Juno: I read that article you posted a while back, and it makes some excellent points. And I'm also aware that you did not mean it as a counterargument to what I was suggesting. Just more food for thought.
However.
I am suspicious of claims that the current processes are now all so far beyond the reach of human hearing that it's not even physically possible to detect any difference. The same thing was said about compact disc players 30 years ago, and look how much converters have improved since then. When people just hand-wave it away, it seems like they don't fully understand. 60 years ago the word was that it was not possible for tube and solid state audio to sound different. Now we know better.
I have experimented with bit-perfect audio playback on my computers, using players (my favorite is MusicBee) that go out through the ASIO or WAVERT driver instead of Direct Sound (which is what Microsoft Media Player and iTunes use), and the difference between the default and the bit-perfect playback is stunning, much like the difference between a lossless FLAC vs. a lossy MP3. Using MusicBee and ASIO4ALL with a Realtek chip sounds better than Direct Sound through a Presonus or M-Audio interface.
Any time information goes through some sort of conversion process, it is being changed (otherwise it wouldn't be going through the conversion process in the first place). This includes "normalization," which many of us do without a second thought. Are all "normalization" algorithms the same? Probably not, we just have to trust our devs to use one that doesn't change the information in a bad way.
It's easy enough to try: using the software of your choice, rip file from a favorite CD at 44. Then use a conversion program to convert it to 48. You will be able to hear the difference between the original 44 and the converted file at 48. Convert it back to 44 and the difference will be even greater. All this up and down sampling is happening at levels that we should theoretically not be able to detect, but it's not even a subtle difference.
So, Mark, it's not just a "moral" choice. I am sure that you, with your musician's ears, can easily tell the difference. Try it! Mix to a FLAC and then mix to the highest quality MP3 and listen to the results on your mixdown system. Listen to the transients, listen to the sibilants, as well as the extreme lows. Try to detect where you placed things in the mix.
Anyway, I am not claiming that there ARE differences in DAW's, or that I can hear them, I'm just saying that it is not outlandish to think that since different ones use different methods to sum the tracks, deal with the plug-ins, etc., that those differing methods might result in something that sounds different. When we get to things like plug-in wrappers (which Mixcraft 64 uses) it gets even less outlandish.
Maybe a given DAW, when you're working with it at 48K, at some point downsamples it to 44K and then back. It's not out of the question, and the artifacts of that process are known and audible. We have no way of knowing what goes on under the hood, and in the absence of knowing, I at least try to keep the number of conversions that I AM aware of to a minimum.
