How many bits are really meaningful?

fas42

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I was referring to your repeated rant about upsampling MP3's and reporting how great they sound. You simply cannot recover the damage done to music once it has been encoded to MP3. You could minimize the damage by using a high bit rate encoding of 256k or 320k, but otherwise serious compromises are made. That you don't seem to notice the damage says a lot about how you are listening and your system's ability to resolve detail. That's fine for you, but don't be making claims about how all other systems should be able to do the same -- even better if they're great systems. T'ain't true.
Bill, in that original post I was referring to a completely conventional PC and speakers: nothing in the slightest bit audiophile about it. IN THAT CONTEXT the replay of the resampled sound was dramatically improved, POINTING OUT that the quality of implemention of the audio replay electronics was all important, at that price level. I have not asserted that a high quality DAC will benefit, though it may; there have been many posts by various people who have had experiences going in one direction or the other. The message is: the intrinsic level of musical data does NOT alter with resampling, but it may make a very significant difference in how well that data is translated into audio, depending upon everything ...

Frank
 

fas42

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Bob, thanks for your kind thoughts. The message I bring is a simple one: that a great deal more can be achieved in audio reproduction, if sufficient effort and focus is applied. Sadly, only a very few seem to want to tune into that, and the manufacturers are not going to be interested, there are no profits in it for them in the near term.

Shame ...

Frank
 

Bruce B

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Was given the task this week of transferring dozens of Blu-ray discs. Most were 70's and 80's rock concerts. Found out that the stereo layer of 50% of these discs were 16/48 while the other 50% were 24/48. Oddly, the DTS 5.1 layer of every one of these discs was 24/48. Don't know why they truncated the stereo layer down to 16bits on some of these

I loaded them into the workstation alphabetically and blindly tried to identify what the bit rate was. Within 10-15 seconds, I would choose what I thought it was. I chose correctly 100% of the time. This seems too easy. Certainly, the bit rate matters more than the sampling rate.
 

jkeny

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I don't go along with the mantra that MP3 files can be made sound wonderful by upsampling, etc. but I do believe that a great playback system can render 16/44 CD material in a way that is as convincing as high-res material. I have heard a dCs stack do this & was amazed at how much information there actually is on red-book CD. Perhaps the better a system is, the better it's ability to render the LSBs, the better the "realism" of the music? Of course, mastering comes into play & there is an argument that could be made that high-res material has a better chance of being mastered with more care (big maybe!) - this better mastering can have a much more profound effect on what we hear than the fact that it is high-res.

Edit: Maybe this has been said already - sorry, I haven't read the full thread
 

fas42

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The mantra, which I started, is not that MP3 files can be made "wonderful" always, rather that if one has a low cost, simplistically engineered playback environment, which has minimal capability of minimising interactions, interference between the digital and analogue side of operations, that if one upsampled offline the MP3 to hi-res then the playback may be improved considerably. This happened for me on a straighforward PC with everyday add-on speakers; the downside is that you end up with massively sized music files, which probably aren't worth it for most people.

The analogy is that if you have a car with lousy shock absorbers, then the more you iron out the bumps on the road before you come to them then the smoother and more pleasant should be the ride. A very expensive way to solve a suspension problem, but it's a demonstration of what's going on ...

Frank
 
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jkeny

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Sure, Fran, I understand & see that you posted this same explanation not so long ago. Apologies for not reading the whole thread.

I must try what you say & see if I concur with your findings.

I agree with you about the playback system - I think most playback systems are not rendering the LSBs & therefore we are effectively not getting full 16bit playback. As I said in my post redbook CD can sound wonderful, given the right system. But if this contention is correct, is the MP3 codec retaining this LSBs or does the MP3 psychoacoustic drop them? I'm not intimate with the workings of the codec to answer this but I suspect it varies & in a lot of instances the LSBs are dropped. This to me is where the "magic" lies - the illusion of realism that we all crave, the sonic tails of instruments, etc.
 

fas42

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The MP3 encoder is retaining the LSB when it thinks we'll notice, and it drops information when it believes our ears won't realise what's going on. At one stage I did quite a bit of playing with the very versatile LAME software, and determined that it allows you to tweak the end result to a point very, very close to the raw, uncompressed music file. If you're prepared to spend the time twiddling, that is ...

My own experiences with fooling with digital replay is that the quality of it has very, very little to do with encoding, bit depth, sample rates; but everything to do with how clean, distortion free, you can make the process of playback. This is not trivial, in spite of all the very impressive specs produced on how good digital is, they do not tell the whole story by a long way.

The "magic" is there, I found it over 25 years ago, as have some others, but you can't walk into a retail hifi shop and buy it off the shelf ...

Frank
 

jkeny

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The MP3 encoder is retaining the LSB when it thinks we'll notice, and it drops information when it believes our ears won't realise what's going on.
That's what I'm getting at - how often does it cut the LSBs because the codec's psychoacoustic model says it won't be heard? I know it's not an easy question to answer by looking at the codec but would be easy to answer by comparing the original digital audio file to the digital audio file recreated from the MP3 .
At one stage I did quite a bit of playing with the very versatile LAME software, and determined that it allows you to tweak the end result to a point very, very close to the raw, uncompressed music file. If you're prepared to spend the time twiddling, that is ...

My own experiences with fooling with digital replay is that the quality of it has very, very little to do with encoding, bit depth, sample rates; but everything to do with how clean, distortion free, you can make the process of playback. This is not trivial, in spite of all the very impressive specs produced on how good digital is, they do not tell the whole story by a long way.
I completely agree here - we are saying the same thing - I'm saying rendering the LSB is required for the "magic" - this can't happen without a low distortion playback system - which is not a matter of published specs - they bear very little correlation to how something sounds

The "magic" is there, I found it over 25 years ago, as have some others, but you can't walk into a retail hifi shop and buy it off the shelf ...

Frank
Well, I cited a dCs stack as the only commercially available kit I have heard capable of this "magic". there may be others? But the dCs stack CD, DAC, clock) costs about €40K. Amplifiers & speakers another €40K
 

fas42

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That's what I'm getting at - how often does it cut the LSBs because the codec's psychoacoustic model says it won't be heard? I know it's not an easy question to answer by looking at the codec but would be easy to answer by comparing the original digital audio file to the digital audio file recreated from the MP3
Unfortunately this is a "how long is a piece of string" question: changing one parameter slightly for the encoder will change the output by a very significant amount, at the detail level. The LAME encoder will allow you encode at rates beyond what's part of the MP3 spec, better than 320, but what good is that if most replay can't handle it. And at some point here you'll be getting very close to an end file size comparable to FLAC, say. So what's then the point of the whole exercise?

I completely agree here - we are saying the same thing - I'm saying rendering the LSB is required for the "magic" - this can't happen without a low distortion playback system - which is not a matter of published specs - they bear very little correlation to how something sounds

Well, I cited a dCs stack as the only commercially available kit I have heard capable of this "magic". there may be others? But the dCs stack CD, DAC, clock) costs about €40K. Amplifiers & speakers another €40K
Yes, indeed, rendering the LSBs correctly is equivalent to low distortion; the trouble is, or at least I have found so, is that this distortion creeps up to equivalent to fairly high level SBs.

Others here may, or will dispute this, but you don't have to go expensive -- it certainly helps, but is not necessary. My "discovery" is that most or all gear is capable of getting there, but it requires heavy duty tweaking of many parts of the system: something that very few people do. The underlying problem is that the digital replay process is very fragile, in that the slightest problem areas in terms of noise, static, RF interference, poor mains quality, inadequate power supplies, throw the quality of digital badly, and unless all those elements undermining the sound are fixed up or bypassed by some means then you don't get the good sound ...

Frank
 

jkeny

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I agree & have done modifications/treatments along the lines of what you are talking about, Frank!
Hearing that low level detail from digital audio is a pretty challenging task. Even just getting jitter to a low enough level (low pS) is a very challenging task & that is just one barrier to good digital audio reproduction.
 

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Glad to see that in print, but nothing new to me. I've mentioned similar perspective's, here and in other forums, but not as well written, of course.
 

Bruce B

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Can't view it: I need Firefox, Opera or Chrome as my web browser in order to do so.

Bruce, what is the exact title of that article?

The Problem with A-B'ing and Why Neil Young is Right about Sound Quality.
by Allen Farmelo on April 4, 2012 - Comments (26)



With news of Neil Young pushing to get studio quality audio out to the consumer, it occurred to me that it would be good to get some perspective on how we go about comparing audio quality. One of the most problematic issues surrounding audio quality and file resolution these days has been that, more and more, we hear about people not being able to tell the difference between compressed audio and full-resolution audio, and by extension some people are claiming that efforts like Neil Young's are out of touch and irrelevant. I couldn't disagree more.

In my opinion, one of the biggest pitfalls in this arena is the dreaded A-B test. In our seemingly infinite desire to manipulate the response of our fellow humans, we tend to stage A-B tests in two ways:

1) Almost-Barely: A-B because you can barely tell, and thus proving that the two items are close enough to be interchangable.

2) Absolutely-Boldy: A-B because you want to show how different two things are (and typically the superiority of one of them, which the test-designer is likely selling).

In my opinion, both types of A-B tests won't tells us what we want to know, but it's the first type, the Almost-Barely tests, that I take issue with here. (And I'm also inadvertently and respectfully drawing into question what, Ethan Winer, (who I adore) has said about "scientific" A-B testing in Tape-Op #88, p.66.)

We've all done it at some point. A. B. A. B. Flip. Flip. Flip, flip, flip. This is B? Ok, A. B. Ok, is that B? Ok, do A? Gosh, they're close!

The Almost-Barely tests seem so objective. Here's A, and here's B. See, soooo close! The conclusion: because most people can't tell the difference, there's not really a difference. And then you stack up the stats on these results, and suddenly we're doing science.

The problem is that these tests assume that because two things are close enough in a quick test that the difference will also be indistinguishable over long stretches of time. However, this assumption totally misses how it is that we tend to actually experience things in our very real lives.

For example, we hear people talk about how one can't make out the difference between a hi-res MP3 and a 24bit WAV file (assumedly a difference similar to the one Neil Young feels is worth fighting for). Admittedly a hi-res MP3 and a 24bit WAV are relatively close enough in resolution that many people will not be able to pick them out in an A-B test.

But, we don't live with music like that. If your'e anything like me, you listen to a lot of music in a lot of styles and - over the course of, say, a month - perhaps you've absorbed well over a hundred listening hours across many different albums on a few different playback systems.

How can flip flip flip replicate what it is to live with that much music for that long? How can a drop of water emulate what it is to swim for hours?

If you want to do a real test of the differences, give people a music collection that's all MP3s for a month, then give them that same collection as 24bit WAVs for a month, and then ask which one's which, and I bet you will start to get some correct answers.

Why? Part of the answer is that, if given enough time, subtle differences will reveal themselves to us. Subconsciously at first, and eventually consciously, we become aware of new details, subtleties, nuances. We humans need time to truly come to perceive things in full detail. But details, once revealed, become important features in the big picture.

For example, when I started living with my partner I introduced her to what I call "good coffee." At first she kind of shrugged it off as my snobbery at work, and she couldn't really taste the difference. But then, after months of drinking the good stuff, she found herself to be a bit of a coffee snob, too. She could taste the difference because she had, simply, spent time with the good stuff. The coffee revealed itself to her, slowly and subtly. Her palate developed. And the thing about good coffee is that it holds more detail, nuance and, therefore, interest.

But it takes a while to become aware of that depth and complexity. Had she done a flip-flip-flip A-B and made her choice to only drink the cheaper stuff because, "you know, they're basically the same," she'd have missed an opportunity to develop her palate.

I think the same thing can be said for the resolution of music, and it breaks my sonic heart to think of the A-B tests out there designed to convince someone that because they can't tell the difference today they won't tell the difference in a month or a year. A-B tests may be designed to show that subtle differences don't matter, but what they really do is shut down the possibility that those subtle differences could be the key to someone's aesthetic awakening.

If you can't tell the difference when you flip flip flip between two subtly different sounds, please know that I'm here cheerleading for you to slow down and be a real human being. Live with the better quality for a while, and see if over time you too start to hear things you never heard before. There is nothing like real life to truly test the quality of something.

So, for this reason, I herald Neil Young (as I always have, come to think of it) for sticking to his beliefs and making efforts to bring affordable and easily accessible hi-resolution audio to the market. It's time we left the MP3 behind and opened back up the possibility of people developing their ears and loving great sound once more. Who knows where such high standards could lead us.

Thank you, Neil.

The aesthetic revolution will be beautiful!

Allen Farmelo
www.farmelorecording.com
 

jkeny

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Yes, a good article & worth reading. One of the possible reasons for the failure of A/B testing to differentiate between two devices known to sound different on longer term acquaintance, is that the focus on A/B testing tends to be differences in the frequency domain. This is also a the focus of most measurements in audio - just look at the use of FFT analysis - it is an averaging measurement that fails to pick up fast transient details!! These time domain differences are what I believe is the big faux pas of digital audio & are only beginning to be considered now.

I believe that the ears are very finely attuned to time differences due to the need for survival - we can locate the direction a sound emanated from because of the tiny timing difference between the sound waves reaching each ear. This has obvious survival advantages & I have seen it referred to as the threat correlation system.

Now audio is an illusion produced by our electrical devices which try to recreate a 3D performance using (in the main) 2D stereo sound. This illusion of depth of sound stage can only be achieved with very precise placement of the elements in the sound stage & no shifting of these elements. This requires that very precise timing differences are created & maintained for the different placement of elements in the sound stage.

Now, finally to my point - the time domain differences are a subtle effect that do not leap out at you immediately like a gross difference in frequency response might but rather reveal itself over time & yes also require a decent system capable of preserving these subtle details (my LSB references above :)). Sometimes it reveals itself as an unease in listening, sometimes as a headache after prolonged listening, sometimes as just an unnaturalness
 

NorthStar

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Bruce, that's why I asked you the exact title; because I finally accessed it! :b

...And you took your precious time to type it all!
Oh my God, are members here at WBF nice or what!

TY so much for the privilege of knowing you.

* And thank you too Don for that alternative link (it works too)!

--> The Expert members here truly rock! :cool:
 

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