Upsampling. Truth vs Marketing

Sean,
basically your "kinda - sorta the "quick and dirty" explanation" is pretty good for the layman.
I have objections on some of the text you posted:
* Audio Note eschews much of the filtering and gets rid of the oversampling, which reduces a LOT of the in-band noise and distortion
How does getting rid of oversampling & eschewing much of the filtering reduce IN-BAND noise & distortion???
AFAIK, anything in-band cannot be touched. It's sacred as it's THE signal we are looking for. If noise exists in-band or if distortion exists in-band, you basically have to live w/ it OR design better electronics. What you wrote will not do the trick.

* At the same time, it can introduce out of band noise and distortion into the equation
what are you referring to here? i.e. when you write "it can introduce....", what is "it"??

* Obviously, the key is to find a way to increase the sampling rates to recover more of the data
Increasing the sampling rate does NOT recover more data. It, however, allows the discrete-time system to follow the original analog data more truthfully. This is evident from your section A, section B example.

On a historical note, Philips is the co. that is to be credited or discredited with the concept of upsampling. The original idea at Philips Reasearch Labs was to somehow get that analog filter order lower & that transition band less steep. In the original redbook spec, the transition band is 20KHz-22.05KHz. Upsampling was the answer from an engineering perspective & from a cost prespective. They really didn't care about the sonic effects back then.
FWIW. IMHO.

Germanboxers,
basically, you are correct in pointing out that 96K & 192K were selected owing to other hi-res audio formats (namely DVD-A). Here the electronics is a multi-rate system wherein it up/oversamples by 160 & then decimates by 147 to change the sampling rate to 48K from 44.1K.

However, if you buy any of SimAudio's products, then you'll find that they oversample at exactly 8X, which is 352.8KHz!!! So, here is one commercial co. that doesn't use 96K, 192K or 384K. There must be others too but I cannot think of them right now.
FWIW.

Thanks for the feedback Sean. Putting your orig. & 2nd post together, I see what you were trying to say.

Eldartford's sentence: "In Sean's explanation the second set of 20 dots in set B should not be random. Those dots should lie somewhere between the two dots adjacent to them".

is exactly correct. One possible location of "somewhere between" could be legitimately the midpoint. There is no problem with that at all. If the waveform looks smooth then what's the issue with that??? How, in the world, do you know that the waveform at this point in the CD is not supposed to be smooth?? There could be a consistently low volume passage or a consistently loud volume passage of 1 particular instrument that creates a smooth area. Entirely possible.

Anyway, the thing to remember in your 2nd example is that when you placed that "random" set of points, you were looking at the output of the digital estimation filter. The output of digital estimation filter is very deterministic & it is designer created. The o/p simply cannot be random - no way!! It lies "somewhere between" the actual sampled data points off the CD along a line determined by the algorithm of the digital estimation filter. This is that (digital) filter that creates all those signature sounds (like Wadia's house sound, Sim Audio's, dCS's, etc, etc) that many love & equally many hate.

In Eldartford's example, I think, that he used a smooth waveform only to illustrate the point. This is the way that it is usually introduced in DSP 101 classes. His particular example is pertains to oversampling. When he shows the repeating of numbers, he has considered a 12X oversampling & when he does the div-by-4, he is considering 4X oversampling. The div-by-4 most probably represents the digital FIR that follows any over (or up) sampling operation.
My only question here is why did the example consider an oversampling of 12X then later decimate to 4X?? Should have just started of with a 4X DAC. Anyway.....

You mentioned "error correction" for the 2nd time. Error correction in redbook CD playback has nothing to do w/ upsampling or oversampling. Error correction is NOT designed to correct the music written on the CD. It is designed to compensate for high-speed read & transmission of the bits where read errors will occur (owing to the high speed read operation). I think Eldartford's succinct explanation is exactly what error correction is all about. Any other idea of it is a mistaken impression.

I have read the recent upsampling verbose text by Moncrieff on IAR. IMHO, I have not read more bull**** anywhere that filled up so many pages. Very little of what he has written is correct. AFAIK, Moncrieff is very lost when it comes to up & oversampling. If you are taking your lessons from him, then I can see why you are mistaken too. Get hold of a DSP text (like Oppenheim & Schaeffer or Rabiner & Gold) & read that. You'll get the correct explanation of upsampling & oversampling.

As long as human beings are analog, the initial & final music will always be analog. What's in between can be digital. Digital is a compromise for an analog signal - no doubt. How good or bad it is depends on how well the digital system is engineered for the 20Hz-20KHz bandwidth. Digital is chosen mostly for its cost effectiveness (scalability of the DSP engines with shrinking CMOS technology) & what Carver Mead once pointed out - its tremendous noise immunity. Corrupting a stream of digital data to the point of making it useless is very difficult as it requires a lot energy to flip a bit. Some bits do get flipped but the overall context of the message is very much retrievable by using various error correction algorithms. This is hardly the case with a purely analog music signal.

Having more sampling points with an estimation filter allows the digital to better track the analog waveform. Whatever benefits one accured with over/upsampling could be lost by distortions in the analog reconstruction filter. Hence, the above mention of implementation. Having somebody engineer a good re-produced sound CDP solution is priceless (for everything else there's MasterCard!).

Yes, if one converts from analog->digital->analog, one does degrade the original sound. That is to be expected as we take only a finite # of samples (hence the term "quantization"). BTW, if we had infinite # of samples, it would analog! In the redbook CD format, the powers-that-were decided in all their infinite wisdom to Nyquist sample the data onto the CD disc . Thus, no matter how much one oversamples, one can never undo this. Hence the rise of "hi-res" music formats. In fact, if the over/upsampling was A1 perfect, you'd get *exactly* what was on the CD, which is Nyquist sampled!! How good is taking just 2 samples of a dynamically waveform music signal? Not very good I'm afraid!
Eldartford cited his experience: 4 samples was worth every effort. I've found 5-8 samples is worth the effort. The difference is that my work is voice-related. Not hi-res by any standards but when people hear another voice at the other end, they do want to recognize it. Need more samples for this.
FWIW. IMHO.

Eldartford,
Good observation indeed! It would be like splitting hair, IMHO, to find out where that analog-digital demarkation is!! I'll let the others "battle" it out while I listen to my music.