equalizing volume, levels?

[canadave]

I'm not a recording engineer, but I do volunteer at my community radio station, so I'm stuck trying to figure this question out

We use Adobe Audition to pre-record things. I want to get it set up so that people are recording as loud as possible with as little noise/distortion as possible.

Now there are two things that influence the level I'm recording at in Audition. There's the input slider on the radio board, which controls the input level of the mics. And there's the recording level I set on the PC's internal sound card mixer. If I adjust either of those two things, the level being recorded changes in Audition.

Beyond that, we also have analog VU meters on our ancient radio board that we use. It does not jive with the recording levels in Audition. In other words, when someone's recording and the VU meter needle shows something like -10db, Audition's meters show something different.

So...I'm jsut trying to understand the proper way to set up recordings, levels-wise. Which of the two input adjustments (board input slider or PC recording level) should I be tweaking? What level should I be recording at? Is there a way to get the levels in Audition to match the levels on the VU meter? And if so, should I be doing that?

Thanks for bearing with what I imagine are highly naive questions

 

[PirateJohnny]

First you need a 1 kilohertz tone generator. That is the standard and makes things easy. If you have one of those Shure four input mixers that are everywhere it has one built-in. This isn't a necessity but it makes things much easier.

Start with the first meter in the chain. I set it at +1Db. Set the next meter at 0Db (100%) and so-on down the line. Setting the first meter at +1 Db gives you a little "head room" before you get distortion farther along in the system.

Grab a CD and play it on the computer in a player that has level indicators. Then compare your recordings to those of the CD. That will give you an idea if your Audition recordings are at the proper level.

There is something important called the Signal to Noise ratio. You want that as high as possible. This method should get you there.

I don't have access to Audition but I would guess it has editing "plug-ins", like other audio software. With those plug-ins you can get rid of background noise and boost the volume of a recording that was recorded too low. In the beginning use the plug-ins sparingly. Things can get crazy fast.

 

[frnkp2000]

Been using an analog board with Adobe for years...here's what seems to work for me:


Run a 1KHz tone on your board until it sits just beyond 100%/0vu---right about +1 where your loudest sounds should be peaking anyway. Run Adobe in record mode and adjust your sound card until you hit -3vu on the digital meter.

Your mileage may vary, but this seems to work for what I have: A 1982 BE Stereo board running Adobe Audition with an old MPC Pentium 4 computer...Seems to work! I don't get any pops or clipping.

If you don't have a source for a tone, have Adobe generate one and record it onto a CD or an analog tape!

If you do all this and it sounds right on test recordings--just set it and forget it! Remember, the Adobe Audition "meters" are registering peaks as well as levels...they're never accurate. I check my calibration every couple of weeks.

PirateJohnny beat me by a few seconds, and I seemed to create an echo in here!

 

[canadave]

Thanks to both of you I'm afraid I'm still not entirely sure, though, of what I should be doing.

I can get my hands on a 1 KHz test tone. But then what?

The signal path goes like this:

Person talking into microphone -> input faders on radio board control input level on mic channel -> recording level meters on PC sound card software mixer -> recording level meters on Windows XP generic sound mixer too, maybe? not sure? -> level meters in Adobe Audition when recording -> VU meter on radio board that displays output from the PC (I think...or does the VU meter indicate the recording level at the second step of the chain instead, pre-PC?)

Anyway, with a 1 KHz test tone...what do I do there? Do I put it on a cassette tape and hold it up to the microphone? (just kidding...I'm clueless, but not THAT clueless...!)

 

[Goat Rodeo Cowboy]

You have been given some good advice here, but it assumes you have a source for tone, etc.

Since Audition will generate tone, create a tone file and play it back into the microphone from an mp3 player that has a speaker, or some other ad-hock lash-up.

Now here is where that system can go wrong, and may be a problem you have run into already in your own experimenting.

What if the line output from your mixer when the (V.U.) meter is reading zero, plus 1 or what ever you choose, that could overload the input of your sound car that expects a lower incoming signal.

At this point you are left to either pretend that -10 or some other value is 0-level for your purposes.

You have my permission to do some experimenting which may violate every text book or magazine article and web site post you have ever read. What you are doing does not require the kind of precision of levels that you might want if feeding into a broadcast chain feeding a transmitter.

Don't set anything in your chain at extreme levels. Not too hot, not too soft, follow Goldilocks and set 'em just right.

make a recording. Leave a quiet place in the recording and at the end of the recording so you can see what your noise level is. You would love to have 50db below the signal when someone is talking into the mic. Give yourself a gold medal if you EVER get that level of separation between noise and desired signal.

Do some recording where in addition to leaving silence (quit talking to the mic! It will forgive you.) Then after a few seconds, turn the gain for the mic channel all the way down. Here we are going to see how much noise is room noise, and how much noise is just electrical system noise when the mic channel is slammed shut on room noise.

In broadcasting (on air chain... not work in the production studio) you want maximum levels. Station policy will tell you what works with the chain between you and the transmitter, but broadcast people always think in terms of hitting zero level with regularity.

In studio work I read all kinds of targets that people have, but when a normal speaker is talking to the microphone you probably want peaks (as you see them graphically on the screen) to reach maybe -6. A lot of people like to work closer to -9.

If only ONE user tends to use this system, you really don't care what all these meters are reading when you are doing recording. You just want a predictable signal to end up in your file with minimum noise.

For my recording I use no other meter than the horizontal peak meter running across the bottom of the screen in Audition. I double check that by looking at the wave form peaks on screen.

If you record a lot of different people and/or instruments, all outputing at their own favorite level, then the use of V.U. meters on the mixer and other points takes on some importance.

 

[PirateJohnny]

Bring the tone in on a line level input on the board -> input faders on radio board control input level on line level channel ->

Set board to +1

-> recording level meters on PC sound card software mixer ->

Set to 0

-> recording level meters on Windows XP generic sound mixer too, maybe? not sure? ->

Set to 0

-> level meters in Adobe Audition when recording ->

Set to 0

-> VU meter on radio board that displays output from the PC (I think...or does the VU meter indicate the recording level at the second step of the chain instead, pre-PC?)

Yes, it displays the PC output if this is the fader you use to playback anything on the computer. Only fade this up after you have recorded something - not while you are recording. Set the fader at the hatch-mark that indicates the "standard nominal setting" and adjust the computer output so the meter on the board reads 0.

 

[Emmett1057]

Alright, lets try to straighten this out. First of all, your Audition meters are an entirely different measurement than the meters on your board. Audition measures PPM at digital full-scale. This is NOT the same thing as an analog RMS meter. Not at all. Before you can do anything else, you need to define your standard. That is, a tone that reads 0dBVU on your board will read what level dBFS in Audition? It's common for this to be as loud as -9 and as low as -22. Pick a number and always stick to it. My home studio reference level is -9, my work studio is -12.

Next, you need to match voltage. This is especially tricky if you're combining pro and consumer gear. The consumer gear voltage spec is -10dBV. The pro spec is +4 dBu. Both are the actual voltage output of 0dB. If you're using a pro console, the output is likely +4dBu. If you're using a stock soundcard, the in is likely set for -10dBV. You need either a Matchbox or a pro level interface/soundcard. Once you have that, run your 1kHz sine wave through your board, set to 0dB o. The board meter. Adjust your input of your interface or soundcard to the reference you've decided on. For instance, if you decided on -12, your Audition meter should display -12 while the board displays 0.

That's all there is to it.

Emmett

 

[PirateJohnny]

As I said, i don't have access to Audition. So I'm not sure how the input and output meters are labeled. I'm familiar with the numbers described by Emmett. I work in TV and we use -24 to be the 0 "zero" I'm referring to. At home I use Audacity and the meter goes from green to yellow to red with no numbers. On Audacity I use the first yellow above green as 0 (zero).

The important thing is to adjust the meters in the order they are in the sequence.

 

[chriscollins]

Look, let's do this with few resources...

With the board first in line, run content or talk with it peaking at about 0...

Adjust recording input slider on computer until that hits about -3 to -3.5 (a little less).

Now you have about 3.0-3.5 dbu headroom. Always keep analog below 0 and you have no chance of clipping.

Very little range sacrificed for what you are doing. You need the headroom in digital... The analog can oversaturate a bit, but you never want to hit 0 dbu on the digital side.

 

[frnkp2000]

Look, let's do this with few resources...

With the board first in line, run content or talk with it peaking at about 0...

Adjust recording input slider on computer until that hits about -3 to -3.5 (a little less).

Now you have about 3.0-3.5 dbu headroom. Always keep analog below 0 and you have no chance of clipping.

Very little range sacrificed for what you are doing. You need the headroom in digital... The analog can oversaturate a bit, but you never want to hit 0 dbu on the digital side.

That's exactly what works for me!