Calibrating system and monitors

[Sammigz]

Can someone give me kind of a "quick" explanation of this?
Ive been working on calibrating my system correctly, to bob katz -20 ref levels, and think i may be having an odd problem (ill post description of problem if the suggestions here dont work). But tonight i decided to disconnect everything and start from the beginning.

My set-up is very simple. I have a FireStudio Project running to a Presonus monitor station, and the speakers, which are M-Audio BX5a's. Thats it! Im going to make sure all of my settings are correct also, as far as buffer size, sample rate, etc.

A few things to note... I would like to know if i am calibrating the speakers first, or the system first?
A lot of people told me they like their monitor level knob (on back of monitor) all the way up (clockwise). I would prefer to have them at the same SPL level if possible.

Im also still trying to figure out if the '0' or the '10' on my MAIN knob on the FireStudio Project is considered unity. I believe its the '0' but its confusingly worded every where i check. I know this is important.
And is other mastered professionally recorded material always blaringly louder than when I am working in my software? What i mean by this when i am working on a song i wrote in Studio One, then open Spotify and play something professional, it is BLARING. Like blasting! I realize i am recording/mixing etc at lower levels to save the headroom, but is it supposed to be THIS much of a difference.

Thanks in advance!

[WillMorgan]

And is other mastered professionally recorded material always blaringly louder than when I am working in my software?

Mastering sure does loud tracks up but I am guessing that you may not be normalizing your mixes before comparing to others and that the two together explain the difference. If I'm right when you turn your outputs up when monitoring so the peaks are near 0db you'll be a good deal closer.

[JWL]

The main thing is to just decide what you want your reference to be at 0dB on the meters, ie, the max loudness. Then if you are doing Katz's K20 or whatever that will give you your average listening levels relative to that -20dB down point.

I use pink noise and an SPL meter to do this in the room.[/quote]

[Injured Ear]

The easiest way I have calibrated my speakers:

Get a cheap SPL meter. Radio Shack makes a great one. You can find decent software versions for iPhone and Android that are accurate enough to balance your speakers.

1) Turn your speakers off
2) Find a way to look at RMS metering on your DAW
3) Play pink noise out of your DAW so that it is feeding your DA converter at a -20 dBFS RMS level
4) Set your monitor controller to 12 O'Clock. Or 2-3 o'clock. Depends on how much louder you want to turn your speakers when you want to listen to low level material Also, this may need to be louder, depending on how powerful the amps are on your speakers. (see step 5)
5) With your SPL meter at where your head would be during mixing, turn on one speaker and adjust its level until it read 82 dB on the SPL meter. Turn that speaker off. Repeat for other speaker so that they have matching levels. You may need to adjust your monitor controller so that maybe 3 O'Clock is your ref level if you are turning the level on your speaker up all the way and you are not achieving 82 dB.
6) Now you should be able to turn both speakers on and achieve somewhere in the neighborhood of 85 dB SPL with that pink noise you've been playing with this whole time.
7) Mark (or memorize) that level as your "reference level" for mixing
Make awesome records

Mastered tracks will knock your head off when you play them at full level through your speakers set to that reference level. It will come out at least 95 to 100 dB.
That's just the way mastered tracks are. Their RMS levels are -12 to -8 dBFS which is way hotter than you should ever make your raw mixes. Just pull their levels down until they match what you are doing before you do any A/B'ing.

[Jarvis]

Sound on Sound magazine has a very good article on this in the May issue. If you don't have a subscription you can buy that article for 99 cents. It's the best walk through I have seen.

[MoreSpaceEcho]

i think the k system is more complicated than it needs to be.

basically all you want/need is a reference spot on your monitor controller where things sound loud enough to you, and if you're mixing (not mastering) you want that spot to coincide with levels in your DAW that are peaking well shy of 0dbfs.....consider -6 your zero....this way you can mix without worrying about clipping/running out of headroom.

the methods others have posted work great.

also, i dunno about anyone else, but to me, 85db SPL is pretty damn loud. i know that's where our hearing is supposed to be flattest, but if i were a mixer, there's no way i could work all day at that level. i forget what i have my system calibrated to, i just know it's "27" on my monitoring DA, but it's probably closer to 80-82ish. my point is just find a spot that's comfortable for you.

and yes, mastered stuff is gonna be way louder. most of it is probably way louder than it really needs to be, but this is the world we live in.

[tjcasey1]

I dunno - I just keep my monitors at a comfortable level and when I'm mixing I make sure the mix is peaking near -3 to 0 db.

I also listen to other music on my monitors in order to get a feel for common levels. When mastering, I compare my levels to these levels. Your non-mastered mixes will always sound quiet.

I think you may be needlessly agonizing over monitor calibration. I can see needing to calibrate a video monitor - you don't want your video to look darker or brighter than most, and otherwise there's no way to tell if the problem is the monitor or the source material. But for audio, I'd just pay attention to the meters.

And buffer size, sample rate, etc - those are for other problems. They won't affect your levels. You're thinking too much about this.

(Level calibration is extremely important when gain-staging - that's when you have two serial level controls in your setup while recording. You don't want to have the first stage really quiet and the second stage really loud to make up for it (or vice-versa) - that would decrease the overall signal-to-noise ratio of your tracks.)

[Nick Sevilla]

Hi Sammy,

I have always calibrated from the playback machine / DAW out to the speakers.

This is how I do it:

First I TURN OFF THE SPEAKERS. You don't want to go deaf.

I have my Pro Tools HD IO calibrated to -18dBFS inputs. This gives me plenty of 0.7746 Volt sine wave signal from my console (I check it once with my voltmeter), and I input it to my PT inputs, all 16 of them, and set my DAW to "calibrate mode" which allow me to use the meters in PT for seeing what dBFS the incoming signal actually is. I then turn down / up each input level to get -18dBFS inside the DAW.
I end up with -18dBFS = 0 dBu (the tape reference). This means that when my input sees a 0 level (0.7746 Volt) signal, it registers as a -18dBFS in my DAW meters. It gives me 18dB of headroom before clipping, when referencing analog gear.

The Pro Tools HD outputs are calibrated to -18 dBFS as well, to keep it equal, and less complicated. I calibrate them by routing a signal from the PTs signal generator to each individual output. The signal I use is a sine wave, set to -18dBFS, and I double check it with my voltmeter to make sure it matches the outside world voltage of 0.09751 V.

Then, to make sure that if I output 0 (Full Scale, or maximum in a DAW) I output from the signal generator the maximum, which is 0dBFS, and double check the voltage, which need to be 0.7746 V. Once this is done on the first channel, I do the rest, back at -18dBFS.

Now, once this DAW is calibrated both in AND out, I can calibrate my console to it.

I just output the -18dBFS signal from all 16 channels to each channel 1-16 on my console, and calibrate each console input to 0 dBu. Because the console still has +18dBu above their zero mark for headroom, this matches my headroom inside the DAW. Now when I clip in the DAW, it also clips in the console.

When calibrating my console Line inputs, I ALWAYS set the faders to Unity (0), and use the input trim to get to my zero level. This means the fader amplifier is neither adding nor subtracting level from the input to the output (mix).
Likewise, I set my MAIN faders to Unity (0).

Now, I can do my monitor speakers. I use 2 reference levels for mixing:

72dBu and 85dBu. Cause I like both. The lower one lets me work a long time, the other is the "standard" where most ears have the most neutral response curve to sound.

I use a sound pressure level meter (SPL meter). I have a radiocrap one, it works great.

I place it on a mic stand, and place it exactly where my head is during mixing, same height, and pointing straight ahead (it is an OMNI mic).

Then, I first make SURE my monitor knob (control room level) to the speakers is all the way OFF before turning on the speakers.

I turn on the speakers, and slowly turn up the control room level knob until I get 72 dBu, and make a mark on some white tape that I have set around the knob base, to mark 72dBu. Then I go up until I get to 85dBu. I do this for my main speakers, and then turn them off, and turn on the smaller nearfields. These have a nice level trim knob on the front. I set them until I get 72dBu, and then turn up the control room knob to 85dBu, and check the meter to see if the nearfields are giving me the same level. If they are not, then I have to move them forwards /backwards until I can match both levels with both sets of speakers.

Once this is done, I get a beer and go home, because my ears are pretty much done.

I do this every time I am going to mix another big project. Most of the time, things are out of alignment only a tiny bit.