To the OP, I would argue against the definition of high-resolution music being expanded to the recording equipment or quality - not because I disagree with the OP's overall view, but rather because I agree with the OP.
To me, "high-resolution" applied to music is a digital concept. It clearly refers to increased resolution aka precision of sampling, via higher sample rates and longer digital words (aka higher bit depth).
The problem with this definition is that it is based on the fallacious "stair-step" concept of digital sampling, which mistakenly thinks (or acts as if it thinks) that sampling any given frequency more than twice will somehow make the sample more "accurate" and therefore increase the "sharpness" or "musical resolution."
This is a false premise, and so expanding that false-premise definition to take into account the frequency response capabilities of microphones and other recording equipment seems like the wrong way to remedy the problem with our concept of "high-res."
Which brings me to Sergei's point, below, which I think exemplifies the flawed assumptions that drive the high-res discussion. Nyquist-Shannon sampling theory remains correct - you need only sample a frequency twice in order to properly encode it digitally and decode it back to analogue. Increasing the sample rate does not reduce distortion.
A finite sample will indeed always produce quantization error, and yes, any "error" in recording/encoding a musical signal can technically be considered to be distortion. But digital quantization error comes from bit depth, not sample rate, and it manifests as noise, not distortion - and crucially, that noise is at an extremely low level; it does not rise to the level of obvious audibility the way that, say, an amplifier's or speaker's distortion might do at high volume or at certain frequencies, or the way a stylus and cartridge might transmit sibilant distortion when tracking certain records, or the way a recording might have distortion from microphone overload.
With dither, the quantization error noise of a digital recording is easily randomized sufficiently that its perceived audibility is even lower; and with noise-shaped dither, it's transformed into something even less perceivable.
If this were not true, then SACD/DSD would be impossible: DSD has a native bit-depth of 1, and a native noise floor of only -6dB! With noise-shaping dither, though, the original analogue signal can be accurately reconstructed from the 1-bit digital samples, so much so that DSD is considered not only high fidelity but also high-resolution.
When it comes to sample rate, any sample rate more than twice that of the highest frequency that needs to be recorded is unnecessary and does not add anything to the recording or to the resulting playback of the decoded analogue signal.
As a practical matter, of course we want a sample rate that is somewhat more than the highest desired recording frequency, so there is headroom to implement digital filtering to prevent aliasing from frequencies in the original source that are higher than the sample rate can capture - this is why the CD standard has a sample rate higher than 40kHz (2x20kHz). But while a higher sample rate provides more room for the antialiasing filter to work, the higher sample rate does not capture the audible-range frequencies any better than the lower sample rate does: 44.1kHz samples frequencies up to 20kHz exactly as accurately as 192kHz does. This is not my listening opinion - this is what digital sampling theory predicts, and that theory is mathematically and empirically verified.
So the definition of high-resolution audio is not so much flawed in and of itself - rather, it's a pretty accurate definition. It's just that what it defines is not especially meaningful when it comes to determining audio quality.
You are probably aware of a rather long thread discussing, among other things, the technical aspects of hi-res:
https://www.audiosciencereview.com/...qa-creator-bob-stuart-answers-questions.7623/.
I narrowed down that thread to another one, and then this newer thread to a single post, which I believe captured the core issue:
https://www.audiosciencereview.com/...higher-sampling-rates.7939/page-2#post-194272
IMHO, the core issue is that, formally speaking, a limited-duration non-perfectly-periodic piece of music can't be Fourier-transformed to a finite spectrum representation, and thus the
Nyquist–Shannon Sampling Theorem, which relies upon Fourier transform, strictly speaking is not applicable to music.
Thus, we must assume that a sampled digital representation of real-life music contains distortions, as compared to the original analog sound. Mathematically, increasing the sample rate and number of bits per sample makes this distortion lower.
Most of the time listeners can't perceive the difference of distortions between 44/16 and, say, 192/24. Sometimes they can:
A Meta-Analysis of High Resolution Audio Perceptual Evaluation.
The 5% figure keeps coming up. Like: the difference can only be perceived by 5% of listeners, on 5% of music. A naive approach is to multiply these probabilities: the thinking goes that on average ~0.25% of listening sessions would be affected by the regular vs high fidelity differences.
But, and there is a big but! For a particular listener in the 5%, whose favorite music genres happen to fall in the 5% too, the number of affected listening sessions can be much higher, closer to 100% actually.
And vice versa, for a listener in the 95%, or for a listener whose favorite music genres happen to be in the 95%, the advantages of hi-res are immaterial. For them, the promise of hi-res is 100% snake oil.