Are you able to articulate why this should sound better, at least in theory? I guess that there are potentially at least several ways one could upsample/upconvert, (a) as you have, by using pre-player software that permanently converts the files, (b) by using software in or associated with the program that plays the files, to upconvert it as it is played, or (c) through the hardware on a DAC that upconverts the bit stream once it gets there. Without getting into the debate regarding whether up-converting is worth while in the first place (of which I can see both sides in theory, yet experienced no appreciable benefit in my system), is there a reason why one method should be better than another? Don't purport to have an answer, just curious.
Background: for the past month or so, I was was using Bit Perfect to upsample during playback by powers of 2 -- ie., 44khz and 88khz files played at 176khz, 98khz files at 192khz, etc. Then I switched it all back to native resolution in integer mode. Think I prefer native, but I spotted very little difference, albeit sometimes more on some tracks than others. Given this option to upsample on the fly with only the click of a button, not sure why you'd choose to permanently alter your files by running them through an algorithm to extrapolate and add in hypothetical detail that was not recorded in the first place -- at which point you no longer have a native file, and the only way to return would presumably be by squashing it through yet another algorithm to "prune" out the large majority of the data to take it back down to 44khz. (Just thinking about that makes me a little squirrely, as I suspect that there is quite simply no going back.) Put differently, if you take a lossy mp3 and upconvert it to 16/44, you're not turning it into a CD-quality file, just a larger one. No? I'd love to hear why that's a great idea -- and perhaps it is in practice -- but having trouble getting my head around it in the abstract. Thanks.
Background: for the past month or so, I was was using Bit Perfect to upsample during playback by powers of 2 -- ie., 44khz and 88khz files played at 176khz, 98khz files at 192khz, etc. Then I switched it all back to native resolution in integer mode. Think I prefer native, but I spotted very little difference, albeit sometimes more on some tracks than others. Given this option to upsample on the fly with only the click of a button, not sure why you'd choose to permanently alter your files by running them through an algorithm to extrapolate and add in hypothetical detail that was not recorded in the first place -- at which point you no longer have a native file, and the only way to return would presumably be by squashing it through yet another algorithm to "prune" out the large majority of the data to take it back down to 44khz. (Just thinking about that makes me a little squirrely, as I suspect that there is quite simply no going back.) Put differently, if you take a lossy mp3 and upconvert it to 16/44, you're not turning it into a CD-quality file, just a larger one. No? I'd love to hear why that's a great idea -- and perhaps it is in practice -- but having trouble getting my head around it in the abstract. Thanks.