# Help Me Understand Monitor Level Calibration



## Gerhard Westphalen (Dec 19, 2015)

As I work on getting my new template set up I'd like to have my monitor levels properly set up. It's my understanding that pink noise at -20dBFS should be 85dB on each channel. What I don't understand is, shouldn't 85dB be the top level? Shouldn't that be 0dB in Cubase? If I calibrate it this way wouldn't everything have to be a lot quieter and not be anywhere near 0dB? How do you know the maximum level that things should be at if you have the 20dB of headroom? I've always used 0dB as my maximum and the clip indicator to know if I'm going too loud. 

If I have my system set up this way and I want to play something on the internet like Youtube where the volume defaults to being all the way up, wouldn't it be really loud since -20dBFS is already 85dB? Would I essentially need 1 level for working in Cubase and 1 level for everything else on my computer?


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## Gerhard Westphalen (Dec 20, 2015)

Thanks for the links. I had seen them but hadn't read the SOS article all the way through. It says this close to the bottom:
"However, I must issue one very important warning at this point. When using commercial CDs as reference material, or when mastering your own material within the peak‑normalisation paradigm, the normal working headroom margin is absent because it has been deliberately stripped off. The signal will be banging against the digital end‑stops and the average level will therefore be some 12 to 15 dB higher than expected in a system aligned for a digital operating level of ‑20dBFS. It's therefore essential that you turn down the monitor level in advance, or attenuate the source material within the DAW to make it compliant with your standard operating level."

So people with calibrated systems are constantly having to adjust the monitor levels (can anyone here attest to that?)? Seems like you could easily blow your monitors if you forget to turn it down. 

I still don't quite understand using the headroom. So there's not really a fixed point of the maximum level you should reach? 

Most plugins like limiters default at working around 0dB so does that mean that everything would have to be brought down to -20? I've seen videos like the drum programming one from Christian Henson where he's working at around 0 and even clipping some of the busses. If his room was calibrated that would be really loud, would it not? 

When working on a calibrated system would you have to normalize everything being put on Soundcloud etc.? If you were using a limiter at -20 you'd have to bring everything up by 20dB in the normalization? 

I don't even know if I can get my system up to this level. I recently spoke to support from Dynaudio and they said that if my interface is +4 the monitors should be set to the -10 level. With these levels, the output on my Fireface at 0 is only getting about 60dBC with the pink noise. Playing music at the normal 0dB ceiling levels is already more than the monitors can handle with these level settings so if I were to turn them up to reach the proper level with the pink noise they could easy get overloaded.


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## d.healey (Dec 20, 2015)

I use the K-system (http://www.digido.com/how-to-make-better-recordings-part-2.html), I only use my monitors for music production so I have no need to adjust them often. I tend to use -14dB as my "0" point, this leaves plenty of headroom for mastering.


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## Gerhard Westphalen (Dec 20, 2015)

d.healey said:


> I use the K-system (http://www.digido.com/how-to-make-better-recordings-part-2.html), I only use my monitors for music production so I have no need to adjust them often. I tend to use -14dB as my "0" point, this leaves plenty of headroom for mastering.



Thanks for the link. I'll give it a read later today. I'd like to get as close to an industry standard as possible so that I can get used to it. 

One solution that I've thought of for dealing with the sound of everything else is to have the busses used by Windows on my RME turned down and then have Cubase use separate busses for its outputs which I leave at 0. This is a rather complicated system which relies on the capabilities of the TotalMix so I'd still like to know how other people who might not have this option deal with this. It also doesn't deal with plugins and instruments defaulting to levels at 0.


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## mmendez (Dec 22, 2015)

Gerhard Westphalen said:


> It's my understanding that pink noise at -20dBFS should be 85dB



That's not true for a home studio, though. That's the recommended level for movie mixing in a movie mixing room, which is way bigger than your home studio. If you set your near-field monitors to 85dB you might damage your hearing.

There are two topics here:

1) The K-system is an awesome concept that will allow you to end up with mixes that will preserve the dynamic range of your piece. That's a good thing because you don't to smash and over compress your music.

2) How loud should you monitor is a personal matter. It has to be loud enough that you can hear the soft and intimate parts in detail, but not so loud that your ears will get tired too quickly. What I've done is listen to my favourite scores and find the sweet spot, volume wise, that allows me to listen to every detail in the piece. A very straightforward way to see where you are is to listen to "In the hall of the mountain king". Can you hear everything in the beginning of the piece? Is the end loud but not deafening? If you answered yes to both then you are on your way. 

The perception plug-in (http://meterplugs.com/perception) can be a very valuable tool to compare your mixes to other people's.

Also remember to take breaks after mixing sessions or you will end fixing things that are not there. It's also good to listen to your mixes in different settings like cheap speakers, ipods, cars and see how they translate.

About the system sounds; yes you want to send them somewhere else, possibly the computer's onboard soundcard so they don't interfere with your DAW levels.

My two cents,

Miguel


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## Gerhard Westphalen (Dec 22, 2015)

This article helped a lot:
http://www.soundonsound.com/sos/sep13/articles/level-headed.htm

What's still unclear to me is the level which things should be at. Should I be trying to keep things up to -20 and only rarely going above that? Say I had a short cue with little dynamic range. Or should it be a higher level that I'm trying to hit like -12? Or does it simply not matter since it'll pushed up to 0 when mastered?


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## Gerhard Westphalen (Dec 22, 2015)

mmendez said:


> That's not true for a home studio, though. That's the recommended level for movie mixing in a movie mixing room, which is way bigger than your home studio. If you set your near-field monitors to 85dB you might damage your hearing.
> 
> There are two topics here:
> 
> ...



I don't understand your comment about damaging hearing. 85dB is 85dB regardless of if it's in a large room or in a small room.


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## synthetic (Dec 22, 2015)

Here’s how I calibrated:

1. Put in earplugs. Turn the control room monitor knob all the way down.

2. Get a sound pressure level meter. Set it to 90dB peak, C-weighting, slow. I used the https://itunes.apple.com/us/app/audiotools/id325307477?mt=8 (Audio Tools app from Studio Six Digital) but I also have the old Radio Shack meter that this is based on.

3. Create a *mono* audio channel in Pro Tools/Cubase/whatever. Pan it to the front left speaker.

4. Insert a test signal generator plug-in. Start generating pink noise.

5. Set the gain of that noise generator so that the *average (not peak) *front left output level shows an output level of -23dB. (Not -20dB, see below.)

6. Turn up the control room knob. Set it to a marked reference point. I have the “85dB” point marked on my output level knob.

7. Look at your SPL meter. Turn the amplifier gain up or down until you see 85dB SPL.

8. Repeat this for the center and right channels.

9. Repeat for the surround channels, but these should each be set to read 82dB.

http://www.meyersound.com/pdf/cinema_technical_papers/cinema_calibration_tech_report.pdf (This article) explains why the meter should show -23dB instead of -20dB (crest factor). They also compare pink noise sources but I couldn’t find their magic WAV file.

Another https://www.smpte.org/sites/default/files/2013-03-12-Standards-Cinema_Audio-Vessa-v2.pdf (calibration article) that I found.


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## gsilbers (Dec 22, 2015)

Gerhard Westphalen said:


> I don't understand your comment about damaging hearing. 85dB is 85dB regardless of if it's in a large room or in a small room.



if you are in a theatre 85db sounds ok since its a big room. that same level might be too loud in your small personal room. so a mix will sound too loud in your room and you will lower the volume on your DAW and when presented on a big room it will sound quiet.


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## d.healey (Dec 22, 2015)

gsilbers said:


> if you are in a theatre 85db sounds ok since its a big room. that same level might be too loud in your small personal room. so a mix will sound too loud in your room and you will lower the volume on your DAW and when presented on a big room it will sound quiet.


85SPL in a small room and 85SPL in a big room aren't the same though because the SPL meter will be a different distance from the source.


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## Gerhard Westphalen (Dec 22, 2015)

synthetic said:


> Here’s how I calibrated:
> 
> 1. Put in earplugs. Turn the control room monitor knob all the way down.
> 
> ...



The calibration process is all straightforward. What I don't understand clearly is the use of the 20dB of headroom.


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## Gerhard Westphalen (Dec 22, 2015)

d.healey said:


> 85SPL in a small room and 85SPL in a big room aren't the same though because the SPL meter will be a different distance from the source.



But the monitors are at different levels so if the amount on an SPL meter reads say 85dB at the listening position in both rooms then the level is the same. Is this not correct?


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## gsilbers (Dec 22, 2015)

Gerhard Westphalen said:


> This article helped a lot:
> http://www.soundonsound.com/sos/sep13/articles/level-headed.htm
> 
> What's still unclear to me is the level which things should be at. Should I be trying to keep things up to -20 and only rarely going above that? Say I had a short cue with little dynamic range. Or should it be a higher level that I'm trying to hit like -12? Or does it simply not matter since it'll pushed up to 0 when mastered?



-20 is just a negative number. since you are into learning this stuff I would suggest the lingo as well. -20dbfs (db below full scale) full scale is 0 db in your daw.

as to what levels to hit, it depends on what you are doing. maybe use waves rms dorrough meter.
or the competition plugs. use the k system and try to hit k-12 or k-14.
also use sample magic a/b to compare.


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## Gerhard Westphalen (Dec 22, 2015)

synthetic said:


> http://www.meyersound.com/pdf/cinema_technical_papers/cinema_calibration_tech_report.pdf (This article) explains why the meter should show -23dB instead of -20dB (crest factor). They also compare pink noise sources but I couldn’t find their magic WAV file.



Coincidentally I came across that paper yesterday while researching something completely different (active acoustic systems)


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## d.healey (Dec 22, 2015)

Gerhard Westphalen said:


> But the monitors are at different levels so if the amount on an SPL meter reads say 85dB at the listening position in both rooms then the level is the same. Is this not correct?


Oh yea, doh! You're right, that's makes sense, I was forgetting that you'd have to bump up the monitor level, so if you're comfortable with 85SPL in your studio then that should be fine for you


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## gsilbers (Dec 22, 2015)

Gerhard Westphalen said:


> But the monitors are at different levels so if the amount on an SPL meter reads say 85dB at the listening position in both rooms then the level is the same. Is this not correct?





Gerhard Westphalen said:


> But the monitors are at different levels so if the amount on an SPL meter reads say 85dB at the listening position in both rooms then the level is the same. Is this not correct?



its about how you perceive the 85db. for my room its too loud so I will tend to lower the fader on say, a drum part because it sounds loud and tiring.
you can stil have it at 78db and provide a good mix that translate on the re-recording stage.


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## Gerhard Westphalen (Dec 22, 2015)

I think my SPL meter is completely out-of-whack. Having my BM5mk3 on the higher setting (+4), my TC level pilot all the way up, my Fireface output at +4 with the faders in Totalmix at 0dBFS, I was just barely hitting 65dBC and it was pretty darn loud. This was both with the dBA/C and fast/slow settings.

I downloaded an SPL app and it was about 12dB higher than my SPL meter shows. With it I can easily his 85dBC with the monitors on the 0 settings and the level pilot not all the way up. I am using it on an old iPod touch so the mic level could be different from what the app is calibrated to for newer Apple devices. I'll test it on my brother's iPhone later today but I'm fairly certain that it too will show that my SPL meter is completely off.

Edit: Both the iPod touch and iPhone showed the same levels for 2 different apps so I'm going to assume that these levels are correct. This means that my SPL meter is around 12dBC lower. The calibration screw on it isn't working so I guess it's pretty much useless.


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## Gerhard Westphalen (Dec 22, 2015)

I believe I've calibrated it all correctly. The 85dB level is a bit too loud for me so I'll work a bit lower. The RME conveniently has the digital level so I can easily switch and compare the levels precisely. The 85dB calibration ends up at the -4 fader value in Totalmix and I think I'll work at the -7 value.


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## synthetic (Dec 22, 2015)

Gerhard Westphalen said:


> The calibration process is all straightforward. What I don't understand clearly is the use of the 20dB of headroom.



Just a cinema standard. But standards are cool! It should theoretically sound pretty close in any cinema with the X-curve and levels and everything. Of course most theaters don't play this loud. 

Trivia: When they mixed "The Fugitive" at WB, they were apparently going in order from the beginning of the movie. They mix the bus crash, and it's a big loud action moment. Then they put on reel 3 and see the train hit the bus and think, crap we gotta remix the bus crash so we can top it with the train crash. Instead, they just recalibrated the recorders (mixing to mag film back then.) They figured out that the analog tape could handle more headroom than they thought. So maybe the 20dB headroom is leaving themselves room for emergencies like that. Or for Michael Bay movies.


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## wst3 (Dec 22, 2015)

The problem is, it isn't simple!

-20 is NOT just a negative number, well, I guess if you write it that way it is<G>!

dB, by itself, is meaningless! The decibel is a power ratio, always and only! It has been co-opted to represent other things, among them levels, and since human hearing behaves in a logarithmic fashion it works nicely.

If you want to calibrate your system (you don't just calibrate the monitors, kinda pointless) you need to understand the different measurements systems, and at the difference between headroom in an analog system, headroom in a digital system, crest factor, and dynamic range.

That's way too much to try to cover in a post, but I'll try to get you started:

There are two standards for describing RMS signal levels in the analog domain:
dBV uses 1Vrms as the reference. For the most part equipment that uses this standard provides single-ended interfaces, and the nominal (0VU) level is -10 dBV.
dBu uses 0.7746Vrms as the reference (for historical reasons - not hysterical). For the most part equipment that uses this standard provides balanced interfaces, and the nominal level is +4 dBu in the recording space, and +8 dBu in the broadcast world. (you gotta love standards!)

For sound pressure level (SPL) we use dB-SPL, which uses for its reference the threshold of human hearing, 20 uPA (micro Pascals, a measure of pressure). This is also an RMS measurement.

Thus far all of our measurements are in the analog domain, and they are all average - or more properly RMS (the square root of the sum of the means squared). We use RMS instead of average because, at least for voltages and pressures, the value can be positive or negative. If we simply summed the values over a period of time the result could be zero, which would not make sense. So we square the measurements to remove the effect of the negative values without removing their contribution, then we take the square root to get back to reality.)

Note that the amount of time you spend making the measurement is not part of the definition. It IS part of the definition when we start to talk about VU meters or PPM meters or whatever meters. The RMS value is always changing, so we do need some rules about measurement periods, but that has no real impact on system calibration, where we use 'well behaved' test signals, e.g. sine waves or filtered noise.

OK, so this is our first major stumbling block - for reasons that escape reason measurements in the digital domain are made for a single sample. OK, I do know why, but that doesn't make it any less annoying! So the first difference between analog and digital that we need to grasp is that digital measurements are, by their nature, peak measurements. And just as (by definition) there is no such thing as peak or peak-to-peak dBu, there is no such thing as an average dBFS.

Do not feel bad if your head is swimming. I've had to work this out for some really talented EEs, it isn't taught in school, and it isn't immediately intuitive. To be perfectly honest, I didn't appreciate the finer points until I had to design a analog to digital converter, and it didn't work well.

Back to dBFS - this is a way of representing how many bits we are using. 0 dBFS means we used all the bits, there is no more room for apple pie! So there can never be a positive dBFS measurement. You can't measure what you can't, by definition, measure. (OK, if you are an uber geek there are ways to predict inter-sample 'overs', but that is WAY beyond the scope of a post!)

And that's the other thing we all know and love about digital - we get to define how much headroom we have. If we set our nominal level to -20 dBFS then we 20 dB of headroom. If we set our nominal level to -12 dBFS then we have only 12 dB of headroom. Which might be enough, if we did not have to account for tansients, peaks, and crest factor. And lions and tigers and bears... oh MY!

There is no standard! The AES, among others, are working on it, but it is complex. In fairness, even something you'd think should be simple, such as which pin is hot, or where do you connect the shield, takes time, because there are competing (and viable) solutions, in addition to the solutions that simply won't work<G>!

Let me repeat... there is NO STANDARD NOMINAL VALUE for digital systems.

So pick one. Given the audio quality of modern converters it is perfect fine to 'thow away' a few bits at the top of the scale. So start with -20 dBFS, or maybe -18 dBFS and see what happens. In real life I do not think you will hear a difference.

Now you need to equate that to something in the analog realm. I think it is smart to match it to 0VU at the output of your converter, but for some folks that is an unnecessary step. If you go pretty much straight to your amplifier or monitors you can skip it. Which leaves sound pressure level (and if I didn't mention it before, that is pressure, not intensity!) In all but the worst rooms you should probably be in the area of 80 dB-SPL to 85 dB-SPL for your nominal level. If you have 20 dB of headroom in your digital system (and your analog system can handle that) then your peaks will hit 100 to 105 dB-SPL, which is stupid loud, but it if happens for only a fraction of a fraction of a second you'll never notice, nor will you damage your hearing or your equipment.

The actual steps:
1) turn your amplifier or monitor level controls all the way down, set your channel level and output level controls in your DAW to unity.
2) find a collection of wave files of sine waves, you'll want at least three frequencies (400 Hz, 1000 Hz, and 5kHz), and you'll want at least three levels for each, and you'll want to find files that tell you the peak dBFS value, or you'll need to use an audio editor to figure it out. You want 0 dBFS, -20 dBFS, and maybe -40 dBFS. -10 dBFS might be handy too.
3) find a collection of pink noise wave files - you'll want wide band and octave band centered on at least those three frequencies, and again 3 or 4 levels.
4) play back the -20 dBFS 1000Hz sine wave.
5) turn up your first monitor until you read 82 dB-SPL on your SPL meter (flat weighting, slow response)
6) turn up the second monitor until you read 85 dB-SPL.
-NOTE- we are NOT going to use this as a reference, but it's a good starting point because it will tell you how well your system behaves under heavy load.
7) turn the output of your DAW back down and load up a favorite song. SLOWLY turn it up until it becomes stupid loud (and yes, that's a technical term). If everything is working properly, and your test song has any dynamic range at all your output fader ought to be in the vicinity of -20. Let's assume things are working<G>!
8) turn the amplifier or monitors down again, reset the channel and output faders to unity, and load up a 1000 Hz sine wave at 0 dBFS
9) turn the amplifier back up again, but really slowly this time, cause it's gonna get loud! (also make sure you aren't tripping the peak indicators on your channel or output strips in your DAW. If you are reduce the channel level until you don't see red flashing lights!)
10) same drill as before - turn up one channel till you read 82 dB-SPL, then turn the other one up till you read 85 dB-SPL. 
11) and again substitute a known music track and see what you think about the level in the room. If you are happy with it then you are done for now. If you aren't you will need to repeat steps 9 and 10 with different test signals, and different levels.

My guess is that most systems with a reputable DAW and reputable monitors will line up pretty quickly. And please note, this is NOT the K system or any other system for that matter, is is basic level alignment, nothing more, although it does try to account for the differences between analog and digital measurements.

It may seem like I took the long way around to describe such a simple procedure, but I think it helps to be at least familiar with the terminology if you are trying to calibrate a system.

And that's all I can type!


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## pkm (Dec 22, 2015)

The good thing about standards is that you have so many to choose from!


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## Gerhard Westphalen (Dec 22, 2015)

Thanks for the explanation wst3!


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