I need EQ help

psyfi

Pie Fly
Messages
14,810
Reaction score
989
Location
Aldermaston
Hello everyone I hope you can help me. I've been trying to get the bass and kick not if possible to meet. I've been working in logic and the checking the wave in cool edit. Everything looks good until I try to EQ the bass. Usually I cut a notch out of the bass where the low end of the kick wants to come through but after having looked at the wave again and listening to the sound it becomes obvious that when the notch it taken out there is a boost in the bass frequencies around is that extend further than the original note length and the runs in to the kick. I stead I have tried a high pass filter to cut the low end of the bass up to the kicks low end. The bass no longer meets the kick but the bass has now lost much of its power even with boosts between 60 to 120 hz. Dos anyone know why this is happening? Or can you tell me a good method to sort this issue out.

Here are some images to explane
This is the bass with out EQ


This is the bass with a notch cut at 62Hz

AAAGGGGHHHHHHH sort me out man.
 

Continuum

Throb Farmer
Messages
7,467
Reaction score
321
Location
Straight outta muthaf***ing Surbiton
If eq's are basically filters it looks like theres some resonance around the cutoff frequency - what eq is this?
 

psyfi

Pie Fly
Messages
14,810
Reaction score
989
Location
Aldermaston
Cheers guys. I thought it was something to do with res but didn't think that it would change the sound so dramatically as is has. I'm going through different EQs at the moment. Any recommendations for transparent and precise EQ plugs?
 

psyfi

Pie Fly
Messages
14,810
Reaction score
989
Location
Aldermaston
I've been playing around with linier shift EQ's on it and I don't get the messy res stuff going on but is there anyway to combat the delay that these EQ's add to the audio of that track or is it just going to be a case of bouncing the track of as audio with the EQ's and then placing the file back in to the song in just the right position to compensate for the delay? If so that’s a bit annoying.
 

psyfi

Pie Fly
Messages
14,810
Reaction score
989
Location
Aldermaston
Cooool. No worries people I have found that this doesn’t happen with the Sony parametric EQ So I'll use that unless some one has any better suggestions. Hooray time to start being happy again..........:party2:
 

Faction

Proto-col
Messages
21,786
Reaction score
940
Location
Bristol
In general, EQ can be thought of as working by selectively shifting the phase of certain frequencies and combining them back with the pre-eq sound; this is certainly how basic analog (R/C) EQ works. A phase-shift can be thought of as a delay whose length is shorter than the wavelength being delayed; with low frequencies (which have long wavelengths) these phase-shift delays are long enough to 'smear' the waveform quite considerably, which is what you noticed. Linear phase EQs don't tend to cause such smearing because they operate on different principles not involving phase-shift (as their name implies).

[EDIT] By the way, if anyone wants to correct my shaky knowledge of analog circuitry please go ahead; it's been a long time since my electronics A-level!
 

psyfi

Pie Fly
Messages
14,810
Reaction score
989
Location
Aldermaston
Cheers Colin for explaining what’s going on. Half off my frustration was not knowing what was actually happening. Now I can incorporate this in to my thinking.:ibiggrin:
 

photonic ballast

Junior Members
Messages
30
Reaction score
0
If that is a VST:
We are not in analog (time-continue) I think., We are in digital domain (Z), and when a filter in digital domain shifts a wave in time, the result is e−(i2pifn0)S(e(i2pif) in frequency. mmm i dont remember what exactly means that :s ... Please does anybody remeber Fourier???

I think that is the answer .... jajaja

Is strange that Fourier would say that a time sihft corresponds to the picture you have...I think (as everybody) that a change of plugin is the better choice.
 

Goran

Forum Member
Messages
312
Reaction score
16
Location
North London
There are 2 kinds of filters, FIR (Finite Impulse Response) and IIR (Infinite Impulse Response):

FIR filters use an "impulse transfer" buffer to do the job (basically a "result" buffer to apply to the incoming signal), so they're accurate, stable, and phase-linear - plus you can describe any response. However, the bigger the buffer the slower they go (lower freqencies have longer wavelength & so need bigger buffers) and need recalculating whenever some parameter changes.

IIR filters approximate the transfer by applying the filter function to the current sample only, using a few previous samples to calculate the new (filtered) value. This is why they're also called "recursive" (output is based on previous output values as well as the input value). IIR filters are very fast and parameters can change without much penalty, so a huge majority of all digital filters are IIR-based. Unfortunately, this method introduces a delay which results in the "smearing" of the frequency response.

If you have Cool Edit/Audition, go to "Scientific Filters": there's an option to show this delay which is a great way to see what's going on. Basically, the stronger the filter (more poles), the stronger the smearing... Especially prominent on bass, as wavelengths run into 20+ milliseconds.

Hope this helps :Smile3:
 
Top