My intent is not to prove anything. Rather, to illustrate why I believe that the hi-res vs CD perception depends a lot on the music genre, on how a particular music piece was captured and mixed, on the listener, and on many other parameters.
To illustrate the illustration of the point. A puzzle by Lewis Carrol (paraphrasing):
- You need to walk to a train station to catch a train departing in 2 hours.
- The distance to the train station is 5 miles.
- You walk with the speed of 3 miles per hour.
- Will you catch the train?
The straightforward answer is: "Yes."
Lewis Carrol retorts: "But what if on the way to the station, a mad bull starts chasing you?"
Morale of the story: the correct answer is "I don't know".
So, the correct answer to the "Is there an audible difference between ... ?", being hotly debated on the forums like this one, is, in most cases, "I don't know", as it has many important parameters under-specified.
Once you concretize the question with the music, equipment, person, and person't physiological condition, then you may get to a more concrete answer. Averaging these answers over many variations of music, equipment, person, and trials, will get you a general answer. If that's what you are after, great. If you are interested in the answer regarding concretely your music and your equipment - you'd need to experiment on yourself.
There are two major uses of splines in audio time domain processing I know of:
(A) Extrapolating between samples during upsampling.
(B) Using them as basis functions for sampling and reconstruction of analog signal.
(A) is not actually the best interpolation approach. It is fast computationally, and produces nice smooth upsampled graphs pleasing the eye, yet it introduces distortions that the sinc-based interpolation and Fourier-based upsampling don't.
(B) is theoretically attractive because it allows to vary between the most accurate representation of the resulting analog output in frequency domain and its most accurate representation in the time domain, by changing the spline order parameter, while getting rid of the infinite character of sinc-based reconstruction.
So, I could imagine using different spline orders for recording an opera singer solo vs a death metal concert. However, (B) is computationally challenging, and requires either using a proprietary end-to-end system (thus interest of MQA creators I guess) or a new set of open standards. IMHO, the latter is not going to happen any time soon - PCM is very convenient and too entrenched.
You still haven't grasped the simple fact that microphones used in 99.999999999999999999999999999999999999999999999999999999999% of recordings simply don't capture anything any where near 96 kHz bandwidth. Specialised mics are required to do this. Your speakers can't reproduce up to 96kHz bandwidth. You can't hear 96kHz bandwidth .This appears to be the case, for me, in the context of the audio systems I had access to, and genres of music I ever cared about. 192/24 is 100% for me, to the best of my knowledge, as of today.
I'd think so too, if all the tracks I cared about, including Gamelan and Prog Rock (I think I could survive without Mariachi for a while ), were available in 192/24. But they aren't Perhaps that's why I'm so passionate about this whole Hi-Res thing: I want more music in 192/24!
I don't mean to be rude, but you are struggling here on 32+ pages of posts to prove something that nobody proved to hear. Let me quote @JJB70 's post again as IMHO that one hits the spot perfectly:
"My honest opinion is that if differences are so marginal that there is a debate over whether they are audible and even if real can only be discerned by a trained listener under controlled conditions (even there with less than a 100% success rate in DBT) then that tells its own story. That story being that it is all meaningless in the real world of using hifi and carrier media to just enjoy music. "
You made a statement without reading all the pages. The proofs were provided.
I don't argue with your opinion. I just said it couple posts ago. However, the trained listeners have other opinions, also worthy of consideration.
You made a statement without reading all the pages. The proofs were provided.
I don't argue with your opinion. I just said it couple posts ago. However, the trained listeners have other opinions, also worthy of consideration.
You made a statement without reading all the pages. The proofs were provided.
However, the trained listeners have other opinions, also worthy of consideration.
No, even "odd-shaped" signals are accurately captured at 44.1 kHz if their bandwidth is less than 22 kHz.44.1/16 PCM would fail to encode correctly some odd-shaped signals in the 11-22kHz range
No, even "odd-shaped" signals are accurately captured at 44.1 kHz if their bandwidth is less than 22 kHz.
3 sound mixers mixed differently. Wow, not exactly surprising. Again, why are all your hints so terribly indirect? If it weren't so small we'd never have to grope about for such flimsy explanations of something really being there. Much more direct and clear differences would be easily found.You have no idea We went as far as recording live performances with three independent sets of ADC boxes: at 48/24, 96/24, and 192/24; and giving the tracks (numbering between 12 and 14) to three different mixing engineers to produce mixes they liked. Also doing downsampling of the resulting mixes from 192/24 to lower resolutions. Also converting them to MP3 and AAC.
The three mixing engineers had quite different preferences for the mixes. One of them would de-emphasize high-frequencies, and in general didn't like transients. Another one liked "crisp" and "punchy" mixes. The third one was somewhere in between. Mixes of the first engineer tended to translate well all the way down to 48/16, as they were "mellow" to start with. MP3 and AAC of these mixes sounded OK too.
Mixes of the second engineer not so much: that's where I could hear the differences between the 192/24 and downsampled 48/24. Interestingly enough, I personally couldn't hear differences of 192/24 vs 96/24 and 96/24 vs 48/24, so it was "two steps down" that would reveal the differences to me. You are right, the differences were small.
And as I mentioned before, for simple music, such as a solo vocalist accompanied by a solo acoustic guitar, there were no differences. The mixes of the three engineers sounded virtually identical, between each other, and at all sampling rates.
Well, I must have missed them, do you care to make a short recap?
As English is not my prime language I will quote again:
"You still haven't grasped the simple fact that microphones used in 99.999999999999999999999999999999999999999999999999999999999% of recordings simply don't capture anything any where near 96 kHz bandwidth. Specialised mics are required to do this. Your speakers can't reproduce up to 96kHz bandwidth. You can't hear 96kHz bandwidth ."
So, your trained listeners are from Krypton, special mikes were used to record up to 96kHz and also special speakers that can reproduce all that bandwidth. Did I get it right?
The simple fact is that a physical event that only lasts 1 microsecond, such as a nuclear bomb explosion, can be heard, because it puts in motion a chain of physical phenomena, which eventually create a sensation of sound. You don't need a hearing frequency range up to 1,000,000 Hz to hear a nuclear explosion, or a sound of lightning, or a supersonic shockwave, or a sharp transient generated by cymbal/xylophone/gamelan.
«The Hitchhiker's Guide to the Galaxy notes that Disaster Area, a plutonium rock band from the Gagrakacka Mind Zones, are generally held to be not only the loudest rock band in the Galaxy, but in fact the loudest noise of any kind at all.»Why are you comparing a nuclear bomb explosion to music replay in the home?
Not the beautiful imagery of Louis Carol, but consider this: In perceptual matters you are never going to draw a fine sharp line. Just as an alternative: how much weight can a human dead lift? We know it isn't 5000 lbs. We know it isn't 1000 lbs. Whoops, wait a minute there have been a couple do 1000 pounds. Okay, but an edge case. Seems out of 7,000,000,000 people a couple can do 1000 lbs. Most can't do half that. An average might be 1.5 times body weight for healthy people.
Now we have good info on the variability in hearing. It isn't nearly so wide a range as dead lift capability.
But as pointed out, microphones, speakers, hearing, conditions of hearing etc etc, the idea you need high, high sample rates is just bonkers for enjoyable music.
Instruments that put out way up in the high ultrasonic range are going to have that sound absorbed by air in a few extra meters.
If we assume that most people use their stereo systems to listen to music at a quality level of a CD (16 bit 44.1Khz determined by the good people of Phillips and Sony, taking the Nyquist sampling rate in account) it should be clear that 20 Khz is the very upper limit of frequencies one will ever
experience.
I'll include once again the chart showing the frequency range of musical instruments indicating frequencies between 30hz and 14Khz
except perhaps a church organ that can go as low as 18hz (average).
Expecting to hear frequencies in music above say 18Khz is unrealistic.