Headroom and dB explained

TheWinterSnow

Den Mørke Natt
Oct 22, 2008
3,087
2
38
34
Sacramento, CA
This is a repost from another thread, but I felt that this topic deserved a thread all its own. This is a post from my audio tech blog corvusaudio.blogspot.com explaining what dB, headroom and proper tracking levels are and why they are important. Figured you guys might like, especially the newbies that have little to no experience with analog gear or proper ITB tracking levels.

Without further ado:

There have been questions floating around the internet asking about tracking levels, what the meter readings (in dB) mean among other things. The internet is a place full of this information but it is very scattered and incomplete so I figured I would set some things in line.

For starters we have to talk about what a decibel is and what they mean in relation to a signal. A decibel (dB) is a unit of comparing a value to another value, in our case, a voltage reference compared to a signal. When we talk about the signal strength in line level audio which is used in professional and consumer based electronics, the input or output signals are not measured in volts, but with a system that allows you to compare the nominal, or average signal amplitude to other systems as well that may not be directly compatible, as such it allows for quick comparison of ratios and signal strength that may be difficult when analyzing voltage or power alone. As the decibel is a value comparison to another value the question becomes, "what is the reference for line level dB"?

The unit is noted as dBu, or "decibel unloaded" and its reference point is the voltage required to dissipate 1mW into a 600 ohm load, or in other terms, about 0.7746v. The equation for calculating decibels is as follows:

dB=20[log(Vo/Vi)]

Where:

Vi = Input or reference votlage
Vo = Output voltage or signal voltage

This means that 0dB is the same as the reference voltage. If the input signal is less than 0.7746V or 0dBu, then the decibel value will be negative and if the input signal is greater than 0.7746v or 0dBu, then the decibel value will be positive. Negative means signal loss, positive means signal gain, zero means no change in signal magnitude. If you have a gain of 2 for example (put 1v in and you get 2v out) you have a gain of roughly +6dB. If you had a gain of 10 as another example, you would have a gain of 20dB.

Most balanced line inputs have a nominal rating of +4dBu which roughly represents 1.228v. Nominal means that is the target signal amplitude that you would be aiming for that would be considered "proper tracking levels". A balanced input means there is a hot, neutral and ground connection, used to reduce noise over the cable run and increase gain. If you connect an unbalanced signal to the input, you will get -6dB of gain reduction as we have an output signal that is half the original signal, the equation 20[log(1/2)]=-6. External noise can also increase by as much as 60dB. The 60dB figure comes from the fact that the common mode rejection, a fancy name for saying how much a noise signal can be attenuated, on average attenuates noise signals up to 60dB and higher with better audio equipment. Without a balanced connection that same noise level is unimpeded.

Now how does this nominal 4dB of the line inputs interface have anything to do with the dB scale in your DAW. Well the dB scale in a daw is a negative range, this is because its 0dB reference is the maximum headroom of the system before clipping. All Analog to Digital Converters, while having a nominal value. Like mentioned before this nominal value is the average signal strength that the manufacture deems you should track your levels at. As a result, all ADCs have much more headroom than that and will typically peak and clip any aditional signal after 24dBu. If you have an ADC that has a maximum input level of 24dBu, a signal peaking at 24dBu will register as peaking 0dBFS in the DAW. In other words the DAW is telling you that the ADC is maxed out and any additional signal will cause is to clip.

"Decibel fullscale", or dBFS is simply the name given to digital equivalents to actual analog signals, and this is the measuring unit in all DAWs. This means that 0dBFS is equal to the maximum headroom level of the audio interface, the line input and in most cases, the line output as well.

Now how does this relate to a line out? Professional line out devices used in audio recording all the way down to small USB interfaces have a range usually in the 12dBu to 30dBu output before clipping. The bigger professional DACs are all 24dB. Lets say that the line output is rated for a maximum output of 24dBu, if a signal that is peaking 0dBFS in the DAW, it will come out as 24dBu. These are extremes though because again the proper tracking levels are listed as the nominal level designated by the manufacture which for compatibility, professional line level is nominal 4dBu as mentioned. This means that if you track your lvels nominal for 4dBu, never raise the faders in the DAW over 0dB, then the line outputs will come out nominal 4dBu as well. The signal will be less than 4dBu if the fader for that track in the DAW was turned down as is typical in the mixing process. There is a lot of headroom left over as a result. If we have ADCs and DACs that have a maximum rating of 24dBu and each track is at most peaking at 4dBu nominal, that means we have 20dB of headroom. The reason this is done is that in Digital to Analog Converters (DACs), the reconstruction circuitry and associated amplifiers become less linear or pick up more Total Harmonic Distotrtion (THD) the closer the signal from the DAW reaches 0dBFS. To keep the signal clean and free of coloration from the ADC and DAC process, the accosiated amplifiers in the converters must have as much headroom as possible..

This is where we get into consoles, their headroom and VU meters. A VU is the same as a dBu in terms of the reference voltage but with an analog console 0VU is the same as 4dBu. In a console you can see up to 24dBu of headroom which leaves you a good 20dB of headroom left over after it receives a 4dBu (or 0VU) signal. Obviously you need this headroom when you sum all the tracks together, you shouldn't be anywhere close to distorting the master bus. If you are working "In the Box" and want to give yourself as much headroom as you would with an analog console, you would take the average amount of headroom you would have left with that analog gear and subtract that from your DAW or, take that 18dB to 20dB out from the 0dBFS. This is why 0VU is commonly referred to as being the same as -18dBFS in the DAW. This value is by no means scientific, as 0dBFS is always the maximum output of the ADC and DAC being used and in less than professional equipment ranges quite dramaticlly. The -18dBFS peak of the tracks in the DAW is usually a safe bet and a good starting point. As always depending on what you are using you will have to adjust your track levels and the input levels on your analog gear in some cases.

The way that the ADC, DAC and console setup in signal amplitude would be as such: if the audio was tracked peaking at 4dBu on a nominal 4dBu ADC, it would register in the DAW peaking at -18dBu. If the DAC has an output of 24dBu that was sent to a track in the console, the console will register a track signal peaking 0VU and when all the tracks are summed together, should not exceed the 24dBu headroom of the master bus. Our tacking levels are happy, the Fullscalse values are low enough to keep THD of the DAC low while the signal strength to the console is adequate. We have a happy system.

There is one unfortunate side to this though, if you are tracking in 16-bit, peaking tracks at -18dBFS is quite low and they will lack dynamic and have an increased noise floor as well as possibly having a somewhat digital sound. Some digital plugins and amp sims may not have enough signal to function properly as well. To remedy this, tracking in 24-bit ensures that you have more than plenty of headroom, resolution and noise. A 16-bit track has only 65,535 different points representing the signal where a 24-bit track has 16,777,215 different points representing the signal. Of course that increase of resolution will not only give you enough headroom to burn, it will be overkill and actually provide you with punchier mixes even if you export to 16-bit as the final format.

If you are using some plugins that still aren't working correctly with the tracks peaking at -18dBFS, then using the preFX trim control in your DAW's channel strip will allow you to adjust as needed. It can also be used to bring down tracks that are too hot and clip or cause malfunction to certain plugins, specifically analog simulated plugins.

Even if you are not going to use analog gear and will stay completely "In the Box", having a good habit of tracking levels is always wise, especially if you use analog simulated plugins which impress that equivalent headroom limit of the real analog gear, basing its headroom in the real world on 0dBFS. If you are using an console simulator, then to get it to color the sound as if you were really pushing that console track to about 0VU, you need to keep the signal in your DAW peaking around -18bBFS and never peak higher than -14dBFS which is the same as 4VU.
 
Short version.
Treat minus 12 as zero.
In PT this means stay in green.
Make sure you have pre fader metering on and adjust output of all plugins as you go so level always stays at -12 through tracks and busses.

I have tried to explain the whole zero=-18 and +4=-14 and have been met with blank stares. Telling them to treat yellow as red gets the job done, even if they can't grasp the numbers :)
 
Thanks for that mickrich, my eyes and brain were starting to flip out with all those words!
 
-18 signal level is quite low for some plugins e.g. ampsims, clippers, compressors. Ampsims don't sound the same at different input levels, and just try to see what gclip shaves off from the tiny peaks...
So I have to boost the level and then pull the fader down to get to -18 on the track. I guess there could be some clipping prefader, or even within the plugin. How can this be avoided?
 
I have tried to explain the whole zero=-18 and +4=-14 and have been met with blank stares. Telling them to treat yellow as red gets the job done, even if they can't grasp the numbers :)

The bigger part of this was more for the curiosity of guys wanting to use outboard gear or use their DAW like an analog console especially if they are using analog plugins which base how they operate on the amount of headroom they have over 0VU subtracted from 0dBFS in the DAW. It therefor becomes important to know how the dB system works.

-18 signal level is quite low for some plugins e.g. ampsims, clippers, compressors. Ampsims don't sound the same at different input levels, and just try to see what gclip shaves off from the tiny peaks...
So I have to boost the level and then pull the fader down to get to -18 on the track. I guess there could be some clipping prefader, or even within the plugin. How can this be avoided?

with digital plugins (plugins not based on real analog gear) this is true that some of them will h ave a signal that is too low. In the end though a digital plugin doesn't change the way it operates in sound coloration and noise but in cases like a compressor or clipper can effect the maximum amount of gain reduction that you can get. Same goes with amp sims, since they are not based on having headroom, the tracking levels that are deemed to be equal to a real guitar and analog amp are subject to the designer's digression. Like mickrich mentioned for the most part, the acceptable digital reference point to 0VU is actaully -12dB. So for amp sims and other non-analog simulating plugins that can be a safe bet.

To answer your question though, if the prefader levels are too low for a plugin or chain, most DAWs have a preFX, preFader trim. It's function was exactly for this purpose. If the input levels going int a plugin are too low, you use to trim to get up to level or bring down to level if you recieve tracks that were tracked too hot. With any decent digital based plugin, you can set the trim to get your track peaking at -12dBFS and you shouldn't have issues with not having enough input. Also with GClip, you use the clip control to set the threshold for clipping, if you have good tracking levels, the clip control is all you will need. As a general rule of thumb you shouldn't ever have to touch the gain control.
 
I totally agree.
I think when "they" were standardising digital levels, they should have put -14 as zero on the meter then everything above that as plus.
That would make better sense since most analogue like being run at +4dBU.
I always get blank looks from my students when I try to explain this but last time I brought my LA-610 pre/comp in and set levels to 0dBU on the meter and it went into PT as -18.
Made it much easier to explain than just telling them that this was the way it is.
Lots of people don't understand that if you hit -4 on your daw meter you are driving your external pres into distortion.
 
Lots of people don't understand that if you hit -4 on your daw meter you are driving your external pres into distortion.

^This.
Even if you're mixing all in the box by recording very hot levels you're really pushing your pre's and converters really hard. Anyone who's done a live gig on a cheaper desk will know that limited bandwidth strained sound you get when the meters start going high into the yellow but before you've technically "clipped"
 
I totally agree.
I think when "they" were standardising digital levels, they should have put -14 as zero on the meter then everything above that as plus.
That would make better sense since most analogue like being run at +4dBU.
I always get blank looks from my students when I try to explain this but last time I brought my LA-610 pre/comp in and set levels to 0dBU on the meter and it went into PT as -18.
Made it much easier to explain than just telling them that this was the way it is.
Lots of people don't understand that if you hit -4 on your daw meter you are driving your external pres into distortion.

Well here is the kicker. I was somewhat wrong but taking a look at some specs of various line inputs on low end high end ADCs, the rating of the interface (-10dB/+4dB) is not the maximum signal level before clipping, no, it is actually the targeted tracking level. ADC and DAC made for consoles and outboard gear, while having the +4dB rating, have a maximum signal of 24dBu. That means in order for the ADC to register a signal of all 1s or 0dBFS, you would need to have a signal peaking at 24dBu. Now looking at a high end ADC's input impedance and the rated output power of an LA-610, its effective maximum output is 10dBu, or giving your ADC 14dB of headroom, which means if you were exactly peaking the LA-610 at its limit, it will register -14dBFS peak, or like you said, about -18dBFS average in the DAW.

This goes right back to what I was saying in the blog post too, if you are tracking and hitting 4dBu or 0VU on the console, if the ADC has the same headroom as the console both being 24dB, then -20dBFS is the same as +4dBu.
 
Ok so I updated and revised the post with more accurate information as well as removing some ambiguous and hard to understand wording. It should be a little big easier to understand.