Do you know how to read your meters? -18dBfs theory?

[Machinesworking]

To be clear, I'm not saying we should have all levels up at 0dBFS but I'm calling bullshit on -18dBFS as being the sweet spot (unless you are using hardware)

I think there is a ton more reasons for setting a low level line like that than just using hardware. If you want say a mix with heavy synth bass, chip tunes or distorted rythmn guitar, for examples of sounds that are by their very nature maxed out db wise, IE not dynamic, then getting a good sounding mix with more dynamic instruments is a lot harder if you're near 0dbfs. -18 might be too low, but at the very least -10 seems smart considering that some really great mixes have been done at that level.

This is one of those things where doing it cause zero losses in literally every way. Whereas not doing it might just make mixing down a bit more difficult. It's a bit like wearing a seat belt.

[phaded]

To be clear, I'm not saying we should have all levels up at 0dBFS but I'm calling bullshit on -18dBFS as being the sweet spot (unless you are using hardware)

watch this: https://www.youtube.com/watch?v=hazcYwkCP2M

especially the part where he start's playing with the 'headroom' knob

still bullshit?

I think a little bit of the sticking point is the actual term 'sweet spot', which is kindof ambiguous I guess. But its just referring to the optimum level for a processor to perform its best at. Its not a mystical thing at all... its as science as 32bit math, yo. 'Sweet Spot' just sounds more rockstar.

If you want some other clear examples of this, check out any of the Waves Signature Series... that little light on all the plugs that tells you what level to set your signal at?

I've got some good money that its probably.... around 0db VU.

(btw that UBK-1 is one of the most wicked things I've put my money down on.. highly recommended if you're a slave to the Ilok)

[lunabass]

watch this: https://www.youtube.com/watch?v=hazcYwkCP2M

especially the part where he start's playing with the 'headroom' knob

still bullshit?

Yeah that UBK does sound sweet. It has a headroom control which adjusts exactly that, less headroom causes clipping. Does he mention anywhere that the levels coming into the plugin need to be at 0VU or something like that because I missed that part. He also mentions that the plug-in still has loads more headroom than a hardware equivalent.

I'm glad you've mentioned Waves plugins as these are what I've tested myself (and why I have the opinion I do)

If I input something close to 0dBFS into an API hardware EQ it sounds terrible, as you'd expect, the EQ is setup for 0VU.
If I input the same level through the Waves plug-in it sounds pristine. Stop and think about that for a sec! Do you really think Waves are going to make their plugins sound terrible to those that dont understand gain structure and decide to input something close to 0dBFS?

I admit I haven't tried the Waves signature stuff yet but I've done a very similar test to Stromkraft using Waves SSL and API and as far as I can tell there is no difference if I input -18dBFS or -2dBFS. Might demo the waves signature stuff and see if they're any different.

Anyway, if I happen to get a plug-in in my collection that needs 0VU I'll definitely do just that, the difference will be obvious just as it is when I use hardware, I've just not heard one that needs that yet

[Stromkraft]

I'm glad you've mentioned Waves plugins as these are what I've tested myself (and why I have the opinion I do)

If I input something close to 0dBFS into an API hardware EQ it sounds terrible, as you'd expect, the EQ is setup for 0VU.
If I input the same level through the Waves plug-in it sounds pristine. Stop and think about that for a sec! Do you really think Waves are going to make their plugins sound terrible to those that dont understand gain structure and decide to input something close to 0dBFS?

Well, this is the Spike first channel of the Waves NLS in mono fed a sine wave bass signal at 0dB volume level in Operator. What was going in was probably a little bit lower, like -0.5dB or so.



This did indeed sound like crap, which I must say is to be expected when the maker have modeled analogue hardware and how it reacts to frequencies at different volume levels. Yes, lowering the drive improves the sound, but I found my "sweet spot" for the sound I wanted feeding it a -12dB signal instead. Which is a peak level I think. The RMS levels of the track after the NLS plug-in was about -9dB and -3dB peak.

Lowering the preamp drive with about 0dB going in doesn't give the same sound as feeding it the lower volume signal with increased drive. I wanted that very subtle distortion this dynamic plug-in can deliver.

When I used the -12dB level in Operator I could have added more dynamic plug-ins that can push the levels up a few dB sometimes later in the chain. With a signal close to 0dB that would be more difficult.

What's simpler, to add gain or remove gain in your signal chain? I find it simpler to add gain from a reasonable maintained volume level trough one signal chain. I think it typically sounds better too.

Of course to really push and overdrive something and put it at a lower volume level in the mix can be really nice, but you probably can't push every track in that way and make it sound good. Unless you're Richard D James anyway and if your are attempting something like that you know what you're after hopefully.

I think using your ears is as a good idea as is using an additional meter like FreeG for getting your RMS levels as that usually is more interesting than driving if not every track, but many tracks, to "peak efficiency" or just. That way you leave no room where to let things grow when you add them together.

[phaded]

Do you really think Waves are going to make their plugins sound terrible to those that dont understand gain structure and decide to input something close to 0dBFS?

You're still missing what's going on here. It's not that they are concerned with the end-user's ability to know proper gainstaging (or reading the manual for that matter). When modeling old hardware units these days, the detail in replicating the signal paths and behavior of units is DEEP. They are modeling saturation characteristics of the inputs&outputs (hopefully) and everything in between, down to the last transistor depending on the attention to detail.

Now ask yourself... HOW would they go about this? By using the same standards that have been developed for over a century now. Check it:

VU means Voltage Unit... its a power measurement essentially

Anything you see VCA based (compressors, synths, etc)... that would be a Voltage Controlled Amplifier

Before the days of midi control, you know how hardware used to communicate and send signals to each other for sidechains etc? CV. Which is... drumroll... Control Voltage

Even how notes and octaves are standardized in synths... Voltage based... from Wiki:

Volts per octave. This standard was popularized by (if not created by) Bob Moog in the 1960s; it was widely adopted for control interfacing. One volt represents one octave, so the pitch produced by a voltage of 3 V would be one octave lower than that produced by a voltage of 4 V. Notable followers of this standard include Roland, Moog, Sequential Circuits, Oberheim and ARP.

http://en.wikipedia.org/wiki/CV/Gate

So the point is in order to model of these units, they are using the same rules as in the analog domain in order to produce the most accurate representation of the hardware in the digital domain. Even more to the point- the onus is more on YOU to know/understand how the plugins operate in order to use them correctly in your sessions.

I'll say this- I'm not arguing one bit the necessity in the end to 'use your ears'. The suggestion is some best practices that will help you understand to a better degree what you are hearing when processing audio, and of course, to make best use of your tools.

Indeed you will find on plugins that driving the inputs hot gives a pleasant sound... but it may clip unpleasantly on the output. Other times you might find a synth or plugin that has some type of soft clipping / limiting / saturation on the output that you can drive the outputs. But without turning things down after you crank the gain somewhere, how can you tell what you've done to your signal?

I'll leave one last thing here for anyone interested... this is a quick rack I put together that I use religiously for these purposes:

http://postimg.org/image/xnu67semx/

I drop whatever plugin I'm using in the middle section of the rack (the 'INSERT' part). Then I set initial levels going in and can adjust inputs/outputs with the macros to get the sound I'm after. I'll typically drop a utility or two to replace the VU meters after I'm done. Took a little while to incorporate into my flow, but takes seconds for me to setup now. Especially if I'm using Hornets VU Meter or Track Utility. Those things are MINT.

[stringtapper]

From reading posts from a few mastering engineers over at Tweakheadz years ago I have always been under the impression that the -18dBfs guideline was about leaving headroom for the mastering process.

[phaded]

From reading posts from a few mastering engineers over at Tweakheadz years ago I have always been under the impression that the -18dBfs guideline was about leaving headroom for the mastering process.

Sortof.. really in that scenario the concern IS peaking/clipping your outputs... they can turn down your track for extra headroom if needed. Somewhere around -6 dbfs is actually perfectly acceptable to send to an Mastering Engineer. You could even go closer to -3 but you're in that sneaky ISP territory that you would want to monitor for.

If you want to go to Church on this topic reading engineers debates, I'd (sortof) recommend diving into the following thread. Be forewarned though, its a bit of a shitshow to begin with before there is some clarity. Def a good read tho:

https://www.gearslutz.com/board/so-much ... tored.html

[login]

Have been learning to mix in the last years and I had never came across this topic, reading this as well as all articles related on various sources and also the famous "Why yout ITB mix doesn't sound as.." thread at gearslutz made for a lot of researching.

My first reaction is surprise that this stuff is never mentioned at video sites as groove3 and lynda.com, and well fater I tried it's way easier to mix with so much headroom now.

I think some sources in this topic aren't clear or provide the whole lot of information, some youtube videos for example tell you to achieve it with the mixer faders LOL!

Others never tech you how to read VU meters or never specify that you must check this plug in to plug in and in to the mix busses so you conserve the headroom all the way to the master channel.

So I think a whole article or series of videos on this topic is kind of missing, many make quite clear the basic point but they go deep in to the details.

[Stromkraft]

So I think a whole article or series of videos on this topic is kind of missing, many make quite clear the basic point but they go deep in to the details.

Well, as someone said the reason this is not mentioned is in many cases because this kind of thing is assumed. In addition it can be tricky to instruct people beyond "making sure we're not clipping" as there are no absolute levels that works for everyone — there are several schools of thought on this even if there are strong tendencies —  and levels and gain staging may not be the main selling point for making the video in question.

That said I've seen a few cases where the instructor makes a point of setting levels with headroom and mastering in regard. In the recent free video from Vespers on snares he talks about the importance of choosing a drum library with samples with some headroom left to go so that you can process and refine these to your liking (using BFD3 as a good example) also pointing out maximized full-on samples also have their place when you just want that specific sound.

While I think it would be nice with someone addressing this in detail and especially using very illustrative and cogent examples so that also naysayers — I've met a few — can embrace something of all this for their benefit, I also think this could just be referred to or just be done when demonstrating other techniques.

There are no right and wrongs here I think, only the results you get and how you act and react on these.

[Verside]

If you’re using analogue hardware such as EQ, Mixers, Compressors etc with your DAW then -18dbFS is excellent advice, if you aren’t, then I believe it to be a non-issue. As Angstrom said, there is nothing wrong with following this guys advice, it’s actually OK, but people rarely take the time to understand or test it for themselves.

Take analogue emulation plug-ins for example. Do you really think Waves will emulate their plug-ins so as to distort at 0dBFS just like a piece of Analogue Hardware would with that much level? Try running that much level through a hardware Neve EQ and tell me how it sounds.

Run a simple test.
1. Take a loud drum loop with peaks close to 0dBFS.
2. Place an audio effect rack onto the track.
3. Place an analogue emulating EQ into it and make a few cuts or boosts. Make it noticeable and ensure if boosting that it doesn’t clip. Name this chain “dBFS” or something that indicates it’s high level.
4. Duplicate this chain (and therefore all of the EQ settings) and name it “VU”.
5. On the VU chain, place a Utility set to -18dB before the EQ plug-in. This will ensure it receives 0VU “the sweet spot”
6. Place another Utility after the EQ set to +18. This enables you to do an A-B comparison between the 2 chains without any difference in level.

Now just solo each one and listen. Do you hear a difference? If VU sounds “better” to you then it’s obvious what you need to do.

For what it’s worth, I’ve done this test (although I did them at -12dBFS as the studios I’ve worked in, 0VU was calibrated to -12dBFS) on a bunch of my plug-ins without hearing any difference BUT that doesn’t mean you won’t get a different answer with your plug-ins, DAW and ears.

Great advice, thank you sir!

[MiMic23]

So I'm glad I came to this because I haven't found the Ballistics for this in the manual at all. Maybe some has. But from my experience, yes, -18db is 0db (unity)for ableton channel strips. I Did a test for it by doing gain staging from a ql5 console signal patching through dante to recording tracks in the daw. After gain staging for the signal through the console of it hitting -18db is 0db to analog which is what is in the ql5 console manual and was told to me by Robert Scovill when I asked him during a workshop. then patching into ableton after this. It hits -18db as well. So to get the perfect amount of bits without over processing is -18db on abletons meters for recording.