upscaling, bits and resampling ??

To be strictly theoretical the number of bits represent an amplitude. Thus for red-book CD the 16th bit represents the highest amplitude. There also the number of samples that divides the audiable freequency spectrum 20Hz...20kHz. 44.1kHz is the sampling rate of the red-book CD. Any represented analogue audiable signal can be placed in the grid of bits and samples.

Imagine that you want to make a copy of the picture and you drow the grid on your empty piece of paper and you drow the same grid on the picture and start re-producing exact copy.
The same thing happens in digital recording. The more density grid the more precise "picture" you will be able to drow.

So finally if you want to know what realy upsampling and upscaling can do:

Upsampling is simply increasing the sampling freequency. It decreases the overall probability of error. One sample from red-book CD decoded by upsampled 192kHz DAC will be decoded by ~4 samples. The resulting probability of error of 4 samples at the same time is much smaller than the probability of error of pure 1 red-book sample.

Upscaling basically will not change even a thing for a red-book CD placing the same way 16-bit amplitude into 20 or 24-bit amplitude. It might only benefit HDCD if coding is present and higher mastered CDs.

New tech made it easier to perform the reading with minimal error in comparison to the older machines but still "former heavy weight digital champs" sound great -- do your research and deside.

Zaikesman is spot on ... ignore the hype and use your ears. If you're interested in how it works go to DCS's web site http://www.dcsltd.co.uk/papers.htm. Beware of answers on Audiogon, because it is my experience that many people offering answers don't really know what they're talking about.

If I'm not mistaken, I believe that the multiple concurrent sampling technique that Marakanetz and Jeff refer to above is actually descriptive of *oversampling*, rather than "upsampling", which has more to do with Jeff's reference to the use of alogrithmic interpolation to synthesize "extra" samples in between the data points to "smooth" the sampling frequency. Both of these are distinct from the adding of extra bits of random dither to "extend" word length from the 16-bit CD data to 24-bits before it is fed to the DAC chips, which is done in an attempt to better linearize the chips' low-level performance. Again, to the best of my (admittedly limited - thanks, Sean!) knowledge, neither of these techniques can actually increase the resolution of the recreated analog signal, but are efforts to extract more subjectively pleasing results out of the digital-to-analog conversion proccess. (I do promise to read that link, Sean - I hope it doesn't show me to be too far off base here!)

I will add that I recently bought a newer generation, popular "upsampling" DAC to audition against my well-regarded older unit, which does only tradtitional oversampling. While I found that upsampling did indeed change the sound, I contucted a controlled test which informed me that - at least for this particular DAC - the results were actually less faithful to the digital input signal it was being fed when upsampling was engaged than when it was defeated. I also wound up preferring, and finding to be most accurate, the performance of my reference model. I intend to write a full review of these sessions and post them separately here on the forum soon, but this experience did once again drive home the fact that it's not the type of technology used, but its implementation, that will often make the biggest difference in performance.

Zaikesman, I'm stuck for now with old machines EAD DSP7000 MKII and Monarchy DT 40b as a transport. I'm also curious what units you're going to a/b.
I'm aiming to keep transport but upgrade the DAC and the best so-far I've heard is TACT RCS(pre-amp with DAC)