24/192 Music Downloads ...and why they make no sense

Actually Thad, there is a lot of Bravado doing rounds on this. From what i know, many of us "young Folks" over 35, we dont hear over 12 kHz..some till 15 ;)

People can check here.

Yep... I am not qualified to comment on the reproduction of sounds much over 12Kh, because, without winding the volume right up, I simply can't hear them. :rolleyes:

Lots of interesting stuff on that site: www.audiocheck.net. Try these: blind tests.

Is it relevant? Nope... as kumarab says, this conversation is all over the place ... but what I like best about these threads is exactly this: all the associated and not-quite-associated stuff it throws up that I might never have seen otherwise :)

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Thanks but seriously... those who want to hear a difference will hear it while others will not. And who is to say who is right?

Does not matter. If you hear the difference, then go for 24/192. Else stick to MP3. Simple, eh?

Cheers
 
I dont deny that digital audio works. I studied all this in college, including the theorem, the proof etc. Now I cant even remember calculus or fourier transorms etc:lol:. I also remember reading about sampling, interpolation etc and a bunch of samples are used, interpolated and smoothed out to give an analog waveform.

So what am I missing here. Dont really want to go through all the theory here, just a small brief would do. No mathematical proof please, I know it exists, but dont think poor me will be able to understand that now. How in real life, two samples would be enough to reconstruct the original analog waveform. Are fts or ffts being used for smoothing? I am not questioning anything here, just wish to learn.

If the theorem works, we should not be able to differentiate between 16/44 and 24/192, other wise the theorem doesnt work.

The theorem works:lol:
 
Here is a very informative document (I picked it from the article posted by OP).
Your questions should be answered in page 24, 25 and 26.

Sampling Theory [pdf]
I cannot believe that I just read all the way through that. The reason I cannot believe it is that anybody that doesn't know me personally cannot believe just how innumerate I am!

OK, all but the first and last pages did have my head spinning (sinc? I can't even handle sin! ...well, ok, I can handle being just a bit naughty...) although the graphs are works of art, and include some pretty colours (remember spirograph?).

I don't imagine the author was writing for the non-numerate, but actually, he has done a really good job of making this almost understandable for us. Those that can easily digest it will enjoy reading all the fine detail. If I read correctly, he places the ultimate sample rate somewhat higher that 44.1, but says, "You do not need "more dots" for better accuracy," counter-intuitive though that may be. If it still seems unbelievable, go suck on his sinc wave illustrations. And stuff. <baffled, brain-steaming smiley>.
For those unwilling to plough through the paper, and I don't blame you, here is some of the author's conclusion. That's the author's conclusion: not mine. I think I see his point and am inclined to agree with it. but I post it here for general conversation, rather than my point of view...

There are reports of better sound with higher sampling rates. No doubt, the folks that like the "sound of a 192KHz" converter hear something. Clearly it has nothing to do with more bandwidth: the instruments make next to no 96KHz sound, the microphones don't respond to it, the speakers don't produce it, and the ear can not hear it.

Moreover, we hear some reports about "some of that special quality captured by that 192KHz is retained when down sampling to 44.1KHz. Such reports neglect the fact that a 44.1KHz sampled material can not contain above 22.05KHz of audio.

Some claim that that 192K is closer to the audio tape. That same tape that typically contains "only" 20KHz of audio gets converted to digital by a 192K AD, than stripped out of all possible content above 22KHz (down sample to CD).

If you hear it, there is something there is an artistic statement. If you like it and want to use it, go ahead. But whatever you hear is not due to energy above audio. All is contained within the "lower band". It could be certain type of distortions that sound good to you. Can it be that someone made a real good 192KHz device, and even after down sampling it has fewer distortions? Not likely. The same converter architecture can be optimized for slower rates and with more time to process it should be more accurate (less distortions).

The danger here is that people who hear something they like may associate better sound with faster sampling, wider bandwidth, and higher accuracy. This indirectly implies that lower rates are inferior. Whatever one hears on a 192KHz system can be introduced into a 96KHz system, and much of it into lower sampling rates. That includes any distortions associated with 192KHz gear, much of which is due to insufficient time to achieve the level of accuracy of slower sampling.

Conclusion:
There is an inescapable tradeoff between faster sampling on one hand and a loss of accuracy, increased data size and much additional processing requirement on the other hand.

AD converter designers can not generate 20 bits at MHz speeds, yet they often utilize a circuit yielding a few bits at MHz speeds as a step towards making many bits at lower speeds.

The compromise between speed and accuracy is a permanent engineering and scientific reality.

Sampling audio signals at 192KHz is about 3 times faster than the optimal rate. It compromises the accuracy which ends up as audio distortions.

While there is no up side to operation at excessive speeds, there are further disadvantages:
1. The increased speed causes larger amount of data (impacting data storage and data transmission speed requirements).
2. Operating at 192KHz causes a very significant increase in the required processing power, resulting in very costly gear and/or further compromise in audio quality.

The optimal sample rate should be largely based on the required signal bandwidth. Audio industry salesman have been promoting faster than optimal rates. The promotion of such ideas is based on the fallacy that faster rates yield more accuracy and/or more detail. Weather motivated by profit or ignorance, the promoters, leading the industry in the wrong direction, are stating the opposite of what is true.

From "Sampling Theory For Digital Audio (PDF)," By Dan Lavry
(He could do, perhaps with a higher sampling rate for his proof reading!)

Anyone not willing to at least try the explanations might just think twice before bashing the conclusions. Having just experienced, if not understood, those lovely graphs, I find "192 > 44.1," true though it is, a rather unconvincing argument. ;)

And yes, of course I'm willing to listen to 24/192, but my interface stops at 96. I thought it didn't, but it does :eek: --- which invalidates all the tests I have tried so far:eek: :eek:. Still: at last testing session I could not convince myself of any difference between 48 and 96, so, for my ears, it probably isn't worth craving an uprgrade.
 
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Why argue? Listen to the difference yourself.

Listening to 16/44.1 v. Higher Definitions | AudioStream

Cheers

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Thanks ventatcr,
If you rip DVD (Video) to Blu-ray scale OR try to convert video file with lower bit rate to higher format you will find NO quality improvement!
Similarly if you have 16bit/44.1khz audio file (as originally recorded) you cannot up-scale to higher format (say 24bit/192 khz) by any sort of conversion. But, if original audio file of higher quality (as recorded for higher qualitysay 24bit/192kz) is available, then it will sound better.
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Setup: (1) ONKYO-HTS3300 / Xtreamer / Samsung 32LED (2) DENON PMA-510AE / Paradigm Titan Monitors /PC- Xonar Essence STX
 
If you rip DVD (Video) to Blu-ray scale OR try to convert video file with lower bit rate to higher format you will find NO quality improvement!
Similarly if you have 16bit/44.1khz audio file (as originally recorded) you cannot up-scale to higher format (say 24bit/192 khz) by any sort of conversion.
I don't think that is disputed
But, if original audio file of higher quality (as recorded for higher qualitysay 24bit/192kz) is available, then it will sound better.
but that is not only disputed, it is denied! The interesting thing is that it is denied with technical justification, in two sources referred from this thread.

The company that Venkat's link leads to appears to be sincere in its offerings, and not just a File->Export As->Charge-More setup. I would have tried their samples, but first, I came across one of those sign-up things where one never receives the email, and then, more to the point, I checked the spec of my interface. The one plugged into my PC; the ones attached to my head certainly don't go to 20KHz, let alone beyond :sad:
 
^^I wouldn't say that's completely denied. Much like the example I gave for the crash cymbal; a pure digital recording made at 24/192 or higher, on ultra high end equipment (recording 30-35K harmonics), with no band limitations is bound to sound better than a redbook made out of it. A Redbook will almost always start deteriorating exponentially after 21Khz or so.

The real question is, was all that extra cost/size burden necessary given the range of human hearing and the limitations of consumer audio gear?

@Thad--On the point of "you don't need more dots"..I haven't read the article yet, so I will go look into it. But as far as I know, you actually do need more dots, it's just that sampling doesn't work in a STILL time frame. Every cycle has a history, the sampler has many historical dots for any given tone. So yeah, there's two for 20Khz right now, but you have a number of previous ones as well for interpolation. Don't really know the technical details, but maybe that's what he's talking about with sinc etc. Heck, even if you get 8-10 dots at 192K, you can still draw an infinite number of waves with them..so which is the right one?
 
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... Beats me! :lol:

That's the problem, maybe: we think that this sampling business is a matter of joining the dots and approximating steps, so the closer the dots or the smaller the steps, the near the end result to the original wave. We even believe that, given this dot/step business, it might never completely match that wave. This stuff is "intuitive" ever since we joined dots to make a picture as four-year-olds. No? That's what I understood too, until I started reading these articles. My only previous objection to being sold this stuff was that it should be on the bat's counter.

Do read the article, especially if you can do a better job of following the numbers than I can: it is interesting stuff about how digital audio actually works.
 
As far as I know, 2 samples per cycle is enough to represent the tone but more are needed for recording accurate amplitude.
Of course this applies only to the highest tone (i.e. 20khz) because the lower tones are already sampled by more than 2x.
So author's conclusion: "48k ought to be enough for everybody" ;) (pun intended)
 
As far as I know, 2 samples per cycle is enough to represent the tone but more are needed for recording accurate amplitude.
Of course this applies only to the highest tone (i.e. 20khz) because the lower tones are already sampled by more than 2x.
So author's conclusion: "48k ought to be enough for everybody" ;) (pun intended)

Aha... thats what I was missing... thank you....
So the frequency gets correctly sampled by more than 2 samples. its the amplitude ie volume at that frequency doesnt get recorded all that correctly. And we are not all that sensitive to minor variations in volume, many speakers having non flat frequency response anyway...
24/192 will give you smoother joining of dots due to more dots. Hence a smoother frequency response, which most of the people will not be able to make out anyway. If you tweak a crossover to change response at a particular frequency by 0.5 db, most of the people wont notice.
 
Amplitude is in the bit depth, rather than the sampling rate, isn't it?

What bit did I miss[-understand] I wonder?

It is tough for the non-numerate to understand all this, but it is interesting and I'm trying!
 
@Thad: Bits are for encoding the sampled voltage levels in to PCM format. 24 bits can represent more voltage levels than 16 bits. They may be used either for increased dynamic range or for finer grained digital copy of the original signal (= reduced quantization error). But this happens *after* the signal has been sampled. What doors666 is talking is *during* the sampling process where signal is being captured.

Think of it something like actual image file being displayed on lower vs. higher pixel density monitor -- while doors666 is saying you need 20 Gigapixel camera to accurately capture the image :D (where as kumarab is saying go get yourself a better lens+sensor instead of wasting time on pixel counts.)
 
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I'd say it is not so much a rejoinder, as just another subjective reaction, of which we have plenty already! OK, probably not many of our members get to try this out using their own recordings of symphony orchestras. :cool:

curious points, though...

My tests are straightforward.
But not blind?
I record live symphony orchestra concerts
ok, ok, vast amounts of cool :eek:hyeah:

but
the 192 File sounds the closest to the DSD file and is the highest fidelity of all the transfers.
Maybe just the manner of writing, but, somehow, given that the guy was there, I'd be more impressed with, "sounds closest to the original orchestra."
 
Just read it..it completely misses the point.

What some people fail to understand is that nobody is making the claim that "192 will never sound better than 44.1 under any conditions". It absolutely can, as I stated earlier, with uber-fi recording equipment and no band limitations which creates severe aliasing problems for 44.1.

Anyway, feels like resuscitating a dead horse, killing it..and then beating it again. A passionate rant is all that article is.
 
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