Where should the mix be peaking during/after mix process?

Big Money Dilly

New member
I was wondering.. if you're going to master after mixing then how loud should your mix be? When i 'attempt' to mix everything is always 'just' under 0dbs only cos i put a limiter on the master channel - but from what ive heard shouldn't the mix leave a few db's of headroom available so that mastering can deal with bringing it up to max. level?

I find i have to keep my mix as loud as it can go without clipping so i can hear it properly and get the feeling that the audio does sound closer to a commercial sound, but now im wondering if im going about it the wrong way.

So what is the mix meant to achieve apart from giving each instrument its place and space amongst each other

Thank you!
 
I hope I don't sound jaded here (we've been through this around 100 times recently).

(A) Yes, you're mixing too loud. 0dBFS is NOT 0dBVU. You're using up all of your headroom. If you're tracking everything that loud, then you're using up all of your headroom at every opportunity to HAVE headroom. And you have more headroom available now than at any time in history with 24-bit digital technology. Why ANYONE would record or mix at hot levels is well beyond reason.

(B) Using a limiter in the mixing stage - You might as well shoot yourself in the foot (see everything above). Doing ANYTHING for the sake of volume during mixing - (just see above).

The point is that somewhere, you should have a volume knob. TURN IT UP.

Mixes aren't supposed to be as loud as finished production masters. That's part of the mastering process (I only wish it wasn't, as the volume war has really gotten out of hand in the last decade or so).

With that, I'll point you here -

https://www.futureproducers.com/forums/showthread/threadid/135252
 
haha I like how John has to point out the same thing in about 43 consecutive threads.
You should become a prophet or something ;)

..made me a believer.
 
The answer is real simple.
Avoid anyprocessing on the stereo output and make sure it doesn't clip.
The rest is just hype and pretty pointless.
Why leave headroom for the mastering engineer, when he doesn't need it?

There's is no doubt that any conversion or processing would benefit form more info(the whole point of using 24-bit anyway)
Why anyone would sacrifice, bit depth and precision for no reason whatsoever is beyond me. There's absolutely nothing to gain(!) soundwise by recording at low levels. Just make sure it's not clipping .

Headroom in the digital domain is a different beast than when you are in the analogue domain.

Can anyone give a technical description of ANY sonic benefits gained from recording tp peak at -12dBFS as opposed to -3dBFS?

When I mix I simply make sure there's no clipping and leave out any dynamic processing. I usually let it peak at around -2 or -3dBFS, and bring it to the mastering-suite. I have never once heard the ME wish for more "headroom"/fewer bits.
His only wish is material that sounds good that gives him freedom to work.
That means no dynamic processing or clipping. Headroom is of no concequense to him as the file can be attenuated accordingly at any time.

Relating nominal analogue operation levels at 0 dBVU to digital is also pointless without taking reference levels into account.
 
Other than reconstructive D-A distortion, the whole point of the "reasonable" levels (NOT "low" - as this is how it's been done since the advent of digital recording) is that EVERYTHING else other than the 1's and 0's themselves are designed to run at a particular level.

Although that level varies from piece to piece, it's almost always somewhere between -20 and -14dBFS when translated to digital.

Recording with the "meat" (not the peaks, but that's fine too) at around -14dBFS is how digital was designed to run in the first place. That was the whole point. Being able to keep the sound relativley "pure" without having to smash anything.

Even 16-bit digital is going to blow away even really good analog tape as far as S/N is concerned. 24-bit has SO much usable headroom... Well over *twice* that of the analog tape that everyone is now trying to emulate.

Can anyone give a technical description of ANY sonic benefits gained from recording tp peak at -12dBFS as opposed to -3dBFS?

Everything above - Ideal operating levels and reconstructive D-A distortion. Adding 10dB of unnecessary gain to every input (just to get a little more level) is permanently adding noise and distortion to the signal. On one signal, that might not be such a huge deal. When you put that noise and distortion on 20 or 30 tracks, it's a different story completely.

And again - I'm not pushing anything new here - This is how it's been done in every professional facility I've been in or have known of for over 20 years. Even back when 16-bit was king. Even back when most converters sucked. This is NORMAL. This is how it was all designed to work. The added detail of 24-bit was to *further* allow for this (attaining the most detail at nominal levels).

I'll give you the other part - Once everything is "in the box" then it really isn't going to matter where everything is (short of full-scale, of course). You could add gain until the peak hits -0.01dBFS without doing any damage to the audio - no matter what your D-A converters might think of it.

Which extends to the mastering phase - When mixes come in at those levels, most M.E.'s are going to turn them down considerably to avoid reconstructive distortion anyway. Whereas, if mixes comes in at normal levels (where the M.E.'s gear is also designed to run at) there isn't a reason for that.
 
now dont take my word for it...but I was always told that your aftermix should peak as clos to 0dB as possible...

thats what I was always told...dont take my word for it though
 
That wonderful information was the beginning of the end of sound quality, IMO.

With older converters and 16bit audio, the reconstructive D-to-A distortion was pretty nasty - That meant keeping the levels around 0dBVU was a good place to be (which again, has never changed). In 24-bit with modern converters that handle reconstruction with a little more "style" there is SOOO much headroom, that getting the peaks in the red doesn't really have any advantage. The M.E. will almost undoubtedly turn it down anyway (back to "gear wants to work at 0dBVU" again) to keep the signal as clean as possible.

So, once a mix is completely "in the box" and there isn't going to be ANY outboard processing or additional conversion, it isn't going to *HURT* to keep the peaks up near FS.

However, it certainly isn't going to help...
 
MASSIVE Mastering said:
Other than reconstructive D-A distortion, the whole point of the "reasonable" levels (NOT "low" - as this is how it's been done since the advent of digital recording) is that EVERYTHING else other than the 1's and 0's themselves are designed to run at a particular level.


I asked about the ADC(recording) not reconstructive distortion in the DA converter.


Everything above - Ideal operating levels and reconstructive D-A distortion. Adding 10dB of unnecessary gain to every input (just to get a little more level) is permanently adding noise and distortion to the signal.

Why are you talking about reconstructive filtering in the DA when we are talking about recording(ADC)

Yes, keep the analogue gear at nominal op-levels, but the DA actually has nothing whatsoever to do with the recording process.
 
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Maybe I side-tracked there a bit - But the point is valid. Once it's in the box, someone is going to have to listen to it (through the D-A) eventually.

As far as the input goes, pushing the analog signal messes with the signal before it even gets to the A-D. A lot of people get their sound down, and *then* set a level (pushing the unit's output much too far) without even noticing the sudden loss of focus and clarity. By the time it gets to the A-D, it's too late.

But again (and again) this isn't anything new - This is how the system was (is) designed to operate.
 
MASSIVE Mastering said:
Maybe I side-tracked there a bit - But the point is valid. Once it's in the box, someone is going to have to listen to it (through the D-A) eventually.


Of course, but that won't be affected by how close it was to 0dBFS when it was recorded. Any detrimental effect to the audio if it was recorded peaking at -0.5dBFS as opposed to -14dBFS will be due to limitations in the source/analogue equipment. NOT the ADC.

As far as the input goes, pushing the analog signal messes with the signal before it even gets to the A-D. A lot of people get their sound down, and *then* set a level (pushing the unit's output much too far) without even noticing the sudden loss of focus and clarity. By the time it gets to the A-D, it's too late.

Exactly my point.
It's not a matter of how close you are to 0dBFS. The issue is with the analogue gear and what happens before AD conversion.

But again (and again) this isn't anything new - This is how the system was (is) designed to operate.

Analogue, yes,
Digital is designed to capture information and does so more accurately at higher bit resolution.
It does NOT work better at-14dBFS(
You do NOT need that headroom once the signal is recorded.

On the analogue issue of trying to get the source sound as best as possible I agree wholeheartedly.
 
I hear where you're coming from, but my point is that if you're recording much hotter, the analog gear is no longer where it wants to be and the sound is being degraded before the conversion even comes into play. I'd argue that the degredation of the core sound is far more damaging than recording at nominal levels (still, where they were designed to be recorded at and have BEEN that way since before 1985). Again, this isn't my idea (although I find it works wonders) - This is the industry's standards. That (tiny) extra bit of resolution, that NO ONE is going to notice in 24-bit, isn't an issue. Distortion is.
 
im sure you can understand why i get confused just by reading the different responses in this thread.

First of all, thank you to all of you. Im trying to build my amateur understanding of what to do and yet im still confused. I dont know how to mix at such low volumes cos then u cant hear everything clearly enough. What is 0dbfs/dbvu? I did some reading and as far as i gather dbfs is just a way of measuring volume where 0 is the loudest and anything below is less. Im using cubaseSX3 and im guessing the volume meter on the master channel is just that then. You know the best way for anyone to learn is by example. If only someone just did a proper tutorial from the beginning of recording, mixing then mastering, showing the sound qualities at each stage. But the likelyhood of it happening is slim.

So again, in cubase i try to mix under the 0db mark on the master, but fail often cos everything sounds pants and low otherwise. whasupwitdat!

Cheers people!
 
Big Money Dilly said:
You know the best way for anyone to learn is by example. If only someone just did a proper tutorial from the beginning of recording, mixing then mastering, showing the sound qualities at each stage. But the likelyhood of it happening is slim.


Cheers people!

I have been thinking this for some time now as I have read until I can't read anymore and feel very comfortable with the theory of everything but I have yet to hear a pro or even semi pro mix before it hits the masterer. So I decided to purchase this book:
http://www.bhphotovideo.com/bnh/con...584&is=REG&addedTroughType=categoryNavigation

I just ordered it but can't wait to recieve it, I am hoping it will step my game up a bit. It just sucks cause if you aren't a intern or are around people whom do it, it seems like your mixes sound good to you but still don't sound completely professional.
 
^^Good purchase in my opinion^^

I like how the book tells you what they're doing and why and allows you to follow the project with the plugin demos.

However, their results sound like crap. It's a shame because the initial sounds are pretty good. The problem is mainly that they end up advising a compressor on just about everything. The waves compressor is good for dynamic control but the compressor isn't forgiving if your attack and release settings aren't right. Then they give your sounds an odd and unpleasant tone that gets annoying fast.

I recommend following all the projects and reading all the sections regardless of which genre you are focusing on. You will learn a different thing in each section and all sections are very useful.

The "mastering" section is okay but by that stage everything has already been compressed and there really aren't much to experiment with dynamics-wise.

I'd recommend following the projects and keeping tabs on the guidelines. Then doing your own independant mix without the literature to follow, just with what you feel sounds good.

Big Money Dilly said:
I dont know how to mix at such low volumes cos then u cant hear everything clearly enough.

Turn up your monitoring volume, that's what it's there for.
 
Ahh turning up the monitors. Will have to get the screw driver out and adjust the back of my KRK V6's then :P

I guess you're right - that way one can mix at low levels and still hear it loud, resulting in a non clipping mix. I may try it that way then, although still you guys all have varying ideas as to what levels you prefer to mix at. So is the 0DB on the cubaseSX3 master out DBFS?

I think a book may be beneficial here then. ill take a look at that one posted above.

cheers
 
I think most of us are probably recording at the same levels just looking at it differently. I believe that if you are recording digitaly with transients peaking anywhere from -6dB to less than -2dB your meat is going to be in the-20db to -14dB range. In which case it is pretty much as hot as it can get without the peaks clipping. I believe once you get the track recorded you can turn it down if you need to when adding effects and mixing during the mix stage. And if you're gonna be burning your own CD's, I do believe you should put a limitor on the two-track.
 
ZeroLatency said:
"I think [we] are probably recording at the same levels just looking at it differently. ...If you are recording digitaly with transients peaking anywhere from -6dB to less than -2dB your meat is going to be in the-20db to -14dB range."

That's it. When Massive Mastering is saying you should record around -14 dBFS, that means average level (RMS or VU; not peaks). Peaks should be no hotter than -3 dBFS, unless you want to flirt with intersample clipping, which won't show up as a normal clip and which takes either keen ears or a digital meter which shows that type of over. There's no need for tracking peak transients to -0.1 dB because analog limiters (save for the Dominator II) are notorious for not being true brick walls... Why risk wasting a take because you were hitting the converter too hot? Furthermore, if you record at higher than 44.1 kHz Fs, and if CD is one of your destination formats (rather than Super Audio or DVD), you'll need the digital headroom for the sample rate conversion, during mastering.

_andrew
 

That's it. When Massive Mastering is saying you should record around -14 dBFS, that means average level (RMS or VU; not peaks). Peaks should be no hotter than -3 dBFS,

when you say record around -14dbFS are talking about the master bus' average levels or are you talking bout recording the individual tracks & them peaking at -14dbFS (i.e. piano, or drums or bass etc.)? So far ive been taking it as meaning the master channel when we talk about peaking.

btw. i had a very quick go at recording at lower levels with my monitors louder (v.quick just to test). I did this with drums, bass and synth brass and had it peaking at around -.2.8db (dunno if its dbfs, it was cubaseSX3's meter) Then i put on an L1 on the mater and boosted it up to about -0.2db. I did come out good, but still loads to learn. To go off on a tangent, if i could get each track to not sound weak on its own, be clean and beautiful it would help about 50%. This must be where compression and eq on what you record from your synth come in.

Cheers!
 
-14dBFS = 0dBVU on every individual track.

Or lower - It really isn't going to hurt.

There is no additional "saturation" running hotter levels as would be the case with analog tape.

There IS a resolution factor - But that is easily pushed aside when you extrapolate that any digital signal that PEAKS above -47dBFS has a higher usable resolution and lower noise floor than a compact disc.

Makes those top couple bits seem pretty insignificant, no? There's a reason for that.
 
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