We all love this topic lol

im looking for deep dive through books or articles on the topic.

intrested in reading up on gain staging for diffrent synths and plugins, mics, intruments, routing ect

something that goes deep into it all.

TIA!

 

I think people make this too difficult. Basically just set your source (first in the chain) to about 0db on the VU and/or -18dBFS on a digital meter, and keep your outputs and inputs at 1:1 at each successive stage.

With modern floating-point digital, it's not always necessary, but it won't hurt.

If someone wants to intentionally slam an input or something, that's cool too, but IMHO they should set everything at unity first, and then adjust to taste.

 

i agree and all that is fine and good but what about finding the sweet spot plug in to plugin or in synths? looking for deep dive with that stuff too

 

My experience is that 99.9 % of all Vsti plugins today follow the same rule as any other and have their sweet spot around -18dBFS. IMO and also like @Rasputin already said there is nothing much to dive in deep regarding this topic and not worth to put in more time than it requires. Gain staging is what it is. Making sure you have sufficient input level into your plugins but still preserve enough headroom on the master bus.

 

It's digital. Just run everything as hot as you can with no danger of clipping.

 

A few VSTs will specifically outline a louder level (almost always -12dBFS) as their sweet spot in their manual (always an excellent place for info), but otherwise I'd assume -18dBFS.

My rule of thumb is start at -18 and raise to -12 while using your ears, but the big caveat is that you have to gain match the output so that you're not tricking yourself with "louder is better" when comparing.

 

The "Sweet spot" is a mix of settings but it is program material dependent. Plugins and virtual synths do not contain electrical components of varying quality unless they are actually modeled/programmed that way. As an example, you may find a channel you are adding a compressor to running hot at it's input side; but by lowering the input gain you are actually able to apply a lower or higher compression ratio, a faster/slower release, more make-up gain or output side gain to get a result that sounds more how you want it. If the input gain is increased or reduced, you have a different amount of those characteristics that you can apply. They are inter-related. So there is a "sweet spot" for each parameter, but only in relation to others. Think about something like using a higher (infinity) compression ratio on a side chained compressor to get it to clamp down more because that one plugin does not have a lookahead parameter and your Attack parameter is already as fast as you can set it.

Every instance of the plugins you use will have various combinations of those parameters, and it's why we spend time going from one parameter back to another inside each plugin. It's caused by the other relative settings, not the properties of some electrical component in a device. You can have two of the supposedly same device vary from each other. I believe the term for this is Unit to Unit variations. I can have two 1176s in a rack and they can actually be different. But my Fabfilter Pro-Q3 does the exact same thing as the one you have installed on your computer; because it is code.

 

This is an interesting listen if you have the time.

https://www.soundonsound.com/techniques/gain-staging-episode-1-podcast

 

Last edited: Nov 23, 2023

There're some misunderstandings here:
- Gainstaging simply means shit in, shit out, level-wise. No matter if it's peak, TP, VU, RMS, LUFSm,s, or i or whatever.
- 0VU doesn't matter for the majority of pluggies because they're not analogue emulations.
- Analogue emus can have any calibration the dev wants them to have. This can be -24dB, -18dB, -14dB, -12dB, ... I've come across analogue emu compressors which worked perfectly fine with 0dB and one that already compressed -44dB - all FS peak.
- That's why I refuse to set my audio files to any specific VU (which, by the way, doesn't even exist in the digital domain) right from the start. But this doesn't mean that I'm not gainstaging.

 

Yeah that's basically it. Especially if you have a high noise floor in your plugins why ever that's the case.

It doesn't really matter in the analog world anymore as well, as long as you do use discrete signal processors.

Or simply use plugins that do not contain distortion.

 

Last edited: Oct 27, 2023

No. Most of us don't like to talk about this topic...
I'm sure you won't find books and articles that go deeper into gain staging because working at a nominal level is part of the basics of the craft. There is not much you can write or tell about it. It's a simple matter.
The nominal level is based on either VU (average level) or MPL (peak level). If you work ITB, it usually doesn't matter which you choose. It's a matter of your own taste and mix philosophy. I set my level mix at the beginning and want the instruments to maintain their average level in the chain, so I go with VU. It relieves my ears a bit. But honestly? Just do your level mix however you want and make sure you have enough headroom on your stereo out.

So, when does it matter?
Quite simply. If you are working with standardized levels, i.e. if you want to make sure that your hardware devices are always driven at a certain level (peak or average). This is the case, for example, if you are working with fixed settings. Or if you have found a good level range for your hardware chain. But if you play around with every device or plugin anyway, then you can confidently do without something like this. Just keep an eye on the peak meter.

 

The great Galileo Galilei decided not to put much if any effort whatsoever expounding on the science of gain-staging. All he had to say of this subject was "The planets don't revolve around gain-staging and neither should you". However, because of this he did run afoul of The Chuch of Woke Audio and spent the last nine years of his life under house arrest. His last dying words were, "The KVR forum sucks".

 

If your plugins are totally linear, where the same level goes in as it comes out, then gain staging is meaningless. If your plugins are not, and there are various non-linear plugins in the chain, then it matters where you want to hit the plugin to get the intended effect. A bit more heavy saturation, or a cleaner more noisy with more headroom for a tape plugin for instance.

I've just done a test on performance using Plugin Doctor on Bluecat Audio Gain Stereo VST vs Airwindows BitShiftGain VST and the latter is around 20 times less CPU intensive, and better performing at perfect 6 dB increments. Nothing in-between though I'm afraid for BitShiftGain. This is great for bus processing where you want to half the volume (double the headroom) on a sum of signals.

 

-20 dB at the beginning of the chain, for some reasons:

- K-20 metering (easy to read visual cues)
- I can boost 20 dB inside an EQ and still have good metering readings (even if the plug is not clipping internally, having a meter that permanently reads 0 dB FS in case of a 20 dB EQ boost, is useless)
- compressors threshold range (and knee) work just fine with -20 dB input level
- saturation is easier to apply - for example on a hot signal I can hear an audible difference between 0 and 1 (out of 15) in izotope exciter. when feeding it a -20 dB signal, I can smoothly add 1,2,3...7 saturation level and I can very precisely set it where I want it.
- and pretty much this sums up the stuff because that's all I have in my go-to (custom made) channel strip

 

I do because:

 

Here, -1TP and peak gainstaging, for various reasons.

 

Sum reasons?

 

Are you talking to me? Are you, you talking to me???

In analogue domain 0VU is the mark, in digital at the end 0dBFS peak is the limit, so why not start with it?
For digi EQs and comps it's easy to handle, when I'm using analogue emu pluggies, I adjust their input according to their meters and don't care at all to which digital level they're calibrated, be it -18, -16, 14, or whatever. Afterwards I raise the peak to -1TP again. Either with the output of the plugin itself, or a levelling plugin.
Additionally, I've set my DAW's meters to prefader metering and peak hold. This way it's monitoring the whole song on all tracks and I can easily see by the red 0dBFS peak LED when I accidentally raised the peak by just 1dB.
And since Reaper is able to do off-line rendering, it's easy AF to check a whole track, subgroup or complete mix from start to finish whether something's clipping or not.
I'm doing this for years now (apart from the off-line thing), with every mix and every master. So far, all analogue emu pluggies are working on their optimal level, no clipping, no problems, no complaints.

 

I mean, sure, there's no ceiling in digital as such, but things like compressors and other dynamic processors, distortions, waveshapers etc., not to mention all sidechain sources will react differently based on input level.

As long as my master out ends up at -0.5dBFS though. I'm not entirely sure how common intersample clipping is with modern D/As or digital mastering processors, but it's just a habit as I read it somewhere a long time ago, and it makes sense to me.

Of course, if somebody requests a certain level, then I'll abide by that. Yeah, I'm not a mastering engineer.

 

Has anyone got a great modern book or some material they can point too on this topic? (on top of the the useful plus funny comments here of course :p) Many Thanks