The tired of the obfuscation on Digital Audio article.

DaveyF

Well-Known Member
Jul 31, 2010
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La Jolla, Calif USA
A few days ago, I was sent this very interesting article on being tired of the Obfuscation on Digital Audio:




http://people.xiph.org/~xiphmont/demo/neil-young.html
All this info can be found on MIT's open courseware. It'll take awhile to digest, so here's a few quotes to whet your appetite.

On Golden Ears:
- Auditory researchers would love to find, test, and document individuals with truly exceptional hearing, such as a greatly extended hearing range. Normal people are nice and all, but everyone wants to find a genetic freak for a really juicy paper. We haven't found any such people in the past 100 years of testing, so they probably don't exist.

- Unfortunately, there is no point to distributing music in 24-bit/192kHz format. Its playback fidelity is slightly inferior to 16/44.1 or 16/48, and it takes up 6 times the space.

- Responses indicate that few people understand basic signal theory or the sampling theorem, which is hardly surprising. Misunderstandings of the mathematics, technology, and physiology arose in most of the conversations,
often asserted by professionals who otherwise possessed significant audio expertise.

Argument for limiting bandwidth to the audible spectrum (rather than up to 196 KHz)
- Neither audio transducers nor power amplifiers are free of distortion, and distortion tends to increase rapidly at the lowest and highest frequencies. If the same transducer reproduces ultrasonics along with audible content, any nonlinearity will shift some of the ultrasonic content down into the audible range as an uncontrolled spray of intermodulation distortion products covering the entire audible spectrum.

- You can't and won't have ultrasonic intermodulation distortion in the audible band if there's no ultrasonic content (like on CDs)---- Just love this one..:D

Intermodulation test for your system that you can try yourself is provided
- If you hear anything, your system has a nonlinearity causing audible intermodulation of the ultrasonics.

- there are (and always will be) reasons to use more than 16 bits in recording and production. None of that is relevant to playback; here 24 bit audio is as useless as 192kHz sampling. The good news is that at least 24 bit depth doesn't harm fidelity. It just doesn't help, and also wastes space ....16 bits is enough to store all we can hear, and will be enough forever .............. a 16-bit noise floor is already below what we can hear ......... Professionals use 24 bit samples in recording and production [14] for headroom, noise floor, and convenience reasons........24 bits keeps the accumulated noise at a very low level. Once the music is ready to distribute, there's no reason to keep more than 16 bits.
- Empirical evidence from listening tests backs up the assertion that 44.1kHz/16 bit provides highest-possible fidelity playback. There are numerous controlled tests confirming this, but I'll plug a recent paper, Audibility of a CD-Standard A/D/A Loop Inserted into High-Resolution Audio Playback, done by local folks here at the Boston Audio Society...............This paper presented listeners with a choice between high-rate DVD-A/SACD content, chosen by high-definition audio advocates to show off high-def's superiority, and that same content resampled on the spot down to 16-bit / 44.1kHz Compact Disc rate. The listeners were challenged to identify any difference whatsoever between the two using an ABX methodology. BAS conducted the test using high-end professional equipment in noise-isolated studio listening environments with both amateur and trained professional listeners. In 554 trials, listeners chose correctly 49.8% of the time. In other words, they were guessing. Not one listener throughout the entire test was able to identify which was 16/44.1 and which was high rate.

- The BAS test I linked earlier mentions as an aside that the SACD version of a recording can sound substantially better than the CD release. It's not because of increased sample rate or depth but because the SACD used a higher-quality master. When bounced to a CD-R, the SACD version still sounds as good as the original SACD and better than the CD release because the original audio used to make the SACD was better. Good production and mastering obviously contribute to the final quality of the music


For a video presentation and some graphic demonstrations of sampling, quantization, bit-depth and dither on real audio equipment using both modern digital analysis and vintage analog bench equipment, see:
http://xiph.org/video/vid2.shtml
 
Can you post the MIT course? I think I need to join the Boston Audio Society. The xiph article has been discussed here before, and the video was posted by xiph this week.
 
DaveyF,

Unhappily the quotes drove me away. The Meyer and Moran paper is not recent any more - it was published in 2007 and debated ad nauseam in the net.

Besides I do not enjoy the style. Sorry.
 
Yeah I feel a lot of it is err debatable, such as one reason for higher sampling rates comes down to the filters and greater flexibility-better design for time and frequency domain, mathematical constraints of real-time DACs,etc.
What I am glad to see is finally a presentation showing the stair step and also how at the end its effect is negligible (if within the constraints), classic digital engineering (when I was taught-used in 80s-90s) followed this in detail (each step and process-function), which is how it appears in the Nyquist transmission paper/Shannon communications anyway.
Also there is a reason for going beyond 16-bit that has nothing really to do with dynamic-bit resolution of the music.

Cheers
Orb
 
Yeah I feel a lot of it is err debatable, such as one reason for higher sampling rates comes down to the filters and greater flexibility-better design for time and frequency domain, mathematical constraints of real-time DACs,etc.
What I am glad to see is finally a presentation showing the stair step and also how at the end its effect is negligible (if within the constraints), classic digital engineering (when I was taught-used in 80s-90s) followed this in detail (each step and process-function), which is how it appears in the Nyquist transmission paper/Shannon communications anyway.
Also there is a reason for going beyond 16-bit that has nothing really to do with dynamic-bit resolution of the music.

Cheers
Orb

Stair step? What stair step?
 
In the video, not going back in but he shows it on one screen before it is truly filtered and outputted as a DA signal.
In fact it can be shown in several of his examples as it changes quite a lot as you go up the frequency.
BTW it is more of a technical point and academic because you do not see this at the output of the DAC once it is an analogue signal again, if he wants to disagree then that is a lot of digital engineers including Paul Miller (who designs digital measuring equipment) that it is going against.
We are talking not at the end but at the reconstruction filter point, even though he says there is no stair-step he can only say that in the context of the final outputted signal after reconstruction-filters, as shown by one screen showing the before signal ("stair step" structure) and the last showing the analogue sinewave (smooth sinewave).

Edit:
It relates to digital requiring a constant time-sampling gate,quantization, and also the aliases (short and very crude reason).
Anyway it is a loose-general term also called stair-case by others.
Cheers
Orb
 
Last edited:
TBH Tom it seems only a very rare amount of high rez music even has a dynamic range of 0dBFS to -110dBFS, usually from what I have seen measured by Keith Howard/Paul Miller this is classical music recordings based on piano solo performances; not only provides a great dynamic range but also much of this by 10khz so we are not talking hard to perceive real high frequencies.

But there is also another reason why we need more than 16 bits and relates to combined dynamics and dither; some academic papers out there if people are really interested searching.
Cheers
Orb
 
Aaahh yes, have you come to terms with understanding that part of digital now Mark? By the way, if one looks with high resolution at an analog (smooth) waveform, with enough resolution, well, guess what, it aint smooth no more.

i'm no digital theory guy, or one to care about the math.

a question; how does MP3 compare to Redbook 16/44 from a math perspective?

does the math or theory side of things say MP3 is also without steps? that it's smooth and continuous? that it has 'all' the information from the source.

as the time gaps between samples gets longer and longer i suppose that the approximations of what should be represented between the samples gets worse and worse.

conversely; it's a matter of opinion and a subjective thing to say how short the time gaps need to become before one's hearing is fully convinced that you have 'all' the information. and maybe it's not the actual time gaps but some other part of the math that get's in the way of the information.

something sure does.
 
Mike, that is down to lossy compression algorithms IMO when comparing MP3 to CD.
The maths are pretty solid in that one can reconstruct a perfect sinewave as long as it is within Fs/2.
If you look at the video posted (although I feel a lot is left out on other related subject matters) you can see the effect of 16khz signal, it looks a right mess before the filters-reconstruction but afterwards you see the perfect sinewave.
This will not change unless one breaks the Fs/2 rule, however one also has to consider that 44.1khz does not provide much flexibility for the algorithms-mathematical computational-etc required nor which is the "right" filter algorithm to use as they are causal in either time or frequency domain and not perfect.

Cheers
Orb
 
In the video, not going back in but he shows it on one screen before it is truly filtered and outputted as a DA signal.
In fact it can be shown in several of his examples as it changes quite a lot as you go up the frequency.

BTW it is more of a technical point and academic because you do not see this at the output of the DAC once it is an analogue signal again, if he wants to disagree then that is a lot of digital engineers including Paul Miller (who designs digital measuring equipment) that it is going against.
We are talking not at the end but at the reconstruction filter point, even though he says there is no stair-step he can only say that in the context of the final outputted signal after reconstruction-filters, as shown by one screen showing the before signal ("stair step" structure) and the last showing the analogue sinewave (smooth sinewave).

Edit:
It relates to digital requiring a constant time-sampling gate,quantization, and also the aliases (short and very crude reason).
Anyway it is a loose-general term also called stair-case by others.
Cheers
Orb

I know what the video shows and I brought it up on the other thread as to why some audiophiles are confused about how digital records analog waveforms and I was told that was all a myth and there really aren't any stair steps when digital records an analog signal.
 
Ah ok MEP.
Well those saying it does not exist either have not been involved with all components/architecture in the digital world of ADC-DAC, or just think of the context of what is outputted and do not consider the integrated functions-processes that are within the DAC (and these are trying to do a lot on one chip-processor) - these were originally separate functions and some may feel sound quality for some products is better due to said separation of those functions to dedicated chips.

Well the stair-case is shown with that real world example link as it has the before and after filter-reconstruction :)
I will see if I have any copies of MIT or Berkeley lecture notes, or something else I can link up here to show that technically it does exist and for the reason I mentioned in my earlier post.
Cheers
Orb
 
They tried to tell me that the stair step we saw on the scope was just something they generated to illustrate a point and not what the sine wave looked like before it was reconstructed in analog.
 
Unfortunately some may see this as an argument to win MEP so I would not get too bothered with them as I doubt anything you can present will change this, just use it for yourself.
I have some pretty complex papers on the subject but the following are simple in showing the "stair-case" and why it exists before the output, but bear in mind the actual analogue output is a perfect sinewave (NOS complicates this though) albeit with causal consideration for the time/frequency domain and ringing-ripples by filters.
Page 23 to 25 (or page 12 to 13 of viewer) DAC in the big picture: http://www-inst.eecs.berkeley.edu/~ee247/fa08/files07/lectures/L16_f08.pdf
Also following and read through the whole exercise as touches on this as well: http://web.eecs.umich.edu/~prabal/teaching/eecs373-f10/labs/lab7/index.html

Also worth noting Don has a large technical section in his own expertise threads.
Cheers
Orb
 
What is wrong with digital PCM

All this endless talk about stairs and steps only evidences that the real problem of digital is the lack of studies of human psychoacoustics by digital engineers who go on studying complicated math’s and algorithms and refuse to look at the human characteristics of sound reproduction. They go on studying books and papers, spend enormous amount of time on their computers and forget to look at the ear.

Some years ago I had to build a stair to access my attic. Very soon I found a famous german book full of architectural tables where I found that the optimum rise-to-tread ratio is 17/29 and absolute values for step tread in various practical cases. Surely this ratio was not established by physicians and mathematicians using mathematical models of human locomotion, but empirically based on human tests carried centuries ago. I also found that some architects hate this type of books, but this is another affair.

So my conclusion it that digital systems still have problems because people still think about sampling theorems, signal to noise ratio and many boring subjects when researching and perfecting digital audio. Very wrong - the sampling rate and number of bits should be established by looking at the best rise-to-thread of the human ear. Only after that we should look at sampling rate and number of bits. Once this important parameter is established and implemented in digital, manufacturers of expensive analog equipment can retire – we will finally have perfect digital sound forever.

BTW, only on the light of this theory we can understand why DSD sounds better as Bruce has been telling us – it manages to circumvent the problem of not knowing the optimum rise-to-tread ratio of time and voltage. But it is still not perfect.

Dedicated to Amir and Don, our digital experts.

Copyright of What's Best Forum. Neither the author or the forum owners assume any responsibility on the inappropriate use of this text.
 
How many audiophiles are electronics backgrounds, and then trying to understand stuff technical without the background, then someone tries to simplify the theory, then the simplification becomes the reality. Hell, theres engineers out there that don't understand how an emitter follower or cathode follower works, just what it does. A big difference.

I would be needing some simplification to try to understand microbiology and then with that simplification I would be confused real fast when they started to talk more in depth about it.

Study what sampling is and what bit depth is and then what a filter does and then it will make sense, if one wants to do that kind of stuff. Otherwise, just have to believe the people who know what they are taking about, just like i have to believe the biologist dudes and dudets.

The process of sampling of course implies discrete samples, (numbers) and when we pull back a number, the next number must be a bit bigger or smaller, and thus, yes, there are discrete levels during the conversion process that we smooth out between them with filters when going back to analog, or (assume proper bit depth) if we had a super high sampling frequency, we could just push out the super insignificant level stair steps (no output filter used or we are in the noise floor) and your ear could not hear them. Thats analog.

It's more than implied, it's real. And the stair steps aren't smoothed out with filters, it's done with dither if I understand the process correctly.
 
MEP, look at those links I provided, helps to clarify the situation.
Cheers
Orb
 
MEP, look at those links I provided, helps to clarify the situation.
Cheers
Orb


I did. I think this was more clear to me in explaining the process:

Dither is essentially a very small amount of white noise (equivalent in amplitude to one quantising level -- about 90dB below peak level in a 16-bit system) which is deliberately added to the analogue audio signal as it enters the A-D converter. Thinking back to the transfer plot, this dither noise effectively 'fills in' the steps in the transfer curve so that a straight line can be drawn through the noisy staircase. When correctly dithered in this way, the digital system behaves perfectly linearly, but there is a small amount of noise always present -- the amplitude of which rises and falls as the input signal increases in level through each quantising band (see Figure 4).

This info came from: http://www.soundonsound.com/sos/jun98/articles/digital2.html
 
Enjoy your digital learning path!

What's that supposed to mean Tom? Of all the digital I have heard in my life, the best sounding digital to my ears is DSD and I can and have listened to it for hours on end. I just downloaded the Opus Sampler 3 DSD128 and I'm looking forward to hearing it. Back to my original point, you are either right or you are wrong with regards to filters smoothing out the digital stair steps.
 
You could take a look at say some measurements by JA for a NOS DAC (no reconstruction filter) and compare its figs to that of one with said filter (oversampling delta-sigma DAC) MEP.
Its late and pretty academic anyway in the scheme of actual digital challenges that I touched upon earlier, so going to let this go myself even though you cannot get away that the DAC must reconstruct and then use algorithms to provide the analogue sinewave perfectly, part of this relates to aliasing artifacts not just quantization and associated errors.
But as JA mentions in his articles and also in measurements relating to NOS, normally this is not something one has to worry about for DACs with said filters.

Edit:
Just to say I have seen the exact same output results for a NOS from both JA and Paul Miller.
Cheers
Orb
 
When correctly dithered in this way, the digital system behaves perfectly linearly, but there is a small amount of noise always present -- the amplitude of which rises and falls as the input signal increases in level through each quantising band (see Figure 4).

This info came from: http://www.soundonsound.com/sos/jun98/articles/digital2.html

That is not correct dither - the correct dither is TPDF which does not give rise to noise modulation and is two LSB in amplitude (not one as claimed in this article).
 

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