# What is gain staging and is it important?



## José Herring (Sep 30, 2020)

I've heard the term a lot and did some research a few years ago on gain staging. I ran across an old engineer trying to explain it and poor dude. It just seemed like he was trying to apply some old analog knowledge and hadn't really caught up to the digital age. 

But, then I'm like but what if? What if it's important?

So what is gain staging? Is it worth learning and doing?


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## Uiroo (Sep 30, 2020)

I heard people pointing out how important it is and I heard people saying it's not at all.

The way I understand it gain staging is relevant for analog mixing desks, that's it. But I have no idea.


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## doctoremmet (Sep 30, 2020)

Mixing with Mike is a rather nice YT channel. Michael has done a number of videos on it, I think. Also, check @Joël Dollié who has an even cooler channel, pertaining to mixing an orchestra in the box! I bet Joel is in the “gain staging matters” corner...


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## el-bo (Sep 30, 2020)

From my basic understanding, it’s especially important when using plugins that are modelled on analogue gear, to get the right volume level going in, as this type of equipment would usually have had a limited sweet spot in which it could perform it’s processing magic.


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## thorwald (Sep 30, 2020)

Gain staging in the analog world is finding the best signal to noise ratio. If your mix (or track) is too low, you get more analog noise (hiss), this is not preferred. If your track (or mix) is too loud, it (your signal) can get saturated, or even distorted, which is also not preferred. The key is finding the right balance, so that the final result has the right ratio. Sometimes, especially with analog plugins, saturation, or even a distorted sound is preferred, likewise a more hissy sound, to give your music an analog feel.

Today, gain staging matters just as much, things above 0 DB can clip (distort), which is usually not preferred. It is generally a good idea to give yourself and your music a bit of head room, usually about -2 DB, just in case things get louder. This means mixing everything at -2 DB. Arguably, you can use a limiter, and let it take care of your volume levels, but this does not sound too great for example on orchestral music. The sound you get also depends on the limiter, and how much you are going over your set goal (usually a LUFS value).

If you are subscribed to any streaming service, such as Deezer or Spotify, you might have noticed that the track has a crazy amount of compression. This can be because it sounds like it originally, which is the case for a lot of hybrid trailer music that is meant to be loud, or because that particular service slapped on a limiter, since your track is louder than the expected LUFS values of the service. This is definitely not what you want, and this is why you see tracks or albums mastered for various streaming services as well.

So, yes, gain staging matters in the digital world just as much ☺️

Hope this helps!


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## SlHarder (Sep 30, 2020)

As a hobbyist who returned to midi 9 months ago after a 15 yr hiatus I found that incorporating gain staging in my mixing process produced a significant improvement in the end result. Gain staging provides a great foundation to base all my other mixing steps on. Once I've gain staged I can efficiently move forward through a mixing session without having to do a lot of backtracking to fix stuff. Try it, what have you got to lose?


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## robgb (Sep 30, 2020)

Gain staging doesn't matter in the digital world unless you're using plugins (particularly analog modeled plugins) that distort when overloaded. If you're using regular digital plugins, the only gain you need to worry about is your master bus. Now I know a lot of people here will go nuts when I say this, but Kenny Gioia proves the point with this video:


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## Voider (Sep 30, 2020)

Gain staging is very important and it doesn't apply to analogue gear only, it's an universal thing regarding your mix.

Gain staging is basically to take care of your volume meters and your "volume budget", the purpose of gain staging is to make sure that you never exceed your volume budget to avoid distortion or clipping, and beyond that, give all of your instruments enough space to shine and sound good.

You could peak at 0dB with just a bassline and a lead synth, or with a full orchestra with 45 tracks. To fit the latter into the same space as the first while still sound clear and powerful, gain staging is inevitable.

More in detail it's as well about the signals between your plugin chain.
If you'd have a chain like this: Synth -> Chorus -> Delay - Reverb, you could easily be clipping above 0dB at the chorus plugin already but lower the post-volume afterwards, which would still lower the overall quality of your signal (if not done on purpose, sound design wise). So with gain staging you make sure that your levels between the plugins are all fine for a clear final sound.



robgb said:


> If you're using regular digital plugins, the only gain you need to worry about is your master bus.



That's simply not true, as explained in my example of a single channel where you could easily run into distortion while your master bus and even that channel's output meter both look fine.

The master bus doesn't contain any information about the clarity of the signal, it just shows you the final output volume.


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## vitocorleone123 (Sep 30, 2020)

I believe, but could be wrong, that if you're 100% digital, then you have more wiggle room on the gain staging, but it still matters (as Voider described). Whereas, if you have even just 1 plugin with analog emulation, or even potentially just one mode of analog emulation, gain staging can matter a great deal more.

It might be worth consider gain staging as a healthy habit, even if fully digital. Perhaps not stressing about it though, in that scenario. But worth keeping in mind.


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## GtrString (Sep 30, 2020)

Gain staging is a fundamental process in all mixing, as mixing is about balancing volumes in creative ways. You gainstage everything from instruments, parts, effects and frequencies. You might even argue that all you can do with audio is gain staging, but ofc a lot of different types of parameters.

In daily mix lingua, gain staging usually is something you do in the beginning of your mix process in order to better manage balances in your daw.

Absolutely fundamental to learn. Not only to find out what others do, but to find your own sweet spots and preferred balances, so you can develop your own sonic signature.


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## Akarin (Sep 30, 2020)

Simple: get a VU meter (surprisingly, Cubase doesn't come with one). Lots of free ones to choose from. Make sure that all your tracks and busses hover around 0VU. To adjust that, don't use your faders but the gain knob. Doing this, you'll hit the sweet spot when using analog emulation plugins. That's all there is to it in the digital world.

EDIT: use the $5 plugin from Hornet Plugins. VUMeter. It does it automatically for you. You just play the track and it will adjust the gain to hit 0VU.


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## Living Fossil (Sep 30, 2020)

José Herring said:


> So what is gain staging? Is it worth learning and doing?



Haven't read through all posts, however, these are the basic things to keep in mind:

- DAWs work internally with higher bit resolutions (normally either 32 or 64 bit floating)
For the DAW itself, overs in the signal chain are no issue, as long as the master doesn't clip.

- however, some plugins might have problems with overs due to their layout.
That's why it's worth avoiding getting over 0 in the signal path.

- as has been mentioned (by @el-bo ) some plugins require a more or less specific dynamic range in order
to do the desired task. If they have insufficient input control, i usually put a Gain plugin in front to adjust the volume that goes in. This can have a drastic impact on plugins like some compressors, tape sims etc.

- When working with busses that you want to automate, you have to keep an eye on the sends of the respective channels. Otherwise e.g. the relative amount of reverb will change when you adjust the volume via the bus automation

- Therefore, the best thing for more included tracks is VCA automation.
In Logic, VCA automation makes gain staging a breeze, since you can combine different ones.
You have busses without overs that sum into an over? -> Easiest solution is just to lower the master VCA.
(BTW changing the volume of the masterbus itself it something i would never do, specially not if there is dithering going on.)

- Methods that rely on concepts from working with consoles make sense if you used to work for thirty years with a console. Otherwise it makes no sense. In the digital realm, no noise is created out of nothing in the signal path and 24bit files maintain their full resolution also at very low levels, so some strategies have no real meaning any longer.


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## NoamL (Sep 30, 2020)

1. Take a good live recording

2. Lower the clip volume so it's peaking at -9 or so in your DAW (this is to account for the mastering of the recording, and to give yourself mix headroom on YOUR mixes)

3. Set up all your orchestral instruments so that they match the reference volume. If a VI needs to be turned down in volume, do it as "early" as possible in the signal chain e.g. CC7 or the mic mix inside Kontakt.

You now have good gain staging... in principle... if you do stuff like add EQs just make sure the processed signal has the same apparent volume as when bypassing that plugin. Some plugins have non-linear response, meaning the effect will kick in more if you have higher input signal. Tape saturation would be an example. If you want to add gain to the input of the plugin that's fine as long as you subtract it again at the output. Some tape plugins do that automatically.

Gain staging does not mean "gradually adding gain as you go through the mix chain," so if you have things set up well at the "earliest" part of the mix chain, you are set.

Mixing electronic music is probably more complicated but for orchestral music, "gain staging" primarily means "accurate gain" for all the instruments + making sure you have headroom.


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## John Longley (Sep 30, 2020)

If you are using a modern DAW, you have up until you truncate below 32 Bit (64 Bit DSP) to turn it down. The only functional problem is that your DAC will clip. If you choose to follow rough analog friendly protocol so you can insert your outboard (my last console was around -16Dbfs = 0 Dbvu (it varies), or you just smash it all into the stereo master and turn it down doesn't matter. Learning to use routing and how to manage your mix is still useful-- but the days of Pro Tools TDM clipping is gone. Learn it out of good hygiene, not fear you will cook your mix.


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## labornvain (Sep 30, 2020)

Gain staging is simple. Bypass all plugins if you have any inserted on that track, then put your track fader at 0db, which should be its to default setting when you create a new track. Then adjust the track source ( master volume on the VI or clip gain if it's an audio part) so that the level reads -12db average with the peaks not exceeding -6db~.

Some people prefer to set the gain at -18db, But I use a lot of analog modeled plugins and they seem to be happiest, at least as a starting point, at around -12db.

It's true that in a modern DAW running at 32bit float, which effectively gives you infinite headroom, gain staging is not the critical function it was in analog days.

But who wants to have their levels and their faders flying all over the place?

By maintaining a consistent signal level across all tracks through the signal chain, it greatly simplifies the work flow.

And it's not just about analog modeled plugins. Let's say you have a track where the gain is really low. So now you want to adjust the send to a reverb. With such a low level on that track, you might have to turn the send all the way up just to get a good level going into your reverb. This would be surely undesirable.

That's just one example of where improper gain staging may create bizarre work flow anomalies.

To me the best thing about proper gain staging is predictability. I know that when I insert a plugin on a track, the levels feeding into that plug in are going to be right. The send levels are gonna be right. 

Also, by having all of my tracks gain staged to -12db, the positions of the faders become useful information. I know that if I have a fader pulled way down low, it's because it's a track that I want to be very quiet in the mix and not a track that is so hot that I have to have to lower the fader to compensate.


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## Nick Batzdorf (Sep 30, 2020)

Note that the term "gain staging" just means setting levels throughout the chain. What people really mean is *good* gain staging - setting the right levels.


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## mscp (Sep 30, 2020)

Nick Batzdorf said:


> Note that the term "gain staging" just means setting levels throughout the chain. What people really mean is *good* gain staging - setting the right levels.



Back in the days of audio engineering school, a lot of my schoolmates were baffled by the notion of parallel compression, but honestly, it's all in those two names. The rest is maths.


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## mscp (Sep 30, 2020)

José Herring said:


> I've heard the term a lot and did some research a few years ago on gain staging. I ran across an old engineer trying to explain it and poor dude. It just seemed like he was trying to apply some old analog knowledge and hadn't really caught up to the digital age.
> 
> But, then I'm like but what if? What if it's important?
> 
> So what is gain staging? Is it worth learning and doing?




It's not as important in the digital realm as it is in analog, but is still important José.

What do you know so far about it?


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## gohrev (Sep 30, 2020)

Akarin said:


> Simple: get a VU meter (surprisingly, Cubase doesn't come with one). Lots of free ones to choose from. Make sure that all your tracks and busses hover around 0VU. To adjust that, don't use your faders but the gain knob. Doing this, you'll hit the sweet spot when using analog emulation plugins. That's all there is to it in the digital world.
> 
> EDIT: use the $5 plugin from Hornet Plugins. VUMeter. It does it automatically for you. You just play the track and it will adjust the gain to hit 0VU.



Hello Akarin, 
I bought VUMeter this summer, thanks for the tip in another thread..  
Do I place VUmeter on every track?

Cheers,
W


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## Ashermusic (Sep 30, 2020)

Less important in floating point than fixed point DAWs, but still worth paying some attention to.


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## Nick Batzdorf (Sep 30, 2020)

berlin87 said:


> Hello Akarin,
> I bought VUMeter this summer, thanks for the tip in another thread..
> Do I place VUmeter on every track?
> 
> ...



If you want, but your tracks already have meters. (They're not actually VU meters when you're using digital audio, but that's a discussion for mastering engineer cocktail parties.)


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## José Herring (Oct 1, 2020)

I'm reading through every thoughtful response. It is taking me time. But, I just wanted to post to say that I am reading and taking in a lot of information.

I did find a course online by Denis Sands one of my favorite all time film music mixers. I'm considering taking it but it's a bit prices for 7 hours worth of material. But, I may just do it. 

Keep the responses coming. Thx


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## jcrosby (Oct 1, 2020)

el-bo said:


> From my basic understanding, it’s especially important when using plugins that are modelled on analogue gear, to get the right volume level going in, as this type of equipment would usually have had a limited sweet spot in which it could perform it’s processing magic.


Yes indeed. This is where gain staging makes all the difference when mixing ITB. Modeled plugins are calibrated to their real world, non-0 dBFS big brothers. So while in theory you can run a mix as hot as you want in a 32 bit float engine, the attitude that gain staging doesn't matter goes out the window as soon as you start to work with plugins that are calibrated to mimic 0VU calibration. There's about as much wisdom in that belief as saying if you want to write a melodic piece it doesn't matter if you play in a completely unrelated key. Sadly that misconception is a lot more pervasive than you'd think...

Does that mean you shouldn't deliberately run plugins hot if that's a particular sound you're after? Absolutely not. People have been deliberately hitting tape hard and over-saturating analog gear for ages... That's what decapitator's punish button does, it adds 20 dB of gain... That said if you do this to every single track in a mix you're just going to wind up with mud, and no transient definition.

If you have a hard time sticking to the rules the hornet VU meter's the best 5 euros you'll spend all month. Not only is it useful for getting over bad gain staging habits, it lets you gain stage an entire template in a single play-through.


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## tc9000 (Oct 1, 2020)

A great question and great replies here! Gain staging is something I keep coming back to to figure out a bit more...

Thinking about levels going into plugins, and hunting for that sweet spot, I like this section of a review where the reviewer uses an ITB signal generator and VU meter to assess how the PA Lindell 254E is calibrated. Its not rocket science but I never thought of doing this until I saw this:



Also, this:


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## el-bo (Oct 1, 2020)

jcrosby said:


> If you have a hard time sticking to the rules the hornet VU meter's the best 5 euros you'll spend all month. Not only is it useful for getting over bad gain staging habits, it lets you gain stage an entire template in a single play-through.





jcrosby said:


> Yes indeed. This is where gain staging makes all the difference when mixing ITB. Modeled plugins are calibrated to their real world, non-0 dBFS big brothers. So while in theory you can run a mix as hot as you want in a 32 bit float engine, the attitude that gain staging doesn't matter goes out the window as soon as you start to work with plugins that are calibrated to mimic 0VU calibration. There's about as much wisdom in that belief as saying if you want to write a melodic piece it doesn't matter if you play in a completely unrelated key. Sadly that misconception is a lot more pervasive than you'd think...
> 
> Does that mean you shouldn't deliberately run plugins hot if that's a particular sound you're after? Absolutely not. People have been deliberately hitting tape hard and over-saturating analog gear for ages... That's what decapitator's punish button does, it adds 20 dB of gain... That said if you do this to every single track in a mix you're just going to wind up with mud, and no transient definition.
> 
> If you have a hard time sticking to the rules the hornet VU meter's the best 5 euros you'll spend all month. Not only is it useful for getting over bad gain staging habits, it lets you gain stage an entire template in a single play-through.



There are a couple of plugins that do this GS thing right, though I’m sure there may be more. Both Ohmforce’s Ohmicide and NI’s Supercharger GT have a lit indicator that works hand-in-hand with the input trim. Like this, it guides the user towards the best level to be hitting the processor.

It’s a shame that more plugins don’t follow suit.

And thanks for the VU meter link. Seen it recommended on a few occasions recently, but never sprung for it. However, even though HS is not too difficult to grok, it still remains somewhat of a dark art. At this price, it even makes sense as a tutorial and checker, ti make sure everything is in it’s right place (...in it’s riiiight plaaace)


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## robgb (Oct 1, 2020)

Voider said:


> Gain staging is very important and it doesn't apply to analogue gear only, it's an universal thing regarding your mix.
> 
> Gain staging is basically to take care of your volume meters and your "volume budget", the purpose of gain staging is to make sure that you never exceed your volume budget to avoid distortion or clipping, and beyond that, give all of your instruments enough space to shine and sound good.
> 
> ...


Watch Kenny's video. Proof.


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## wst3 (Oct 1, 2020)

I hate to state the obvious, but at some point your signal is going to be analog. If you are recording a live source then gain staging is required in the analog stages, and the first stages in the DAW. The remaining stages will benefit from a proper gain structure, but yeah, you can cheat like crazy and probably get away with it.

Oh wait... if you signal is going to hit a D/A stage you need to worry about gain staging. A D/A converter has an analog half. And any analog equipment that follows your D/A conversion will also care about levels.


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## JEPA (Oct 1, 2020)

I'll leave this here...


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## Voider (Oct 1, 2020)

robgb said:


> Watch Kenny's video. Proof.



I've tried it and to my surprise I must admit, it works with real time VSTs, but if you record something, do render in place of your synths/vsts you'd still bake that distortion into the audio file. And I am not sure if it always works with real time VSTs as there are pre- and post gain knobs and different constellations could lead to different results.

Here are 4 beautiful piano notes (I'm kidding, before anyone believes that), I've rendered them in place while they were peaking above 0dB, now I've turned the channel fader of that audio down so that it peaks at -1.4dB on the master fader - you can still hear the distortion from the piano being rendered above 0dB.


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## robgb (Oct 1, 2020)

Voider said:


> I've tried it and to my surprise I must admit, it works with real time VSTs, but if you record something, do render in place of your synths/vsts you'd still bake that distortion into the audio file.


If you're rendering in place, the audio is still being drawn from the master bus, so I'm not sure why you would be baking in any distortion unless you're overloading the master.

As Kenny says, we're talking only about mixing, not recording. Recording input levels are still important.


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## ProfoundSilence (Oct 1, 2020)

I turn down a professional mix I like a little and then use that as my reference for what "loud" is supposed to be like. I always warm my ears up a little with that sound - so that when I'm adding things - they make sense volume wise. 

With orchestral instruments - gain staging should be done for the desired body/peaks during loud sections... gain staging each individual instrument to a certain level is absolutely silly, given a clarinet is not as loud as 3 trombones.


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## NoamL (Oct 1, 2020)

ProfoundSilence said:


> given a clarinet is not as loud as 3 trombones.



agreed, but they have the same "reference loudness"


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## SlHarder (Oct 1, 2020)

labornvain said:


> the positions of the faders become useful information. I know that if I have a fader pulled way down low, it's because it's a track that I want to be very quiet in the mix and not a track that is so hot that I have to have to lower the fader to compensate.



This.

The positions of the faders then represent artistic decisions you've made in the mix, not technically driven decisions.


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## vitocorleone123 (Oct 1, 2020)

Side note: TBProAudio mvmeter2 vumeter plugin is free - there's no technical reason to pay for a VU meter plugin for a couple years. It's just a matter of preference and money.


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## Kent (Oct 1, 2020)

NoamL said:


> agreed, but they have the same "reference loudness"


Yep. Set two mf clarinets to one mf horn, two mf horns to one mf trumpet, and you're in the Rimsky-Korsakov ballpark. 

(this is an oversimplification obviously please don't hate on it)


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## rudi (Oct 1, 2020)

There are several factors to take into account.

If you are recording an audio signal, staging is vitally important to ensure the best noise to signal ratio.

However, once the signal has been recorded, it is stored in your DAW as a series of numbers.

In the early days of digital audio for technical reasons (e.g memory, processor speed, maths operations etc.) the resolution was limited. From 8, to 12, to 16 bits. As computers became more powerful audio could be stored and processed at 24bits, and even 32bits or more.

Each increase in bit size gave you a greater range i.e more bits > more steps > more dynamic range. You jump from 255 steps and ~48dB at 8bits, to 4,294,967,296 steps and ~192dB at 32 bits - which is not far off the maximum loudness range ~210dB.

But no matter how much dynamic range you have you still need to be careful not to go over the maximum possible resolution. You also have to be careful when adding gain (multiply), reducing gain (divide), or mixing (adding) to avoid rounding errors and exceeding your maximum 0dbFS ceiling.

But what if you could you increase your audio resolution without the above pitfalls? The good news is that you can - by using some clever binary and maths manipulation.

We'll stick to 32 bits integers. Instead of storing an integer as a single value, you can divide the 32 invidual bits to represent different parts of a number, similar to scientific notation when dealing with very large of very small numbers. There are many ways of doing that, but one of them is by using a floating integer.

In our example we'll divide our 32 individual bits as follows:
- a sign for + or - (1 bit)
- an exponent (8 bits)
- a mantissa or floating point value (23 bits)

This gives you an incredibly large range of possible values. You jump from a range of 0 to 4,294,967,296 for a fixed 32 bits integer, to a range of ~1.2 x 10(-38) to ~3.4 x 10(38) values which is incredibly large! In terms of dB range it works out at ~1528dB!

It is also a signed range so you can go from hugely negative to hugely positive values. Even if your recorded signal hits its maxinum dB range in 32 bits integer ~192dB, you can still increase or decrease its gain by ~770dB of headroom!

Of course, when you output your end signal to an audio file, it will be scaled down to whatever format the file uses, but in the intervening stages in your DAW you won't have to worry about distortion.

Regarding FXs that have a "sweet spot", they are generally modeled after analogue pieces of equipment which distort if you overload them. If the internal resolution of the FX uses fixed integers, they will still distort the signal - but this is usually done by design and for aesthetic choices.

https://en.wikipedia.org/wiki/Audio_bit_depth
https://www.hdvideopro.com/columns/audio-assist/what-makes-32-bit-floating-point-audio-powerful/


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## NoamL (Oct 1, 2020)

kmaster said:


> Yep. Set two mf clarinets to one mf horn, two mf horns to one mf trumpet, and you're in the Rimsky-Korsakov ballpark.
> 
> (this is an oversimplification obviously please don't hate on it)




hmm sort of... I have no idea how to communicate the idea of "reference loudness"... it's like, if instrument X is averaging at X dB at mezzo forte then instruments W, Y, Z will all have a correct loudness at mezzo forte relative to that...


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## DS_Joost (Oct 1, 2020)

José Herring said:


> I've heard the term a lot and did some research a few years ago on gain staging. I ran across an old engineer trying to explain it and poor dude. It just seemed like he was trying to apply some old analog knowledge and hadn't really caught up to the digital age.
> 
> But, then I'm like but what if? What if it's important?
> 
> So what is gain staging? Is it worth learning and doing?



José, I know you are a Reason user, but this tip also applies to most other DAWs:


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## Snoobydoobydoo (Oct 1, 2020)

Sonimus has written an Article about Gain staging,
and their Plugin's Britson / Satson are an interesting approach.
*Sonimus, what is gain staging.*


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## jaketanner (Oct 1, 2020)

Been mixing professionally for well over 15 years...from analog to now ITB...here is what I have found. The plugins that emulate analog units..example: Acustica Audio's Acqua plugins...they are sample modeled...not algorithms but also also plugins that emulate analog like the BX console from Brainworx should also be gain staged to get the most from them...-18 is standard. Many plugins like Slate Digital's "trim" within their rack is calibrated at -18 = ZERO...so if that trim is at the end of a plugin chain, you want to make sure it is hovering around zero, so that whatever you put on after it, is properly calibrated.


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## Akarin (Oct 2, 2020)

berlin87 said:


> Hello Akarin,
> I bought VUMeter this summer, thanks for the tip in another thread..
> Do I place VUmeter on every track?
> 
> ...



Yes. As the first insert. I play the track and let the autogain work. Then, I adjust the balance with the volume faders.


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## Nicholas (Oct 2, 2020)

I've found that mixing became a lot easier for me when I started "gain-staging" everything correctly. Many VST-emulations of analogue gear "expect" a certain input level and sound best (subjectively of course) when you hit them within a specific range... Also, not having to pull your faders down so much gives you a lot more precise control over the mix, since it all tends to get fiddly with lower fader levels (naturally). So, since I try to pay attention to correct levelling before it hits the level-fader in the mixer, it all falls into place a lot easier for me.


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## Mattia Chiappa (Oct 2, 2020)

How does it apply to an orchestral template that has so many more instruments than the average song and can cover such a wider dynamic range? 

I've always balanced my template using the top layer as my ceiling, but most of the time my music tends to sit on mp with the occasional ff. That obviously almost always results in having a much lower than optimal output level. I haven't tested this myself but surely if you make all the tracks hit -6db max, the output must clip on a ff moment. On the the hand when you don't you do that it gets nearly impossible to average -18db on every track. Of course you can always reduce or the gain at the end using a gain on the out but if that's the case what's the better option? Or should one use multiple templates depending the overall dynamic level of the piece?


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## Ashermusic (Oct 2, 2020)

Nicholas said:


> I've found that mixing became a lot easier for me when I started "gain-staging" everything correctly. Many VST-emulations of analogue gear "expect" a certain input level and sound best (subjectively of course) when you hit them within a specific range... Also, not having to pull your faders down so much gives you a lot more precise control over the mix, since it all tends to get fiddly with lower fader levels (naturally). So, since I try to pay attention to correct levelling before it hits the level-fader in the mixer, it all falls into place a lot easier for me.



This. Also true with certain FX plugins.


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## SlHarder (Oct 2, 2020)

Akarin said:


> Yes. As the first insert. I play the track and let the autogain work.


Got VUmeter on the strength of your mention. Loaded a 55 track project which had taken quite some time to gain stage originally. Track by track finding the highest level and playing just that section a couple of times while watching meter and tweaking gain.

Printscreened the gains and reset to 0. Took 5 minutes or so to insert VuMeter on all the tracks, hit Auto and played the mix once all thru, then compared to my original gain staging. Most were within 1 or 2 db, a couple varied more but nothing off the charts. A couple were spot on. Plenty good enough to start mixing with.

So what was previously a long slog will now be easy peasy on my next large project. Well worth the US$7.34.


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## BassClef (Oct 2, 2020)

I use Logic and various orchestral and synth virtual instruments. Now for controlling track volume... could someone please explain the difference between:

1) using the virtual instrument's own volume control in a given track... and 
2) using Logic's "gain" plugin on that same track... and 
3) Using Logics mixer and the individual volume slider for that same track 

My normal work flow is:

1) leave all of Logic's mixer volume faders at "0"
2) record instrument track (by regions) via controller keyboard.
3) adjust track volume within the instrument (as in Kontakt or Play or SINE volume control)
4) either during or after recording, use cc1 and cc11 for dynamic contrast within each region.
5) sometimes when needing larger volume changes, I'll use Logic's gain plugin
6) after panning, EQ, reverb, other plugins, etc... make final balancing adjustments via Logic's mixer volume sliders. 
7) if needed (only after balancing with no clipping) add Logic's limiter plugin (set to -.5) to the final mix bus. I don't do this often, especially since much of my music has very soft passages that I do not want louder. 

Thanks


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## jaketanner (Oct 2, 2020)

Mattia Chiappa said:


> How does it apply to an orchestral template that has so many more instruments than the average song and can cover such a wider dynamic range?
> 
> I've always balanced my template using the top layer as my ceiling, but most of the time my music tends to sit on mp with the occasional ff. That obviously almost always results in having a much lower than optimal output level. I haven't tested this myself but surely if you make all the tracks hit -6db max, the output must clip on a ff moment. On the the hand when you don't you do that it gets nearly impossible to average -18db on every track. Of course you can always reduce or the gain at the end using a gain on the out but if that's the case what's the better option? Or should one use multiple templates depending the overall dynamic level of the piece?


Hey Mattia...As you know, orchestral music is far from being slammed like pop music...so you have nothing to worry about in terms of trying to get "hot" levels...From a mixing standpoint...it is utterly important to set your MASTER bus level to no more than 75%...so this works with any DAW, keep it barely hitting yellow. Your individual tracks can certainly peak red from time to time...there is plenty of headroom and it's normal. In the analog world, we used to hit red because it sounded better...LOL BUT the master bus was always reasonable. I would not worry about individual tracks, but rather your STEM tracks in terms of level...then that level into the master bus. 

The -18 comes into play when you need to feed analog type plugins. Most analog plugin emulations sound best when fed with -18 DBFS which is zero in analog, but good practice for digital as well. This is optimum for the plugin, NOT for your tracks. Your track output can certainly hit red when mixing. So as Akarin pointed out, use a VU meter as your first insert but set it to RMS...lower your SOURCE audio wave (clip), not the track level to hit the meter at -18. You can also use a trim plugin BEFORE the meter plugin to get your levels...this might be easier. Truthfully...what I do, is use one instance of a VU meter...once my level is set, I move it over to the next track...no need to keep the VU meter on every track once it's set, as this will not change unless you manually increase the audio clip. It is certainly a bit of a pain to do but worth it...again, this is ONLY during mixing once your tracks are rendered. If you don't render...you should. You will have much more flexibility to manipulate the audio than you do MIDI.

In terms of multiple templates: The principal of level still applies no matter what. The difference in dynamics becomes relevant in the mastering stage...how hard are you limiting. Orchestral/movie scores are very dynamic and have very little processing...trailer/hybrid tracks and scores like a Zimmer type or Goransson track can certainly be heavier on the processing in the mastering stage. But anything BEFORE mastering, should always fall within the proper gain staging (usually -14 at this point)...this is of course for best sonics.

If you have a very transparent limiting plugin like the (WEISS MM-1), then it should be easier to get your levels up without sacrificing dynamics.

Hope this makes some kind of sense and answers your question. Sorry for the winded reply...you have helped me with BBCSO and answered some of my questions, just wanted to return the favor.


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## storyteller (Oct 2, 2020)

Firmly in the "gain staging is still important" crowd. Most of the reasons have been covered in this thread regarding analog-modeled plugins. I did want to add one more point I didn't see mentioned yet (apologies if overlooked)...


While it is true that new 32bit interfaces can provide an insane amount of headroom and recovery potential, objectively, proper gain staging is still important to get the best sound out of the analog parts of the signal chain. You should view those 32bit interfaces as tools to help fix something that occurred with proper gain staging in place... like recording dialogue and then there is a louder-than-expected siren or explosion in the background that would normally distort the audio being recorded. Or... say a singer gets really explosive with a certain part of the song that was set properly during demo takes but would clip in a traditional 24bit signal flow. Use the 32bit capabilities to recover a potentially great take.


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## jaketanner (Oct 2, 2020)

SlHarder said:


> Got VUmeter on the strength of your mention


also look into Waves WLM Plus...this is much more comprehensive and very useful for your master bus as well.


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## jaketanner (Oct 2, 2020)

Wanted to add, that gain staging is important of course, but the recording of the audio signals is equally as important. So if you track your MIDI with the proper levels to begin with, then gain staging is far less of an issue.


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## Mattia Chiappa (Oct 3, 2020)

Thanks @jaketanner really appreciate it. I never render or hardly even mix my music tbh. I kinda like to do it once, when building a new template and then just make really minor adjustments. This thread though as been helpful as I'd like to learn more about these technical things.


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## Ben (Oct 3, 2020)

I gain-stage with the "Gain-fader" in Cubase (recordings; for most virtual instruments I simply mix via sub-mixes in VEP). You can open it by clicking on the MixConsole toolbar, click Racks and activate "Pre(Filters/Gain/Phase)". This adds the Pre-Section on top of the mix-rack.

I simply set the gain-level so the channel-meter is around -10dB during playback.
this makes it easier to mix with the faders, most plugins (analog-simulation, compressors) work with this input-level - including the presets, and I have enough headroom for the mixing channels without digital clipping/distortion.


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## jaketanner (Oct 3, 2020)

Mattia Chiappa said:


> Thanks @jaketanner really appreciate it. I never render or hardly even mix my music tbh. I kinda like to do it once, when building a new template and then just make really minor adjustments. This thread though as been helpful as I'd like to learn more about these technical things.


If you don't really mix, then don't sweat the levels really. As long as your master is not peaking constantly, you're good. A trick I like to use: GROUP all your stems, and lower them all evenly keeping the appropriate balance...this way your master bus is always at zero as it should be.


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## jadedsean (Nov 24, 2020)

I’m in the middle of building a new orchestral template and wondered if you guys use gain staging only in the mixing stage or for example you utilise VU meters on the tracks? So basically have your gain staging pre built into the template? 

My idea was to use VU meters on the busses of each section. If I use VU meters on every track on a large template my CPU would be choke, so that makes no sense to me. Is this correct? I find building a large template and balancing each section daunting enough without the added confusion of gain staging. 

I watched a video recently were a guy used a VU meter on all tracks, first the VU meter and I think it was set to -18. Then he applied the necessary processing, (most emulation plugins) and then added another VU meter at the end of the chain also set to -18. Is this correct? Seems quite confusing.

I’ll most likely use analog emulation plugins so I know I will have to feed them a certain amount volume and I won’t be using only VI’s so I’m know it’s necessary when recording instruments live. Can anyone make this clearly for me?


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## jaketanner (Nov 24, 2020)

skip the VU on every track. put it online on the buses going to the stems. If you need to feed an analog plugin, then put the meter before the plugin, but then just shift the plugin to another track when you need it...unless you are tracking your entire template at the same time, you don't need a VU on every track. Also, doesn't your DAW have various different meters for the tracks? I work in Pro Tools, and we have like 14 different variations of meters...LOL But I still use the Waves WLM plus to see LUFS...but check or set your DAWs meters first.


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## jadedsean (Nov 25, 2020)

jaketanner said:


> skip the VU on every track. put it online on the buses going to the stems. If you need to feed an analog plugin, then put the meter before the plugin, but then just shift the plugin to another track when you need it...unless you are tracking your entire template at the same time, you don't need a VU on every track. Also, doesn't your DAW have various different meters for the tracks? I work in Pro Tools, and we have like 14 different variations of meters...LOL But I still use the Waves WLM plus to see LUFS...but check or set your DAWs meters first.



Hey Jake thanks for the advice, however I am a little confused when you mentioned to shift the VU meter to another track after using it. Do you mean I put the VU meter first then send in the required volume to the plug-in and when I achieved this take the same VU meter and apply to the next track? 

Would this not defeat the purpose as once I take the VU meter off the track the volume will automatically change? Also do you set up a template with gain staging or only while mixing?


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## jaketanner (Nov 25, 2020)

jadedsean said:


> Hey Jake thanks for the advice, however I am a little confused when you mentioned to shift the VU meter to another track after using it. Do you mean I put the VU meter first then send in the required volume to the plug-in and when I achieved this take the same VU meter and apply to the next track?
> 
> Would this not defeat the purpose as once I take the VU meter off the track the volume will automatically change? Also do you set up a template with gain staging or only while mixing?


Hi. The meter plug-in doesn’t do anything but show you how loud the signal is. If you’re talking about placing a TRIM as your first plug-in that’s different. But a VU meter plug-in does not change the signal at all. So you would use the output of whatever is feeding your stem mixes to set the level. So after you have it set, you can move the VU. I do not have the individual tracks gainstaged. That is pointless because it’s the stem tracks or audio you use to mix that needs to be set for the plugins. At least for me...I mix professionally, and do not gain stage instrument tracks until I’m ready to mix audio. No real right or wrong way, just how I do it. But feel free to love the VU once the level is set.


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## Andrew Souter (Nov 25, 2020)

Most DAWs have 64-bit Floating Point signal paths these days. IF (be careful of that word -- it's a tricky one) you are ONLY usiing the DAW mixer gains and staying inside the computer you could:

* Add an audio file to a track that already peaks at 0dB, somehow gain boost it +100dB 
* Send it a group, boost that group +100dB
* Send it to the master and reduce the master by 200dB

...and there will be zero distortion. ZERO. The track is "clipping" by 100dB, and the group is "clipping" by 200dB according to your meters (not really true, but your meter will say so), but zero disotrion.

If you are working with a (*LINEAR*) plug-in that processes in float (most do), and even better, one that keeps the signal path double precision 64-bit float intnerally you can insert 1, or 10, or 100 etc plugs into the above track/group and the result is still fine. Try it with our Breeze or Precedence plugs. No problem. The result will be identical to keeping gains at 0dB.

If all processing is done in float (and ideally 64-bit float), and is *linear*, and the MASTER does not go over 0 gain staging is basically irrelevant.

Here is when it becomes relevent:

1) If any plug-in in the chain if processing in fixed point. (Some Waves plug-ins used to. Not sure recently. There are likely others, but they are rare these days. These can/will clip if signal exceeds 0dB)

2) Any plug-in in the chain uses any kind of NON-Linear processing. i.e. wave-shaping, distortion, saturation, etc. (anything "analog modeled" likely falls into this group.) eg: the result of f(x) = tanh( g * x) is highly dependant on the value of g (gain). 

in other words in a LINEAR process such as simple gain:

w = g * x;

can be undone with another (inverse) gain:

y = (1/ g) * w
y = (g /g) * x = 1 * x = x
y = x // the same!

but a non-linear process can not be undone via a later gain:

w = tanh(g * x)
y = (1/ g) * w
y = tanh(g * x) / g
x != tanh(g * x) / g
y != x // NOT the same!

So for any non-linear transfer-function based process the result will depend on the gain and various analog models use such things and have expected ranges of optimal signal levels.

Note however, most plugs that have any kind of non-linear processing have input gain (or "Drive") and an output gain specially to allow you to control exactly what levels are hitting the non-linear part of the process.

3) You are integrating real-world physical processors that are going through fixed-point/integer DA/AD conversion.

4) You are integrating real-world physical processors that are connected digitally and the signal is 24bit (or any bit-depth really) FIXED-point/integer, such as standard AES/spdif/toslink etc.


The various recommendations to keep tracks peaking at around -12dB or similar is because of these 4 considerations. Gain staging in this manner is most important if/when these apply to your workflow and tools.


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## Andrew Souter (Nov 25, 2020)

robgb said:


> If you're rendering in place, the audio is still being drawn from the master bus, so I'm not sure why you would be baking in any distortion unless you're overloading the master.
> 
> As Kenny says, we're talking only about mixing, not recording. Recording input levels are still important.




... it is possible his DAW is writing 24-bit fixed/integer files for render-in place. If you clip a fixed-point file it is exceptionally hard to undo it. There are some "unclip" dsp products/processes, but they are not perfect. For sure simple gain adjustment post clip will not do anything about the distortion from clipping. So I guess his temp files are fixed point. What DAW is it? Prob there is a preference to use float files for temp files somewhere in the DAW. Use that, and then no problem.

Additionally, note you can "psuedo clip" even the master out of the daw if you are rendering 32bit or 64-bit float, and then gain reduce it somewhere else before it hits the DA converter. The float file itself has no problem at all having levels over abs(1.0) i.e. 0dB. 

I.e. make your master peak at +12dB. Render a 32-bit float file. Load it in iTunes and turn the gain town to 25% or less. Should be fine. Magic. 

I am NOT suggesting to do that. Just trying to point out that it is important to realize the relevence of float vs fixed point files in these topics.


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## MarcusD (Nov 25, 2020)

Actually gain staging is just useful in the digital world when it comes to mixing. 

When the faders are at unity (and you've set the gain for each track to an approximation for each channel) you'll have more precise control over fine volume adjustments when increasing or decresing the fader volume by fractions of a db.


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## jadedsean (Nov 25, 2020)

jaketanner said:


> Hi. The meter plug-in doesn’t do anything but show you how loud the signal is. If you’re talking about placing a TRIM as your first plug-in that’s different. But a VU meter plug-in does not change the signal at all. So you would use the output of whatever is feeding your stem mixes to set the level. So after you have it set, you can move the VU. I do not have the individual tracks gainstaged. That is pointless because it’s the stem tracks or audio you use to mix that needs to be set for the plugins. At least for me...I mix professionally, and do not gain stage instrument tracks until I’m ready to mix audio. No real right or wrong way, just how I do it. But feel free to love the VU once the level is set.





jaketanner said:


> Hi. The meter plug-in doesn’t do anything but show you how loud the signal is. If you’re talking about placing a TRIM as your first plug-in that’s different. But a VU meter plug-in does not change the signal at all. So you would use the output of whatever is feeding your stem mixes to set the level. So after you have it set, you can move the VU. I do not have the individual tracks gainstaged. That is pointless because it’s the stem tracks or audio you use to mix that needs to be set for the plugins. At least for me...I mix professionally, and do not gain stage instrument tracks until I’m ready to mix audio. No real right or wrong way, just how I do it. But feel free to love the VU once the level is set.



Sorry Jake I should have been clearer, yes my VU plugin also has a trim option, I was under the impression if the track volume exceeded 0db I can then use the trim option. I seen people actually set a trim on tracks at -18 so they have plenty headroom when the tracks hit the master. I think this is used so we don’t actually move any faders within our mixer, maybe it’s comparable to clip gain. Sorry im a newbie when it comes to mixing. Am I completely confused here?


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## jadedsean (Nov 25, 2020)

Andrew Souter said:


> Most DAWs have 64-bit Floating Point signal paths these days. IF (be careful of that word -- it's a tricky one) you are ONLY usiing the DAW mixer gains and staying inside the computer you could:
> 
> * Add an audio file to a track that already peaks at 0dB, somehow gain boost it +100dB
> * Send it a group, boost that group +100dB
> ...



Thanks Andrew for the detailed explanation, I’m new to all this so might take a while to get up to speed with the lingo but I will research all the info you have supplied.


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## SlHarder (Nov 25, 2020)

Laborvain said:
the positions of the faders become useful information. I know that if I have a fader pulled way down low, it's because it's a track that I want to be very quiet in the mix and not a track that is so hot that I have to have to lower the fader to compensate.





SlHarder said:


> This.
> 
> The positions of the faders then represent artistic decisions you've made in the mix, not technically driven decisions.



Yesterday I returned to a 60 track project that I hadn't opened in 6 months. Because I had used gain staging I could look thru the fader positions and easily remember the overall mix sound that was my goal. The fader positions told a story that I could easily understand.

Imo gain staging keeps all the techie issues from messing up a visual representation of the acoustic story you are trying to tell.


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## jaketanner (Nov 25, 2020)

jadedsean said:


> Sorry Jake I should have been clearer, yes my VU plugin also has a trim option, I was under the impression if the track volume exceeded 0db I can then use the trim option. I seen people actually set a trim on tracks at -18 so they have plenty headroom when the tracks hit the master. I think this is used so we don’t actually move any faders within our mixer, maybe it’s comparable to clip gain. Sorry im a newbie when it comes to mixing. Am I completely confused here?


Maybe I misunderstood what you are trying to gain stage...if you are thinking that gain staging the individual VSTis will do anything, it won't...this is a waste of time. The gain staging needs to occur in two spots...when feeding a plugin (more so with true analog emulations), and when you have audio that's been recorded hotter than -18 DBFS also when monitoring your master bus...but this really isn't gain staging as much as it is achieving proper levels. The FINAL mix, should also not exceed a high level because then the mastering engineer will not have any headroom to add more processing (even though they can simply clip gain down...this is not a desired approach though). With the already recorded audio, the VU meter just needs to be present to get the proper level...use clip gain in this case. Then you can move the VU meter to the next channel and so on. Hope this is making sense..LOL 

The MOST important time to worry is during mixing though...and in the end, if it sounds good, it's good...if you are -14 instead of -18 and it sounds better at -14 then leave it. I wouldn't go too nuts.

Also...some plugins DO have input controls...this can be used as well to monitor the incoming signal. I often use the Slatedigital VMR, that has the TRIMMER tool.


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## tim727 (Jan 7, 2022)

I read through this whole thread and honestly still find myself *hopelessly* confused. I'm a hobbyist composer and mixing is far from an area of expertise for me. I did read once a few years ago that tracks should be kept peaking around -12 dB. I have to admit I didn't fully understand the technical reasons but knew it had something to do with avoiding clipping and so started adopting the process. 

I use one Kontakt instance for every articulation of every instrument in my orchestral template. Every time I add a new patch/articulation to my template, I lower the volume level to around -12 dB in the Kontakt instance. After reading through this whole thread I'm still having trouble understanding if what I'm doing is good practice ... or if it's not needed at all. My life would be a lot easier if it were not needed at all to be honest since having to lower the volume level every time I add anything to the template makes it a much bigger pain to add new libs to my template. 

Could someone be so kind as to answer a few simple questions for me:

(1) Should I keep lowering every individual Kontakt instance to -12 dB (or some other value)?
(2) Is there any appreciable difference in directly lowering the Kontakt instance to -12 dB vs lowering the fader (in the Cubase mix console) for the track to -12 dB vs lowering the gain on the track by 12 dB in the pre-rack in the mix console?
(3) Let's say I have 6 different tracks, each containing a different horn articulation, and that all these tracks route to a single horn group track. Instead of changing the volume to -12 dB on all the individual tracks, can I leave all those at 0 dB and just change the volume on the group track to -12 dB?

Thanks for any help. I find this to be a very confusing topic so any input from those in the know would be greatly appreciated!


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## liquidlino (Jan 7, 2022)

Whilst in theory you don't need to "gain stage" any more, in practice you do, for two reasons in my view:

1. Many "analog" plugins expect -18db signal, if you go higher, then they start to get non-linear.
2. Many presets on insert plugins are built for -18db.

So when it comes to mixing time, first thing I do is pick the thing that I'm going to hinge the whole mix around (e.g. kick drum), and get it to -18db on the VU meter, and go from there, gradually bringing in the extra parts. And same for group busses, bring them back down to -18db before putting buss inserts on, and then again at master, I'll use a gain insert to get it to -18db before starts going into the master inserts. Definitely makes like easier not having to adjust input gain / preset settings etc all the time.


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## tim727 (Jan 7, 2022)

liquidlino said:


> Whilst in theory you don't need to "gain stage" any more, in practice you do, for two reasons in my view:
> 
> 1. Many "analog" plugins expect -18db signal, if you go higher, then they start to get non-linear.
> 2. Many presets on insert plugins are built for -18db.
> ...


Forgive me if this is a stupid question, but when you say that some presets on plugins are "built for -18 dB" what does that actually mean in practice?


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## liquidlino (Jan 7, 2022)

tim727 said:


> Forgive me if this is a stupid question, but when you say that some presets on plugins are "built for -18 dB" what does that actually mean in practice?


Just that I've found preset settings seem to be built around a -18 input, eg compressor thresholds. Usually only requires adjusting the input gain control or the threshold if it has one though. But some don't. It's a minor thing, I mean you're going to end up changing settings anyway. Point 1 about analog inserts is much more noticeable.


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## waveheavy (Jan 20, 2022)

Gain staging is just as important in the digital world as it was in the analog world. The main difference between digital and analog is there's no tape 'hiss' problem in digital. But setting the right levels is still important, because even when mixing at 24 bits, if you don't use as much of those bits as you can, your result will be inferior.

See this article by Waves that explains this. (In their plugin manual for each plugin, they have a gain recommendation for the sweet spot processing of that plugin.)









Gain Staging in Your DAW: Good Levels, Better Mix | Waves


Getting a clear, punchy mix is not just about processing and effects – it starts with setting the right gain levels at which you track and mix. Get tips on how to achieve proper gain staging – including how a good, simple VU meter can help you.




www.waves.com


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## from_theashes (Jan 21, 2022)

tim727 said:


> I read through this whole thread and honestly still find myself *hopelessly* confused. I'm a hobbyist composer and mixing is far from an area of expertise for me. I did read once a few years ago that tracks should be kept peaking around -12 dB. I have to admit I didn't fully understand the technical reasons but knew it had something to do with avoiding clipping and so started adopting the process.
> 
> I use one Kontakt instance for every articulation of every instrument in my orchestral template. Every time I add a new patch/articulation to my template, I lower the volume level to around -12 dB in the Kontakt instance. After reading through this whole thread I'm still having trouble understanding if what I'm doing is good practice ... or if it's not needed at all. My life would be a lot easier if it were not needed at all to be honest since having to lower the volume level every time I add anything to the template makes it a much bigger pain to add new libs to my template.
> 
> ...


It doesn’t matter where you set your volumes (Kontakt level vs mixer level)... you can even group tracks and lower the level of that group.
And it doesn’t need to be -18db on every track as long as your whole mix isn’t over -18db.
I use a simple way: I have a VU-Meter on my master bus when mixing, before all my mixing-plugins. That way I can ensure, that I don’t go over -18db before hitting some emulations of analog gear (consoles, compressors, etc).


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## gohrev (Jan 21, 2022)

from_theashes said:


> It doesn’t matter where you set your volumes (Kontakt level vs mixer level)... you can even group tracks and lower the level of that group.
> And it doesn’t need to be -18db on every track as long as your whole mix isn’t over -18db.
> I use a simple way: I have a VU-Meter on my master bus when mixing, before all my mixing-plugins. That way I can ensure, that I don’t go over -18db before hitting some emulations of analog gear (consoles, compressors, etc).


Hi! I have two questions to follow up:

I thought preventing your tracks from going above -12dB was "the new standard" for digital?
If your entire mix sits at -18dB max, what would the max volume be for your master? -18db, too?
I struggle with understanding what dB-level my finished product should be at..


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## waveheavy (Jan 21, 2022)

gohrev said:


> Hi! I have two questions to follow up:
> 
> I thought preventing your tracks from going above -12dB was "the new standard" for digital?
> If your entire mix sits at -18dB max, what would the max volume be for your master? -18db, too?
> I struggle with understanding what dB-level my finished product should be at..


dBFS is what your Peak meter in your DAW is reading usually by default. RMS is a average loudness, and often what digital studios use to set the max loudness of a mastered track.

Some of the loudest songs are in the Pop, Heavy Metal, and Dance genres. Lot of those usually hit around -8dBRMS. Other styles like Rock, Blues, Country, Folk, etc., hit around -14 to -10dBRMS, Jazz and Classical being the quietest.

You can load songs off a CD into your DAW and put an RMS meter on it and see the song's max RMS, and this way you can get a general idea of what level it was mastered to. Can't really do that with a downloaded streamed song, because the song website has its own standards for loudness using the LUFS measure. To upload a song on their sites, they have recommended LUFS levels, otherwise they'll change it for you.

One should record into the DAW around -18dBFS, but mixing is different. Always mix/master at 24 bit minimum. That's a studio standard. 48kHz/24 bit or higher for cinema/DVD is the standard.

What I was taught by a pro in Nashville:
. try to get all track faders in a sweet spot as to loudness, leaving room to raise the fader. Generally, this will mean your faders running in the yellow. In the old analog days, each channel had a level pot where you could adjust the fader range, putting all the faders in the best control range.

. shoot for a max of -6dBFS peak on the Master bus at the hottest part of the song. (Classical or Jazz might be even lower). The idea is to leave headroom for the later mastering stage.

. OK to put a Limiter on the Master bus when mixing, set low to like -1 or -2. But bypass it occasionally during the mix to see if any "overs" are happening. If so, find the track that's causing most of it and fix it, or group all the faders and pull them all down together until no more 'overs'. Do this at hottest part of the song. If you shoot for a -6dBFS Peak level on your Master bus, you shouldn't have this problem.

. most plugins have a sweet spot range they work best at recommended by the manufacturer. Read the manuals. Waves plugins usually have a Peak meter on the out that turns red for "overs".

. do a rough mix first, trying to get a good sounding balance between the tracks, no EQ, no Compressors, no Reverb, no plugins at this stage. Do only Volume and Pan. This is your initial mix level that you want to stay around.

. When adding EQ, or plugins, note the start output level of the track. If you just slap a plugin on a track and don't gain stage, you'll lose that rough mix balance you worked so hard to get at the first. The way to treat this problem is to note the beginning volume level of your track before the plugin, and 'match' that level when you're finished with the treatment. This is one of the main uses for your A/B button on the plugins. It's also the best way to actually hear what the treatment did to the audio.


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## KEM (Jan 21, 2022)

Gain staging is like 75% of the mix in my opinion


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## gohrev (Jan 22, 2022)

waveheavy said:


> dBFS is what your Peak meter in your DAW is reading usually by default. RMS is a average loudness, and often what digital studios use to set the max loudness of a mastered track.
> 
> _(respectfully snipped)_
> 
> . When adding EQ, or plugins, note the start output level of the track. If you just slap a plugin on a track and don't gain stage, you'll lose that rough mix balance you worked so hard to get at the first. The way to treat this problem is to note the beginning volume level of your track before the plugin, and 'match' that level when you're finished with the treatment. This is one of the main uses for your A/B button on the plugins. It's also the best way to actually hear what the treatment did to the audio.


THANK YOU very much, @waveheavy for taking the time to write a post that I immediately bookmarked for future reference.

The bit about doing a rough mix first is something I'll change immediately in what I call my "midi template" in Cubase — the template where I compose and construct my work in. Some of the instruments and instrument groups already have plugins and what have you not slapped on them. I should save this for the mixing stage (where I export all VST instrument group tracks to audio stems).


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## waveheavy (Jan 22, 2022)

gohrev said:


> THANK YOU very much, @waveheavy for taking the time to write a post that I immediately bookmarked for future reference.
> 
> The bit about doing a rough mix first is something I'll change immediately in what I call my "midi template" in Cubase — the template where I compose and construct my work in. Some of the instruments and instrument groups already have plugins and what have you not slapped on them. I should save this for the mixing stage (where I export all VST instrument group tracks to audio stems).


You're welcome for any help I was able to give.


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## Tod (Jan 24, 2022)

I didn't read the whole thread so maybe this was covered, but I didn't see anybody talk about using their ears. Gain staging is fine, but it won't get you to that ultimate mix without using your ears. Whether it's orchestra, rock, or anything in between, your ears are the most important tool you have. I came from the old tape days so I know, we pushed things to the max for only one reason, S/N.

But in these times that's not necessary. However, I can understand if you're new to all this, you probably need or want someplace to start. And after you've started, where do you go from there.

For me, the first thing is to group my instruments, each section has to sound good on it's own. Of course it's not that simple but I think you know what I mean. Each section will be sent to a sub-bus for mixing in later. 

For me I don't even think about gain staging, however without thinking about it, I will probably start gain staging somewhere along the line in order to keep from hitting FX or whatever to hard. The thing is, if I've got all my sub-buses together in a meaningful way, so the mixing becomes easier. I might end up bringing down a sub-bus or 2, or maybe all of them to get a better handle on the output. 

So I'm probably gain staging, but I don't think about it.


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## gohrev (Jan 24, 2022)

Tod said:


> I didn't read the whole thread so maybe this was covered, but I didn't see anybody talk about using their ears.
> 
> _(respectfully snipped)_
> 
> So I'm probably gain staging, but I don't think about it.


Tod, how do you balance your instruments within one group, say woodwinds? Would you adjust the output in the instrument itself (e.g. Kontakt), via Pre-Gain or using the channel's fader?


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## wst3 (Jan 24, 2022)

Gain staging is simply a convoluted term for optimizing, and optimizing is essential for analog or digital audio. Many of the previous posts have great info. I am replying because I did not see one that summarized all the factors. I will try. (I may not succeed.)

In the analog domain there was a noise floor that you wanted to bury as far as you could, and distortion from exceeding the capabilities of the power supply, which you wanted to avoid.

In the digital domain the noise floor is the noise in the audio itself, the digital path will not contribute noise. This is a big plus, which comes with an equally big minus. 

Exceeding the clipping point in an analog system could be inaudible for the first few dB, and then it could be thought to add some character for the next few, and then it becomes a fuzz box.

Exceeding the clipping point in a digital system means exceeding 0 dBFS, in other words it is no longer possible to capture the level. Again if you exceed that threshold for a very brief time only the goldenest of ears will detect it. If you exceed the threshold by only a couple of dB it may also be tough to hear.

There is more to gain staging but that's the important part. And the only way you can reliably keep the signal in the sweet spot is to have your analog monitoring aligned with your DAW. Otherwise you have no idea what is going on.

To do that it helps to understand the difference between peak, peak-to-peak, average, and RMS measurements. Helps, it is not essential. I still think you should dig into it, but that's mostly my curiosity<G>.

So, pick a reference level, -12 dBFS works, -18 dBFS works, for some material it is possible that -6 dBFS would work, not sure I'd want to walk that tightrope though.

Generate or play back a sine wave at 1004 Hz and set the output level of your DAW to drive the analog system to +4 dBu or -10 dBV (sorry, that's a mess that our ancestors created). These are both reference levels in the analog domain.

The way I do it is I feed a +4 dBu sine wave into my monitors, and set their level adjustments such that the SPL measured at my listening spot is 85 dBSPL (I wear hearing protection while doing that<G>). Next I playback that digital reference level and if all is good in the world I should measure -18 dBFS on the DAW output meters, and +4 dBu at the output of my D/A converter, and 85 dBSPL at the listening position. Everything is lined up, I know that if I end up sending 0 dBFS that is +22 dBu (most pro gear handles that easily), and well, ear bleed loud at the listening position (102 dBSPL if you are curious, it will hurt!)

There is, of course, no one that enjoys listening to single frequency sine waves. Music is far more dynamic, with peaks that can exceed the reference level by a lot (hence the -18 dBFS reference). The point being if the RMS level of your track sits around -18 dBFS you'll be monitoring at a reasonable level, there will be no clipping, and you are well above the noise floor. All is good in the universe. There will be peaks that approach, or exceed, 0 dBFS, (or +22 dBu) but they short enough that they should not cause problems. 

If they do then you adjust your mix so that they don't, and since you have aligned everything the results will be reproducible, and reliable.

Old farts like me remember a time when it did matter where the fader was set, heck, it mattered where every control was set. This was the result of non-linearities in the circuits, and event the components. We also had to worry about the reference level we used when we aligned the tape deck.

Digital audio has eliminated most of those concerns. That doesn't mean it is not a good idea to understand them.

Sorry for the length of the post, I just could not edit out any more (you should have seen the first draft<G>)


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## Tod (Jan 24, 2022)

gohrev said:


> Tod, how do you balance your instruments within one group, say woodwinds? Would you adjust the output in the instrument itself (e.g. Kontakt), via Pre-Gain or using the channel's fader?


The first thing I do is get each instrument in Kontakt setup to output a decent level, usually around -0.0dB. I can usually do that with Instrument Options where I can set CC7 to 1 of 5 stages each being 6dB apart from a -12dB to a +12dB. If you don't know where that is, open the instrument with the wrench, then open Instrument Options up near the top left. Click on "Controller" and you'll see it. Some instruments are set so that you can't open them, at which point I just use what's available.

The important thing for me is that I have enough output volume so that I have good control in the midi editor with the various controller envelopes (CC7, CC11, CC1 etc.), because that's where I make nearly all my adjustments of the individual instruments to get a good relationship between them. 

If I have to make volume adjustments in Reaper (my DAW) in order to get them set up, then no problem. 

So that's pretty much it goherv, the whole purpose for the CC controllers is for not only getting a good sound on the instrument itself but to also get the right balance between them.


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## gohrev (Jan 25, 2022)

Tod said:


> (respectfully snipped)
> 
> So that's pretty much it goherv, the whole purpose for the CC controllers is for not only getting a good sound on the instrument itself but to also get the right balance between them.


Thank you. I am struggling a bit with making sure instruments deliver a healthy output of around 0 VU, but then I end up with a Flute in low register as loud as a Frenc horn! So I figured while it's good to have that "pure output" of 0 VU, there's still some balancing to be done right after.


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## Tod (Jan 25, 2022)

gohrev said:


> Thank you. I am struggling a bit with making sure instruments deliver a healthy output of around 0 VU, but then I end up with a Flute in low register as loud as a Frenc horn! So I figured while it's good to have that "pure output" of 0 VU, there's still some balancing to be done right after.


Yeah, make the adjustments where they need to be. For me the most important thing is to get levels set so that I have complete control with all the CC controllers, that's where the magic happens.


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