhang.

 

As for the power amplifiers, their damping factor is the result of several factors at work - number of power devices in parallel (when two are put in parallel, their output impedance halves), power supply output impedance (this includes both the power transformer and the filter capacitors, though their overall behavior is rather more complex), open loop response and negative feedback levels. All of them result in some damping factor spec, and you have no way of knowing just how.

 

Also, the amplifier’s output impedance has something to do with current delivery; the higher the output impedance, the less current it will be able to deliver, though this is not a linear function, and admittedly, it’s not too important these days. Finally, a good damping factor enables the amplifier to influence the speaker’s Q (quality) factor. In very simplified terms, a high damping factor means that the amp will grab the speaker and will not let it go, will not allow the bass driver(s) to go into overhand and to behave flabbily. Unfortunately, it does not guarantee good sound. Too many amps were on the market with wild damping factors which still sounded just awful.

 

By and large, go for damping factors of 100 and more to 1 for 8 ohm loads. If you accept the minimum of 10:1 as the lowest allowable value, then also assume that value into 1 ohm. That would mean 20/40/80:1 into 2/4/8 ohms. With an 80:1 damping factor into 8 ohms, you should be safe and sound. Anyway, most modern power and integrated amps in the serious class (i.e. 50+ watts per channel into 8 ohms) have damping factors of 100 or more, at 80-100W per channel this being at typically 150 and more. Rotel offered a few models with damping factors of 1,000:1, a figure which I, with all due respect to Rotel, seriously doubt you will ever actually see in real life. Remember, the amp’s output binding posts are not the end, there’s the cable in between.

 

Load impedance

 

Load impedance simply denotes the impedance of speakers the unit is rated for. This is typically 8 ohms only on el cheapo units, 8 and 4 ohms on anything attempting to be better and upwards. To the best of my knowledge, nobody rates their amp for less than 4 ohms, though some do give specifications for 2 ohms and less. While their amplifiers may be able to handle that without blowing fuses, I doubt the actually encourage you to use two 4 ohm speakers sets in parallel.

 

In case of preamp outputs, this also denotes the minimum load impedance in kiloohms the preamp can deliver its rated output to, with stated performance figures.

 

Slew rate, voltage

 

This was covered in the above section on SID.

 

Slew rate, current

 

Just as there is a rate of exchange for voltage, so there is a rate of exchange for current. You will very, and that’s “very” as in VERY rarely see anyone specify this. For most, it’s because it would be a rather small number, and because specifying anything is like taking an oath. Most are not willing to take the oath, because they know only too well what they are making and selling you. Furthermore, to be able to specify it meaningfully, you need to have great control over your production, always using the same quality components - and component consistency is THE manufacturing problem today. In the everlasting search for ever lower prices, many technical specifications are simply overlooked or ignored. This unfortunately applies to much revered component manufacturers as well.

 

I have been in the situation to have a high quality, very expensive capacitor right alongside an all but unknown manufacturer. Because I was under the subjective impression that the very expensive cap wasn’t doing anything significantly better than the cheap one, I started measuring them. What I got was so similar, with most differences down to +/- 1%, that I was appalled – the famous name one costs 2.7 times the cheap one! That was when I said goodbye to the Far East and went back to German manufacturers, with whom I at least know what I am paying for.

 

Basically, current slew rates would be expressed in amperes per microsecond. I have seen this a declared a few times, like 9 A/uS. It would tell you how fast the amplifier can track the signal in terms of current. Declaring this should, in my view, be as binding as declaring voltage slew rates.

 

Rise time

 

Rise time tells you how long it takes a unit to rise from 10% to 90% of its output as a reaction to the incoming signal. While hardly unimportant, this is not too important a factor after you get it below a certain limit. Specifically, a 100 kHz signal has a rise time of around 7 microseconds, so anything smaller than that is good enough.

 

However, figures can be misleading here too. It all depends on HOW it was measured - with or without input and output filters. Obviously, filters will slow the unit down, never mind if it can actually do twice as fast; any filter will do that. Some manufacturers, such as for example Harman/Kardon, will specify this WITH the filters on line, and this is creditable practice, because that’s exactly how you will be listening to the unit, with the filters on line. Less reputable manufacturers will simply give you a figure and let you guess how they got it - so, its’ very safe and prudent to assume it was measured without the filters on line, and obviously, somebody is trying to pull wool over your eyes. Most will play deaf and dumb and will state nothing - you work it our why.

 

S/N ratios

 

Quite simply, this tells you the difference between the noise floor and the loudest signal you can have, assuming the signal you want is at least 6 dB, or twice as loud as the noise. Obviously, you want as good S/N ratios as you can get. Bear in mind there are two general types of specifications quoted. There are linear specs, those which measure absolute values as is, and there are weighted specs, which use some form of filter in an attempt to simulate our hearing (in full knowledge that we do not have linear hearing, much to the consternation of the high end audio gang, which dumps out tone controls in an attempt to convince us that we do have linear hearing - but who regularly use weighing to improve their linear figures). In real life, the “A” weighing filter is generally accepted and used, though pros tend to use CCIR (Consultative Committee of International Radio) weighing, which I also believe to be superior. But life is life, and the “A” filter is the king of the road in consumer electronics, and that’s that.

 

Anyway, for line inputs, you should look for at the very least -90 dB A weighted figures. Remember that the 16-bit format corresponds to -96 dB A S/N ratios. You will see specs going down to -120 dB and more A weighted - but referred to full rated power, which is something like 200W per channel. Nice, but hardly practical, and unfortunately, few manufacturers will give you the figure that is meaningful to you, the one at 1W output power. Why? Well, obviously because that will be a much poorer figure, which will not sell well.

 

To be fair, I must say a number of paper mags have started measuring this at 1 W output, which is creditable practice I wholeheartedly support. In real world terms, this is far more important to you than at full blast because you will NEVER listen at full blast - try it and you will get a whopper of clipping which will burn your speakers to ashes inside of 10 seconds.

 

At 1 W, S/N ratios vary from about -80 to about -92 dB. However, those 12 dB mean 4 times more silent. The only thing you can do about improving your S/N ratio figures without rebuilding your equipment is to buy a decent line filter. If good, it will improve your S/N ratio quite significantly, and you won’t need any equipment other than your ear to verify this. I would guess that with some units, that’s your only option, that or changing the unit.

 

Sensitivities

 

This tells you how much voltage across how many ohms or kiloohms you need to have for full rated output power. On the face of it, this looks rather immaterial to a buyer, but you might find it’s not so at all. Basically, much matching can be done here. The reason is that to the equipment designer, this is necessarily a big trade-off. On the one hand, he’d like to have sensitive equipment with very large input impedances because then he’d be sure he has great signal transfer with little loss, but this would also open the door wide to all sorts of airborne and other interference. This fact makes him reduce input impedances and lower sensitivities, but on the other hand, he can’t go too low else he will introduce signal transfer losses.

 

So they compromise. Typical input impedances vary from 22 to 36 kiloohms, though you will find as low as sub-10 kiloohms (Monrio) and above 100 kiloohms (Electrocompaniet).

 

You need to look at this for three reasons. One, the obvious one, is to make sure your other electronics can supply the required voltages. Two, to make sure your input impedance is at least at the lowest point suggested by the output device manufacturer, so you don’t make the upstream unit work too hard, or above its rating. And three, because this can help you match good interconnects well, or at least better than without even looking at this, and consequently buying blindly and wildly.

 

Example: say you have an output impedance from a CD player of the nominal 2V, with a source impedance of say 500 ohms, while your preamp input sensitivity is 1V and input impedance is say 47 kiloohms. Regarding voltages, no problem there, you have twice what is required. Regarding impedance matching, your 500 ohms is 94 times smaller than your input impedance, so you hardly have any problem there. Now, you know you have a highish input impedance, so if you pick an interconnect cable with a say highish capacitance, you could experience some cut at high frequencies. If your cable has a capacitance of 50 pF (effective in your length), with a 47 kiloohm impedance it will start to modify your frequency response at 6.8 kHz, well within the audible range. Obviously you will need to change cables if you want a fully linear signal transfer. And this was just capacitance, there is also the inductance, plus their interaction.

 

By contrast, if you cut your cable to half, it will have 25 pF of capacitance. If your input impedance is say 30 kiloohms (very typical these days), response modification will occur at 21.2 kHz - outside the audio band (where your digital brickwall filter is probably wreaking havoc anyway).

 

Overload margins

 

This tells you about the maximum allowable levels at the input before the unit saturates and starts to distort badly, possibly even clip. In most cases, this is not a problem with line level inputs, though exceptions do occur, my Yamaha amplifier being a strange one. When both defeat and CD direct buttons are pressed, its CD input will saturate at around 2.8 V! This is way too low, but if I switch off the CD direct button, the overload level climbs to around 12 V, which is more than enough. Yet, that same Yamaha will deliver a 1W S/N ratio of -92 dB “A” weighted! Silent as death. Adding a line filter will improve this already outstanding figure, especially so for a cheap mass produced product.

 

In case of phono headamps, this becomes an important consideration. Just to be clear on this, I will not analyze the RIAA curve, but will limit myself to the 1 kHz point, assuming it’s made to the standards. Maximum voltage should be whatever your cartridge outputs/cm/second, times maximum tracking ability. In case of my Ortofon, it deliver 0.7 mV/cm, and will do 37 cm - hence, its output voltage is (0.7 x37) 26 mV. And it would probably be so if vinyl was as hard as steel, but it isn’t, so I have pops and cracks, which will give outputs 2-4 times the nominal quite easily. So, being the old sceptic, I go for the max, 4 times - 4x26=104 mV at 1 kHz. Thus, I would not buy any headamp, much less make one, which would saturate at less than 120 mV, and preferably more. That’s for MM cartridges.

 

For MC cartridges, the situation is much the same if they output a fully amplified RIAA corrected signal as a linear signal, but is different if the MC section is merely a headamp used to boost the uncorrected RIAA signal to the following MM RIAA headamp (God, how I hate that sort of hodge-podge slap-it-together stuff!). There, you have to worry about your MC pre-preamp and your phono headamp!

 

In both cases, you need as good a S/N ratio as you can get, but because of the lower levels of input signal and the required frequency correction, don’t expect figures like those of the line inputs. An MC spec of -75 dB “A” is acceptable, however, -78...80 dBA is better. An MM figure of -84...86 dBA is good.

 

Weight (mass)

 

Weight? What’s weight got to do with sound? More than you think. Consider. Say somebody is trying to sell you an integrated amp weighing say 7 kilos, and with an output power rating of say 75/125W into 8/4 ohms continuous. So, let’s see where we stand. A 300 VA TOROIDAL transformer weighs about 3 kilos. Four single, or two heat sinks of any quality will add another 2 kilos. Filter capacitors, stand-by transformer, relays and other auxiliary electronics will contribute with 1-1.5 kilos. Connectors and fundamental electronics with pots will add

 

another 1-1.5 kilos. So, add it up and you have 7-8 kilos, and the case is nowhere in sight. Assuming the empty case weighs in at just 1.5 kilos including the aluminium front plate, that amp would have to weigh at least 8.5 kilos. So we have about 1.5 kilos missing.

 

This means smaller and flimsier heat sinks, possibly a lower power transformer rating. Whichever, forget it, it’s a mass produced product not likely to be able to deliver any serious power levels.

 

The message is this - very lightweight, yet nominally very powerful units just don’t cut it, too much saving inside. But this can work the other way as well. I have seen units weighing say 20 kilos, where the cases were made of very thick aluminium (5 and 10 mm), where the cases alone weighed around 8-9 kilos. Very impressive in terms of size and finish, but that’s some damn expensive aluminium you are buying there. A case like that will cost around Euro 200-250 all by itself, and by the time it reaches you, the consumer, using the typical multiplication factor of just 2.5, you end up paying no less than 500 Euro for just the case. So, try to peek inside, to see if you’re buying a lot free real estate, or if there is actually some electronics inside.

 

Dimensions

 

Much the same goes for the outer dimensions. If it’s a small yet nominally powerful case, something is amiss - they might be using switchmode power supplies, or are using hybrid integrated amp modules which require very little space, or some such. One thing is for sure - their heat sinks are not much, small heat sinks simply cannot cool off bags of power.

 

Prices

 

Touchiest subject of them all. Many, many variables. If somebody asks for Euro 350 for a 2x100W/8 ohms amp, be very careful. There are some around actually worth the money, but they are oh so few. To work properly, 2x100W/8 ohms simply costs money. Unfortunately, with a few notable exceptions like NAD and Rotel, most feel that centiwatt amps are upmarket products which must be packaged fancily, and hence expensively. Yes, NAD products look VERY bland, but they perform, they were intentionally made that way. Rotel is a bit better in looks, but just as good in performance. Yet, their hundred watt products cannot be bought for less than Euro 500, about the lower limit of what is possible with serial production.

 

On the other hand, I sometimes wonder what the high end people put in their products - I already know about capacitors hand wound only by virgins in a special village on the Andes, and only with a westerly wind. Is it the power devices? No, I pay retail prices of only Euro 3.5 for a pair of Toshiba 150W output devices. If I was a manufacturer, I’d be buying them by the hundreds every month, and paying less than Euro 2.5 per pair too. A single unit custom wound toroidal transformer, rated at 400 VA, costs Euro 40, so Euro 80 for two. OK, heat sinks are expensive, Euro 35 each, so Euro 70 for two. So far, I just spent Euro 150. A decent case retails for Euro 40-45, say 45. Capacitors Euro 8.5 each, and I need 8, so Euro 68 for that. Electronics Euro 50 at most, power transistors Euro 28. All told, Euro 341. Say Euro 400 everything included. Also say labor costs would work out to what I’d save if I was buying wholesale as I would be in production. Add Euro 200 for all sorts of costs (administrative, advertising, etc) and it’s still Euro 600. Times 2.5 to get it to EEC customers, and I have a street price Euro 1,500 for a well made 2x100/200/350W power amp into 8/4/2 ohms.

 

All I can think of are two things - one, price of labor is so high in the developed countries that it’s really painful, which is probably why everybody is moving out to Asia for manufacturing. And two, advertising costs are way higher, as this includes graft in its many forms, from “gifts” to outright payoffs. I’m not suggesting everybody does it, but I am saying many do it. We live in merciless times. Consumer pays.

 

Note:

 

This text is NOT intended to be technically precise and/or all embracing. It has been put in loose and popular terms, its intention being to offer only an insight into the matter, not to educate. "

 

Din pacate nu mai stiu de unde anume din internet l-am salvat ...

Sper sa nu apara un autor care sa se laude ca a descoperit raza gaurii din gogoasa !

[hpavictor]

Va mai prezint un alt articol , de data asta de la Audio Note , care dezbate problema efectelor Reactiei Negative :

 

" The Negative Effects of Feedback

 

 

The Problem

 

 

Before his successful invention of recorded music, Thomas Edison had to arrive at the fundamental realization that sound can be entirely characterized in two dimensions. His first cylindrical recording was nothing more than a rough approximation of the changes in pressure amplitude (caused by the modulation of his voice) plotted against a constant timebase (generated by the steady turning of a crank). Crude as his technique may have been, sound was recorded and quite recognizably, reproduced. The limiting factor in Edison's first experience was not his idea, but his hardware; further refinements in the field of audio could come only from improvements in technique.

 

Unfortunately, in the more than hundred years that have passed since the birth of recorded sound, engineers have clouded the simple definition of sound as changes in amplitude versus time by the use of such irrelevant (to music at least) ideas as frequency, phase, harmonics, intermodulation, etc. Such factors serve only to complicate the basic function of audio recording and reproducing equipment, and, to a certain extent, have stood in the way of the development of truly accurate audio components. Although the art and science of sound reproduction has progressed to a point Edison could never have imagined, only after shifting engineering priorities back to a study of the real components of real signals, will further substantial improvements be realized.

 

Once, having established that all audio signals can be expressed as a change in amplitude over some period of time, or DeltaA/DeltaT, the function of all sound system equipment can be easily defined. For instance, an amplifier performs simple multiplication resulting in an output signal which can be expressed as G(DeltaA/DeltaT) or GDeltaA/DeltaT, where G is the gain of the amplifier and DeltaA/DeltaT is the input signal. From this representation it can be seen that all changes in amplitude must be magnified by the same factor (i.e. the Gain of the amplifier) and the time base (DeltaT) must remain unchanged, independent of all other considerations. This leads to the discovery of the only two families of real distortions that can and do exist in audio systems.

 

Such amplitude distortion can assume two forms, harmonic and non-harmonic. Harmonic distortion (the most commonly and easily measured anomaly in audio components) is generally caused by non-linearities in the electrical characteristics of the amplification devices. Such distortion is "harmonic", as the number of zero crossings in the error wave form in an integral multiple of the number of zero crossings in the fundamental. Additionally, the value of the distortion signal will always be zero at the zero crossing point of the fundamental. A small amount of this type of distortion is inaudible as it does not drastically alter the shape of the waveform and does not affect the zero crossing point.

 

Non-harmonic amplitude distortions are generally caused by network anomalies. Such phenomena as slew rate limiting, clipping, and transient distortions result in non-harmonic distortion components which not only alter the shape of the signal waveform, but can change the zero crossing point, as these elements may have some real value, when the input signal is at zero.

 

This leads to the second major family of distortion; time base distortion. Time base distortion occurs when the DeltaT term of the signal equation is altered. The zero crossing point displacement described above is a form of time base distortion. Modulation of pulse width, or a change in the delay time between signal "events" also constitute time base distortions. These distortions are the most audible as our auditory system can more easily detect duration and delay than amplitude.

 

 

 

The Audio Note No-Feedback Real Audio Amplifiers.

 

The Audio Note amplifier range does not make use of any kind of feedback. As a result, they were neither designed for vanishingly small harmonic or low intermodulation distortions, but instead for minimal non-harmonic and time base anomalies.

 

The Audio Note amplifiers are all using directly heated power triodes in their output stages, and miniature double triodes in their high-gain and driver stages. Their function were defined before their circuitry was conceived, as constant multiplication of amplitude over a totally non-varying time base, with a view to maintaining power output into a varying load.

 

During the development of these amplifiers using direct heated power triodes, most accepted amplifier design practices had to be ignored, as investigations into their implementation showed circuits with variability of gain with amplitude, time and signal duration, as well as variability of time delay with amplitude, signal duration and signal delay. What has resulted are amplifier circuits which operate optimally and non-varyingly for all signal and load conditions. Where compromises have been necessary between maximally linear amplitude response, and optimum time base performance, the design parameters have always been adjusted to favour the latter. With the superior linearity and load characteristics of the directly heated power triode, whose circuit configurations naturally lend themselves to the defined functions.

 

The design practices most obviously eschewed in the development of the Audio Note Real Audio amplifiers (using direct heated power triodes) is the use of negative or local feedback. Negative feedback, quite simply, is the application of an inverted portion of an amplifier's output signal to its input terminals. This "extra" signal is subtracted from the input and serves to reduce the effective amplifier gain (as the input signal is then smaller). In addition, steady state distortion is thought to be reduced as the out-of-phase distortion components contained in the feedback signal cancels out some of the errors created by the amplifier circuitry.

 

This scheme presents two very obvious problems. Firstly, all amplifiers introduce some delay to passing a signal from its input, to its output and then back to its input. During this delay period, a feedback amplifier is operating at its natural (referred to as "open-loop") gain. It is not until this initial delay period is over, that the circuit begins to exhibit its intended operating ("closed loop") gain characteristics. There must be, by the very definition of a feedback system, some change in the gain factor G, during the transition from open to closed loop operation. This gain modulation would probably not be audible by itself, as the propagation delays of most good amplifiers are quite small, except that the increased gain of the amplifier during the initialization period results in a decreased maximum input capability before overload. Simply put, an amplifier which utilizes 20 dB of feedback (a relatively modest amount by modern standards) and requires an input of two volts to clip during closed loop operation, would overload with only two tenths of a volt input during the forward delay period. Once the amplifier is overdriven, it may take many times its delay period to become fully restored to normal operation. The distortion created by this condition has been commonly referred to as Transient Intermodulation Distortion (TIM), Dynamic Intermodulation Distortion (DIM), and Slew Induced Distortion (SID).

 

In addition to this obvious form of feedback induced distortion, there exists another more subtle effect of signal regeneration. Because all amplifiers have some forward propagation delay, the fed back portion of the output signal will always lag behind the input. There is therefore a constant introduction of "out of date" information into the amplifier. Under transient conditions (which is what music is; transients), this results in the presentation of an error correction signal intended to reduce the distortion of an input signal which has already passed through the amplifier and is either already out of the circuit or well on the way out of the circuit. The signal present at the input by the time the feedback has arrived may bear no relation to the previous signal and thus will not be properly acted upon by the regenerated information. The current input signal is then distorted once, through the subtraction of an erroneous feedback waveform, and again by the amplifier. Additionally, the error signal present in feedback is passed through the amplifier and again fed back, with all of the newly created distortions, to make yet another trip through the circuit, until it is allowed to decay through successive attenuation. Thus, a distortion signal which originally may have lasted only a few microseconds, can pass through the amplifier enough times for its effective duration to have exceeded the threshold of human audibility. The mechanism originally designed to reduce audible distortion, actually, under transient conditions, serves to regenerate, emphasize and, in fact, create distortion.

 

Because our Real Audio triode amplifiers operate totally without signal feedback, such distortion regeneration does not take place. The circuits have been designed for maximum linearity without corrective mechanisms, and thus responds as easily to transient signals as it does to steady state waveforms. The amplifiers make no attempt to reverse the path of time in order to correct their own errors. Those distortions created by these circuits (which are almost entirely harmonic in nature) are allowed to pass only onto the loudspeaker, and not back to the input.

 

Despite the absence of feedback, the forward propagation delay of all our amplifiers has received much attention. All our output transformers have been designed using this criterion, obviously with a keen eye on cost. It is obvious that if this delay is not absolutely invariant, for all conditions, the DeltaT component of the input signal will not be accurately preserved. Thus, those factors which determine delay have been carefully observed and stabilized. In addition, the operation of all amplification stages at nearly constant power, independent of signal conditions, i.e. Class A operation at every stage, greatly contributes to the symmetry and linearity of our circuits.

 

It is, however, not enough for an amplifier to operate linearly by itself. In order to minimize audible distortions, the device must be able to operate as well into a real loudspeaker as it does into a laboratory resistive load.

 

In order to adequately control the cone excursions of the loudspeaker and to optimize power transfer, the effective output impedance of the amplifier should be as far below the impedance of the load as possible. The ratio of these two impedances is referred to a damping factor, usually referenced to an eight ohm speaker. Thus, a damping factor of eighty reflects an amplifier output impedance of one tenth of one ohm. The design of the output transformer is extremely critical, and taps on the output are normally provided to match the load impedance best possible.

 

A problem in the normal expression of damping factor is that its measurement is performed using steady state signals. This results in a factor relying quite heavily on the action of an amplifier's feedback. The damping ability of an amplifier under transient conditions, before the feedback mechanism has been able to reach, is only accurately expressed as the steady state damping factor divided by the feedback factor. Thus, an amplifier with twenty decibels of feedback and specified damping factor of one hundred, has a damping value of only ten under transient conditions. This not only reduces the amplifier's ability to control the cone movement, but allows voltages created in the speaker voice coil to mix with the output signal and enter the amplifier's feedback system. In this condition, distortions created by the speaker's motion are not only unattenuated, but are emphasized through feedback regeneration.

 

Audio Note Real Audio, no-feedback amplifier's damping ability remains constant at all signal conditions.

 

An investigation is underway to fully explain the relationship between damping factor, real world power output and amplifier feedback. This will take some time. In the meantime, the only amplifiers available on the market to fully fulfill the criteria set, amplitude against time, with as little change as possible, are the non-feedback amplifiers from Audio Note.

 

Peter Qvortrup from http://www.audionote.co.uk/ "

[hpavictor]

Rog userii care dispun de cunostinte solide de limba engleza , de timp liber suficient si foarte multa bunavointa sa ne traduca aceste articole din limba engleza in romana , pentru a fi la indemana tuturor userilor .Stiu foarte bine ca regulamentul Elforum are o prevedere expresa care obliga userii sa scrie doar in limba romana , dar aceste articole nu merita sa fie traduse simplist , cu Google Translate , datorita faptului ca pot apare erori . Sper ca moderatorii acestei sectiuni sa inteleaga si sa accepte argumentele mele icon_jook !

[franzm]

The Audio Note amplifier range does not make use of any kind of feedback...Because our Real Audio triode amplifiers operate totally without signal feedback

Si reactia interna a triodei? A eliminat-o si pe asta (cascoda? scheme?) sau doar o ascunde de publicul mai putin stiutor?

[hpavictor]

The Audio Note amplifier range does not make use of any kind of feedback...

Because our Real Audio triode amplifiers operate totally without signal feedback

Si reactia interna a triodei? A eliminat-o si pe asta (cascoda? scheme?) sau doar o ascunde de publicul mai putin stiutor?

Textul evidentiat cu rosu s-ar traduce astfel : Amplificatoare Audio cu triode adevarate ........

Triode adevarate adica triodele pur sange , nu pentode sau tetrode legate ca triode etc .

Audio Note se lauda ca aceste amplificatoare nu au reactie negativa globala ...asa interpretez eu textul de mai sus .

Reactia locala cu siguranta nu poate fi eliminata din schema , in special reactia interna din orice lampa .

Daca am gresit , va rog sa ma corectati !

[Vizitator]

Datorita capacitatilor interne mici ,reactia negativa se manifesta doar la frecvente mari dar depinde si de factorul de amplificare a triodei (efect Miller)

[Vizitator prutus]

Ca veni vorba de Lynn Olson, ceea ce voi scrie aici este nu despre masurarea distorsiunilor ci cate ceva despre minimizarea lor in domeniul vintage prin urmare voi fi oarecum off-topic...I auditioned many SE-DHT amplifiers after that, but in all honesty, none came up to the mark set by the Ongaku, although some came pretty close. I met the designer of the Ongaku a couple of years later at the CES, and Kondo confirmed my impression that the circuit of the Ongaku WASN'T ANYTHING REMARCABLE. It was the implementation - the all-silver signal path, especially the hand-made silver coupling cap and the all-silver output transformer, that gave the Ongaku its distinctive clarity and insight. Kondo-san said that building one on the cheap would just result in a quite ordinary SE amplifier - the Ongaku could be thought of as the ultimate parts-tweaker amplifier, a design that would sound completely different if all parts weren't exactly as specified.Asta e un mesaj pentru cei care au ochi sa vada.Candva am spus ca avansul tehnologic si atentia la detalii pot rezolva uneori mai mult decat niste topologii indraznete si foarte ingenioase.Bineinteles ca si aici exista nuante...Transformatoarele AudioNote actuale nu au doar sarma de argint ci folosesc si combinatii de tole diferite(tole destul de speciale si ele) intr-un singur transformator plus intrefier metalic , in genul pastei care umplea intrefierul din dubu-c-urile rusesti , o tehnica intalnita si in constructia capetelor magnetice de hard-disk-uri(MIG sau metal in gap) si toate acestea par a duce la un transformator cu intrefier variabil functie de intensitatea si frecventa semnalului ceea ce ar fi cu adevarat marea inovatie tehnologica din domeniul transformatoarelor audio dupa AudioNote.Atfel transformatoare cu doua primare si un secundar nu ar fi posibile in high-end.Cum eu sunt un fan declarat Kenwood din care se trage practic si AudioNote voi fi intotdeauna si un fan al distorsiunilor infinitezimale de orice fe ar fi ele .Pe site-ul AudioNote veti gasi date oficiale foarte clare despre majoritatea tehnicilor folosite in constructia transformatoarelor de iesire.Totusi startul adevarat in aceste studii l-au avut firmele constructoare de magnetofoane si deck-uri din care Studer impreuna cu VAC, IBM , Sony(Aiwa) si Nakamichi au fost cele mai proeminente.Am inteles ca si rusii foloseau tehnici asemanatoare , dar cum eu am mai putine informatii las pe altii sa se exprime.