Digital Recording Levels – a rule of thumb

Okay, I mentioned this as one of my tips in a previous post, but there’s confusion and many heated debates out there about the ideal level to record into your digital audio workstation.

I’m just summing up the information readily available elsewhere (if you are willing to wade through endless online debates and the numerous in-depth articles), for people who just want to know right here and now what the best level is to record into their digital audio systems.

So I’m going to start with just a quick easy rule of thumb for these people, followed with a little bit more detail after that to explain why I’m recommending these numbers.

I apologize for simplifying some of the math – but if you’re really interested there are plenty of texts and in-depth articles available with a bit of searching. I’ve included a few references and links at the end of the article.

The rule of digital thumb

  1. Record at 24-bit rather than 16-bit.
  2. Aim to get your recording levels on a track averaging about -18dBFS. It doesn’t really matter if this average floats down as low as, for example -21dBFS or up to -15dBFS.
  3. Avoid any peaks going higher than -6dBFS.

That’s it. Your mixes will sound fuller, fatter, more dynamic, and punchier than if you follow the “as loud as possible without clipping” rule.

For newbies – dBFS means “deciBels Full Scale”. The maximum digital level is 0dBFS over which you get nasty digital clipping, and levels are stated in how many dB below that maximum level you are.

Average level is very important – people hear volume based on the average level rather than peak. Use a level meter that shows both peak and average/RMS levels. Even better if you can find a meter that uses the K-system scale.

Some common questions:

Q: Why do we avoid going higher than -6dB on peaks? Surely we can go right up to 0dBFS?

Answer 1 – the analogue side.
Part of the problem is getting a clean signal out of your analogue-to-digital converter. Unless you have a very expensive professional audio interface, or you like the sound of the distortion that it makes when you drive it hard, then you’re going to get some non-linearities (ie distortion) happening at higher levels, often relating to power supply limitations and slew rates.

Most interfaces are calibrated to give around -18dBFS/-20dBFS when you send 0VU from a mixing desk to their line-ins. This is the optimum level!
-18dBFS is the standard European (EBU) reference level for 24-bit audio and it’s -20dBFS in the States (SMPTE).

Answer 2 – the digital side.
Inter-sample and conversion errors. If all we were ever doing is mixing levels of digital signals, we would probably be fine most of the time going up close to 0dBFS, as most DAWs can easily and cleanly mix umpteen tracks at 0dBFS.

EXCEPT there are some odd things that happen;

  • Inter-sample errors can create a “phantom” peak that exceeds 0dBFS on analogue playback.
  • When plug-ins are inserted they can potentially cause internal bus overloads. These can build-up some unpleasant artifacts to the audio as you add more plug-ins as your mix progresses. They can also potentially generate internal peaks of up to 6dB – even if you’re CUTTING frequencies with an EQ, for example.
  • Digital level meters on channel strips seldom show the true level – they don’t usually look at every single sample that comes through. It’s possible to have levels up to 3dB higher than are displayed on the meters.

Keeping your individual track levels a bit lower avoids most of these issues. If your track levels are high, inserting trim or gain plug-ins at the start of the plug-in chain can help remove or reduce these problems. Use your ears!

Q: Aren’t we losing some of our dynamic range if we record lower? Aren’t we getting more digital quantization distortion because we’re closer to the noise floor?

Short answer. No.

Really, both of these questions sort of miss the point, as we shouldn’t be boosting our audio up to higher levels and then turning it down again. So there’s nothing to be “lost”.

It’s the equivalent of boosting the gain right up on a mixing desk while having the fader down really low, giving you extra noise and distortion that you didn’t even need. You should leave the fader at it’s reference point and add just enough gain to give you the correct audio level. This is what we’re trying to do when recording our digital audio as well – nicely optimizing our “gain chain”.

The best way to illustrate this is to throw a few numbers up;

Each bit in digital audio equates to approximately 6dB.
So 16-bit audio has a dynamic range of 96dB.
24-bit audio has a range of 144dB.

With me so far? Probably doesn’t mean a lot just yet.

Now, let’s look at the analogue side where it becomes slightly more interesting.

The theoretical maximum signal-to-noise ratio in an analogue system is around 130dB.
Being awesomely observant, you picked up immediately that this is a lot less than 24-bit’s 144dB range!

In fact, the best analogue-to-digital converters you can buy are lucky to even approach 118dB signal-to-noise ratio never mind 144dB.

So – let’s think about this.
If we aim to record at -18dBFS, how many bits does that give us?

24 bits minus 3 (each bit is 6dB remember). That’s 21 bits left.
What’s the dynamic range of 21 bits? 126dB
What’s the dynamic range of your analogue-to-digital converter again? 120dB-ish.
Less than 20 bits.
One bit less than our 21-bit -18dBFS level.

The conclusion is that when recording at -18dBFS you are already recording at least one bit’s worth of the noise floor/quantization error, and if you actually turn your recording levels up towards 0dBFS, all you’re really doing is turning up the noise with your signal.

And most likely getting unnecessary distortion and quantisation artifacts.

Apart from liking the sound of your converter clipping, there’s NO technical or aesthetic advantage to recording any louder than about -18 or -20dBFS. Ta-Da!

Mix Levels

If you’ve been good and recorded all your tracks at the levels I recommended, you probably won’t have any issues at all with mix levels.

The main thing is to make sure your mix bus isn’t clipping when you bounce it down.

Most DAW’s can easily handle the summing of all the levels involved, even if channels are peaking above 0dBFS. In fact even if the master fader is going over 0dBFS, there’s generally not a problem until it reaches the analogue world again, or when the mix is being bounced down.

Most DAWs have headroom in the order of 1500-2500dB “inside the box”. You can usually just pull the master fader down to stop the master bus clipping.

Saying that, it’s still safer if you keep your levels under control.
Like I mentioned before – a key problem is overloads before and between plug-ins. If your channel or master level is running hot and you insert a plug-in, it could be instantly overloading the input of the plug-in depending on whether the plug-in is pre-or-post the fader. So use your ears and make sure you’re not getting distortion or weird things happening on a track when you insert and tweak plug-ins.

Try to use some sort of average/RMS metering, and try to keep your average mix level (ie on your Master fader) between about -12 to -18dBFS, with peaks under -3dBFS.

Mastering will easily take care of the final level tweaks.

To conclude – when recording at 24-bit, there is a much higher possibility of ruining a mix through running levels too high than having your levels too low and noisy.

As Bob Katz says, if your mix isn’t loud enough – just turn the monitor level up!

PS – say “no” to normalizing. That’s almost as bad as recording too loud.

References:
Bob Katz’ web site.
Plus Bob’s excellent book “Mastering Audio – the Art and the Science”.
Paul Frindle et al on GearSlutz.com
A nice paper on inter-sample errors

Download a free SSL inter-sample meter (includes a nice diagram of inter-sample error )

18 thoughts on “Digital Recording Levels – a rule of thumb

  1. Just to clarify for a newbie please. Do I use the gain stages of my preamps to bring down the level to say -20 as seen in the in the DAW meter, or do I use the channel fader in the DAW to bring down the level, i.e having set it to the yellow zone in the preamp. Both will result in that 20 dB of headroom from the digital perspective. Thanks. TJ

  2. Hi there – the channel faders in your DAW usually only control your monitoring level, so the recording level should be set using the preamp gain. Hope that helps!

  3. I think I’m still not quite clear on something: Is this digital clipping actually going on in the hardware converters themselves? If so then yes, I can see why I would definitely need to trim back the preamps. After all, the DAW is ultimately just recording numbers and in the final analysis you could theoretically scale them how you want. i.e. I could for example use a trim or a gain plugin if I wanted to attenuate at the DAW stage. Sorry to be be such a clutz!

  4. Hey there – don’t worry – there’s no stupid questions – it’s not the easiest thing to get your head around.

    The “ideal” is to also avoid any potential clipping or unwanted colouration coming through your converters, but if your recording sounds great but it’s just recorded a bit too loud, then yes, you simply stick a trim/gain plug-in first in each channel strip to bring it back down.

    Note that some engineers love the sound of the “soft clip” in their Apogee converters so will deliberately record loudly through them to get the sound they want. So this is how they should also be trimming it back down in the channel strip to a reasonable working level in the DAW.

  5. Oh and also meant to say – if you turn up your preamp gain too much and then have to subsequently turn it back down in the DAW to get better levels, you’ve just added more noise unnecessarily. ;o)

  6. I always thought that dBFS implicitly means “peak level”, and that the output of a 0 VU preamp with a sine wave was equal to “- 18 dBFS” (peak level). So I record at a peak level between -18 dBFS and – 8 dBFS (depending on the type of sound, almost 18 dBFS for sounds like bass or cello, higher for percussive sounds). On the table “Reference Levels – analog and digital scales”, are the three right columns in dBFS to be read in “peak level” or “RMS Level” (I think “peak” but maybe I am mistaken)? A friend of mine swears it’s RMS level, not peak level. Can you tell me what is right? Thank you in advance for your answer.

    • Hi, actually you’re both right. dBFS is simply a scale; it’s mostly just meter ballistics that determine whether it’s measuring peak or average levels. So the scales in the table work either way.

      In a practical sense, you’re watching both your average level and the absolute peak for setting your optimum recording level – which it looks like you’re doing perfectly.

      Working out what the average level actually is on peak meters is tricky though – it’s nice to have a proper RMS or VU meter as well as the peak meter to help with that. If you have later-version Pro Tools you can try some of the numerous built-in meter options. Other DAWs typically have some metering options as well, or there are plenty of good metering plug-ins available (some even free).

      • Hi ! Thanks for your answer. What do I have to do if I want well calibrate my converter with a sine noise at 1 kHz ? For 0 VU, do I have to read -18 dBFS in the peakmeter in my DAW input ? Or do I have to read – 18 RMS ? The RMS level is 3 dB lower than the peakmeter level. Or maybe it’s (quite) the same ? Thanks again for your help.

        • In theory it shouldn’t make too much difference – VU meters can be calibrated against a sine wave or square wave and I think it does show up as a 3dB difference between the two.
          Unless you’re genuinely worried about sudden huge peaks, it’s probably better to aim for the higher version of the two VU levels.
          I’d just use the peak meter level for setting the signal to -18dBFS.

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