Thanks Jeff.

Like Bitflipper, I don't like having to lower everything else when something is too hot. Wouldn't it be nice if someone invented some kind of grouped control where you could lower all the tracks, at once, (not the Master). Perhaps that is simpler than I think it is, to do it.
 
Also wouldn't it be nice for someone to come up with an inverse volume control where as you turn one thing up, it lowers everything else (that you assign to that). Sometimes that would be very useful, too.
I look forward to someone telling me this is already available! Maybe I just haven't thought it through. Instead of grumbling when I have to, I could be seeing this as an opportunity to approach the issue from a new, better informed place.
Michael

Well, it's not too hard to do - group all tracks and drop the Gain on all of them at once.
 
Of course, any tracks that have compression on them will have to have those compressors' thresholds adjusted accordingly. Any tracks that have an distortion/exciter/amp sim might similarly require tweaking. And you still might have an issue with a plugin distorting because of its own internal gain, as was the case in the anecdote that began this thread.

I believe in console view you just grab a fader while holding down ctrl and (in X3 anyway) all faders are momentarily grouped. I use on occasion when the 2bus is too hot.

Working with Nebula I have HAD t track everything at -18db, went to the trouble to calibrate my Dorrough meter to a specific output bus in my Lynx that never gets touched and is always on the master 2 bus.
 
It is amazing how much some of the plug-ins tweak the input/output level...Nebula will vomit in your audio if you hit it to hard, it is painful so I have had to go in and learn how to track, then making sure the plug-ins are spot on as well...reaper makes some plug-ins that can help you set the gain staging and know exactly how much a plug-in is messing your levels theres a two part "pre" and "post" gain plug that will calibrate your signal, once you've learned what the plug-in is doing your gold.
 
There is a growing industry push for the -18db as standard protocol...it gives you plenty of headroom and everyone should find a consistent gain-staging plan...
 
One of my headaches is I set the zoom on my track meters at the extent of -18 but I have to do it for each track, maybe there is a way to set it as default but i haven't found it yet...not that I was looking...it's just habit now to set my zoom level so I can see what is happening in my clips.
 
Davids video was decent, it heps see it to make sense...https://www.youtube.com/watch?v=EaHQJ0T9iHg

i have to say, i think my typical RMS during tracking is about -20....
so, close to that -18
 
but i allow peaks as high as -10, so i give a nod to dynamics
 
in mix down, i mix into a compressor across the 2-bus....
usually doing about 2-3 db of reduction, at only about 1.5:1
 
i change attack and release based on the song requirement...
 
but point is, that -10 peak here and there, gets a bit more tamed during mixdown,
but my final stereo file (pre master) is typically -22rms with peaks at about -10, so quite a crest there....
and that works for me, to keep dynamics, but still have a pretty loud master without crushing
 
 
 
the most common mistake i see amongst other folks' mixing, is getting those individual faders too HIGH across the board to begin with, not watching the master bus close enough for levels, and then adding automation, including effects, and getting painted into a corner with no headroom.

I have read this thread and the SOS article. I am dismayed. Too many references with too many associations.
 
Fur shur, lots of good advice for all of us to follow. And I welcome all attempts at helping developing safe-recording practices.
 
But too many of the comments comparing the many different aspects of analogue to digital to playback to processing to summing to exporting to conversion to actual sound are just plain confusing.
 
A few of my loud verbal responses:
 
Recording to 24 bit wav files is not the same as playback of these files.
 
Zero on the sonar meters is NOT 0dbfs.
 
Zero on the meter of any track/bus/main that is output to the input of an audio converter CAN be an indication of 0dbfs.
 
There is no headroom above 0dbfs: FS means full scale.
 
A good reason to track low, and closely monitor plugins, is because some transients are in the nano second range. Meters, and our eyes, do not respond this quickly. But plugins can.
 
(Reference Tape-op http://www.tapeop.com/interviews/btg/102/bill-cheney/)
 
I am not suggesting that seeing-no-red is a bad thing. I am suggesting that item by item comparison of all aspects of digital recording and playback with analogue recording and playback is causing confusion. 
 
Paying attention to not pumping 24 bit plugins beyond their happy zone is reasonable.
 
But to say we should not use the possible 1000 db of headroom provided by floating point calculations seems silly to me.
 
If 24 bit wav files are limiting our use of floating point math should we instead use 32 bit files?
 
Maybe Bob Katz recommendation to always use dither when bouncing to clips and exporting would fix some of what is being discerned as loss of fidelity.
 
Confusion reigns supreme. 
 
 

In terms of VU metering the same applies with analog and digital mediums. In analog, the 0 dB VU represented a level that was well above the analog noise floor but still had some headroom above. The headroom above was just built into the analog system naturally. eg tape.
 
In digital 0 dB VU is simply a level well below 0 dB FS that is all. eg -14 or -20 dB FS. But in terms of what the meters are telling you, it is the same for both systems.
 
If you keep an eye on rms levels going in and out of plugins you will never cause any clipping inside the plugin itself.
 
If you are VU metering on buses it is easy to use the meters to just create the bus mixes and end up with a perfect buss mix and the meters just hitting 0dB VU.
 
batsbrew has the got the right idea keeping the whole system well down. In terms of K system -20 is the closest level to where batsbrew is with his final mixes. -20 is a magnificent reference and sounds beautiful.

jm24
But to say we should not use the possible 1000 db of headroom provided by floating point calculations seems silly to me.
 
If 24 bit wav files are limiting our use of floating point math should we instead use 32 bit files?
 
Confusion reigns supreme. 

The point of the original post is that the "1000 dB of headroom provided by floating point" isn't always there in practice. A surprising number of otherwise quality plugins just can't handle > 0dB signals without distorting.
 
I used to believe that 32-bit floating point made it unnecessary to worry about positive decibel values, but discovered that I had been incorrect in that assumption. That was the simple point of this thread, which then took some ancillary side-trips regarding best-practices and metering.
 
 
 
 
 
 
 
 

jm24
If 24 bit wav files are limiting our use of floating point math should we instead use 32 bit files?
 
Maybe Bob Katz recommendation to always use dither when bouncing to clips and exporting would fix some of what is being discerned as loss of fidelity.
 
Confusion reigns supreme. 

Not trying to "toot my own horn" - but here's a tutorial I wrote a while back that addresses these issues:
 
http://soundbytesmag.net/ofdigitalbitsanddecibels/