# Digital recording - are you recording too hot?



## Scott Cairns (Aug 1, 2008)

Ok, Ill admit it, I didnt know until recently that Ive been recording too hot when going to digital all these years..

For me some of it is a hangover from analog; overcoming the noise floor to get some nice distortion as you pass 0dbvu saturating the tape.

There's also the hangover from 16bit recording where you had to record hot enough too overcome the noise floor and have enough definition in bits.

But now we're 24 bit, and I realise that Ive been recording way too loud - even if the meters arent peaking.

The ideal recording level seems to be somewhere around -18dbfs - which is what most converters are calibrated to. (the documentation should tell you) Even lower is good; -22 to -24dbfs still gives you a noise floor roughly 100db BELOW that. AND it gives you enough headroom for;

a) - transients
b) - digital processing; eq, compression etc

Here's an interesting post from Hugh Robjohns, the technical editor of Sound on Sound Magazine (sorry for the crosspost to another forum, but the info is good); 

Digital Recording Levels

This quote is particularly relevant;

_In contrast, when a digital system runs out of headroom, the distortion products result in aliasing, which folds the high frequency harmonic components back into lower parts of the spectrum. The distortion becomes anharmonic where the artefacts are mathematically related to the sample rate. This is completely unnatural and so sounds most unpleasant._

Forgive if this is all bleedingly obvious to most of you, but I admit - by habit all these years, Ive been recording as loud as possible into my DAW - and its not only uneccassary, it can actually _harm_ the quality of your recording.

So, as Im an old analog guy from way back, I expect to see a meter nudging into the red, for me it looks like a nice healthy signal. It just doesnt seem right to watch the meters barely climb halfway up the scale and not at least turn yellow. :D 

So here's what I did in Cubase, my meters are set to start into the yellow at around -18dbfs and into the red at -10dbfs;
http://www.sca-soundstudios.com/images/Cubase_meters_01.jpg (http://www.sca-soundstudios.com/images/ ... ers_01.jpg)

Not only does this make an old analog guy like me feel more comfortable, its also a good visual warning that Im sucking up my remaining headroom when those meters start showing red; http://www.sca-soundstudios.com/images/Cubase_meters_02.jpg (http://www.sca-soundstudios.com/images/ ... ers_02.jpg) 

Lastly, I have enough headroom for the summing of the tracks and for any FX I want to add in.

The very last thing I do, is to master the music in a seperate process, up to the level expected on CD these days.


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## Alex W (Aug 1, 2008)

Wow, really??

I've always thought that with digital, as long as you don't clip, you're gold.



Edit:

Just read the forum, nice info in there, thanks Scott!

Luckily I've always naturally left a lot of headroom as a habit anyway. And I'll continue to do so.


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## bryla (Aug 1, 2008)

Guidline could be recording with peaks hitting -11 - -10dBFS

Also keeping all peaks between plug-ins, channels and busses at a maximum of -6 - -3dBFS


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## Scott Cairns (Aug 1, 2008)

Hey Alex, I always thought that a good strong signal (without clipping) would be fine too.

I havent had a chance to try it yet, but some guys reckon they've recorded an acoustic as loud as possible without clipping, and then recorded it at around -20dbfs - the claim is that the acoustic sounds thin in the loud recording and is full of body in the lower take.

Certainly leaving headroom for transients, summing and fx processing makes sense, Im still not sold on the audio fidelity being markedly better cause its recorded quietly - *yet*


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## synergy543 (Aug 1, 2008)

You are unlikely to hear the difference in audio fidelity on a single channel. However, if your tracks are slammed to the max, and you sum several tracks, then you risk getting unwanted distortion from the summed channels.

So why live on the edge? You don't drive fast, go without 8 hours of sleep, or use a computer without a redundant RAID backup system, so why risk recording hot tracks? 

Honestly though, your sample recordings are already tamed by your friendly sample developer so recording samples and live instruments (with dynamics...whoah) are two completely different beasts. Samplists have it so easy! If you watch your levels during each link in the chain, you should be fairly safe.

Hugh is suggesting "nominal levels" of -18dB. This has pretty much always been the case (PCM1630s used for mastering had nominal marks at -14dBFS I think) but peaks of course could go much higher. If you're interested in this stuff, you should read http://www.digido.com/bob-katz/level-practices-part-2-includes-the-k-system.html (Bob Katz articles on metering and the K-system). The K-system is probably more important for composers on this forum that want to still work on music in their old age and PSP has a nice K-system with their http://www.pspaudioware.com/download/docs/PSPxenon.pdf (Xenon Precision limiter).

And Hugh's comments about headroom in the old pro audio gear really ring true. If you've ever had a chance to work with a real Lexicon 480 or 960 you'll know how far you can push the headroom before it overloads.....its completely different from lower end semi-pro gear. In fact, I'd say this is the single most important difference between pro and semi-pro (aside from the engineers).


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## Nick Batzdorf (Aug 1, 2008)

"In contrast, when a digital system runs out of headroom, the distortion products result in aliasing, which folds the high frequency harmonic components back into lower parts of the spectrum."

The waveform gets squared off if you run out of headroom. That's quite different from aliasing, which you get when you record a frequency over 2x the sampling rate (and those freqs are filtered out by the anti-aliasing filter in your A/D converter anyway, so you don't have to be concerned with it).

Or is he saying that the distortion *products* are what result in aliasing? That's an interesting thought.

***
I'd recommend that everyone stay away from precision meter crap and even Bob Katz' site - great as it is - until their mixing skills are at an advanced level. My experience is that musicians get bogged down on nose-in-the-air mastering engineer bullshit that has only a passing relationship to music.

As synergy says, as long as you watch your levels at each link in the chain you're going to be okay. I used to get the most ridiculous questions about digital fly poop when I was at Recording - "do you think I should get a such-and-such in order to use 32-bit processing instead of summing in my mixer?" and so on. Meanwhile: what kind of mic are you using? "Radio Shack."


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## synergy543 (Aug 1, 2008)

Nick, I understand the "nose-in-the-air" attitude is a big turn off, however, the idea of getting people to mix at consistent and reasonable volumes could save a lot of ears...and that's why I recommend it.

I can't tell you how many musicians and engineers I've met that can no longer hear. I've even received many private messages from SEVERAL members on this forum (yes, our vi-members!) asking for mixing suggestions due to hearing loss. Its really a very serious and pervasive problem. Very, very sad.

So as the prominent editor of our trade bible, you really need to be aware of this situation as I'm sure people don't mention it to you face-to-face nor want it published.

And I know your gonna post one of your "eh-what-was-that?" jokes (love your humor btw), but this really is a serious problem that people can easily address if they are aware.


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## ComposerDude (Aug 1, 2008)

Nick: The input filtering on A-D are supposed to omit any frequencies beyond 1/2 the sample rate (Nyquist limit).

By creating a sharp corner in the waveform, digital clipping effectively creates an "infinite" series of high frequencies at relatively high level, having no particular harmonic relationship to the rest of the music.

While perhaps these clipping products are so loud they make it past the brickwall filtering and cause some aliasing, it's also possible that they just sound lousy due to their inharmonic relationship to the rest of the audio -- even within the passband. (A brickwall filter is 'supposed to' prevent anything above the Nyquist limit from getting through.)


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## ComposerDude (Aug 1, 2008)

Greg, I'm just addressing Nick's issue of clipping distortion, without prejudice against digital. I love digital when it's properly done.


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## Nick Batzdorf (Aug 1, 2008)

"Nick: The input filtering on A-D are supposed to omit any frequencies beyond 1/2 the sample rate (Nyquist limit)."

Ja. I said that. 

"While perhaps these clipping products are so loud they make it past the brickwall filtering and cause some aliasing, it's also possible that they just sound lousy due to their inharmonic relationship to the rest of the audio -- even within the passband."

Could be, but surely the main reason clipped digital audio sounds like ass is not because of the aliasing, it's because the waveforms are squared off.

***

synery, what you're talking about is the monitoring level. As you know, that has almost nothing to do with the levels inside a DAW! Yes, hearing loss is a very serious problem with musicians, most of whom have some degree of tinnitus (because of cochlear damage) by their late 40s. I calibrate my level (with a Radio Shack meter) for 85dB pink noise @ my listening position.

The point I'm making is serious, though: people get bogged down with incredible nonsense, in fact that thing about the 32-bit processing is only half an exaggeration. Of course I'm all for people understanding levels, and you're right to recommend that. But I really have fielded calls from people barking up the most absurd trees you can imagine, all because they read some cocktail party wank on the internet!


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## RickD (Aug 1, 2008)

I stopped recording things too hot when I read this discussion at PSW

http://recforums.prosoundweb.com/index.php/t/15038/0/


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## Scott Cairns (Aug 3, 2008)

Thanks for the link Rick.


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## Scott Cairns (Aug 3, 2008)

Nick Batzdorf @ Sat Aug 02 said:


> I calibrate my level (with a Radio Shack meter) for 85dB pink noise @ my listening position.



Ok, Im going to ask a really dumb question (never stopped me before!) - is there a way to gauge *roughly* how much 85db is without a db meter?

For example, the average volume of someone talking equates to a certain db level. The volume of your average acoustic guitar must be in a certain db range when played moderately...

Im just wondering if there's a frame of reference; like - can you hold a conversation with someone when your music is playing at 85db?

I dont monitor too loud now, but unless I get a db meter, it would be nice to have a rough idea of where im at.

Edit: here's an example of what Im talking about;

_If you whisper, the sound will measure about 20 decibels. Talking in normal voice will measure around 60 decibels. _ http://www.lpb.org/education/classroom/ntti/lessons/html2003/3sbFun.html (http://www.lpb.org/education/classroom/ ... sbFun.html) 

I guess that alone gives a good idea - if somone speaking normally is @ 60db, then certainly listening to music @ 85db aint too loud. o=<


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## Ashermusic (Aug 3, 2008)

Either way, there is no point in recording as hot as possible in a DAW. In the old days of analog, we did it to reduce the noise floor, but in a DAW if you are recording 24 bit, you have full bit resolution even at fairly low levels and the noise floor is not a factor.

I was a small part of that PSW thread and some pretty high priced talent, including Terry Manning, absolutely convinced me that you will get abetter sounding mix if you do not record really hot and do not mix with pre-fader with everything pinning the meters, even in a 32 bit float app like Logic and especially with a fixed point app like ProTools.


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## Scott Cairns (Aug 3, 2008)

Hi Jay, yes i read the entire thread and saw you post in there.

I dont need any more convincing - i already re-calibrated my converters to -20 dbfs -theyre being fed a line level signal from the pre-amp the output of which is set to 0dbvu :wink:


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## RickD (Aug 3, 2008)

Scott Cairns @ Sun Aug 03 said:


> Hi Jay, yes i read the entire thread and saw you post in there.
> 
> I dont need any more convincing - i already re-calibrated my converters to -20 dbfs -theyre being fed a line level signal from the pre-amp the output of which is set to 0dbvu :wink:



Hi Scott, there's lots of good information on that website, whenever I want info. about gear I do a search there. when I was working on acoustics, I found a link to Ethan Winers page that was all about acoustics, really interesting and good stuff there.


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## synthetic (Aug 3, 2008)

85dB is pretty loud. Its the dialog level for feature films (usually played softer in your local cineplex) and there is 20 dB of headroom above that for feature films in Dolby Digital. I don't monitor nearly that loud unless I have to. 

The recording at 0dB trend for digital started with 16-bit ADATs. You don't need to do that for 24-bit recording, and those kinds of levels clip everything else in the signal path. If you're mixing and processing a bunch of tracks recorded around 0dB, they don't have anywhere to go. And if you patch in analog gear, they're not built for those high levels.


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## Nick Batzdorf (Aug 3, 2008)

Hold it. We're talking about 85dBFS and 85dB C-weighted pink noise - two completely unrelated things. And what you hear in a theater that's supposedly "calibrated to 85dB" is measured at 10 miles away from the speakers; I'm measuring 5' away from them.

My ears are very intolerant of loud noises, in fact I almost always have to use my custom moulded earplugs when I go to see a film. Believe me, 85dB is a very moderate monitoring level.

By the way, ADATs are calibrated for 18dBFS.

"Ok, Im going to ask a really dumb question (never stopped me before!) - is there a way to gauge *roughly* how much 85db is without a db meter?"

I don't think that's possible, Scott. "A comfortable monitoring level that allows you to feel the impact of heavy metal without blasting the frick out of your ears" is the best description, and that's a pretty wide range. 

I bought my cheapo Radio Shack SPL meter I bought in the early 90s and don't remember how much it was, but it's not a lot. Cheap meters aren't accurate at low levels, but at higher levels like 85+dB they're just fine.


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## Nick Batzdorf (Aug 3, 2008)

"you will get abetter sounding mix if you do not record really hot and do not mix with pre-fader with everything pinning the meters, even in a 32 bit float app like Logic and especially with a fixed point app like ProTools"

Propaganda from the superfreak who doesn't even think Logic needs a Quantize key command.


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## Nick Batzdorf (Aug 3, 2008)

One of the nice features of my Blue Sky BMC monitor controller is that being digitally-controlled analog, it lets you set a reference level and then toggle between it and wherever else you have the knob set at the push of a button. When mixing I twist it to a pretty low level so I can check that everything can still be heard.


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## Nick Batzdorf (Aug 3, 2008)

To quote the indignant Nick Batzdorf, Jose:

"And what you hear in a theater that's supposedly "calibrated to 85dB" is measured at 10 miles away from the speakers; I'm measuring 5' away from them."

Point being exactly what you said.


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## synthetic (Aug 3, 2008)

You used to be able to download the room design and calibration materials from the Dolby website, I'm sure it's still up there. It's an interesting read and worth checking out if you plan to mix for cinematic release. It teaches you about A versus B chain, X-curve, all kinds of stuff. 

The Rat Shack meter is around $40. PITA purchase because you'll rarely use it, but it's one of those studio tools that's nice to have around. You can even align your home stereo system with it when you're done in the studio.


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## José Herring (Aug 3, 2008)

Nick Batzdorf @ Sun Aug 03 said:


> To quote the indignant Nick Batzdorf, Jose:
> 
> "And what you hear in a theater that's supposedly "calibrated to 85dB" is measured at 10 miles away from the speakers; I'm measuring 5' away from them."
> 
> Point being exactly what you said.



Yeah but you missed my astute observation about "inverse proportion", dude. :wink: 


@Dave 8)


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## Scott Cairns (Aug 3, 2008)

Thanks for the information guys. Id get a spl meter, but Ill probably use it once and then it'd gather dust. 



josejherring @ Mon Aug 04 said:


> Also, congrats on getting a decent recording chain.



Thanks man, I think you're somehow to blame - it all started with a talk about software compressors, and I now Ive bought a tube mic, mic pre and a Liquid Mix. :lol: 

(I'll tell my wife that you made me spend all this money, she will be calling to rag on you) :mrgreen:

On a serious note though, now that Im song-writing more than composing, Im recording more live instruments than ever before. Its a neccassary upgrade to my studio, so a big 'thanks' goes out for everyone's input.


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## Nick Batzdorf (Aug 3, 2008)

> Yeah but you missed my astute observation about "inverse proportion", dude.



True, you're thoroughly ass toot. My apologies for missing that.


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## José Herring (Aug 3, 2008)

Scott Cairns @ Sun Aug 03 said:


> Thanks for the information guys. Id get a spl meter, but Ill probably use it once and then it'd gather dust.
> 
> 
> 
> ...



Money well spent bro. You'll be utterly amazed at the amount of difference it will make in your recordings. With some well chosen outboard gear and a modest amount of money we can move our studios out of the "home studio" range and into something that sounds like pro studios.

Jose


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## José Herring (Aug 3, 2008)

Nick Batzdorf @ Sun Aug 03 said:


> > Yeah but you missed my astute observation about "inverse proportion", dude.
> 
> 
> 
> True, you're thoroughly ass toot. My apologies for missing that.



Better to be an ass toot than an ASS W...... oh never mind.

:? 

I really have nothing to come back with. Good one Nick!


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