# Gain staging templates and quiet samples



## Mrmonkey (May 16, 2021)

Been setting up a new template in logic and based on the Thinkspace Education template lesson it wants hit -18 db for each track with expression set to max, and dynamics at 64. So basically - everything in the middle should hover around there with more headroom for max dynamics.

Sounds great but some samples, in particular Spitfire ones, seem to come up way waaaaay short of that. I get they don’t normalise their samples. The suggestion was to add a simple gain insert on the track rather than try changing the internal Kontakt or instrument volume. Is that really the best way to go? It just seems strange having to add 10db gain to a sample just to hit -18 on the track rather than trying to do it in the instrument.
And second is -18 a reasonable sweetspot?
Also, is it really a good idea to balance to an average value or with different libraries all over the place would it be better to balance to their maximum? And finally, does anyone actually do this? Or do you just sort out volume differences as they arise?


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## Trash Panda (May 16, 2021)

I use Hornet’s The Normalizer or VU Meter for gain staging and it works perfectly every time for me.


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## Mrmonkey (May 16, 2021)

I’m using the stock logic level meter and then I used an SoundMeter X to balance -20 to around 67db from monitors.

trouble I have is that say Hz strings violins legato at 64 modulation comes out at about -24, and the BBCSO harp comes out at freaking -35 at 64 velocity. Seems freakishly quiet based on the need to get to -18


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## Chris Harper (May 16, 2021)

I usually just use a gain insert for the sake of convenience. It’s much faster than opening the plugin and remembering where the volume sliders are for each different plugin. It’s also nice to have a separate knob to make overall adjustments to the mix without changing CC7 or fader automation that has already been added. If I used a universal template, it would definitely be worth the time to balance everything. I think I would use average volume and leave headroom for when you want something to stand out. If you use CC7 a lot then you might want to avoid changing the instrument volumes in Kontakt too drastically because CC7 will change those, I believe.

-18 dB is a good starting point to shoot for, but it’s not a rule that absolutely must be followed. I sometimes set gain slightly higher or lower if I know something will be louder or quieter in the mix. I like to have my faders close to zero because the adjustments are more sensitive. I have heard that some plugins, especially those that emulate analog hardware, work better at -18, but that is definitely not universal. For stock DAW plugins I don’t think it matters as long as it doesn’t go off the default scale of the meters.


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## jcrosby (May 17, 2021)

The question is peak or RMS... Big difference between -18 peak and -18 RMS. Analog plugins that are modeling real life gain staging are modeled for either -18 or -20-ish RMS (Depending on the device's spec this was its equivalent to unity gain - i.e. 1 volt in = 1 volt out; what was considered optimum fidelity). So when people say -18 and you really want to know which one they're referring to...

As far as the post above... Generally it's true. DAW plugins typically process internally with tons of headroom. 32 bit float for example has hundreds and hundreds of dB. However... Anything that creates any kind of distortion (i.e your from your DAWS generic distortion plugin, to a plugin like Izotope trash, to an amp or pedal plugin, etc) Should be gain staged properly. Distortion is level dependent, modeled or not.

The perfect and easiest to understand example is a guitar amp... If you want a clean sound and you overload the signal you won't get clean... Distortion, be it real world or ITB is level dependent. You can also test this on any distortion plugin... The hotter the signal the crunchier it gets, even if it's a waveshaper...

So while in general it _doesn't matter_, it actually kind of does as if you're unaware of how level affects any kind of plugin that create any kind of distortion, analog modeled or not, you're overlooking a scenario that can have a huge impact on the tone you get out of a plugin...

So basically good gain staging is just good common sense for avoiding scenarios that don't just revolve around _analog modeling_, but also impact anything from FF Saturn to Cubase's built-in multiband distortion. Not to mention this leaves your dynamics plugins more headroom for you to dial things in as little or as much as you want...


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## Voider (May 17, 2021)

Mrmonkey said:


> And second is -18 a reasonable sweetspot?
> Also, is it really a good idea to balance to an average value or with different libraries all over the place would it be better to balance to their maximum? And finally, does anyone actually do this? Or do you just sort out volume differences as they arise?


Usually you want to have different volumes for different instruments so it's a bit uncommon to make sure everything hits the same volume, you should rather make sure that you just get -8dB headroom on the master channel for the mastering session after your track is done. Those 8dB headroom ensure that you (or a mastering engineer) have enough space to add plugins that come with a gain in volume and / or doing the final polish.


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## Tralen (May 17, 2021)

I think it is best to do the basic gain staging at the source, otherwise you will get vastly different fader positions and behaviour in the DAW mixer. Handling one fader at -18dB and then the next one at 0B is totally unintuitive, the faders will be scaled differently.

For Kontakt libraries, I use Kontakt's master volume to get them close relative to the rest.


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## Per Boysen (May 17, 2021)

I too prefer to do the gain staging at the source. I run all Kontakt instances in Vienna Ensemble Pro and enjoy how much time it saves me to have audio coming into the DAW at a "decently mixable" level. I also do some EQ in VEP and this makes my Vienna template get better with each new project I use it in. Having a cultivated sound coming into your DAW also makes all dynamic plugins work optimally.


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## Tralen (May 17, 2021)

I forgot to mention gain automation.

If the faders are too different at the DAW level, it becomes difficult to copy automation from one track to another, or even to judge what the automation is doing at a glance.


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## youngpokie (May 17, 2021)

Mrmonkey said:


> Thinkspace Education template lesson it wants hit -18 db for each track with expression set to max, and dynamics at 64. So basically - everything in the middle should hover around there with more headroom for max dynamics.


But this lesson is not meant for orchestral instruments, correct? 

I am trying to wrap my head around for example a solo flute in mid low register and trombones a3, each set to -18db. This is so crazy, so wondering what I missed here....


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## Tralen (May 17, 2021)

youngpokie said:


> But this lesson is not meant for orchestral instruments, correct?
> 
> I am trying to wrap my head around for example a solo flute in mid low register and trombones a3, each set to -18db. This is so crazy, so wondering what I missed here....


This comes from the recording world. -18dBFS ~= 0vu, which is the alignment level for analogue equipment. This is the ideal spot for the signal to be processed, in particular, for non-linear processing like saturation.

If you are not doing any processing to your VIs at the channels, you can simply use the DAW faders to balance the instruments according to your expectations (Tuba louder than Flute), just make sure you leave some headroom for mastering; -6dBFS for peaks at the master is enough.

If you do want to use plugins that expect that alignment level at the channels (if the plugin doesn't provide input/output control) use a gain plugin after the VI (and before the new plugin) to set the level to about -18dBFS (use the gain plugin meters), then use the DAW faders to balance the output of the channel according to your expectations.


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## NoamL (May 17, 2021)

Mrmonkey said:


> based on the Thinkspace Education template lesson it wants hit -18 db for each track with expression set to max, and dynamics at 64


I think this is really bad advice actually. A "rule of thumb" run amok.

There are two golden principles you should follow when balancing your template

1. The _orchestra as a summed sound_ needs to have lots of headroom left in the meter even when it's playing fortissimo. This preserves the dynamic range for your score mixer. There's no point having the music use up all the volume because the dubstage is just gonna tuck it down under dialog and sfx anyway.

2. _The orchestra needs to be balanced_. I would never set the volume of an individual instrument based on what some meter says, I would set it to the volume that sounds realistic next to the other orchestral instruments.

The best way to balance your template is to buy a John Williams orchestral suite conductor's score, mock up all the instruments and figure out where you need to place each instrument so that the total sounds realistic and so that the summed sound is well short of peaking.

Be aware that orchestral instruments balance totally differently at loud and quiet dynamics, that's why a complete piece of concert music is a good reference because it will put your instruments through their paces and let you test the orchestral balance at lots of different dynamics.

The best way to set your volumes is with CC7 not the DAW fader. If you put a little MIDI region at the beginning of each instrument track in your template you can set up the CC messages you want each instrument to automatically obey.

There may be an "ideal volume level for plugin processing" with regards to synths or guitars or whatever, but the orchestra in real life cannot possibly deliver a consistent level. Orchestras are hugely dynamic, way more dynamic than pop music or electronic music.


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## Tralen (May 17, 2021)

NoamL said:


> I think this is really bad advice actually. A "rule of thumb" run amok.
> 
> There are two golden principles you should follow when balancing your template
> 
> ...


I agree with what you are saying, but if you are going to use non-linear effects as inserts at the channel you should set the signal level to -18dBFS pre-FX. The final volume of the track, either by CC7 or the DAW fader, can be set to taste as you exemplified.


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## jcrosby (May 17, 2021)

That's why I assumed what I did as well. -18 is most commonly used to refer to RMS levels, typically in explaining the correlation between 0 VU and 0 dBFS.

As far as template balances though, 100% agree. I gain stage at the source... The other gotcha about gain staging via faders, is that faders by default are set up to be post-insert, meaning they have no affect on gain staging into an insert plugin on that channel.

Some DAWs, (Logic for example) let you switch the faders between pre-fader and post-fader gain, but some DAWs don't, and people often don't understand the difference until they've been mixing for a bit... That said, post fader does play a role as you feed channels into a bus.. So basically one more vote for gain staging at the source... (Ok I'm done carrying on about staging )

But as a rule of thumb, no... I've seen the idea of setting all channel levels the same tossed around here at least once before. I can see why it might work for someone if it's based on their particular workflow, but I don't think it's useful to suggest it's some kind of rule of thumb, which it isn't...


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## Tralen (May 18, 2021)

jcrosby said:


> That's why I assumed what I did as well. -18 is most commonly used to refer to RMS levels, typically in explaining the correlation between 0 VU and 0 dBFS.
> 
> As far as template balances though, 100% agree. I gain stage at the source... The other gotcha about gain staging via faders, is that faders by default are set up to be post-insert, meaning they have no affect on gain staging into an insert plugin on that channel.
> 
> ...


I think the people that give the rule of thumb advice (to set everything to -18dBFS) are really confusing the input and output level of a track.


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## wst3 (May 18, 2021)

jcrosby said:


> That's why I assumed what I did as well. -18 is most commonly used to refer to RMS levels, typically in explaining the correlation between 0 VU and 0 dBFS.


Not to pick nits, but there is no such thing as RMS for full scale measurements. dB FS is, by it's nature, a peak measurement. You may want to read this *tech note* for more detail.


jcrosby said:


> As far as template balances though, 100% agree. I gain stage at the source... The other gotcha about gain staging via faders, is that faders by default are set up to be post-insert, meaning they have no affect on gain staging into an insert plugin on that channel. <snip>


There have been, since the days of tape, two primary camps.

The first camp would set the gain for optimal S/N and distortion with the gain control, or the output level of the tape deck. The output fader is then used to balance the mix. I tend to prefer this approach, but I am but one voice. The advantage is that you can optimize every insert, and every send. That is worth something.

The second camp sets all the faders for unity, and then adjusts the signal level with the gain knob. There are a lot of advantages to this approach, not the least of which is that resetting the mix is a snap. It can also optimize the levels driving the 2-mix bus, in certain cases. In this case you now have to set gain on every element, every time.

All of this is true for a DAW as well. If you know how the signal flows you can choose to optimize levels within the virtual instrument or audio track, or with the gain control at the track input, or with the faders. 

If you are just getting started with mixing I think the first approach will be the easiest to grasp, but as you gain experience you really should try both, and every hybrid in between<G>.

Have fun!!


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## jcrosby (May 18, 2021)

wst3 said:


> Not to pick nits, but there is no such thing as RMS for full scale measurements. dB FS is, by it's nature, a peak measurement. You may want to read this *tech note* for more detail.


Correct. My point if it wasn't clear enough already is that Setting levels between -18 and -20 via an ITB RMS meter puts you in the ballpark of the out-of-the-box equivalent to 0 VU, a good baseline level if mixing ITB and working with plugins modeling analog behavior.

I'm also not suggesting that it's not a rule that shouldn't ever be broken either. Deliberately overloading a something can be just what a a track or instrument calls for. But I am suggesting it's a smart baseline to start form. You have the flexibility to scale things back or dial things up however you see fit.


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## wst3 (May 19, 2021)

Please explain, I think I'm missing somwthing. How does setting levels with an ITB meter help? It is not itself calibrated against anything except number of bits, which we all (I think) agree is only valuable in as much as you can avoid overs.

Maybe I am overthinking things?

In my studio I happen to have a very accurate RMS voltmeter (several in fact). So I will create a sinewave (1004 Hz just to be safe) and then play it while fiddling with level controls in the box, and measuring the electrical output outside the box. This way I can say that -18 dBFS corresponds to +4 dBu. We are still comparing a peak measurement with an RMS measurement, but that's kind of the point.

Stepping backwards just a bit, before I do this I have already feed a 0 VU = +4 dBu sine wave into my monitors and adjusted their gain to provide 86 dBSPL at my listening position. I repeat that with pink noise, expecting the SPL to be about 3 dB lower. 

All of which provides me with a known reference with which I can work.

Is there an easier way?


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## Will Blackburn (May 19, 2021)

Bus them all to their respective groups (strings / woods / brass etc then to a SF master group. Same applies to other devs you may be using. Gain stage there. They need processing regardless so kill two birds with one stone and slap something like Cassette 2 on the SF main buss. That little beauty of a plugin makes SF samples sound miles better anyway! Then just conform to K metering adjusting your sub groups as required (SF Strings / SF Brass etc). K20 is recommended for orchestral but personally i prefer k14. It works for everything, Hip hop to Classical to Pop to Heavy Metal. And it makes your mixes really consistent across different genres. .


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## Markrs (May 22, 2021)

Will Blackburn said:


> Then just conform to K metering adjusting your sub groups as required (SF Strings / SF Brass etc). K20 is recommended for orchestral but personally i prefer k14. It works for everything, Hip hop to Classical to Pop to Heavy Metal. And it makes your mixes really consistent across different genres. .


Just went down the rabbit whole of looking into the K-System. Makes a lot of sense and when working in a DAW you only need to reduce on the mixbuses as tracks themselves don't clip internally only what goes to the final output clips.
















K-System Metering 101


The K-System is a metering and monitoring system invented by grammy award-winning mastering engineer, Bob Katz, in an attempt to curb the loudness war. It he...



www.meterplugs.com


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