dBVU, dBFS, noise, and YOU! (correct mixer gain)

  • Thread starter Thread starter tarnationsauce2
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Im quite late into the discussion but unfortunately I feel as if im not understanding something. Actually I feel completely clueless as to what i really know.

So i run my mic into my dmp3 and then into my audiophile 2496.

I have the db meter in my program reading -18db .(just speaking into my mic)
Heres where i get confused ... my dmp3's VU meters arent really moving.

Now what am i minunderstanding completely . I feel like i should be able to get my preamps VU meters peaking at 0 db (because thats supposed to be its "comfort spot" .. i think) .. but once i get my meters to 0db , my software indicates that im completely clipping. I want to learn this much better than i know it right now. Any help would be really appreciated. I'm sorry if asking the personal question is selfish and digressing off topic a bit. Thank you.
 
Your M-audio card has -10db inputs, your preamp has a +4 output. That is the problem, you will have to go with the software level.
 
So farview , youre still trying to say that in my situation with dmp3 to 2496 , my best case scenario would be to run my signal in at -18dbfs regardless of how low it reads on my dmp3? sorry this is taking a while to sink in my brain.

as lame as it sounds , i couldnt fall asleep last night because of it. I would get out of bed cause i wanted to read more about it. A/D conversion is AWESOME!

EDIT: also , would i be wrong for saying..in my situation.. 0dbvu = -4dbfs
 
Operating levels aside -

Recording "too quiet" will not damage the audio. Recording "too hot" will. If you have the extra room, you can bring it up to a point - But at -4, you're still going to have to turn it down considerably before mixing -

Given the situation, I'd err to the conservative side.
 
Erockrazor said:
So farview , youre still trying to say that in my situation with dmp3 to 2496 , my best case scenario would be to run my signal in at -18dbfs regardless of how low it reads on my dmp3? sorry this is taking a while to sink in my brain.

as lame as it sounds , i couldnt fall asleep last night because of it. I would get out of bed cause i wanted to read more about it. A/D conversion is AWESOME!

EDIT: also , would i be wrong for saying..in my situation.. 0dbvu = -4dbfs
Click to expand...
See if your dmp3 has a switch to change the operating levels.

You would have to send a test signal through the dmp at 0dbvu and see what your daw reads. There are too many things that can be out of calibration to tell you exactly what everything should read.
 
My dmp3 has a switch where i can change the signal levels within the software. My options are "consumer" or "-10db".

I feel like im seeing it backwards. I always have more digital signal than analog signal. From what im reading i should be seeing less digital and more analog signal.
 
Erockrazor said:
My dmp3 has a switch where i can change the signal levels within the software. My options are "consumer" or "-10db".

I feel like im seeing it backwards. I always have more digital signal than analog signal. From what im reading i should be seeing less digital and more analog signal.
Click to expand...

do you mean the 2496 has the option to switch in the software? The DMP3 is an analog unit.
The consumer/-10dBV option is for the outputs, not the inputs.

What you're seeing is exactly what should be happening. The inputs on the 2496 are looking for a weaker signal. In other words, it doesn't take much volume to push the meters in your software. But it takes a lot to push the meters on the DMP3. So the signal the DMP3 is putting out is really hot. And if you are putting a really hot signal into something else that is looking for a weak signal, you're going to overdrive it easily.

Unfortunately it doesn't look like you're going to be able to change settings in the software or on the DMP3. You're just going to have to either live with the mismatch of the meters or get a new preamp/soundcard.
 
I am going to ask what is probably an odd and stupid question. My amateur recording rig has no way to accurately measure levels. My mic preamp doesn't have anything but 4 LEDs to show when it's officially clipping and the firebox has no meters, either.

Here is my question:

Is there some kind of meter I can run in my signal to assess and measure signal levels going into my firebox? Maybe something simple and small with a screen and a needle?
 
zenpeace69 said:
Is there some kind of meter I can run in my signal to assess and measure signal levels going into my firebox? Maybe something simple and small with a screen and a needle?
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Doesn't your daw have any meters?

If you are using the meters in the computer:

Sustain-y sounds like distorted guitars should sit around -18 with the palm mutes pumping up around -10.

vocals, sustained notes need to sit around -18, consonants and such can go up to around -10.

Drums, just try to get them to peak below -6, it's OK if it goes higher as long as it doesn't get to zero.

The volume of all instruments vary as you play them. The sound of an acoustic guitar has two parts. The sound of the pick plucking the string and the sustained note. The pick will be much louder on the meters than the sustained note but the sustained note is what we are trying to get at -18dbfs. The fact that the meter jumps every time you pluck a string doesn't matter, that's what the headroom is for. To properly set the level, you need the sustained (average) level to be at line level (0dbVU, -18dbFS). It's as simple as that.
 
Farview said:
Doesn't your daw have any meters?

If you are using the meters in the computer:

Sustain-y sounds like distorted guitars should sit around -18 with the palm mutes pumping up around -10.

vocals, sustained notes need to sit around -18, consonants and such can go up to around -10.

Drums, just try to get them to peak below -6, it's OK if it goes higher as long as it doesn't get to zero.

The volume of all instruments vary as you play them. The sound of an acoustic guitar has two parts. The sound of the pick plucking the string and the sustained note. The pick will be much louder on the meters than the sustained note but the sustained note is what we are trying to get at -18dbfs. The fact that the meter jumps every time you pluck a string doesn't matter, that's what the headroom is for. To properly set the level, you need the sustained (average) level to be at line level (0dbVU, -18dbFS). It's as simple as that.
Click to expand...


I've been using this really simplistic program from Cakewalk called Guitar Tracks. It doesn't have a meter in the program. I have Tracktion on the way, but it isn't here yet. So I can expect it to have a meter I can read the levels on, huh?
 
I just wanted to thank the guy who suggested Reaper. I've been using now for over 2 years and it's the most intuitive multitracking program I've ever encountered. I've saved a ton of time on learning curve and have been able to spend more time on making music.
 
That was the only thread I've ever really read from start to end...3 pages..anyways just wanted to post to revive this 3 year old thread cuz it was so interesting and informative.
 
Actually, this thread was started some 5 years ago. The only reason I bother to bring that up is because there have been some changes in the last 5 years, and if this thread is to be bumped, I guess we should make sure the info is up-to-date.

I just wanted to point out that there was a lot of mention of of a conversion factor of 0dBFS analog converting to something in the -14 to -18dBFS range, depending on the converter. While this is still true, and -18dBFS is still often cited as the ballpark average for conversion, in the past few years more equipment has come on the scene with even lower conversion factors. It's not too hard to find boxes these days that convert 0VU to -20dBFS or even -22 to -24dBFS at times, making the average unity gain digital recording level even lower. -18dBFS is still a good number to quote and to go by when one doesn't want to explain the whole story, and there's still many (perhaps a majority?) of devices that use that conversion rate, it is truly more of an "average" value these days, and one that seems to be dropping by a dB or two every couple of years as new gear keeps coming out and the older gear gets retired.

G.
 
One reason I've always sort of "pretended" I was at -24dBFS ever since 24-bit recording became the norm.
 
Thank you nbtech for bumping this great thread. Until an hour ago (I too read the whole thing), I remained among the misguided "Why wouldn't I want to use all of my computer's range? As long as I don't hit zero, where's the harm, and look at how far I am above my noise floor!" dumbasses. This in spite of an electrical engineering degree and having been messing around with this stuff for, shall we say, a good little while now. Explains many situations where I have felt my vocals sounded "saturated" despite no red lights!!! May also explain why my stuff has often sounded "less than clean and clear" in a way I couldn't quite describe. I can't wait to get to work on my next project!

Thanks again,
J

And thanks to the many folks who contributed to this thread. Seems to me it should be stickied or something with the heading "Required reading for newbs" or some such.
 
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