Here's a vid from Fab Filter showcasing their Q2 plugin and the "cramping" of bells in typical EQ plugins at lower sample rates, due to the Nyqvist filter cut off, versus the Q2
This has nothing whatsoever to do with the sampling rate for recording.
Most plug ins today use internal oversampling, so when you apply the plug in, there is no sound difference whatsoever between material recorded with 44k, 88k, etc.
Nothing against the (very good) Fab Filter EQ, but their advertising video makes it sound like it's the only EQ on the market that uses internal oversampling, and offers a linear phase option (linear phase is not inherently better, it's better for certain purposes, worse for others..).
I doubt that anyone can actually distinguish between audio recorded at 44k and 48k. If you think you can, do a double blind test: take any track recorded at 48k, and bounce it, once to a 48k target file, once to a 44k target file. Then do a double blind test, to hear if you can distinguish between the files (and please tell me what you find - I'm genuinely curious..).
To do a double blind test, you can either use Apple's free AUlab (if you're on a Mac - how-to video here), or any kind of ABX test app.
Slightly off topic: most DAWs today work at 32bit float internally. In a recent video I watched, Andrew Scheps explained that you can apply a Trim plug in to any DAW track, boost the signal by 80dB, so everything in the DAW's mixing board looks like it's peaking and distorting like crazy. Then you insert another Trim plug in to the sum that reduces the gain by 80dB. Everything will actually sound fine, no distortion whatsoever, due to the way calculations are done internally. Counterintuitive, if you think in analog gain staging terms...