"In the world of audiophiles — where provenance is everything and the quest is to get as close to the sound of an album’s original recording as possible — digital is considered almost unholy."
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!
if your DAC is a DAC, truly (that is not a high bar) then the audio that comes out of it is a perfect reproduction of the audio that went into the ADC that produced the digital audio, provided it is a true ADC (also not a high bar at all).
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.)
The Shannon-Nyquist sampling theorem only defines the sample rate to perfectly reproduce a signal of a certain bandwidth.
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.
I'm unfamiliar with the mathematics involved, but this is what Wikipedia says the theorem states:
> 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?
Yes, a perfect recording of the amplitude of a function at a frequency of at least 2B contains enough information to perfectly reconstruct that function.
Reaching this frequency is not that hard, but getting a perfect amplitude measurement is. This is where "noise, distortion and dynamic range" come in, they act as disturbances to the amplitude.
Yes, now we just need mathematically pure materials...
Like..
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.
I agree. I was not arguing against digital storage. Only against the sentiment that "just do digital and all problems are solved".. well, the storage, transport and editing is solved.. The analog problems are not solved, nor solvable, only optimizeable to a degree where further optimization becomes irrelevant, microphones/ADCs/DACs/Amplifiers/Speakers still have analog components that are inherently imperfect.
As both a sound and an electrical engineer the issue here is the if. Many DACs claim 24 Bits, while their power rails are so noise that they will meaningfully reproduce much less. The you'd also have jitter on the DAC clock etc.
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.
DAC jitter is not the major problem people make it out to be. even the lowly, uncontrolled, $2, +-20ppm frequency stability crystal oscillator has less jitter than is audibly perceptible to a human, especially given that the jitter is dependent mostly on temperature variations.
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.
I used to believe this, based on the mathematical proof referred to.
But every proof relies on assumptions, treated as axioms, ideally (but rarely) all noted.
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.
Lots of things are not especially difficult for someone paying attention to get right. But that doesn't mean they are always got right. Thermodynamically speaking, it is easier to have a digital system that fails to reproduce a sound waveform accurately than one that achieves it, because there is only one way to the right output, and uncountable ways for it to be wrong.
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.
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.
I used to have this exact problem with my cables! For this reason, I've spent the last six months building quantum shielding for audio cables, with genre-specific time crystal coatings. It is now available in my Qtsy store, if anybody is interested!
I'm not an audiophile, actually. There's no point, IMHO the most limited component of the system (with the highest actual threshold) would be my ears, and I know why: lots of exposure to loud music of various kinds over several decades.
The sample rate 44.1 kHz is objectively "low" in the sense that if you have to want a passband of 20 kHz, then extremely sharp low-pass filters must be inserted before the ADC.
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.
While the ideal filters have an infinite long inpulse response and are therefore „non realizable“, shorter filters simply increase the noise level. With filters of 100ms length you are already at SNRs that is far from what any human can handle (think of hearing rustling leaves while standing next to a jet engine).
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.
I don't agree with the characterisation in your first sentence (nothing is ever perfect in analogue), but the video was very interesting, thanks for sharing!
So you could claim to be able to reconstruct output of that filter. Which still is not the case — you cannot have signal to be both aperiodic and band-limited: either in time domain or in frequency domain signal will be periodic.
So no real (that is, finite) signal satisfies requirements of Nyquist theorem.
Not only that, your DAC needs to have infinite lookahead to reproduce the same input that was passed to ADC.
Okay you're saying mathematical perfection is not achievable in the real world. Fine.
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.
Absolutely agree. Even more, once you are in digital, you can have actual mathematical perfection, that is reproducible and does not degrade with time.
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.
You may be speaking to better informed suckers than I am - as of the people I know who are “into audio”, they all get the magic ceramic quantum nanomagnets for their cables to go on, they all swear that 8-track was the pinnacle of sound quality, and that they can “hear” digital noise on CDs.
Honestly, it’s 1% “I like music” and 99% snake oil salesmen taking advantage of fragile egos.
I agree completely. I have a large collection of original shellac 78's some as early as 1930's as well as true vinyl LP's from 1970's. They all sound terrible, what make them cool and collectable is the originality, not some magical thing about them being on shellac or vinyl.
The new digital sound quality is far superior, hands down.
Buying a reissue vinyl is a waste of money IMO.
"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”
> While some in the audiophile scene like going analog, none of them debate against the truth that objectively, vinyl is far lower quality.
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>
Many decades ago, before the appearance of any digital audio, most audiophiles agreed that vinyl sucks badly, because the records are degraded after each listening (by mechanical abrasion).
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.
And eventually even that kind of tape will wear out, too. Though I suppose it might be more of a concern in commercial usage than at home…
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.
Exactly - the easiest way to prove this is you can record Vinyl to a CD, do a double blind test between the Vinyl and the CD copy, and you can't tell the difference.
> Vinyl doesn't have nearly the resolution or nearly the dynamic range that a high-quality digital recording will have.
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.
"The fallout of the MoFi revelation has thrown the audiophile community into something of an existential crisis. The quality of digitized music has long been criticized because of how much data was stripped out of files so MP3s could fit on mobile devices. But these days, with the right equipment, digital recordings can be so good they can fool even the best ears. Many of MoFi’s now-exposed records were on Fremer and Esposito’s own lists of the best sounding analog albums...“One of the reasons they want to excoriate MoFi is for lying,” says Howarth. “The other part that bothers them is that they’ve been listening to digital all along and they’re highly invested in believing that any digital step will destroy their experience. And they’re wrong.”...And Randy Braun, a music lover, Hoffman message board member and lawyer in New York, hopes that, in the end, the MoFi revelation will prove what he’s been saying for years, that the anti-digital crowd has been lying to itself: “These people who claim they have golden ears and can hear the difference between analog and digital, well, it turns out you couldn’t.”"
That whole line of thinking just blows me away. It's possible to accurately sample and reproduce any vinyl recording in the digital realm. It's absolutely not possible to accurately record every digital recording onto vinyl. For instance, it's not hard at all to make bass so loud that it causes the needle to leap out of the track, or treble so high that the needle can't accurately track the high frequency vibrations.
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.
I absolutely despise most of the vinyl community. Don't get me wrong, I love my records. I have hundreds of them and I play them regularly. For some reason, almost everyone I come by has this delusion that vinyl is objectively better than a good digital recording. Anyone who dissents from that narrative gets gaslit:
"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.
I remember when CDs were very very new the first review of them in an audiophile magazine a friend subscribed to. The reviewer was quite unhappy.
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?
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.
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?
"I notice many of them seem to have very poor taste in music."
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.
I love records (78s and vinyl) for the tactile and visual experience-- and the fact that I can get something for a few bucks at the thrift store and not feel too bad about my toddler ruining it trying to put the needle on. To that end I have stuck to cheap, sturdy, school-grade equipment and mostly secondhand records.
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.
in the case of the 78s without the need for anything electrical in between.
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.)
I do have one of those Edison diamond disks in my collection, though no playback device that can handle the vertical signal in the groove. My ~1917 Silvertone phonograph does fine for me with the more conventional 78s.
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.
There is a certain ceremony involved with choosing and playing vinyl that cannot be recreated with digital media. That, to me, is the magic. In modern times a full digital setup is going to provide a more "faithful recreation" than anything analog.
Been there. Browsing vinyl involves flipping fore and aft and is very tactile. Browsing tape and CDROM involves running your eyes over them but you can run a finger along them to get another sense involved.
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!
I used to just look at the titles on the spines of my records (back when I had them) to decide what to play. I'd look at one and listen to the music in my head for a few minutes before deciding whether to actually put the record on, or else look at the spine of another record and do the same thing. I could spend hours doing that, and by that time I had to go do something else. I got a whole concert with no equipment at all. That's when I decided I wasn't an audiophile.
I have a few songs ripped from CDs over the years that I’ll be very unhappy to lose if iTunes Match ever goes away, since I’ve lost the CDs and as far as I know they simply aren’t online. Such a great service.
If your Apple ID gets accidentally suspended because of a false positive from the "can't turn it off" clientside CP local file scanning that Apple is planning to roll out to all of our devices, you won't get any warning.
There is also some with CDs, but vinyl's the real deal. The record cleaner brush. Inspect the needle for dust. Be sure the record is properly seated. Sit back and enjoy.
I'm not going to say the date but my birthday is exactly 4 months before a certain holiday. So some years back (too many, alas), on this holiday and 4 months after I turned 33, my friends put on a party to celebrate both the holiday and my 33 1/3'rd birthday. Naturally, LP records were played.
Watching the reels spinning at different rates and the tape moving on an intricate path, in a high-end open-reel tape recorder with multiple motors and multiple heads, e.g. a Revox, was even more satisfying.
I recently got a record player and have been amassing a small collection of my favourite albums. Including the reasons you gave, my excuse for buying/listening to records is I like having a physical representation of albums I like, and something that won't disappear because a contract expired with a streaming service.
Stick with it. I discovered some great music buying random stuff from the <$5 section. You'll meet fun people at the right record stores (if you're in Minneapolis, I have some suggestions). Every once in a while one of your LPs will turn into a nice investment piece. If you haven't tried repairing/restoring your own turntable, I recommend it. The engineering inside all the little mechanisms in an old turntable are fun to tinker with.
There definitely was a short period in college where I needed to spend money I barely had to upgrade my system. This was when CDs first came out. I could tell, when listening through my stereo (as compared to plugging my headphones into the CD player) that there was a ton of extra noise/distortion and most importantly, a very high noise level (audible hiss). Ended up having to go to Salvation Army to buy a $25 used receiver and the speaker store to buy a pair of $200 speakers before I got decent results. Even after that, I had no end of problems with the analog cable. it took a long time to switch to digital.
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.
That's the main reason I really like vinyl. Physics prevents the mastering engineers from going overly crazy. Vinyl often sounds more carefully mastered, even though nothing would technically prevent you from releasing the same thing in digital form.
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.
It's still true that it takes a skillful ear/hand to do that mastering. I've heard terribly mastered older (not abused) records and modern releases that are excellent. A good original copy of Rumours is about as good as it gets IME.
Vinyl _does_ also provide a nice soft upper limit to the loudness wars.
Couldn't we characterize vinyl limitations and capture that as a digital filter? And if so would you be able to tell the difference between the vinyl and filtered digital?
It wouldn't work with albums where the mastering is different, especially if the digital release is compressed more. You cannot undo squashed dynamic range really. A famous example from a while ago is the album stadium arcadium by the red hot chili peppers. Comparing the two versions of a song should sound notably different even to a casual listener. And you can really see the difference just looking at the wave forms.
The premise in this part of the thread is that the mastering engineers will not make a record that exceeds the limits of the medium, and what fits on the medium sounds nice.
Assuming this is correct, and that the digital mix intended for cds or streaming/downloads outside vinyl's range, you might be able to filter the digital mix to fit in vinyl's range and that might approximate the vinyl mix, but maybe the mastering engineer would make different choices (of course, maybe they just use a filter). It'd be interesting to explore with actual data.
I listened to "Disintegration" last week exactly like that, but with Apple Music. It was the first time I'd listened to an album start-to-finish in years, and I highly recommend it.
I like not having to flip the record halfway through, or being able to listen to "The Wall" without 3 flips.
The structure that side/disc limitations impose is part of the listening experience, to me - I like having to take a moment to consider the sub-narrative of the side as a whole before moving on. Even when streaming I’ll often look up the vinyl track listings to get a sense of the artist’s intended schema.
> I like having to take a moment to consider the sub-narrative of the side as a whole before moving on.
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.
Even if the master wasn't digital, how many vinyl mastering engineers used a digital reverse RIAA?
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)...
Don't forget that playback progressively damages the medium. A true audiophool keeps a stockpile of virgin vinyl so they can playback once and throw it away.
Most of the time people will argue that vinyl has "warmth" and this makes it subjectively better. This is true, but it's actually because of the pleasing distortion record players add to the music. Being distortion, even if it improves your listening experience (and this depends on genre, though it generally does), it is certainly objectively worse in terms of accurate reproduction.
> 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.
But if analog is what you knew, that wasn't "distortion". That was "natural", because it was what you grew up with. That was the way things were supposed to be.
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.
On a high quality record with a high quality setup, you can have both. It's because the distortion is really subtle, and hits all the right harmonics. On bass-heavy music it makes the bass sound really nice. The other parts of "warmth" aren't really in my wheelhouse, but it makes the bass in bass music better
An instrument amplified by vacuum tubes does have a distinct sound, "vinyl" also has its own signature.. magnetic tape is pretty analog too. All of these signature sounds can be digitized/discretized at sampling rates that preserve the vinyl crackle or tube warmth or combinations. To me it seems like marketing and the consumers justification of their susceptibility.
Not necessarily (well I guess it depends what you mean by "pushed"), but I think tube amps naturally boost the even-numbered harmonics at all volume levels. This sounds good but is distortion.
That’d show up in the Total Harmonic Distortion figure for the amp, and for any halfway hifi quality oriented amp it’ll be well below 0.1%. Perhaps “golden ears” can hear that? I’m not convinced. Most hifi audio amps typically run at well down into the single digit percentage of their rated output at ordinary listening levels in most systems, so they’re an order of magnitude or two away from clipping. If the feedback circuit and the output devices (transistors or tubes) can’t keep the response perfectly linear the thd down under 0.1% across the entire audio frequency band in that power range, it’s just badly designed.
You can also bake analog sounding warmth into your hi-res digital master and it's done all the time. There are a wide variety of different digital mastering plug-ins which do remarkably sophisticated analog modeling.
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.
> Individuals can prefer different sonic characteristics encoded in an output but that's an aesthetic choice.
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.
Absolutely, personal sonic character preferences can vary per content, listening context and the play back device (plus play back signal chain like DACs, pre-amps etc). And it's not just personal taste, most human's bio-based input 'hardware' varies enough to matter, especially for those of us who are out of warranty (ie >40 yrs old). There are even interactive apps which will give you a quick hearing self-test and build an EQ profile based on your individual response curves. I use one bundled into Android and it definitely improves my listening experience.
I've heard the recent Kemper Profiler uses advanced analysis to generate fairly convincing simulations of analog equipment, given its response to specially designed test signals. I haven't tested it myself though.
The finding some years back was that the harmonic distortion in tube circuits was even harmonics and in solid state was odd harmonics (or maybe vice versa). This could cause tube electronics to sound better than equally good (by measure of thd %) solid state electronics.
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?
That reminds me of a thread here from a few weeks ago about how somebody was training an ML model to add noise to video, so that the noise-free version of the video can be compressed better but then the noise can be added at playback to maintain the authentic feel of old video.
Sure you can, but then you can't call it 'analog' anymore. But you can reproduce the distort...I mean, the "warmth" with digital filters or whatever and it will pass any blind test.
I still think that "warmth" is a way to describe not actually hearing the separate digital samples subconsciously, and just hearing the sounds gliss into each other. I also think it's an anachronism that comes from way back when the digital sample rate was often a lot lower (and people would insist that their live digital effects sounded as good as analog, when you could sometimes even consciously hear the samples jumping into each other.) Other than that, it seems like something that people project onto music with bassy reverb.
You cannot hear samples, or the "steps" between them. This is a silly idea that comes from people seeing pictures of the waveform as it is digitally captured and imagining they can hear it.
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’s really funny about this is that although CDs objectively can present a greater dynamic range, the loudness wars have created a lot of highly compressed CDs that do sound worse than a lot of vinyl, where it often seems that more cars is taken with mastering because of the limitations of the medium. Additionally, a lot of early 80s albums sound better on vinyl because digital mastering wasn’t as good then. It is an absolutely night and day difference between my vinyl copy of Unknown Pleasures and my CD.
Yep. This is why I like buying original 80s CDs where they just chucked the master tape onto a digital disk, rather than post-2000 reissues where they slap a limiter on the master and jack it up +8 dB.
modern mastering does not optimize fidelity in a car at the expense of other modes of listening, and these articles do not indicate as such.
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).
> It's absolutely, 100% fine for someone to say that they subjectively prefer the sound quality of a record player to a CD.
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…
Flat earth, Scientology, dowsing rods, ... willful ignorance is hard to fight.
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.
Agreed. So much of the early negativity around "digital" and "compression" was due to avoidable mistakes made when mastering some particular pieces of media back in the early days.
There's no a priori reason why any recording medium needs to have a flat response curve, so long as the curve is known so it can be reversed. The reason for the RIAA curve is that most sound has an inverse relationship between amplitude and frequency. The RIAA recording curve limits the overall amplitude of the groove, allowing for better management of dynamic range and distortion in the cutting and playback processes. The slight lumps in the curve are a concession to practical filter technology of the 1950s.
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.
Indeed, that's true. The vinyl thing is probably more about the vinyl trend.
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.
> it's not hard at all to make bass so loud that it causes the needle to leap out of the track
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.
Yes! That exact song is perhaps the canonical demonstration of the limitations of vinyl! Here are some pics of the grooves on that LP: https://imgur.com/a/veVB0
Was going more for: “Rabbit hands Duck a phonograph of a nuclear detonation. Duck carefully places the needle on the record. Duck sits quietly in front of the phonograph's horn. Blastwave from nuclear detonation is emitted from the horn while Rabbit looks at the viewer smugly.”
> For instance, it's not hard at all to make bass so loud that it causes the needle to leap out of the track
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.
They're not referring to amplified sound. If you press a record with too much bass on it the 'wobble' of the groove itself can end up so extreme it'll make the needle jump out - no speakers required.
The thing I miss with the Apple audio chain is a user-controlled equalizer. With my hearing issues, I really need to boost treble, and I prefer to boost bass. I haven't yet discovered a way to do that. Apple thinks that flat is best, and so should you, apparently.
On OSX Music, window>equalizer. You can also grab Rogue Amoeba’s “SoundSource” to have volume controls for every noise-generating app on the computer, and apply effects chains to them, including EQ.
I love SoundSource so much. I bought it so I could use the volume keys on my keyboard to adjust the system volume on my HDMI monitor. I was blown away when I found out I could plug my collection of VSTs into the output pipeline.
Great, thanks very much. All their products look useful.
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.
And it's insane that this hasn't changed for at least a decade! Apple's own PowerBeats Pro sound like garbage without bumping up the low frequencies, but with an equalizer they sound great. Or maybe this is just subjective and you can file it under "accessibility".
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.
Yes, I've used a couple of those iOS apps, but the end result is not boosted enough in the higher frequencies. My $500/pair of MDhearingaid Airs work a lot better, but they don't connect to my Apple devices.
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.
That’s the idea, in theory. But because analog components such as the resistors, capacitors, and transistors will not be identical[0] between the master and player, it’s not perfect.
[0]: you can get very low tolerance resistors and capacitors, but they’ll cost you
To a point, but remember that you're dealing with low-bit floating point math. That boosting stage creates information that wasn't originally there. That's not so bad in the treble case where original * boost / filter is pretty close to the original. But in the bass case, original / filter * boost might be significantly different than the original. That's usually not a huge problem because the low bass frequencies are pretty forgiving, but it's something to consider.
If they were idempotent, yes. But they're not. Analog processing can't be perfect, and amplification is an inherently noisy process. The RIAA curve first cuts the signal on the bass end of the spectrum, throwing away information in the process (because vinyl doesn't have infinite resolution). Then it amplifies that degraded signal.
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.
No; it's really fine - same with tube amps etc. It's OK to have a subjective preference for some kinds of distortion. It's not OK to say it's "truer to the original sound", which is where this usually comes off the rails.
You're not wrong in the general "audiophile" world where fools and their money are parted over snake oil magical cables claiming benefits only those with the most golden of ears can enjoy.
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.
Right, but this subthread is about it being fine for people to prefer the sound of less accurate systems, just not to claim those systems are objectively superior.
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.
A digital recording has to be mastered with the understanding that it could end up getting played on a car stereo, on a bluetooth boombox, on cell phone speakers, in bluetooth earbuds, at a dance party with a DJ or on a hi-fi stereo setup that looks like the one from the old Maxell "blown away" ad. A recording on a vinyl record doesn't. The person mastering a record doesn't need to care about bluetooth earbuds, car stereos, cell phone speakers, boomboxes, and only marginally about DJs at dance parties. They can tune their mixing and mastering decisions to just the case of a hi-fi stereo setup with an enthusiast listening. They have to make fewer compromises because there are fewer cases to handle. The format enforces those constraints. You can't just ignore that.
> They can tune their mixing and mastering decisions to just the case of a hi-fi stereo setup with an enthusiast listening.
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 there's a "high bitrate" (24-bit, 96kHz FLAC or something like that) master, often enough that will be the case. You can reasonably assume that someone who's buying that version of the track has something capable of playing it back sanely. Same with the "high def audio" formats (SACD, there was another one...) - they often did, in fact, sound better. Not because of the high resolution, so much as because they were mastered to be listened to on a competent system, not on the subway on earbuds.
Nobody who is mastering records gives a damn about bluetooth boomboxes or phone speakers; and there is no way to optimize for a phone speaker and the "Maxell blown away" stereo setup.
Yes, because the sriracha can be confirmed to exist.
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.
Not only can you not tell the difference between analogue and digital, you also cannot tell the difference between lossless and properly transcoded MP3.
Most audiophile mania is poorly informed at best and astrology at worst.
> you also cannot tell the difference between lossless and properly transcoded MP3
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.
Many mp3 encoders cut off really high frequency stuff - somewhere between 16khz and 20khz depending on the encoder. Often this isn't changed by any of the quality settings, e.g. on lame it's still enabled on the "insane" preset without manually disabling it.
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.
^ yeah exactly. Also, you really do need a pretty good sound system to hear it. I was with the poster above you for a long time, but then I did blind A/B testing on my buddy's $10k monitors, and the difference was clear.
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.
Not the person you replied to, but Hotel California’s Hell Freezes Over version is generally very well regarded in mastering.
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.
Thanks, I’ll try and take the test asap. (I need to find my earphones and usb c to 3.5mm adapter first). Is there any way I can contact you after taking the test, perhaps email?
> Hotel California’s Hell Freezes Over version is generally very well regarded in mastering.
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.
As a former orchestra player I can say that, with a good hi-fi system, the difference is audible, in terms of soundstage size/depth, not in details.
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.
Badly transcoded mp3s are the worst. For most artists I couldn't tell the difference between a v2 and a flac. There were a few who you could but they're mostly not what people are listening to (intentionally very noisy artists). Realistically if I just heard them blind I don't think I would have cared.
As an anecdote I've noticed that while driving, the bass line in YouTube music jazz can be incredibly hard to hear despite my shelling out for the premium Bose sound system that can shake the vehicle. So I wonder whether this is the encoding or whether it's really supposed to be that quiet.
There's definitely some difference in quality with some music. For instance, the reverb will be slightly better with lossless, in my opinion. Some might say it's not much to bother with but it's definitely there.
MP3 is not monolithic, and has changed drastically since it was released.
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)
> MP3 is not monolithic, and has changed drastically since it was released.
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.
True, though I think variable bit rate MP3 is fine enough for most.
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.
> unless you go to a source like Qobuz which serves FLAC.
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.)
While I likely wouldn't notice the difference, digital space is so cheap now that it costs me very little to store everything I want as lossless copies. Even if the difference is negligible, there's little reason not to get the best quality I can.
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).
>The fallout of the MoFi revelation has thrown the audiophile community into something of an existential crisis.
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.
Cables are all well and good, but the quest for perfect audio will continue until somebody invents a gadget which makes you 15 years old again, hanging out in your friend's basement, having just tried pot for the first time and hearing Led Zeppelin III on a cheap Sears record player. I don't know exactly what this gadget will look like, but it's the only thing that will get audiophiles the sound they've been chasing.
I think I'm in the right vintage for that, but did cassettes ever sound good? I could get nostalgic for the sound (especially fast forward/reverse, which is lovely), and I probably never had good equipment, but tapes were always kind of warbly for me. I had access to older records too, which could sound good when there wasn't too much hissing and popping and they weren't skipping. Some records were maybe off balance? and wouldn't play at consistent speed either. Again, I didn't have great equipment.
>I think I'm in the right vintage for that, but did cassettes ever sound good?
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.
>Yep plus early ‘discman’ didn’t have anti skip and battery life was terrible.
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.
> 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.
You don’t need to attack any person or community - especially given that rich people buying expensive stuff is hardly hurting others directly. Please make good faith arguments.
I have never heard of Wayne Gretzky; Google tells me he's a hockey player. I don't think I have ever seen a match of hockey in my entire life. I'm not annoyed by what I don't know.
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.
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.
That's a little simplistic. You can certainly tell the difference between a good analog recording played on a high quality system vs. an MP3. But once you throw enough samples and bit depth at it (more than are on a standard CD), you get quality that's equivalent to, or better than, most analog media.
44100hz sample rate is enough to reproduce well above the highest frequency even young humans can hear, and 16 bits of depth is enough for 96db of dynamic range, enough to make the noise floor of any consumer system totally inaudible.
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.
So it's been many years since I looked at this stuff, but the reason for higher sample rate isn't to do with humans being able to hear over 20khz. If you sample at 44khz you need a high pass filter at 22khz to avoid aliasing. Since that's pretty close to 20khz, it needs a pretty sharp drop off. Having a filter with a sharp drop off can introduce artifacts.
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 :).
You are correct; with a 96 kHz sampling rate, you don't need such a brick wall filter. But then you dohn't have to actually keep the data at 96 kHz; it can be reduced to 44, this time using a digital filter.
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.
That's the "further processing" notion the grandparent comment talked about. A correctly processed signal at 44.1kHz is more than sufficient to reproduce all audio. But if you're going to do something with the sample series (like low-pass it for aliasing protection in your example, or to resample it for conversion, or for any effects or mixing step really) it's good to have extra data in there to prevent headaches. Likewise sampling at 24+ bits prevents accumulated error in repeated processing, etc...
There's a space in the audio world for more bits of data. But the final output format isn't where it belongs.
[ EDIT: contents of this post were factually incorrect. What Kazinator wrote in response is correct, and I don't want to leave incorrect ideas floating around. ]
If the goal is audio capture and reproduction, the filter used when sampling at 96 kHz still starts rolling off at around 20 kHz. The wrap-around aliasing artifacts do not begin until around half of 96 kHz, or around 48 kHz. So the filter only has to be steep enough to hit a large attenuation at 48 kHz: full signal to virtually nothing over the space of 28 kHz.
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.
Sure, but going to disagree about the numbers there. I have a DAW interface which is exceedingly linear (+/- <0.15 dB) up to 35 kHz. Don't need to start filtering so low to hit those results.
This explanation loses me at "infinity per dB" ... and I'm an EE. I think you're trying to cover too much ground here, it's really confusing to try to understand what you mean.
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."
Nyquist isn't about such assumptions at all. It is an information theoretic value, based on the provable claim that there is zero information present in the original analog waveform that is not represented in the digital version.
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.
There is zero information present in a waveform that is not contained in the digital version, if that original waveform is confined below the Nyquist limit. Either it is that way already, or else is derived from an original-original waveform that isn't, by low-pass filtering.
That's an excellent coda/qualification of what I said. Put into a more listener centric context: there's no audible information in a waveform that is not captured by sampling it at 44.1khz.
That's fine - but, to the extent that's true, it's true only about the recording and mastering process. CD, as a delivery mechanism, is totally fine and perceptually flawless.
> 44100hz sample rate is enough to reproduce well above the highest frequency even young humans can hear
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.
Not only that, but what you hear at those upper frequencies isn't tone anyway. the highest harmonics of tonal sounds live in that region, plus aperiodic signals (the hiss from sibilants, cymbals crashing, and so on).
The content can be faked, and this is used in low-bandwidth codecs and such.
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.
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?
Normally, it’s “2”. Music is mastered to transfer you to the place where it’s recorded. Mastering is just tuning the sound to fit into your music system limits, while keeping the atmosphere as much as possible.
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.
I worked in guitar audio effects for a while (the exact opposite of this discussion), but I started playing with stereo guitar rigs and then thought about three speaker setups and beyond.
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.
That's a real thing, to be sure. The Loudness Wars have completely ruined any number of otherwise decent recordings. The worst I ever heard was Dropkick Murphys's "The Meanest of Times" which had the dynamic range of an air conditioner in Phoenix.
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.
Metallica's Death Magnetic was another truly dreadful example of the loudness war - Thankfully a better mastered release came out in 2015. I've just got my hands on this release and comparing it against my original copy it is a night and day improvement - especially on "All Nightmare Long".
However, you can’t change the mastering of the album you have bought. When the CD version of the album you have bought has subpar audio quality, the abilities of the medium has no value and importance.
"When the CD version of the album you have bought has subpar audio quality"
Yes, but these mastering choices could apply to the vinyl version too. In other words, your response had nothing to do with a comparison of the two media.
Nope, you can't apply the same mastering to the vinyl, esp. the latest stonewalling compression/normalization techniques because of RIAA equalization [0].
Without RIAA equalization, it's not possible to keep the needle in the groove with that amount of bass and low end. You can't keep the needle on the track with stonewalling even with RIAA equalization.
As a result, you are limited by the medium itself, hence you have to make milder choices.
Obviously the mastering differs. So your point is that the mandatory mastering limitations specific to each medium should be factored into comparisons between media?
Perhaps, but what about optional mastering choices like brick-wall limiting with normalization (which is what I think you meant)? Maybe I'm misunderstanding your point.
Telling this out loud upsets a lot of people, but I’m on the same boat with you.
A good lossless album, regardless of the medium played through shows a significant difference in soundstage given the system can handle the resolution thrown at it when compared with a MP3/M4A file.
