bipolar (BJT) transistor families for audio power output stages

10% = 0.1 = -20dB,
1% = 0.01 = -40dB,
0.1% = 0.001 = -60dB,
0.01% = 0.0001 = -80dB,
0.001% = 0.00001 = -100dB,
0.0001% = 0.000001 = -120dB.

dB value = 20 * Log [voltage ratio]
If the voltage ratio >1 the dB value is +ve.
If the voltage ratio is <1 the dB value is -ve.
Most calculators get this right.

1% = 0.01 = -40 db...in voltage ratio..you did mention a
power of 1 W...let s take it as an exemple..

1 W rms / 8 OHM load is 2.83 V rms...
distorsion at -40 db makes 28.3 mV...
(0.0283^2 )/8 = 0.0001 W = 0.01% of 1 W
 
since you re using a power as reference to be
manipulated :
db value = 20log(votage ratio)^2 = 2 X 20log(voltage ratio)
thus a voltage ratio of 100 is a power ratio (literaly) of 10 000
and so on..
if distorsion is at -60 db in voltage ratio, then, in power ratio
it will be at -120dbW
 
For me it is amazing, that the currently types of old motorola MJ series (e.g. MJ21193/MJ21194 without DC-SOA in data sheet) provides perfect sonic transmission by only 4 MHz ft. What is in real life the advantage for 10 times higher values of ft for audio power applications (possibly depending on whether the emitter follower or common emitter is used)?

And so, returning to the first post, the question was the value of a high Ft. If I understand at least a little of the past few pages then the benefit of a larger Ft is actually for use with NFB. The higher bandwidth, if supported by the preceding amplifier stages, allows the amplifier to retain good gain into much higher frequencies because now the output devices have more 'headroom' in terms of their speed so that even when they are operating outside of their ideal range in terms of current/voltage (such as Class B perhaps) they don't become too slow. This allows the NFB to reduce distortion at higher frequencies.

But I am now wondering whether amplifiers are being designed with adequate NFB factors and sufficient error amplifier accuracy at high frequencies to take advantage of this.

By some coincidence, I have a design on the drawing board with excessive OLG so I may do some experiments in this direction. It's essentially a JLH amplifier topology, so it has the benefit of a Singleton (as Anatech calls them) error amplifier, although doing the experiment properly means making a variant with low OLG and low NFB. I'ts perhaps a rather old experiment that many of you have already done, but I guess each of us has to hear it for ourselves at some time. If all goes well, it will be built sometime over the winter, when Class A brings fringe benefits to homes in the Great White North.
 
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Erm, except for the high order products that it introduces...

Often the concept of "introducing new distortion products" makes no sense though. Mathematically, if we have a static nonlinearity and can express the input-output relationship as a polynomial, we can by inspection observe the highest-order term of the polynomial and conclude that's the maximum order of the harmonic distortion product. Consider the tanh() function that shows up in BJT diff amp analysis. It has a power series representation, the Taylor series, but that thing has an infinite number of terms. So it generates an infinite number of harmonics. Even though the ones of very high order may be extremely difficult to measure, they are there nonetheless. Therefore the concept of "introducing new distortion products" makes no sense in this case, because a simple diff amp already generates an infinite number of odd harmonics. I haven't tried this, but I doubt that feedback would introduce even harmonics to the mix.

This is by no means an isolated example either. Ever do a DC sweep of an optimally biased class AB output stage in simulation and try to fit the resulting data to a polynomial using least-squares techniques? I tried and gave up at order 9. The results weren't even close. I don't have proof of this, but I've concluded that it's going to take a huge number of polynomial terms, possible infinite, to fit the curve properly. This is due to having small, fine-grained wiggles near the zero crossing (but not discontinuities), in effect straightening out to a gradual bend as the output voltage and current get larger. You can simulate the effects of feedback on this circuit as Bob originally did, producing a Baxandall-like plot of relative harmonic levels vs. feedback. The results of that analysis, found in the Cordell feedback thread, are counterintuitive. Even small amounts of feedback reduce all harmonics, and larger amounts of feedback reduce them more, in a monotonic fashion.

The conclusion is that the effect of feedback on individual distortion components is rather subtle to grasp, highly nuanced and sometimes counterintuitive, depending a great deal on the nature of the open-loop nonlinearity. I'm reminded of a saying by H.L. Mencken. "For every complex problem, there is a solution that is simple, neat, and wrong." Just replace the word "solution" with "explanation" and you describe much of what's been said here about feedback and distortion.
 
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And so, returning to the first post, the question was the value of a high Ft. If I understand at least a little of the past few pages then the benefit of a larger Ft is actually for use with NFB. The higher bandwidth, if supported by the preceding amplifier stages, allows the amplifier to retain good gain into much higher frequencies because now the output devices have more 'headroom' in terms of their speed so that even when they are operating outside of their ideal range in terms of current/voltage (such as Class B perhaps) they don't become too slow. This allows the NFB to reduce distortion at higher frequencies.

This is in accordance with partly of my observations, described here about post # 292.

Some statements do change off-topic in the meantime - regarded the NFB actually this is the right thread:
http://www.diyaudio.com/forums/soli...-nfb-negative-feedback-me-wrong-question.html
on the other hand, the clear assignment is problematic because of the numerous overlaps
 
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Piercarlo,
He's incorrect in attributing this problem to the feedback itself instead to the BAD USE of feedback. Timing problems are out of worries from at least a 40 years, as planar devices become largely available on market.
Lumba Ogir, as many others, don't know - or forget to know - that if most higher frequency we can ear is about 20 kHz, then we can't ear time slices shorter than of the corresponding period. Actually appear that real abilities in detecting time slices of human ears are at most limited to the frequency band where its sensitivity peaks - 2-3 kHz, roughly equivalent to a 0.5 ms... when typical throughtput timing of transiting signal in not exceptional audio electronics are AT LEAST 500 times shorter! Time is not anymore a problem in audio electronic, either analogical or digital (the so called "jitter problem", in the form as often publicized in audio reviews, is a not problem at all...).
All this is rubbish.
 
Time is not anymore a problem in audio electronic, either analogical or digital (the so called "jitter problem", in the form as often publicized in audio reviews, is a not problem at all...).

This is indeed rubbish. The time domain is the most understudied one in all audio analysis. Thanks to Fourier.

And jitter is one of the biggest issues of digital reproduction. Have you ever asked yourself why Reel tapes sound better? No A/D and D/A conversion = most of the damages to the signal are avoided.
 
andy_c , Taderbam and other unbelievers,

It should make sense to you. Global feedback is the exclusive cause of all dynamic distortions, transients by nature with nonharmonic content like switching distortion, much worse than harmonic distortion.

Dear Troll,

You definitely need to go back to school - any basic EE Circuits and Systems course will tell you that any time invariant non-linear system is unable to generate non-harmonic components.

Otherwise, keep feeding here with your junk. As other mentioned, you are an invaluable deposit of confusions, misunderstandings, urban legends, with a certainly entertaining value.
 
In terms of entertainment value, check this out. It's a novel use of output devices if you ask me.

An interesting possibility opens up if the output devices driving the speaker have a different Ft than those driving the feedback loop.

The ALTMANN SPLIF Amplifier Topology

A variant of the classic "feedback loop eating it's own tail" urban legend.
 
andy_c , Taderbam and other unbelievers,

It should make sense to you. Global feedback is the exclusive cause of all dynamic distortions, transients by nature with nonharmonic content like switching distortion, much worse than harmonic distortion. Have you ever heard about Otala?

Lumba we sorted out the "other" frequencies just go back and read the thread. Someone should post an update of Otala's graph with all of them labeled. Please note, Otala himself did not instigate the absurd interpretation of his data.
 
Piercarlo,
Second armonic above 7 kHz is just the only DIRECT armonic you can hear (pheraps you can hear a well boosted third if you have really - but REALLY, in physician terms, not audiophile terms! - good ears... not more). Other armonics, if not processed by intermodulating non linearities that downmix them in audible band, are for human beings simply inexisting.
The TRUE reason for which SOME instrumentally bad amplifier (not ALL as usually many guru claim for) soun "better" of some other really better amplifiers i've yet explain in the previous reply and, adversely the beliefs of many actracted by alleged "audio misteries", there is nothing of misterious: is simply psichoacustics.
All this is rubbish.
 
Telstar,
I think I disagree on the bolded part above. If the distortions are pushed low enough they do merge into the noisefloor. Why shouldnt they? Point is that the noise floor must be low for a start, damn low.
This is not quite clear to me. It is not about noise floor, it is very importantly about harmonic spectrum.
Hmm this sounds weird
That's the reality we are facing. The the ear`s distortion is not a disadvantage in sound reproduction.
Once more (sigh)...
I dont hear the ultrasonic frequences by themselves, but the armonics of lower frequencies and all this content is elaborated by the brain. Feel free to not believe it (i know most people dont).
Until some more studies will be done on this subject, i think it is a moot point.
No doubt. Several studies have been done showing that. The high frequency range must be in good shape for good performance in the audible range. Wide bandwidth is one condition of low high frequency distortion.
 
Trederbam,
It is a mathematical fact that feedback multiplies and spreads harmonics in a non-linear system.
Not sure this is always "safely outside the measurable range" though.
In that case, the measured distortion is not lowered. The transformed energy (if you like it that way) is by definition distortion, but we better don`t discuss its audible impact (could trigger dismay).