Up and Over sampling EXACTLY the same thing

Martian,

If you like how the Belcanto sounds, what's the problem? The sound is the most important part, at least for me.

Upsampling and oversampling are indeed the same thing. Both are sample rate converters. Both increase the number of samples by some factor (2, 2.18, 4, 4.35, 8, or whatever). Both will need interpolation to assign a voltage (amplitude) value to the newly created samples. Although a separate process, both may increase the word length from 16 bit to 18, 20, or 24 through the appropriate use of dither (basically white noise). Both may use a dac of various resolution (18/44.1, 24/96, 24/192 or whatever).

As Charles Hansen (research director for Ayre) said in the article I posted above, "Upsampling and oversampling are the very same thing and anybody who tries to tell you differently is misstating the case... And there's nothing unusual about putting two digital filters in a row -- virtually every digital filter is a cascade of 2x stages, because it costs less than accomplishing the entire filtering process in one go".

The reason why turning off upsampling on a capable player changes the sound is that you are taking one of the digital filters out, Particularly if it was designed to utilize the computing efficiency of two digital filters, it is likely to sound worse with only one. Again, Charles Hansen: "upsampling almost always makes a difference, and it can make an improvement".

In the end, it's the sound that matters. If it sounds better and the mfg happens to market using the "upsampling" hype, then fine. I do appreciate mfg's who talk straight about the topic though and if two products sounded identical (or maybe I should say the final subjective value I assign to the two products are the same), I will award the straight talking mfg with my business. That's just my philosophy.

Germanboxers, The VHS/Betamax story is much more interesting. As I recollect: The original betamax was a superior medium for encoding audio along with video. However, in the best spirit of technical competition, a multiplexing scheme was developed for VHS that kept them in the running quality wise. The format wars ensued with better quality being produced at every iteration. This was great reading back in the days when people were deciding what type of machine to buy. Ultimately Betamax lost out due to SONYs shortsightedness with regard to exclusive patent rights.

BTW, Judith, Beta (super~) was used in advertising all the same. Better of course, than vhs with good enough resolution to air, and cheaper to use than 35mm.

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...)