How do Anti-Aliasing-Filters in audio signal processing work?
I think you make this needlessly complicated. Let's do a specific example: let's say we want to go from 48 kHz up by 4/3 to 64 kHz and your signal bandwidth is 20 kHz. 

In the first step, you would up-sample by a factor of 4 by inserting 3 zeros between each sample. Your new sample rate is 192 kHz. Your original spectrum is preserved and you get mirror spectra centered around 48 kHz &amp; 96 kHz.

To eliminate this you need a lowpass filter with cutoff of 20 kHz at 192 kHz sample rate and sufficient attenuation at 24 kHz.

Finally you just down sample the result by factor of three by simply throwing away every second and third sample. This works without aliasing, since the lowpass filter has already eliminated all content that would alias.

Regarding your graphs: in general I would avoid drawing anything past the Nyqusit frequency. It's just periodic repetition and doesn't add any useful information and it can get confusing.