Hello community!
I noticed that even if i low cut an instrument or the master at, say for example, 60hz slope 24, the sub still appear on the analyzer (like nothing or few changed). How this is possible? Scientific and over nerd responses are welcome

Thank you!

 

the signal is reduced 24db per octave gradually starting from 60Hz and below. at 60Hz probably only 0.1db is reduced (depending on the filter type, Q value etc), at 45Hz which is half octave the frequency content is reduced around 0.8db, at 30Hz which is exactly 1 octave below 60Hz, 1.25db is reduced making the total level removed in the entire octave is 24db. It's not like it's reducing the level of the signal as much as 24db per frequency! So it's not like the filter removes everything like what shown in the graphic, of course all the subs will still be there you dont really "remove" it, you just reduced the levels of content around the 60Hz and below 1-2db, more like dampened it a bit.

 

you need to switch your EQ's Analyzer to POST, instead of Pre, so that you will see only the change made by the eq. See in @Stevie Dude's above screenshot at the bottom of Pro-Q3 where the Analyzer shows Pre+Post? You are looking at both simultaneously. If you only look at Pre, you will not see any change.

 

I am not an audio engineer or an expert in analog circuits, but as far as I know, the purpose of filter design is often not to completely eliminate certain frequencies, but to reduce or increase certain frequencies (due to the design flaws of the filter and the limitations of digital processing precision, we may not be able to manufacture perfect filters). In addition, the resolution of your spectrum analyzer also affects your judgment, and a lower resolution spectrum cannot accurately present the results of the Fourier transform. Moreover, you mentioned that the Filter Slope is 24, which means that the filter itself has allowed some frequencies lower than the cutoff frequency to pass through. In sum, the design of modern filters is not perfect, and our analysis is also not perfect, hope this can help you

 

Yeah that is true, I used to understand it like that until someone corrected me. He said the sine wave test can't be accurate because it's only a single frequency, and the reduction didn't change when another sine wave is added is because the total amplitude actually doubled 2 sine waves and the filter (affected the signal) doesn't make them pure sine waves anymore. It make sense but I stopped there and didn't bother to check. You sure about this ?

 

might have to run my own little test then. I trusted him without verifying because he's a programming guy, there was a lot more things he said that made sense at the time, but I couldn't understand a thing, that's why I didnt bother to dig deeper.

 

it might simply be that the analyzer does not show the levels linear.

-24db at an F of 30 Hz sounds a lot (because you know it is a difference of 87.5% linear), but if the GUI has a db scale of -120 to 0, -24 less than before is a difference of only 25%. ;)