Gain Staging in Your DAW Software

Gain Staging In Your DAW Software 9/21/15, 2:33 PM
Gain Staging In Your DAW Software

If you don't understand gain structure, you may be undermining
your recordings and mixes without even realising it.
Matt Houghton
Despite the immense power and

flexibility available in modern digital
audio worstation software, many people
still find that the mixes they craft
entirely 'in the box' sound unsatisfying.
Why is that? There are obviously many
things that can go wrong, and Mike Senior's Mix Mistakes article in SOS
September 2011 (/sos/sep11/articles/mix-mistakes.htm) did a better job than
I have space for here of exploring many common pitfalls. In this article, I'll
focus on one fundamental issue that blights many tracks sent to SOS for help
or advice: poor management of levels throughout the signal chain. In other
words, poor gain staging.
On the face of it, gain staging couldn't be simpler: you ensure that you feed an
appropriate level from the first stage of your signal path to the next, and repeat
this from the second stage to the third... and so on, all the way from your
instruments, mics and preamps to the final stereo mix bus. By 'appropriate',
I mean an ample level, which ensures a healthy signal-to-noise ratio (the
difference between the wanted signal and the noise floor), while leaving
enough headroom that you needn't worry about whether the signal might be
clipping.
I've seen plenty of online discussions in which seasoned engineers are shocked
by the ignorance of many would-be engineers about managing signal levels.
I'm not surprised: to anyone who learned to mix with analogue consoles and
http://www.soundonsound.com/sos/sep13/articles/level-headed.htm Page 1 of 13
outboard gear, this is all second nature, because the gear effectively forced
good practice on you.
Even when an engineer is hitting the maximum level on the meters of

professional analogue gear, there's bags of headroom left to accommodate
stray high-level peaks, or particularly loud passages: if your signal is
measuring 0VU on an analogue meter, then, assuming it's a professional-
standard device, where 0VU = +4dBu, there'll be at least 20dB of headroom,
and often considerably more; the signal can peak well above 0VU and cause no
problems. Even with particularly high peak-to-average ratio sources such as
a loud snare drum, the peak may be 'gracefully' clipped by the analogue
circuitry — in the digital domain, by contrast, clipping produces anharmonic
distortion which sounds very ugly.
Furthermore, at one time, much of the signal processing (EQ and sometimes
compression) was built into consoles, and the levels leaving one stage had been
designed to 'just work' with the next, from a user point of view; you'd have to
try very hard to get unacceptably high levels on the main mix bus. Even where
outboard processors were employed, they would be plumbed in and the meters
calibrated, so that when you were mixing you didn't need to think about
anything but mixing. When cheaper 'project' consoles such as the Mackie 8
Buss came along, there might have been a little less headroom than in the
Neves, APIs and SSLs of this world, but the situation was essentially the same.
In the modern software-based production environment, though, many people

don't completely appreciate what appropriate levels are, or even correctly
understand the notions of headroom and dynamic range. This is
understandable: after all, at least one generation has now grown up trying to
teach themselves mixing using no more than a computer, a piece of software
and the vast resource of conflicting, variable-quality advice on the Web.
Bad advice aside, I believe there are also several reasons why the DAW
software itself leads people to make

mistakes. Partly, I think it's due to the
huge amounts of 'internal headroom'
offered by modern 32- and 64-bit floating-
point software — you could theoretically
describe a dynamic range of a whopping
1500dB in a 32-bit floating-point system
without causing problems. That enables
you to apply serious amounts of gain if
you wish — in theory! In practice, many
people, including me, believe that the When you're using a professional audio
interface, -12dBFS is broadly equivalent to
summing engines in different DAWs don't the nominal signal peak levels on an
always produce the same results when analogue console, so there's no need to let
your DAW's channel levels peak anywhere
summing lots of very high-level signals, near 0dBFS.
and that these differences can be audible.
The probable reason for this, to put it simply, is down to differences in how the
32- or 64-bit floating-point calculations are rounded to create the 24-bit audio
part of the floating-point data.
More commonly, problems arise when you introduce plug-in effects and
processors, because some are just not designed to operate well when presented
with very hot signal levels. I'm not referring exclusively to cheap plug-ins,
either: there are some fantastic-sounding plug-ins by the likes of Waves, Slate
and Sonnox that audibly suffer if you overload them.
Analogue-modelling plug-ins, in particular, seem susceptible to this sort of

abuse: they often do a great job of mimicking the sonic behaviour of their
hardware equivalent in its normal operating range, but start to fall down when
you push them outside that range. My assumption (which I've yet to verify) is
that this is because modelling the increasingly complex non-linearities of
analogue gear as you feed hotter and hotter levels into it takes up so much DSP
power. If the tool is used as intended, there's just no need to model that
behaviour accurately, and users will often prefer a more resource-efficient
plug-in in any case.
Happily, most plug-ins offer input and output level controls, and often meters
too, so you can boost or attenuate at any stage. What this all boils down to is
that, despite the apparently forgiving nature of 32-bit and 64-bit floating-point
processing, you still need to be aware of and manage the levels at every stage of
your mix, and to be aware of the impact these levels have on your software's
sonic performance.
Your DAW's channel meters are of little

practical use when mixing with suﬃcient
headroom. Some plug-ins, including the
pictured one by Zplane, enable you to mimic
the more useful analogue VU and PPM
meters — but there's no practical way to
display these readings for all your mixer
channels simultaneously!
The good news is that all of these problems can be avoided if you work with
audio at sensible levels in the first place! In other words, if you adopt the
analogue approach and leave plenty of headroom right the way through your
digital signal path, everything should be just fine. Meanwhile, if you find that
you need things louder when you're mixing, use the volume knob on your
monitor controller, amp or speakers, because it's perfectly possible to increase
the loudness of a mix during a mastering process, but it's not possible to rescue
an overloaded mix by turning it down. Working in this way, when you find that
you need to raise a channel fader, apply an EQ boost or do whatever other
process that adds level, you have room to do so without needing to think about
it. In fact, that's a good way of looking at this whole subject: gain-staging is
about providing you with a safety buffer.
There's nothing inherent in 32-bit and 64-bit DAW systems that encourages
people to use ridiculously high-level signals. So why do they? Obviously no-
one's told them, or the message hasn't got through, but quite apart from that
I can see some factors that might convince someone that it's a good idea. One
is about adhering to outdated practice, while the others relate to the DAW
software itself.
First, there's the archaic practice of recording levels as hot as possible, in order
to maximise the signal-to-noise ratio of the digitally captured audio. This was
important in the days of 16-bit digital recording, where the dynamic range of
the recording medium was relatively restricted, and when converters didn't
apply dither correctly — or even at all! It's certainly not the case now: DAW
projects pretty much all default to 24-bit recording and I'm not aware of
a modern DAW that doesn't at least offer that option. Digital recordings at 24-
bit resolution offer a greater dynamic range than most users need — ie. the
digital system's noise floor is so far below the wanted signal that you just don't
need to think about it any more. It's easy to achieve an ample signal-to-noise
ratio while leaving bags and bags of headroom. This advice is more commonly
given now, but it's not that long since I heard more than one music production
tutor offer poor advice on this point, so it's well worth repeating!
In case you're unconvinced, let's compare dynamic range in the digital and
analogue realms. A top-notch analogue mixer has a noise floor around -95dBu,
or maybe a little better, and clips at around +24dBu, giving it a total dynamic
range of around 119dB. A typical mid-budget computer interface will deliver
exactly the same dynamic range, and even budget ones will manage 115dB or
more. The very best might make 125dB, which is a little better than most really
good analogue systems. So, since analogue and digital systems have exactly the
same dynamic range, there's no problem in using them in exactly the same
way, with the same headroom margins, to deliver the same kind of sound
quality.
In the case of the DAW software, I can see two major issues. The
first is 'shouty' presets. Load a virtual synth in your DAW and run
a programmed MIDI loop through it with notes set to a velocity of
100. The chances are that the meters are already into or
approaching the red. The sound won't clip its own DAW channel —
so it won't cause problems if you're just using the synth to do a live
performance, for example — but if you run this patch through
some plug-ins, or route multiple such patches through your mix
bus and any mix-bus processing, you can easily run into problems.
The faders
Many people like to start a track with a software drum machine
on most
and overdub other parts as they build a composition and mix, all DAW mixers
as part of the same process. If that's you, lesson number one is to are
designed to
use the output-level control on your software synth to turn it down allow finer
to a sensible level and claim back that headroom! Don't use your control over
gain/attenuation
mixer faders for this, because they won't affect the level flowing at around
from your synth to any insert processors you add. Again, if this their unity-
gain
sounds too quiet, turn up your monitors.
position, so
try to set
Related to this issue is the use of mastered mixes as mix your levels
references. Using references is a great way to train your ears, with the
faders at or
become accustomed to your speakers and help get your mixes into around that
the right sonic ballpark. However, the mastering process strips off position.
the headroom margin that was present during tracking and mixing. Anyone
unaware of this might assume that the aim is to produce a raw mix that hits
0dBFS and has an average level up around -6dBFS, or

whatever! This is most definitely not the case. Lower the
level of your reference CD tracks to regain that headroom
and make them more comparable with your mix level.
The second way in which DAW software can mislead is in

its digital sample-peak metering. That's used by default in
all DAW mixers (although, thankfully, progress is at last
being made in meaningful loudness metering on the
stereo mix bus). The sample-peak meter indicates the
amplitude of the highest audio sample at any moment in
time, and provides an approximation of the actual peak
level of the reconstructed audio waveform. The approximation is perfectly
adequate for use on the stereo bus, or any channel where you plan to send the
signal out into the analogue domain — if you're working with a sensible
headroom margin — since any amplitude errors are non-critical.
I'm not entirely sure why DAWs feature large sample-peak meters on the main
mixer channels at all these days, though, unless you use them while tracking.
They're only really of any use when mixing if you fail to leave sufficient
headroom in the first place, and they provide you with very little useful
information that a couple of LEDs couldn't, yet they take up vast swathes of
screen real-estate! It occurs to me that the sample-peak meter is a throwback
to the early days of digital recording, when, as I've mentioned, it was necessary
to record as 'hot' as possible without clipping — and that no-one has thought to
abandon or replace them since that time.
The point is that this sample-peak metering

is very different from the metering you'd
find on a typical analogue console.
Analogue meters, whether they take the
form of a moving-coil device or a stack of
LEDs, are always 'integrating' types of one
sort or another: they're designed to display
the average level of a signal calculated over
a defined window of time, and the analogue
electrical and mechanical components
mean that they effectively have a built-in Some DAWs allow you to customise your
channel meters. Here, Cubase's channel
'attack time'. They're deliberately designed meters have been configured to show red
not to react to the briefest transients that at -10dBFS, yellow down from -10 to
-18dBFS, and green below that. Aiming for
would register on a sample-peak meter —
yellow will give a good healthy signal while
in the analogue domain, these are not a leaving suﬃcient headroom.
problem, because of all the headroom. The
VU meters of analogue consoles calculate that average over a relatively long
time window (about 300ms), and even though the PPMs beloved of broadcast
professionals use a much shorter window (about 10ms), they still don't display
true peaks, which is why they've often been referred to colloquially as 'fake
peak' or quasi-peak meters. The benefit of both these types of meter — at least
in relation to digital sample-peak meters — is that they provide a more
effective indication of how close or far away the signal is to/from a suitable
nominal level. (Note that this is different from 'loudness' metering of your
stereo bus output, discussed in the TC Electronic LM2 review elsewhere in this
issue.)
Metering within DAWs can also mislead if you're unsure exactly what is being
metered! Your DAW meters default to showing the level at a certain point in
the signal chain, and different DAWs have different defaults: the channel's
input signal (ie. before any insert processing) may be being monitored, or the
signal could be post-panner (and post-inserts) or post-fader. It's entirely

possible for your meters to look OK but for you to be feeding massive levels to
the mix bus — if, for example, you've used compression with make-up gain or
have applied EQ boosts.
Most DAWs allow you to change where the meter signal is tapped from, and
some allow more options than others. What none do, as far as I'm aware, is to
indicate the level between the insert slots. If that's what you want, you'll need
to use the meters on your plug-in processors and effects, or dedicated metering
plug-ins. That's a shame, as you don't have any indication at a glance from
your DAW's mixer screen without having the plug-in GUIs open. I've often
thought it might be useful to have an (optional) input clip light on each DAW
insert slot and at the top of each channel, not as a lazy means of protection, but
as an aid to tracking down problems more quickly.
In short, then, when it comes to metering, the digital sample-peak meters of

your DAW's mixer window tell you very little. If you leave headroom, just as
you would in the analogue domain, you don't really need them at all.
The advice in this article all boils down to

the need to allow headroom on all your
DAW channels, but, due to the difference
in metering types used in the analogue
and digital domains, this can seem more
complex in practice than it sounds. You
could, if you wished, insert virtual VU or
PPM meters on every channel of your mix,
but I'm unaware of a DAW that allows you
to change the channel metering in the When allowing generous headroom, beware
hot signals coming from other sources, such
mixer to an averaging type, so there's no
as operating system alerts, mastered
way that you can see at a glance what all reference tracks and the horrible sound of
those insert meters are doing. We have to full scale noise if your interface stops
work with the tools at our disposal, working. A hardware volume or mute control
really is a must!
although if you're working entirely in the
box you don't need to mimic the analogue signal flow so precisely. You don't
even need to think about the headroom of your interface's A-D or D-A
converters, except in as much as it interfaces with your monitoring chain
(another subject about which you'll find half-baked advice all over the
Internet!). However, you can make use of those almost-useless peak meters.
If you take the sound with the highest peak levels and set it so that it peaks at
between -12 and -18dBFS, you shouldn't run into problems with plug-ins or
summing on the mix bus. If these figures look different to those I discussed in
relation to analogue gear, remember that the meters are different, and you're
actually leaving about the same 20dB headroom. I wouldn't expect to see peaks
on the channel meters reading more than -8 to -10dBFS (and lower is often
better). The highest level track in your mix will vary, but in rock and pop it's
usually the kick or snare. Set a rough balance of your other tracks in relation to
that track and you should be good to go.
There are various means of setting that initial level and which is best will
depend on the DAW you're using. Some allow you to select and
boost/attenuate all the clips in the project's arrange window. I find that a good
tactic when working only with audio files, as all the mixer controls remain
neutral, but it won't take account of any virtual instrument channels. In some
DAWs, there's a dedicated gain control at the top of the console channel strips,
and you can set each track to the desired level, or gang the controls to preserve
the balance while tweaking. In others, you'll find that you need to insert
dedicated gain plug-ins in the first insert slot of each channel, and adjust them
to set an appropriate signal level. In those cases, linking the controls may be
more difficult. In extreme cases, where you have a project in which all the
sources are incredibly hot, you might think about batch processing the files by
lowering all the faders and bouncing the results in place (freeing up an insert
slot in the process), but I wouldn't suggest doing this as part of your routine
mix project setup.
You still need to think about your plug-in chain. Bear in mind that shouty
presets are offered by many plug-in processors: I've observed plenty of decent
plug-ins where the designer has craftily added a dB or two to the output even
when it's not processing anything (we're conditioned to think that louder is
better). Just aim for broadly the same level coming out of a plug-in as is going
into it, and you won't go far wrong. In other words, try to stick close to the
'standard operating level' between plug-ins, which means peak sample-meter
readings of around -10dBFS. You can bypass each plug-in in turn to check
levels at each stage. Given that you have a little headroom, a wee bit of gain
here and there from an EQ boost or something like that won't matter, but don't
limit the bejesus out of a source and apply so much make-up gain that you
need to drag your channel fader right down.
Most DAW faders are designed to have greater 'resolution' around the unity-
gain position (their default position in most project templates), in the sense
that small fader movements in this region produce small gain changes, while
elsewhere in the fader's travel, small movements produce larger gain changes.
(So this is really about manual tactile controllability, not 'mix resolution' as
such.) If you plan on mixing by tweaking faders, either with a control surface
or a mouse, your initial aim is to work towards a static mix balance that leaves
all your faders at or around this unity-gain position — simply because it will
give you much greater control when you need to tweak the faders as the mix
progresses.
To do that, I suggest that when you set your initial channel input levels, you do
so with all your faders at or around the -6dB mark. Why -6dB? Well, as the mix
progresses, you'll often want to nudge the odd track up here or there to reveal
it in more detail at certain spots in the mix. Starting with the fader at -6dB
means that when you apply those tiny boosts, the part you're trying hardest to
draw the listener's attention to has its fader right in the region where it's most
controllable. (If you plan on mixing entirely with mouse-created automation,
this tactic is less critical.)
So far, I've assumed that you're doing all your mixing in the box. However,
many people like to incorporate some outboard gear in their DAW setup,
creating a 'hybrid' system, as some people call it. For example, I'll often use a
hardware compressor on individual channels in my Cubase projects,
employing the compressor as an External FX plug-in. In such scenarios, you
should treat each channel in a similar way as you would your stereo mix bus:
because you're routing your signal through your D-A and A-D converters, you
need to avoid clipping like the plague! Note that your DAW's sample-peak
meters will not catch 'inter-sample peaks,' where the true reconstructed
waveform can reach 3dB or more above 0dBFS between the sample values
(this is why the 'True-Peak' meter — which is an oversampling meter — was
included within the ITU-R BS1770 loudness metering specifications). If you're
only using the project with your own interface, that's less of an issue — because
you should be able to hear any problems — but if you're sharing your project
with bandmates or anyone else, it could cause headaches.
Nothing I've described here is rocket science, and neither is it magic — you still
need to put the hours and years into learning the craft of mixing. But
understanding how to manage your gain structure could make a huge
difference to the quality of your mixes — and avoiding fundamental mistakes
early on in your mixing career will save you countless wasted hours and a lot of
frustration as you learn. .
Most professional converters are designed to produce +24dBu for a 0dBFS

signal, which is the clipping level of most analogue gear. For those who mix
'hot', with average mix levels around -6dBFS, the gear is handling signals that
are around 18dB hotter than it was intended to cope with, which is why it often
sounds hard, brittle and strained. Keeping a 20dB headroom margin in the
DAW avoids that problem and won't compromise the digital noise floor,
because that is still 95dB lower — in other words, it's roughly aligned with the
analogue noise floor. However, note that there's an inherent risk when working
like this. If your DAW's analogue monitoring chain is set up expecting signals
averaging around –20dBFS, it will come as a nasty assault on the ears if you
import and play a mastered CD track peaking 0dBFS. For this reason, as well
as for panic applications if the computer goes into howl-round and freezes,
I urge the use of an analogue monitor controller with easily reachable volume
and mute controls! Hugh Robjohns

Gain Staging in Your DAW Software

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Gain Staging In Your DAW Software 9/21/15, 2:33 PM

Gain Staging In Your DAW Software

Despite the immense power and

Even when an engineer is hitting the maximum level on the meters of

In the modern software-based production environment, though, many people

software itself leads people to make

Analogue-modelling plug-ins, in particular, seem susceptible to this sort of

Your DAW's channel meters are of little

0dBFS and has an average level up around -6dBFS, or

The second way in which DAW software can mislead is in

The point is that this sample-peak metering

signal could be post-panner (and post-inserts) or post-fader. It's entirely

In short, then, when it comes to metering, the digital sample-peak meters of

The advice in this article all boils down to

Most professional converters are designed to produce +24dBu for a 0dBFS

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