FabFilter

Also to add to this, the help says

"but because more incoming samples are needed to calculate a single spectrum, dynamic changes are less clearly visible. The Low value corresponds to a resolution of 1024 points, Medium to 2048, High to 4096, and Maximum to 8192 points." "A fast release shows dynamic changes more clearly, while a slow release gives you more time to examine the spectrum before it disappears. Higher resolution settings generally work better with slower release speeds."

sounds like it should say a fast release would be best for higher resolution settings. Unless there is something I am missing here.

Hi Andrei,

It's not that one type of resolution is always best. If you like a fast response, the Low resolution mode provides that. If you like a detailed look at all harmonics, use the High or Maximum mode. However, because it needs more samples to calculate the spectrum, the response is slower. This is why it works well with a slow release speed. Anyway, it's up to you to mix and match the options to your liking!

Regarding linear phase vs. minimum phase, have a look at this tutorial:
www.youtube.com/watch?v=efKabAQQsPQ

Cheers,

So just to clarify, maximum resolution is more accurate to what is actually going on while low is a quicker response, yes?

Also I still don't understand why a slow release is good with maximum, if maximum is slow, wouldn't you want a fast release? I tried it on very slow and whenever a note changes the spectrum just mushes together even if nothing is playing any more. I am concerned with accuracy because I work on harmonics a lot, what settings would you suggest for this? Thanks for your time.

Apologies for butting into this post, but:

Whenver the frequency resolution is increased, the temporal resolution will suffer, and vice versa. This is why it is a slower (smoother) update rate when there are many more frequency bands being measured.

As for the comparison between the first and second images you linked, this is to do with the way in which frequencies are derived from the waveform. A Fourier transform is used, which means that the longer the time is taken to sample the sound, the more accurate and finer the frequency measurement will be (the bands will be thinner and there will be more of them visible.) The shorter the time period that we sample the waveform, the wider the frequency bands will be, and also the faster the measurement can be done.

I find that the basic level works best for most things, and the next setting above that works great for analysing harmonics on instruments like Pianos, Guitars, Synths, etc. Anything higher doesn't tend to help much more and the update rate suffers.

One suggestion I'd like to add here would be to give a 'Wide' mode to Pro-Q the same as with Saturn. It would help greatly when notching out resonant frequencies that are closely packed together and high in number.

Thanks for the post. What I am confused about is that if you're cutting in notches, and can't see the notches, won't that cause issues? A lot of time I EQ just to have to general frequencies not get muddy because of two instruments playing them but if it turns out that there is nothing there how would I know in a basic resolution?