The 'fun' part is that you are showing signals that don't exist in any music files. All 44.1 recordings should have passed through a steep brickwall filter on the encoder side (when down-sampled properly from the recording format) so frequencies that are reaching Nyquist should not be there. Square-waves that were recorded through the analog input or are down-sampled won't have such steep rising and falling edges.
Besides, the upper treble 'energy level' in recordings (almost) never reaches near FS. So stuff that 'mirrors' back due to slow filtering also will be lower in amplitude (unless one is daft enough to use filterless at 44.1/48)
That's why I suggested to use (recorded) music and nulling software if you want to find out where the audible differences are.
...
As I mentioned earlier, with most music I can't hear the difference between these filters. But there are a few tracks where I can. The 500 Hz square wave is one of them. What's interesting is the perception of the difference is the opposite of what I expected: on the slow filter, this signal sounds like it has just a touch more "edge" or "buzz", similar to hearing a pure midrange sine wave that has a barely perceptible (say 1%) distortion. Yet the FFT (and the FR curve) shows the reverse, the sharp filter having slightly more HF energy. Perhaps what I'm hearing is a phase shift that is accentuating certain frequencies, which wouldn't show up in FR or FFT.
Also, the step responses from REW are more similar than the actual square waves. They don't show the slow filter's over-shoot and longer ringing. This makes me wonder if deriving the impulse response from the sweep (which REW does), while mathematically exact, is less than exact in the real world. So you get a mathematically predicted step response that doesn't necessarily match actual step response. Maybe there's a different reason they don't match.
The square wave response here shows that the slow filter on my DAC is minimum phase. But not all "slow" filters are minimum phase. Some are still linear phase, but use a wider transition band (like filters #1 and #5 on the WM8741). This leaves me wondering what that would look like in measurements. From a math & engineering perspective, that would seem to be the ideal filter.
Anyway, I normally listen to music, not test signals. On those few musical tracks where the difference can be heard (such as the Drums & Bells snippet I posted earlier), it's so subtle I honestly can't say which is more realistic or express any preference. So the experiment shows the switch actually does
something, but it doesn't make any difference
to me in actual musical listening to
music. So the experiment was educational, but I'll leave it in the sharp position simply for peace of mind knowing it's more correct.