If a track is already too loud, EQ is an answer. Because it is not only a matter of level, but a matter of TOO MANY frequencies.
They will "mask" your mix. So remove them ;)

Another solution is to use an expander (opposite of compression) Fabfilter pro-G, pro-MB, Waves MB..
Or using a transient modifier, to recreate some transients.
They will lower RMS.

 

I am aiming for between -18 and -12 on each instrument track (depending on the density of the piece) while trying to keep peaks to no higher than -6. This way it frees me up come mixing time, I don't need to sweat over overs and I have no ceiling stress.

 

Proper gain staging is not only about the levels inside your daw..
https://www.gearslutz.com/board/mastering-forum/1024968-k-system-try-t-system.html

 

Very nice post, just something I want to clear up about this "master fader 0db" thing, it's true. Well it was. It's not since quite a long time now. Originally, this rule was told in the digital world at the 16-24 bit era. Multiple reasons linked to the fact we used 24 bit DAWs bringed this rule. A master fader at -6db would give weaker DR, may lose some bittrate quality, etc...

But then, 32 floating points DAWs came out. Nowadays, most of them are working at 32-bit. The DR is almost twice bigger, very low volume get twice better bittrate, etc... So with 32 bit DAWs, you can actually touch the master fader without damaging the mix (but is it really useful in a mix situation? Doubt it), you can also destroy this 0dbFS limit withing DAW's mixer without any distortion. But again, is it really useful?

So I would say the same as you to struggling producers, dont touch the master fader, as there is tons of better ways to tweak what you want to tweak, don't make your tracks red, as it will F up your balance and other delicate things.

But I just wanted to clear ou the fact that nowadays, our DAWs are capable of getting of those lines.

 

I think it's saturation. Here's why:

Just a couple of days ago i was making a beat. Added some guitar and keys. (all fresh kontakt sounds and banks)
I wanted to work on the keys to make em sound dirty. Added a saturator plug.
Then keys were way too loud. I then lowered the output of the saturator drastically (i.o.w. gainstaging; keys were at -6db before and after saturation) but still the keys felt louder than the rest.

Another session: again making a beat. Played some Kontakt third party sounds. All of them had some kind of processing.
(In other words, in kontakt they added filters, compression and or saturation.)
I then played some drum sounds from my own vinyl collection. (record from 1973 straight from vinyl no further processing)
I had to really crank up the volume of those drums to get them closer to the processed kontakt sounds.

Session three: With just a loop to keep tempo in my DAW i recorded some vocals using my LA2A. I really pushed it to the limit cuz i luv that grit and dirt. I then started to make music using samples from my vinyl coll.
Wow did i have problems getting the vocals to sit in the mix instead of soaring above it. Eventhough the samples were playing at near 0db. And the vocals were still at -6db.

Now i'm no philosopher, or a doctor or a scientist. But it seems to me that the only logical explanation for this is saturation/ compression.
I know it isn't stereo vs mono cuz i make everything mono (or mono compatible) if it isn't already.
And it's certainly not gainstaging in my case cuz i watch that like a hawk.
Simply put, a sound that reaches -6db on your master fader can sound louder then another sound peaking at -1db.

There is ofcourse the possibility that something is going on that i do not understand. If i've learned anything in life it's that i do not know everything.

 

Well yeah but when I first started I didn't even know what clipping was. I half thought alright it's red that means its really loud. I'd want to turnt he volume up and really not know which of the 10 volume knobs on my chain I should change.

 

Thank you very much for all the answers again!

Oh yeah I saw that. I did not watch it yet, but now I did. I have to push myself a little bit to be able to resist that pink noise on max volume through the mixing session, haha. But yeah if it helps definitely worth it. Thank for your very much for your answer and for your reminder!

Thank you for your answer again! By the way I would love to hear what your thoughts are about Voxengo Span compared to K-metering?

If you process the sound with eq and stuff and you make sure that any volume increases or whatever is already compensated with for example the output gain knob. Do you still recommend to use a new utility plugin after it?

Oh Klanghelm. I am hearing so many positive things about it. By the way that VUMT plugin looks really nice. And it's cheap aswell. Do you use it? Anyway also if I am right gain staging can be done with every plugin as long if there is a gain knob?

It helped for sure! I am working with FL Studio. And yeah, that's indeed possible.

I know you can do that in FL Studio through the sampler, but it only will make the sounds already louder by boosting it to 0dB. Altough any reasons why some people are doing that? There is probably a good reason for that.

Are compressors and limiters, etc falling under normalisation aswell? Or is normalisation still different and seperate from this?

Alright thanks!

Thank you aswell for your explanation!

But now you know you will never forget it, right?

Actually one thing about transient shapers I always ask myself is it not doing kinda the same thing as the ADSR envelope? Also how can transient shapers help with lowering RMS? If I am right they are mostly used on drums for example to give them more punch and helping it with cutting though the mix. Just like a compressor? That makes it louder, right? I quess it also depends a little bit on the situation and what you want to do with sounds.

And if it peaks higher than -6. How do you make sure that it's going to sit back on the right volume? By turning down the levels?

Thanks for this! Very interesting article!

If you should not touch the master fader, then do you know what the reason is why the master fader is still there? I only used it a few times when I thought that's the best way for getting headroom really early at the beginning.

That's really normal of course. Nobody knows everything. That's also why you're never old enough to start or learning new things. Anyway I am not sure, but I thought differences between any sounds can be caused by pitch and tone, but also through waveforms and the amount of energy it have aswell.

 

@Nick12

I said "I am aiming for between -18 and -12 on each instrument track (depending on the density of the piece) while trying to keep peaks to no higher than -6. This way it frees me up come mixing time, I don't need to sweat over overs and I have no ceiling stress".

You said, "And if it peaks higher than -6. How do you make sure that it's going to sit back on the right volume? By turning down the levels"?

I say, "The vast majority of what I am recording is live takes so I am observant of my levels as I proceed along from the very beginning. The performance is the key and mic tecnique is really important, be it a vocal, acoustic guitar or percussion track. Anything that has a wide dynamic sway needs be reigned in else it is like trying to herd in a bunch of housecats with coyotes come mix time, shit is just too outta hand and could take too much time trying to temper it all down later on. It is for me much better to print tracks conservatively. I have been using compressors on the way in just to catch whatever pesky peaks, as well as have transformers in the gain path which act upon the signal as a certain kind of compression. If you find the sweetspot where there is just a hint of saturation, that will shave a bit of the transients off the top of the signal, besides doing what I want the transformer to do, how it behaves, it is also pleasing to my ear.

As I mentioned before, and it might sound odd, mic tecnique is very important when recording acoustic guitars (as well as many other instruments such as winds, reeds, brass and others, but chances are probably greater for folks around here to be using guitars more than the others mentiones. After getting the sound that you are after with a microphone with whatever techniques that are available to you (one mic, various different stereo arrays, etc.), the artist just can't really be moving all around the place because with close micing there is a sweetspot that is rather finite and unforgiving of a good capture if they stray from it. If you are depending more on a room mic or mics and using the close just for focus it is not as critical (sort of), though it still will make a difference.

The pizza just came to the table, yum! I hope this helps rather than "hellps".

 

Voxengo SPAN is great! I use it.
But it's a frequency analyzer (with the different K-meters built-in). But Bob Katz K-metering system (mixing loudness) is something different. Read up on it here:
http://ranchstudio.com/student/bob katz levels.pdf

 

My solution was to get one of the acoustic stands for live playing, and use it while mic'ing. Keeps the performer (usually me, lol) from moving the guitar around and leaving the sweet spot.

 

I'm not familiar with FL Studio, so I couldn't tell you why its normalization only allows you to set it to -0 dB (though it's not unheard-of). First try setting the fader on your VI track so that its peaks are hitting around -6 dB, if possible, then bounce the VI's audio to disk. This will at least put you in the ball park. If you have a two-track editor -- let's say Acoustica or whatever -- you should be able to load up that bounced VI's audio track into it and perform a normalization to whatever volume you want. You should at least be able to change the gain on the audio file, so if Acoustica says the file's peak is -4.9 dB, just lower the gain by -1.1 dB, render it, save it and load it back into your FL Studio project.

No. Think of a compressor as a "sound vise". It basically squeezes down on the entire track, so that the difference between your peaks and the softest parts of your audio track are closer together. Once that happens, your louder parts are pushed down closer in volume to your softer parts, which means you have to "make up" that gain by turning the compressor's output up and making the whole track louder. And think of a limiter as a hard ceiling. Instead of squeezing down on peaks, like a compressor does, you're basically just pushing your overall audio signal straight up into the limiter's threshold without going over it. You can literally just keep pushing the audio up in volume until the waveform is a solid bar. Both of these processes are pretty much the exact opposite of what you're trying to do here.

 

Something like this, LOL!

 

Oh, you're recording? Interesting. In the future I would actually like to do that aswell, but then more as sound design. Going outside for example and recording interesting things. Afterwards all editing and stuff. Sounds so nice. But yeah future thoughts. First paying my attention now to my mixes and working and trying to improve them with all the amazing help that I got/getting

Anyway nice explanation! It helped for sure. Thank you!

It makes indeed sense to me aswell about mic placement. It's probably also a lot of trial and error to see what works and what doesn't. Also because every mic and room are different from each other. And even if you want to use multiple mics there is coming a lot more into it, because of any phasing issues that can arise for example. Have you ever experienced something like that? With all the mic placements and the techniques I think there could be a lot of sweetspots with different characteristics and soundings, right? Are your mic placements everytime the same or also still different sometimes?

By the way I assume you're using and recording guitars and if I could do a try to help somewhere you maybe can check out Bias Amp from Positive Grid. It have a transformer aswell if I am right and I heard it's just so great for using it for guitars.

I see the differences now. Thanks!

Thank you for your tips and for giving clearness!

 

you could try normalizing the sources when prepping your mix

 

Sorry to take so long to reply, I literally lost track of the thread...shite...gotta run, I will try to answer after supper, they just came to pick me up....

 

Thank you for your answer!

Oh, no problem. It' s fine. You don' t have to apologize. I am already grateful for your help that you gave me in the first place