I set the trim knob, to the highest possible setting without clipping or peaking.
On the Seasound (the a/d converter) I set the line input levels so that in Sonar it is as high as possible without going over 0.0db.
Is there anywhere I should be leaving any headroom, or am I doing this correctly?
The only reason I use the external mixer is because I can increase my amount of input tracks from 1 to 2! (exciting, yes I know.)
First, turn everything down on the Be#^&* and Seasound, and center the channel tone knobs.
Also, not sure what "instruments" you are running through this, but they should have a good level on their outputs, but sometimes too high a level can generate noise. Too low a level will force you to turn the trims too high, generating noise. Check that.
Set the Be%$&*#r's channel and master faders to their optimal settings first. THEN adjust the trim on the inputs. You don't want channel and master faders too low OR too high, or trim too low or high. You want those faders optimal, and the trim is only used for as much input as necessary.
Next, if you insist on using mixer EQ, set those, and adjust the levels accordingly, because adding or attenuating, especially bass, will affect the overall level.
Next, adjust the Seasound inputs. Headroom there will depend on what bit/sample rates you use. If using 16/44.1, you might want a bit higher signal, but you also want to make sure it isn't too high. One big over, and the track is toast.
I don't know if the Seasound has input volume knobs (I think it does)? Do those move the software input faders when you turn them? If not, set your software faders to an optimal level (fairly high), and then adjust the knobs. You may have to experiment back-and-forth to get a level where neither the software or hardware faders are too high or low, and listen for noise.
If they do link to the software faders, just set them.
I'm not sure I understand the statement about "I can increase my amount of input tracks from 1 to 2!" Doesn't the Seasound have two inputs? If you only want two inputs, why not remove the weakest link...that mixer...and record directly to the Seasound? I'm not sure that mixer is doing you many favors in terms of noise or tonality.
"Next, if you insist on using mixer EQ, set those, and adjust the levels accordingly, because adding or attenuating, especially bass, will affect the overall level. "
When you say adjust the levels, do you mean on the master fader, the channel fader, or the channel trim?
Quote:
"Next, adjust the Seasound inputs. Headroom there will depend on what bit/sample rates you use. If using 16/44.1, you might want a bit higher signal, but you also want to make sure it isn't too high. One big over, and the track is toast. "
What would be the difference in headroom guidelines be between 16/44.1 and 24/44.1? (I am rather newb). As a rough guide, how many dbs should I leave between the peak of the track and 0.0?
Another question.... I read somewhere online that you would want different amounts of headroom based on each particular intrument and also where you want it to sit in the mix. Any truth to that statement. What I have done so far is pretty much set everything as high as I can without clipping.
Sorry about the beh*^$*&^, its on loan (for free) from a friend. Good knowledge on the Seasound, but alas, I bought that used for very cheap because the phantom power switch doesn't work, so I don't really have two inputs. Using the external mixer is kind of nice because I use the L/R inputs to record two mono tracks or one stereo. Unless there is a better way of doing things, I don't know it yet without buying more/new equipment. I am only using this as a project studio, not anything professional...but I might as well get as best results as possible.
Thank you again for your help!
Codemonkey Recording Org Pro Audio Group
Joined: Dec 11, 2007
Posts: 771
Location: Scotland, UK
"how many dbs should I leave between the peak of the track and 0.0?"
Heard a couple of people say -6dB which leaves you room for "enthusiasm".
Do you need phantom power on an instrument? If not, then drop the Behrwhatsitdoingerr. TBH I can't imagine needing to apply phantom power on a guitar channel unless it's going into a DI.
If you ask me, software track levels are for the mix. Get the track in at an optimal value to work with - then turn it up or down.
_________________ Curious button pushing Church sound guy.
1) When you say adjust the levels, do you mean on the master fader, the channel fader, or the channel trim?
2) Another question.... I read somewhere online that you would want different amounts of headroom based on each particular intrument and also where you want it to sit in the mix. Any truth to that statement. What I have done so far is pretty much set everything as high as I can without clipping.
3) ....I bought that used for very cheap because the phantom power switch doesn't work, so I don't really have two inputs.
1) It's a balancing act. The reason you want the channel and master faders started at optimal is to GIVE them a bit of breathing room. They don't have to stay perfectly at "0" (or whatever), but none should go too far off. Once you get the input trim right with flat EQ, and then change EQ, use a bit of channel and/or a bit of master to tame levels, if necessary..
2) Get good recording levels-with some headroom- first, to allow for the best possible use of the entire resolution of the AD converters. Then you can bring down the levels when mixing...whichyou'll probably need to do anyway since all the tracks added together through one stereo buss will be louder than any one track alone. you've captured it at its best S/N ratio, (theoretically), and when you turn each track down in the mix, you (theoretically) lower the noise floor. this is more dramatic on a tape, but it works kind of like that also in digital.
3) You don't need "phantom power" for anything but condenser mics, so anything else you can plug in...DI's, keyboards, mixers...shouldn't matter. Unless a channel just doesn't work. Just because it doesn't have phantom power doesn't necessarily mean the channel is useless.
2) Get good recording levels-with some headroom- first, to allow for the best possible use of the entire resolution of the AD converters. Then you can bring down the levels when mixing...whichyou'll probably need to do anyway since all the tracks added together through one stereo buss will be louder than any one track alone. you've captured it at its best S/N ratio, (theoretically), and when you turn each track down in the mix, you (theoretically) lower the noise floor. this is more dramatic on a tape, but it works kind of like that also in digital.
I don't know about how much headroom by "some headroom"... do you mean on the scale of like 20-40dbs, or 3-10dbs or like .5 to 1dbs. I don't have any kind of starting point besides the answer of 6ish by the guy who replied above (thank you btw).
What do you mean by 'S/N Ratio'?
What determines a good S/N Ratio, just by listening?
At some point, it helps to know how to use a very special research tool in order to learn more in-depth answers to questions that were briefly answered. Here's a link to it:
Oh man. I am an idiot. I hate when people don't google things. Now I am a hypocrite. I wasn't thinking clearly, I was actually sitting here waiting for a response, when I just could have looked. I got swept up by the magic of a forum. Seriously though, thanks.
I could tell you were getting amped by the number of posts in a short period, and just had to give you a gentle "wink, wink, nudge, nudge" to try to calm you down a tad. Search function at the top of these forums is very useful, also.
Kapt.Krunch.
Codemonkey Recording Org Pro Audio Group
Joined: Dec 11, 2007
Posts: 771
Location: Scotland, UK
I can't give you a solid figure, mainly because I record a live band. Levels get ridden on the way in, to keep it above the rotten noise floor, and below the clipping point.
Kapt.Krunch:
"when you turn each track down in the mix, you (theoretically) lower the noise floor."
Surely by piling tracks together it combines and raises the noise floor anyway.
20-40dB of headroom? My noise floor is at -50dB on a good day.
_________________ Curious button pushing Church sound guy.
Kapt.Krunch:
"when you turn each track down in the mix, you (theoretically) lower the noise floor."
Surely by piling tracks together it combines and raises the noise floor anyway.
Exactly. If you have a track that is peaking at just under "0", and it has a small amount of hiss in it, and you turn it down to, say, -12, for a mix, then you are lowering the track instrument level AND the noise level.
If you have 16 tracks, all with some noise, and mix them together, the noise will add together at their respective levels. But, since many of the tracks were probably lowered from a recorded higher level to mix, the noise won't be as much as if the tracks were all kept high and mixed with the noise at its original level. You bring the track down a bit, you bring the noise down.
But, you want to use as much of the resolution of the AD converters as safely possible when recording (especially at lower rates) to achieve a better representation of the waveform.
Back in the tape days, they slammed the tapes for two reasons. To get a nice little compression out of it, and because they knew that when they lowered the track in the mix, the noise also lowered with it. Then, when they added it all back together, the noise was lower than it would have otherwise been, even though all the tracks added their own noise in.
Noise can be handled several ways. If the signal is loud enough, it masks the noise, somewhat. Just trim the noise from when an instrument is not playing. But that can be tricky.
Use a noise removal program on a track, lightly, to remove the worst of it. Then trim, being careful not to cut off tails, etc.
If you clean up each track, delicately, before mixing, you'll have less noise in the quiter spots, and the music should mask the rest, somewhat.
It's best to eliminate noise as much as possible before hitting "Record", but it's not always possible. All electronics generate noise, and all microphones may pick up unwanted noise. All you can do is get signals and levels as good as possible, and deal with it.
Open to getting clobbered if I have anything wrong.
A hardware noise gate should only be used before time-based effects, or to create a "gated" instrument. Cheap ones, especially, can be difficult to set and control. But if an unwanted noise does pop out during a planned silent passage in a recording, you can always nuke it.
If you are using a keyboard with onboard reverb/delay, or are running a mixer signal or guitar with reverb/delay into a recording, you shouldn't use a noise gate after reverb/delay.
There are noise reduction programs in some software packages. On a single track, I'd prefer a bit of that, with some careful hand-editing of the waveform after, if needed.
Kapt.Krunch
Codemonkey Recording Org Pro Audio Group
Joined: Dec 11, 2007
Posts: 771
Location: Scotland, UK
I found a gate indispensible on some guitar tracks I had. But the guitarist was cutting his strums short - no ring, creating a dead stop to let the vocalist carry the line. So I gated, EQd, etc. and it worked, although I had a bit of jiggery pokery at the start and end involving chaning the threshold to below the noise floor. During the song was fine.
BTW you can get Gated Reverbs but I have no idea why anyone would want one.
_________________ Curious button pushing Church sound guy.
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