Best way to increase track loudness after production?

[Paulie]

From the Taxi TV chat to the forum...

I have several finished tracks that don't pass the "volume/loudness" test. What's the best way to bring up the overall volume, or signal to noise ratio? I still have all of the Logic project files, but Id like to avoid remixing entire tracks if possible. But, if that's the only option, so be it and I can just zero out the faders and dtry again with proper gain staging.

PC

[mojobone]

I had a detailed post on this subject recently, and the internetz ate it, so I'll be happy to take another crack at it. In uh, a...couple hours.

[andygabrys]

Basically limit the crap out of them.

A well (or better) leveled mix with everything well level controlled by using compression, distortion, and limiting on individual tracks is going to have an easier time of being made "loud" (and it's always genre specific too - it can make some sounds sound rather like poo).

But even at that, several stages of gentle (I.e. 1:1.5 ratio compression with appropriate attack and release times) plus appropriate distortion (saturation) plus several stages of limiting each only taking of a db or actual gain reduction can make things silly loud.

Naturally - doing it well is an art to develop.

Over to you Mojo!

[mojobone]

There are basically two ways to increase apparent loudness and the first one is real simple; it's called digital gain change or peak normalization. Peak normalization tells your computer to find the highest peak in your file and raise it by a specified amount relative to OdBFS. Simple, easy, and effective but...when you raise the gain, you'raising everything that's in your file, including low level noise that maybe wasn't so bothersome before, so it's a good idea to try to maximize signal to noise ratio beginning with the tracking stage. This is easy enough if you have a loud instrument, even in a not so quiet recording space, but distant-miking a quiet lap harp in that same room could prove to be a challenge if you have to deal with traffic or cicadas or if your mic or preamp exhibit self-noise. You could record everything direct, but you might find your productions to lack a little 'air' if you do too much of that. You could fix that with some tight artificial ambience from a digital or analog reverb, but that can sound uh, artificial. You can move the mic closer so you hear more of the instrument than the room, but if your mic is cardioid, it will have some proximity effect, which means the bass response will increase as the mic distance decreases. which brings us to method two, compression.

Essentially, a compressor is a tool that reduces dynamic range which is the distance between the loudest and quietest portions of your file. But hold on a second, aren't musical dynamics a good thing to have in a performance and isn't that something we want to preserve, rather than reduce? Well sure, but there's a catch; the listener is not gonna be in a treated room-she might be at the beach or in a car. Shoot, the car windows might be down; your low-level passages might be completely lost! That quiet interlude you sweated over for hours has been reduced to "whooooosh". So yeah, compression, baby; bring it on! But like with proximity effect, almost every kind of compression (and there are many flavors) will affect the tonal balance of the material in some way, usually emphasizing bass at the expense of trebles, depending on the settings, but often compression will also enhance vocal sibilance and string squeaks.


So what's a poor studio rat to do? Well I'll tell you what I do; I strap a compressor across the two bus and mix into it. Because bus compression will affect different instruments/tracks differently, we know it's going to affect the balance of levels, so as soon as I have a rough balance I start compressing and adjust the levels as I go and compensate as needed with some judicious EQ. Now for a little detour, I wanna talk about meters; we have meters that read peaks and we have meters that read averages, labeled RMS which stands for "root' "mean" and "square" the method of averaging the electrical energy in your signal or more properly what that energy is gonna be when your parcel of ones and zeroes gets decoded. I trust you've seen a visual representation of a waveform featuring peaks and valleys, right? The peaks represent transients; they're called that because they're just passing through and they don't last long, but they have a lot of energy; some in fact happen so fast, they can occur between samples-more on that later. These transients are collectively responsible for the punch and presence in your mix, so if you reduce them too much things can quickly start sounding dull, but we want to reduce them so we can raise the overall level; this means the peak levels and the average levels will be closer together. Essentially, we're trading headroom for greater average level and that sounds louder because our ears have a slow integration time; they respond better to changes in average level than to peaks.

Back in the dear old analog days there were so many ways for noise to creep into the recording process that a lot of recordists who maybe had only a paltry few channels of hardware compression available would use them at the tracking stage as well to keep their signals above the murk. Man, I miss the murk; it hid a lotta flaws. And a lot of edits. A lot of that juicy murk had to do with consoles, transformers and channel leakage, but in a well-maintained facility most of it was down to the process and the formulation of magnetic tape; without getting too deep in the nanoWebers, I'll just note that simply passing signal through a tape deck results in a smooth, natural compression with subtle level-dependent saturation, slightly increased second and third-order harmonic distortion and a little EQ bump around 90-100Hz. All this analog sweetening came free with your thirty thousand dollar Studer A80, but nowadays we need about fourteen plugins to do all that, considering that those tape tracks were then passed through a second tape deck when they did the mix and yet another at the mastering stage. These multiple stages of compression worked out great (save for the noise and the murk) because it turns out that working a single compressor too hard almost always sounds bad.

How hard is too hard? Well, the very best, cleanest, most expensive hardware compressors in the world can do about 4.5dB of gain reduction before you start to hear artifacts. (and if your monitor situation ain't up to snuff, you might not hear them and that's no bueno) A common old school solution is to chain together a compressor and a limiter; the compressor raises the average level and the limiter clips off the peaks. Today we also have software compressors, look-ahead limiters and mastering suites with a collection of tools for your enloudening pleasure. Usually you'll have a linear phase parametric EQ, a multiband compressor, (two or more bands of compression separated by one or more crossovers) another linear phase parametric EQ, maybe followed by a limiter or two. There is also a new third method of increasing apparent loudness through psychoacoustic means that I don't really comprehend, but it works and it's called Oxford Inflator.

For further reading, I recommend Mastering Audio by Bob Katz or this free resource.

[elser]

I have two mastering plugins that I like that have both compression and limiting built in. The Slate FG-X is somewhat unique in the way that it increases loudness without absolutely squashing it. And lately I've been using the Lurssen Mastering plugin from IK Multimedia which has more of a distinct characteristic sound but gets the job done quickly and I usually like the way it sounds. Or you can use the stock stuff that comes with Logic, there are several mastering presets that will get you in the ballpark. Or if I'm in a hurry I'll send out a rough and just check the Normalize box in Logic. It won't do any compression or limiting but it will make it louder.

[Paulie]

Great stuff... thanks!!!

[TheElement]

Great question as I have had that same problem recently. Some good answers above.

I'll provide a simple solution that seems to work at least for EDM. I use Cubase so not sure if all DAW's work the same but look for the pre gain on the master buss. it will be set at 0 but you can raise the volume using the pre gain on the master buss. I used to select all the tracks and bring them all up in volume but the pre gain seem to do the same thing. I guess it increases the stereo signal coming into the master buss.

It seems to work for me and I have not noticed an degrade in audio quality. In fact I think things hit a little harder. I increased the pre gain by +3.8db on this track and hitting -3RMS.. I think it sounds ok. https://soundcloud.com/elementproducer/drop-it

You could limit but you might just end up squashing the track and making it sound terrible. definitely if you want to increase RMS try the pre gain.

[Len911]

Because frequency sensitivity is relative to loudness, it's unlikely you might have ideal results. That's kinda the theory of Fletcher-Munson. Wavelab 6 that I have has a "Loudness Normalizer" that makes an effort to compensate, but it isn't perfect because it is more scientific than artful,lol!

Once I found the sweetspot on my headphone volume amp, I don't have as many under or even over mixing loudness, because it's the reference for everything going in or out. When I did, I would whip out the Sonnox Oxford Limiter and go through the presets. Though it's really probably better to just remix, there may not be much that needs adjusting, and because your mixing skills have also progressed, the mix will benefit as well.

[cardell]

I always find this subject a little amusing, to be honest.

We work hard to produce dynamic tracks at 24bit or greater depth....
...then we just squash them.

It's kinda like, "I need this track perfectly in tune...so I can use a chorus effect on it [in order to de-tune it]".

I swear I'm not being sarcastic here, I really find it amusing at times.

Stuart

[Paulie]

I totally agree! I thought you wanted to see a nice peak and valley graphic for your audio files. Now it's basically a solid bar from right to left.