For those who gain stage and use many analog moddled plugins or just gain stage by workflow, do you wrist makeup gain by ear, rms level difference, LUFS Meter before and after or even plugins with automatic gain compensation etc
Just make sure that at every stage your peak values are hitting around -10dBFS. Alternatively you could use a VU meter calibrated to something sensible like -18 dB or -20 dB (then just make sure individual stages are around the 0 mark). The idea is not to overload at any single stage so plugins work in the optimum range. Here’s a good video for Mixbus which works equally well in Ardour: Mix Tips - Gain Staging in Mixbus (v3,4,5) - YouTube.
the thing with ‘analog’ model plugins is that they introduce ‘artifacts’ (example: saturation, slight eq changes, among other artifacts) on the input or output… Different plugins have their own unique vision of what is making that ‘analog’ sound. Implementation is different for different plugins and vendors… therefore you cannot rely on simple metering. The best way is to
- determine the difference by null test at various levels
- determine whether your current work will benefit from this or not (example: snare gets punchy, but the slight eq change might not be what you want)
Gain-staging shouldn’t be a problem if there is no excessive make-up gain at the end of every plugin instance,. If you are looking for ‘analog’ sound, you should crank input stage (and compensate output), so the plugin can recieve the ‘hot’ signal, and therefore, add more ‘analog’ whle keeping the output signal similar to input (without major difference in levels between input and output).
I hope this helps
I respectfully disagree and I think you are perhaps overthinking the OP’s question. Simple gain-staging of peaks hitting around -10 dbFS at every stage will be absolutely fine.
but that truly depends of whether someone wants more or less of that ‘analog’ treatment… not on the peak nor rms value… perhaps you would want to drown the drum bus into analog saturation? it depends on the material, really… It’s not real analog…
what do i know after all
The point is that whatever effect/volume you get out of your plugin, the trim into the next “stage” should be peaking around -10 dBFS
well, i’d like to agree, but:
what happens if a plugin has lower input threshold than that?
what happens if a plugin introduces ‘analog’ mimickry at much lower levels?
I think you are missing the general point. Whatever a plugin’s input threshold, the audio that goes to the next stage should be peaking around -10 dBFS achieved with the trim knobs on the bus, for example. If a plugin introduces ‘analog’ mimickry at lower levels then you adjust your plugin input gain knob. It’s not rocket science.
Let’s say you have a track with a chain of, say, 5 plugins (4 “digital” + 1 “analog”) leading to a bus. The input into first plugin should be around -10dbFS, likewise maintaining that through the entire plugin chain by adjusting the input gains of the various plugins. Let’s say that in order to get the sound you want from your “analog” plugin you do need to increase/decrease the input somewhat. No worries. Whatever the final output, the trim knob of the bus re-adjusts that to -10 dBFS peak so that the input into the bus plugin chain is also starting at an appropriate level. That is the whole point of gain staging.
Beth is right, and I say this as someone who uses a boatload of Airwindows Tape saturation (ToTape6). -10db per track at its peak, with saturation/distortion.
Good points overall. I know some plugins have automatic gain compensation that’s based on rms level difference, but I still think they adjusting it manually helps more. Also sometimes after I eq something I realize that at times if I try to match it with the meter it doesn’t sound the same like it’s as if the volume has to be higher but sometimes with certain sounds I don’t have to adjust make up gain with eq too much all depends on how I processed the eq.
I guess I try to match it 99.9 percent as I gain stage so I was wondering how others do it.
My advice? Don’t fret too much. If you did that for every plugin in your project you’d never get any work done Either “ear-ball” it or stick to those -10 dBFS peaks and enjoy the freedom “gained”
So since I generally aim for -18 which is -12 db on a regular peak meter should it be -12 db peaks
It doesn’t work out that neatly due to the way the two meters types react differently to incoming audio but, yes, -12dbFS peaks would obviously be fine.
I’m chuckling at this, mostly at myself, because before I did digital, I did tape. And gain-staging to tape, I tried to get the hottest signal possible for each track without clipping. So my signal chain would go through a pre-amp, then hardware compressor/expander (or noise gate to compressor, but the expander I liked better) to sort of tame it and get the best dynamic range I could, and then to tape.
So, naturally, when I started playing with daws, guess what I did? Aimed for as close to 0db as I could get of course. Heck, I even kept the preamps and hardware compressor expanders for feeding into the digital interface! I’m amazed I didn’t blow anything, because my first digital interface wasn’t even external, but was a PCI card in the PC with a D connector and a rats nest of inputs and outputs and midi stuff coming out.
Anyway, digital ain’t like analog. In the analog world, I was trying to get the best signal to noise ratio by making the signal as loud as possible before I started mixing. But in the digital world, standard 16, 24 or 32 bit recording gives you so much dynamic range that a signal at -10 or -12 is perfect. I generally aim for a range between -8 and -12.
I am new to Ardour and new to this forum. I own Mixbuss7 32C as well.
As a serious hobbyist trying to better hone my DAW skills I so appreciate conversations like these.
I understand the importance of and utilize gain staging as good as possible based on my limited experience. So when I read about gain staging your VST’s, if I read that right, i am wondering where and how one would do that?
At the sound card input or on the volume fader after one applies the effect?
How do I adjust the input gain of a VST?
Sorry for the newbie question but I guess everyone has to build from where they are if they are to learn anything.
Unless you mean creating a separate effect channel where one would have better control over an individual effect?
A well designed plugin (regardless of whether it’s LV2, VST, or AU) will have input and output gain knobs inside the effect control parameters, and the goal is the set that up so that you’re getting between -10 and -12db of gain on both input and output.
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