matthewbarnhart

If you're keeping Pro-L in your workflow, you can use it as a clipper.

Mastering engineer here. What you're talking about is not mastering, it's mix buss processing. If I only had Logic tools to use for actual mastering, it would be the Channel EQ and the Adaptive Limiter, and maybe the Compressor at a very low ratio and slow attack/release settings. Anything else I would use would be determined by specific problems that are present in the mix. If you're doing extreme stuff on your stereo mix (100 Hz HPF? 4:1 compression w/ <1ms attack?) then I would suggest fixing those problems in the mix. Linear Phase EQ is inappropriate for mastering except in a few very specific circumstances. It isn't "higher quality" than the regular channel EQ.

MacBook Pro M1, Ventura

Yep, noticed that here too.

And I guess to clarify my POV: are you sure it’s an actual problem that the beat drifts from 120 to 126? Are you sure bleed from amps/etc is a problem? When I started recording I thought all of that stuff had to be eradicated. I was soon fortunate enough to work on some multitracks recorded by my heroes, which were full of bleed and tempo drift, but the final mixes were amazing. Really turned my head around. Again, it will depend on your style of music, but you might find it illuminating to try quantizing some of your favorite recordings from before, say, 1979, and see what happens!

Just wanted to chime in as an anti-click guy: the basic tracks for most of my current band's records are recorded in our basement during practices, all playing together, without a click. When I start mixing, I beat-map as described here -- not to align everything to a grid, but more of convenience when setting delay times, selecting bars/beats for automation, etc. I typically start beat-mapping the first downbeat of each section, maybe going down to every bar or two depending on the tune. On the occasion there's an errant hit (usually my bass playing, ha!), the beat-mapped grid is helpful for quickly nudging things into place. This works well for our style of music, which is indie-dad rock. Other styles might need more rigidity, but I would suggest starting very lightly with quantizing if you aren't playing to loops/samples/tracks/etc. One thing I love about Logic's workflow for beat-mapping (and flex pitch) is that it leads me to do less messing around and get more natural, pleasing results than I might in other DAWs. My $0.02!

Seconding the recommendation for Nova. It's actually my primary digital mastering EQ these days. (I use the paid GE version, but used the free version for months beforehand.) It is relatively processor-heavy but certainly does the trick.

Because often it doesn't make a difference. Have you watched this video? Or this one? O

To be clear: I actually work at Chicago Mastering Service, which is owned by Bob Weston of Shellac, Albini's band mate. As such, most of my tape work these days is playing back 1/2" and 1/4" masters. I do happen to live at Electrical Audio (long, weird story!) and work there as a freelancer from time-to-time. Both rooms are fully-analog (along with Pro Tools systems) and Steve still only works on tape. The studio I co-owned for 20 years had both analog and digital capability, and most of the records I made there were 100% analog. This is by no means an appeal to authority, but hopefully it's of use to the curious!

Well there's your problem: the supposed analog modeling of these plugins is based on the RMS/average signal level, not Peak. Therefore, the general aim of all this gain staging stuff is to reach -18 dBFS RMS/average (0 VU), not -18 dBFS Peak. To attempt to clarify further: In your setup, 0 VU = -18 dBFS RMS/average. Not -18 dBFS Peak. Analog systems typically used VU meters and RMS/average levels for measurements. (Analog peak meters exist, of course, but day-to-day studio work used VU meters.) But, transient-heavy material (drums) will have very large peak levels compared to the RMS/average values, so VU meters (aka RMS/average) were not as useful for these types of signals. You can see this in the first attached screenshot from a Logic session of mine, where I've adjusted both the Snare drum and Guitar channels for an RMS level of -18 dBFS. (i.e. the assumed ideal for analog modeling plugins.) As you can see, the Snare drum has a Peak level of +3.1 dBFS. In an analog system, this would clip the circuitry and likely cause distortion. (It will also clip the output of your converters, which you don't want.) The Guitar has a Peak level of -9.6 dBFS. So, if you normalized all regions to -18 dBFS Peak, you'd get what you see in the 2nd screenshot: the guitar would have an RMS value of -26.4 dBFS, a good 8.5 dB lower than the "ideal" level for your analog emulation plugin. You'd be missing out on whatever harmonics and distortion they're trying to add. Again, in a DAW none of this matters. It does matter in an analog system, because the closer you are to the ideal gain staging the better signal-to-noise ratio (and performance in general) you have. And this is why I, a person who has 30 years of analog workflow ingrained in him, normalize drums to -6 Peak, and everything else to -18 RMS (or -23 LUFS, which is close enough for government work.) But a digital native really doesn't need to bother.

Right, it seems like you still aren’t grasping the difference between Peak and RMS/average levels. Have you looked at Multimeter as I suggested? A VU meter only shows RMS/average levels. Audio with lots of fast transients (drums, acoustic guitars, percussion, etc) have a relatively low average level compared to their peak value. Look at a waveform of a snare drum and a distorted electric guitar and you can see the difference clearly. In the analog tape world with only VU meters, we would usually try to hit 0 VU for bass, vocals, electric guitars, and synths; for drums, we’d start around -10 VU and listen for distortion, as the high peak levels weren’t represented by the VU meter. All that said, as many others have said, this is all an intellectual exercise in a modern DAW. It’s fine if you want to learn all this stuff (and there are many excellent resources out there — I started with the Yamaha Sound Reinforcement Handbook in the 90’s) but none of it will make a practical difference in your workflow today. If you just want to make music, forget about it.

True, but those records were also made by well-trained engineers who understood mic selection, mic placement, arrangement, analog gain-staging, tape recorder calibration, acoustics, etc... all recording a lot of really excellent musicians. Sgt. Pepper's was indeed made on a 4-track, but it sure wasn't made on a PortaStudio!

Agreed. I started recording nearly 30 years ago, so proper analog gain staging was drilled into my head early on as my workflow. There's no reason for digital natives to bother, so long as their converters don't clip.

-16 RMS will correspond to 0VU w/ your calibration. -16 Peak is a different measurement -- that's what you see on the meters in the Logic mixer. Check out Logic's Multimeter plugin, you can see both RMS and Peak side-by-side to get a better idea of how they correspond.

I finally got home from a long day, opened up Logic, and realized the Logic compressor's VU meter only does gain reduction -- not input or output level -- so my suggestion won't work at all. So sorry for the distraction!