What really happened i'm reading so many comments wishing well to Paul but i have no clue at all ?

@@Badassvidsz type in google - Boulder, Colorado shooting. PS audio resides in that city, as well as Paul himself and his family. I think that there's been some Ask Paul videos from that mall but I'm not sure.

@@freekwo7772 Thank you i'll give it a try .

Exactly.

@Matthias Keysermann: You still can't. "High resolution" audio is only a marketing term, a gimmick. It doesn't exist.

@@Historia.Magistra.Vitae. Paul provided a very good explanation for having high res audio. He did not claim humans could hear above 20k, neither will I. It's about the filters and I can respond adequately now.

@@mkeysermann : You should watch and listen some actual audio engineers like this guy ( kzfaq.info/get/bejne/aap6gM2Q25PNYI0.html ) instead of the snake oil salesman Paul.

Lijepo ste to napisali Ante!

the dynamic range is almost irrelevant for playback. 16bit gives you something like 96dB of dynamic range, not ideal, but not bad. 24bit is really excessive for playback, but it is invaluable for recording, the extra headroom is a MUST. The biggest problem with lower bit depths IMO is simply the potential for quantisation errors when going from 24bit down to 16bit.

that is my understanding as well.. human audible spectrum is 20Hz to 20kHz and you typically need to sample at multiple times of the audible signal to accurately capture all the nuances of the actual waveform for processing and playback. Sampling at 192kHz means that capturing the waveform of 20kHz part of the spectrum gives you approx 10samples (9.5samples). This video should have mentioned sampling not just say Nyquist and signal phase

No, you're not wrong and with all respect. Sample rate rate is a speed of sampling when is recorded you can keep it, while output measurment does not have (in this theory) any connection, "same language" or however you call it, together. You're completely right!

Sample rate has NOTHING to do with frequency, it's just how often a snapshot is taken measured/recorded per second. The higher the rate the more likely is your music sounding better if your setup is up to reproduce it.

Do you mean 192kbit/s? I.e. not hertz but kilo bits per second.

@@johan.mp4 No, hi Res 192kHz files can be several thousand kilobits per second. I've got Pet Sounds in hi Res and they are something like 6,000 kbps , huge files.

Please take a little time and search the academic credentials of the major industry figure which you refer to.

@@IsmaelMartinezPR Why?

@@RobWhittlestone Respectfully, this is what I found about Mr. McGowan. I quote: "I am a self-taught analog electronics engineer and have designed most of the analog product built by the company". I will reserve my assessment of such statement. However, there are other resources where technically trained professionals address this same question available. Best.

For a deeper dive, the fabfilter people published a very good video on the topic a year or so ago. It’s called Samplerates: the higher the better right? (Is here on YT). It is very illuminating.

@@lindsaywebb1904 I can refer you to Amir Majidimehr channel a video called Antonio Forcione looking into files of Hi-Res Audio.

Do you refer to the new GaN FETs?

@@VladoT yes. :-)

Indeed so. I'm surprised Paul confused the two.

That’s exactly I was thinking

That's simply not true. Let me try and explain. A 44.1kHz sampling rate is bandwidth limited to 20kHz (half the sampling frequency). 192kHz is similarly bandwidth limited to 96kHz. Now, having said that, obviously we cannot hear 96kHz, but that was never my point. What we want is to have a gentle filter so that we don't suffer audible phase shift in the music. That cannot be done with lower sample rates but it can with a higher sample rate. Put another way, if we were dealing with strictly analog circuitry, a preamp that was bandwidth limited to 20kHz would sound rolled off because of the affects of phase shift that extend down well into the audible frequency range. An analog preamp with a bandwidth of 50kHz will sound much more open and less limited because of this. That's the difference - and higher sampling rates allow for wider bandwidth in the audio band.

@@Paulmcgowanpsaudio Paul, thanks for replying. This means that all the information I have read elsewhere online about sampling frequencies is wrong! Thanks for your ever boundless knowledge.

@@davidtomsett Hard to say if it was wrong but I think in general there's a bit of misunderstanding out there. The higher the sample rate the higher the possible bandwidth of the audio signal. What the recording engineer decides to after that is up to them.

FINALLY!!

Finally

yes , our ears are just part of our auditory/ scanning (lol) system. The hairs on our skin pick up much higher frequencies.

Just because a speaker can go above 20KHZ does not mean it is highly resolving , Magnepans and speakers with electrostatic or ribbon or AMT (mid and high frequency drivers) are way more resolving than most other types of speakers.

@ReaktorLeak No kidding! It's like some kind of pathology.

Exactly!

Good recordings are what we are all looking for. A poorly recorded piece of music will sound like a turd even if that recording is recorded at the best possible bit rate. You just end up with a very pricisely and shiney turd. A great recording can sound fantastic, even on a casette tape (some of my best master material originated on a simple 4 track cassette). However, re master a great recording at 24 192 studio master quality, and you can really reap the rewards if tge mastering is done correctly. Ive recently moved to Roon as my front end, so now i have access to Tidals mqa studio masters. Some of these are trully fantastic, but some of them are not sympathetically mastered and decenly masterd cd quality version will sound better despite the bit rate. Good recordings can sound better with a more precise digital version at higher sampling rates, but a turd will always sound like a turd no matter how many bits were used to digitise it.

@Kim Atong: Wrong. Digital audio and sampling does not work that way. The greater the sampling rate, the greater the potential frequency response since higher sampling rates allow higher frequencies to be recorded and played back. Has nothing to do with fidelity whatsoever.

120 Khz?

Maybe it's for the bats?

the issue is when you record you cant let anything above the nyquist frequency hit the A/D converter in the first place, hence the need for an aggressive low pass filter right at the nyquist frequency when using the standard, relatively low sample rates.

@@PAPO1990 Yes I know, but your missing the point that the majority of the filtering is now done digitally and that greatly reduces the discrete analog filtering needed.

@@chrisharper2658 you seem to be missing the fact that the filtering NEEDS to happen BEFORE the signal ever hits anything digital. I literally got a diploma in sound engineering studying this stuff, you can’t process it digitally if you haven’t digitized the signal yet, and you CAN NOT digitize the signal if you don’t filter out EVERYTHING above the nyquist frequency

@@PAPO1990 But your talking about recording and the subject is playback as in the DAC

@@chrisharper2658 but in recording, the damage would already be done. Plus, down-converting always results in rounding errors.

More bits and higher sample rates are used in the recording studio to give editing headroom but are pointless for playback.

@@geoff37s38 how is the headroom not defeated by 16 bit conversion?

@@graxjpg The bit length determines noise floor, this is all it does. 16 bits gives a noise floor below hearing threshold. This is why silent sections on a CD are perfectly silent. You cannot improve on perfection.

@@geoff37s38 thank you for your succinct responses!

@@graxjpg Just to clarify. 24 bits may be used by the recording engineer to ensure capture of full dynamic range, especially on a live performance where dynamic range is unknown before performance starts. 24 bits gives engineer room to capture loudest sound without risking horrible distortion on these peaks. The recording can then be put in a 16 bit frame which is more than enough for the finest playback equipment.

youll still _know_ when it happens.

@Rick C I’m feeling a bit better now. I had my hearing tested at a medical center last month and mine drops off at 12Khz. But then, I’m “only” 66.

@ReaktorLeak you can see the shannon nyquist theorem, you need infinite digital precision and we cannot have this in real world. It's fact

@ReaktorLeak he is saying exactly what I'm saying : we do not have infinite precision (we have a finite number of bits), so we cannot have 100% accuracy in conversion. The DA conversion add aditional noise because of that. If we respeted the shannon theorem, there should have 0 noise in DA conversion (which is the theorical 100% accurate conversion that cannot exist in real life) And to reduce the effect of quantization error, increasing the frequency is a solution. That's why I'm saying : "192 khz gives a better result thant 44 khz because the shannon theorem cannot be applied in real world, it's a theorical result" Now I hope you understand better how it works

@ReaktorLeak no a 8 bit digital does not allow it does not because the digital errors create digital noise, so the conversion cannot be perfect ;) And yes there is no such thing as perfect conversion because there is no digital signal with infinite bit depth. That's what i'm saying since the beginning of this thread ^^

@ReaktorLeak i don't say it's wrong, i say it's a simplification of the real world. As a lot of theorems are. And i say it again, the more bit you put, the closest it is from the infinite accuracy needed for the shannon theorem. A lesser bit means higher digital inaccuracy/errors and thus higher digital noise. It's quite simple. Which part don't you understand ^^ ?

@ReaktorLeak it needs infinite precision (no quantification error) Because if you have quantification error, you cannot reproduce exactly the signal. Because you will reproduce the signal with errors (which is noise and distorsion). It's the reason why we use dithering : to mitigate this. I say it again : to reproduce perfectly a analog signal, you need the digital signal with 2x freqmax, and with no quantification approximation, which means a inifinite level of precision (number of bits) ! It's basics We dont have infinite level of precision, it's impossible to reach.

Like the older Power Plants... ;-) I recapped my P-600, and wow, what a difference!

I have done AB comparisons between a CD and a MFSL half speed mastered vinyl (which technically is hi-res ) and the vinyl wins easily but that was on an expensive highly resolving system.

@@leekumiega9268 Vinyl is nowhere near "hi-res" - nowhere.

@@AntPDC Point #1 vinyl is capable of preproducing frequencies around 122 KHZ (see you tube anadialog sound above 20 KHZ ) , in show more the last link shows that MoFi was able to cut the lacquer at 122 KHZ but over heated the cutting head so it is limited below that but still Hi-Resolution. Point #2 the 70's quadraphonic records had the rear channels modulated on a 30 KHZ carrier with the frequency going up up to 45KHZ , and a good cartridge with a shibata or microline stylus could track that high, so vinyl is easily capable of Hi-Resolution. All of MoFi's vinyl records are in fact high resolution.

True.

@ReaktorLeak So maybe, just maybe, the "wider dynamic range" is why, subjectively and to my ears very obviously, there is "more detail?" Certainly with an analogue system an extended dynamic range reveals low-level details and ambiance etc. previously masked by the system noise floor. Maybe, just maybe, "a lot more bits a lot more often" allows for easier error correction (as there are more check bytes) and lower jitter? Thanks for the reply - enjoy your day and stay safe

@ReaktorLeak Probably not as it appears you've already made your mind up you won't. "There are none so blind as those who will not see (or hear!)" Stick to measurements and theory and leave the subjectivity to us loonies ;0)

@ReaktorLeak Firstly you may, for all I know, be partially deaf. The fact that I can easily and consistently recognise HiRes variants of familiar recordings doesn't necessarily mean the next man can and, I have to say, have literally a lifetime's critical listening experience under my belt. For a visual analogy a colour blind person could not judge a painting, for instance, the same as you or I and, in likelihood,would be incapable of the same analysis (and response to) colours and colour balance . Accurate, consistent measurements don't lie; I agree, but perhaps you're measuring the wrong things. A van Gough may measure the same as some art student's scrawl; same canvas size and weight, same paint consistency, similar number of brush strokes.... How do you measure a Miles Davis solo against an indifferent but competent player's? Same eight note scale, same acoustics, similar instruments, identical volumes..... How do you measure beauty and inventiveness? How do you measure Shakespeare against a schoolboy essay? Same word count, similar language, all words composed of the same 26 letters of the alphabet.... How do you measure the person you love against a similar human? Same weight, same colour eyes, same age, similar education and background......How do you measure great culinary cooking against a Big Mac? Same calorific value, similar breakdown of carbs and sugars, similar ingredients of wheat, meat and vegetable matter... How do you measure what one man hears against the other? In a very crude sense you can test auditory response in terms of volume and frequencies but how do you measure his interpretation and experience? How do you measure his social conditioning and emotional response? I'd happily go blindfold a/b 'ing against standard 16 bit / HiRes files. Would you accept a consistency of correctness as a "measurement?" "There are more things in heaven and earth, ReaktorLeak, Than are dreamt of in your philosophy" to paraphrase the Bard.(who,incidentally, is no greater writer than Harold Robbins. I've measured a sample each of their books and they both weigh the same with similar page counts and pictures on the cover. This proves conclusively that Shakespeare is no more substantial or influential writer than Robins as they both measure identically). At the end of the day we're just going to have to agree to disagree my friend and, in the great scheme of things, it really doesn't matter does it? I'll continue to get pleasure from my HiRes files and you continue to enjoy your stance which I actually envy as that way I wouldn't have the expense of replacing my existing Red Book recordings! Take care, stay safe and thanks for a lively and thought-provoking debate ;0)

@ReaktorLeakTTo quote Mr Zimmerman; "You go your way (and I'll go mine)"

@ReaktorLeak Just because you can't, or "have never met anyone who could" isn't a definitive argument; it just casts aspersions on your own dogmas, closed mind, abilities and those of your associates. I can't run a minute mile or walked on the moon nor have I ever met anyone who can but that doesn't mean people who have don't exist. You're perfectly entitled to your opinions; go off and fill your boots with my blessings but please remember they're nothing but that; opinions.

youll be amazed how much (i know i was) how much 'love' the BHK-pre will give your system. they give you an in home trial. what have you got to lose? oh... and dont get me started in the DirectStream DAC. _mind blowing_

@@googoo-gjoob No doubt PS Audio makes awesome stuff. I use the Stellar S300 in my dining room at the moment. But preamps are gone from my setups as I archived my vinyl discs and have gone 100% streaming.

Most of the time i am when im watching this guy talk

"I'm high alright, but not on false drugs. I'm high on the real thing: powerful gasoline, a clean windshield, and a shoe shine. Over." -- Pastor Rod Flash, a.k.a. The Firesign Theatre (1970) Or words to that effect.

@@stephensmith3111 yeaaa lets slonk juul gasoline and windshield cleaning liquid 🥴🤤

While that is sometimes just a marketing ploy other times is is better but you need a highly resolving system to hear the difference.

@@leekumiega9268 Are you speaking from experience? What system are you using?

@@AntPDC Yes I speak from experience , the system was my friends very expensive system ,the speakers used 4 electrostatic panels for the mid and high frequencies.

@@leekumiega9268 : The only way things can be better, is if they are mixed and mastered differently.

@ReaktorLeak Yes, I'd like to know that too.

@ReaktorLeak Looks like KZfaq may be stripping out my reply with links, so I'll just have to give you generic details - it's on Prostudiomasters in the US, and on Qobuz for most of the world. Qobuz gives you the benefit of multiple file formats and qualities, whereas PSM gives you just FLAC hi res. Just search for Solo: A Star Wars Story Deluxe

@ReaktorLeak Why on Earth would I want to do that? I already have the full 24-bit / 192 kHz complete album, as well as the CD quality 16-bit / 44.1 kHz, and when I play both files through my system I can tell the difference. I have already "judged" for myself. I don't need to listen to your adulterated files when I own the originals and can easily listen to them, without having been downsampled or otherwise manipulated. What a daft waste of your time and efforts. Oh, and you're welcome for the links by the way - sheesh.

@ReaktorLeak They are the exact same source - as you'll see from Qobuz, you can download in multiple file sizes/options - I downloaded both the high res files and the CD quality files. It's exactly the same source material.

@ReaktorLeak You seem to have missed my original point - I HAVE the 192/24 bit version and the 44.1/16-bit versions, both from Qobuz, both bought at the same time. I am able to do my own listening test, and have done so - played back through my Node 2i the difference between the 192 and the 44 is clear.

@PAPO1990 : If you understand how sampling works, then you should also understand why 16/44.1 or 48 is more than enough for the human ears.

@@Historia.Magistra.Vitae. If you really understand how sampling works, you'd know how wrong you are, especially once to take into account the aggressive low pass filters required to make things work while recording, and the phase problems those create. If you only have 2 samples per cycle on a wave, there is a good chance those samples won't line up perfectly with the peak and trough of the wave, creating errors. Add to that the staircase shaped digital waveform and you can SEE 44.1/48Khz isn't "more than enough" it's a bare minimum. As for 16 bit, it's ALMOST enough for playback, but it is BAD for recording. If you are recording you NEED 24bit, the extra headroom is invaluable because if there's even a TINY bit of digital clipping, it's going to sound ATROCIOUS. A little extra dynamic range is nice for playback, but I'll admit 24bit is overkill, HOWEVER, the conversion to a lower bit rate can lead to errors, yes, they are minor, but with storage and bandwidth being so plentiful these days, there is NO reason to down convert the bit rate just because you don't need the dynamic range.

@@PAPO1990 : Unfortunately you are incorrect. 2 samples is actually enough to sample a high frequency. Also there is no such thing as a "staircase shaped digital waveform". It doesn't exist. All signals with content entirely below the Nyquist frequency (half the sampling rate) are captured perfectly and completely by sampling. When it comes to bit depth, 16 bit is 96 dB worth of dynamic range and 120 dB if dither was used. There is no song nor album in existence that would come even close to having 96 dB worth of dynamic range. For example one of the most popular songs, PSY - Gangam Style, only has 8 bits. You are lucky if you can find a mainstream song that uses over 10 bits. The 16 bit noise floor is already below what we can hear however. For recording, sure, 24 bit is just fine. For playback, it's useless. I highly suggest you check out a video here on youtube called: "D/A and A/D | Digital Show and Tell" since that explains these things way better than I do.

@@Historia.Magistra.Vitae. I literally studied this stuff, everything below the nyquist frequency is not PREFECT. Dither is literally noise being added to the signal. And while most mainstream releases are compressed to use very little dynamic range, high quality classical and orchestral music often uses significantly more, even necessitating more than 96dB in some cases. As for stair casing. That is literally what the digital representation of the waveform becomes. The higher the sample rate, the finer the stair case.

@@PAPO1990 : Then you should ask a refund since they teached you bullshit. As long as we keep using a sampling rate that is double the desired frequency, we can perfectly capture and reproduce those frequencies. Simple as that. If you have an oscilloscope, you can actually test and prove it yourself. When it comes to dither, it is shaped noise and replaces the original noise produced by quantization. Again, there is no recorded music in existence that would come close to 96 dB or exceed it. Again, there is no such thing as "stair casing". Watch the video I told you about already, since otherwise you are just wasting your time talking nonsense.

To me that it more of an analogy for power amps. Perhaps a better analogy would be digital cameras. Say you wanted to crop an image. You'll get a higher quality picture if you crop a 10 megapixel image than a 5 megapixel image and then later make a large print. By having a higher resolution audio signal you are able to crop (filter) the extraneous audio signal and still have good sounding audio.

@@Commiefornia 👍

Maybe you should check the Wikipedia article about Nyquist theorem before posting. Or actually watch this video where this relationship is explained 🤦‍♂️

@@johan.mp4 dont worry about the theorem I learned it at school long time ago in my electronic degree. You clearly dont understand how it is apply to hi-res audio if you think that the goal is to get frequency over 20Khz, it is to get more precise audio reproduction of the signal, combine with higher bits it create an higher fidelity reproduction of the analog signal. Of course higher sampling will be able to record higher frequency, but most will be attenuate by the speakers, amplifier and internal dac filter.

@@DrLoveQc never wrote that was the only goal. It was you writing it has NOTHING to do with audio range. Chill.

@@DrLoveQc : You have no clue what you are talking about. Please go back to school or at least educate yourself about this topic before commenting.

@ReaktorLeak No one heard the high frequencies but some people could tell when the high frequency content was there or not and some reported what they could hear sounded better even though they could not actually hear the high frequencies .

That's why it has a closed comments section, ey? Though shall not question the gospel of CD PCM! P.S.: We can hear multiple concurrent soundwaves in the audible 20Hz-20kHz, likely very many, possibly even the entire audible range. Just like our eyes can see the entire visual spectrum at the same time, it may be possible that our ears and bodies could hear & perceive the entire audible spectrum at the same time. That's how hi-res audio makes sense. It's not about the bs misdirection of hear 100kHz soundwaves, but about perceiving many of the soundwaves in the 20Hz-20kHz at the same time.

Except that this has been done on multiple occasions, and published warts and all. As an example have a look at The Guardian, which did a test and published their findings way back in 2014, shortly after Neil Young launched his Pono tech: www.theguardian.com/technology/2014/aug/21/mp3-cd-24-bit-audio-music-hi-res Really would be better to do some research before claiming no journalist would "ever, ever agree to participate".

@@richardt3371 Thank you for your comments - even your acidulous concluding remark. I totally stand by my comment. The Guardian article you cited did not feature a live video of the tests they ran; nor were they conducted under double-blind AB listening conditions. No, the write-up was done post-facto, in close collaboration with the other participants (although it is fair to say that the conclusions reached lacked any apparent bias). This is a cost/benefit issue for prospective purchasers of audio equipment, including those whose enjoyment of their existing kit may well be undermined by hi-res marketing claims, thereby encouraging them to shell out on costly and probably quite unnecessary replacement gear. A live video featuring double-blind AB listening tests involving the industry and its audio journalist co-conspirators would assist in this process - especially when the outcome of 50/50 (ie a guess) can safely be predicted, judging by previously published tests conducted by the AES under controlled conditions. Best wishes.

@@AntPDC Good heavens. So they did not do it exactly as you wanted them to? Even though it was done in 2014, and was - as I stressed - just one example of audio tests carried out by journalists and their findings published? There was no bias in it, no axe to grind - it was a fair summation of the experiences of 3 music journalists. But apparently only if they do it double blind (which wouldn't work, and would be impossible to carry out, suggest you look into what double-blind would actually entail, as it would mean both participants AND those doing the test wouldn't know which was which, which is daft) will you consider their experience worthwhile. Son, you can't always get what you want.

@@richardt3371 : You just have no clue what you are talking about. Please go back to school or at least educate yourself about this topic before commenting.

Yep.

oversampling is exactly what he's talking about.

@@easystreetphoto2401 he said nothing about over sampling. His explanation was muddled and confusing.

@@geoff37s38 oversampling is literally the only thing he talked about.

@@easystreetphoto2401 oversampling is used to denote a process used in the reconstruction phase of digital-to-analog conversion, in which an intermediate high sampling rate is used between the digital input and the analogue output. Here, digital interpolation is used to add additional samples between recorded samples, thereby converting the data to a higher sample rate, a form of upsampling. When the resulting higher-rate samples are converted to analog, a less complex and less expensive analog reconstruction filter is required.

@@geoff37s38 yes. that's the one thing he talks about here.

Where it ends depends on how steep you can make a filter that doesn't significantly distort the music.What the Nyquist-Shannon theorem does is to define which frequencies contain the quantization noise. If you have a sampling frequency of 44kHz, then the quantization noise is at frequencies starting at 22kHz. It is very difficult to create filter that allows frequencies of 20kHz to pass unattenuated while blocking the noise above 22kHz that do not add their own distortion -- particularly phase distortion. However. If you sample at 192kHz, the noise is all above 96kHz. It is much easier to make a non-distorting filter in the gap between 20kHz and 96kHz.

@@timharig its hilarious that you consider filter that allow frequencies of 20KHZ to pass...lol why does it matter exactly... lol ?

@@taroistok8751 Because if you do not, you end up with the samples as a kind of square wave instead of the recreated analog wave form.

@@timharig nyquist explained this long ago..... at 44hz the human ears cant see the difference at all it perceived a analog wave of sound and cannot hears the step or know its squares, for frequency up to 20 khz. The problem is audiofool paying thosand of dollars for no reason..... When you go to the cinema do you look at a movie and think you see the individual frames? No your eyes cant, same with ears and high rez

@@taroistok8751 You are mis-reading the theorem. Unfortunately, most sources only state Shannon's part of the theorem about the Nyquist limit rather than actually showing the mathematics behind it. Like all mathematical theorems, the Nyquist-Shannon Sample Theorem has criteria that determine when it applies and when it does not. In the case of Nyquist-Shannon Sample Theorem, that if case specifies that the theorem ONLY works for bandlimited systems. This is why DAC's always have a low pass filter. It is referred to as a reconstruction filter, an anti-aliasing filter, or an anti-imaging filter: en.m.wikipedia.org/wiki/Reconstruction_filter The output of the DAC will not be a square wave or step. Once the signal goes through the low pass filter, all of the sampling noise above the Nyquist limit will be removed and a perfectly smooth recreation of the original sampled sound will be output. All this works because Nyquist part of the theorem isn't just about the Nyquist rate. What it really shows is how the sampling rate determines the higher order aliases that are produced by the sampling pulses. It shows that an infinite aliases occur at integer multiples of the sampling frequency in order to create the sharp sample boundaries.(Remember short pulses in the time domain equals broad bandwidth in the frequency domain.) Low pass filter to remove all of the aliases, and you are left with just the original smooth signal from the baseband.

Too many people concentrate on number crunching and measurements as opposed to letting their ears do the judging.

That question doesn't even make sense. Bit depth and sampling frequency goes together. High bit depth makes a lot sense when making and mixing music. But when it's time to play it back 16 bit is enough.

It doesn't, and that wasn't the question. 16-bit *cannot* sound better than hi-res, but you could argue that hi-res doesn't sound better than 16-bit. It's all in the mastering.

It's like asking how fast is one kilogram.

CD quality is most than any human may need. Maybe 48K sample rates to accommodate for not such as steep filter. However don't forget that the music industry is simply packaging the same music and selling it again in a different format. Audiophiles and others buy a DAC that reads 768K and you want a file that lights up that LED so you feel you got the best. All placebo.