Upsampling. Truth vs Marketing

Sean...The sampling (your first set of dots) is at 44.1KHz. The highest audio information that exists at this sampling rate is around 20KHz, and at this frequency the music signal amplitude is very small. Therefore, unless the signal is momentarily a constant (two adjacent points the same) the in-between points will lie between adjacent points. Of course this is all overlaid with random noise that will blur the quantization staircase.

The CD recording protocol has been cited as an everyday example of the application of CRC error correcting technology, and I have seen descriptions of the CD protocol as having interpolation as a "fall back" procedure when the CRC error correction fails. Of course the second "fall back" is to abort playing the disc, and this ought to be the only time that the process is easily heard.

To tell the truth I have never actually read this infamous "Red Book" which defines the CD spec, and so am relying on what others have reported. How would I get a copy?

Bombaywalla...Your check is in the mail :-)

Sean....Homework is to read.. http://en.wikipedia.org/wiki/Reed-Solomon_error_correction
Test on Monday :-)

Sean, in fact, Jeff Kalt of Resolution Audio was marketing and using "upsampling" in his players at the time, so yes I do think some manufacturers will be honest when asked directly.

But more to the point, what do you think is so magical about 96kHz or 192kHz? Why not 88.2 or 176.4 or 352.8? I think the obvious answer is that the high rez format in DVD-A is either 96kHz or 192kHz...marketing anyone??

If you could, would you please contrast your upsampling dot graph with the equivalent oversampling dot graph? Remember that to get to 96kHz from 44.1kHz in your example you have to increase the number of dots from 20 to 43.5 dots. What you described is essentially a 2x oversampling routine with linear interpolation. The graph cannot get any smoother than the original unless you use something other than linear interpolation. Yyou are just connecting a series of dots in a line between samples, otherwise.

The main reason for adding points between the original samples is ultimately allow a more gentle analog filter. The original (really bad souding) CD players used no oversampling and analog brick wall filters to avoid the problems associated with the Nyquist limit for 44.1kHz sampling (22.05kHz) and the spurious images that get reflected back in band. These sounded horrible and led to 2X, 4X, 8X etc oversampling moving these images well beyond the audio band and allowing more gentle (better sounding) analog filters.

To paraphrase Charles Hansen, adding another digital filter (upsampling) to the chain will affect the sound; however it is certainly possible to design a single digital filter with exactly the same composite characteristics as the two cascaded filters, usually justs costs a little more money.

Anyway, not trying to be a pain in the a$$, I just think the marketing component of the choice for 96kHz or 192kHz needs to be pointed out.

The connect-the-dots metaphor is really unfortunate, because a lot of audiophiles buy into the idea that that's what a DAC does. But reconstructing an analog wave is nothing like connecting dots. More dots DOES make it easier to DRAW a wave. But as long as you have enough samples for the bandwidth, a DAC can reconstruct that wave without more information.

(BTW, the example given above didn't have enough information to do so, because it called for only two samples per cycle. You need fractionally more than two to reconstruct the wave properly.)

Imagine that, instead of a wave, you were trying to trace a straight line. The more dots you had, the easier it would be to do this freehand. But a graphing calculator would only need two points.