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As I have mentioned there are two components to a signal. The rms level and the peak level. Normalising is a waste of time and basically you don't need to do it. Normalising only takes into account the peak level. (unless you can get rms normalising but Studio One does not do it)

One track might have an rms level of -25 but a peak that goes to -6. Normalising that track will only take that track up 6 db. So the rms level of that track might still only be -19 db rms. Another track might have very little peaks e.g. a cello track doing a slow legato. Its peak might reach -14 and its rms might be -17. So normalising that track will take the cello track up to -3 db rms which is way too loud. Normalising will put the rms levels all over the place and they won't be consistent.

Look at the above example. One track after normalising ends up at -19 rms and the other at -3 rms.

VU meters are slow to respond to peaks so they don't show them only the rms level. rms levels are roughly equivalent to volume and how loud a track actually sounds. So the best way to get consistency over all your tracks is rms wise and with a VU meter. Another good thing is all the tracks will have roughly the same waveform heights and will be easy to see everything.

I don't like the all faders at unity concept either and recording tracks at their mix level. Some will end up way to low. (you won't even see some waveforms) Faders at unity tell you nothing about the mix. When track levels are all consistent rms wise, then the faders should and will be varied and all over the place. Instant visual feedback as to what tracks are soft in the mix and what tracks are loud. Just as they were in the analog days.

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by SwitchBack on Tue Mar 19, 2019 4:40 pm
Funkybot wroteThese days, I like to keep my levels low-ish. Fun challenge: try to get your levels where your mix already sounds pretty good with all faders at unity. With virtual instruments, it's as easy as doing a pre-mix using the output level of each instrument, but with acoustic instruments and anything else you're mic'ing up or bringing in via a DI, it's not a bad idea to track around the same level you expect to mix at. The added benefit: it makes mixing easy as hell when everything already sounds pretty good before you move a fader.

Now, why even bother? Why not record everything as close to 0dbfs as possible? Well, 1) your converter was likely calibrated to operate with more reasonable reference levels in mind (probably close to -14dbu), and 2) if you're using "analog modeling" plugins, you probably don't want to slam their input unless you're intentionally trying to drive a track into distortion. Remember: most analog modeling plugins are generally calibrated with a reference levels between -18 and -12dbu. If you go hit them a signal peaking close to 0dbfs, then you're going to drive these plugins way harder than anyone in the analog domain ever would (unless they were intentionally trying to distort the signal).

Ideally, you'd do your gain staging on the way into the DAW, but doing it after the fact will still have benefits if you're using analog modeling plugins. So doing the trimming in the DAW or via gain plugins isn't entirely without benefit.

To avoid any confusion: I’m not suggesting to set levels at 0dBfs, because in analog gear that would be the absolute maximum level, on the brink of clipping.

So it should be lower than that, -12 to -18dBfs sounds about right, also depending on the number of channels contributing to the mix. But don’t go too low or it may become difficult e.g. to get the next compressor in a nice working range.

NB. dBu is a meaningless unit in a DAW because it is referenced to a voltage level.
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by Jemusic on Tue Mar 19, 2019 6:30 pm
Where dBu is important though is the spec for your audio interface. It is worth looking at what your interface puts out at 0 db FS for example. The better ones pump out some serious voltage levels at 0 dB FS e.g. +18 or +20 dBu at 0 dB FS.

What this means then is at +4 dBu the internal digital ref level is around -16 dB FS. If you were working at say -18 then you would know you can output at least +2 dBu which is a lot. It means you have got some serious headroom on the analog output side of your interface.

Many USB powered interfaces running on 5V supplies will be lucky to put out +4dBu at 0 dB FS which means down at -18 FS ref levels output levels of -14 are only possible. At higher outputs less analog headroom results.

Specs i5-2500K 3.5 Ghz-8 Gb RAM-Win 7 64 bit - ATI Radeon HD6900 Series - RME HDSP9632 - Midex 8 Midi interface - Faderport 2/8 - Atom Pad/Atom SQ - HP Laptop Win 10 - Studio 24c interface -iMac 2.5Ghz Core i5 - High Sierra 10.13.6 - Focusrite Clarett 2 Pre & Scarlett 18i20. Studio One V5.5 (Mac and V6.5 Win 10 laptop), Notion 6.8, Ableton Live 11 Suite, LaunchPad Pro
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by frank.crow on Wed Mar 20, 2019 8:36 am
kaosikosmos wroteIf the point is to get all tracks at the same volume, why not just select all audio events and normalise, then drag them down to -18 or whatever you like? Will doing this change the waveform or destroy the dynamics?


I actually run my audio tracks through the “stereomonizer” plugin (I think that’s what it’s called). It has a normalize function. You could actually set it to -18 and then just pull everything into a new session.

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by Vocalpoint on Thu Mar 21, 2019 1:15 pm
Over here - from time to time I simply run my trusty 1KHz calibration sine tone thru the Mixer and into S1 - tweaking the gain on all relevant analog (mixer) channels so the Mixer metering shows 0dbFS while the Studio One meters show exactly -18dbFS

After that - start recording and do not worry about levels. Beauty headroom and the track almost mixes itself most times.

I apply the same rules to ITB VSTis etc. Drum kits are bit more lenient with peaks of kit pieces rising up to -10 or -8. But I keep a tight rope on all levels.

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