Has anyone been able to define well or measure differences between vinyl and digital?


It’s obvious right? They sound different, and I’m sure they measure differently. Well we know the dynamic range of cd’s is larger than vinyl.

But do we have an agreed description or agreed measurements of the differences between vinyl and digital?

I know this is a hot topic so I am asking not for trouble but for well reasoned and detailed replies, if possible. And courtesy among us. Please.

I’ve always wondered why vinyl sounds more open, airy and transparent in the mid range. And of cd’s and most digital sounds quieter and yet lifeless than compared with vinyl. YMMV of course, I am looking for the reasons, and appreciation of one another’s experience.

johnread57

Showing 26 responses by thespeakerdude

@asctim , here you go. https://www.audiogon.com/listings?q=test+record

RCOA test systems record has a frequency sweep. So does the Stereo Review one but it is pretty old.

Some newer ones here:

https://store.acousticsounds.com/g/48/Test_Record

 

Perhaps you can post how to do your phase test and the frequency roll off. I wonder if that is unique to your setup?

Reading the answers, @asctim ’s most resonates with me. Some of my favorite recordings are on vinyl. Some of my favorite recordings are digital. I don’t feel a need to declare vinyl better than digital to like it. Neither is my first born :-) I have never seen any argument that holds up that supports vinyl having any property better than digital. IMHO there is probably something to the magic that @asctim describes, flaws or not. Vinyl also ties one had behind the back of the person working on it. Does that force them to pay more attention to what they are doing? Groove space is a limited quantity. Make the most of every one and do it right or it will sound terrible. Digital? Throw whatever you want down, it will fit and come out the other end.

@sns , I cannot speak for audio equipment in general, but for professional speakers (that some consumers buy), except for lunch room conversations, analog as a storage medium is not something that ever comes up. We never say, "we better feed this with a record player or tape player in case we missed something" though some people bring in needle drops.

IMHO as well, I think the MOFI debacle should put to rest much of these discussions. It won't, but it probably should. We could be spending effort figuring out where the magic comes from, not making up properties that always fall apart under the magnifying glass.

 

@fair ,

I see one major flaw in your logic. CD and two channel DSD is just that, two channels. When I am in a room, out in the wilds, or anywhere, there could be an infinite number of sound sources, that all contribute to that data you mention. When I am at home, there is only 2 sound sources. They may bounce off the walls, the floor, the windows, but there is only 2 sources. In another thread we are talking about ATMOS with 9, 11 or more speaker which still only simulates all that we can hear.

Use that 11 speaker example at CD data rates. The rate is 7.8 mbits / second. 11 speakers is not enough. 24? Now 16.8 mbits/second. Well beyond your 3.5 - 4 mbits/second.

I don't think you can correlate the data rate for the cochlea with the brain, which I suspect is a WAG, from sound information that comes from all directions, with what comes out of 2 speakers.

 

Auditory processing circuits in the brain drastically compress this flow of information: this explains why lossy encoding works so well. Still, if some part of the original 3.5. to 4.0 MegaBits per second flow is arbitrarily removed, artifacts may occur.

As we can see, CD falls about 5.7x short of the target of complete digital transparency. DSD64 falls about 1.4x short.

DSD128 encodes more information, by factor of 1.4x, than the nerves running between cochlea and brain can transport. DSD256 exceeds the sufficiency threshold by factor of 2.8x, and thus shall be considered far more than enough.

@johnread57,

We are in the early days of ATMOS and Auro3D. I have heard some bad recordings and some exceptional ones. I think it will be difficult to find a sweet spot with users, most will not put in that many speakers. In theory the object oriented nature of ATMOS takes this into account, but my gut is it may be difficult to translate recorded live events with multiple microphones to a different speaker array. Of course, where all the speakers get a signal manufactured for them through mixing, then it likely will work fine.

These formats on headphones will be popular far before they are for speaker systems.

Can you imagine the remixes from older material? Scary!

I do not know what you mean is the image of 2 channel speaker systems equal?

@akgwhiz , some good thoughts. Dither provides added dynamic range in digital. Is there an equivalent with hearing and analog noise? Don't neurons have discrete trigger levels?

@fair 

I am familiar with binaural representations of full immersive audio space. It is still only a representation of a full immersive space and sound sources. Do you have a link to these data points? I think I have seen that before or had it mentioned to me and there is another error in your interpretation, probably more significant than the one I mentioned.

The hair cells each respond to a range of frequencies and triggers the nerve associated with it. The cell next to it does the same. There is a large overlap of the frequencies between cells / nerves, but each one generates a signal, or data based on your inference. Because there is overlap and the range of frequencies is large for each cell/nerve, then there must be massive data redundancy. There may be what looks like 3.5-4.0 megabits of data, but most of it is redundant, and if combined to generate a single data stream, the amount of data would be far less.

We cannot hear over 20Khz. In a listening room, 90db exceeds the full range of human hearing from the quietest detectable sound above the noise floor to the loudest sound we can tolerate, and at those levels, much of it distorted. Are you familiar with Shannon Hartley formulas, C(bps) = B * log2(1+SNR) ?

This states that the maximum data that can be transmitted in a channel is Bandwidth * log2(SNR+1). This provides the CD data rate.

@clearthinker ,

The MOFI debacle I think is the best counter to your argument about digitizing of an analog signal. That they could do this for over a decade, while creating some of the best rated vinyl for sound puts a hole into the argument that useful information is forever lost. Can you explain your concern with chopping time? What do you think is lost when this happens knowing we can only hear a limited range of frequencies?

Movies and video chop images in the time domain.

@fair ,

There are far too many errors and misinterpretations in your post. It will be highly misleading to someone who is not familiar with digital audio.

Start off with sampling theory and window functions. The requirement for infinite time is only required for infinite precision. We obviously do not need infinite precision as our ears do not have infinite dynamic range, and audio does not extend to 0Hz. Purely practical, the inherent noise of the quietest rooms and the onset of pain sets hard limits on what we need. Hence we do not even need infinite time. The windowing function does its required job. Sampling theorem is absolutely applicable to music. These theories are tested day in and day out. All our communications are based on them.

Short term Fourier Transforms are analysis functions primarily. They make pretty graphs, and are used for signal analysis. The data that comes out of them is bounded by the window width, which defines the lowest frequency that can be represented, the sample rate, which sets the upper bound, and both which define how fine of frequency analysis can be done. They do what they do accurately, understanding their limitations.

This is exact equivalent of DSD encoding, only its frequency is 64 times lower. Correspondingly, the highest frequency that we can hope to encode with similar fidelity as DSD will be 44,100 / 2 / 64 = 344.5 Hz. Say goodbye to the "micro expression of transients and micro transients"!

I am going to highlight this last paragraph. This is 100% false. That is not how DSD works. The single bit in DSD is not equivalent to a single bit change in PCM. No direct comparisons can be made. Hence you conclusion cannot be made and can be assumed false.

There are two flaws in your statement of equivalence 11 bits and 0.03% distortion detection. More like 3 flaws. That distortion limit is at full scale. Assume your stereo is set for 100db peaks, which is fairly loud and you have low distortion playback. There is a particular distortion level evident at that volume. In your analysis, you are claiming to be able to hear distortion at the bit level, on sounds that are only 70db. Are you claiming to be able to hear 0.03% distortion on a 70db peak signal. Not average, peak. That is a low volume level. If you are a very quiet room, 25db, that is only 45db above the noise of your room. That is 8 bits. So there is still 3 bits of addition digital range below the noise floor. Further, CD is dithered. Dither improves the dynamic range where our hearing is most sensitive for added noise where it is not. That extends the dynamic range to where we are most sensitive to 110db. Your argument fails with that information.

 

So, for faithful reproduction of a symphony we would need 90 / 6 = 15 bits for encoding the dynamic range, and 14 bits for encoding the shape of the signal. 15 + 14 = 29 bits.

This is obviously not at all accurate. You are stacking flaws in your understanding of how digital works to come to incorrect conclusions. The digital bit depth only needs to be large enough to encompass the full dynamic range. By shifting noise, we don't even need that many bits for the dynamic range. DSD has 1 bit depth. The noise is shifted to provide large dynamic range. CD has 16 bits. The noise is shifted to increase the dynamic range.

 

However, in order for the quietest signal to be still distinguishable, it only needs to be 6 db, or 1 bit, above the noise floor. This leaves an equivalent of 11 bits for dynamic range, which is more than twice of the 5 bits of the usable CD dynamic range.

 

You are basing this conclusion on a stack of fundamental flaws. It does not represent reality. More accurate is that we can hear below the noise floor. Vinyl has a signal to noise ratio of about 70db, sometimes higher, but the dynamic range can extend 10 or 20db. CD almost beats this with a raw dynamic range over 90db. The dithering extends this to 110db far higher than vinyl.

 

Viewed from this perspective, LP has twice as wide usable dynamic range in comparison with CD. But higher noise and distortions.

This is also based on a stack of flawed assumptions. It is incorrect

 

@clearthinker , I don't think there is anything to support you current view. I am not sure issues in playback when CDs came out was due to jitter, the brickwall filters, or just poor quality playback equipment. I have not delved into it extensively and I am more interested in the here and now. When I can look at a jitter test on a Schitt Modi+ for $129 and even on Toslink the jitter components are -130db, and better on other inputs, jitter is not something to concern ourselves. I have read even old studio equipment had good clocks for the ADC even if the DACs did not for the stuff we bought.  Dither can be added analog or digital. It extends the dynamic range. It works. What is your concern about it?

CD reproduces to 20KHz.  It have been many decades since my ears heard 20KHz. What do you think we are losing by not having higher frequencies. 24/96 would solve that issue.  Someone posted above that with vinyl as you move to the inner grooves, there is not much above 10KHz and it is all distorted.

I am still back with @asctim. All the points he lists are real. I am more inclined to believe that things that are real and identifiable, including mastering, are why we often like vinyl, as opposed to interpretations of digital audio that are either incorrect or have no evidence to support. Is that the simple explanation? I think so.

@fair ,

 

I believe at that point I provided enough explanations. Your reactions are quite typical of engineers who consider the classic DSP based on Fourier Analysis the only true paradigm.

 

From where I am sitting you have not provided one explanation because every single explanation or example you have used is wrong, stacking misunderstanding on top of misunderstanding. Fourier analysis is not a paradigm, it is a mathematical translation from time to frequency, it just is. The accuracy, as I previously wrote, is based on suitable bandwidth limitations, and appropriate windowing functions, much which occur naturally in audio, but are still supplemented by the appropriate analog filters, over sampling, and digital processing. People are not just guessing at the implementation and not considering what the underlying waveforms can and do look like. Let me break just one section down to illustrate your logic flaws and misunderstandings. It carries through to the rest of what you have wrote:

Imagine an audio signal which is a sinusoid of frequency 12 KHz, with amplitude described as piecewise function of two segments linear on dB scale. First segment goes from 0 to 100 dB SPL during first half cycle of the the sinusoid. Second segment goes from 100 db SPL to below quantization noise during the next four cycles of the sinusoid.

Try to sample it with 16/44.1. Then try to reconstruct the signal from the samples. Then shift capture time of the first sample by 1/8 of the sinusoid period. Repeat the exercise.

What you’ll find is that, first, reconstruction will be pretty rough, and second, that it will be wildly changing with even small shifts of the first sample capture time.

You start with a flawed premise, proceed to a flawed understanding of digitization, and finish with an incorrect understanding of reconstruction.

Flawed premise: 12 KHz sine wave do not suddenly appear, starting at 0. As I previously wrote, we are dealing with a bandwidth limited and defined system. You cannot go from 0, silence, directly into what looks exactly like a sine wave. That transition exceeds the 20KHz (or whatever we are using). Also, the digitizer, filters, etc. will have been running and settled to required accuracy by the time this tone burst arrives. Whatever you send it, will have been limited in frequency, by design, by the analog filters preceding the digitizer.

Flawed understanding of Digitization: As written above, the digitizer was already running when the tone burst arrives. Whether the sample clock is shifted globally the equivalent of 1/8 of a 12KHz tone, or not, will have no impact on the digitization of the information in the band limited analog signal.

Flawed understanding of reconstruction: When I reconstruct the analog signal, using the captured data, whether I use the original clock, or the shifted one, the resulting waveform that results will be exactly the same. In relationship to the data file, all the analog information will be shifted by about 10 useconds. That will happen equally on all channels. The waveforms will look exactly the same either case. One set of data files will have an extra 10 useconds of silence at the front of them (or at the end).

 

As I highlighted, the approach you are advocating doesn’t address the need of having some bits left available for encoding the shape of signal faithfully enough to be perceived as distortions-free.

 

I am sure you believe this, but you used flawed logic, a flawed understanding of the waveform, and a flawed understanding of digitization, reconstruction, and the associated math.

I went back and looked looked at the research. In lab controlled situations, humans can detect, a very specific signal up to 25db below the noise floor, A-weighted. That is not listening to music, that is an experiment designed to give a human the best possible chance. For vinyl, that means in a controlled experiment, maybe you could hear a tone at -95db referencing 0db as max. With CD, the same would be true at -110db (or more) due to the 100% use of dithering.

 

It a signal consists mostly of harmonic components quickly changing their amplitudes, non-harmonic transients, and frequently appearing/disappearing components, dithering is not as effective.


I cannot respond to this as it is wrong, and I am not certain where exactly you have gone wrong. As I wrote above, you have made errors of logic and understanding in all the areas critical to digital audio. As I wrote previously, dithering can be applied in analog prior to digitization.

 

The noise considerations started to amuse me lately. Practical examples were a trio of class-D power amplifies, highly regarded by ASR. I bought them over the years, evaluated, and quickly got rid of, due to intolerable for me distortions.

To be sure we are on the same page. Class-D amplifiers are analog amplifiers. They are not digital. I will correct you. Perception of distortion. You are making an assumption of something that is there, without proof it is there.

 

Not on assumptions. On theories. Fitting experimental facts. The theory I use is more sophisticated than the classic one, taking into account analog characteristics of human hearing system.

Which theory is it that you are using? I noted many flaws in your understanding of critical elements of digital audio, and assertions that are also incorrect. I have already falsified your theory.

 

Perhaps not important to this discussion, but 16/44.1 is a delivery format. From what my colleagues tell me, is has not been used as a digitization format in decades, and depending on your point of demarcation, it has not been used as a digitization format since the 1980’s, as all the hardware internally samples at a higher rate and bit depth.

@fair do you think that the people working on this technology never do comparisons of input and output waveforms with real music?

 

@fair ,

One last point,

For higher frequencies, at 44.1 KHz sampling rate, this may correspond to only a few samples. The shape of a quickly changing signal can’t be faithfully captured by such small number of samples.

This demonstrates, succinctly, a lack of understanding of the topic we are discussing. Perhaps the problem is you only see a few samples. What you should see is a few samples, taken at very precise time intervals. The information is not only the sample value, but the exact relative time of each sample. Both are critical.

If there is any interest, this is probably the best single article I have discovered that explains digital audio. It is not light reading nor heavy reading. Dan, who put it together obviously put a lot of time into it. It is almost 20 years old so comments about processing power are no longer relevant, but everything else is. I have come across many articles on digital audio written by less technical people. They get the basics right, but they often make mistakes and they never go into the depth that this article provides. You may need to read it 2 or 3 times to understand well enough, but if you do, it will dispel a lot of misconceptions about digital audio.

https://lavryengineering.com/pdfs/lavry-sampling-theory.pdf

 

If you have any questions about the article I will try to answer them.

@prosdds , there is value in a blind test, but there is no point doing that test for vinyl and CD. The noise, even the faintest clicks/pops, will quickly identify the vinyl. Even if it did not, you would need 1 of each mastered exactly the same. Does that exist? Is there a test record that matches a test CD?  After that you need a perfectly set up turntable. Nice to test to aspire too, practically impossible.

@asctim , @grislybutter , I think they are on the right track. @asctim has provided concrete differences between CD and vinyl. @grislybutter talks about learned responses or learned likes. If you tie your self worth to your likes and purchases, then you may be inclined to argue that those likes and purchases are inherently better, not just for you, but for everyone. Maybe it is just the mastering, but I am inclined to believe it is the flaws in the vinyl that often give it that special magic. Not always, not even 50% of the time for me, but when it works, it works really well. For me, it does not need to be superior technically for me to like it.

 

For discussion accuracy, vinyl, tape, and analog are not the same thing. Vinyl and tape are storage mediums that are predominantly analog in nature. What would perfect analog sound like? Digital!! :-)

If anyone cares my patience is officially at an end :-)

 

Nyquist is not a paradigm or a guess about how things work. It is not about engineering. It is a well understood, well researched, to this point not disproven mathematical theory. Nothing in the last 30 years has changed that. How it is applied is also well understood including translating real world limitations to accuracy. Those are not engineering principles, they are math principles.

However, my professors, from leading European universities, and their teaching assistants, had other opinions, giving me straight As on all courses related to Fourier Analysis and DSP.

Call me skeptical. No that is not right. I flat out don’t believe you are telling the truth.

 

Yet there is more, which came mostly from research conducted by others over past three decades. Unfortunately, too much of it is still widely dispersed in numerous peer-reviewed papers, rather than concentrated in a few engineering handbooks.

Well isn’t that convenient. Hate ta break it to ya but I learned all the theory in an applied math course. Because little of this has to do with engineering. It’s applied math.

 

This depends greatly on the nature of the band-limiting filter used.

That is totally irrelevant to what you replied to.

 

Implemented with analog circuitry, yes. But, at some point inside a class-D amp analog signal is transformed into a sequence of discrete +V and -V segments, starting and ending at analog time boundaries.

Wrong. Very wrong.

 

 

for some mysterious reasons, want to use digital formats providing higher information density.

For processing not delivery just like I said

 

(e.g. I own several nice 32/768 ADC/DACs),

No such thing as a 32 bit ADC or DAC. Purely a data standard ... And a marketing ploy. Maybe you will find some rounding errors in those bottom 8-9 bits.

 

By the way, 0.1% THD corresponds to a width of one pixel on a typical laptop display, if a graph like that is enlarged to fill the whole screen.

I am only storing that last paragraph for posterity.

 

Shhhh don’t tell anyone, but vinyl is terrible at <20hz and pretty awful in practice >20khz .... But I can’t hear it so I don’t care

@johnread57 ,

 

I did not present an opinion. I presented verifiable, researched, well understood, mathematical facts. Facts not disputed by those with the deepest understanding of the underlying math, and those able to adapt the math to practical implementation.

 

Below is an opinion. It misinterprets personal opinion, narrow market popularity, and different to "something". That something is only described in easily falsified claims, falsified with math, not an appeal to narrow market popularity.

Perhaps @Fair, can enlighten with at least 2 or 3 of these research papers he claims are hard to find? A new paradigm with 3 decades of research that legitimately calls into question all current signal processing and hearing knowledge should have many available sources to reference.

 

I see it differently. The old paradigm is falsified by phenomena for which it gives invalid predictions. For instance, according to the old paradigm, LPs shall be long gone, the way of cassette tape recorders and VCR video tapes. Yet LPs persisted, and the classic paradigm produces no convincing explanation as to why.

Perhaps @fair , can enlighten with at least 2 or 3 of these research papers he claims are hard to find? A new paradigm with 3 decades of research that legitimately calls into question all current signal processing and hearing knowledge should have many available sources to reference.

 

Still I wait for this. A meta analysis of purely digital sources, some too old to be relevant due to hardware limitations and others with experimental flaws, does not support your hypothesis let alone suggest there is any new paradigm. I do appreciate the repartee as it demonstrates the vinyl argument.

 

This is just like the tube discussion.  Even though there are significant, identifiable differences between typical tube amplifiers and SS amplifiers with good design practices, differences that are highly audible, every discussion devolves into a debate between those who point out those differences and those who believe in some unseen, unmeasurable property that "must" exist.

 

@johnread57 , I was out a few messages ago.

 

I'm telling you that the errors are signal dependent, you are quoting this, and then are asking me to tell you exactly how big the errors are? 

You know, serious people run long simulations to answer this question for specific digitization schemes and sets of representative signals.

 

To quote @cleeds these are fanciful imaginings, not serious discussion points. Serious discussion points would come with serious analysis or serious links showing that analysis. 

@johnread57 , I came across this NPR interview again, has some good comments on this topic. I am going to copy some and a link to the whole article.

https://www.npr.org/2012/02/10/146697658/why-vinyl-sounds-better-than-cd-or-not

 

METCALFE: Well, I think it has a lot to do with the fact that I'm primarily a recording engineer, as far as working with music. And it's - the closer thing to what I'm sending into the recorder is very much what I'm getting back out. With analog formats, although the sound can be very pleasing in certain styles, it's definitely imparting its own sound on it. And I think, to an extent, it's that sound that some people are really drawn to. But it's nice as an engineer to have the confidence of knowing that what I'm putting into - in most cases these days, the computer - is pretty close to what I'm going to get out.

 

OLIVE: Well, I mean, I grew up listening to records up until about '85, when the CD was already out. And I was involved in testing loudspeakers up at the National Research Council in Canada. And we were testing cartridges at that time, and it was quite apparent that the amount of distortion coming out of these devices was very high compared to CD. So what we found was that vinyl was a limiting factor in our ability to do accurate and reliable listening tests on loudspeakers, and we had to find a more reliable and more accurate medium.

 

 

@johnread57, this is not at all about vinyl, but about analog tape recording from the perspective of a recording engineer. There are some great sound bites to listen to. Summary: Digital comes out just like the way you put it in. Analog recording does not sound like what you put in. It may sound better. It may sound worse. This guy has 9 albums nominated for Grammy’s. He is not a hack.  Vinyl has a lot more distortion than digital and it rises with frequency. Add this to the qualities of vinyl that others has posted here.

http://recordinghacks.com/2013/01/26/analog-tape-vs-digital/


Here is another from a musician (60 recorded works) and recording engineer (100 albums)

https://aestheticsforbirds.com/2021/04/07/an-audio-professionals-take-on-vinyl/

There is MOFI, and this, http://drewdaniels.com/audible.pdf

The test results show that the CD-quality A/D/A loop was undetectable at normal-to-loud listening levels, by any of the subjects, on any of the playback systems.

 

Step 1, accept that properly implemented digital, even CD quality, has no sound or so little to be ignored. Step 2, accept that vinyl and other analog formats have a particular sound, or many sounds, and that we like it because it has those sounds that appeal to us, as living, breathing humans. Step 3, figure out what those sounds are and encourage audio companies to work on them and recording and mixing engineers to make better use of them. Step 4, accept that we are mostly a group of old buggers, and maybe young people don’t like the same thing. I tried to research more on this last item but I didn’t find a lot of work on it.

 

 

@cleeds  I included the weasel word little. I linked examples where analog sources were passed through a digital chain of CD quality and the listeners were not able to detect it. I could waste hours on the web finding examples of suitable implemented testing comparing CD quality audio to hi-res audio that would conclude no detectable difference. Can we agree that if there is a difference between CD quality done right and high resolution that the difference is very small, and hard or very hard to detect?

Play any example of vinyl ever made and the closest CD and everyone will be able to tell them apart. Maybe you will find some obscure set where that is not true. Can we say 99.9% of them?

Take the last two paragraphs and put them together. The difference between vinyl and CD is bigger, much bigger than CD and high res.  I am working from the assumption that high resolution digital is good enough to be perfect. Two-four times the bandwidth of CD, 20db or more of added dynamic range and hard to tell the difference from CD. I would say it is near perfect.

I did not say what that particular sound is. Cross-talk was mentioned. No matter what you do, that is there. When you are getting to the inner grooves there is unavoidable distortion. I am not up on the latest in vinyl, but my memory says the best distortion from vinyl, especially at high frequencies is several magnitudes higher than even CD. Maybe it is a combination of the cross-talk and the mastering, and nothing else?  Maybe my turntable setup that I think has a flat frequency response does not?

 

Every vinyl versus digital argument seems to devolve into an attempt to find some mysterious flaw with digital that cannot be supported with math, engineering, nor experiment. Maybe there is some flaw at CD quality that we can possibly detect. If there is, it is very small. The differences between CD and vinyl are not small. Some progress in understanding would be nice. It is not going to happen by starting with an unjustified conclusion and working back.

@cleeds ,

I did not mean you with respect to "digital confusion".

I do think vinyl done well, excellent pressing (clean), good turntable, good cartridge, all properly setup sounds very good. I don't think you need to spend $50K either. $10K maybe. I also think if you listen to that side by side with CD, you will always be able to differentiate them, even if its the slightest tick. Anytime I have been in situations where they are compared side by side, they are always different. I won't claim the level matching was perfect. If the vinyl frequency response was not flat, I am not sure that is possible.

I am open to it being just mastering, mastering and cross-talk, maybe my FR is not as flat as I think it is, etc. 

I think my only point, at this point, is that for people who have a vinyl preference, there are simple and probably obvious reasons we could find if we looked closely at their system or the music they listen to.  You say you are an analog guy, but you have not commented on your preference or thoughts about why?

We have internally used a blended metric similar to the Gm Gedlee metric for quite a few years. I know at least 2 of our competitors do as well. I agree on his basic premise that mechanical systems (speaker drivers) do not have as much higher order distortion, but on a practical basis, with multiple drivers, how distortion presents is more complex. One missing element is frequency weighting.

In one of the links, 2nd order distortion was described as euphonic. That may be true at a high enough level. Using the Gedlee metric, at the levels indicated, it would inaudible or close enough. Almost any DAC today would have a Gm close to 0. Not just expensive ones, all of them.

The vinyl vs CD article talks a lot about vinyl, not much about CD. One small, near useless section. I lost confidence in the author over a few items. Changing the load on an MC cartridge (135 ohms), from100pF to 200pF reduced the 3rd order distortion? I have done enough DIY electronics to know that smelled funny. That would change an RC filter from 12MHz to 6MHz. I say bad amp, bad switch setting, or multiple plays. Raises an issue with phase in filters on CD while ignoring that vinyl likely has phase shift too. Show digital samples that look bad, say it needs a reconstruction filter, then don’t show one? Bio attached to the article. Really impressive! He must be an expert? BS-Zoology, BS-Psychology, MS-Physiological Psychology, Ph.D.-Neuroscience. Welcome to 2022, everyone is an expert.

Thank you @johnread57 . I don’t normally put mich stock in his videos but he does have access to the process. He seems to support my claim above that vinyl is noticeably different no matter how perfect the vinyl it is still changing the sound.

@johnread57 , by posting that link, is that not an admission that all the differences are due to the recording and flaws in the format?