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"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 — digit

• iamacyborg 4y ago
  > 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.
• naikrovek 4y ago
  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.) https://xiph.org/video/vid2.shtml https://xiph.org/video/vid2.shtml
  • jstanley 4y ago
    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!
  • timpattinson 4y ago
    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.
    • e63f67dd-065b 4y ago
      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?
      • karelpeeters 4y ago
        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.
      • dusted 4y ago
        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.
  • yobbo 4y ago
    Audiophiles sometimes claim they can hear artefacts of the "low" sample rate 44.1 kHz, and only 16 bit samples. Nowadays there exists audio files recorded in up to 96 kHz with 24 bit samples.
    • adrian_b 4y ago
      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.
      • smat 4y ago
        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.
    • 4y ago
  • quickthrower2 4y ago
    This is trivially correct if you assume: 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.
    • SideburnsOfDoom 4y ago
      There's always a threshold though. 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.
      • quickthrower2 4y ago
        Audiophiles talking about quantum fluctuations in their analog recordings :-)
        • dr_kiszonka 4y ago
          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!
        • SideburnsOfDoom 4y ago
          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.
  • atoav 4y ago
    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.
    • naikrovek 4y ago
      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.
  • ncmncm 4y ago
    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.
    • naikrovek 4y ago
      it's almost always far more than good enough for human ears. 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.
      • ncmncm 4y ago
        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.
  • 112233 4y ago
    > Nyquist theory proves it. 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?
    • dreamcompiler 4y ago
      The input of every competent ADC process is always a well-designed multipole lowpass analog filter below the Nyquist frequency, so yes. EDIT: This also applies to sigma-delta sampling. Although there you can often get away with a simpler single-pole input filter.
      • 112233 4y ago
        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.
        • dreamcompiler 4y ago
          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.
• madaxe_again 4y ago
  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.
• dmitriid 4y ago
  > 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>
  • adrian_b 4y ago
    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.
    • nikau 4y ago
      Worth noting that the "tape recorder" was reel to reel not mainstream compact cassette. 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.
      • iggldiggl 4y ago
        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.
• zw123456 4y ago
  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.
• deleted 4y ago
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• mellosouls 4y ago
  Reminds me of the famous John Peel quote: "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”
• Shorel 4y ago
  > But with good recordings and mastering 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.
  • twangist 4y ago
    > Sound engineers are making sure this is not the case That's pretty offensive and ignorant. The loudness wars ended at least 10 years ago.
  • nikau 4y ago
    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 forced better mastering.