Up and Over sampling EXACTLY the same thing

Germanboxers, «...it's the sound that matters.» Of course you're right. And I'm about to believe your right with up-/oversampling, too – until now I didn't find any clue that the interpolation with upsampling is based on a sine-wave algorithm instead of a linear one (as with oversampling). But I won't give up completely.

Instead I've found this article (on that site):

Upsampling CDs to DVD-A high-bit standard is just the same as oversampling. Or is it?

by ANDREW HARRISON

While there is certainly overlap (but never ‘uplap’!) in the use of the terms ‘oversampling’ and ‘upsampling’, some guidelines can be given to differentiate the processes.
Oversampling is typically used to describe a technique used when transferring between the analogue and digital domain, where a signal is sampled many times over and above that actually required by the sampling frequency.
Oversampling in the context of the D-A process involves multiplying the sampling frequency by a whole number, typically between 4 and 32, or even higher. For example, in ‘8x oversampling’, CD’s base rate of 4.4.1kHz is raised to 352.8kHz by introducing seven new ‘empty’ samples between the original data samples. These new samples, though, are often not just empty strings of noughts, but based on mathematical models to assist the DAC to work more linearly with the extracted data.
Oversampling, as well as easing the workload of the anti-aliasing filter, which can now operate more gently at a higher frequency, can also reduce distortion created when those analogue signals are first turned from continuous, analogue waveforms into stepped, digital, stair-like curves. This quantization noise is now spread over a larger band after oversampling, and can even be somewhat shifted out of the audible envelope by the technique of noise-shaping. Sony/Philips’ Direct Stream Digital, as used
in SACD, takes this idea to its limit, in order to dump high levels of digital noise up to higher frequencies than are not directly audible.
Upsampling is a solely digital domain process where the data stream is also stretched out by interpolation — guessing the points in between, again mathematically — and is typically used to refer to small, non-integer changes, such as from 44.1 kHz to 48kHz. When the change is larger than this, such as 44.1 kHz to 192kHz, ‘upsampling’ is a more popular term.
'There is apparently no extra information in the upsampled signal that was not present in the initial signal,’ says Mike Story of dCS. ‘With a 44.1 kS/s input, both the input data stream and the upsampled data stream will only contain a spectrum that must be between 0 and 22.05 kHz and is probably only between 0 and 20kHz.'
'This conventional analysis starts from the viewpoint that the behavior of the ear can be described in mathematical terms using Fourier analysis. This assumption is probably pretty good — it means we are interested in frequency responses, for example, and these do provide good guides to the performance of equipment and to descriptions of what we hear. The analysis was right at the heart of the definition of the audio coding used on CDs.'
‘For those working with audio, it is also apparent that theories based on these descriptions are not completely adequate, and that there can be significant differences in the performances of pieces of equipment with similar "conventional" specifications. It seems that two things are going on here: the ear may have more than one mechanism at work; and sine waves may not be the best function to use as the basis for analysis. On the mechanism front, it seems highly likely that the ear has a sound localization mechanism ("where is it?") that is fast, and independent of the mechanism that says "it’s a violin", and that is related to transient response. There may also be a third mechanism at work. On the analysis front, it may be that some form of wavelet is the best basis for mathematical modelling. The problem here is that sine-wave theory is relatively simple, and has been fully worked out by generations of mathematicians, following on from Fourier. Wavelet maths is just plain hard work, and does not yet have anything like such a solid core of mathematical results to call upon. Our ears, however, are not waiting.’

It's me again...

Having read this site a second time, I'm convinced that the sampling rate conversion makes the difference between up- and oversampling – and possibly in sound quality. One thing that's rather clear is that such a conversion involves a sine-function interpolation (!). And that's exactly what probably makes the sonic advantage. (Without engagement...)

Martian, while I agree that the term "upsampling" is generally used in conjunction with 96kHz and 192kHz, both non-integer multiples of 44.1, I disagree that there is any fundamental difference from oversampling. I do believe that 96kHz and 192kHz are used specifically to wrap the manufacturer (and convince the consumer) that this is somehow as good, the same, or close to the same as the higher resolution format DVD-A (true 24 bit, 96kHz recordings).

"Sample rate conversion" simply means changing the sample rate from one rate to another. If you recorded a concert at 88.2 kHz DAT, the sample rate conversion was a simple 1/2 of the original rate to burn it to redbook CD. Most DAT's, however, are 48kHz or 96 kHz, so the sample rate conversion to Redbook CD is a non-integer conversion.

The "bits and pieces" (pun intended) of actually changing the sample rate are the same whether it is a non-integer or integer change. Extra samples are created and voltage values are assigned for these extra samples based on interpolation of samples before and after. Several techniques are available for the digital filter designer to choose from, one of which would be a linear interpolater. But an interpolation algorithm based 2nd order, 3rd order, or sine function can be used as well. This is the same for both integer and non-integer sample rate conversion (over/upsampling) and is just one of many factors a digital designer must consider.

I will leave you with this Wadia quote from the article you linked:

When used to convert a CD signal to a higher sample rate, the process of sample rate conversion is mathematically synonymous with over-sampling. Whether this process is performed in a digital filter housed in the same chassis as the D-to-A converter or in a ieparate chassis has little bearing on performance. Any advantage that can be claimed for a rate-conversion system can equally be achieved in a sophisticated over-sampled system such as the Wadia DigiMaster.

The concern with the Redbook standard that Up, Over or Down sampling miss...is that the biggest concern with all of your/my/our CD players is still where it was when we were all buying Sonic Frontier, Levinson, Classe DAC's and things like the Genisis Lens and SF Ultrajitterbug...the two areas that really have big impact on your sound, gentlemen, is jitter and the audio section of your player.

Why isn't this addressed(marketed) by CD/DAC manufactures..well it's because the cures are expensive. It is much easier to buy the latest Burr-Brown DAC...(which is a chip) and tout that, rather than do the serious structural/transport/receiver work that is the jitter end, or speak of the jfets or tubes(let alone the wire, caps, resistors..etc. that we all worry so much about in pre-amps etc.) or whatever in the audio section. Go to look at the ad/flyer on-line for some of these older DACs and see where they address jitter and the audio section.

Most of our/your most used source piece..the CD player has no better jitter than prior generation units and usually poorer audio sections..even very $ players are using op-amps for the audio signal. Kinda silly to worry about the tube vs. SS or silver vs. copper wire..etc. when the audio signal in most systems is starting in an op-amp that none of you would think much of if it was the audio section of your pre-amp!

So, my friends, up and over...is a new chip to put into a player or DAC that is easier to do and market..than the more costly..and harder to do improvements in jitter and audio signal quality.

Wadia oversampling advocating:

«...Any advantage that can be claimed for a rate-conversion system can equally be achieved in a sophisticated over-sampled system such as the Wadia DigiMaster.»

I can follow you. But at the same time it's clear that the interpolated curve resulting from an upsampling (non-integer sample-rate converting) system isn't identical with the curve from an oversampled interpolation – after smoothing by the low pass filter. Though we can't say which one is better or more adequate, there is a difference. It can also be deduced from the reviewer's (Andrew Harrison) judgement, which btw. clearly favors the «upsampling» dCS against the «oversampling» Wadia. No deciding argument, of course, due to the very different devices.