#1
When recording with my sm58 there's always this airy background noise. I've tried turning off the a/c, turning off laptop fan, etc, but it's always there. I know that mics have this sound a lot, so I was wondering how you can make the recordings clear. I've also tried some noise reduction on audacity and mixcraft, but I can't get rid of it without effecting the quality of the recording. maybe I just don't know how to tweak it. Any help, thanks
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#2
How are you connecting the microphone to your laptop?

Are you running the mic with high pre-amp gain?
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#5
It might. I'm very inexperienced, so I'll try to figure out a noise gate.
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#6
Just EQ it. You're putting a low pass filter on there, right? If not, start at 15000, and if it's still there, you can drag it as low as 10-12000 without really hurting your sound quality. This is pretty standard stuff.
#7
Quote by KevinGoetz
Just EQ it. You're putting a low pass filter on there, right? If not, start at 15000, and if it's still there, you can drag it as low as 10-12000 without really hurting your sound quality. This is pretty standard stuff.

15k will have little to no effect on anything. That's near the upper limit of human hearing. Fan noise is probably around 1-3k (judging from a quick listen to my laptop; I'm probably off by a bit). If EQing could remove noise effectively, we'd have no reason to try to eliminate it.
#8
I use reaper to do this. you can use the built in plugin ReaFir, set it to subtract mode, put the check mark in the box that appears and then turn monitoring on for a few seconds and then uncheck the box. Don't just use it like that though. Look at the frequencies and then with the normal EQ mode, try to EQ it accordingly. A little goes a long way. Always do A/B tests and Hi-Pass and Low-Pass.
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#9
Quote by Cavalcade
15k will have little to no effect on anything. That's near the upper limit of human hearing. Fan noise is probably around 1-3k (judging from a quick listen to my laptop; I'm probably off by a bit). If EQing could remove noise effectively, we'd have no reason to try to eliminate it.


That's not at all accurate. Sorry, but if you look at the frequency response of an SM58, it's got a huge bump around 12k. That's the "airy background noise" he's hearing. I've tested this repeatedly with my own 58. If you place your LP between 13-15k, the rolloff curve will cut that noise. If it's a particularly bad bump due to poor gain staging, you may need to place it lower, closer to the 13k end of the spectrum or maybe as low as 10-12k. But there IS noise in that area, and that IS the noise he's hearing. Try it. Low pass at 15k, then take it off. You'll feel a difference, a claustrophobic sort of feeling, especially if your mix features prominent cymbals or distorted guitars. There IS information there that needs to go away.

Also, the human hearing range typically tops out around 20k, not 15k. Saying 15k is too close to the limits of hearing to be of any use, is literally like comparing 2k to 7k, or 5k to 10k. Pretty drastic differences between those points.
#10
Quote by KevinGoetz

That's not at all accurate. Sorry, but if you look at the frequency response of an SM58, it's got a huge bump around 12k. That's the "airy background noise" he's hearing. I've tested this repeatedly with my own 58. If you place your LP between 13-15k, the rolloff curve will cut that noise. If it's a particularly bad bump due to poor gain staging, you may need to place it lower, closer to the 13k end of the spectrum or maybe as low as 10-12k. But there IS noise in that area, and that IS the noise he's hearing. Try it. Low pass at 15k, then take it off. You'll feel a difference, a claustrophobic sort of feeling, especially if your mix features prominent cymbals or distorted guitars. There IS information there that needs to go away.

True, but EQing out the air, again, is also removing information that doesn't need to go away. I'm a fan of that airy vocal mix that some albums have; when the instruments cut out, again, it sounds more open and free.
Still, in this case, there's not much choice.
Quote by KevinGoetz

Also, the human hearing range typically tops out around 20k, not 15k. Saying 15k is too close to the limits of hearing to be of any use, is literally like comparing 2k to 7k, or 5k to 10k. Pretty drastic differences between those points.

I take issue with this. Frequency is measured on a log scale for basically everything, because that's how our ears interpret it. Octaves double the frequency, not offset it. It's like comparing 2k to 2.7k, not 7k. Or, following that pattern, 100 to 133, rather than 5100.
Last edited by Cavalcade at Sep 18, 2013,
#11
Quote by KevinGoetz
Saying 15k is too close to the limits of hearing to be of any use, is literally like comparing 2k to 7k, or 5k to 10k. Pretty drastic differences between those points.

You don't know how human hearing works do you? the difference between 15k and 20k is actually the same as the difference between 1.5k and 2k, or 150 and 200, or even 15 and 20. Everything to do with how we perceive sound is logarithmic, so it deals with multiples of 10. That's why if you look at any frequency graph, it clumps together at the high end. Because everything is in multiples of 10 (next time you see one, look at the distance between 100, 1k, and 10k, they should be the same).
#12
Quote by chatterbox272
You don't know how human hearing works do you? the difference between 15k and 20k is actually the same as the difference between 1.5k and 2k, or 150 and 200, or even 15 and 20. Everything to do with how we perceive sound is logarithmic, so it deals with multiples of 10. That's why if you look at any frequency graph, it clumps together at the high end. Because everything is in multiples of 10 (next time you see one, look at the distance between 100, 1k, and 10k, they should be the same).


Fair enough, I stand corrected, and I apologize. Though while I may not be well versed in the science of human hearing, I'm very familiar with audio engineering itself, enough to know that if he's hearing a hiss and hasn't low-passed his source, check there first. It's not fair to say that LP'ing 15k wouldn't have any effect. That's simply not true, because the rolloff curve does tame the noise below it as well, and will also cut into the octaves of certain fundamentals that perhaps do more harm than good.

Edit: You're not REMOVING that information, only lowering it so it doesn't overpower the rest of the signal, which is what's happening here. I'm not saying to cut by 24db in addition to the low pass or something absurd like that, only let its rolloff do what it was meant to do and bring those high frequencies into alignment with the rest of the presence region.
Last edited by KevinGoetz at Sep 18, 2013,
#13
You're all arguing about fixing a problem that shouldn't exist in the first place!

The 58 is a pretty quiet mic and even the cheapest m-audio interfaces don't have overly noisy preamps. Seems like some sort of USB grounding issue, or maybe dodginess in the interface.

A realistic possiblity is the windows audio drivers adding extra gain on top of the interface. If I record with directsound instead of ASIO in a DAW I get a far hotter signal - assume it's due to processing Windows does on microphones.

An unedited sample would be very helpful.
#14
You need to silence all possible acoustic noise that is actually from the mic hearing it in the background. That's a no brainer.

If you're getting noise from the device having a loud noise floor, then you need to gate it out.
#15
Here's a question: how close are you to the mic.

If you're far away and singing really quietly you'll have to turn up the gain a load which will decrease your signal-to-noise ratio.

TS are you singing close to the mic? Like an inch away? Sorry, I assumed you're singing because it's a 58. What are you recording and how close is the mic to it?
#16
Quote by tim_mop
Here's a question: how close are you to the mic.

If you're far away and singing really quietly you'll have to turn up the gain a load which will decrease your signal-to-noise ratio.


In a way, this is true, but in another, it's not. There's almost 2 stages of noise floors going on, at least on my interface and from my experience.

Turning up the preamp will raise your signal away from the digital noise floor, but will raise the analog noise floor.

Basically calibrate the preamp to where being close to the mic is as loud as possible while giving it a bit of a buffer to prevent clipping. This will give you the best signal/noise ratio.

I've found that using a compressor effect to raise your volume instead of the mic gain creates a lot more noise. I try to use a balance between the preamp and a compressor effect to give me a nice buffer to be loud without clipping the mic.
I put a bit of a noise gate in the beginning of the effects chain to kill any noise that does get through.
#17
Thanks guys. I have a good ear, but I'm honestly not an audio engineer. I don't understand a lot of terms.

I know an sm58 is primarily a vocal mic, but it's all I have right now so I've tried to record acoustic guitar with it. But I have fiddled with the placement of the mic and it has the background "air" sound no matter where it is, and no matter what instrument/vocal I use it for. I also don't have a preamp, the m-audio only has the mic input knob. So basically the only tools I have for removing the noise are audacity and mixcraft. So does anyone have any program specific instructions? Thanks.
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#18
The M-Audio IS a preamp, an interface is (essentially) just a preamp and AD/DA converter in one unit. If noise is an issue then you (probably) want to turn the mic input up as far as possible without clipping (there should be a light on the interface that will tell you when it clips, otherwise mixcraft will show it on the volume meter).