I'm confused about why the methodology that they talk about is innovative. They talk about shifting when a sound event of a given frequency arrives, relative to

I'm confused about why the methodology that they talk about is innovative. They talk about shifting when a sound event of a given frequency arrives, relative to another sound. But with digital audio, esp that you can prepare in advance, isn't this pretty simple? Like, given signals A and B, apply a band-pass filter to A (for the desired frequency window), truncate off the first k samples based on the time shift of interest, and add to B. Even if you do need it in real time, why isn't this equivalent to delaying B by k samples? I must be missing something, because the researcher makes this sound like a cool new technique.