I was reading the thread RE: 32 vs. 24 bit recording. In it someone made a reference to gain staging. As to not derail that topic, I’ve created this one.
Gain staging has been a topic a friend and I have been discussing for the past while. I used to aim to have my input levels hit -3dB (applying compression on the way in) – of course against the advice of all pros. My friend pointed out with 24-bit recording (rather than 16-bit), printing such loud levels was no longer necessary (because of the headroom 24-bit offers, I think). Heeding his advice, I made several changes. I:
used -6dB as my peak input level
started using the gain staging in the EQ portion of the channel settings to reduce SD2.0 (and other VSTi) output levels as to not exceed -6dB
started playing through the entire project and lowering the Gain (not the fader) on the Master channel so the peaks maxed at-6dB
While all this seemed silly to me and I didn’t really understand it (because of the seemingly infinite headroom internal processing seems to allow), I noticed my mixes sounded a lot better. They were punchier. The clarity was better. The imaging was better. And that’s still after applying mastering plugs (Maximizers, et al) in the same way.
He suggested it’s, at least in part, because plugins, especially those made to emulate vintage gear, want to see levels similar to what they saw back in the day (which I guess were substantially lower). He went on to say he uses -18dB for his peak input level when recording, and that lowering channel gain as I’ve been doing defeats the purpose or isn’t as effective.
What are your rules about levels at the input, bus, and master bus (pre-Maximizers et al) stages? And what’s your reasoning?
I have not the time to give a lenghty qualified answer right now (I’ll try tomorrow).
But for a starter, I suggest you read up on the good old VU meter, and learn how to use it.
Why the standard is set to, 0VU = +4 dBu = 1.23 Volt in pro audio equipment (often set -18 dBFS in the digital domain), and so on…
I suspect the cleaner portion of your recording, is first and foremost that you are not driving your analog chain (pre-amp) to hard.
0VU is there for a reason. That level (1.23 Volt) is a standard were the most pro audio manufacturers calibrate their equipment to have the best S/N Ratio (Signal to Noise). The best middle ground between noise (lower han 0VU) and distortion (higher than 0VU) added to the signal. That’s why 0VU is a standard.
The same goes for the newer quality analog emulating plugins (Sonnox, Slate, Softube etc.). See my comment in the 24 vs 32 bit thread.
If you then add that you don’t ever have to worry about missing headroom, what’s not to like.
Most people tend to forget the analog chain, pre converters. This is the most important part, as once you have captured a signal in your DAW you can’t take away whats already there (Whether added noise or distortion).
Not that lenghty, but I think you got all my reasons as well. For the most part anyway
IBM, yeah those are the exact things to which my friend was making references. It’s so great to have access to such knowledgable folk here! So, your point, distilled, is 0VU (analog) = -18dBFS (digital) and the optimal level for input as far as plugins are concerned. Is that correct?
I look forward to your fuller explanation. Thanks!
0VU is the most optimal level for the most pro analog gear (pre-amps, compressors, eq’s etc.).
Most digital DAW’s and plugins are calibrated to 0VU = -18dBFS, and therefore many plugins that emulates analog gear has been calibrated the exact same way.
We are obviously talking about RMS levels, not peaking. Hover your signal around 0VU and you are good to go.
The easisest “translation” (my rules of thumb when teaching) if you do not use a VU meter (you really should) are these:
For percussive sounds (drums/sharp transients) is to use a peak level at -6dB .
For Signals like distorted guitars, long sustaining organ parts etc, use peak levels at -12 to -18 dBFS.
This is due to the nature that:
Long sustaining (and compressed) signals will (and should) have a lower peak level while reading the same 0VU as short transient full signals.
Added bonus: Following the above “rules” you will be able to have you faders closer to 0 in your DAW, providing you a much better resolution (more precise fader moves) than lower down the scale (due to the Logarithmic dB scale).
There are excellent VU meter plugins out there (will provide tomorrow).
By for now, it’s soon midnight in Norway. Good night.
preamps (interface or external) have a sweet spot, probably met around -18 dbFS in Cubase (see documentation of preamps/interface)
recording to 24 bit or 32 bitFP (32 just makes sense when recording with plugins or extended usage of offline processing but also doesn’t hurt much, see other thread) offers lots of headroom indeed
the internal headroom of the Cubase mixer is around +1500 db (of course your converters’ headroom isn’t) so internally nothing can clip, but…
…plugins have a sweet spot regarding leveling just as the preamps and converters - this is different from plugin to plugin. Serious emulations of vintage equipment certainly have a standard they’re calibrated to (see documentation) - good idea to make wise decisions of input and output levels here (= not abusing plugins, unless it’s a consciously made artistic decision)
due to the internal 32 bitFP processing the Cubase’ channels can exceed 0 dbFS without damage to the signal but as there are a lot of them summed it makes sense to keep everything far below just like on an analog desk (ok, analog sounds ‘hotter’ when driven hotter, digital doesn’t without extra efforts like saturation tools/console emus etc.)
of course the master out(s) should never exceed 0 dbFS as the master level hits the hardwares’ converters
That’s all the magic: keeping things in the sweet spots of whatever hard- or software (plugins) used. Otherwise the master out becomes sort of a bottleneck, with everything hitting against the brickwall.
I go around -12 to -10 dbFS on the way in. This has proofed to be a well working leveling on my hardware. It happens that some signals might be hotter but also lower. As long as the pure file sounds good, everything is good that far. For the mix, the channel gain can be lowered or raised to get to whatever your idea of an ideal level is (-18 to -10 db seems like a good range…).
When mixing I’ll do some gain staging anyway, almost always using Slate’s VCC usually as the first plugin - I set the gain of every channel to hit 0 dbVU on VCC’s meter. If I don’t use it, it depends on the plugins I use. Most ‘formal’ EQs are pretty forgiving (probably working in 32 bitFP internally), ‘vintage’ EQs might produce different colorations when driven more or less hot. Dynamics need proper gain staging (input levels) anyway - technically. ‘Non-proper gain staging’ can be the method of choice as well if the result is what you’re after.
About busses: same story, usually below 0 dbFS, also depending on the potential bus plugins. If the channels feeding the bus have a good relation but are too hot/too low for whatever reason - that’s what the group channels’ input gain is for. Same thing for the master, which is just another bus.
Oh - I was late to game, iBM already said some wise words
I also use Slate VCC (and the rest).
And as one “offset” to the 0VU = -18dBFS, Slate has set the calibration to 0dBU = -14dBFS by default for their VTM (Virtual Tape Machine).
So as marQs say, read your plugin documentation for best use.
Oh yeah, UAD Ampex ATR-102 and UAD Studer A800 are calibrated to -12 dbFS @ 0 dbVU. Stuff like that’s essential to know… for both proper use and proper misuse
Man, you guys are awesome, iBM and marQs. Thank you for your extremely useful and on-point responses.
I was sifting through the Cubase manual to see if the meters could be set to RMS, but it looks like only the Master bus can do that. Bummer! I’ll look around for the VU plugins. I thought PSP had a free one back in the day…
I think your (iBM and marQs) posts, the fourth and fifth, tell me all I really need to know in terms of application. Thank you for taking the time to provide that info. That said, I have my eye on this thread so if you choose to add more later, I’ll catch it.
So I was forwarded to this link via techical support. I don’t know why I can’t get direc answer to my question because it can be dependent on software version. But anyway, if somebody from the pros or more experienced user can comment I would be glad. I have read this thread above, but din’t get the answer to my question: Hi, after mixing obviously with too low levels and freaking about the digital clipping, I got bit frustrated trying to find help for gain staging in Cubase 9 manual. I couldn’t find there any mention what are the meter fader true levels.These things are left out from Artist version. But when I added a software VU-meter to master bus, I noticed that 0 at master bus means actually -18 dBFS. Can you confirm this to me? So there should be plenty of head room when I reach 0 with master fader? This is a very important basic things that should be explained in a manual and I would be interested.
The matter is not about plugin. I wanted to know what Cubase Artis 9.0 master fader output level actually is. It looks like 0 means -18 dBFS. It is hard to make any gain staging if you don’t know the actual output level.
