Chris, when I was in high school I took a few years of German. At one point my mother decided that she wanted to try to help me with my homework, so she picked up my textbook and started trying to read a few sentences. Not knowing any German, she came across the word "die," which is simply one form of the definite article (like our word "the," to which it is closely related). But to her, this was irrevocably the verb form of "death," and she became convinced that this had been part of the problem with Germany in the Hitler era--that they threw this terrible word around so casually, they became desensitized to it.
In German, of course, the word "die" has nothing to do with death. But my point is, you can come up with some pretty wild conclusions, as my mother did, and feel that you have clear evidence for them, if you interpret something in the wrong framework of meaning.
You are doing a similar thing by interpreting sampled audio as it was still a continuous-time, analog signal. According to the behavior of continuous-time signals, everything that you say makes sense. However, a discrete-time representation of a signal is different, and must be interpreted differently. I'd like to suggest that you read a little about Shannon's sampling theorem, because I think that otherwise we will only be able to talk in circles. As I recall, Wikipedia has a pretty good article on it.
(added the next morning): I also wanted to say that some recording equipment can indeed sound different when run at different sampling rates--but when that occurs, it isn't necessarily because of those sampling rates alone. As an example, at the end of the 1980s the Sony PCM-2500 was by far the leading studio DAT recorder; it could record at 32 kHz, 44.1 kHz and 48 kHz. For many engineers it was the first piece of digital recording equipment that allowed them to compare the three sampling rates--or so they thought.
There was general consensus that there were audible differences, and that the 48 kHz rate sounded better than the 44.1 kHz rate. However, not only the sampling rates but also three (or six, for stereo) complete, separate analog anti-aliasing filters were also being compared, since the ICs which were later introduced for digital filtering didn't exist yet. And it turns out that if you bypassed the A/D and D/A converters in the deck--effectively isolating those filters and simply listening through just them and the other analog circuitry of the deck--they sounded rather different. The 44.1 kHz filter in particular could have audible levels of a kind of hard-edged distortion when it was pushed. (Apogee, the company which is well known now for its A/D converters, got its first foothold in the studio market by offering better-sounding replacement plug-in filter modules for the PCM-2500 and other similar equipment.)
Many people formed their own, honest opinions about sampling frequencies, like I think that yours are--but those opinions weren't based on what those people thought they were based on, like I think that yours probably aren't. Similar stories could be told about "tube vs. transistor" comparisons (e.g. Neumann U 67 vs. U 87) and "CD vs. vinyl" comparisons where extrinsic factors that people didn't know about had a large, hidden influence on their decisions. It's not always easy to set up listening comparisons so that you really are testing just for the one thing that you want to test for. Unfortunately for consumers and even most studio engineers who can't get into the insides of the equipment, it is sometimes quite impossible for them to do so at all.
--best regards
Topic: Transfer from Analog to Digital (Read 8565 times)