terrible_buddhi
New member
I am looking for a list of useless eq...like...on vocals anything below 70hz and above 10,000 hz should be cut completely. Does anything like this exist?
useless eq?
- Thread starter terrible_buddhi
- Start date
PapillonIrl
New member
Get an SM57.
actually it can be a very smart thing to do in the digital world.
digital converters actually make frequency nodes, just like a room does. Thats why mastering houses normally do a low cut on the mix at 30hz, and a high cut at around 16kz. This is actually the reason why 96k recording sounds better, because the nodes are tighter. Also, at 44.1, anything above 12k is in the form of a square wave, and 96k goes beyond that, so it sounds more natural.
But, in general its not usually worth worrying about it if it sounds alright. But try it out.
Also,
a list of these frequencies mean nothing. just use your ears, high and low cut until you start hearing major differences in the sound. Thats the best way.
digital converters actually make frequency nodes, just like a room does. Thats why mastering houses normally do a low cut on the mix at 30hz, and a high cut at around 16kz. This is actually the reason why 96k recording sounds better, because the nodes are tighter. Also, at 44.1, anything above 12k is in the form of a square wave, and 96k goes beyond that, so it sounds more natural.
But, in general its not usually worth worrying about it if it sounds alright. But try it out.
Also,
a list of these frequencies mean nothing. just use your ears, high and low cut until you start hearing major differences in the sound. Thats the best way.
terrible_buddhi
New member
I think I was looking more towards something to the effect of...everything below 50hz on the guitar will always just be mud in a mix, and everything above 10khz (I am just making this up for an example) is just noise.
if such things don't have hard and fast rules, then so be it.
if such things don't have hard and fast rules, then so be it.
nessbass
New member
No, I'm afraid there are no rules. I nevertheless find myself quite often low-cutting guitars or vocals, for example. But I almost never hi-cut things.
Excuse me, but I don't believe in that stuff. The reason being this: Theoretically, everything above 12k is somewhat square-like. BUT we don't hear it as such, because the difference between a sine wave and a square wave are the harmonics, right? Well, the first harmonic of a 12k square wave would be @24k. I don't know about you, but I don't hear that high.
But this is just another opinion, maybe I'm wrong as well.
So, anyone else?
David
Excuse me, but I don't believe in that stuff. The reason being this: Theoretically, everything above 12k is somewhat square-like. BUT we don't hear it as such, because the difference between a sine wave and a square wave are the harmonics, right? Well, the first harmonic of a 12k square wave would be @24k. I don't know about you, but I don't hear that high.
But this is just another opinion, maybe I'm wrong as well.
So, anyone else?
David
terrible_buddhi said:
You have to approach this kind of thing on a situational basis, and use a little bit of common sense . . . along with some actualy knowlege/education about this stuff.
You may or you may not know that most guitar amps aren't even capable of reproducing anything below 50 hz. Or at least anything meaningful. But I'm sure there are some rare exceptions, depending on the type of amp and the genre of music. This is where you kinda' have to use some common sense and ask yourself something like: "I'm playing blues.
Is my fender champ going to have some crucial sub-bass content that I'm going to be stripping it of if I high-pass it at 50 hz?" Probably not. But is there possibly going to be some hum, rumble or other mostly non-audible junk going on? Think about it (your heating, trucks passing by, rattle from the mic stand, etc.). In most home recording situations, yes, so it certainly can't hurt to high-pass filter a guitar amp. That's why they have low-cut switches built in to a lot of mixers and mic pres.
Son of Mixerman
Mix Junior
Its not true that anything above 12KHz is a square wave, maybe it looks that way because of monitor resolution.
A 20Khz wave can be decimated and reconstructed with 44.1Khz sampling rate. Part of DAC uses an filter called an interpolator, its responsible for taking the samples a reconstructing the waveform.
Mastering Engineers don't typical cut above 16KHz or below 30Khz, if there is nothing in those regions maybe. I often boost at 14Khz to 16Khz rather than cut.
Its true that some converters may cause problem because of distortion caused in the anti-alias filter section. Ive haven't heard of any converter creating nodes unless the opamp has an hpf's with too much overlap. Usually these circuits are tested an plots created to make sure it works to spec. The way software is written is probably more of a cause of artifacts than the actual converter. I was reading a tech paper on voice recognition algorithms, lots of interseting mathmatical phenomena.
SoMm
A 20Khz wave can be decimated and reconstructed with 44.1Khz sampling rate. Part of DAC uses an filter called an interpolator, its responsible for taking the samples a reconstructing the waveform.
Mastering Engineers don't typical cut above 16KHz or below 30Khz, if there is nothing in those regions maybe. I often boost at 14Khz to 16Khz rather than cut.
Its true that some converters may cause problem because of distortion caused in the anti-alias filter section. Ive haven't heard of any converter creating nodes unless the opamp has an hpf's with too much overlap. Usually these circuits are tested an plots created to make sure it works to spec. The way software is written is probably more of a cause of artifacts than the actual converter. I was reading a tech paper on voice recognition algorithms, lots of interseting mathmatical phenomena.
SoMm
Son of Mixerman
Mix Junior
Its not true that anything above 12KHz is a square wave, maybe it looks that way because of monitor resolution.
A 20Khz wave can be decimated and reconstructed with 44.1Khz sampling rate. Part of DAC uses an filter called an interpolator, its responsible for taking the samples a reconstructing the waveform.
Mastering Engineers don't typical cut above 16KHz or below 30Khz, if there is nothing in those regions maybe. I often boost at 14Khz to 16Khz rather than cut.
Its true that some converters may cause problem because of distortion caused in the anti-alias filter section. Ive haven't heard of any converter creating nodes unless the opamp has an hpf's with too much overlap. Usually these circuits are tested an plots created to make sure it works to spec. The way software is written is probably more of a cause of artifacts than the actual converter. I was reading a tech paper on voice recognition algorithms, lots of interseting mathmatical phenomena.
SoMm
A 20Khz wave can be decimated and reconstructed with 44.1Khz sampling rate. Part of DAC uses an filter called an interpolator, its responsible for taking the samples a reconstructing the waveform.
Mastering Engineers don't typical cut above 16KHz or below 30Khz, if there is nothing in those regions maybe. I often boost at 14Khz to 16Khz rather than cut.
Its true that some converters may cause problem because of distortion caused in the anti-alias filter section. Ive haven't heard of any converter creating nodes unless the opamp has an hpf's with too much overlap. Usually these circuits are tested an plots created to make sure it works to spec. The way software is written is probably more of a cause of artifacts than the actual converter. I was reading a tech paper on voice recognition algorithms, lots of interseting mathmatical phenomena.
SoMm
Son of Mixerman
Mix Junior
Its not true that anything above 12KHz is a square wave, maybe it looks that way because of monitor resolution.
A 20Khz wave can be decimated and reconstructed with 44.1Khz sampling rate. Part of DAC uses an filter called an interpolator, its responsible for taking the samples a reconstructing the waveform.
Mastering Engineers don't typical cut above 16KHz or below 30Khz, if there is nothing in those regions maybe. I often boost at 14Khz to 16Khz rather than cut.
Its true that some converters may cause problem because of distortion caused in the anti-alias filter section. Ive haven't heard of any converter creating nodes unless the opamp has an hpf's with too much overlap. Usually these circuits are tested an plots created to make sure it works to spec. The way software is written is probably more of a cause of artifacts than the actual converter. I was reading a tech paper on voice recognition algorithms, lots of interseting mathmatical phenomena.
SoMm
A 20Khz wave can be decimated and reconstructed with 44.1Khz sampling rate. Part of DAC uses an filter called an interpolator, its responsible for taking the samples a reconstructing the waveform.
Mastering Engineers don't typical cut above 16KHz or below 30Khz, if there is nothing in those regions maybe. I often boost at 14Khz to 16Khz rather than cut.
Its true that some converters may cause problem because of distortion caused in the anti-alias filter section. Ive haven't heard of any converter creating nodes unless the opamp has an hpf's with too much overlap. Usually these circuits are tested an plots created to make sure it works to spec. The way software is written is probably more of a cause of artifacts than the actual converter. I was reading a tech paper on voice recognition algorithms, lots of interseting mathmatical phenomena.
SoMm
Well, if you dont believe me, look up the Nyquist theory.
I dont understand it very much, but it is the same theory why 96k recordings also tighten the bass up frequencies up.
As far as i know, it is some of the first rules of digital recording.
Does it really matter much if it sounds right?
Maybe it does, if you have a better understanding of how digital works then you might be able to figure out ways to make it sound better.
Anyway, everyone will have their own opinions on it all.
Danny
I dont understand it very much, but it is the same theory why 96k recordings also tighten the bass up frequencies up.
As far as i know, it is some of the first rules of digital recording.
Does it really matter much if it sounds right?
Maybe it does, if you have a better understanding of how digital works then you might be able to figure out ways to make it sound better.
Anyway, everyone will have their own opinions on it all.
Danny