[QUOTE=Jaledin]
Not sure about that review, Sam Stone. For example, if “finger cymbals were confined” to a channel on a particular mastering, and on another, “spread across the stage,” it’s obviously not anything to do with the format. And the last paragraph of that review? Unless gongs aren’t actually hit by mallets, what is a recording of a gong supposed to sound like?
Nothing from me to add of substance, except disappointment at yet another nonsensical review not English in competent written.
[/QUOTE]
Yeah, this is obviously a contentious and very subjective topic, which is why it has been debated endlessly in audiophile circles. There are generally two camps - the raionalist, engineering types who understand information theory, sampling theory, the Nyquist frequency, yada yada. Then there are the audiophiles who may have little to no technical training, but “know what they hear”.
I’m in the first camp. If it can’t be measured, it doesn’t exist. If it flies in the face of existing theory, I’m hugely skeptical. And most of the time, I think the audiophiles are wrong, and are victims of subjectivism and wishful thinking.
However, we are dealing with a very complex environment here, and sometimes just saying, “The Nyquist frequency is all that matters!” is just too simplistic. In the real world, filters are not perfect. Recording and mastering practices for CDs may be different than for other sources. There are interactions between components, both on the recording side and the playback side.
The fact is, there are a lot of blind tests that have shown that people can easily distinguish between a 44.1K CD and a 96K sampled wave file. But there are some that don’t show that at all. So it probably comes down to the underlying environments - the master, the type of equipment used to reproduce the sound, etc.
Another thing that may affect the sound (just as a guess) is the require to downsample studio masters in an uneven increment. Studio masters these days are usually done at 96K or 192K - this is done to get a very signal to noise ratio so that the noise floor is still plenty low after adding in reverb, other effects, compression, etc. Then the resulting mix is downsampled for CD. I wonder if the downsampling process in a lot of studios just isn’t that great when going from 96K to 44.1. Maybe 48K, a nice even multiple, would be more accurate. In any event, presenting the output in native 96/24 eliminates the downsampling step entirely.
Complicating matters even further is that SACD in particular uses an entirely different encoding mechanism for the bit stream, which has its own pluses and minuses.