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Your recording levels are WAY too high.


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Yes. That low. "Zero" on the meters for Pro Tools is typically at -15dBFS. On some systems, it's set to -18dBFS.

 

The number one issue I have been seeing with recordings lately is printing WAY too hot. It not only potentially causes issues for the individual track(s), but for the mix itself. Change that one thing - start recording with your average signal levels at about -15, and your recordings and mixes will more than likely improve - oftentimes rather dramatically. :)

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Yes. That low. "Zero" on the meters for Pro Tools is typically at -15dBFS. On some systems, it's set to -18dBFS.


The number one issue I have been seeing with recordings lately is printing WAY too hot. It not only potentially causes issues for the individual track(s), but for the mix itself. Change that one thing - start recording with your average signal levels at about -15, and your recordings and mixes will more than likely improve - oftentimes rather dramatically.
:)

 

:cool:

 

It makes perfect sense.

 

When you say 'average signal levels', you mean to have most of the peaks hitting around -15dBFS, with the occasional stray going a dB or 2 higher, right?

 

I had been following Bob Katz's advice for the last few years, and peaking at no more than -6dbFS. Even coming down to that level had made a huge difference to my mixes.

 

They taught us, in a fully accredited college in 2004, to always aim for the highest possible signal level without clipping :facepalm:

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When you say 'average signal levels', you mean to have most of the peaks hitting around -15dBFS, with the occasional stray going a dB or 2 higher, right?


I had been following Bob Katz's advice for the last few years, and peaking at no more than -6dbFS. Even coming down to that level had made a huge difference to my mixes.

 

No, I mean having the average signal level in that -15dB range. You'll have peaks that exceed that, and Bob's suggestion of not exceeding -6dBFS with peaks may be a little conservative (IMO), but it's generally a good idea. Like I said, I don't like to see mine ever go past about -3dBFS, but they're generally lower than that. It depends on the nature of the sound source. If it has a wide dynamic range, and lots of transients and you don't want to squash it going in (compress when tracking), you may need to go even lower with the RMS (average) levels so that you don't clip on the peaks.

 

They taught us, in a fully accredited college in 2004, to always aim for the highest possible signal level without clipping
:facepalm:

 

That's probably a holdover from back in the 16 bit days. :)

 

I grew up in the analog era, and we used VU meters a lot back then. You learned pretty fast that you couldn't record every type of source at the same "levels" on the meters - while a synth pad might be fine there, try to print a tambourine at 0 or +3 on the VU meters and it's going to be really distorted because you saturated the tape. Sometimes that was a good thing... but you generally had a certain amount of room "above" zero on the meters where some things sounded really good. It compressed and distorted in a audibly pleasant way. You can't do that with absolute zero (full scale) in a digital system. You exceed that 0 dBFS point, and it's all over. It sounds terrible.

 

With analog tape, the noise floor (tape and machine hiss / hum) becomes an issue if you track with your signal levels too low. You wind up with a noisy recording. At the other end of the scale, if you hit the tape too hard, it starts to compress and distort. The basic idea was to hit the levels hard enough to optimize the signal to noise ratio, but not so hot as to cause audible distortion in the recording. With digital, the distortion is generally at the low end of the scale... things "chatter" as the signal dies out (unless you use dither... another topic for another time.)

 

With earlier 16 bit digital systems, a lot of engineers believed that printing as hot as you could short of clipping was generally a good idea. You had a theoretical SNR of 96dB, and the idea was that by using as much of that as you could (short of clipping) you optimized the signal to noise ratio. However, those digital tape decks and early DAW systems were generally connected to real physical consoles, and they often had sufficient headroom or could be trimmed to handle the hotter levels.

 

Today, when even most inexpensive interfaces offer 24 bit recording, there's really no reason whatsoever to try to slam those levels. Your converters are probably doing, what? 105 - 115dB on the signal to noise ratio? Your room noise and converters, mic and preamp noise and everything else are already conspiring to put your noise floor significantly higher than that. 24 bit recording theoretically supports 144dB - much better than even the best converters. Trust me - you've got plenty of "resolution" - by tracking with the right "reference", you're optimizing the system, and giving yourself some headroom to work with. To me, most converters sound bad when pushed hard. Back the levels off a bit, and suddenly everything opens up. :)

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I just read this article lat week. Interesting stuff. Out of curiosity, are you guys using the meters on your DAW or other software? I just picked up a Presonus Audiobox VSL1818, and the included driver/mixer software has a software mixer with meters for each channel. They read a bit different from the meters on GarageBand. So what do you guys prefer? I'm thinking the Presonus should be more accurate, since it's made specifically for that hardware.

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No, I mean having the average signal level in that -15dB range. You'll have peaks that exceed that, and Bob's suggestion of not exceeding -6dBFS with peaks may be a little conservative (IMO), but it's generally a good idea. Like I said, I don't like to see mine ever go past about -3dBFS, but they're generally lower than that. It depends on the nature of the sound source. If it has a wide dynamic range, and lots of transients and you don't want to squash it going in (compress when tracking), you may need to go even lower with the RMS (average) levels so that you don't clip on the peaks.

 

Hmm, that's pretty much exactly what I've been doing this last few years. My mixes are hitting -20dBFS RMS without even having to check the meter on the mix buss. That's all down to the K-System. Sounds like I'm on the right track. :)

 

S/N is the reason I asked what you meant by 'average'. I'd have problems with system/room noise if I kept peaks at or below 15dBFS.

 

That still doesn't answer my questions in the other thread!

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Just out of curiosity, was it in Ireland?

 

 

Yes in Ireland.

 

In hindsight, I should have picked one of the other colleges here to get qualified. But in the long run, I'm not sure how much difference it makes - you learn more on the road with bands, making endless demos, and doing constant research than you ever would on a City & Guilds accredited Music Tech/Sound Engineering course.

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Good read Phil. the only thing you might have left out is the cause of running hot signals.

Most people use their ears when doing these gain adjustments and really dont know two clicks about db's and meters.

The cause will be either the monitors or headphones being set too low. They are the end of the chain and if they

arent set up for optimal levels.

 

Many who start out dont have the best gear and have an even harder time setting it up.

If thay have an interface with .1W headphone amp built in and a cheap set of headphones with a really low spl level,

thay are going to have a hard time hearing what they play or sing through. so what do they do to compensate? They crank up

the input levels at the front end and saturate the gains so what they are hearing through the headphones is acceptable.

 

We have to keep in mind alot of these guys play live music pushing 110db. give them some cheezy headphones and they

just havent developed the ability to focus on their playing at lower levels. They are used to feeling the music as much as

hearing it and playing their instrument, and often times for guitar players pushing speakers to their limits sounds better to them.

 

The fix is a good quality headphone amp with low noise and good studio headphones with amble headroom.

This way thay will be forced to run their mic gains lower and it still sounds powerful in the headphones.

 

A good set of monitors for direct tracking works wonders too. I intentionally run my monitors louder so I wont push the

tracks too high. I have a 200w reference amp and several sets of passive monitors that are as loud as I'll ever need them

for mixing. I can flip a switch and patch in the PA system for another 2000w too if I want to hear a mix at live levels too.

I dont use them very often but a good mix at 85db should still sound even better through a high end system.

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Good read Phil. the only thing you might have left out is the cause of running hot signals.

Most people use their ears when doing these gain adjustments and really dont know two clicks about db's and meters.

The cause will be either the monitors or headphones being set too low. They are the end of the chain and if they

arent set up for optimal levels.


Many who start out dont have the best gear and have an even harder time setting it up.

If thay have an interface with .1W headphone amp built in and a cheap set of headphones with a really low spl level,

thay are going to have a hard time hearing what they play or sing through. so what do they do to compensate? They crank up

the input levels at the front end and saturate the gains so what they are hearing through the headphones is acceptable.


We have to keep in mind alot of these guys play live music pushing 110db. give them some cheezy headphones and they

just havent developed the ability to focus on their playing at lower levels. They are used to feeling the music as much as

hearing it and playing their instrument, and often times for guitar players pushing speakers to their limits sounds better to them.


The fix is a good quality headphone amp with low noise and good studio headphones with amble headroom.

This way thay will be forced to run their mic gains lower and it still sounds powerful in the headphones.


A good set of monitors for direct tracking works wonders too. I intentionally run my monitors louder so I wont push the

tracks too high. I have a 200w reference amp and several sets of passive monitors that are as loud as I'll ever need them

for mixing. I can flip a switch and patch in the PA system for another 2000w too if I want to hear a mix at live levels too.

I dont use them very often but a good mix at 85db should still sound even better through a high end system.

 

 

Some good info here as well... So as kind of a newb to decent equipment, where do you guys reccomend setting the gain levels on your monitors? I'm using KRK Rokit 8's, and while I've had them set to 0db, I'm thinking I might up that a hair...

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Some good info here as well... So as kind of a newb to decent equipment, where do you guys reccomend setting the gain levels on your monitors? I'm using KRK Rokit 8's, and while I've had them set to 0db, I'm thinking I might up that a hair...

 

Go to www.digido.com and search the documents there for 'level practices'.

 

What you want to do is use a SPL meter at the listening position, and have your monitors producing 80-85dBSPL (the lower end of that SPL range if the room is small) at that position for playback of -20dBFSRMS stereo uncorrelated pink noise.

 

You can also calibrate for higher levels of pink noise, such as -14dBFSRMS and -12dBFSRMS, and mark the positions on your monitor gain knob. These higher positions are useful for applying 2 buss limiting to get the levels up to 'commercial' standard, or if you have some creative reason for mixing something 'loud'.

 

If you follow the gain structuring/level practises in Phil's article in the OP, and keep an eye on an RMS meter on your mains, you'll be hitting -20dBFS average RMS on everything you mix, within a matter of weeks.

 

For tracking, you may need to use even higher monitor gain. To paraphrase WRGKMC, turn up your monitors/headphones, not your input gain.

 

I'm happy to be corrected in any of this. This is where I am with level practises right now :)

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With earlier 16 bit digital systems, a lot of engineers believed that printing as hot as you could short of clipping was generally a good idea. You had a theoretical SNR of 96dB, and the idea was that by using as much of that as you could (short of clipping) you optimized the signal to noise ratio. However, those digital tape decks and early DAW systems were generally connected to real physical consoles, and they often had sufficient headroom or could be trimmed to handle the hotter levels.

 

 

So should I continue to print short of clipping?

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No, I wouldn't recommend it, even with a 16 bit system. Record with your signals calibrated and referenced to whatever "Zero" is on your system - that doesn't change just because you're using a 16 bit system instead of a 24 bit system.


What are you recording with / into?

 

My soundcard is an M-Audio Delta 1010. I'm using an old PC from 1999 that runs on Win98SE and Cubase VST. The setup has been dedicated all these years without any updates, so it's really stable. I don't use any midi with it, I just track audio for my demos. It doesn't even have usb 2.0, so I have to save to CD-R to transfer files :o

 

I also use Garageband on my iPad2/Alesis ioDock or on my MacAir on occasion, but only if I need to be portable.

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My soundcard is an M-Audio Delta 1010. I'm using an old PC from 1999 that runs on Win98SE and Cubase VST. The setup has been dedicated all these years without any updates, so it's really stable. I don't use any midi with it, I just track audio for my demos. It doesn't even have usb 2.0, so I have to save to CD-R to transfer files
:o

I also use Garageband on my iPad2/Alesis ioDock or on my MacAir on occasion, but only if I need to be portable.

 

I used that same kind of system for years before upgrading my computer, in fact I still have it as a backup DAW.

The main problem with an older system is its speed processing. A 16 bit system is just a matter of hard drive storage,

not the Audio quality. Win 98 has 32 bit hard drive compatibility and drives can be formatted to take advantage of the

larger partitions. XP and up use NTFS which allows for very large drives. Its a matter of work space on the drives not

the quality of the audio obtained from the sound card so using the proper meter levels holds true on any system, if not more so

on an older system. Its speed processing and saving the waveform may benifit from using lower levels.

 

You can use that 1010 card in a newer computer and it has drivers that work with XP and Win 7.

I bought a dual core 3.2G system a few weeks ago for $50. It even had the liscence for Win XP pro

if I wanted to use it but I installed win 7 on it. The main benifit of A newer system is it will let you run

newer software, and of course take a huge jump in speed. That old copy of Cubase 32 was good in its day

but you can do a whole lot better. Cubase 32 will run on XP and Win 7 too. I'd likely try Reaper as a freebee

if you dont choose to upgrade the DAW program right away.

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