• Techniques
  • Hitting the red: I was wrong about this (p.2)
2014/07/11 19:50:43
drewfx1
The problem is that there is no consistent answer, so you have to know how each of your plugs behave. Unfortunately, many people can't seem to wrap their head around the idea that you have to gain stage each individual plugin.
 
Of course I like to go into the red just because it upsets some people. 
2014/07/11 20:54:04
bitflipper
gswitz
Bit, some plugs are designed to start distorting before you reach zero. They model overloading before the signal is clipping.

Yes, that's intentional harmonic distortion. Lots of plugins do that, on purpose, and it usually sounds OK. They're emulating the way tubes and transistors behave as they're driven outside their linear range. But that's not the kind of distortion I'm talking about.
 
Digital clipping and foldback are just ugly. They generate tones that are not harmonically related to the signal. They don't emulate anything in the real world. They just sound bad. Fingernails on a chalkboard bad. Technically, neither should happen in a floating-point system. I can only speculate about what's going wrong within those plugins.
 
Values are passed into and out of the VST interface as floating-point data. However, the spec calls for audio to be represented using values between -1 and 1. That means 1.000 is valid, but 1.001 may not be.
 
Now, there is no reason 1.001 can't be stored in a floating-point variable and manipulated exactly the same way as 1.000. But if the developer assumed that no value would ever exceed 1.0, it's possible that his math could be thrown off, or some logic circumvented when values unexpectedly went over 1 or under -1. You'd have to examine their code to know what's happening, but I think it's reasonable to say that it's either a mistake or an oversight.
 
 
2014/07/11 21:47:57
drewfx1
There are some plugs that add limiters/saturation at ~0dBFS. Some PSP plugs are like this (EDIT: the PSP stuff does this mainly as protection, not as a saturation effect). And I know Reaktor won't let things go above 0dBFS, but my theory is that they do that to protect people from accidentally creating an ensemble that goes to 11.
2014/07/11 21:57:48
Jeff Evans
If you have to turn things down too far on track faders before you mix you are still recording a little hot. All the more reason for K system and VU metering.  The use of the K system for me extends right back to setting input signal levels arrivinga at the DAW.  It is easy to get a K system ref level recorded on a track for any input source.  Apply it at the time of turning soft synths into audio ensures the audio is also at the ref level.
 
If you track at your reference eg K-14, then later when you are mixing, the faders are much closer to unity to still achieve a K system ref levels on any buss those tracks are sent to.  I prefer the amount of gain  adjustment around unity compared to anywhere else on a fader.  Tracks only have to be shifted downward slightly from unity in order to create a perfect balance and the right level on that buss. A buss master can also easily fine tune your K ref level on any buss.  That is what it is for.
 
With plug-ins if you make sure your levels are K-14 or 0 dB VU just prior to going in, you know you have got a nice level going into any plugin without clipping.  Then make a slight adjustment with the VU level on the way out too.  You can get those two levels (IN and OUT) very well matched and you can be sure nothing bad is happening inside the plugin itself. (things getting near 0 dB FS for example)  If you are still concerned, drop down to K-20 and you will never have an issue with internal distortion. (with plug-in chains you can always insert VU meters between all the plug-ins in a chain to ensure they are ALL working at a perfect level too. Not all plug-ins match the OUT level exactly to the INPUT level, you need to keep an eye on that. Output levels vary from many plug-ins and need to tuned back to your ref level sometimes.) Matching plug-in input and output levels with a VU also means when you bypass or test for what a plug-in may be doing in your mix, the level won't change and that sound sits perfectly in your mix, in or out.
 
With soft synths, they all vary by a lot.  Some from very quiet to very loud.  All you have to do is take a VU meter reading straight out of any soft synth and just adjust that for your K ref or 0 dB VU.  You will never hear distortion again from any virtual instrument when you get into this habit.  Some of mine are quiet and I have to add gain to them to achieve the desired K ref level.  Others need to be turned down quite noticeably.  As much as 9 or 10 dB.  Most offer output variable level adjustments.
 
By doing this all your soft synths will be at the right level prior to mixing too.  It is good to render them at the correct level, one less step to do in your pre mix work prep.
 
Reading a VU meter also takes a little practice too.  Signals vary over a wide dynamic range sometimes and it is a matter to understand what the VU meter is showing you.  We are looking at the loudest bits and over longer times as well.  After a while it is easy to see where that average point is with the meter ballistic and line that up as best you can around 0 dB VU.
 
2014/07/11 23:02:53
backwoods
Hey Jeff. I'm very interested in knowing your methodology with respect to VU meters. I know you posted it sometime before but I can't seem to track it down. Could you please repost it?
 
I bought the PSP meters and have been wondering exactly what it is I am supposed to be looking/aiiming for.
 
Thanks Jeff.
2014/07/12 09:07:42
Guitarhacker
Epiphanies are defining moments. You never know to much to have one every now and then. 
 
I've noticed that Melodyne can distort and clip in a really nasty way and you don't have to be in the red to do it. I forget the exact circumstances but the wave before was nowhere close to clipping, yet in the processing, Melodyne slammed it to the flat top..... I went back and used undo to get back to the pre-processed wave, then reduced the gain and tried again..... waaa laaa.
 
I always try to look to be sure I'm at the proper levels and check the processed output before moving on to the next thing.
2014/07/13 12:06:24
57Gregy
Hitting red on older programs is -6 dB, but I see in X3 -6 dB is orange. I haven't hit red in X3 yet. Haven't recorded with it (no interface/no FireWire port for that W7 computer).
2014/07/14 11:18:37
batsbrew
MY TYPICAL TRACKING LEVELS:
 
Peak = -6 to -8
RMS = -22
 
that's a big crest factor.
 
 
i don't feel the need to EVER get into even the yellow with my track levels,
and most of you have heard the results.
 
happiness,
is clean tracks.
 
2014/07/15 10:19:26
TremoJem
BatsBrew,
 
So are you saying that when you check your incoming levels on Sonar...you do not exceed -6dB?
 
I just read (on another site) that you should bounce the master to a 24 bit stereo track and then bring that into a mastering session, I guess, meaning that you copy this stereo, 24 bit track to another project and then use mastering tools to master the project.
 
I really did not know this. I am wondering what your thoughts are.
 
Here is my workflow.
 
Track musicians/instruments.
 
Create busses where applicable...like for overheads and drum set.
 
Mix each track.
 
Mix each buss.
 
Mix the master, as all tracks or busses go to the master.
 
I then use the master track to master on, using tools like Ozone5.
 
I could not tell you if there is, or is not a benefit to bouncing the master to a 24 bit stereo track and then copying this stereo track to a new project and the mastering it...but it does kinda sound like it would be.
 
What do you think?
2014/07/15 11:17:42
batsbrew
tremojem
 
i'm talking strictly about TRACKING.
 
during the tracking phase, i like to try to have my individual tracks never exceed -6 on PEAK.
 
typically, it's actually -12.
that is the sweet spot.
 
when i combine all of my tracks together to do a mix,
i try to keep the MASTER bus with peaks of no more than -8, and RMS of no more than -20.
 
this has to do with matching analog scale of 0dbu at digital scale -20 dBFS
 
so when i pull my final mixes into WAVELAB to do my mastering,
i have a file that usually has a Peak level of about -6, and a RMS of about -22, if i did my job right.
 
 
then i master it, and typically bring my levels up to peak of -0.2 and RMS of about -10-12,
aiming for a dynamic range of about 10.
 
i can push my masters louder, to about DR8, and NOT get into trouble,
but i like the sound of it less squished.
 
 
if i could put out my mixes WITHOUT mastering, i think those sound the best of all.
 
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