Me being an amateur, I would normalize by habit. I don't think I'll be doing it ever again in Reaper. Coupled with some advice about not touching the faders until the end of mixing, my tracks now have a more rounded feel. Keep up the good work!

Have you done an article about getting a greater RMS output on basic mixing software?

I use normalization in two ways:
- at the end of my podcast where i am the only person talking to bring it up to a -1 db level to get some consistency. All files are treated this way so there is a known level to it, but I also control surrounding and such.

- when I do interviews, I do them through skype and though I try to level both sides before the interview, it sometimes does not work and I acnnot fix my side as they both are recorded on the same track. I use a software called levelator, which is a freeware, to bring usually my voice up or the male voice down. This takes a toll especially with my side to increase the noise level, but if I have the choice of unbearable volumne levels or a bit more background noise with mine, it increases the quality.

Most (all?) DAWs have a normalize feature built in, though .. With Sonar, it's on the "Process|Audio" menu, Cubase has it on the "Audio" menu. Reaper puts it under "Edit|Item Processing".

And, of course, you can do it "manually" by playing the file through once, noting the peak signal level, and raising the track's gain by the required amount.

(Thanks, Adam)

Multiplication and bit-depth can get be a pretty complicated affair, so I'll use an [arbitrary] example to illustrate. Say you've got a 16-bit WAV file where the peak sample value is ~0.713684, or 23386/32767 (32767 being the number of possible positive values represented by 16 bits), and you want to normalize it to -0.1db FS. That will require a gain of ~2.9db, which will make that peak sample becomes 32391.919252…/32767 . To be representable by 16 bits, that extra stuff after the decimal place needs to be either dropped or rounded off (also know as truncated or dithered, respectively) for every sample in the file. Your other option is going to a greater bit-depth that can more precisely represent those remainders.

Hope that makes sense.

I suspect I've got a forehead slap in my near future, but: You can't normalize past full-scale, so there'll never be overflow in the multiplication. Why expand the word length when you know you're always gonna throw the extra bits away?

Or, if it's just done as a matter of course (i.e. normalize and gain adjustment use the same code,) then why dither when the truncated bits are always 0?

> Hopefully my explanation doesn't betray a complete misunderstanding of digital audio! :)

Heh, maybe I've gotta reign in the hyperbole a little :-p

The important thing to note is that 99.9% of the time normalizing in 16 bit is a lossy operation. I suppose the real question is can anyone tell?