Pro Tools Normalization plugin

Normalization can't be done in real time....ie, via "plug in".

There is no difference in "quality"--it's literally finding the peak and turning the volume up or down until that peak is the given value. It's not altering the audio anymore than your fader is.

So if using Normalization is non destructive then there is absolutely no reason at all to record loud levels because you can always turn them up later come mix time using Normalization . Ronan Chris Murphy said some converters even before 0 start to distort a little so even getting near the top is not good .

You could pretty much just leave your mic pres gain knob all the way down if you wanted all the time .

What level when recording at 24 bit is actually TOO low . This would be based on the mic pres noise floor ?? Lets say a CAPI VP26 . How low can you record with a VP26 before noise becomes a factor ?



Heres a beautiful chart I found ,

I'm gonna admit something embarrassing. I don't really know what "normalizing" is and I've never done it.

Thank you.

Ragan

Do you use pro tools ?

if you do go to the  audiosuite plugins , and you'll find Normalize . Highlight a section of audio and if the plugin is set at 0 it will then change the highest peak of that audio to hit right under or at 0 . It's just quick volume control , but for doing comparisons of stuff like mic pres you can match levels exact . 

I've never understood the normalization thing, mostly because when I learned, the guys that were tracking were always setting their levels to tape so that when they put all the faders to 0 on the console - you had your basic rough mix.

I've been doing it like that ever since.

Normalization IS destructive. This is especially obvious with lower level values--even if those values are just the components of a waveform near 0 dB. Let's use an example.

Assume your normalization scan indicates the overall signal should be boosted by 1dB. This means that it should be multiplied by 1.122 and change.
Now assume you've got part of a signal that has 4 bits of magnitude. Expressed as integers, you can code values from 8 to 15.
Multiply those values by the boost of 1.122. A value of 8 turns to 8.976. A value of 15 turns to 16.83.
But there's no such thing as a fractional bit. Those values must be rounded or truncated (they're usually truncated). So our original value of 8 stays at 8 and our original value of 15 now becomes 16. We haven't just changed the gain of the waveform--we've distorted it. This is less obvious in the louder parts of the signal, since the error is lower. But for every time a wave hits a max, it also passes through a min.

If you absolutely feel like you've got to change the gain, then clip gain is better. It's non-destructive and the math will happen in floating point, pushing the error waaaay down.

But the real question is why you want to normalize in the first place. Maximum signal level is only loosely connected to perceived loudness. The loudness you perceive in a track is a function of the actual content as well as the tracks before and the tracks after. There's no reason to introduce distortion into a track--especially when you're making it permanent. Final adjustment of overall levels should be moved to as late in the process as possible.

Clip gain ......you mean in Pro Tools the Digirack "Trim" plugin ?

So it looks like Clip Gain came out in PT 10 . I'm running 8 , so I don't have it .

I really don't like the implications of "if you absolutely feel like you've got to change the gain".

You DO, IME. Unless you recorded the tracks to your studio nominal level....or someone else recorded to theirs AND theirs is close enough to yours to not bother.

If you DON'T....then you have to immediately take all the faders to their bottom 1/3rd of their throw where there's far less granularity in movement....you can't insert your analog gear via hardware insert because your IO is calibrated to your studio's level. Actually--I would argue that's usually what DEFINES your studio's nominal level. You can't use analog modled plug ins that expect a certain level as X voltage.

Anyway-tape return channel trim has been a function of literally every mixer I've worked on my entire life. Analog or digital. Done for somewhat different reasons on analog than digital, but it's always been step on of mix time to get all the channels getting their feeds at the right levels. So, the implication that it's a some weird harebrained idea is....patently historically incorrect.


THAT SAID....file normalization is not the way to do it for a different reason. I honestly don't KNOW that it damages the sound in any meaningful way....why would it not be the same 32bit volume math as doing it any other way? I guess theoretically it could be that writing the 24bit file back embeds something "lossy".... But, I've never tested because I've never done it for a different reason: The reason it's not the best way to do it is that it gives too much "value" to an absolute momentary (potentially one time) digital peak level. Which isn't how levels are set. OP has noticed this by his assertion that he wants drums "normalized" to a different level than guitars....this is actually WHY normalization is sort of a dumb way to do it. A Distorted guitar and a snare should be roughly set to a pretty similar window, but because of the nature of that snare it will mean there are some higher absolute peaks....where the dirty guitar doesn't have much in the way of high absolute peaks.....it's average peak will actually BE pretty near it's peak....where the snare's average peak might be 10db below some occasional peaks.

So, the discussion to be had is what's the least destructive way to adjust the gain of tracks at the beginning of a mix....PRE fader/insert/channel. I used the DSP24's 56bit fixed mixer's channel trim....then when I moved to software, Cubase's input channel trim, which is right at the top of the mixer channel next to the polarity like I'm used to from analog and later digital mixers....Logic and Protools not having trim/polarity functions as part of the mixer is a really shit move, IMO, from a position of workflow. While it's easy to say "just insert a Gain (in Logic that handles both trim and polarity) across all the channels--that still doesn't tell me from looking at it what's being done. Is the snare bottom reversed polarity? Now I have to open a plug in to see? I suppose I could use two....and just know the first is polarity and the second is trim....and default them to deactivated....I digress.....point is, the channel input trim is how I've always done this. I would love to hear if there's abetter way--I've taken to using the Region Gain, which is effectively the "clip gain" that Cubase and Logic have had for decades without a catchy name....since Logic requires a plug in to trim....