Brickwall EQ

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well, one example: when synthesizing kicks or nice resonant filter modulation, cutting everything below 30hz really frees up headroom and tightens up the low end. I was actually amazed at how much stuff is down there.

I don't think that the OP really means a true -infinity HP but something north of 24db slope.


@OP There also was that thing called Engineer's Filter that deadmau5 mentioned. Win only i think.

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TDR Nova offers very steep slopes, the free edition up to 72dB/oct, the GE (€40.00) up to 120dB/oct.

I agree that the term brick-wall is nonsense, these are just very steep filters.

Sound wise, 24dB/oct is already pretty strong. Steeper stuff is more about phase distortion related sound design (i.e. make a "boom" out of a "poc").
Fabien from Tokyo Dawn Records

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Do you remember Rubberfilter? It goes up to 384 dB/octave

http://www.kvraudio.com/product/rubberf ... de/details
It's easy if you know how

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i remember rubberfilter. i still use it and i still don't understand what it does :?:

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inkwarp wrote:i remember rubberfilter. i still use it and i still don't understand what it does :?:
Rubberfilter is a plain Low Pass Filter, High Pass Filter, Bandpass Filter or Notch Filter.
The buttons at the top declare the mode, which activates the middle row for "highpass" (aka LowCut), and the bottom row for lowpass (aka high cut). Bandpass activates both bands, and "Notch" pretty much inverts it's position.

You can also trigger this filter into Stereo Link Mode, else only one of the stereo channels is affected.

It has since been discontinued.



Melda Production's MBandPass is the spiritual successor of it.



Still, as meleco_go and FabienTDR already pointed out, why such steep filter usage can only cause issues. Unless wanted on purpose.

Else, yes - "brick-wall EQ" is a nonsense term, just like "Multi Band Level Meter" is (*cough* 1/3rd Octave Signal Analysis). People are getting really creative these days. Until someone picks up that term, then declares it as "fact", and others spread it as the next best thing since slicing and toasting bread at the same time.
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mnmlfrk wrote:well, one example: when synthesizing kicks or nice resonant filter modulation, cutting everything below 30hz really frees up headroom and tightens up the low end. I was actually amazed at how much stuff is down there
Yes, but you dont need to cut it off with infinite slope, there's rarely even need to cut it at all, using shelf eq is much better suited usually...
Soft Knees - Live 12, Diva, Omnisphere, Slate Digital VSX, TDR, Kush Audio, U-He, PA, Valhalla, Fuse, Pulsar, NI, OekSound etc. on Win11Pro R7950X & RME AiO Pro
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Kicks absolutely need some sub content in the 20-30hZ range otherwise, even if you don't think you can hear it, you certainly won't FEEL it.

Unless you like weak kicks, ofcourse...

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Oops. Never mind.

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meloco_go wrote:Such EQ would have ringing for ages.
What do you mean by "ringing"?

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Chris-S wrote:
meloco_go wrote:Such EQ would have ringing for ages.
What do you mean by "ringing"?
The easiest way to understand is to see for yourself. Take something like distorted guitar. Cut it so it starts and ends abruptly. Now apply any kind of filter with steep responce (LP, HP or notch, doesn't matter). Zoom to the end of waveform and see. If the filter is steep enough you will see ringing. If the filter is linear phase, you'd see ringing before as well.
On a more fundamental level signal "exists" in two domains (time and frequency) which are (inversly)related. A signal which is localized in frequency domain (such as simple sine) is infinite in time domain, a signal which is localized in time (such as impulse) is infinite in frequency domain. Ordinary signals are somewhat limited in both. But if you try to limit a signal in one domain, it will spread in the other. E.g. if you take sine and cut it abruptly, a lot of "junk" will appear in frequency domain, and if you take a sharp signal (like a drum hit without much ringing) and limit it in frequency domain (with a steep filter) it will spread in time domain.
Hope I was clear enough.

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come again??

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inkwarp wrote:come again??
Um, so there's two (at least) ways to look at a signal -- a waveform (i.e. dependence of amplitude over time) and a frequency spectrum (amplitude over frequency, for a full description a phase must e involved too but I'd skip it for simplicity). They are connected. If there is a sharp change in a waveform (a transient) there would be a lot of components in a frequency spectrum. And if there is a sharp change in a frequency spectrum of a signal (a solitary sine is an example but if works the same for a sharply filtered signals too) the waveform would be sort of "diffuse". I.e. there is a minimal possible length of the signal which is inversely proportional to the bandwidth (probably I'm oversimplifying things a bit, but it's like that). The narrower the bandwidth, the longer the minimal possible length of the signal. There are all sorts of the consequences here. One is plainly obvious when listening to low-res audio -- first things to suffer are transients. Another consequence, is if signal is EQ'd with a very steep filter or high-Q peak there would be ringing.

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hey
i was only half-joking in my reply, but thank you for the explanation. very clearly explained...

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Not judging, but thats a very steep cutoff!!.

Another technique you could use is filter the track(s), sum them in a buss, filter that, then finally finer on your main buss, if necessary.

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inkwarp wrote:hey
i was only half-joking in my reply, but thank you for the explanation. very clearly explained...
That was embarrassing. Nice way to thank someone for being helpful. :wink:

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