In the signal follower are two sliders: LP Filter and HP Filter
If the HP filter value is bigger than the LP filter, all sound (or most of it, in most cases) is filtered out.
I understand how it works, but the design is kind of “cheap and barely good enough.”
They should at least be linked, so that if you slide LP to the left and pass the HP, the HP automatically moves with it. Vice versa for the other slider.
Optionally, design a new GUI slider that can do “range” selection?
Hmm, you really think so? I see it as 2 filters in serial (and their only purpose is to narrow down the signal that the Signal Follower should react on). Haven’t thought of it before, but if I wanted to automate these, wouldn’t that be problematic then if the LP affected the position of the HP and vice-versa?
My point is that when you overlap LP over HP, you cancel everything out.
From an interface perspective, it doesn’t make much sense. If you are a newbie and don’t know what you are doing, you could spend a lot of time having no sound and not understand why.
PS: Not that important, i’m not going to cry if this doesn’t get changed, just wanted to point out that it could use more polish from a UX perspective.
Optionally, I was the newbie trying to figure out where my sound went for a few minutes…
How about if LP is lower than HP then you get notch rejection? It’s what these values would seem to indicate (although may be slightly confusing and would almost certainly produce a click if automation caused it to switch modes by going from one to the other, not that I can imagine people often wanting to on purpose.)
I like it the way it is: more automatable and more responsive in this respect than a range finder. Also takes less thought to deliberately overlap the two when you want a quick blockout.
this prompts a question though: What are the specifics of these filters anyway. Does anyone know what the curves are?