Out of context of the rest of the article, this is total nonsense. While some in the audiophile scene like going analog, none of them debate against the truth that objectively, vinyl is far lower quality. Vinyl doesn't have nearly the resolution or nearly the dynamic range that a high-quality digital recording will have. Not to mention the higher noise floor. You lose a lot of the original audio with vinyl.

I dislike peddling it as somehow the objectively superior option. Or more "natural" or close to the original. That's nonsense.

Don't get me wrong, DACs aren't perfect. But with good recordings and mastering, a good DAC and overall system, it'll sound great.

If someone still likes vinyl knowing its faults, that's 100% fine. Nothing wrong with that, everyone in audio likes the sound a bit differently. Some people absolutely love the sound of vinyl and having a cool physical collection with gorgeous cover art. Power to them!

Nyquist theory proves it.

The Ogg people have a fantastic video explaining this. Technology Connections on YouTube has a video about it as well (the Ogg video explains things better, imo.)

https://xiph.org/video/vid2.shtml

It says nothing about noise, distortion, dynamic range. In these areas it is impossible to create a "perfect" DAC, although granted the best DACs are indistinguishable from perfect as far as human perception is concerned.

> If a function x(t) contains no frequencies higher than B hertz, it is completely determined by giving its ordinates at a series of points spaced 1/(2B) seconds apart

I took this to mean that it's any continuous function x(t), including amplitude information. I took a quick read through the proof and that's correct as far as I can tell.

Does that not mean that "noise, distortion, and dynamic range", as they are all encoded in the continuous function that is air pressure over time, can be perfectly captured and reproduced? All you need to do is throw out all information outside of human hearing range to have no frequencies higher than B hertz, and that's all you need for perfect reproduction.

If there exists a transformation f(x(t)), then said transformation can also be captured by the same sample, can it not?

Wires without resistance, capacitance and inductance. Resistors without capacitance and inductance. Capacitors without resistance and inductance. Inductors without capacitance or resistance.

While we're at it, semiconductors with perfect linearity and so on and so on..

I'm not going to argue that audiophiles generally achieve much of these, or even that it's especially important for the perceived audio quality.. But, perfectly recording and reproducing anything is still not possible, not in audio, not in video.

Which is an argument in favor of getting the signal into the digital domain as early as possible and maintaining it there for as long as possible.

The audiophile's stereotypical myth of analog supremacy is the assumption that the opposite is somehow true. Weird, but everybody needs a hobby.

There are costs associated with digital that, like with analog, must be managed.

Just because it says 24 bits on the datasheet of the used DAC IC doesn't mean your circuit will output a voltage that represents your input with 24 bits of precision. Designing and layouting PCBs with high precision DACs on them is certainly something where a lot mistakes can be made.

But you are right, if the PCB is done correctly, the signal exiting the DAC would indeed be an truthful representation of the digital data.

That being said: nowaday reaching enough precision for even the most critical listener should be not that hard/expensive.

you are right about bit depth. a jellybean 24-bit ADC has around 8 bits of noise floor, maybe more. To improve that, you sample at an insane rate (making jitter even less of a significant factor in the final audio) and average your measurements, and you can get 2-3 more bits out of that.

my overall point is that these problems you hear from people trying to sell you audio hardware are almost all not problems at all, as far as human ears and human perception are concerned.

This particular proof assumes that the clock is perfect, with no jitter or delay between one place it is used and another. There are more ways for a clock and uses of it to vary than you probably imagine possible.

It also assumes the A/D and D/A converters produce exactly correct results, which none do. We expect them to be close enough.

So in fact the output is as close to the same as the input as anybody cared to ensure, subject to cost, component tolerances, and amount of attention spared.

In practice, it is almost always as close to exact as anybody listening cares about (and better than a phono needle could manage). But there are ways for digital systems to produce bad results, by happenstance, laxness, or just normal aging.

Few ever check the calibration of their equipment.

audiophiles' continuous search for "perfection" is pointless because human ears are not perfect. not by a long shot. I don't care who you are.

just like how your eyes lie to you, so do your ears.

48kHz is not a difficult clock rate to maintain perfectly. at all.

all you need is a DAC of regular every day commodity quality, and an equalizer if you think "warmth" is a quality of vibrating air.

So, even leaving aside gross incompetence (is that smart?), anything that could be sufficiently perturbed thermodynamically, e.g. by age or decay, could throw off your results.

The great advantage of digital electronics is that they make most of the system relatively insensitive to commonly encountered thermodynamic effects, within limits. Most such effects that exceed limits make your thing just not work anymore. (We have all experienced this.) But some can have a more subtle result, some of those without even exceeding those limits.

Fortunately, most of those involve only a few components. Those components are mostly only in your power supply, your amplifier, your DAC, and the clock circuit driving the digital stuff, including the DAC. Of those, the ones that make a difference to sound quality are mostly in the power supply and amplifier, which usually makes sound obviously bad, and the clock, which can make sound subtly bad.

Fortunately, clocks involve very few components and those are not operated anywhere near physical limits, so they rarely go bad. Furthermore, most when they go a little bad don't affect the clock's output in any way that affects what it drives.

Unfortunately, when they do, the effect on the sound you get may be hard to describe beyond "not right". Then, fortunately, you can swap out the whole subsystem with the clock in it to see if that is at fault.

So the most usual danger from a dodgy clock or other source of subtle wrongness is that, as for the case of a needle dragged down a wiggling groove, you might get used to how that sounds, and want it all the time. And that can be hard to match in another system.

Perfect reproduction means error less than some threshold e.

“True” means has enough sampling frequency and accuracy to ensure the error threshold is less than e.

Proof by contradiction: if there's no threshold, then sound fidelity can in theory be infinitely precise. But nature does not work that way - at Planck length distance breaks down, at the width of a carbon atom the resolution of a vinyl surface ends, etc.

Nowadays there exists audio files recorded in up to 96 kHz with 24 bit samples.

So sharp low-pass filters are extremely difficult to implement as analog filters with reproducible characteristics.

Even when they are implemented as digital filters, as in most modern equipment, i.e. the audio ADC uses a much higher sampling frequency and its output signal is interpolated and decimated digitally to 44.1 kHz, it is still difficult to design the very sharp low-pass filter so that it will not introduce any audible artifacts on transient signals (the sampling theorem is based on ideal non-realizable filters).

Raising the sampling frequency to be distant from the 40 kHz minimum value removes all difficulties with the design of the low-pass filters making it easy to guarantee that the digital signal reproduces exactly the analog input.

When the audio CD format has been designed, the digital technology was much less advanced and storing a lot of bits was a far more difficult task than later.

So they have made the trade-off of requiring very expensive analog low-pass filters in order to minimize the amount of bits stored on the disc and transmitted over the digital interfaces, because the digital processing and storage was even more expensive anyway.

Nowadays, it is much more convenient to just use higher sampling frequencies for audio signals.

You will however need to accept delay from the lowpass filter, which is fine for music reproduction.

So while these effects are measurable with very expensive equipment, they are in all metrics orders of magnitude better than any analog equipment.

24/192 is becoming common, and you can even find 384 kHz files for sale (transcoded from DSD recordings.)

Only if your signal is infinitely repeating in time, with no start or end. Otherwise, it's spectrum is not band limited.

Is your ADC input such a signal?

EDIT: This also applies to sigma-delta sampling. Although there you can often get away with a simpler single-pole input filter.

Not only that, your DAC needs to have infinite lookahead to reproduce the same input that was passed to ADC.

It doesn't matter.

What matters is that with a digital signal processing chain you essentially have a knob you can turn to approach mathematical perfection as closely as you wish, at least until you reach the point where the inherent limitations of your analog components begin to matter. And that point is far, far beyond where human hearing can tell the difference.

But with analog signal processing you reach the limitations of analog componentry much quicker, while you are still well within the noticeable range of some humans to tell the difference. This is because the entire chain is analog and thus no part of the chain is immune to information loss or added noise. With a digital chain, only the ADC and DAC stages require high quality analog componentry. Everything in between is pure math.

Neither process is capable of overall mathematical perfection, but digital can approach it much more closely than analog.

I just seem to take issue when Nyquist is getting inwoked. That is a purely mathematical result. It is sufficient (and I think required, too) to plug source into ADC+DAC, measure the output and show that the difference is insignificant. And much less than, say, with tape.

Honestly, it’s 1% “I like music” and 99% snake oil salesmen taking advantage of fragile egos.

"Somebody was trying to tell me that CDs are better than vinyl because they don't have any surface noise. I said, 'Listen, mate, life has surface noise”

You conveniently forget history. Audiophiles embracing digital media is a relatively recent thing. It was vinyl and everything analog until about 10-15 years ago when it became <any combination of tech. nonsense>

When the audio CD's appeared, their main claimed advantage was not that they offer a superior sound over vinyl (early units had actually modest characteristics, with 14-bit DACs, modest low-pass filters, and possibly also low accuracy and high jitter of the sampling frequency), but that you can play the digital disc as often as you want without quality degradation.

Before digital audio, real audiophiles also had a tape recorder, and as soon as they bought a new vinyl disc, they played it only once, to transfer it to a magnetic tape, and then they stored the disc safely and they played only the tape, for fear of degrading the vinyl disc when listening to it.

At that time, most audiophiles believed that vinyl discs that have been played more than 3 times are no longer suitable as a master source from which to make a copy on a tape, as they were already too degraded in comparison with a new vinyl disc.

Reel to reel usually ran the tape almost 4 times faster with over 3x wider tape vs mainstream compact cassette, so it was very high quality.

One interesting fact I've come across is that e.g. when Bob Dylan's "electric trilogy" first came out on CD, they apparently had to do new remixes of Bringing It All Back Home and Blonde On Blonde not because they just felt like it, like they so often seem to do these days for artistic and/or sales-tactical reasons, but simply because by that time the original master tapes (including the safety copy) had worn out somewhat through continuous re-pressings of the original vinyl albums.

That's the issue here. The loudness war made any CD mastering a race to make the most compressed track, just to sound louder than the competition.

Technically, digital should sound better. The technology is better. Sound engineers are making sure this is not the case.

Vinyl forced better mastering.

That's pretty offensive and ignorant. The loudness wars ended at least 10 years ago.

Not to mention that setting up a tonearm just right appears to be a feat of wizardry, or being sure that the recording was properly mastered for vinyl.

Hard to imagine a better A/A-test.

In fact, part of the record player spec is the "RIAA equalization"[0] that attempts to account for the fact that vinyl isn't good at recording a wide range of frequencies well. On recording, you cut the bass and jack up the treble. On playback, you boost the bass and reduce the treble. Without that, vinyl would be even worse than it is today.

It's absolutely, 100% fine for someone to say that they subjectively prefer the sound quality of a record player to a CD. I don't, but that's fine: it's purely a preference thing. But to claim that vinyl is objectively better in some way is just ludicrous.

[0] https://en.wikipedia.org/wiki/RIAA_equalization

"oh what kind of turntable do you have? oh that's not good enough, there's your problem right there."

"what speakers do you have? what about your phono + power amps? oh that's not good enough, there's your problem right there."

"how did you place your speakers? oh, you should have them at 45 degrees, not 60."

"is it a re-pressing? from whom? oh, there's your problem right there."

The end result is a bunch of nerds ruining their credit scores because someone told them they need gold-plated hydraulic bearings in their tonearm to properly listen to their mom's copy of Led Zeppelin IV.

Does it sound different? Yes. Does it sound better? If you're really trying to get the most out of the music itself, no. Warm ⊄ good.

Vinyl is great because you aren't just listening to music. The medium demands something from you. It's tactile. The album artwork is a complete ensemble, rather than a little icon on your phone. In my case, I had to put in effort to restore the equipment I play my records on, so there's a bit of pride in it. It's a more intimate experience.

The next month he reviewed a different player. He basically retracted his opinion of CDs but said the player from the previous month was obviously terrible, at least the particular one that he did his review on, if not the entire brand.

And for audiophiles themselves, I notice many of them seem to have very poor taste in music. Like do you really need $50,000 of equipment and sand weighted speakers that weigh 700 pounds each to play 1968 bachelor pad music?

He could have been right both times. Many early CDs were just plain awful. Same with the first couple of generations of CD players, even some pricy ones. The early development of MP3 and other lossy audio codecs recapitulated the history of digital audio pretty well: lots of truly crappy recordings got passed around.

Like do you really need $50,000 of equipment and sand weighted speakers that weigh 700 pounds each to play 1968 bachelor pad music?

I've never understood audiophiles. How can it make sense for me to spend more money on my equipment than the freaking recording studio spent on theirs? Do these people think studios use $10,000 power cords on their sound boards, and atomic clocks to drive their ADCs?

Great response and critique until that point. I'm old (52) enough to have seen quite a few media changes too. I recall that when you were off your tits at a party, and discussing music, you tried to find common ground. That could lead to some impressive contortions!

Can't say I ever spent much on playback hardware. I generally went second hand for speakers/amps etc. Once the Walkman was invented and then rather a lot of Chinese (and other copies were available) clones appeared then you could ruin your hearing close up with seriously decent quality sound on the move. I was still walking around with a mini cassette player with wired ear plugs in my jacket's breast pocket until around 1993ish.

I did lay out something like £200 on a pair of decent headphones for use at home. They make anything from O Fortuna to Ghost Town via say Schehehererherereherezade, Finlandia and One sound rather ... Special.

It is different than someone who appreciates music, which it seems like you do.

It's so fascinating to me that an inert physical disc is engraved with sound waves that are reproduced by dragging something along to return them to sound, in the case of the 78s without the need for anything electrical in between. It's an added bonus that it has zero opportunity to inspire screen time for my kid, who absolutely doesn't need to stare at a screen while listening to music.

For fully mechanically recorded records you have to go back to pre-1927 Edison Diamond Disks. Or the real deal, Edison cylinders.

Edison cylinders are actually rather good recordings. The mechanical playback system. with no amplifier, introduced most of the distortion. Modern playbacks, using a modern record pickup with a suitably large-sized stylus, are quite good. Here are some from cylinders made around 1900.[1] Those include Sousa's band playing Sousa's marches, led by Sousa himself.

The ultimate playback device is IRENE. This was built for the Library of Congress. It does a full 3D scan of the entire disk or cylinder surface, and a detailed 3D model of the surface is created. That's turned into audio by a program that simulates a stylus tracking through the grooves. IRENE is used on very old, fragile items. It's even possible to put broken shards back together in 3D and recover the audio.

For real authenticity, there are reproducing pianos. These are player pianos, but with a better recording system, one capable of recording key pressure while a live pianist played the piano. While the Duo-Art system had binary-coded pressure values, the Welte-Mignon system was analog. Mostly. Here's a Welte-Mignon roll being played.[2] The holes at the outer edges of the roll are encoding the key pressure. The ones in the middle indicate which keys. MIDI, from over a century ago.

If you're going to go retro, go all the way. If you just want to listen to music, it doesn't get any better than 24 bit audio. (16 bits may not be enough for really soft passages, where all the high bits are zero and you may be down to 6 bit audio.)

[1] https://www.nps.gov/edis/learn/photosmultimedia/recordings.h...

[2] https://www.youtube.com/watch?v=U8K5Yh0aLhA

My grandpa used to tell me that they donated the clockwork out of their record player as scrap metal for the war effort, so they listened to music by spinning the records by hand and holding a fingernail in the groove. That's something you can't do with pretty much any other recorded medium, aside from print.

Loading a record is quite the event: slip out of the card sleeve, then slip the dic out of the waxed paper sleeve and pop it on the spindle without scratching it. There is of course the correct way to handle them that if you are around or older than my age (52), you already know - the delicate fingers on the rim and your thumb on the centre.

I can remember my Dad going to the "biggest NAAFI in the world" (Rheindahlen) and coming back with a brand new record player in around 1978 or 79. This thing could handle something like five LPs at once and play them sequentially.

Pissing around with Spotify isn't quite the same and I do understand why people enjoy the theatrics but to be honest: I turned my CDs into FLACs and ditched the tapes a long time ago. I still prefer to buy my music by the CDROM and rip it to FLAC or mp3 or whatever - now that is me showing my age!

It’s very convenient especially now that lossless is an option.

I hope we get plenty of warning before they axe it. Imagine that being a google service?!

Sometimes especially on Zoom I look at my cabling and think it's amazing that images of other people and their voices are going through that cable.

Only on Zoom and similar video conferencing. For all other uses I don't even think about it.

https://www.imdb.com/title/tt0110622/

Hilarious and well put! Speaking as an erstwhile musician, this has been my theory about Guitar Center's business model since the 90s.

Nowadays I can get excellent audio quality watching good Youtube videos or streaming from my preferred music site (I only listen through headphones now).

I'm super glad I was able to get to the quality level I liked for fairly cheap.

For the record, I like your approach to music enjoyment.

The other thing I like is that it really takes more effort to put on a record than start a track on Spotify. I really only do it when I can take the time to do so. Pick a record from the shelf, carefully take it out of the sleeve, carefully put it down on the turntable, and then really listen to a full record while sitting or lying on the couch doing nothing else. Again nothing you couldn't do with Spotify or YouTube music, but somehow you (well, at least I) never do.

Vinyl _does_ also provide a nice soft upper limit to the loudness wars.

I like not having to flip the record halfway through, or being able to listen to "The Wall" without 3 flips.

That's a great way of putting it! And something I hadn't considered before I bought a record player and started listening to it. There's that point in time where you realize the A-side is done, you get up to go and flip it, and you have a half-minute or so of reflection on the first half. Then the B-side starts and it's like the second act of a play is starting after intermission.

But some vinyls actually come from a digital master anyway.

I have a ton of vinyl records myself, never understood why anyone would say they sound better. Vinyl is a compressed format, even if the compression is analog. There is significant loss of signal in the whole signal chain. You need to do reverse RIAA (compression), then you fabricate a master, then you press vinyl from this master which is only good for a few thousand presses, then you have the pickup and at last the RIAA preamp (decompression)...

> But a few specialty houses (...) have long advocated for the warmth of analog. “Not that you can’t make good records with digital, but it just isn’t as natural as when you use the original tape,” says Bernie Grundman

which is absurd. "Warmth" is a kind of distortion, and if a signal is distorted then it's less accurate or "natural"!

They can't claim "fidelity" and "warmth" at the same time. It's one or the other.

So "fidelity" means "fidelity to the way I remember" (and like).

Just like people buying yellow lightbulbs because that's how incandescent bulbs used to always be -- actual natural sunlight isn't that yellow. Good LEDs are more "natural", but less "familiar" to people.

Only if it’s pushed into distortion.

Guitar players (generally) like tube Amos because of the way they distort compared to transistor amps.

If you are not pushing the amps into distortion, they will only sound different if they’re badly designed.

Ultimately, this boils down to signal which vibrates speakers. As a thought experiment, if you use some kind of theoretically perfect surface-sampling laser to capture every movement of the speaker surface at sufficient frequency and fidelity to reproduce all of the information in the original signal and speaker surface vibrations (ala Shannon->Nyquist), then a digital playback of that signal which vibrates the same speaker surface identically will sound exactly the same.

Individuals can prefer different sonic characteristics encoded in an output but that's an aesthetic choice. The entire signal chain creating that output is the result of creative and technical choices. It goes from guitar string to studio acoustics to microphone to mixing board to outboard processing gear to recording medium to duplication to distribution to playback to amplifier to speakers to room acoustics to human ear. Most of those elements significantly color the sound. Yes, mistakes can be made in the downstream signal chain which diverge from the creative intent. However, those mistakes are exceptions, and not inevitable. Done correctly, there's no reason a digital step in the chain shouldn't be completely undetectable.

I find that listening to one of three different sources (earbuds, gaming headset which is EQ'd via open source software, and inexpensive-but-fancy headphones) causes me to adjust to that particular set of headphones' sound. It seems to be like the equivalent of our brain continuously auto-white-balancing our vision.

Physics doesn't care about people's beliefs, but sadly many people also don't care about physics.

In practice, if I give you a random analog box, transforming it into an accurate DSP algorithm it's an extremely difficult problem.

Which is why people instead of doing that, they recreate the "idea" (reverb, delay, ...) digitally, but it's not a copy.

The popular Kelvin Tone Shaper https://www.toneprojects.com/kelvin-tone-shaper.html

Maybe you want tape? London Acoustics Tapei https://www.londonacoustics.com/product/taipei-studio-tape-r...

As for freeware, the comeback of Variety of Sound with https://varietyofsound.wordpress.com/2021/11/30/tesslase-mki... or the less subtle https://kitplugins.com/pages/burier-free

This is just a part of (good) saturation plugins from 2021...

This says nothing about the difference between analog and digital recording media unless you think that disc lovers shun solid state electronics or something like that. Do they?

There wasn't much digital sound around when McLuhan said that "The medium is the message." I think that's the substance of this whole brouhaha. What's wrong with rejecting digital media when one deplores the inundating tide of changes in society's tempo, cultural content, aesthetics, and forms of dis-function that have been carried so far and wide by digital media? Isn't change always impugned?

Though I would expect it'd be more of an eq profile thing.

https://www.analogueseduction.net/record-clamps/SHUNMOOKprod...

You cannot, it is impossible.

Even with an unrealistically low sample rate (say 500Hz instead of 44.1kH), this will not sound like 500 beats per second as the "steps" occur, but rather will just sound extraordinarily dull, as the Nyquist frequency has dropped to 1kHz, a fairly low tone, and all info above that has been lowpassed away.

When the signal is changed to analog by the DAC in order to come out of a speaker, the "steps" become smooth curves like any analog waveform. You cannot hear "digital steps" because they cannot be audible - sound is vibration, and it cannot be made into steps in the analog realm.

what is bassy reverb?

I know this was a typo, but tools like https://shop.audified.com/products/mixchecker (with a literal "sedan" button) and articles like https://ask.audio/articles/the-final-mix-using-your-car are sadly indicative of the emphasis in modern mastering on sounding good on car speakers - which is the entire problem in a nutshell.

modern mastering does take into account a highly popular listening environment, to ensure that the mix sounds good there as well.

the second article you posted is talking about using a car as one of several environments in which to test a mix. because it’s a “known” reference in a real-world environment (your ears are used to listening to mastered music on a car stereo).

I love listening to vinyl. But I’m not gonna lie to myself and say “it sounds better”. The engineer/scientist in me knows it doesn’t, and the pragmatist/experimentalist in me knows that even if I can reliably detect differences between a good digital recording and a good vinyl copy of a song (which I’m not convinced I really can) there is nit nearly enough difference ever to say one is better than the other.

I like the ceremony of playing vinyl. It adds to my enjoyment, but in the kind of way that a Japanese tea ceremony “improves the taste” of tea.

Also, a long time back I realised that listening to hifi _gear_ leads only to criticism and unhappiness, while listening to great music leads to joy. So I mostly have given up critically listening to “gear”. A great song or album can be a thing of joy coming out of your laptop speakers. Coltrane or Ellen Allien or Polyphia are no less amazingly talented in shitty phone earbuds compared to my thousands of dollars of living room hifi. I’m too old to want to argue about “clarity of soundstage” or “time alignment in the tweeter crossover”, I just want to rock out or dance or groove along to great tunes. I can do that with a 100% analog signal path in my lounge room, or in cheap Bluetooth earbuds inside my motorcycle helmet with a 97db noise floor…

Anyone with basic understanding of A/D conversion and sampling theory will understand that you can always exceed analog with digital, given enough bits (and 2*16 @ 44.1 kHz is already pretty darn good).

In the early days of MPEG Audio, Level 2 (not mp3) I spent a lot of time listening to samples and I was taken aback when I finally traced flaws back to the CD itself (Cranberries, Zombie, 1994). Many of the early CDs really weren't that great, but that wasn't due to being "digital", just a sloppy production.

Other analog broadcast and recording media have similar curves, called pre- and de-emphasis.

Digital doesn't need it because it has effectively zero distortion, and dynamic range to spare.

An advantage of vinyl is that there's a physical deliverable, making it possible to monetize it. I think that's why a lot of indie bands use vinyl.

I go to a lot of "indie" performances, and have the musicians directly: How do I buy your stuff, so you make the most money? The answer is always the same: Buy the CD right here.

Reminds me of the LP my band teacher in high school had. It was the 1812 overture, and the groves for the cannon shots at the end were easily distinguishable with just a casual examination. Based on what you're saying, I suspect that recording was pushing the limits of vinyl.

With the satisfying full warm tone vinyl is known for.

I was a drum and bass dj when it was all still vinyl. Even in a club with huge speakers, I've never seen the needle jump due to the bass (although some places used to tape a penny to the top of the headshelf). The weighting on the tone arms can be adjusted so if you've got a needle dancing around it's due to your set up, not the bass, or Jungle music wouldn't have lasted too long.

I've tried Sound Source and not only do the eq's work, but I can boost the volume so I'm listening at 6/10, where before I was at 9/10 and had no headroom.

I knew I'd get an answer by saying something wrong on the internet.

I was jailbroken on iOS 14 for the longest time solely for a system-wide equalizer for Apple Music. With a new phone it's a major step back, and I've switched to Spotify for their (worse) equalizer and somewhat worse overall experience (Spotify doesn't even have an equalizer on desktop!)

It's really unfortunate that such a straightforward feature is so limited on Apple platforms, and for no discernible reason.

Settings -> Accessibility -> Audio/Visual -> Headphone accommodations

I also have the latest AirPod Pros. With Apple Music, they are not too bad, but Apple is preventing me from having enough volume. I switched to Tidal - the sound is better, and their overall audio out level is higher than Apple. (Tidal also has notes on the music and the bands like the liner notes in the old days.) I tried Idagio, but their output level is even lower than Apple's Music. Basically, I couldn't hear the music well enough to bother with them.

If I play music on my sound system at a comfortable level, and ask people, "Is it too loud for you?", they say, no, it's fine.

The other problem I have with the AirPod Pros is their transparent mode doesn't provide any gain or much EQ for external sounds (even with the audiogram), so I can't hear people nearly as well as I can with my cheap hearing aids. I still have to remove the AirPods and put my hearing aids on, if I want to talk with people in anything but a silent room.

What I need more than anything else is compression. I can hear loud sounds fine - in fact, they bother me. What I can't hear any more is quiet sounds. That's what I need amplified, along with treble boost to compensate for my aging ears.

Apple is really missing a market here for OTC hearing assist. They have all the tech needed, even already built into existing hardware - AirPod Pros are an engineering marvel. They are holding back for some reason, probably some contractural arrangement we're not aware of.

Sounds like if you combine the two steps you end up with the identity operator :)

  signal -> [HF boost] -> [needle adds HF noise] -> [HF cut] -> same signal, less HF noise

Dolby B for cassette tapes is the same idea (but only for the highs).

[0]: you can get very low tolerance resistors and capacitors, but they’ll cost you

> It's absolutely not possible to accurately record every digital recording onto vinyl.

Outlining an idempotent process of recording and playback does not help there.

You can simulate this pretty well with computer speakers. Turn down your audio outputs so that it's barely audible, then turn up the speakers as load as they'll go. Noisy, isn't it? In theory that should be the same as turning the outputs to their maximum clean level and turning your speakers way down, but in practice it's absolutely not. Well, every vinyl record made does exactly that to the bass.

There's a lot of weirdos out there, but these ones aren't hurting anybody.

It's not intellectually fine, only fine in the sense that it's well within the bounds of free speech.

Where you're getting pushback is that this thread is talking about vinyl records and tube amps, things where the noise and distortions introduced by their operation are well known, well documented, audible to anyone with normally functional ears, and regularly utilized artistically. These are not the inventions of grifters with a product to sell, it's just an older technology that is less precise in a certain predictable way which some people find pleasant.

The poster I'm replying to is then acting like it's some weird audiophile belief that records sound different from other media.

On the topic of the overall article, of course the people who believe a digital processing step makes things inherently worse are absolutely off their rockers.

“Not without a double-blind test you don't!”

This is what audio purists should be asking for. Let's pay the creative production team to create and release a high-fidelity digital mix optimized for us. No concerns about radio play, boomboxes or loudness wars. Just their original creative intent.

If you say that waving your hand over the shriracha while murmuring a magic incantation makes it taste better, then that may not be so intellectually fine any more.

Most audiophile mania is poorly informed at best and astrology at worst.

http://abx.digitalfeed.net/

I'm not sure what you mean by "properly transcoded," as though there's a scourge of bad encoders out there and everyone has nutty encoding practices, but I really, really bet you can. Listen to an mp3 of a rock/pop track. Try to focus on the cymbals. See if you can isolate them aurally from the rest of the music. Can you hear how mp3 encoding completely munges high frequencies to sound like digital glass breaking? Then you shall finally understand why mp3 encoding has always sucked, and Napster really, really should have won.

While that's on the very top-end of human hearing - most by middle age can't detect anything above 14-16khz, but it's certainly possible (especially for those younger) to have frequencies all the way up to 20khz as audible.

So it is possible for some people to hear a difference with some mp3 encoders, no matter the bitrate or quality settings.

And the psychoacoustic model is going to focus on the "average" listener, so it's perfectly possible that an lower bitrate encoding is transparent (IE: Completely indistinguishable) to one person, while another may be able to notice higher frequencies that have been cut off to provide more quality in the more noticeable parts of the spectrum. Or even the same person at different ages. This is completely physical difference, and no amount of training or harder listening would be able to bridge the gap.

And I find many of the higher frequencies that are hit by such things are often not particularly noticeable unless you're actively looking for them, being able to tell there's a difference and knowing what to specifically look far isn't the same as saying the recording is somehow less enjoyable due to quality differences.

I wouldn't call it "night and day", but I had no trouble distinguishing even 320 MP3 from FLAC on his system by focusing on the very high frequencies (3/3, small N I know I know, but I felt like I could've gone on indefinitely). The MP3s lost some clarity in the highs which led to less dimensionality/sense of space ("soundstage"), because our spatial hearing is very attuned to minute transient differences in high frequencies in particular. But you need to be listening on a system that is adequately equipped to reproduce very high frequency transients accurately, which most people don't have access to.

Do you have a favorite rock track we can do an ABX test with?

If I’m not mistaken one of those A/B/X test websites also tested that, and while I was using a very simple pair of headphones, the difference was noticeable.

https://abx.funkybits.fr/test/the-eagles-hell-freezes-over-h...

(obviously there's a bug on the tracks display after round 1)

Never heard this, but I know the original track is used for reference in studios from LA to NY, and is generally regarded as among the best mixes ever engineered.

Another commenter suggested high end audio equipment and $10K monitors... not necessary. With any off the shelf hifi, $100 worth of equipment, one can easily hear mp3 high frequency problems. Focusing on the cymbals just makes it obvious.

I have a couple systems, one is a proper hi-fi system. If the album is mastered without brick-wall normalization, the difference is easier to hear.

When I listen the same album via my DAC, from MP4, I enjoy it. When I listen the same album, from a CD, I sit in front of the system like a rabbit blinded by lights.

The DAC is the same. It’s a Yamaha CD-S300 with a proper iPod interface, which carries the signal digitally till DAC.

This ABX test is the same. You need a high fidelity chain to enjoy it. I have a nice sound card, but the speakers connected to it can’t handle the resolution put out by it.

I'm not a fan of analog, but a high-end analog system will sound far different than a similarly priced digital one. It's a very different type of sound.

Whether someone likes it or not is personal preference. But it does sound night-and-day different to listen to a vinyl from a turntable and using a DAC.

As for MP3 vs lossless, that's mostly true assuming a modern variable bit rate MP3 on OK hardware. The better the audio, the better the equipment, the more noticeable it gets. But yes, for 95%, it's basically not significant.

You're going to get a pretty terrible experience with MP3 128kbps, whereas modern variable bit rate schemes can actually be quite decent. Personally I prefer to always go lossless, but for most people on most systems, the difference isn't noticeable. (Moreso the better the system gets, ofc)

To some extent certainly, though I think that e.g. the handling of short sharp transient remains somewhat of a fundamental problem of MP3 no matter how much bitrate and encoding brain power you throw at it, and was only really improved upon in the subsequent generation of codecs, i.e. like AAC, Vorbis or Opus.

But the main reason is that it's so ubiquitous. If you're buying music, it's almost always MP3 unless you go to a source like Qobuz which serves FLAC. Same with devices - basically everything supports MP3. Vorbis and Opus support is far patchier.

Apple is another odd one out there by only selling music as AAC, and Bandcamp gives you a comprehensive choice of all sorts of formats.

> Same with devices - basically everything supports MP3. Vorbis and Opus support is far patchier.

True that, though AAC support might also be somewhat reasonably common-ish these days. (Though I've found that apart from also having a relatively low folder count limit in general, my car radio also has some difficulties with reading the metadata in some files, and as far as I can tell I think the affected files are all MPEG-4 files. It's only some MPEG-4 files that are affected, though, but I haven't yet bothered to find out what exactly else they might have in common.)

Oh? Why's that?

The main reasons why not would be streaming (currently Qobuz is the only one I know of that does it properly, and services like Roon + Qobuz can get pricey) and bluetooth (which doesn't support lossless at all).

Don't worry, the crisis will not last long. Audiophiles will find a way to get over it and buy some better cables that would have revealed they were using a Digital source all along.

Hence the cassette revival for those of us of a certain vintage.

Compared to CDs? Definitely not! Not to mention how many people listened to homemade mixtapes and copies, which made the audio quality even worse.

Did they sound good enough that you could dance, relax, have a good time, be consoled when you were unhappy, or whatever people did with music? Absolutely.

Even when CDs did eventually come out, I far preferred a WalkMan to a Discman. I could put a Walkman in one pocket and a cassette in the other. That wasn't easy with CDs.

Cassettes we’re easy to remix, record, share.

The physical mixtape was a cultural touchstone of romance and friendship for a while there.

So yeah, I guess ‘good enough’.

I had completely forgotten about the development of anti-skip. And now that you've made me remember it, I suddenly feel old.

I knew somebody (with very rich parents) who basically had a shoulder bag (read: man purse) so he could carry around his Discman (early Sony version, no anti-skip), an extra CD case or two, and an extra set of batteries. It was awful.

Superstition is always bad.

(https://thevarsity.ca/2022/03/20/athletes-superstition/)

But the "superstitions" that the linked article mentions, are rituals, lucky charms. They're much less dangerous than going around pretending that something exists, that provably isn't there.

Audiophiles are moon-landing deniers. Some might think it's funny or harmless. I think it's terrifying.

Sorry, but I do not follow how you came to this conclusion, could you explain it?

> They're much less dangerous than going around pretending that something exists, that provably isn't there.

By your definition is every person in the world who believes in god therefore dangerous?

In part it goes to prove Mark Twain’s quote that it’s easier to fool people that to convince them they have been fooled. So many people build their identity around being something like an audiophile and something like this demonstrates the are mostly frauds…not all of it is a bad-faith fraud either which makes it so much more difficult for those people who believed they had a genuine passion and talent to accept.

On the other hand there are people who really don’t give a shit and just shill themselves in a given industry to make money as self-proclaimed experts. It’s part of a sickness social media pushes on people where they think they are brands…as opposed to actual humans. These people have no allegiance and will readily move on the the next money making opportunity so long as they can…probably high priced wines.

Standard CDs might not be good enough for archival and further production work (where you also need wiggle room for further processing or format changes), but they have plenty of headroom to reproduce any kind of sound for playback purposes. Any more is just placebo.

With 96khz sampling rate, you don't need as a sharp drop off in your filter.

The above is mostly accurate, hopefully someone else will comment and tell me how I'm wrong and correct the inaccuracies.

Edit: to anyone reading this, read the replies if you want the correct explanation. To everyone that replied, thank you :).

A sharply cutting off analog filter is expensive to produce. It has multiple stages to create the multiple poles. High precision resistors and capacitors have to be used to get all those circuit stages to line up. The filter will have phase distortion.

That's the basis of "supersampling": sampling at a higher rate with a simpler filter with less of a cutoff, then completing the job with a digital filter to get to the target sample rate.

This can be done in reverse, in reproducion. Take, say, a 48 KHz signal, and digitally interpolate it to a higher sample rate like 96 KHz. That is fed to the DAC. The filter after the DAC then doesn't need such a steep cutoff after 20 KHz. A greater bit depth can be used; like 16 bit samples interpolated to 24 bit at a higher rate, fed to a 24 bit DAC.

The digital filter or interpolator doesn't care about accurate resistors, capacitors or drift in component values over time or due to heat; it does the same thing with the same data every time.

Digital filters can look at future values also. The state of an analog filter is determined by only the current and past values of the signal; but digital signal processing can delay the signal a little bit and look at a "box" around the current value. I think that is key to preserving phase relationships.

Monty (of xiph) has an article on the subject (well on 24/192 but it applies all the same) which goes into the gory details: https://people.xiph.org/~xiphmont/demo/neil-young.html

There's a space in the audio world for more bits of data. But the final output format isn't where it belongs.

If you're sampling at 48 kHz, the signal has to be severely attenuated already; it has to go from 20 kHz to deep cutoff in just the space of a few kHz.

At 96 kHz can achieve the effect as if you were sampling at 48 kHz, with a steep filter. You sample at 96 kHz with a milder filter, and then purely in the digital realm, you down-sample to 48 kHz. There is an overall filter consisting of the original analog one plus the digital processing.

That is cheaper and more reliable than doing it all in analog.

An analog filter with a steep cut off will be challenging in mass production because of the strict component tolerances.

Sure, you could use a steep filter with 96 kHz also. Say, a steep filter that starts cutting off at 30 kHz. It would still be a less demanding filtering application because of the margin that you have in the frequency domain. The multiple poles of the filter don't have to be lined up as well. E.g. if the first pole starts rolling off at around 30 kHz, and then next ones at 31, and the third one at 28, ... it doesn't matter because you're still hitting the absolute target of there being next to nothing at 48 kHz, and nearly the full signal at 20 kHz.

I believe the comment you're responding to is talking about the analog filter that is needed to avoid aliasing -- as the first words of your comment correctly note/explain.

And in particular, the original comment seems to be noting the phase distortion (in frequencies near the cutoff) that analog brick-wall filters will cause. This has been a design contention for decades, really, ever since the CD format was introduced.

It's a big design space, with options for gentler analog filters, followed by very fast digital sampling, and further tricks with filtering in the digital domain, where you don't have to worry about getting great capacitors, etc.

It may be out-of-scope to lay all that out in one paragraph!

You're right that its a big design space. The key takeaway is that "yes, higher sample rates can actually make a difference, but almost entirely down to the filter design, not because Nyquist moves ... and you probably cannot hear the difference."

Its vwry “spherical cow”. It assumes all your filters, DACs, are perfect, and that time is infinite.

Yes, your filters, DAC and speaker wire could still create a less than ideal listening situation, but the fundamental aspect of the Nyquist frequency is not concerned with any of that.

And you can use oversampling to correct for the filtering issues. That's pretty standard in modern DACs and ADCs.

It's enough to reproduce well above what we can hear by the normal mechanism that we use for most of our hearing (sound vibrates the eardrum, which vibrates some tiny bones in the middle ear, which cause fluid in the cochlea to ripple which disturb sensory hairs which we perceive as sound).

Ultrasonic sounds can be conducted to the cochlea via the surrounding bone bypassing the eardrum and middle ear. Those bone conducted ultrasonics can be perceived. See [1].

This is probably not important in music except in maybe a few rare cases.

I'd still like to see better frequency handling though, so that our music wouldn't sound terrible to the dogs and cats that are forced to listen with their owners. We can't normally hear much past 20 KHz (more like 12 KHz for many of us), but they can.

There are several musical instruments that have significant ultrasonic output. It would be an interesting experiment to see if dogs and cats like those instruments live better than they do recorded (or dislike live less than they dislike recorded). Then compare to the same experiment but with instruments that don't have ultrasonic output.

[1] https://pubmed.ncbi.nlm.nih.gov/23384569/

The content can be faked, and this is used in low-bandwidth codecs and such.

https://en.wikipedia.org/wiki/Bandwidth_extension

This is a very simplistic view of things, on many fronts, but I'll try to be as concise as possible.

In a classical wind/bow orchestra, there are more harmonics at play than the 20Hz-20KHz band, and these harmonics affects our perception of the sound and soundstage. However, recreation of this is very hard, because you need both recording and playback chains which can handle these harmonics as well.

To faithfully reproduce such orchestra, you need a speaker for every instrument, ideally with the exact air movement capacity of the instrument you're mirroring. This is not practical. Instead we mic all of them, mix all of them, and add room echo to the mix to capture harmonics as best as we can.

However, this can't replicate some instruments anywhere around its real sound.

Examples are Turkish qanun [0] and Chinese guzheng [1]. These instruments sound bland, flat and shallow on all recordings, but listening them live, directly with your ears is a goose pimlples inducing experience. Higher end stringed instruments have a similar vibe to them.

So, a 16bit/44KHz signal at 20Hz-20KHz band cannot reproduce these instruments with any faith.

Standard CDs, high quality vinyls, lossless files carry a lot of information, but not all information we can process with our ears. Like how even the best digital cameras cannot reproduce the colors we can see with our eyes.

Reaching these resolution levels are neither cheap, nor practical, hence we use what's practical.

As a result, your master-pressed-vinyl possibly can't carry this information either, because the recording chain was not able to capture that amount of information, even if you went all the way to install your own power pole to feed clean power to your impossibly expensive audio equipment with all that capacity to reproduce that sound.

IOW, you can't extract the sound which is not there to begin with.

Your friendly ex-orchestra player reported from its AKAI-AM2850.

[0]: https://en.wikipedia.org/wiki/Qanun_(instrument)

[1]: https://en.wikipedia.org/wiki/Guzheng

It has never been clear to me what the goal of home playback is supposed to be.

Let's simplify from a full orchestra to just a solo piano. Is the goal

1. to sound like it would if that piano was being played in my living room, or

2. to sound like what I'd have heard if I was there in the concert hall sitting in a good seat when the recording was made, or

3. to sound like what it would sound like if a replica of my living room was built inside the concert hall but with walls that do not transmit sound, with my speakers replaced by speaker-sized holes in the wall behind where the speakers normally sit, and I was sitting in that replica living room during the concert?

ABBA’s sound is mastered to fit into AM radio and jukeboxes for example. To make the music broadly listenable, for example.

Today everything is so blurry, because there’s no studio per se. It’s just sequenced, vocal is added, mastered, compressed and released (today’s pop). Rock and other stuff is still track recorded, tho.

At a certain point the speakers really start to interfere with each other and you have to worry about the stability of the image at each point in the room...

It occurred to me at that point that you're really starting to approach 3. You're trying to recreate the sound pressure patterns within the room.

That's about the point I moved on to something simpler. Granted my explorations were super simple, but it was a VERY weird thought that got me thinking about the question you've asked.

(And also much of the digital gear in the 80s when the first wave of CD releases was pretty crappy)

Sorry, but CD (16bit PCM) has better dynamic range than vinyl. The best vinyl systems can have dynamic range of around 80dB. 16bit PCM is 96dB.

Still, you can always take a great analog recording, pipe it into a good ADC, and listen to that sample forever without degradation. I think almost everything can play FLACs now.

Eventually got the better mastered version which was leaked from guitar hero or something.