CHANGSTAR: Audiophile Headphone Reviews and Early 90s Style BBS
Main Deck => Discussion for Registered Members Only => Topic started by: Marvey on September 23, 2014, 10:41:45 PM
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Not too many speculative threads here, but I wanted to throw out something for consideration. Keeping this in the members section since "nobody" will believe what I suspect anyways. If Jason or Mike want to contribute at times, that would be great, but I don't expect them to reveal any secrets or make any bold claims - just doesn't seem to be their style.
As many of you already know, the few of us who have heard the Rag consider it an exceptional amp. So this got me thinking why? From a topology point of view, I bet it's not much different from the Mjolnir, yet the Rag stomps all over it, eats its, and chews it out. In fact, the Rag does that with virtually every other SS amp I've heard so far.
The obvious major difference is the microprocessor controlled bias, offsets, operating factors, etc. Could this be the difference? Is there something Jason knows that we don't know. I mean, if we think about it, why would anybody do anything so crazy and unnecessarily complex? I remember asking Jason... why? You guys are nuts! Unless of course there's a huge sonic dividend - past the marketing. (At least Schiit doesn't imply they invented 7000 series aluminum like Apple.)
So this gets me thinking about the closed form digital filter for the upcoming Yggy? So again, I'm thinking... why? That's nuts. Why make your lives so difficult? Why the crazy emphasis on accuracy? (The "accuracy" aspect from the guys at Schiit is subtle, but its very much there if you've been reading carefully.) And then we when go back in time, with Theta, and their DSP filters, reading old Stereophiles, product information, marketing materials, interviews, you realize that Mike Moffat has a thing for accuracy.
So now I'm wondering... Maybe there's something to it other than just fancy technology that can be marketed. That these guys might know something. Like maybe more accuracy = better sound quality.
I'll leave this up for you guys to discuss. Just some thoughts. I did own a Theta DAC back in the day.
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TBH I have no idea.
For example, not sure what it's meant by closed-from digital filter... Best guess is that they mean they don't approximate some types of filters with a finite number of IIR or FIR coefficients and do some other strange stuff? Also dunno how certain signal integrity aspects are being optimized for accuracy. Or even, taking a step back, what it's meant by accuracy in their context.
I know Mike M thinks poorly of certain delta sigma implementations and don't blame him at all for that. But I also think (incorrectly perhaps) that he thinks highly of ladder DACs (or prefers them), and from what I know, they aren't perfect either. I personally prefer something like a combination: oversampling multi-bit... which I think is what AKM (and Schiit) sometimes use.
Based on their product offerings, it seems class-A amplification is favored, discrete solutions are favored, hybrid topologies are well regarded, JFET output stages seem present in some of their products, so are AKM converters, and circlotron topologies are at the core of some of their flagships.
And that's about the extent of my piss poor knowledge.
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So this gets me thinking about the closed form digital filter for the upcoming Yggy? So again, I'm thinking... why? That's nuts. Why make your lives so difficult? Why the crazy emphasis on accuracy? (The "accuracy" aspect from the guys at Schiit is subtle, but its very much there if you've been reading carefully.) And then we when go back in time, with Theta, and their DSP filters, reading old Stereophiles, product information, marketing materials, interviews, you realize that Mike Moffat has a thing for accuracy.
So now I'm wondering... Maybe there's something to it other than just fancy technology that can be marketed. That these guys might know something. Like maybe more accuracy = better sound quality.
I'll leave this up for you guys to discuss. Just some thoughts. I did own a Theta DAC back in the day.
For me, what Schiit is doing with the Ygg is much more impressive than the Rag. The Rag may very well be super king big nuts of SS headphone amps, but I know from first hand listening that there are a lot of ways, A LOT, to make a great sounding SS amp, from Valvet's ultra simple floating bias pure Class A design with a single bipolar output transistor to Accuphase's awesome Class A amps to the absolutely incredible stuff from Vitus and BALabo. All great, all very different design ideas.
What I'm saying is, the Rag destroying its competition in the headphone arena I think says more about the state of the competition than it does about the Rag.
There isn't the same separation when it comes to DACs. A DAC is a DAC, you don't have headphone DACs and speaker DACs. It would've been the easiest thing in the world for Schiit to slap some TI chips on a board and call it a day, or to make yet another Sabre DAC. Just about everybody else is doing it at anywhere near the Ygg's price level. You even have Charlie Hansen from Ayre calling out DSD as a bunch of dumb BS, and then making the QB9 DSD with a Sabre DAC because that's what the market wants.
The Ygg seems to me to be about as far in the other direction from that as you can go, and it's definitely the hard road to take. Considering how ambitious of a product it is, it should probably be priced like an Empirical Overdrive. The fact that it's going to be about a third of that is crazy.
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Wait Marv-I thought the Rag was just a Moar Powerful MJ With Speaker Taps!? ::)
I'm also infinitely excited for the Yggy-maybe moreso than the Levi honestly. Seems like Schiit is about to change the game w/ both of these statement pieces. Good for everybody.
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I'll take a shot at responding on the amp side. I suspect Mike will weigh in on the DAC side, sooner or later...maybe later, since I know he and Dave are doing some tweaking on the Yggdrasil tonight, and are in full "make sure it's fully ready for the show mode" right now.
First, a disambiguation of Ragnarok and Mjolnir topologies. They're actually quite different, though both are circlotron-output amps. Simply scaling up a Mjolnir wouldn't work for Ragnarok, mainly because the output stage has to have a low enough impedance to control complex speaker loads. The key differences are:
Mjolnir: ALPS 4-gang volume pot, single JFET gain stage, JFET-buffered and capacitively coupled to the MOSFET circlotron output, with MOSFET bias via a differential DC servo.
Ragnarok: Relay-switched resistor ladder attenuator, JFET input differential stage, plus BJT VAS and driver stage DC-coupled to the MOSFET circlotron output stage, with bias set by microprocessor. The MOSFET outputs in this case are MUCH bigger than Mjolnir, and require drivers. There is feedback that combobulates the VAS and output stage into their own "output local" loop. There's no overall feedback, and feedback switching at different levels uses differential localized feedback, rather than changing the output local loop.
So why does this sound good? Well, let me get back to that. Because we're entering into the "IMO" realm here, and lot of people won't agree with what I'm going to say right now, as their belief system lies along the "well, if it's got PPM (parts per million, or 0.000X% levels) distortion at 20kHz, it's the most accurate it can possibly be."
And designing an amp purely for super-low distortion is no easy task—but, surprisingly enough, it's not due to the topology, it's due to the actual physical layout of the boards. Topology-wise, designing a PPM-level amp is well-known. It just takes a bunch of gain and a lot of feedback, with attention paid to minimizing commonly known distortion mechanisms in the open-loop design. This will mean lots of parts. However, once things get on the board, a trace run the wrong way can increase distortion by orders of magnitude. In addition, most of the distortion will come from the output stage, and how it interfaces with the speaker load. This can cause all sorts of unpredictable behavior with complex loads (which are not typically measured.) So, a couple of board errors, and a reactive load...and you gotta wonder if designs that deliver low THD in simulation are really the be-all-end-all.
We don't design for PPM distortion, though low overall distortion is one criterion. So why does Ragnarok have a good chance of sounding good, in our opinion?
I think it comes down to really one thing: avoiding overall loops, of any kind, that feed stuff that isn't the signal back to the front end of the amp.
Huh?
Well, traditional feedback takes the output of the amp (at the speaker taps) and feeds it back all the way to the front end of the amp. This scaled signal sets the gain of the amp...and it also brings back any distortion generated by the interaction of the output stage and the speaker reactive load to correct the distortion of the overall system. The more feedback, the lower the distortion.
Also, most DC servos do the same thing--sample the output, bring the DC correction signal back to the input. Of course, this also means you have a phase-shifted, frequency-dependent part of the output coming back as well, because a DC servo doesn't just amplify DC.
But--and here's where I'm way out in la-la land, where the "THD is all" crowd will crucify me--you're also feeding back stuff that was never in the input signal in the first place...nonlinear distortion from the output stage, plus frequency-limited and clipped garbage from the servo. And this stuff goes through the full gain stage of the amp.
The best amps I've heard (and designed) in the past did two things:
1. Kept the output stage as its own loop (via Hawksford error correction) with no feedback back to the gain stage. (Hawksford is a neat way to get low output impedance and very linear output stages, without having to do overall loop feedback.) So, in effect, it "decoupled" the output from what the rest of the amp was doing.
2. Paid careful attention to making sure the DC servo was as close to DC as possible. This was done, in the past, with heroic multi-pole filtering, and careful choice of time constants. But still, if you amplified the output of a DC servo, you could hear the original signal, even after this kind of filtering, so it wasn't perfect.
So, Ragnarok is an extension of this. We aren't doing Hawksford error correction, but instead relying on a close-coupled local output feedback loop between the VAS and the output stage. Similar results overall (but slightly higher distortion). None of this is taken back to the input of the amp. In addition, we simply threw out the DC servo, because microprocessor-controlled bias means it isn't necessary (at least after we got the Ragnaroks to stop blowing up.)
Are there other topologies that could express this exact design goal (output isolation, limited feedback, no servo)? Sure. A neat topology for a single-ended amp would be a very linear front-end stage with local feedback from the drivers, coupled with a Hawksford-linearized output stage with bias set by microprocessor and a voltage controlled voltage source. It would work for a balanced amp too, but a SE amp in this configuration would be very affordable to make.
(And, the usual disclaimer--this is blue-sky stuff, not necessarily hints at future products.)
Annnndddd...who knows, I could be completely full of it, or barking up the wrong tree. It could be possible that my conclusions are entirely wrong (look at Vali, capacitively-coupled and relatively high THD...and it sounds good.) However, the output stage does not have any overall feedback to the front end...and nor is there a DC servo...
Hope this helps a bit!
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A neat topology for a single-ended amp would be a very linear front-end stage with local feedback from the drivers, coupled with a Hawksford-linearized output stage with bias set by microprocessor and a voltage controlled voltage source.
That's a funny coincidence; I'm building something like this in my garage. Sans microprocessor.
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For example, not sure what it's meant by closed-from digital filter... Best guess is that they mean they don't approximate some types of filters with a finite number of IIR or FIR coefficients and do some other strange stuff?
I could be full of it or completely wrong... Maybe it works by hanging on to the original samples where an initial output from the digital filter is compared to originals. Error is calculated. Output function is tweaked to minimize or effectively zero out error. Then that output is used. Sort of like closed loop operation on an engine - where the computer measures emissions and tries to maintain a certain AF ratio for fuel economy.
Lots of variations of how this can be done.
With the talk of quantization (rounding) error in the chatbox, I can understand why the DSD thing is an anathema to the Schiit guys. The DSD noise shaping thing to push the 1-bit crap up to higher frequencies. Of course "noise shaping " is another word for adding adding RNG noise to the original signal. Dynamic range is expanded in the audible region, but total error is going to increase.
Sort of goes against Mike Moffat always talking about the accuracy requirements of cruise missiles and medical equipment.
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I think it's important to mention that we need to differentiate accuracy and precision. Electronic components (D/A, A/D, opamps, etc.) often advertise themselves as "precision" devices. And they are in the sense that the results they produce for a given input are tightly bound together. You get consistent outputs, but not necessarily accurate outputs.
So I'm going to stick my neck out there and say that modern delta-sigma DACs are indeed accurate for reproducing sine waves. Or most any simple periodic function. It's not hard to verify this empirically.
The reason why these ladder DACs are used in industry is not because they're necessarily more accurate for AC signals, but rather they are designed to be accurate against a certain reference (DC precision). If you want to set the levels of your MRI scanner, you want to use a ladder DAC with the correct feature set. Generally meaning a reliable part that is stable with temperature, long-term use, low noise, low error, etc.
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When dealing with DACs, the stored sample is already digitized (amplitude quantization) and sampled (time quantization).
Suppose we have the luxury of infinite precision. Then we would still be missing the data in between samples due to sampling, and would have to do our best effort in guessing what was going on there. The relevance of what Nyquist said about this problem was that there is a unique solution for all the missing data points between samples if the signal minimum period is constrained to be at least two times more than the time distance between samples. It's a pretty powerful theorem, but it does have it's real world implementation problems (like pretty much everything).
As far as amplitude quantization, AFAIK not much we can do about that if things were already digitized to 16 bits. Usually there is no original signal to compare with. One could encode an error (or residual) signal channel, and do some sort of interesting stuff like it seems DTS-HD does. But if music was encoded 16-bits and that's all there is left, then don't know how one can beat up 24-bits of resolution out of it. Damage is done.
A hi-res file could be done at 96 kHz and with 24 bits of precision and then it may be up to the equipment to handle the file gracefully. Still, at some point going for 125412588 bits will likely result in doing an awesome job at capturing every single background and thermal noise nuisance of the recording equipment. I guess one could check where noise floor usually lies on recording equipment and derive requirements from there.
The job of most digital and analog filters in the DAC signal path (AFAIK) is to do as best of a job as possible in guessing missing signal between samples (time quantization). Not much it can do about number of bits except to not further reduce precision. It's not much different from what an analog filter needs to do, except a digital filter doesn't have to deal with component tolerances which can render a high order analog filter unstable or weird, and certain component non-linear behavior. Dealing with loss of precision is not an easy task though. Specially when dealing with very long digital filters. The good news is that simulation in the digital world yields sort of predictable results. Analog ICs (and board layout) are much more unpredictable.
As far as boards and amps I get a lot of what Jason is saying (not an expert in those areas though). I appreciate the fact that Rggy has to deal with less than 8 ohm loads and things will have to be different. I also get the feedback discussion, and optimal setting of operating points. There is a lot I still don't understand about interaction between amp and non-linear loads though. While I do get how feedback can correct for transistor turn on stuff due to class B or A/B operation, I don't see how it can correct for non-linear load issues.
As far as 1 bit stuff, I can say that GPS is QPSK modulation which is 1 bit per quadrature. It is also spread spectrum. Lots of military equipment use FSK, BPSK and QPSK. Usually no need for 88939 bit DACs (or even ADCs) there. Some applications are PAR constrained.
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As far as delta sigma, that's a different problem than 20/48 or 1582/52581. 1-bit delta sigmas can suffer from limit cycles and other issues, but n-bit delta sigmas can deal with some of those problems (ala AKM). And they can be very accurate.
This is going to sound over the top, but for digital and analog communications EEs one important concept is the one about channel capacity (the Shannon Limit). It is given by BW*log2(1+SNR) where I guess BW is related to oversampling and SNR is related to n-bits. When given the choice between BW or SNR, one should increase BW (though one should really increase both if sky is the limit). Delta Sigmas, Spread Spectrum, Error Correction and other crap work and correlate highly to the channel capacity idea.
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The best amps I've heard (and designed) in the past did two things:
Exactly. I think you guys are missing one of the major points involved with how the Rag and potentially the Yggy perform which was always a critical component to the mission of this site. Correlating how it sounds by ear to how things perform or measure. A true non audio engineer will be quick to point out all that they know from a few blurb spread by non experts across the internet or various other channels and perpetuate the myth of chasing a few basic metrics like decimal places that many have for more than 40 years and get nowhere. This is the everything that isn't broken sounds the same crowd. The real audio engineers have the education and experience to know what the don't know and use induction to correlate their vast experiences into audible results that may or may not mesh with accepted parameters of various audio lynchmobs who don't know what they don't know.
I think Jason's post shows the proper approach to design which is comprehensive and non prejudiced. It's no wonder that many of the vendors that are often favorably reviewed hear often have similar philosophies and approaches to design and engineering. The proof is simply in the pudding. Sadly there will always be those that don't like pudding.
On the amp itself, one of the most impressive features for me that I've taken issue with in about every SS amp out there is the apparent lack or diminished role of thermal drift impacting sonics so far.
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Despite Jason mentioning designers will probably disagree with what he is 'claiming' I don't think they will.
I don't in any case.
Firstly PCB design is very important (even more so in digital IMO) and certainly when higher currents are in play (speaker amps)
Also I agree with his feedback assesment (crap happening in the amp being 'nulled' with a counter input voltage at the input stage)
Furthermore I don't think topology/components used is that important by itself.
Accuracy (on a number of aspects) is important to get good results.
I use non overall and local (if any) feedback in my designs myself and even though it is having higher amount of distortion I don't think those levels of distortion are detrimental to the sound if they remain low enough under real world circumstances (isn't the same as on a test bench).
IMO opinion an amp should be tested under 'real loads' with real music and nulling input and output and listening to the result (the nulled) signal can give a good indication of how good an amp is.
test bench results are handy to show some basic and comparative numbers that should be viewed as a whole (and not as individual parameters) and should also be accompanied with some plots to show relations.
It is impossible to test out any speaker amp with all speakers and see which may not be driven well and standard measurements (which could be used for comparing aspects) WON'T show all aspects.
It is NOT that it cannot be measured or quantified or is impossible to measure but there are no 'standards' on how a speaker or headphone should/must behave.
They certainly aren't linear by a long shot.
So... yes .. measurements don't say it all and not because differences cannot be measured BUT because there is no standard to which they should comply and can be compared.
Having a 'standard' speaker load emulation (a selection of inductors, coils and resistors and perhaps even back EMF ?) will not be enough as amp designers could 'cheat' and get that to perform well but react poorly to other loads.
There is really no way to test/compare it all. Even a differential amp cannot show everything because it also reacts to phase differences which may not be detrimental but do show up as an error signal.
Listening to a differential signal under different difficult loads (NOT a digitally recorded dif signal) is a really good way to asses how an amp performs in real life situations with music.
Having more than 24 bits won't enhance resolution in practice simply because of S/N ratio and accuracy/linearity of components the DAC is made of are the limiting factor.
Ladder DACs with real 24 bit linearity simply cannot be created simply by lack of accuracy (tolerance) of parts.
This is easier to do with oversampling as higher frequencies aren't as difficult to make as higher accuracy.
Audio isn't voodoo or an art form it's just that not all designers know what matters the most and some of them (mostly bigger companies) only design to earn a living and make devices that must be able to do more or other things than other brands. Also in order to increase sales.
To me it appears as though Schiit doesn't give a schiit about it but wants to bring well made stuff for 'low end' prices.
They could easily move their schiit into a much higher price range (at slightly higher production costs) and sell it for 10x as much.
In that case they would likely sell 10x less schiit and thus not get richer so I REALLY appreciate their philosophy of good affordable schiit for the audio minded masses.
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The little I know about thermal drift is that as things get hot, optimal linear operating points change and it seems the Rggy has a uP specifically to deal with that.
I also liked the pudding, and pretty much every Schiit amp I heard so far. A lot.
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There are a lot of crap sounding SS amps out there that claim to represent the pinnacle of electronic transparency possible yet pale in comparison to the performance and sound of the Rag. I fail to see how those designers would agree with Jason's approach. I can think of at least 5-6 of them off the top of my head who are commonly referenced and purchased if you want me to name names.
I, like you, would like to think do but I simply don't see it based on the sort of product proliferation I see out there.
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You mean Self (among them ppls)? Nah. I think he is or was with Cambridge Audio. To him anything not the Azur 840A is proly crap, unless your name is [insert some random dude Self admires].
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Anax brings up a good point with regard to different designers coming from different backgrounds having converging philosophies. It seems to me the best amplifiers pay close attention to the implementation of the feedback loops, DC servos, biasing conditions, board layout, etc. Ti Kan, Cavalli, Gilmore, etc. (all of which make amps that pirates think favorably of) subscribe to design philosophies which aren't too far removed from Jason's. Albeit with quirks here and there.
It's really the Self's and Nwavguy's that are the minority here. They just have a disproportionately louder voice due to books, blogs, seminars combined with rabid minions set on pushing a parallel agenda.
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Can you guys tell me what's WRONG with D Self's designs in a technical sense ?
(aside from the use of overall feedback, which can work quite well IF executed properly)
Could it be that someone who BUILDS his designs on breadboard, or makes a PCB himself, NOT being aware of proper layout could F-up the design in such a way that it sounds bad ?
Is that the designers (D.Self's) fault and does it make him someone that has no clue about audio ?
Or is he a bad designer because he measures/researches things to death and has a highly 'objective' standpoint because of it ?
The O2 design is just a simple C'Moy with the output stage doubled.
Could the PCB layout be improved ? yes
Would it have sounded better ? unlikely... it's just a C'moy and that's all what you can expect of it.
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Can you guys tell me what's WRONG with D Self's designs in a technical sense ?
(aside from the use of overall feedback, which can work quite well IF executed properly)
Could it be that someone who BUILDS his designs on breadboard, or makes a PCB himself, NOT being aware of proper layout could F-up the design in such a way that it sounds bad ?
Is that the designers (D.Self's) fault and does it make him someone that has no clue about audio ?
The O2 design is just a simple C'Moy with the output stage doubled.
Could the PCB layout be improved ? yes
Would it have sounded better ? unlikely... it's just a C'moy.
So in the end the famous O2 is basically a simple C'Moy with a bit more power.
Is it arguably the 'end all amp' even though he claimed it was ?
Nope... it's just another C'Moy and that's all what you can expect of it.
Thousands of people love it to death... just not some guys here.
Self's designs all work very well from a THD and stability perspective, based on testing with steady-state sine waves. It's entirely possible that many of them could be butchered on the board, because it doesn't take much to turn 0.001% THD to 0.1% with high currents running around.
The grand irony with his work is that he simply codified the steady-state distortion mechanisms that any competent amp designer already knew about, and were typically already working to eliminate.
The main problem I have with him is that--unlike other writers like Cordell--he acts as a demagogue, declaring anything that deviates from his designs as "undesirable" or even not "allowable." See his latest rant on Class A amplifiers. Any topologies or approaches that deviate from his, whether CFA, folded cascode, single voltage gain stage, no overall feedback, tube gain, tubes, transformers, Class A, hell, even just using JFETs in the input stage--are instantly dismissed, and in sharply judgmental terms.
In my opinion, this kind of demagoguery not only causes divisiveness (by the acolytes convinced that there is One True Path) and actively inhibits any real advancement, since, hey, if the 5532 is the ultimate expression of what is needed for audio (per Self), why ever try for more?
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I agree with the 'attitude' thing... it's unpleasant and far from 'constructive'.
It's similar to Harold's attitude and why he isn't liked much but the O2 will always be an O2.
Bob's books are indeed fun reading material.
Still, I don't think that some, well laid-out, D-Self designs can't sound good just because he is a 'know it all'.
I never built any of his designs though nor read his books/articles thoroughly, but when browsing through them I don't see a 'poor design' methodology which would result in bad sounding amps when properly built.
It seems that people not liking those 'characters' (and their disciples) as well as their attitudes, reflects on how their designs are (possibly) perceived.
I also agree about 'the One True Path' (super specs on the test bench is all that's needed) is plain insightful and wrong.
There isn't a 'One True Path' nor a 'best amp' there are LOTS of ways of creating designs and LOTS of good sounding amps (as well as crappy ones ;D )
On the other hand the all subjective 'lets replace this or that diode/resistor in the power supply and it sounds better' path is just as wrong IMO.
There is something in the middle which is why I like Schiit products.
No claims, no BS (IMO).... nice looking and good performing and affordable designs ... just how I like it.
Just not everyone thinks that way.
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Still, I don't think that some, well laid-out, D-Self designs can't sound good just because he is a 'know it all'.
Where did you see anyone say that?
The problem that myself, Jason and others have stated is that while they might produce something that sounds acceptable or 'good', their approach dismisses the potential for anything better. Theirs is the 'reference' approach and any design that deviates is sacrilege to physics and basic science. Not to mention hearing anything to sound better is also tantamount to any combination of ignorance/placebo/confirmation bias/psychological delusion.
Also, no one here has ever said you can measure anything too much. In fact, most of us have been adamant that many of the so called self proclaimed measurements crowd aren't measuring enough or are measuring the wrong things. This is because many of those folk haven't bothered to correlate actual listening and musical experience to such measurements. Listening and hearing is untrustworthy. The ears and brain are always deceivers. Hell, many profess most sensory input to be basically illusory. Most read a forceful blurb, article or a few notes and think to themselves, "Ah hah! I got it all figured out now." That attitude and approach is anathema to progress and basically human interaction tbh.
It's simply easier for some folks to feel secure in a blanket of half truths rather than go naked into what often times is a wilderness of ignorance. In that I mean no matter how much we know, we are all ignorant in an absolute epistemological sense so we should approach the world as such. That's really the core of the scientific approach and method. Assuming that we know nothing, rather than knowing everything and confirming that bias.
Overall though, as you say I think most of us here are largely in agreement.
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Yup. Him-Self designs are not necessarily miserable POSes IMO. But, for example, I started reading one of his books sometime ago, and a WTF moment hit me when reading one of his bookes Chapter 1's section on "Science and Subjectivism".
There is this attitude by some folks about "the right way" or "the rules" which I find a bit restrictive and progress inhibiting. It maybe understandably comfortable for some though.
It is what it is. Take the good and leave the bad.
I also consider myself much more of a learner at this point than anything else. Some of the discussion here pushes me to read and try shit out. I truly enjoy it.
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I also consider myself much more of a learner at this point than anything else. Some of the discussion here pushes me to read and try shit out. I truly enjoy it.
Yup!! Same. What's odd is that I detest deconstructionist philosophy as pretentious and self-serving yet the same approach is critical for science. Question everything!
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Questioning can be pretentious, but it's not always that way IMO. For example, not following "the way" is proly a result of honest questioning of "the rules". Good things can come out of that.
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I've built amplifiers that followed the "blameless" template, more or less. Not because of Self, but because they're what any competent amp designer would do when designing a typical speaker amp. They're definitely going to be stable (but that's less topology and more compensation), and they'll measure well. They can sound good.
Its just...just the out-of-hand dismissal that irks me. "JFETs are unsuitable for an input differential amp," for example. Oh yeah, I'd rather use BJTs and deal with all the RF garbage that's going to be rectified by the PN junction in a modern, cellphone-infested environment. Fun fact: Ragnarok would be a complete no-go with BJT inputs, since the input ladder switching would cause HUGE transients due to the fact that they are current-input devices. JFETs, no problem.
But yeah, sounds like I'm with most of you here...still learning. My conclusions to date may be wrong. They may be revised in the future. I haven't talked about other factors, like high rails for linear operating points without feedback, open-loop gain being greater than the audio band, etc...which also may be factors. I may blather away a little more about that in the future.
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You can blather all you want Jason - I (and I'm sure others) appreciate the insights into what you've learned, what you've built, and whatever else. Audio / Audio Electronics keeps me engaged because I don't see an end to what I can learn. Thanks for helping 'learn' me :-)
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The main problem I have with him is that--unlike other writers like Cordell--he acts as a demagogue, declaring anything that deviates from his designs as "undesirable" or even not "allowable." See his latest rant on Class A amplifiers. Any topologies or approaches that deviate from his, whether CFA, folded cascode, single voltage gain stage, no overall feedback, tube gain, tubes, transformers, Class A, hell, even just using JFETs in the input stage--are instantly dismissed, and in sharply judgmental terms.
In my opinion, this kind of demagoguery not only causes divisiveness (by the acolytes convinced that there is One True Path) and actively inhibits any real advancement, since, hey, if the 5532 is the ultimate expression of what is needed for audio (per Self), why ever try for more?
As far as JFETs I see what you mean. Just skimmed through this bit:
http://www.eetimes.com/document.asp?doc_id=1278805&page_number=1
http://www.eetimes.com/document.asp?doc_id=1278866
http://www.eetimes.com/document.asp?doc_id=1278963&page_number=1
http://www.eetimes.com/document.asp?doc_id=1279027
Not sure, but I guess he likes the 5532.
"With horrible inevitability, the very popularity and excellent technical performance of the 5532 has led to it being criticized by subjectivists who have contrived to convince themselves that they can tell op-amps apart by listening to music played through them. This always makes me laugh, because there is probably no music on the planet that has not passed through a hundred or more 5532s on its way to the consumer." - Douglas Self.
Not sure why DS is laughing. The fact that the 5532 is widely used has proly more to do with economics and not cuz the 5532 is a wire-with-gain. Likely many consoles impart their own characteristics to the recorded music. Perhaps even more significantly so, music goes through room coloration at a studio and sometimes very desirably so. And weren't many well regarded recordings made with tube amp ridden consoles, which were specifically picked for their sound qualities? Think I saw that in the Sound City video or something.
Also, as far as I could tell there were BJT OpAmps that didn't do too great (OP270) and JFETs that kicked some ass (Difet OPA627), by DS own standards. Only thing to bitch about the OPA627 I guess is the price. Again, that was a JFET OpAmp that kicked some serious ass, price aside. Also, don't think by his own measurements, that the OPA2134 did bad at all in some configurations.
I see a bit of pulling the "too expensive" card if random OpAmp kicks beloved NE5532's ass, but just in case there is a possible new "champion": LM4562 (Nw's OpAmp to rule them all)
And BTW, what does the Cambridge Audio 840e preamp has for headphone amp and perhaps balanced inputs? Apparently the awesomely awesome NE5532! (http://www.highfidelity.pl/!ev/artykuly/21_01_2009/ca.html)
According to The Absolute Sound (http://www.theabsolutesound.com/articles/tested-cambridge-azur-840e-preamplifier-and-840w-power-amplifier/):
"By any standard the Cambridge Audio Azur 840E and 840W rank as excellent components and at roughly $4200, a supernatural deal."
All those supernatural savings proly coming from the awesomely awesome <$1 NE5532.
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According to The Absolute Sound (http://www.theabsolutesound.com/articles/tested-cambridge-azur-840e-preamplifier-and-840w-power-amplifier/):
"By any standard the Cambridge Audio Azur 840E and 840W rank as excellent components and at roughly $4200, a supernatural deal."
All those supernatural savings proly coming from the awesomely awesome <$1 NE5532.
HAH! Just like Cambridge's $5K transport killing 840 CD player. When TAS decides to pump up a brand, they go all in.
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"With horrible inevitability, the very popularity and excellent technical performance of the 5532 has led to it being criticized by subjectivists who have contrived to convince themselves that they can tell op-amps apart by listening to music played through them. This always makes me laugh, because there is probably no music on the planet that has not passed through a hundred or more 5532s on its way to the consumer." - Douglas Self.
Yup, I wonder how much he clings to that logic considering the number of tracks that have passed through tube gear on their way to all those consumers...
Nvm, we know it doesn't because logic is just too inconvenient sometimes. I also find the idea that most of the music that reaches consumers being reference quality to be laughable. Someone should start listening and stop talking.
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According to The Absolute Sound (http://www.theabsolutesound.com/articles/tested-cambridge-azur-840e-preamplifier-and-840w-power-amplifier/):
"By any standard the Cambridge Audio Azur 840E and 840W rank as excellent components and at roughly $4200, a supernatural deal."
All those supernatural savings proly coming from the awesomely awesome <$1 NE5532.
HAH! Just like Cambridge's $5K transport killing 840 CD player. When TAS decides to pump up a brand, they go all in.
I had the Cambridge Audio Azur 351C to test side by side against the PS Audio PWT, Oppo BDP-105, Tascam/Teac CD200, and both Marantz 5004 (modded) and 6004 against my vintage Denon. It was nothing really all that special. It did come in third behind the Marantz 5004 and my Denon and had rather sketchy build quality.
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Let me highlight the inflammatory and judgmental aspects of this quote, then I will shut up about it:
"With horrible inevitability, the very popularity and excellent technical performance of the 5532 has led to it being criticized by subjectivists who have contrived to convince themselves that they can tell op-amps apart by listening to music played through them. This always makes me laugh, because there is probably no music on the planet that has not passed through a hundred or more 5532s on its way to the consumer."
For example, does the above do anything that this impartial rewrite does not--besides incite anger?
Although the 5532 delivers excellent measured performance, and has been used widely in mastering consoles, some listeners believe that it is not a truly transparent device, and is bettered by other op-amps."
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Let me highlight the inflammatory and judgmental aspects of this quote, then I will shut up about it:
"With horrible inevitability, the very popularity and excellent technical performance of the 5532 has led to it being criticized by subjectivists who have contrived to convince themselves that they can tell op-amps apart by listening to music played through them. This always makes me laugh, because there is probably no music on the planet that has not passed through a hundred or more 5532s on its way to the consumer."
For example, does the above do anything that this impartial rewrite does not--besides incite anger?
Although the 5532 delivers excellent measured performance, and has been used widely in mastering consoles, some listeners believe that it is not a truly transparent device, and is bettered by other op-amps."
There was another dude who recently made waves by writing and making claims EXACTLY like that before tucking tail and disappearing.
It always completely AMAZED me that people could write like that and others just suck it up like non biased objective truth. Seems dangerous to me many people are so eager to accept or ignore such behavior.
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Thing is, the NE5532 is measurably worse than the best audio opamps you can get these days. Anyone still using it in high-end designs is either pinching pennies (seriously, even the "expensive" opamps are cheap) or just too self-righteous to admit otherwise. I just feel it's kinda embarrassing when you open up certain "high end" A/D or D/A converters and you see them using decade old parts that aren't even spec-ed for audio. The implementation is going to matter more than anything but you're not helping yourself out by sticking to what really is old IC tech.
I'm sure the engineering departments at TI and AD are full of evil "subjectivist" tweekers who should have given up a long time ago. Now they're just bored so they're trying to pull one over on all of us ::)
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"This always makes me laugh, because there is probably no music on the planet that has not passed through a hundred or more 5532s on its way to the consumer." - Douglas Self.
Some of the best most immediate sounding recordings I have (Louis Armstrong, Frank Sinatra, orchestra pieces w/ Stanley Black / LSO, etc.) passed through zero NE5532s. NE5532 didn't exist back then. In fact, I'm not even sure they could make ICs cheaply yet. I don't think the NE5532s even existed during the heyday of Blondie, Rick Springfield, Talking Heads, or Kool and the Gang.
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Thing is, the NE5532 is measurably worse than the best audio opamps you can get these days. Anyone still using it in high-end designs is either pinching pennies (seriously, even the "expensive" opamps are cheap) or just too self-righteous to admit otherwise. I just feel it's kinda embarrassing when you open up certain "high end" A/D or D/A converters and you see them using decade old parts that aren't even spec-ed for audio. The implementation is going to matter more than anything but you're not helping yourself out by sticking to what really is old IC tech.
I'm sure the engineering departments at TI and AD are full of evil "subjectivist" tweekers who should have given up a long time ago. Now they're just bored so they're trying to pull one over on all of us ::)
Benchmark says they are bestest:
Here you go: http://wiki.benchmarkmedia.com/wiki/index.php/NE5532_Opamp (http://wiki.benchmarkmedia.com/wiki/index.php/NE5532_Opamp)
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It can achieve very good measured performance (with a few caveats) when placed in a good implementation. No question there.
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Benchmark used to say a lot of things and support much of the Self philosophy. Oddly enough they seem to be pushing newer updated models that supposedly already surpassed the capability of human ears. Hmmm.....
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Let me highlight the inflammatory and judgmental aspects of this quote, then I will shut up about it:
"With horrible inevitability, the very popularity and excellent technical performance of the 5532 has led to it being criticized by subjectivists who have contrived to convince themselves that they can tell op-amps apart by listening to music played through them. This always makes me laugh, because there is probably no music on the planet that has not passed through a hundred or more 5532s on its way to the consumer."
For example, does the above do anything that this impartial rewrite does not--besides incite anger?
Although the 5532 delivers excellent measured performance, and has been used widely in mastering consoles, some listeners believe that it is not a truly transparent device, and is bettered by other op-amps."
Dunno, Doug's unbiased approach is moar dramatic. Emotional. It makes me lol, weap, cough, and sneeze. All at the same time.
It can achieve very good measured performance (with a few caveats) when placed in a good implementation. No question there.
"The NE5532 is power hungry, it has high input bias currents, and high offset voltage"
>So in other words, pure win.
"The 5532 should not be used with low signal levels, and it should only be used in low gain circuits."
>So moar win.
"It may surprise you that the NE5532 was selected for transparency and not on the basis of cost."
> ???
Benchmark used to say a lot of things and support much of the Self philosophy. Oddly enough they seem to be pushing newer updated models that supposedly already surpassed the capability of human ears. Hmmm.....
They are surpassing human ears capability version 69.0 this time though.
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It can achieve very good measured performance (with a few caveats) when placed in a good implementation. No question there.
LOL, Benchmark ended up replacing NE5532 with LM4562 in the USB and PRE versions.
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It can achieve very good measured performance (with a few caveats) when placed in a good implementation. No question there.
LOL, Benchmark ended up replacing NE5532 with LM4562 in the USB and PRE versions.
$3 for a dual channel audio opamp? Obvious snake-oil.
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Pfft, I'm gonna build my own discrete opamps and roll in the moolah that amp makers will throw at me.
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As with a lot of components....
It's HOW you use them that counts !
For example it didn't do so well (while clipping) in a C'Moy testrig.
http://www.mediafire.com/view/p2hnepv9u8dlbje/Portable_op-amp_headphone_amplifier_test.pdf
When NOT loading it with a headphone it does a bit better.
The LM4562 has long been surpassed by other opamps and the newer and higher spec'ed opamps are considered to sound better by some and sound worse by other 'experts'.
I see the Douglas 'point' about it ;D
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UB, was he listing the 5532 being "power hungry" as a positive? If so, what is his argument against Class A amps? Is he actually saying they sound worse or was it an argument based on efficiency?
What amps has Self designed, if any?
I see where you all are coming from regarding the way he presents his opinion, but I think the other side of the spectrum (subjectivist quacks, tweakos, etc) are far more ridiculous and nerve/mind grating.
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The power hungry claims were from Benchmark (see linky in Marv's post (http://www.changstar.com/index.php/topic,1782.msg48015.html#msg48015)). Definitively a negative, but seemingly waved as a minor issue.
Haven't looked into his beef with Class A, but proly has to do with his "uber-efficient" + "uber-accurate" Class XD stuff:
http://www.cambridgeaudio.com/blog/class-xd-explained
http://archive.cambridgeaudio.com/assets/documents/ElectronicsWorldNov06840Aweb.pdf
http://worldwide.espacenet.com/publicationDetails/originalDocument?CC=GB&NR=2424137&KC=&FT=E
As far as subjectivism, I actually see some of Self's views as examples of Self-serving subjectivism.
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Fascinating! This is way out of my knowledge base, but the scientist in me enjoys reading this stuff. I must tip my hat to Jason + crew.
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class XD is basically this: http://tangentsoft.net/audio/opamp-bias.html but applied to a power amp.
Nothing revolutionary there and simply shifting 'the problem' elsewhere.
I believe biassing a class A-B amp higher into class A will also shift 'the problem' similar to class XD but more symmetrical where in XD the 'problem' is just shifted (and still there).
The idea is to shift the crossover distortion at a higher output voltage level (where the speaker is loud already) and 'mask' the distortion (however small it is) by the distortion the speaker already has.
I think (feel free to correct me here Jason) that the Rag (and Sumo 9) also bias class-AB output stages in 'partly' class-A
It should be noted that a lot of amplifiers use this trick.
All you need is enough cooling surface basically.
With class XD the dissipated heat may be smaller though.
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class XD is basically this: http://tangentsoft.net/audio/opamp-bias.html but applied to a power amp.
Nothing revolutionary there and simply shifting 'the problem' elsewhere.
I believe biassing a class A-B amp higher into class A will also shift 'the problem' similar to class XD but more symmetrical where in XD the 'problem' is just shifted (and still there).
The idea is to shift the crossover distortion at a higher output voltage level (where the speaker is loud already) and 'mask' the distortion (however small it is) by the distortion the speaker already has.
I think (feel free to correct me here Jason) that the Rag (and Sumo 9) also bias class-AB output stages in 'partly' class-A
It should be noted that a lot of amplifiers use this trick.
All you need is enough cooling surface basically.
With class XD the dissipated heat may be smaller though.
You're right on with Class XD (as far as it not being anything revolutionary). If you want to see some interesting examples of conceptual circuits that go into some really wacky biasing arrangements (sliding bias, mode-shifting amps, etc)--check out Broskie at tubecad.com. His stuff usually isn't worked out to be the point of being "build it and go," but he has a lot of very interesting (and unique) ideas.
In terms of the Class A/AB question, it's slightly more complicated.
For Class A, there's really three kinds of "Class A."
1. High Bias Class AB. Most "Class A" speaker amps are simply high-bias Class AB designs. Think Krell, etc. Conventional complementary output stage, run with high bias. These kinds of Class A amps can go out of Class A mode and enter Class B at some operating point. Also, most small-signal preamps and such that are billed as "Class A" use this same model.
2. Real Class A. This is usually a single-ended design with current-source load. The current source sets the quiescent bias, and the amp can never go out of Class A. It also limits the total current it can source, so power and efficiency are much lower. Asgard 2 is a real Class A amp, for example.
3. Bullshit Class A. Manufacturers like to put "Class A" on everything under the sun, including things that are really not Class A, like op-amps with Class B output stages. This is why "Class A" claims have to be looked at with deep scrutiny. It may mean nothing.
Class AB is where it gets interesting, and it illuminates why we use MOSFETs for a lot of stuff:
Most amps are run in Class AB, where, in some operating range, they are working in Class A, then transition out into Class B. This is a good compromise for performance vs power draw. Magni, Vali, Mjolnir, and Ragnarok all use Class AB output stages.
However, there is actually an optimum bias point for a Class AB BJT design, at which crossover distortion is minimized and Gm doubling has not taken over as the primary distortion mechanism. This bias point is actually fairly low. Turn up the bias randomly on a Class AB BJT design, and the linearity can actually go down. So, if you're going to do a Class A amp of the high-bias AB type, you need lots and lots of output transistors. Now you know why Krell amps look the way they do.
Note: Cordell, Self, etc cover the optimal bias point for a BJT amp in their work, but it was well-understood before it was codified by them.
For MOSFETs, it's different--and simpler. The higher the bias, the greater the linearity. Period. Higher bias sounds better. Period. Of course, you still can't exceed the device's thermal limit, and you have to take into account the positive temperature coefficient of vertical MOSFETs, if that's what you're using.
But in general, a MOSFET amp can have a simpler output stage with fewer paralleled devices...which is good, because paralleled devices are always a sonic compromise. The more devices, the bigger the compromise. Yes, even when "matched."
You could say that the ideal output stage has single matched devices of the same type (N-channel or P-channel)--which, by the way, can only cleanly be implemented with a circlotron topology. Of course, this limits you to about a 60/100W amp into 8/4 ohms. Kinda like Ragnarok.
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Also, most DC servos do the same thing--sample the output, bring the DC correction signal back to the input. Of course, this also means you have a phase-shifted, frequency-dependent part of the output coming back as well, because a DC servo doesn't just amplify DC.
I never realized this about servos - thinking they were magical circuits which nulled the DC offset. Looking at a few circuits in TI documentation, a servo is basically feeding back the output via a low-pass filter no?
Just thinking out loud, for some reason, I always got the idea that servos were harmless, more or less.
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If you are indeed feeding it back to the input of the (entire) amplifier, then yes. Most clever designs that I've seen connect the servo (opamp's output) somewhere else in order to null the offset.
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Looking at a few circuits in TI documentation, a servo is basically feeding back the output via a low-pass filter no?
Yup, the used opamp works as a differential amp and compares the low pass filtered signal, which basically consists of DC + very low frequencies (think TT rumble).
Just thinking out loud, for some reason, I always got the idea that servos were harmless, more or less.
Certainly not harmless but they can alleviate some DC drift issues (caused by different temperatures mostly).
The amplifier basically has an extra feedback point with a very high gain for DC and lower gain the higher the frequency goes + the regular feedback (be it AC or DC).
Problems one may encounter using DC servo's are added noise (as the output of the used opamp is connected to the input of the amp (be it the inverting or non inverting) or somewhere in the middle of the circuit.
That opamps noise MAY become audible IF proper measures aren't taken.
I have seen circuits that actually weren't designed properly.
The DC output (over longer periods) is as good as the opamps input offset + leakage of used caps (usually high capacitance values)
Another thing that's already been touched is the used low pass frequency.
It's a compromise.
Make the frequency really low and you have great LF extension and no influence in the lows BUT the circuit is sluggish and on start-up may actually cause DC to appear on the output. Also when DC is suddenly present on the input it may take a long while before it is 'removed'.
Make the circuit react too fast and low frequencies are affected (basically roll-off) BUT the circuit is very fast and no DC will be on the output and TT rumble is filtered somewhat.
So.. fine to use DC servo BUT it must be designed properly, have no (very low) noise on the output, react fast enough BUT not affect the lowest octaves that contain useful info.
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For Class A, there's really three kinds of "Class A."
1. High Bias Class AB. Most "Class A" speaker amps are simply high-bias Class AB designs. Think Krell, etc. Conventional complementary output stage, run with high bias. These kinds of Class A amps can go out of Class A mode and enter Class B at some operating point. Also, most small-signal preamps and such that are billed as "Class A" use this same model.
In fact all opamps that are loaded so the output current is less than the bias current in the output stage is class-A
Clipping is symmetrical.
The amps that have idle currents exceeding the max drawn current (and thus get very hot) I would not call high bias class-AB but simply class-A (thus in push-pull) as there is nothing AB about it even though the output stage can be similar to an AB amp.
A LOT of 'class-A' amps are indeed 'partly class-A' and do the rest in AB.
2. Real Class A. This is usually a single-ended design with current-source load. The current source sets the quiescent bias, and the amp can never go out of Class A. It also limits the total current it can source, so power and efficiency are much lower. Asgard 2 is a real Class A amp, for example.
Slippery slope there... the term real class-A I mean...
The high bias class-A (when the quiescent current is always higher than the max drawn PEAK current (= higher than effective current) are very real class-A amps as well, nothing fake or unreal about it.
... I would call it Single ended class-A instead as it just as real class-A as the PP version.
A disadvantage of the circuit is asymmetrical clipping which can cause DC on the output when the current source is over-driven and the output stage isn't current limited, but this can be tackled easily.
3. Bullshit Class A. Manufacturers like to put "Class A" on everything under the sun, including things that are really not Class A, like op-amps with Class B output stages. This is why "Class A" claims have to be looked at with deep scrutiny. It may mean nothing.
I used to work at Technics... They also had a few 'BS'-class-A designs iterations.
New Class-A (synchro bias) basically is a standard AB amp but the bias voltage to the output stage came via germanium diodes (for softer switching) and there was also a 'fixed' bias voltage applied to the output stage via another set of diodes.
In class-AB the output devices turn off completely when the other 'side' conducts.
In this topology the devices that turned 'off' always remained a tiny bit 'on' instead.
This was done to lower (not eliminate) crossover distortion significantly as the transistor didn't go completely 'off' any more and thus goes 'on' much faster as the Vbe didn't reverse any more.
Later on they made 'computer drive new class-A' which basically is the same but the idle current was controlled by a processor that 'pre-heated' the power stage and then controlled the bias current.
There was also class-AA which basically was a class AB amp with a high voltage rails which 'drove' a low voltage (literally a few volt!) output stage with a very high idle current. That low voltage amp fed the speakers as it floated along with the AB amp which provided the actual voltage swing.
I still have some 'own design' built versions of both principles in the attic. They might even still work when powered up.
Of course BS class-A as the output current had to go through the AB stage as well but the crossover distortion was 'compensated' by the class A part.
A few iterations were there with BJT(AB) + MOSFET(A) and all BJT and later the class-A stage was a small module instead of discrete.
These 2 amps were joined with a bridge circuit.
It was thus sort-off current dumping with a twist where the overall feedback point was the actual output signal where in the current dumping it is coming from another point.
However, there is actually an optimum bias point for a Class AB BJT design, at which crossover distortion is minimized and Gm doubling has not taken over as the primary distortion mechanism. This bias point is actually fairly low. Turn up the bias randomly on a Class AB BJT design, and the linearity can actually go down. So, if you're going to do a Class A amp of the high-bias AB type, you need lots and lots of output transistors. Now you know why Krell amps look the way they do.
Indeed... most DIY'ers aren't aware !
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But in general, a MOSFET amp can have a simpler output stage with fewer paralleled devices...which is good, because paralleled devices are always a sonic compromise. The more devices, the bigger the compromise. Yes, even when "matched."
Does this also apply to dacs which stack a bunch of chips together? (a big generalization I realize)
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But in general, a MOSFET amp can have a simpler output stage with fewer paralleled devices...which is good, because paralleled devices are always a sonic compromise. The more devices, the bigger the compromise. Yes, even when "matched."
Does this also apply to dacs which stack a bunch of chips together? (a big generalization I realize)
Going to depend on the chip. With current output DACs that are well matched I wouldn't be afraid of paralleling them. Someone else could provide some insight into any negative side effects (there's always pros and cons in engineering)
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But in general, a MOSFET amp can have a simpler output stage with fewer paralleled devices...which is good, because paralleled devices are always a sonic compromise. The more devices, the bigger the compromise. Yes, even when "matched."
Does this also apply to dacs which stack a bunch of chips together? (a big generalization I realize)
Usually this was done to improve linearity.
The older 16 bit ladder DAC chips had problems with linearity (you were lucky to get 14 bits real resolution out of them).
It is VERY hard to make the MSB with such a tight tolerance compared to the LSB.
Paralleling those chips averaged out the errors of many chips (and increased the total output current) and linearity got a bit better.
In some low noise pre-amps some parallel input transistors, in this case the theory is that the noise (being random) doesn't 'add' as much as the signal which is common for these devices and thus effectively increasing S/N ratio.
For power amps it is needed to spread currents, or power dissipation (less heat/chip), over more than one device for technical reasons.
The O2 uses 2 IC 'halves' to increase output current.
I once made a power amp with a load of (high current) opamps in parallel.
Simply to increase output power and spread heat dissipation.
(http://diyaudioheaven.files.wordpress.com/2013/03/opampamp.jpg?w=921&h=933)
Just for the fun of it and because I had the parts lying around.
You can also parallel lots of tubes to increase output power
(http://www.soundstage.com/revequip/pics/atmasphere_ma2ii3_rear.jpg)
So there are plenty of reasons to parallel devices.
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The reason I asked about parallel dac chips was due to reading about the DDDAC1794 project, and this note here... http://www.dddac.com/dddac1794_other.html
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I actually got curious about them op-amps. What ya guys think of this one:
http://www.ti.com/lit/ds/symlink/lme49726.pdf
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A nice theory that assumes jitter in each component either varies or is linearly spread among the chips.
Of course it is just HIS theory because he needs to explain something that he heard and appears to be getting a lot of 'heat' from others.
It would be nice to see actual shots of the final risetime with say 1, 10, 60 or 240 chips showing jitter, caused by DAC chips having different 'trigger points' AND slew rate improves with more (TDA1543) chips used.
It must be one hell of a clock and data driver being able to 'charge' the capacitance of so many paralleled DAC chip inputs/wiring not to mention the local decoupling and power supply which would be a real pain to get right with so many 'switches' switching at the same time :D
I would love to see the RFI emission that DAC shows.
Anyway... The spread that is in there will not be linear (as per his explanation drawing) but Gaussian BUT the jitter caused by the DAC chips NOT 'switching' at the same time will be improved for sure.
Too bad this jitter is much smaller than the jitter coming from the drivers that supply the clock (my assumption)
So yes... he may well perceive an improved SQ with more chips.
The question remains WHAT that would be causing.
I reckon it would be VERY EASY for such a DAC pioneer to show some real life scope pictures showing the improvements.
Now lets hope his scope also always 'triggers' at the exact same level :)p17
Having more chips in parallel also creates other problems such as power consumption, power supply decoupling, the current needed to charge capacitances, length of wiring.
The higher the frequency goes (1794 is oversampling) the more problems you get in return.
Therfore I think it is impossible to predict how multiple 1794 DAC chips in parallel will yield a more accurate signal / affect subjectively determined SQ.
Most likely it will be an improvement to those that do.
But as with everything to get more (real) improvement you need to double the amount of chips every time you want more 1-2-4-8-16-32-64-128 etc.
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I actually got curious about them op-amps. What ya guys think of this one:
http://www.ti.com/lit/ds/symlink/lme49726.pdf
Very nice, alas only suited for 5V supply rails (+/- 2.5V) so great for low voltage DAPs that need to drive 8 Ohm or 16 Ohm loads.
Can't be used for +/-5V or higher supply voltage rails. :-[
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On DAC linearity. I'd suppose better linearity correlates to better SQ.
I hacked up a DCX2496 (digital EQ with 6 channels) and paralleled the 3 AKM chips' output. This seemed to increase resolution from a subjective POV.
Would be interesting to measure DAC linearity. Plot X vs. Y where X is code (let's start at 16-bit) and Y it output voltage. Might be an interesting measurement.
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Just injecting a caveat here. One of the problems with many making a subjective claim of 'better' IME is that they often just mean different (i.e. I now hear more of this but have lost that in doing so). It's human nature to perceive new=better. One needs to frame comments in context of how experienced is the listener, how clearly do they communicate, is there an agenda, how much are they being paid or compensated, what do they stand to gain, etc. That way we can better parse what is valuable data from the chaff for constructive dialogue.
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Just injecting a caveat here. One of the problems with many making a subjective claim of 'better' IME is that they often just mean different (i.e. I now hear more of this but have lost that in doing so). It's human nature to perceive new=better. One needs to frame comments in context of how experienced is the listener, how clearly do they communicate, is there an agenda, how much are they being paid or compensated, what do they stand to gain, etc. That way we can better parse what is valuable data from the chaff for constructive dialogue.
Yep, that's one of the reasons our dev cycle is much longer than it used to be.
You do have to sit back, listen, test, and get perspective on new products, especially since, as designers, we're so close to them. It's thrilling when you first fire something up and have a listen...but you may be too close to the product to be impartial at that moment in time.
It's not unusual for us to intro something 18 months after first fire-up now. But there's a lot of stuff that's been on the burner...
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Just injecting a caveat here. One of the problems with many making a subjective claim of 'better' IME is that they often just mean different (i.e. I now hear more of this but have lost that in doing so). It's human nature to perceive new=better. One needs to frame comments in context of how experienced is the listener, how clearly do they communicate, is there an agenda, how much are they being paid or compensated, what do they stand to gain, etc. That way we can better parse what is valuable data from the chaff for constructive dialogue.
Yep, that's one of the reasons our dev cycle is much longer than it used to be.
You do have to sit back, listen, test, and get perspective on new products, especially since, as designers, we're so close to them. It's thrilling when you first fire something up and have a listen...but you may be too close to the product to be impartial at that moment in time.
It's not unusual for us to intro something 18 months after first fire-up now. But there's a lot of stuff that's been on the burner...
By the way, you mentioned having an epiphany a little while back. If you explained what it was, I missed it. Could you share it?
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240 TDA1543s? It's kind of like advocating 1000 grit sandpaper for a slip 'n' slide. Yes, it's better than using 40 grit, but still... why??
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The problem that myself, Jason and others have stated is that while they might produce something that sounds acceptable or 'good', their approach dismisses the potential for anything better. Theirs is the 'reference' approach and any design that deviates is sacrilege to physics and basic science. Not to mention hearing anything to sound better is also tantamount to any combination of ignorance/placebo/confirmation bias/psychological delusion.
Also, no one here has ever said you can measure anything too much. In fact, most of us have been adamant that many of the so called self proclaimed measurements crowd aren't measuring enough or are measuring the wrong things. This is because many of those folk haven't bothered to correlate actual listening and musical experience to such measurements. Listening and hearing is untrustworthy. The ears and brain are always deceivers. Hell, many profess most sensory input to be basically illusory. Most read a forceful blurb, article or a few notes and think to themselves, "Ah hah! I got it all figured out now." That attitude and approach is anathema to progress and basically human interaction tbh.
It's simply easier for some folks to feel secure in a blanket of half truths rather than go naked into what often times is a wilderness of ignorance. In that I mean no matter how much we know, we are all ignorant in an absolute epistemological sense so we should approach the world as such. That's really the core of the scientific approach and method. Assuming that we know nothing, rather than knowing everything and confirming that bias.
What the hell are you on about?
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What the hell are you on about?
You know exactly what I'm talking about. You can start here. http://www.head-fi.org/t/685704/oppo-pm-1-a-new-planar-magnetic-headphone/2490#post_10914588
Or you can start at every post this clown or clowns like him and his type of crowd have ever made.
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What the hell are you on about?
You know exactly what I'm talking about. You can start here. http://www.head-fi.org/t/685704/oppo-pm-1-a-new-planar-magnetic-headphone/2490#post_10914588
Or you can start at every post this clown or clowns like him and his type of crowd have ever made.
Yep. Sometimes I wonder who's objectivist and who's subjectivist after a while. There are too many people relying on selective memory to make argument.
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You know exactly what I'm talking about. You can start here. http://www.head-fi.org/t/685704/oppo-pm-1-a-new-planar-magnetic-headphone/2490#post_10914588
Or you can start at every post this clown or clowns like him and his type of crowd have ever made.
Lol OMG that is so funny as I wasn't thinking of that clown,
but it is so be fitting lol!
They are running rampant over there.
Too many fools like that being stumbling blocks to members learning progress in this audio industry.
Makes me think they are haters.. of hearing.
Whatever happened to "Hearing is believing"?
They can't trust their ears so they stumble on incomplete measuring tools and fool the ignorant.
Makes me remember a clasd long ago where the dam college professor was expounding how feedback made the opamp the best design ever.
I was lucky to have had a teacher to show me his best measuring instrument, in which he relied on and which his livelihood relied on...
his EARS..
He was a recording engineer and used his ear to make judgements on microphone placement and never looked at dial or slider positions.
He told me if he didn't hear what he needed to keep adjusting, keep moving that dial untill it sounded right..
To TRUST YOUR EARS..
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Since becoming part of this community of obsessives, I think I am of the "trust, your ears, , but also look at measurements" crowd. They don't always explain the sound you hear, but they do give you more information than just your ears. However, you must have competent people doing the measurements and also understand what they might mean. Seems some folks on certain sites just repeat what they read elsewhere. Why say you haven't heard something, but expect or just say it sounds good, or bad because"."..............???????
It still comes down to what makes you get drawn into the music. I must agree, the people on certain sites really lead newcomers down a path of spending money and dispensing mis-information. Had it happen to more than once in the past . At least here we can call bullshit. And in that spirit I call bullshit on myself. :)p13
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Makes me think they are haters.. of hearing.
Whatever happened to "Hearing is believing"?
They can't trust their ears so they stumble on incomplete measuring tools and fool the ignorant.
Nobody hates hearing/Listening. If they did, they'd set up the gear, set up the music, check the "measurements" ...and go out.
He was a recording engineer and used his ear to make judgements on microphone placement and never looked at dial or slider positions.
He told me if he didn't hear what he needed to keep adjusting, keep moving that dial untill it sounded right.
Which invalidates measurements ...how? If he didn't know his measurements he wouldn't even know which dial to tweak.
No doubt you are right that a little knowledge can be worse than none. I have less than a little knowledge, but try to learn about the measurements and, for instance, how to use EQ to compensate for the frequencies I don't hear very well. I also very occasionally make some half-arsed attempt at blind comparison of something where I'm not convinced, or I think I'm hearing something that should not be there. I'd do it a lot more if I wasn't extremely lazy.
The brain not only filters what the ears feed it, but I think there is some feedback system too: the brain also tells the ears what to hear. Don't trust your ears: decide if you can trust your brain. That's the important thing.
But, above all, and most of the time... put all this admittedly extremely interesting technical stuff on one side and just enjoy music.
That's what I do. But, sadly, I find that there are places on the net where even a trace of sympathy for anything objective at all leads to one being branded as some kind of monster who wouldn't know the emotion of music if it kneed them hard between the legs and, as that kind of audiophile is usually not very good at listening (how do they ever enjoy music?) it's not much help telling them that, hey, music is what I am into to, not graphs and numbers.
hey-ho... just another rant, and probably off-topic. Feel free to consign to mental garbage :)
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...
hey-ho... just another rant, and probably off-topic. Feel free to consign to mental garbage :)
There has to be a balance and growth without one ideology dismissing the other, guided by common sense, not irrational extremes which end up getting nowhere by being at odds...
kinda like science and religion being placed at odds when they naturally would benifit each other, reaching further understanding if they were together.. .
blab done.
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There has to be a balance and growth without one ideology dismissing the other, guided by common sense, not irrational extremes which end up getting nowhere by being at odds...
Totally agree. Part of the problem is that they have become "ideologies," rather than simple practical combination of engineering, tools and experience.