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Oh geez they could have left this statement out: "When I wrote my previous article, class-D audio was fairly new, and audiophiles were still arguing about whether class-D amps could sound as good as class-AB or class-A units.

Nobody is having those arguments anymore."

Those arguments are still front and center when it comes to amplifier design, both on the manufacturer and consumer sides.

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I'm not convinced audio engineers are having that argument anymore (which is more the audience of IEEE). I get the impression from an engineer I know who's specialized in digital signal processing, and who's done a significant amount of analog work, that anyone who's still having some of these arguments doesn't understand some of the basic maths of the problem space. That basically, you can always digitally encode an analog signal at a higher resolution and fidelity than you can reliably read said signal at any given technology level/pricepoint. Which means that using an ADC -> DAC pipeline will almost always give you better audio output with better signal to noise ratios than strict analog -> analog systems (though possibly at the cost of latency).

Where it gets tricky, is the arguments for how things like tube amps sound. There is nothing that analog components add to a sound that can't be simulated with digital transforms, but the trick is figuring out how various analog components effect the signal. For instance, my friend found that certain tube amps would overdrive components and force them out of spec, and this caused a non linearity in the response of the signal coming from the component that gave the amp it's distinctive distortion effect. Once he figured that out he was able to reproduce the effect digitally. Incidently, he's found that some guitar amps can never be made today, because they were built around a specific batch of a given component, and newer batches didn't have the same out of bounds responses. But this seems to be why so many people are still so big into analog amps, because it takes considerable effort (and apparently engineering skill) to reproduce some of the classic sounds.

Mind you, I'm biased by one of the best engineers I've ever met, so I could be wrong about the general consensus with audio engineers. I'm also mostly using a sample size of 1, though I have corroborated some of his assertions with some of the blogging I've read on the topic. One of the best (I wish I'd bookmarked), was a good overview of encoding by a prolific codec writer, who pointed out that even though a digitized sine wave looks lossy at a low sample rate, it's trivial to rebuild that sine wave perfectly, with just the digitized version (so long as the sample rate was above some multiple of the Nyquist frequency?).

Likely that was xiph on why excessively high sample rates don't do much for the listening experience: https://people.xiph.org/~xiphmont/demo/neil-young.html

Amplifier designers most certainly are. Class D has significant energy efficiency advantages, and significant sonic challenges. It’s taking an analog waveform, whether decoded by a DAC or starting as pure analog, and chopping it into a series of pulses (pulse width modulation). The amp is switching transistors on and off very quickly, rather than using them as a variable linear gain stage. It’s not the same as PCM encoded digital audio, and shouldn’t be confused with it.

The energy efficiency advantages in class D are incredible. When my old Technics receiver stopped working because it started overheating, i replaced it with a class D amp on a half-credit card sized circuit board from dx.com and taped it to the back of the TV. 20W per channel, which is plenty for my casual TV watching. The Technics was the size of several stacked pizza boxes, with fans and heat sinks etc.

> Amplifier designers most certainly are.

Amplifier designers have to sell to a market that is rife with misinformation and a very poor signal-to-noise ratio between technical quality, profit margins and overall market success. I wouldn't expect many of them—especially the ones who have any name recognition—to be concerning themselves with objective technical performance.

Whereas the guy who designed the amplifier circuits for a product where the customer isn't being placeboed by the spec sheet—like powered studio monitors, or consumer devices like the Apple HomePod—is far more likely to be focusing on objective measurement rather than faulty market perceptions.

The thing is that transistors’ behvior is more linear and controlled when fully on or off rather than when used as continuously variable. Class-D can and does sound _amazing_ if built with good theoretical understanding. Check out Hypex amps designed by Dutch engineer Bruno Putzeys on strong control theoretical foundations.

That’s precisely and exactly the opposite of what ‘linear’ means when talking about the operating point of transistors. I think you should rephrase your argument.

I'm going with an understanding similar to what is presented e.g. here: https://www.mwrf.com/components/learn-meaning-amplifier-line...

"The term linearity derives from an amplifier's linear relationship of input power to output power which, in an ideal amplifier, would be precisely related by the gain of the amplifier."

I see it as matching my argument.

Could I ask you to give a definition which clashes? Thanks!

Yes, the definition of linearity is that f(A) + f(B) = f(A+B), which you have contradicted in your GP post. If a transistor is fully on, then doubling the input voltage to it will not double the output, because it's already at maximum. That is very precisely not a linear device. That's why linear amplifiers, like Class A, operate the transistor in a partially switched state, where increasing the input signal (about some bias point, and with feedback to turn a real transistor into a more ideally-behaving one) gives a corresponding increase in output signal.

Class D (switching) amplifiers are _not_ linear, they trade linearity for efficiency. The article you linked to actually says this:

"Over the years, a number of different biasing schemes have been developed for amplifiers, from extremely linear Class A operation, in which the active devices essentially remain powered on through the full sinewave cycle of an input signal, to more power-efficient schemes, such as Class D, in which multiple transistors are switched on and off to conduct different portions of an input waveform."

I think what you are doing is confusing the linearity of the transistor with the linearity of the whole audio amplifier treated as a black box, which is why I suggested you rephrase your argument rather than say you are categorically wrong, however what you said was:

"The thing is that transistors’ behavior is more linear and controlled when fully on or off rather than when used as continuously variable."

... which I'm afraid is really just wrong.

Thanks but no thanks.

When a transistor is off, it’s off. Then it takes a certain input voltage to get it going. Then there’s a varying curve to full output.

In contrast, fully switching on or off “gets rid of the curve” and tranforms the problem into one of timing, and one of a modulation scheme for a series of on or off states. In that respect I contend that it is an useful thought to consider transistors’ behavior (in an amp) as more linear and controlled if used in only the on or the off state. The output quality of Class-D amps in relation to price, size, complexity, and power, and cost suggests to me that the gains are not only in power efficiency.

Class D amps do leave the operation of transistors in their linear domain on the table, and do gain efficiency by that. Overall linearity as a control mechanism is something I stand by saying is increased. You may present an alternate take of course.

It's an interesting contradiction to build a more linear amplifier by intentionally using the transistors themselves in a non-linear fashion.

It is :) But it’s also only a contradiction on the surface of the words. The amplifier is quite linear even though the transistors aren’t used in what’s called their “linear region”, but I’d say they’re still used in a linear fashion.

I think you are confused. To get a linear amplifier, you operate your output transistors in a state of Saturation. This gives a linear transfer function to the drain based on the Gate to source voltage and you use a method of choice to hold the source constant. This is what a class A amplifier does. An AB amplifier has 2 transistors to extended the range of inputs on the gate for which the output is linear at the cost some distortion when switching between the A and B (hi and low side) transistors.

A class D amplifier uses a smaller amplifier internally (which can be a single transistor or more complicated) that is driven open loop so that it is either hard off or hard on but very fast and efficient. The hard on state is known as full compression because the output power will not change with input power. This is purely non-linear behavior and can be very easily verified by measuring the high IMD components it generates. Class D amplifiers employ significant filtering to suppress these non-linear terms and allow you to recover the signal. Mathematically, they violate the linearity condition of F(ax) = aF(x)

Also you should reread the article you learned from. It directly contradicts you.

It’s not about me; The “you”-ing is incredibly offputting.

We are simply and obviously applying the word “linear” a little differently. I am aware that it is a specific term for certain behavior / a certain operating mode of transistors. I’m referring to it in another wider context on purpose. As another example “linear” in “linear algebra” isn’t about saturation or gates. I’ve gone through this before in this thread and believe my perspective is now pretty complete and clear here and won’t engage further.

Actually the opposite is true. Transistors are approximately linear in half-open state, as in Class A, and non-linear at the extremes (when either fully open or fully closed).

Tangentially, the placebo effect is everywhere. If it can affect tasting (wine) and hearing (audiophile gear), I'm surprised not to have noticed its affect in discussing photography.

Will "computational photography" (as touted in the latest iPhones) that alters photos based on criterion of perceived quality end up bringing these kind of inane discussions there too?

Edit: Never mind, I didn't factor in all the arguments about Photoshop altering.

While there are endless debates about manipulation, when it comes to actual capturing of images, the photographic community (in my experience) has a healthy dose of "the best camera is the one you're carrying" and "a great photographer can take great photographs with any camera but even a great camera won't help a lousy photographer take great photographs" to go with its love of gear (who can help but love gear?)

Camera nerds are the worst. They're worse than audiophiles, if that's possible. Good photography requires nothing more than a bit of reason and observable standard practice, applied to aesthetics. You don't have to look hard to find someone with a $5000 camera/lens combo who doesn't know what the Rule of Thirds is or how to apply it.

Last weekend, I was in DC, and snapping pictures of the Washington Monument on my phone. My sister was shocked at how good they were. It's not hard, really.

Girl at work has 20k camera with 100k megapixels, but then again Huawei just asked to use her pics for their phone screen savers, so...

I made my living as a mastering engineer for two years...

To do good work in mixing and mastering, it's important to study the human perceptual mechanism and its propensity for illusion, and it's also important to understand masking effects and how to amplify certain components long enough to tweak them before dialing them back to recede into the gestalt.

It's also important to understand that in sound reproduction the vast majority of the distortion on the analog side comes from transducers (microphones and speakers), not amplifiers. (I'm not even going to bother with the article, because amplifiers aren't an important problem: buy something decently engineered and overpowered, and you're basically all set. Then worry about your speakers and your room, because they're the hard part.)

But it's likewise crucial to grok that the distortion introduced by digital components is disproportionately disturbing because it is enharmonic. And that practically speaking a lot of the damage that gets done in practice happens because people don't handle gain staging right and often don't understand the fundamental principle of preserving high resolution throughout production and only exporting to lower resolution as a final step.

In summary, there are a whole lot of interesting conversations to have on the subject of audio and how perception relates to production. But it's basically impossible to find a community to discuss these subjects with. Either it will be overrun by audiophiles and their pseudoscience, or well-meaning but undereducated skeptics will dominate the discussion with plausible but misguided debunking.

So every once in a while I throw up a piece like this into the middle of a giant discussion, with about as much hope as tossing a message in a bottle into the ocean.

This is pretty fascinating. I can't add to a discussion, but I would certainly read more about this if you have produced it somewhere.

I'm also curious, what made you leave your job as a mastering engineer?

The profession of audio production is extremely competitive because there is a perpetual oversupply of labor. The total amount of money in the music industry is also deceptively small compared to its high public profile.

I left the field (back in about 2003) because although I felt that I was good enough to compete and survive, I didn't love it enough to grind out a life as a workaholic and just get by.

Most of my writings from that time were forum posts with the perspective of the well-meaning but undereducated skeptic and wouldn't be that interesting to read today. Also, what was probably my best writing happened in forums which don't have public archives which have survived 15-20 years later. (Which amuses me as a former mastering engineer because one of our primary objectives is long-term archival.)

PS: not "enharmonic", "non-harmonic" dammit. The point is that while analog distortion is mostly harmonically related, digital distortion is not and therefore is much more poisonous in small quantities. Mixing especially (though mastering is similar) is all about making changes near or below the threshold of perception which sum to a cumulative perceptible effect.

I feel your pain. I produce and mix records - at an amateur level, but enough to get quite good at it. It makes me want to toss all this "objective", "measurable" bullshit from the skeptics right out the window - the subjective experience is all that matters. But it also makes me want to toss the audiophile voodoo out the same window. "Accurate" reproduction? I just want to show them how records are mixed.

My pet example is that I put high-pass filters (12db slope) at 250-300hz on almost everything that isn't a drum or a bass instrument. I'm slaughtering the bottom two octaves of guitars! And if you solo a guitar track, you can hear it. But in the context of a mix, all those bottom two octaves do is make mud and confusion, distracting from the clarity that the kick and bass (guitar || synth) needs. And I need that clarity, because I'm not mixing for $500,000 audiophile systems. I'm mixing for car stereos, boom boxes, Alexas, cheap earbuds, and worse.

So all that matters for me with audiophile systems - and I love audiophile systems - is that they sound good. That doesn't mean they sound accurate.

I agree that your high-pass filters are an application of masking principles. It would be challenging, especially for an untrained person but even for someone who is trained, to perceive enabling/disabling of those filters in the context of the full mix. Instead we solo to isolate the track being adjusted so the change is easily audible -- i.e. no longer masked by the other tracks -- and adjust the filtering anticipating the counter-intuitive but artistically desirable effect when the filtered track is blended back into the mix.

Despite that, your hostility to the skeptics illustrates that you and I still aren't really on the same page (though I don't doubt you produce compelling art, don't question your techniques, and agree 100% about ensuring that aesthetic gestures survive the transition to imperfect end-user environments).

To me, "objective accuracy" is a worthy and desirable intermediate goal for sound artists, a tool which can be leveraged for creative ends the same as a baker understanding food chemistry or an animator understanding Newtonian physics or a modern origami folder understanding programming. However, chasing "objective accuracy" means understanding the wildly non-linear, even bizarre human perceptual mechanism -- which is more of a pattern recognition machine and definitely not a measurement device -- in addition to all the strangeness of acoustics.

The problem with audiophiles is that their approach to "accuracy" is unscientific and doomed from the start. The skeptics on the other hand, have the right idea -- but those who have not done deep study often fail to appreciate just how difficult the problem is and how far off their intuitions are.

I've had some wonderful conversations with individuals whose perspectives are similar to mine, but it still seems impossible to find a hospitable public forum.

When I say "objective", it definitely needs air quotes, because the well-meaning skeptics love to toss it around for some decidedly non-objective opinions. My pet example is quoting "THD" as if it were a useful figure. It's not. The advantage of THD isn't that it is meaningful, but rather that it's easy to measure. Run a 1khz sine wave into the front of the amp, tap a resistive load on the back of the amp, measure the difference. Manufacturers like it because it makes for a competitive-sounding spec. But it has approximately zero to do with how an amplifier behaves in the face of complex musical signals, driving complex speaker loads.

When it comes to recording, my "objective" reality is the sound I hear in the room before it ever hits a microphone. But what comes out of the other end of the microphone is already deeply subjective. The objective ability of hi-fi systems to reproduce that subjective, colored microphone signal (plus whatever processing is involved downstream) is about as objective as they get. But the recording itself? Outside the world of purely electronic music, it's a subjective experience.

I play acoustic guitar every day, listen to unamplified singers/musicians (other than me) at least weekly, play electric guitar at least weekly, gig regularly. These direct, objective experiences of the natural sounds of instruments color my interpretation of both magical-thinking audiophiles and pseudo-scientific skeptics. I know intimately what the natural sound is, I know intimately what the recorded/reproduced sound is, and I know how the entire recording process works, from miking to mixing to mastering. Laws and sausages. The audiophiles and skeptics are both missing that perspective.

So, as a recording/mixing engineer, as a producer, I'm looking not to deliver audiophile accuracy, but rather to deliver the intended intellectual and emotional experience of the music. What feeling is the artist trying to convey? How do I manipulate the sonics to emphasize the musician's intent, as expressed through the lyrics, the composition/arrangement, and the natural tones of their instruments? That's the fun part, for me.

(As an aside, I'm currently working on a particular song of my own, destined for my long-delayed solo album. I wrote the song based on a nightmare I had in great distress. A few years back, I suffered an illness that left me nearly unable to speak, much less sing, and my voice is permanently damaged (I have surgery a few times a year on my vocal cords to keep speaking and not die of suffocation). I recorded the song right after writing it, just a couple of weeks before my first surgery, with a near-useless voice. "My broken voice won't fill the air / I know that you don't really care / You may not need my song but I need to sing it anyway". Fitting with the broken voice is the melody repeated on a piano that hasn't been tuned in over 50 years and has been flooded multiple times, so it's, um, colorful. To both the audiophile and objective mindset, neither the voice nor the piano were worth recording at all. But as an artist trying to communicate an emotional experience, they're vital.)

Whenever I see the term audiophile I mentally replace it with "over priced bullshit"

It's just someone being sold snake oil.

It's already here, for displays. I've seen lots of internet comments claiming I need the >450ppi screen on the new high-end iPhone, not the miserable old cheapo 326ppi one.

I think it might boil down to your philosophy on sound reproduction. Do you want accuracy with some potentials for ugliness if you clip, or do you want coloring and musical distortion? I don't think AM radio sounds good at all, but it is sometimes emotionally comforting to listen to. Fidelity isn't everything.

That said, many high end audio folks are straight up whack jobs.

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