My answer to this is NO, not obviously better but …
To me a hi-rez file reflects more TLC taken in the recording/processing and I suspect that this is more responsible for a better sounding result than any magical mix of high numbers. Various older items, reformatted do sound cleaner but against that, I’m often surprised to discover a very nice sounding album is merely 16/44.
Hi-rez files do hog disc space although hard drive prices are very reasonable these days so I use that excuse to choose to download these bigger files if there is a choice. That said most newer recent releases are very very good so hi-rez is a bit redundant for them.
As others already said, you always need TLC for good mastering, regardless of format. As far as I know, recording and post-processing is always done in hi-res, so the format artists and sound engineers release in is no indication of the kind of creative tools they used.
After a lot of listening over a few years now i came to the conclusion that hi res (above CD quality) sounds no different. To me that is. I am sure there are some sharp ears that can discern. I can’t.
For some tracks, i can’t tell the difference between CD quality and MP3 .
Vinyl is still my go to option for the best sound quality.
We should interpret high-res as being a better audio container. It’d take really high end recording equipment and mastering where the audio has so much actual quality information that storing in a high-res container makes a difference to storing in a low-res container.
A respected sound engineer said they and many other professionals can only hear up to 96kHz in high-res, but that’s really pushing the limit of recording quality. Probably they needed high end and very analytical studio monitors to do this experiement.
If you do photography it’s similar to high mega-pixels vs sensor size. An image can have low mega-pixels but if it were taken by a medium format camera, the image will be shockingly clear and vivid. Many phones have high mega-pixels but still small sensor size, so the images are not anything special despite an industry leading mega-pixels specification. The key to image quality is a camera’s sensor, not its mega-pixels.
hahahaha misinterpretation. Wait lemme check this and write what I meant exactly. They are calling it 96 kHz sample rate. I really don’t know how to tell it.
I understand it might be difficult to believe that high-res might make a difference but I should share a story that was told by Bowers & Wilkins, and go ahead to call them a fraud if it feels better to do that. I don’t own their speakers, or am invested in them in any way.
During the time when B&W was developing a speakers for Abbey Road Studios, the sound engineers reported to the speaker designers that they heard some noise in a track that weren’t supposed to be there. The speaker designers from B&W were saying it’s probably caused by something else outside of the studio or something, but when the Abbey Road Studios engineer played back the track to the speaker designers, and were able to recreate the sound each time. After some investigation, they found that their latest iteration of speakers were so transparent that it was revealing unintended flaws in their old recording that were never heard with their previous speakers.
This was the B&W 800 series speakers and they are fairly cutting edge throughout its lifetime. Just that there’s been an improvement to it recently. I understand that high-end vinyl weren’t that amazing before the 2000s period which gave rise to CD, it was just recently that we were able to play out the vinyl tracks in much higher fidelity than before.
96kHz sample rate (or fs = 96kS/s) is good enough to reproduce frequencies up to 48kHz (fs/2) which is way beyond the audible range.
TL;DR: In general I believe that, all other things being equal there will be a sampling frequency beyond which no benefit will be seen. That sampling frequency will almost certainly be less than 96kS/s. I would personally think that it is far closer to 44.1kS/s for the vast majority of us.
In general for a given sampling rate (fs), a digital to analogue conversion will re-create the original analoge (audio) signal for frequencies up to fs/2 but it will also add reconstruction artifacts at frequencies above fs/2. These reconstruction artifacts, will, if allowed to propagate through your amplification stages, do your speakers or headphones no good whatsoever (but you wont be able to hear anything until your speakers fail). As a consequence, after the actual digital to analogue conversion, there is always an analogue low pass filter present in a DAC to remove these unwanted artifacts.
For red book CD (fs = 44.1kS/s), this analogue low pass filter has to have an exceptionally sharp cutoff frequency in order to preserve all of the audible signal. Thus, if we accept that the limit of human hearing is 20kHz (although that is too high for most of us), a perfect filter for red book would have to go from pass at 20kHz to 100% rejection in the region above 22.05kHz. This sounds like it might be quite a lot of room (2.05kHz) - but in actuallity this is just one tone of the western musical scale (in other words the difference between an A and a B on the musical scale or the notes produced by two white keys on a piano keyboard that are separated by a single black key) which makes this a very demanding filter requirement indeed.
In practise, the filter used will not be perfect and will, itself, introduce one or more of the following artifacts:
Start cutting off frequencies that are in the (extreme high) audible range - at least for those of use with very good and very young ears.
Allow some reconstruction artifacts (frequencies higher than fs/2) to propate.
Have a non-flat frequency response in the audible range.
Have a non-linear phase response (meaning that different signal frequencies get delayed by different amounts).
It is a matter of debate as to whether all of these filter artifacts will be audible. It would certainly be possible to imagine (1) and (3) being audible for some people. I think the consensus is that (2) would not be audible. The audibility of (4) is very uncertain - my knowledge in this area is a bit thin but I believe, in general, human hearing is not sensitive to phase but there may be some secondary affects related to the presence of multiple frequencies that could conceivably be audible.
However, if you use fs = 96kS/s, then the reconstruction artifacts (which start from fs/2 = 48kHz) is a long way (> 1 octave) away from any audible frequency. Thus an analogue low pass filter with a much gentler cutoff can be used and this allows hardware designers to choose filters with much better characteristics in the audible range (more appropriate cutoff frequency, better [flatter] frequency response and more linear phase response).
In principle, higher still sampling rates (fs = 196kS/s and beyond) gives even more lattitude to the hardware designer to design a very good analogue filter - but as with everything it is a law of diminishing returns both in terms of the hardware designers ability to design better filters and the listeners ability to hear the difference.
As a consequence of this, I can believe that it may be possible for some people to tell the difference between RedBook CD and 96kS/s sampling rates. However, even 48kS/s gives more room for filter design and so it would be quite possible for those that can hear the difference between 44.1kS/s and 96kS/s could not so easily tell the difference between 48kS/s and 96kS/s.
Edit: Repositioned TLDR
Edit 2: Removed references to quantisation noise/artifacts and replaced with ‘reconstruction artifacts’ to address the point made by @WiWavelength below.
44.100kS/s (44100S/s) divided by 2 gives 22050Hz or 22.05kHz. Thus the difference between this and the assumed 20kHz limit of audibility is 2.05kHz. It is in this 2.05kHz region that the filter has to, ideally, attenuate the signal to inaudible levels.
Yes you are correct although quantisation noise still has frequency components at (2N+1)fs/2 (for N >= 0) which must be removed although they naturally attenuate as you go higher in frequency.
In fact, if you look very carefully, you will spot that I originally started talking about frequency aliasing and then changed to simplify things - but not entirely successfully. There is still one place where the term ‘signal images’ is used
I tried to use the looser term ‘quantisation artifacts’ which Intended to include both aliasing and quantisation noise (which still crept in by mistake). ‘Digitisation artifacts’ would have been an even better terminology.
from 43:18 onwards. It’s a story that I’d like to tell others, so do watch it if you able to.
