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About fas42

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  1. Well, that's the first time I've seen that Self calls it class B - the usual understanding, that most go by, is using AB to describe it ... it makes things messy, if even the experts can't agree on the 'right' term to use ... 😁.
  2. Note this post just made, Turns out that systems at every cost level benefit, and, those at the bottom rungs get the most benefit! Why? Because those components usually have the lowest level of engineering applied internally to give good isolation, and are compromised badly by poor quality mains, etc. Which is why doing extensive tweaking externally to shield the gear from the environment can often have the greatest audible impact for low cost items - if the internal circuitry is "good enough", then they can shine in terms of getting the subjective qualities of the sound in good order.
  3. In part. Pass is correct in saying that FB won't automatically solve all the problems, and the classic output stage operating in class AB is a perfect example of this. The theory of feedback says that the circuit will be able to compensate for any discrepancies in the perfect behaviour of the signal through its path, by "counter-distorting" at the input of the circuit - an error signal is added to the true input; the amplifier is in effect always amplifying a distorted version of the waveform, which is neatly reversed as it passes through the signal path - a 'pure' version of what you want is then what the speaker sees. The fly in the ointment, in most implementations, is that the signal drive to the output stage, that which is pre-distorted, remember, now has very fast slew rate parts at the crossover region, intended to "undo" the crossover artifact - but anyone who has tried to get a power output stage to have a very high effective bandwidth knows that this is far from easy, the characteristics of high power transistors in every way conspires against being "fast"- it can be a losing battle. Very clever, and very fast designs can get around this - think the Spectral brand here. Higher level, but pleasant distortion can easily be superior to listen to, than quite low level, but irritating, and disturbing artifacts - this is why, say, class D can easily do a better job of getting the subjective SQ right, if the implementation is good enough - no crossover headaches. Well done class A, extremely fast class AB, and good quality class D are all fine solutions - the last is where the future is, for obvious reasons.
  4. Depends on who he is designing for. His DIY stuff is known for it, and this thread, as an example, https://www.audiosciencereview.com/forum/index.php?threads/nelson-pass-amps.8034/, describes his tilt on the game, from the POV of various observers - his aim is to "make it sound good", above all else. Yes, class AB has a major weakness in its concept - it is relatively difficult to completely negate the impact of crossover distortion; I have played with Spice simulations which show that even in ideal circuit setups that the glitch in the waveform is very hard to make completely vanish. Some very sophisticated designs can make it happen, but you then pay for that in that the circuit has a very high parts payload. Explored this area over many years - it started because the original Perreaux had duff smoothing caps for the job, even though they looked like they could jump start a vehicle, 😉. Symptoms were loss of high treble quality when the PS was working hard; quite easy to hear - improved this area over several iterations, by throwing away those caps and introducing a design which had extremely low ESR - treble problem disappeared. Spice does the maths for me, there's a thread over on diyAudio, that ran years ago where I was very active, that I've pointed to a couple of times - it's quite easy to show that real world supply circuits have trouble with voltage sag, because the parts are not capable of delivering the energy needed when the speakers are being driven hard. I do what's necessary to get a particular setup to reach an acceptable standard - the original Yamaha, Perreaux, etc setup had sufficient raw quality, and was exceedingly simple; enough was in place so that it reached the convincing SQ level without that much needing to be done. Lower cost, and quality items have more shortcomings; it was a challenge to see if I could also push them to that point as well - and good learning. Key is ability to be able to hear what a system is getting wrong, and make good judgement calls of what should be worked on - every setup is going to have a unique set of weaknesses; no point in wasting time 'fixing' stuff that doesn't need fixin'. They have got close - but no cigar. Problem is in the electronics; still too much low level distortion for my mind to be able to ignore. Don't worry 'bout it then ... 😉. And, some news!! A new toy is on its way ... the Behringer active monitors I got some years ago turned out to need too much fiddling, tweaking to be a good candidate for a value for money, tarting up exercise - and, just came across something, literally today, that is a more visually acceptable, and capable active speaker, meaning digital in, 🙂 ... cross my fingers that this can deliver the goods, at a low effort cost. Also, starting to play with some circuit simulation of Putzeys's recent class D amp modules - will be interesting to get a handle on what makes them tick ...
  5. I would agree that if one has separate subwoofers that smart DSP correction is invaluable for ensuring that the two spatially separated speakers systems properly integrate, as least for some areas of the room.
  6. Agree with what you're doing here - damping, and changing the nature of the vibration by loading with very high mass are highly effective tools in audio.
  7. Yes, room correction has never interested me - I find that my brain does an excellent job of compensating all by itself, if the SQ is good enough ... 🙂. Demonstrations of what happens, say with the DEQX unit, leave me quite nonplussed. I have come across one clip, apparently of the M33 in operation - the positives were that detail was well expressed, nominally nothing was missing; but what I could also hear was the classic problem of a type of edginess being in the sound, where bursts of strong treble were troubling - this is a key area that I deal with in my optimising; getting rid of that particular distortion artifact. Now, this might be very unfair on the NAD unit; the circumstances of the recording might have caused this - so, wait till more comes in, to get a better handle on its behaviour, as you say.
  8. Noting discussion on fuses, I worry about fuses, but have never tried "audiophile" variants - the trouble with a fuse and the fact that it normally is just slipped into a fuse holder, where it relies on fairly rough and ready metal to metal contact to ensure contact integrity, is that it has non-linear behaviour when you look at it over all possible conditions, and time periods. Yes, shouldn't make a difference, but the power supplies are in many ways marginal in their performance, especially in power amplifiers - so, enough non-ideal electrical behaviour gets through to make their presence audible under some circumstances. A solution that works well enough for me is to ensure high quality contact between the fuse and its holder - either solder, or silver contact enhancers have been adequate.
  9. Just noting this new unit, from NAD, https://nadelectronics.com/nad-masters-m33-bluos-streaming-dac-amplifier-becomes-first-integrated-component-to-feature-purifis-ultra-quiet-amplification-technology/ - would be close to ideal as a integrated solution. Technically, this would tick the boxes for all the measurable parameters being as good as it gets; so any shortfalls in subjective performance are then due to weaknesses in the implementation, the integration of the various parts - it makes zero sense to buy anything more expensive than this ... need to see what the subjective reviews have to say, which will indicate how well NAD have 'debugged' the engineering of the combining of the various elements in one box ... highly likely to be an excellent base for optimising to achieve convincing SQ - if it can't do this in 'raw' form.
  10. Just noting Chris's review, . This is a good sign ... the manufacturers are indeed learning, and it will steadily ripple out, and filter down to lower cost units - what it means is that far less tweaking will need to be done to extract decent SQ from a rig; because the components are intrinsically engineered to a much higher standard. At the moment we are in a transition era, where absurdly costly items are showing the way, and far more price effective units are in the process of incorporating the lessons learned from breaking down what it actually is in the expensive components that makes the better sound happen - bling is not a prerequisite for a special listening experience. Note the clear improvement in what is now going on - the older, less capable gear blurs, homogenises the fine detail; the improved variant brings forth everything that is actually on the recording, but not in a manner which is disturbing - the 'humanity' of what was captured is now far more apparent, and makes the listening so much more satisfying.
  11. Multiple, separate in time passes through the same playback chain will yield slightly different waveforms, each time - people using, say DeltaWave, can see this quite clearly ... so, it's trivially easy to show the 'imperfections'; the real question is whether it's audible. Yes, amplifiers plus speakers cause issues - but the analogue side of the DAC portion of the chain also contributes, IME. The real debate is whether the variance is below audibility. I have zero problems believing it is audible, because my first good rig, decades ago, had a 100% repeatable characteristic that its SQ could either be holographic and convincing, or merely a conventional stereo presentation. The only thing that varied was the time since the last resetting of the electronics. The rationale is that every rig - even mine, 😜 - are not "perfect" ... what one hears when they sound different are the precise qualities of the distortion artifacts; the brain can quite easily focus on the one or two things that in a consistent pattern mark one version of the sound of a particular recording from replay on another system, or a variation of the same system - this is often talked about in a positive way, people say that the rig had a "warm" signature, or a "clinical" signature, or whatever. The cables/regenerators/filters have, or should have, maximum impact on the signature of the distortion - hence, make the SQ vary ... the ideal is to attenuate the signature of the system to zero; which then will mean that any extra fiddling with these type of tweaks will have zero audible impact.
  12. This may or may not help in understanding what I'm after .. .posted in the blog about 4 years ago, about a first attempt to record using a brand new, cheap USB mic, And this is the latter track, "Everybody Rock", from the source, very recently uploaded - starts at 2:30 in the above capture of playback, Now obviously huge differences, volume, acoustic because the sound is being captured from that bouncing from the wall in front of the speakers, and thence reaching the mic, behind the speakers ...but I feel the playback has presented the essence of that piece, that which makes it worth listening to.
  13. Yes, that is the term that I normally use.
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