The less the content changes, the less noticeable will be sample rate differences.
The Nyquist Theorem says that given an INFINITELY repeated waveform, the maximum possible frequency is half the sample rate.
That means that synthesized music that has only very slow changes will likely be better tolerated at low samples rates than sparse vocals and acoustic instruments with rich harmonics and rapidly varying sounds.
As many have quoted here over the years, tests have shown that most listeners would be hard-pressed to tell the difference between samples rates, presumably after only one conversion between the two. So there may not be much point in making your music available in anything greater than CD quality (44.1kHz 16-bit).
However, processing at different sample rates throughout a whole project is another matter, as there is lots of processing going on. That is why Universal Audio internally upscale to 192k on most of the UAD plugins to ensure the accuracy of their emulations. It is engineering common sense, as one cannot expect to have an accurate output if performing thousands of processing steps at the same accuracy as expected at the end.
With our music, we are mainly recording sparse vocal and acoustic instruments using condenser microphones, which pick up EVERY little sound. I record at 192k, so I can clean up the best with RX3. It also allows the reverbs to create the vocal richness (UAD EMT140 plate) and realistic room sound (REVerence LA Studio).
Of course, higher rates require more system resources, which is giving me aggravation when mixing down on my current system. The solution for me is only upgrade!
Also, if using samples, most of which are at 44.1, they will be up-sampled on the fly EVERY time, using more CPU resources. So it is best to freeze or bounce to audio once an instrument part is not likely to change.