So I read a few people say that recording and mixing at -18db helped them significantly. What exactly does this mean? Crank your monitors and mix w the master bus being at -18db? Or leaving the master bus alone, and having the individual tracks at -18db? I understand the concept, its too leave headroom because as you mix and process tracks loudness increases (potential risks of clipping) amirite? Can someone explain to me exactly how to do this and why?
Thanks guys
No, leave your 2bus untouched. What it means is - have your tracks sitting around -18dBFS before you process them with plugins, and have all channel and bus (including 2bus) left untouched at zero.
In analog world idea is/was to have your tracks around 0 VU (+4dBU), which, depending how your converters are calibrated (two most often settings are -18 and -20dB), translates to digital realm to have your tracks around -18dB RMS (
not peak, although quite a few guys here think it's like that; having tracks peak at -18 is not wrong per se, since noise floor in 24bit audio is so low, but it is not necessary, and in this case is misunderstood). That could mean that your tracks should peak around -6dBFS or something similar, it depends, it's not linear (for ex. a snare track will have higher peak level comparing to distorted guitar track, even though their RMS level is the same).
Every once in a while a thread like this yours pops up, but nevertheless - have your tracks peak between -12 & -6dBFS (if a project was not tracked by yourself, set all tracks to sit between these two values before any further processing and balancing; if you don't have trim knob on channels in your DAW of choice, use a trim plugin like FreeG for that). Try to maintain that level throughout whole plugin chain on every track. Meaning, if you for ex. hit a compressor with like -6dB, try to set its output so that it gives you similar dB value after the signal has passed through the compressor (use your ears for that). The same goes for ever other plugin on the track. Now you don't have to be anal about that, just don't clip input any of your plugins. I use FreeG plugin for very this purpose, if a plugin doesn't have output knob, like some compressors. Try and maintain roughly the same level throughout the whole chain.
Don't use compressors to increase loudness, use them to alter dynamics and envelope of your audio. While I am at it, don't use any of your plugins to increase loudness (we're talking about mixing here). There are channel faders in every DAW for a reason. Use them for level balancing between tracks.
Even though the only thing that ultimately matters is to not clip your DA converter, most of VST plugins work best around -18/-20 dB RMS, so try and use that to your advantage.
edit:
to keep it simple, -18dbfs in the digital realm is equal to 0dbfs in the analog realm. this means that if you record a track into your DAW that peaks at -6, it's +12 in analog...aka clipping like fuck
el colonelo, you are wrong. Read my post. You are mixing up RMS and peak. And there is no dbFS in analog realm anyways, it's strictly the digital 'thing'.