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Copyright © 2014 - 2015 by Mozart & Friends Limited
Written by Marc Mozart
http://www.mixedbymarcmozart.com
Cover and Book design by Peter "Fonty" Albertz
https://www.facebook.com/Fontysound
Editing by Peter "Fonty" Albertz, Marc Mozart, Tim Lochmueller
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write to the publisher, addressed “Attention: Permissions” at the address below.
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whether such errors or omissions result from negligence, accident, or any other cause.
Licensed to flemming pedersen. Email address: flemming.oe.pedersen@gmail.com
THANK YOU
3
As strategically planned this book might look, it started out by many of my friends asking me the following
question, or similar, on a daily base:
„Hey bro, you know I’ve been doing this for a while,
and I’m really happy how my songs turn out, and feel
I’ve come a long way as a producer. But my mixes
suck. Do you have any basic advice how to improve
my mixing?“
The content of this book is what I tell my friends.
In classic Grammy-fashion, I would like to dedicate this
book to friendship, loyalty and the following people:
Andy Henderson, Andy Kostek, Alex Geringas, Artur
D'Assumpção, Chris Edenloff, Hazel Ward, Ina Sentner, ISC crew, Jack Ponti, John Brandt, Jos FungJorgensen, Kyle Kraszewski, Manager Tools, Mattia
Sartori (forum.sslmixed.com), Michael Dühr, Nick
Hepfer, Ömür Akman, Val del Prete, Swen Grabowski,
Terance Amalanathan, Thanh Bui, Tim Lochmueller,
Ulrich Mizerski, Vincent "Beatzarre" Stein & Konstantin Scherer
The amazing graphics design and logo for this project
was done by Peter "Fonty" Albertz, whose work and dedication catapulted him into my personal „book of legends“ in no time.
Finally, my incredible family for putting up with me
for such a long time.
Marc Mozart, January 2015
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01
02
03
04
05
Monitoring, loudness and your ears
•
•
•
•
•
Steal mom’s kitchen radio!
From small window to magnifying glass
Listening Levels
Switching Speakers
Essential Monitoring Setup
Creating the “good enough” mix room
•
•
•
•
Room Acoustics – Ghetto Style
Speaker Setup, Absorbers & Bass Traps
Improving the low-end of your room
The 10-minute room test
Preparing your Mix Session
•
•
•
•
•
Micro- & Macro-Management of your mix
Building a DAW-template
Importing the files
The concept of "handles" and complete control
References and A/Bing
The Magic of the 1st listen
•
•
•
•
•
The 1st listen experience
Mixing for someone else
1st listen - emotional
2nd listen - analytical
Mixing your own song
We are mixing! The Foundation,
Bass and Gain Staging
•
•
•
•
•
Low-end analysis and balance
Shaping Kicks, mixing bass
Drum Replacement
Side Chaining
Gain Staging
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06
07
08
09
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It’s all about the Vocals
•
•
•
•
•
A quick excusion into vocal recording
Building the Lead Vocal-chain
Parallel Compression
Vox against the world
Backing Vocals
Continuity
• Correcting levels pre plug-in chain
• Careful Limiting
• Nulling your faders
Parallel Processing
• Blending clean and compressed sounds
• How to check if your DAW software can do this
• Parallel Processing on vocals and drums
Colors, Dimension + the Dynamic of your Mix
•
•
•
•
•
EQing: Pultec-Type EQs, Console EQs, Liniar Phase
A different look at compressors
Magic Chains, Reverbs, Delays, Modulation
Stereo Bus magic?
Automation
Stems, Mastering and Delivery
•
•
•
•
•
Client Feedback on your Mix: ready for anything!
Deadlines: how to survive them
Stems, Mastering and Delivery
Client: „There’s one more thing…“
Facebook Support Group
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CHAPTER 01
MONITORING,
LOUDNESS AND
YOUR EARS
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CHAPTER 01
MONITORING,
LOUDNESS AND
YOUR EARS
D
iscussions around studio monitor speakers should
be added to the unholy trinity of conversation topics (sex, politics, religion). Most people swear by the pair
they use, and roll their eyes upon listening to anything
else.
Let’s just start by saying that you probably all have a
„standard“ pair of nearfield studio monitors. Most likely
some type of Yamahas (with the iconic white woofer), or
Dynaudio, KRK, Adam, Focal, Genelec, Tannoy, Mackie.
You have learned to deal with them, more or less. And
you won’t hear a recommendation from me at this point.
None of them are „perfect“ - for mixing, nearfields are
just one of several reference points. What I’m proposing
is that you add another pair of monitors, and not only
will those be the most important monitors you will ever
own, they will also help you to draw better conclusions
from what you hear on your pair of nearfields.
„Another pair of monitors?
Oh shit - thats expensive, right?“
No, it’s not. Maybe 50 bucks on eBay. Not more. What
you need to add is a very honest small portable - „honest“ meaning from the time before manufacturers started using „psychoacoustic“ digital gimmicks from „super
mega bass“ to „maxx bass“. There are thousands of different models that would fit the bill, and you might have
to try a few.
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Actually,
look
at
what mom is listening to in the kitchen.
And steal it.
If mom insists on keeping the kitchen radio where it
is, there’s a good chance you can get one for cheap
at a flea market/garage sale.
Here’s the profile for your new monitor system:
•
•
•
•
•
•
•
•
•
•
1990s portable audio system, can be mono or
stereo
not the „super bass“-type breakdance ghettoblaster…
single woofer, no ported speaker design (1-way)
keep EQ flat
switch off „bass boost“ or similar
smaller = better
mono or stereo, both works
small enough to fit in a 19“-rack
external AUX input (1/4 Inch-jack or RCA)
less than $ 50 on eBay
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EXAMPLES FOR
SPECIFIC MODELS:
I keep getting e-mails asking about specific models
that fit the bill. But really, go on eBay, buy a few and
test them. I use a Sony ZS-D7 that cost me €40 and
does a great job, but there are many other types and
brands.
Some of you might have heard about the Sony ZSM1 that Chris Lord-Alge uses - these are getting
super expensive now and are almost impossible to
find - I think Chris bought them all.
Sony ZS-M1 (impossible to find, CLA bought them
"Tivoli Audio"
(entire range - expensive hipster kitchen radios)
http://tivoliaudio.de
"clones of Tivoli Audio aka „kitchen radio“
(Aldi, Lidl discounters sell these on a regular base)
The old masters of engineering used the original
"Auratone“-speakers for similar purposes, and there
are a bunch of Aurotone-clones around
(active/passive, and even some DIY-plans floating around the web). Don’t get too nerdy on this
though…
http://www.trustmeimascientist.
com/2012/02/06/auratone-avantone-behritone-review/
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You: „nothing to worry for me, I’m al
ready sorted - I always listen to my
mixes on a pair of computer speakers!!“
No! Not modern computer speakers or
iPod/iPhone-dock speakers!!
You: „why not?“
•
The inexpensive ones are generally lacking in quality.
•
The great ones (like B&W Zeppelin) don’t have that 1-way
speaker „kitchen radio“ vibe.
•
many use psychoacoustic technologies that make your mix
sound better than it actually is
( boosting bass and treble, fake
stereo width, etc.)
•
most involve a cheap frequency
crossover (separating tweeter
and woofer information), which
at this pricepoint, introduces
bad phasing and comb filtering.
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I WANT YOU
TO MIX LIKE
SOMEBODY
WALKING BY
Here’s what I recommend:
monitor on the portable as long as you can, at the
lowest possible volume.
You will find that when you’ve build your mix on
the portable, then later on switch to the Nearfields,
you have far less to correct than when you do it
the other way around.
ONLY use the Nearfields when absolutely necessary.
The difference is similar to being ON a stage with
a band versus walking by along the street seeing a
band play from a distance.
I want you to mix like somebody walking by.
OVERVIEW:
MONITOR SPEAKERS - FROM SMALL
WINDOW TO MAGNIFYING GLASS.
•
•
A.
Different size studio monitors are tools for SPECIFIC
purposes.
the following assumes listening levels that don’t require you to raise your voice for a conversation
1990S PORTABLE
AKA MOMS KITCHEN RADIO
Your „window to the world“ - this is how most people
will hear your mix.
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+++ POSITIVES +++
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13
•
great for all of the big decisions and the overall balance
•
good for 8+ hours of mixing/day without experiencing ear fatigue
•
much less susceptible to room acoustic
problems (simply because they don’t have
extended low-end)
•
use 50% of your mixing timeNote: place these
on the side or behind you
- - - NEGATIVES - - •
not suitable for adjusting or revealing lowend issues (under 60 Hz)
•
not suitable for dialing in the final amount
of top-end you want in your mix (12kHz and
higher)
NOTES
•
place them behind or to the side of your
sweet spot
•
check how full and loud the best mixes by
worldclass mix engineers sound on this
•
keep switching back and forth between a library of references and your own mix
•
find out where your own mix stands relative
to top quality mixes. (we’ll be talking more about mix references in Chapter 3)
I want to point out that once you decided for a certain
model of portable/kitchen radio, you have to listen to
a lot of reference mixes on it and just spend time with
them. I'm listening to a lot of chart stuff on my portable
and know exactly how my references sound on it.
It can take a few weeks to really dial in on a system,
and it takes longer, the more expensive and bigger the
speakers (as they are more complex to understand).
MOM’S KITCHEN RADIO SPECIAL TIP:
1.
Listen to some proven famous reference mixes
and turn volume until it collapses (distorts)
2.
Compare how much you can turn your own
mixes up until they distort.
3.
Adjust! (If your mix allows for more undistorted
loudness, it doesn’t have enough low-end)
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B.
STANDARD NEARFIELD SPEAKERS
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15
•
what most musicians are used to work with
•
many people assume that maximum linearity and
neutrality would be ideal for those.
•
actually, it doesn’t matter when they’re a bit coloured.
•
the classic Yamaha NS-10Ms have agressive high
mids that put vocals in the spotlight and worked
well for engineers over the last 30 years.
•
what you chose as your main nearfield speakers is a
subjective choice.
•
many people have two different types of nearfields,
or add a subwoofer to them.
+ + + POSITIVES + + +
•
•
good for 4+ hours of mixing/day without experiencing ear fatigue
use 30% of your mixing time
1
C.
FULL RANGE AUDIOPHILE
HIGH-END SPEAKERS
•
I wouldn’t say that this third pair of monitors is optional, but they add considerable cost to your setup.
•
These are not speakers you can carry around
in a flightcase, or setup quickly by placing them on the meterbridge of the console.
•
your „magnifying glass“ - unveals tiny details difficult to spot on the smaller systems
•
not suited for more than 1 hour of listening
•
use 10% of your mixing time
They’re either
•
flush-mounted speakers (build into the wall) that
have been part of the studio’s acoustic design
•
or free-standing floor-speakers
16
+ + + POSITIVES + + +
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•
great for EQing sources, surgical finding and
removing unwanted resonances
•
judging and adjusting low-end balance
•
finetuning
- - - NEGATIVES - - •
not suitable for starting a mix (you will rely
too much on a low-end that doesn’t work in
the real world)
•
not representative for what 99 % of people
listen on
LISTENING LEVELS
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„Listening at loud levels is a form of drug consumption.“
• listen at the lowest possible level (example: you can
still hear the fan-noises of your external harddrives)
• you should always be able to have a conversation without having to speak up while the music is playing
• at these levels you will be able to do 8 hours of mixing
without a long break
• you will find that the mixes you create at these levels
sound full and huge at loud levels
Warning:
• the more you start turning levels up, the more your
judgement will be clouded
• listening at loud levels is a form of drug consumption:
the joy of loud levels overrides your ability to objectively
judge the mix
Disclaimer:
Keep in mind: we are talking about mixing here - I’m
totally aware that during production, „beat-making“,
songwriting, arranging etc. many of you are inspired by
loud levels, and there is nothing wrong with that.
Mixing is not creating though - we’re trying to improve
the sonics of something that has already been created,
and to make sure it translates to a wide variety of situations.
There’s no better feeling than turning up a mix that has
been created at low levels - I know you’ll be tempted.
Resist it!
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When listening at loud levels works:
• five minutes before you have a break
• at the end of your work-day
SWITCHING BETWEEN SPEAKERS
Remember what I’ve said above - divide mixing time as
follows:
50%
moms kitchen radio
40%
nearfields
10%
full range audiophile speakers
1
•
that said, switch back and forth between the different systems A LOT
•
you need to be able to A/B/C-switch between these
3 sets of speakers
•
they need to be level-matched (= when you switch
the speaker set, the perceived level remains exactly
the same)
•
a „dim“-functionality that allows to switch between
two pre-set listening levels is very helpful
Traditionally, in professional studios this is something
the Console Centre Section of a large format console
does.
20
1
•
shocking, but DAW manufacturers have forgotten to
include the functionality of a console monitor section, although there are a few exceptions
•
there are a number of „monitor controllers“ on the
market in all price ranges
•
a small mixing console can also do this job, note
that used recording consoles can be a bargain these
days
21
If you don’t have a console or monitor controller, passive speaker switchers are inexpensive and do the job.
ESSENTIAL MONITORING SETUP
The minimum recommended setup can switch between
two speakers:
1.
your main nearfield speakers, optionally with a
subwoofer
2.
a small portable stereo or kitchen radio with aux input (one way speakers, no speaker ports, traditionally studio have been using the Aurotones or a small
Sony Portable)
An extended setup would utilize 3 or more sets of speakers.
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FAQ
1
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Why place the portable / kitchen radio off to the side
or behind you as opposed to in front of you?
It's a philosophical thing - the portable shouldn’t be in
your mental focus. The mix needs to catch your attention on semi-muddy speakers that are not pointing to
your ears directly. Thats the whole point of the little
kitchen radio in an odd place.
Good Mixing translates the production for the consumer.
As I said in this chapter, I want you to mix like somebody
"walking by“, not like you’re the performer on stage. We
need to stay very objective and use many tricks to not
get involved too deeply. Even more important when we
are mixing your own production or song.
That said, once you got the basics sorted out on the
portable, by all means move on and get the subs working! More on that in a later chapter.
Do you usually start your mix listening on the portable?
During the first 50% of the mix, I do anything on the
portable that doesn’t require a „magnifying glass“ into
the low-end (below 60Hz) or high-end. Later on, more
nearfield and full-range, but even when you end up
much less than 50%, make sure to always keep coming
back to the portable.
How do you deal with „ear/mind fatigue“?
Take a walk, breaks are important. Leave the studio environment and reset your brain as often as possible - for
me every two hours, up the point when the mix really
starts coming together and I’ll dance around the room
like a mad man.
I've always done my mixes at quiet levels but never
understood why, aside from listening/ear fatigue.
What is the science behind it?
Room acoustics are linear, frequency curves of a room
stay the same regardless the levels. The difference is in
the perception.
By mixing at low levels you have a competitive advantage - you want the consumer to identify how great your
song/mix is even when they hear it somewhere on the
street, in the kitchen, in the background at a restaurant,
at any level. You mix from a consumer perspective. It
makes you focus on the things that are important in the
mix. Rarely do we get the chance to sit a consumer between two great speakers that are fully turned up.
It's a real challenge to get a kick to have punch and attack when listening at a low level, and to level the vocals consistently so you can understand every word. But
once you get these right on the portable, you have a real
winner in your hand.
Aurotones and their clones - should I buy one and use
them in mono, or get two for stereo?
both works - and if you buy two still keep them very
close together, so you can test your panning/stereo effect decisions in a narrow window.
What about listening in your car?
Listening in your car is worth talking about. As many of
you already know - it works very well for checking mixes.
1. if you’re listening to music in your car a lot, you know
how it’s supposed to sound and will spot problems in
your mix instantly
2. you’re listening outside your studio environment, and
similar to consumer habits
3. cars are pretty good listening rooms, even from a
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room acoustics perspective (no parallel walls to begin
with)
4. probably the highest quality listening for consumers,
considering Hi-Fi culture is not mainstream any more (as
it was in the 80s)
Do you recommend having fixed listening levels and
how would you achieve that?
I personally use the DIM button to switch between lower
and higher listening levels. It's great to be able to program the relative DIM level.
My general recommendation is to listen as low as possible, for as long as possible.
I wouldn’t go OCD over measuring exact listening levels.
That said, here’s an article where Bruce Swedien touches on the topic:
http://www.prosoundweb.com/article/print/make_
mine_music_part_2
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2
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CHAPTER 02
CREATING THE
“GOOD ENOUGH”
MIX ROOM
CHAPTER 02
CREATING THE
“GOOD ENOUGH”
MIX ROOM
H
ere comes a HUGE DISCLAIMER at the beginning
of this chapter: this book is focussed on mixing, not
room acoustics. Room acoustics is a complex science,
and if you are planning to get nerdy over designing a
perfect mix room, and you’re prepared to stop making
music for a year, or you have put $ 50.000 aside for that
purpose, then this chapter will not teach you all of what
you need to know!
Just know that professional studio design starts with
building a room that has mathematically perfect room
ratios, so its frequency response and reverb times are
100% predictable.
That said, what you will learn in this chapter is totally aligned with what a professional studio and room
acoustics consultant will tell you.
2
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NOW THAT WE GOT THAT OUT OF
THE WAY – HERE WE GO:
•
Lack of Acoustic Room Treatment is the single biggest cause of frustration during the mixing process
•
Records are made inexpensive at people’s homes
today, and with the music industry on a “Titanic”course, this won’t change any time soon.
•
The majority of producers are set up in an untreated
small or midsize bed- or living room.
ROOM ACOUSTICS - GHET TO ST YLE
The TOTAL COST of a Basic Room Treatment I’m discussing in this lesson is $ 300. And I bet you it easily
beats many $ 1000 pre-made room treatment set you
can buy. This article is not gonna give you an in-depth
theory-lesson on room acoustics – instead I will show
you the cheapest way to give your mix room a massive
improvement – without breaking the bank.
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FIRST RULE
2
29
The speakers and you have to build an equilateral triangle. If possible, face the small side of the room, and
set the speakers up so that the distance to the left and
right side-walls are equal. Self explaining, right? I’m sure
you’ve seen this before!
EARLY REFLECTIONS FROM THE SIDES
This picture explains the problem best… right side is
treated, left side is untreated:

Ideally, we only want the direct sound that comes from
the speakers (-> green arrows) to reach our ears.
But unfortunately, some of the sound goes to the sidewall and bounces back from the wall to your ear (-> red
arrows), and that journey adds a millisecond of a delay to the direct signal, which causes some nasty phase
cancellations – NOT GOOD!
The graphic shows an „early reflection panel“ mounted
on the right hand side to the listener (producer/engineer), and you can see how it doesn’t reflect back anymore (-> yellow arrows).
SOLUTION
A relatively basic early reflection panel does the job. This
could be a self-made wooden frame covered with cloth,
and some Rockwool behind (or Owens Corning 703 fiberglass boards).
Note: this is NOT, and doesn’t have to be a Bass Trap.
One layer of Rockwool is sufficient.
$ 20 half a pack of rockwool
$ 20 squared timber
$ 5 cloth
$ 5 misc.
–––––
$ 50
EARLY REFLECTIONS FROM THE CEILING
Yep, the ceiling has the same effect as the side-walls.
Sound bounces back from the ceiling and mixes with the
direct signal when arriving at your ear.
That's why we put another absorber above your head.
2
30
It’s called an “Acoustic Ceiling Cloud”.
2
31
For my own mix room I have built a 3 x 3 meter ceiling
cloud made of a timber frame, covered with cloth, and
adding a layer of Rockwool on the back of it.
I made it that large mainly because the console underneath is almost 4 meters wide. The one thing you need
to be extremely careful with, is assuring that the ceiling
cloud can’t fall down on your head. In my own room we
drilled 6 massive hooks into the structure of the ceiling
and used them to attach the ceiling cloud.
The cloud in your own room doesn’t have to be that
large - if it covers your listening area above the speakers,
it will do the job.
Breakdown of the material list would be similar to the
two Absorbers:
$
$
$
$
20 half a pack of rockwool
20 squared timber
5 cloth
5 misc.
–––––
$ 50
IMPROVING THE LOW-END OF YOUR ROOM
This is easy;
•
make sure you have at least 50 centimeters of free
space between the wall you’re looking at, and your
equipment/speakers
•
stack 4 large packs of Rockwool in each corner (8
packs total)
•
find a way to secure the Rockwool-packs, so they
can’t fall on your equipment
•
if you (or your spouse) don’t like the look, mount a
curtain to cover them
8 packs of Rockwool are about $ 200, which adds the
total cost of our Ghetto-Style room treatment up to
roughly $ 300.
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THE FINAL PL AN FOR YOUR N EW ROOM
2
33
Optionally, as you can see in the drawing, you can plaster the entire back wall with Rockwool. It will add a few
hundred $ more to your bill and can further help to
smooth out the low-end in your room.
As a rule of thumb, a minimum of 20% of the 5 treatable
room surfaces should be covered.
THE 10-MINUTE ROOM TEST
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Here’s a link to a YouTube-clip with test tones I’ve created with Logic Pro – you can playback this clip to do
the test I’m describing below, if you want to skip setting
this up in your DAW.
Be careful – as this clip starts with subsonic frequencies
you won’t hear in most rooms!
In about 47 seconds you will learn something about
your room. Listen to see if the volume of the test tone is
perfectly consistent.
http://youtu.be/8Olibnhhm7A
1.
Find a test tone generator in your DAW software.
Most DAWs come with an oscillator or test tone
generator for creating a basic sine-tone (if in doubt,
google „test tone generator daw“ + the name of
your software). The frequency of the oscillator can
be set. (You can also use a synth playing a sinewave, THIS LIST shows the range of notes needed)
2.
Turn the volume of your speakers fully down. to not
destroy the speakers or your ears, as test tone generators can produce some nasty high tones! (which
we don’t need for what we’re doing today)
3.
Set the test tone generator (or synth) up.In Logic
Pro, for example, the oscillator can be inserted as a
plug-in, in any track or even output. Set the output
level of the test tone to -18dB and start with a frequency of 100Hz (G2 on a keyboard).
4.
Turn your speakers up slowly, until you can just hear
the low 100Hz tone clearly. Do not turn it up too
much – at a low volume it will be easier for you to
notice changes in level, which will be important.
5.
6.
Bring the frequency of the oscillator down, slowly –
step by step. Take your time, change the frequency
gradually from 100Hz down to the lowest available
frequency, but take a minute to do that. Notice how
the level of the tone keeps changing? On some
frequencies the signal might seem to disappear
completely, on others it’s louder than the original
100Hz tone. If you are listening on small speakers,
the tone will disappear completely when you reach
a certain frequency, this could be around 40/50Hz
if you don’t have a subwoofer. Keep going down to
perhaps 20Hz, even if you don’t hear the tone any
more.
Take notes of what you hear. I’m now explaining
you how to write down what we hear. Finish reading the rest of this article before you start. It’s quick
and easy but there are a few steps involved. Starting with a test tone of 20Hz, we will gradually move
up the frequencies – and make notes about how we
hear the volume changing. We take notes using the
simple categories
“NO TONE”
– you don’t hear a test tone
“LOW”
– you can hear it, but it’s lower than the average
“NORMAL”
– the average level of the test tone throughout all frequencies
“HIGH”
– when it’s louder than average
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Check out the chart below. (the left arrow going up
shows the level you are hearing, the arrow in the middle
going to the right is for the frequency of the test tone,
starting at 20Hz, going up to 293Hz)

You can draw this chart in 2 min or print it from here:
http://i1.wp.com/www.mixedbymarcmozart.com/
wp-content/uploads/2014/09/room_measurement_
LQ.jpg
Start going up with the frequency from 20Hz all the way
to 300Hz making corresponding little crosses on your
printed chart.
The only slight challenge is that you have to look at your
computer screen to see the current test tone frequency
displayed in the test tone generator, while making a subjective judgement of the perceived volume.
Here is an example of how the chart of my own test
looked:

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36
(for example, at 20Hz I heard “NO TONE” hence the
cross on the bottom left, at 130Hz the tone was really
“HIGH”… you get the idea)
BTW, just in case you are wondering why we won’t go
above 300Hz… the low end spectrum is most important
for the overall balance of your room, and it takes a lot
more effort to fix this, while getting the mids and treble right (in comparison) is almost a trivial task when it
comes to room treatment.
7.
Once you’re done, connect the dots, and voilà –
here’s the low end frequency curve of your room!!

Note, this is the frequency response at your listening position. The measurement curve can look different by changing your listening position even a
few inches. Also, the graph doesn’t tell you anything
about the reverb-time of the room which is equally
important (we’ll measure that in the next part). For
the graph above I did the test in my home office,
which is a completely untreated and small square
room with a pair of cheap active computer speakers.
The results are really bad, the peak at 130Hz as well
at the notch around 207Hz would make this room a
serious headache to mix in.
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ADDENDUM
ROOM MEASUREMENT TOOLS
•
•
REW Room EQ Wizard (free room acoustics software; PC/Mac, Java-based)
FuzzMeasure Pro Audio and Acoustical Measurement Application (Mac)
MANUFACTURERS
•
•
•
ROCKWOOL (Insulation Products)
REALTRAPS (Bass Traps, Diffusers, Acoustics Articles/Videos)
GIK ACOUSTICS (Bass Traps, Diffusers, Acoustics Articles/Videos)
FORUMS
•
•
•
Gearslutz Studio Building / Acoustics Sub-Forum
The Acoustics Forum at PRW (moderated by Thomas Jouanjean/Northward Acoustics)
Recording Studio Design Forum (by John Sayers)
STUDIO DESIGNERS
•
John H. Brandt Acoustic Designs
ROOM ACOUSTICS THEORY
•
Wikipedia Article
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FAQ
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39
Room Acoustics are linear.
The laws of physics are stable. While - as described in
lesson 1 - mixing at low levels is highly recommended,
it does NOT take the characteristics of your room out
of the equation. The difference you might perceive is
rather related to the Fletcher-Munson curves, aka the
ISO certified Equal-loudness contour.
http://en.wikipedia.org/wiki/Fletcher–Munson_curves
http://en.wikipedia.org/wiki/Equal-loudness_contour
You can’t have „too much“ bass trapping.
More bass trapping does NOT reduce the level of bass,
but it makes it more balanced and reduces it’s reverb
time. Which is always a good thing.
Bass Trapping is most efficient in room corners. Note:
there are 12 corners in each room.
Please be aware that when looking at commercial studios, the „walls“ that you see are often fake walls made
of fabric or wood slats with gaps. Behind those, a lot
more bass trapping might be going on. In order for subsonic frequencies to not bounce back from a wall, you
need rock wool or hanging bats several foot deep from
floor to ceiling.
Health Considerations
Glasswool is made from sand. Rockwool from basalt. It
may be irritating while you install it in your room, but it
is NOT carcinogenic. The binder used in each formulation may have some toxicity but if you air out the stuff
before you put it in your room it won't affect you. If you
have allergies or are asthmatic, take precautions - cover
well when working with it. If you want to ensure that no
fibers escape from the bass traps you build, first apply
one layer of ticking fabric before the 'dress' covering of
burlap or your choice of material.
The fabric that you choose must have a very low gas
flow resistivity (GFR). Place the cloth you wish to use
over your face and if you can breathe comfortably - use
it. If you have difficulty breathing through the fabric,
don't use it.
Buying ready-made acoustic panels/bass-traps
Whatever ready-made acoustic panels or bass traps you
buy, make sure to find some testing data from an independent lab. Testing labs use standardized procedures
so that you can compare different materials with each
other.
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42
CHAPTER 03
PREPARING YOUR
MIX SESSION
CHAPTER 03
PREPARING YOUR
MIX SESSION
M
ixing is 80% preparation and 20% inspiration: there
is a good reason why I made this the subtitle of this
book. However, mix preparation is NOT mixing. Ideally,
it is a process separate from the actual mixing. Many top
engineers delegate this to their assistants, to be able to
start the mix with fresh ears.
It's NOT a trivial task - those assistants working with the
world’s best mix engineers are top engineers in their
own league!
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WHAT IS MIX PREPARATION?
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Mix preparation is drawing the final line between the
definite end of production, and the start of the mix, and
has two primary objectives:
1.
Micro-Management: to check the individual audiotracks of the production „under the microscope“
and fix obvious flaws
2.
Macro-Management: to make the mix manageable
example: song structure
example: tracks organized by colors and instrument
groups
MIX PREPARATION - BASIC FIRST STEPS
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Before we go into the details - here are the basic first
steps:
•
downloading the files: files will usually be supplied
through a download-link or a shared web-space
like Dropbox, Google Drive, OneDrive, box.com or
similar. Do not worry if the files comes as a .zip or
.rar-file. Those compress the size of the files into a
smaller package, but once decompressed, the audio
files are the same as the original.
•
ALWAYS import the audio files of the production
into an entirely new session
•
work in 24bit (or higher) and in the sample-rate that
was used by the producer, most commonly 44.1 or
48kHz
•
you may get files in a „high sample rate“ (88.2, 96,
176.4, 192 or even up to 384kHz), it all depends on
the CPU-Power of your computer
•
if in doubt, convert down to 44.1kHz, as constant
„CPU Overload“ messages will seriously disrupt your
workflow
MIX PREPARATION - MICRO MANAGEMENT
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This first part of Mix Preparation only takes a lot of time
if the producer hasn’t done his homework - it’s easy for
the producer to do this at his end of the production, as
part of printing the individual track files.
Oh, maybe you are the producer. We will have some
thoughts on mixing your own production in Chapter 4,
but for now just keep in mind that it is all about making
production and mix two completely separate jobs.
The tasks involved here are overlapping with what many
producers do when they print their files for mixing:
•
removing unwanted noises and clicks
•
if necessary, replacing clicks and transitions between noise/silence and signal with little fades or
crossfades.
•
removing low-end rumble with a High Pass Filter.
Be careful not to cut too high, and use a linear phase
EQ. An analyzer can help to visually show you what’s
useable signal, und whats just useless low-end rumble.
•
fix obvious flaws related to tuning and timing, unless they are intended. Keep in mind that there is
a fine line between fixing a flat or sharp note, and
overdoing tuning.
•
you can do some very broad EQ-ing at this stage, if
you feel it’s necessary - just as an example, tracks of
Electric Guitars, Piano and Bass Guitar typically lack
High Frequencies. If you feel confident about doing broad EQ strokes at this point - go for it. Don’t
touch EQ of drums or vocals though. They’re too
complex to make broad decisions on at this point.
•
every now and then I receive vocals that are severely over-compressed and/or over-EQ’d, and you
can always spot that in the quality of the „Esses“ aka
sibilance. If the vocals you have to start with already
have a problem in this area, they need extra attention
•
print all the changes you’ve made here to a new audio file, and re-import it
•
if you find ways to reduce the numbers of tracks
you have to deal with: do it! Example: convert two
mono backing vocals into a stereo-file hard-panned
L/R.
CONCEPTS FOR FIXING VOCALS THAT SEEM
HOPELESS:
Fixing a vocal with excessive Sibilance (Esses)
The problem is often that when the lead vocal has already been compressed during tracking, and has excessive sibilance, a DeEsser can’t reliably detect the Esses
in the recording
1. first create different versions of the vocal-track you
want to rescue using a De-Esser, next print the DeEssed
versions to new files as follows:
A: untreated
B: light De-Essing applied
C: extreme De-Essing applied
2. Stack the audio-files as tracks underneath each other
in your DAW, keep all the tracks selected and cut them
at the same time to separate the Esses out.
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You can see in the waveform-graphics where an „Ess“
starts. Always try to cut at a point where the waveform
crosses the „zero“-line in the middle.
3. Select the „Ess“ that sounds most natural to you, and
move it in place to the original one.
DE-COMPRESS VOCALS
Yes, you can bring back lost dynamics into over-compressed vocals. You simply lower the parts in the audiofile that are too loud.
Depending on what your DAW offers, there are two
ways:
A. Cutting between syllables is an option when your
DAW offers an individual volume/gain setting for each
audio „snippet“. Of course you have to create very short
crossfades between the snippets, to avoid clicks.
B. Automation nodes are available in most programs.
Where in example A you would just make a cut, you will
have to create two automation nodes.
This technique for de-compression can of course be
used for any type of recording, not just vocals
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MIX PREPARATION - MACRO MANAGEMENT
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There is a presumption amongst many musicians that
mixing is a very complicated process. I guess this stems
from pictures of massive studios with 80 Channel mixing-consoles, or the mental overload created by a DAWsession that has 200+ tracks.
However many tracks the source-material has, the objective is to reduce the DAW session to a size that allows
us to make quick and intuitive decisions. As soon as the
creative mixing process starts, time works against us.
This is because objectivity disappears the more we get
emotionally attached to the song we work on. You can’t
just mess about for hours and not make progress.
DAWs have turned audio material, tracks, and many
more aspects of a production into visuals. This can be
both useful and distracting.
When it comes to getting organized, a good visual representation of your song structure and the type of audiotracks in the song is very useful.
On the other hand, 200 tracks can turn into a nightmare
if you are not properly organizing the material in your
DAW.
Let me repeat: this is a process for which the best mix
engineers in the world use their (world class) assistants. I
do of course not assume you have one of those at hand,
and my recommendation is to do this a day before the
actual mix. At the very least schedule a break after. It can
be done in 10 minutes.
BUILDING A DAW TEMPLATE
Once you have done the process of mix preparation a
couple of times, you should start creating a template
that you use every time you start a mix. The template
will have things set up that are pretty much the same
for all your mixes, from the way you graphically display
the songs structure along with the tracks, subgroups or
VCA-groups already configured, colors for these groups
assigned, to aux sends and returns for your most commonly used FX pre-configured.
LET ME GUIDE YOU THROUGH IT.
• you need to create quick access to the structure of
the song: a template can have a standard song structure
already set up, so you can move pre-labeled markers
„You can’t just mess about for hours and
not make progress.“
around (INTRO, VERSE, BRIDGE, CHORUS, etc.), once
the audio is imported. In Logic Pro, the song position of
those markers can directly be navigated to via the Keyboards numerical keypad. Extremely useful during mix.
• setup basic routing and grouping: you can route
groups of instruments (e.g. drums) to the same audio
subgroup, and same DAWs offer VCA faders - I personally route groups of instruments to dedicated inputs on
my SSL console, and group them using the VCA grouping on the console.
• setup a selection of your most important FX sends/
returns
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Here’s an example:
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Send 1
Standard Reverb (whatever you’ve used for years)
Send 2
Standard Delay (1/2 or 1/4 Note)
Send 3
Chorus, HPF at 300Hz
Send 4
Small Room
Send 5
Mid Size Concert Hall
Send 6
Long Church Reverb
Send 7
Large Concert Hall
Send 8
Speciality Reverb (e.g. 3D, Gates Reverb)
Send 9
Slap Back Delay (1/16 Note or shorter)
Send 10
Special FX Delays (Ping Pong, etc.)
other options would be specialised reverbs for Snare,
Amp Chambers, etc. - I personally have set up 3 banks
of 10 Sends each, for a total of 30 Send FX, send 1, 11, 21
are standard reverbs, send 2, 12, 22 are different delays,
you get the idea...
• set up most commonly used plug-ins on the stereo
buss by default, but keep them bypassed for now
• save the empty template (without any audio) to your
harddrive to use with all your future mixes; keep improving your template as you use new plug-ins or change
your workflow
IMPORTING THE FILES
• import or drag the audio-files into your template (in
Logic, dragging them all into the arrange windows creates blank tracks - very useful.)
• save the mix-session to your primary data drive, always make sure that all required files for your project
are included in one folder (to be able to back them up to
another drive by just dragging one folder over…)
• backup the folder above with all files on an external
drive every few hours
• bring the imported audio into a defined order of tracks
in your DAW software, build groups for the following instrument groups, label the tracks clearly and color-code
them as follows:
Drums red
Sound FX yellow
Bassdark green
Guitarslight green
Keyboards blue
Orchestra orange
Backing Vocals
purple
Lead Vocals pink
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(yeah use different colours if you want, but know I’ve
consulted an expert in colour psychology to find those
colours)
•
use meaningful track names (Routing, Instrument Name) and Icons -> BUS 33 Vocals Lead
•
generate triggers tracks for Kick and Snare for use
with Drum Replacement Plugins - just generate the
triggers for now, it is not the time to pick drumsounds
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If you use a console (like I do), you can assign the most
important elements of the mix to individual channels,
but make sure stereo outputs 1/2 go to untreated channels of your console that are set at unity gain. With everything in your DAW routed to stereo outputs 1/2 at
the beginning, you can start using this default input on
your console, and only route selected tracks to individual console channels. Your reverb returns for example
would still come through the default stereo outputs 1/2.
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THE CONCEPT OF „HANDLES“
AND COMPLETE CONTROL
Starting with Chapter 5 of this book, we are going
through the various dimensions that can be controlled
for every sound-source in the mix, and also on busses,
subgroups and the 2-bus/master.
JUST A FEW EXAMPLES
•
•
•
•
•
•
tonal balance - EQ, Filter
dynamics - Compressor, DeEsser, Gate
transients - Limiter, Saturator, Transient Shaper
volume - Faders, Routing, Gain Staging
room placement - Reverb, Delay, Panning, Stereo
Width
modulation - Chorus, Flanger
Each of those require a dedicated processor (plug-in or
outboard device). While some of these can be inserted
as needed, you can save a lot of time by including the
commonly used standards in your template, that are bypassed by default. You bypass them by default as when
the signal fits in the mix as is, you might not need anything in the signal chain.
This is what I refer to creating a „handle“ on a specific
aspect of your source.
Creating a handle never consists of a single „knob“ or
control - even drastic changes usually consist of a number of processors with subtle settings and also, even
when you apply drastic settings, subtle corrections need
to be made.
Basic terms like „transients“ or „dynamics“ are very technical, and rarely used by people to describe sound.
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Think about how you would deal with the following requests:
•
•
•
•
„The vocal needs more attitude“
„The sax doesn’t sound confident enough“
„Can you give the kick more knock“
„I want more of a 60s vibe to it“
You need a handle on all of these. The lead vocal won’t
magically sound like „more attitude“ - there is no single knob for that. The „handle“ on „lead vocal attitude“
would consist of a number of processors combined that
push the sound in a certain direction.
While terms like „attitude“ will never be 100% objective,
on this particular example I would first create a rich and
full bottom to the sound, then add a lot of mid energy,
probably around 2, 4 and 8kHz with an analogue EQ,
and after that push the signal into a compressor that
adds harmonics and saturates, even distorts the vocal.
A final brickwall-limiter on the vocal chain would make
double-sure that the vocal is always pushing and upfront, like someone with an attitude would stand right in
front of you and not back off an inch.
More „knock“ on the kick? Well, perhaps knock a wooden door, and think about what frequencies are involved?
These are just two examples for the concept of handles,
but once you have mastered a variety of audio tools you
will be able to create a handle on everything you need to
manipulate, even when attributes used to describe the
desired sound are not technical, but rather referring to
emotions.
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REFERENCES AND A/B-ING
What do I need to reference?
You need to be able to switch between
•
the rough mix of the song you are mixing
•
the untreated and mastered versions of your stereo bus
•
a selection of references
•
if you have a console or monitor controller, utilise
the external inputs (drag reference songs into your
mix sessions, send them to a spare pair of outputs
that are patched into the external inputs)
•
if you don’t have external inputs, use a software solution like "Magic AB" for that - it's saves you a ton
of money purchasing extra converters, hardware
etc. - you just use it as your last plug-in on the
master, and it handles all your A/B-comparisons
and references.
What songs to pick as references?
•
•
•
•
•
build a library of reference songs
include some of the worlds biggest radio hits in your
library, to have a general reference for current loudness and frequency curves
include some well known songs in the genre of the
song you’re about to mix
include some of your personal favourites
if the client gave you a reference for the mix, add
that too
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57
I use the plug-in Magic AB, which allows you to compile
a selection of references, and save them as a plug-in
setting.
•
Magic AB will be inserted as the last plug-in on
your stereo bus, and allows you to not only A/Bcompare between your own mix and a reference,
but also between several references with one click
•
you can save a plug-in setting for different
genres, and it will load all the songs with one click
•
you can add all types of audio-files, even iTunes
AACs that you purchased
•
the audio-files can remain at their original locations on the hard drive
•
you can loop specific section of a reference song
•
each reference can be level-matched to your own
mix by ear
•
you can globally compensate for the added loudness the references will have due to the fact they
are mastered
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FAQ
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60
What kind of DAW computer do you recommend for
mixing?
CPU types:
•
•
•
•
•
•
•
in general, Intel i7 outperforms i5 and i3 by far because of virtual cores
CPUs in Macs are usually clocked quite low (for less
power consumption and reliability)
higher clocked models can be build to order at the
Apple Store
CPUs with 6, 8 or 12 cores, including Xeon and Haswell-E are clocked slower than those with 4 cores.
The machine with more cores is not always the better choice, especially considering the higher cost.
all current MacBook Pro’s and iMacs with i7 CPU are
great computers for mixing
all versions of current Mac Pro of course fantastic machines, but very pricey and you have to add
Thunderbolt to PCIe-converter if you want to use a
PCIe sound card
if you want to spend less, consider building your
own computer around an i7 Haswell-processor.
Harddrives:
•
•
•
•
•
•
1x SSD drive (at least 120GB) for the operation system and apps
1x SSD drive ONLY for the current projects you’re
working on
Harddrives for backups, can be internal or external,
even USB
Audio Settings:
I usually start with 256 samples buffer, if needed increase to 512 samples later
buffer-size can be large for mixing, as we do not
•
•
•
•
track or play virtual instruments
Plugin Latency Compensation set to „ALL“ (otherwise the Aux-busses and groups are adding a delay)
using the „Track Freeze“ feature on every track that
has plug-ins inserted
Freeze (in Logic Pro X) renders your pre-fader channel signal with all plug-ins as a 32-bit file on the
harddrives - that track will not require CPU-power,
only a fast SSD drive if you freeze a lot of tracks;
current SSDs are capable of playing back 100+
tracks in realtime
note: auxes and busses always need realtime CPUpower: you can’t use freeze on those
Audio Interfaces:
•
PCIe and Thunderbolt-solutions perform best, I recommend to avoid USB but it depends on your budget
I produce house music with Logic Pro X, and often
create „creative fx“ via aux send/return which play
an integral role in the production. How do I deal with
these?
Creative FX in house music or techno - often involving
automation/filters etc. - are not the "standard" category
of reverbs/delays. I recommend to bounce them as an
"FX only"-stem. You can either achieve this by soloing
the track on which you use the send and sending the
dry signal to "no output", or you could use the following trick: create a track for the FX aux return in your arrangement, then put an empty MIDI region on that track.
You can then use the menu FILE -> EXPORT -> "All tracks
as audio files..." and it will include your "FX only"-audio
file in the folder of bounced files.
Question on how to route to busses & fx - example:
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61
you balanced the drums and added fx to them... Do
you send the fx returns also to your drum buss? If
not, don't you need to adjust your sends if you adjust
volume on the busses? What about that if you like
sharing fx (drums and others in the same room etc...)
Use pre-fader sends, use the drum fx exclusively for
drums, and route their returns to the drum bus as well,
and/or put them on the same VCA group. Same goes for
parallel compression busses, the sends to them are prefader, not following automation, but the returns follow
the automation of the source - I do that via the Drum
VCA group, but an audio subgroup would also work. I
personally don't have a drum-bus, instead using VCA
Groups on the SSL. The main point is to keep those aux
returns that you use for drums (could be reverbs, parallel compression, etc.) exclusive to the drums. Even if the
setting might be similar to the vocal reverb, I would be
copying the setting and group all the tracks that are related to the drums, to the drum subgroup, or in my case,
drum VCA fader.
By keeping all FX separate and assigned to the instrument group they belong to, you get the added bonus
that you can treat the FX returns for each group of instruments separately. We want total control in mixing.
Also, you don’t need to create a new send for every reverb - you can send, for example, to 20 different snare
reverbs from „Send 5“, and then just unmute and mute
them at the aux return, one at a time, until you find what
you’re after; this is a very fast and intuitive process, and
gives you the option to use one or several reverbs at the
same time, and level their balance at the different reverb
returns.
What are your thoughts on inserting reverbs right on
the channel strip, and adjust with the mix parameter?
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3
Nothing wrong with it. Only disadvantage I see is that
you can't EQ or compress the reverb-signal separately.
When using references, does the song I’m referencing have to be in the same key as the song I’m mixing?
Wrong direction, you're looking at referencing too literally. Also, don’t ever reference against just one song.
Look as references as another way to distract you from
getting too focussed on the track you’re mixing.
analogue/console vs. ITB: when do you recommend
mixing on a console, and are there cases where you
would prefer mixing ITB („In The Box“ = completely
in the DAW)
•
•
•
•
•
•
•
•
•
great mixes can be created on both
„console“ always means hybrid (plugins ITB + tracks
fed and summed in console)
even amongst the world’s Top 10 engineers, it’s
50/50 now
all of these learned on analogue consoles though,
and replicate that routing ITB
ITB is often an economical decision, and recalls are
very easy
some people are fast with a mouse, some like knobs
and faders = matter of taste
gain staging is more critical ITB
rock mixers lean towards consoles
EDM producers are mostly ITB
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64
CHAPTER 04
THE MAGIC OF
THE 1ST LISTEN
CHAPTER 04
THE MAGIC OF
THE 1ST LISTEN
U
p until the early 1980s, Mixing commonly hasn’t
been a process separate from production. The one
guy that first put the specialised profession known as
Mix Engineer into the spotlight was Bob Clearmountain,
and around the same time, huge SSL-Consoles and the
advent of sampling and harddisc recording, all of which
are concepts and technologies that resulted in the modern DAW (Digital Audio Workstation) as we know it.
You will see that one recurring theme in this book is the
dividing line between production and mixing. We cannot
mix what hasn’t been produced, and blurring the line is
affecting the quality of your mix and the final product.
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65
THE 1ST LISTEN EXPERIENCE
A little philosophical discours before we dive into the
technical details of mixing.
Think about this: the greatest experience for anybody
involved in music production is the „first-listen experience“. When you are listening to a song for the first time,
it is the only time you are hearing it like the consumer
out there, the person who we want to win over with our
final product.
But what happens when you’ve already worked on a
production for a bunch of days, and maybe you are even
a co-writer on the song and played/programmed most
of the parts?
You know where I’m going with this one, right?
By the time your roughmix is done, you’re married to the
song. In the worst way. The song keeps spinning in your
sleep and you could not be further away from hearing
your production with a fresh ear.
At this point you do not hear it like a record-label A&R,
a radio programmer, and certainly not like a consumer.
You are emotionally connected to your work. You
might be in love with it, proud of your drum-sound, the
amount of bass in the mix or some of the production
tricks you’ve used, or even worse, the opposite, you
could be totally frustrated and tired of the damn thing!
Solutions?
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66
A.
MIXING FOR SOMEBODY ELSE
When you mix a song for somebody else, you are indeed
having the great advantage of the
“FIRST-LISTEN EXPERIENCE”
A great mix engineer, within seconds of his first listen,
will understand where the producer wants the mix to go,
spots problem areas of the mix, and as part of his craft
has the skill and the tools to translate his impressions
quickly into a mix with improved audio quality.
Add to that other advantages a paid mix engineer has
that help him to get the job done.
"What else helps the mix engineers judgement?
•
An environment specifically created for the task of
mixing
•
He is mixing records to a high standard EVERY DAY.
•
He gets paid! Wow, who would have thought that!
„Why should that make the outcome of his work any
different?“
you might ask.
Well, this one is easy. Let’s look at exactly what a mix
engineer gets paid to do…
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67
(just some examples of what clients request)
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68
The mix engineer gets paid to improve the audio quality while respecting the integrity and intention of the
production.
His job is NOT to judge or criticise the quality of the
song, production or instrumentation. Somebody who
spends money on a mix engineer usually BELIEVES in
what they deliver, while they are realistic enough to
know that there is room for improvement
I’m sure you can see why a paid mixer (whose only focus
is on mixing the song for somebody else) is in a much
better position to do a quality job compared to someone doing the mix as a task within a bigger project.
To summarise, before you start to do any work on the
mix, ask the producer to provide you with a rough mix
of the song, the one the client likes best so far, the one
that represents where the producer wants to go with it.
Your first listen is for emotional input from the rough
mix. The second listen is analytical.
1ST LISTEN – EMOTIONAL
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Have a listen to the roughmix through the speakers you
are most familiar with – this could be your portable stereo in the kitchen as well as your car stereo or your main
studio speakers. Try not to listen with an analytical or
technical ear, don’t focus on anything specific, just relax
while you’re listening, and await your natural emotional
response.
Now go and do something else. Have a coffee, eat, do
your tax return. Something fun. Let the first impression
settle.
2ND LI STEN – ANALY TICAL
If the rough mix made you enthusiastic and singing along, dancing or playing air-guitar, there is not
much to worry about for now.
For the second listen, switch back and forth between
your different sets of speakers as discussed in chapter 1.
The mix might not work on all systems, which can be a
hint for areas of improvement. Find out why it works so
well, and be careful – going forward – not to fix things
that are not broken. Anyway, it will be fun to explore the
multitracks, and you won’t have to re-invent the wheel
on this mix.
ROUGH MIX SUCKS?
In the more common scenario, the 1st listen left you
with mixed emotions. Maybe not exactly giving you a
headache, but close to that: a feeling that something is
not quite working in this rough mix.
On the 2nd listen, focus on why it’s not quite working.
You might not get to the bottom of the problem immediately. But further observations will get you closer.
Make notes as you’re listening. Switch between different
speakers.
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TAKE NOTES!
Here are some observations that I would typically come
up with during the 2nd listen - take notes!
(these notes are taken straight from projects I worked
on in the last few months)
•
No lift when chorus hits. Impact even drops! First
vocal note in chorus out of tune and too late.
•
Vocaltiming doesn’t sit tight on the beat. Esses conflicting with hi-hat and throwing the groove off.
•
Several Kicks and bass synth fighting for the low end.
•
Bass notes vary in loudness.
•
Drums and vocals sounding one-dimensional and
squashed.
•
Instrumental hook too far in the back. Doesn’t shine.
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B.
MIXING YOUR OWN SONG
There is a lot of confusion around the point that separates production from mixing.
The general answer to that is very simple: production
and mix are done by two different people looking at the
process in a different way.
Every producer is EQing, compressing, grouping, applying mixbus-treatment etc. and ending up with the best
mix they could do.
The mix engineer is then supposed to take that and get
it to another level.
Producers usually send me all of their instruments EQ’d
and compressed as in the rough mix. That includes
the dynamics of sidechained instruments in electronic
genres - the vibe of the sidechain is certainly part of the
production. Reverbs and delays are not printed, unless
they create a very specific vibe that is part of the sound
(example: spring reverb on a guitar).
Drums and vocals would be an exception - I prefer to
receive them with all plugins bypassed.
A lot of my mix-work involves „recreating“ the vibe of
the rough mix, but at a better quality and higher resolution.
For those of you who write, produce + mix their own
music, I recommend wearing two „different hats“ for
those tasks.
When wearing the „producer hat“, don’t be concerned
with the stuff we’re discussing here. Create a vibe, go to
extremes, try a lot of things, be innovative, open 100 virtual instruments, etc… your rough mix can be messy and
purposely ignoring a lot of „classic audio rules“.
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But when wearing the „mix engineer hat“, the focus is
on retaining the vibe of the rough mix, improving the
audio quality and compatibility to the various mediums
the music will be played on.
That involves a lot of deconstructing and rebuilding.
With that, let’s dive into the details of this issue.
PROBLEM –
NO 1S T LISTEN EXPERIENCE
•
there’s no „1st listen“-experience; you will never
hear your own song the same way as someone else
when they hear it for the first time!
SOLUTION –
DISTANCE YOURSELF FROM YOUR WORK
•
at the point where the production is done, and ready
for mixing: STOP listening to it AT ALL. PERIOD.
•
work on a few other songs, get away from it as long
as you can, the longer the better!
•
even in case of a pressing deadline, DO SOMETHING ELSE for a moment, listen to lots of other
music, take a walk, get a coffee: your musical brain
needs a reset!
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BEFORE THE MIX
•
decisions around instrumentation are part of the
production, NOT the mix
•
print all tracks to audio-files, this will not only force
you to make decisions, but will free up your computer from CPU-load you need for mixing
•
before you print the individual tracks, make absolutely sure, timing and tuning are where you want
them, and unwanted noises and clicks are removed.
•
you can print some tracks as groups, for example
bounce down 12 doubled backing vocals as one
stereo file; you will end up with less files in your mix
session and make it more manageable – reducing
the size of your sessions always wins!
•
open the printed files in a completely new and empty session, ideally a template costumized for mixing-duties (we discussed this in Chapter 3)
AFTER THE MIX
•
when your mix is done, consider moving your DAW
into a great mastering studio and spend a few hours
WITH the mastering engineer to finetune your mix.
Typically, low end and vocal levels, amount of reverb and bus compression are things to discuss with
the mastering engineer, plus you will hear your work
in a different (and great) listening room. If you use
a lot of analogue gear and outboard, you can print
stems of your individual tracks and take those into
the mastering session. Once all issues are resolved,
leave the mastering engineer with the track for half
an hour, and he’ll sort out the master.
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CHAPTER 05
WE ARE MIXING! THE
FOUNDATION, KICK,
BASS AND
GAIN STAGING
CHAPTER 05
WE ARE MIXING! THE
FOUNDATION, KICK,
BASS AND
GAIN STAGING
D
uring this chapter, we are building the foundation
for the mix. What we’re doing now can be described as a frame in which we’ll be able to neatly insert
a picture (the vocals) in the following chapter.
While there are many elements that we can finetune and
treat „on the fly“, the boundaries of our mix are defined
by the low-end foundation and our gain staging skills. If
the low-end doesn’t work, whatever sugary cream and
icing we put on top will not make up for it.
Before we’re starting, please get a quick idea of which
key and harmony-progressions the song is using, the
notes/frequencies played by the bass, and where the
kick sits relative to this.
If you don’t have a lot of experience with this, here’s
an example on how an analysis like this can be done.
The following was part of a mix analysis of the song „All
About That Bass“ by Meghan Trainor, a song nominated
for a Grammy in 2015.
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KEY OF THE SONG + CHORD PROGRESSION
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The song is in A major, based on the chord progression
A major / b minor / E major / A major
BASS NOTES USED + THEIR FREQUENCIES
The lowest notes the bass plays are (from lowest to
highest):
E1
41.2 Hz
(root note of E major, the dominant chord)
G1
48.9 Hz
(minor 3rd to E major, used as a blue note)
G#1
51.9 Hz
(major 3rd to E major)
A1
55 Hz
(root note to A major, the tonic chord)
B1
61.7 Hz
(root note of b minor, the minor parallel of the
subdominant chord; also 5th to E major)
C2
65.4 Hz
(minor 3rd to A major, used as a blue note)
C#2
69.2 Hz
(major 3rd to A major)
D2
73.4 Hz
(minor 3rd to b minor)
E2
82.4 Hz
(5th to A major)
F#2
92.4 Hz
(5th to b minor)
BASS SOUND
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The bass sound used is a double bass played pizzicato
style. The frequency-spectrum of the double bass is
dominated by the 2nd harmonic (octave up, see below)
and 1st harmonic (the frequencies listed above). Other
than a tiny and very short plugging noise that happens
around 4-8kHz, there’s not much that can be boosted
to bring the bass upfront in the mix. Compare that to a
bass guitar, where you can feature the mids or even run
a parallel track through a distortion pedal or guitar amp.
FREQUENCIES OCCUPIED BY BASS + KICK
This is the complete list of frequencies the bass and
kick in “All About That Bass” occupy in the mix:
41.2 Hz
48.9 Hz
51.9 Hz
54 Hz 55 Hz 61.7 Hz
65.4 Hz
69.2 Hz
73.4 Hz
82.4 Hz
88.4 Hz
92.4 Hz
98 Hz
103.8 Hz
108 Hz 110 Hz 123.4 Hz
130.8 Hz
138.6 Hz
146.8 Hz
164.8 Hz
185 Hz
E1 (fundamental frequency)
G1 (fundamental frequency)
G#1 (fundamental frequency)
KICK (fundamental root note)
A1 (fundamental frequency)
B1 (fundamental frequency)
C2 (fundamental frequency)
C#2 (fundamental frequency)
D2 (fundamental frequency)
E2 (fundamental frequency)
2nd harmonic of E1
F#2 (fundamental frequency)
2nd harmonic of G1
2nd harmonic of G#1
KICK (pressure point)
2nd harmonic of A1
2nd harmonic of B1
2nd harmonic of C2
2nd harmonic of C#2
2nd harmonic of D2
2nd harmonic of E2
2nd harmonic of F#2
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LISTENING
Now, have a listen to the song (best to buy the track
on iTunes as it’s a great reference song) – that bluesy
bassline played by the double bass is totally dominating
the low end frequency spectrum of the mix, and a purely
theoretic frequency analysis confirms that.
LET’S LOOK AT SOME FREQUENCY CURVES!
This curve is made from the intro of the song and shows
JUST bass and vocals (using LogicPro X’s Match EQ
“learn”-feature):

There is definitely a High Pass Filter (HPF) used that cuts
even into the lowest bass note. The HPF plus the notch
just below 200Hz give the bass a very defined place in
the frequency spectrum of the mix.
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Let’s look at the Kick Drum in comparison:

The fundamental note of the Kick sits at 54Hz, just below
the root note of the song key, with the pressure point an
octave higher, at 108Hz.
The Kick is very compressed and sits “behind” the bass,
to never dominate the low end of the track. There is another bump below 200Hz which works perfectly as the
bass has a notch at the same place. The Kick is a very
rich sound with lots of noises and higher harmonics,
so it still sticks out, but definitely not because anything
would be boosted in the low end.
Note how there is nothing notable happening below
50Hz.
Here’s what I’ve done to find out at what frequencies the
Kick exactly sits, which wasn’t as easy as usual.
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Logic Pro’s Channel EQ Analyser, looking at the track
in a section where there’s no bass.

Next, I’m creating a very small EQ band, fully boosted
(keep your speakers at low level to not blow them), then
slide the frequency through the low end until I find the
exact resonance frequency. I repeat this with a second
EQ band and find the next big resonance above. Obviously looking at the analyser helps. When you do this,
watch how the analyser reacts. Between 54Hz and 55Hz
there was only a small difference, but clearly the resonance sits closer to 54Hz.

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When you change the EQ gain from + 24dB to -24dB,
you can completely suck the life out of the kick with an
EQ notch, further proof that we found the right spots.

In my own mixes, I’m going through the same process, take a look at the song key, the frequencies and
plan how to fit Kick and Bass together.
Sometimes the Kick lacks clear distinctive frequencies
for fundamental note and pressure point.
In that case, I locate them, and boost a little bit.
Sometimes though, I receive Kicks that have too much
of a boost on the fundamental note or pressure point.
They can be easily tamed by setting a notch EQ on the
frequency and then just backing off 1-2dB.
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DRUM REPLACEMENT
•
the whole point of Drum Replacement is to have
access to a high quality recording of a great drumset with multiple microphones that can separately
be accessed. You can add just a pair of overhead or
room mics to give the existing drums some depth
and dimension, you can add the bottom snare mic
for some noise component, bring in a Kick that
has defined and tuneable fundamental note, etc.
•
Drum Replacement Plugins can extend the repertoire
of sounds you have at hand for the mix. They can help
to add excitement and punch to the mix. Use them
to add to what’s there, not to completely replace.
•
that said, if your original Kick is badly recorded or
out of tune, you will always find a sound that has a
similar vibe and works better; in that case, replace!
Drum Replacement really means two separate elements:
1.
the trigger generator, a plug-in that turns the audio
tracks of your existing drums into dynamic MIDInotes if everything is set right, the trigger generator
captures the full dynamics of a real drummer
2.
secondly, a drum plug-in that provides you with
multi-channel outputs that are identical to the multitrack recording of a VERY GOOD drum recording a great drummer playing a great fine-tuned drumset
in a great acoustic space, the microphones set up by
a great engineer, recorded with the best pre-amps
and the best console through very good converters.
These plug-ins usually provide a huge number of different samples in all dynamic ranges, multiple samples for
each dynamic range, so that even if you play the same
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16 MIDI-notes with the same velocity in a drumroll, it
will still sound organic as no sample was used twice.
Of course, you can use drum replacement to completely
replace the original drumset (I personally rarely do that).
The trick is to use it very subtle so that even the producer of the song won’t notice that samples are involved.
Examples:
•
bottom snare mic, overheads, room mics
•
adding an organic touch to synthetic drums
•
provides you with a set of organic signals
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SHAPING KICKS
Next up, let’s deconstruct the components of a Kick
sound, and look at what type of „handles“ you can create to control it in the mix.
The Kick is of extreme importance, and it’s worth – as
part of building your mix – to spend a few minutes creating the „handles“ to control the parameters we’re talking about in this article.
What you need to control will develop as you’re progressing with your mix, but it’s important to have those
controls at hand when needed, and to know how to use
them.
KICK TONES: FREQUENCIES + TUNING
Think of the Kick Drum Sound consisting of 3 different
frequency components!
1.
the fundamental root note in the low end (45 – 75 Hz),
2.
the pressure point (an octave higher, 90 – 150 Hz)
3.
harmonics and noise (anything above).
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Think of them like an ensemble of tones and frequencies, like a string section that has a bass, celli, violas and
violins.
To place the Kick perfectly in your mix, it helps to look at
and treat those 3 distinct components separately.
Some producers use different samples for these components, and I often get them as separate tracks for my
mix sessions.
But even if your tracks come with a single Kick, you can
create a copy of your Kick track on another track. By
cutting (HPF) everything below 150Hz you can exclusively treat the harmonics and noise-component (3.).
Experiment with various treatments from Tape Simulation to Distortion, Compressing/Limiting, then add that
channel to the original sound.
Just as an example for the separation of the 3 distinct
frequency components: the screenshot below shows
the fundamental removed with a HPF, pressure point
(138 Hz) dipped by 5 dB, and a broad boost of the “noise”
of + 5.1 dB at 2500 Hz.
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The fundamental (1.) and pressure point (2.) are both
an essential component of modern music and need
their resonance frequencies for themselves to really cut
through the mix.
The harmonics and noise components (3.) are very important to „find“ the Kick in your mix, especially on small
consumer devices like laptops, smartphones, small portable stereos and cheap earbuds.
I sometimes separate those three components to 3
distinctive tracks that sit on 3 large faders on my SSL
console – top engineers have done this for many years
and it’s quite easy to achieve on an SSL, you just send
the signal to the routing matrix via the „small fader“, and
„mult“ it to another 2 faders. Thats without even considering added parallel compression channels, and virtual
overhead and room mics generated from a Drum Replacement Plugin (another future post).
808 Kicks typically have a strong fundamental, but
lack harmonics and noise.
Synthetic Kick Drums, like of a Roland TR-808/909 have
a very clean and defined fundamental tone. It often gets
bend down after the initial attack though (an Envelope
Generator modulating the pitch), which means the fundamental tone of the sound sweeps through the low end,
for example from 80Hz down to 10Hz. A bend can make
those Kicks more difficult to mix, and I recommend to
always carefully watch and use an analyzer while listening for the right balance. This is where a good sounding
room is a huge advantage.
There are all kinds of boxes (or their plug-in versions)
that generate harmonics – from subtle Pultec EQ, Fairchild Compression, or Analogue Tape Simulation, to
more extreme treatments like a Guitar Amp or Distortion Pedal Simulation.
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Sometimes creating a copy of your Kick through a distortion pedal or amp simulation, mixed slightly in the
original sound brings up exactly the harmonics needed
to make your Kick easier to „find“ in the mix.
Real Kick Drums are a lot more complex than 808/909
-sounds.
A real Kick Drum is a small physical space in itself, with
two resonating drum heads interacting within the recording room, which also adds to the sonic characteristics. The result are very complex harmonics, and sometimes a few neighbouring fundamentals and pressure
points compete for each other in the mix.
While the complexity of tones modulating, adding to
and cancelling each other adds to what we perceive as a
real sounding Kick Drum, there is a lot more that can be
done to those sounds in the mix.
Use a digital EQ with a very high Q-Factor (a narrow band) to identify the fundamentals and pressure
points of the sound.
Yes, there can be several competing ones, and which
ones to boost and cut is depending on many factors, like
the key of the song, where the frequencies of the bass
guitar or bass synth sit in the mix, and of course most
importantly your taste and the genre of music.
Even a Kick that is totally out of tune with the key of the
song and the rest of the instruments can be spot on in
the mix, if it fits the genre and has the right attitude.
However, tuning those frequencies in harmony with the
key of the song is something many successful producers
and engineers do. For more on this, have a look at my
recent blog posts analyzing the low-end of the two biggest hits of 2014, „All About That Bass“ (Meghan Trainor)
and „Shake It Off“ (Taylor Swift).
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BTW, „convert audio into sampler track“ is a sensational
feature in Logic Pro – it basically imports your audiotrack into Kick Notes in Logics EXS24-sampler, and triggers them via internal MIDI.
Now that they’re in the EXS24, you can tune them in the
sampler plug-in. Works great if your Kick is for example,
a semitone too high, colliding with the key of the song,
you just transpose it „-1“ in the EXS24 and your low-end
alignes magically - sometimes 10 cent of a semi-tone
can make a huge difference.
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Keep in mind, low end instruments have a lot of energy,
and you don’t want them to collide and modulating by
not being properly fine-tuned.
Regardless of how you end up balancing the different
components of the Kick, make sure to find the frequencies of the 3 main components so that later on in the
mix you have handles in place to balance your Kick. You
can easily identify those frequencies with a digital EQ,
and an analyser obviously helps.
In Logic Pro X, the Channel EQ in conjunction with the
built in analyzer works great for finding resonances (as
the Channel EQ is creating strong resonances like an
analogue EQ), however, to boost or reduce them in level
I recommend using the Linear Phase EQ, which works a
lot cleaner and precise which is important when balancing the low-end.
Concepts to use here are to pinpoint EQ-bands to the
exact frequencies of the fundamental and pressure
points, and boost or cut them depending on whats
needed. Also, you can make the Kick more defined by
cutting the frequencies between the fundamental and
the pressure points – make sure to use very high Q-factors! Again, a high-quality Linear Phase EQ is your friend.
The one in Logic Pro is great, but there are many others
on the market.
One more thing…
The final boosts and cuts should be performed when all
tracks of your mix are playing, or at least Kick, Bass and
Vocals.
“The final boosts and cuts should be performed
when all tracks of your mix are playing, or at least
Kick, Bass and Vocals.”
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KICK DYNAMICS:
AT TACK AKA THE TRANSIENT
The transient is the initial attack of your Kick. On a reallife Kick Drum, a pedal-controlled beater hits the drumskin, and that short initial impact creates the transient in
the audio.
On electronic Kick-sounds, the transient is either the
fast ramp-up of an envelope which results in a tiny click
noise (808/909) or an added noise generator with a very
short envelope. In both cases, a very short click noise is
the result.
The added click noise was first used on a drum-synthesizer by Dave Simmons in the early 80s, if you’re old
enough you might remember the SIMMONS Drumset
which was basically a 5 or 6 channel synthesizer with
one voice for every drum-instrument.
The sounds were created by mixing short click noises
(simulating the stick or beater hitting the drumskin) with
tones that could be tuned and connected to a pitch envelope generator, plus longer noise sounds added to the
tones.
You might have heard some of those iconic „laser-gun“
or “spitting”-type of sounds from the 80s.
The SIMMONS Kick-sound was somewhere in between
808/909 and a modern progressive metal Kick.
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TOOLS TO CONTROL TRANSIENTS
If you want to bring the transient in the Kick up or down
in the mix, there are 3 ways to achieve this:
1. Transient Designers
There are several of those plug-ins on the market, the
most popular being SPL Transient Designer, which is
based on the original 4-Channel hardware unit. Many
plug-in companies offer tools to achieve similar results,
I personal like the original hardware best (and there is a
plug-in version of that as well)
2. Noise Gates
If you copy the Kick to a second DAW track and use a
noise gate to make it extremely short, you can turn the
transient up by mixing the noise gate track with the original sound. If you phase reverse the noise-gated sound,
you can remove the transient or at least soften it slightly.
3. Sampling
If your Kick is in a sampler, you can just shorten the envelope of the sound in the sampler plug-in. See the feature we discussed above (converting an Audio Track into
a Sampler Track)
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KICK DYNAMICS 2:
LENGTH AKA DECAY TIME
To control the length of the KICK TONE is of course of
utmost importance.
The Decay Time of the Kick can make or break the
groove, and again – it’s helpful to be able to control the
three tone components of the Kick separately in length.
The tools to use are somewhat similar to the Transient
Control, but of course you can just go into your Waveforms and cut it where you want tone Kick to end, and
apply the desired Fade to the audio.
Another useful feature in Logic Pro…
“Remove Silence from Audio Region”

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Automatically separates the silence between the Kicks
below the threshold that you can set.

Applying a fade to the separated Kicks.

Finally, the Kick sits in the middle, mono, period.
At least the direct and low end component of the Kick.
If necessary, split the Kick at 300Hz, keep the fundamental and pressure point mono and only leave noise
and harmonics from 300Hz upwards Stereo.
Set up a Sidechain.
I recommend to always feed a sidechain bus with your
Kick. In my personal mix template, the first send in the
channel strip is set to 0dB (unity gain) and sends to Bus
64. Bus 64 doesn’t send to an output, disable the output.
The busses 64, 63, 62, 61 are my standard sidechain busses for Kick, Snare, Vocals, and whatever else needs to
push other elements to the back of the mix). I’m not always using it, but when I need it, it’s already setup by
default.

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Without going into detail here, just know you would assign Bus 64 as your sidechain, most commonly on the
bass guitar or bass synth, and push the bass a few dB
back, triggered by the sidechain signal.
Again, Logic Pro’s standard compressor works great for
that, but this technique works of course very well with
a multiband compressor/EQ like the „Waves C6 Sidechain“.

These techniques will give you the handles on important
parameters, and it will be a lot easier to get a great balance between Kick and Bass.
Please keep in mind that having the handles in place
doesn’t mean to have to use and tweak all of them.
There are some genius producers out there who deliver
Kicks that are sitting perfectly in the mix.
In that case, turn it up and be done.
BASS
Getting the bass to sit right in the mix is largely depending on how much space the Kick takes, and where the
bass-notes are sitting relative to that in terms of frequencies. If you haven’t fully understood that concept,
please go back to the beginning of this chapter, and
analyze a few songs the way I showed it.
In addition to that, here’s a shortlist of things you should
look at:
•
it requires an extremely good listening room,
and full range speakers to judge the level of
each bass note in the mix - you can’t trust most
rooms, and while analyzers and meters are a
helpful tool, you shouldn’t trust them either
•
here’s actually a great application
phones: listen through the song
to end, and level the bass - each
vidually - for an even level across
•
the Bass does usually not have a lot of energy at the subsonic range - and if it has - it might
be clever to clean it up or at least control it
for headfrom top
note indithe song
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•
while speaker systems can handle short impulses of sound below 30/40 Hz. any sustaining sound that has excessive amount of subsonic
needs to be reduced to an amount that doesn’t
interfere with the rest of the signals in the mix.
•
one way to make subsonics more dominant in the mix is to add harmonic distortion that happens in higher frequency bands
•
even if the fundamental notes of the bass are happening below or around 60Hz, the biggest role in
how loud you perceive the bass in the mix happens
an octave higher, at the so-called „1st harmonic“
•
in Chapter 9, we are looking into using compressors for generating harmonics, a technique that is very relevant to bass-sounds
•
Sidechaining, unless used as an effect, should
be a „last resort“-type of solution. You shouldn’t
completely rely on it to balance Kick and Bass.
•
Especially in electronic genres, the key to a good
balance is correct fine-tuning between kick and
bass. Dissonances between the two will cost
you valuable head-room in the low-end, and you
won’t get your mix as smooth as you will want it.
•
reserve the low-end for instruments on which you
can hear it. You should know without looking which
elements of your mix have components of low end
frequencies (below 120Hz). It shouldn’t be many - if
you can’t hear the low-end on an instrument within
the arrangement, filter it with a High Pass-Filter. We
need that space.
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The Goal at this point is to have a healthy overall balance
between Kick and Bass, note, we are not determining
the final amount of Low End that you want to have in
your mix. Stay away from boosting any low frequencies
at this point. Whether the overall amount of low-end in
your mix is correct or not will be determined at a later
point of time. It’s not the time for those decisions now.
GAIN STAGING
(DAMN – IF ONLY YOUR BANK ACCOUNT
HAD ENOUGH HEADROOM!)
But you’re hitting “red” all too often, start calculating and
things become uncomfortable.
Sounds familiar?
„Digital Red“ means you’ve maxed out the
bank account of your available dBs
– no more credit!"
Exactly how people feel about their Gain Structure in
mixing – „Gain Staging“ being the strategy you chose to
use your bank account of dBs, if you even have a strategy…
Like your bank account, the Gain Structure of your mix
needs enough headroom to be able to deal with crazy
or unexpected events.
The Drums are buried in the mix, and the artist wants
them „waaaay“ upfront?
OK, you start turning them up, and by the time they are
where you need them, the drum-levels are up 6dB – hitting digital red on the mix bus!
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I’ll start with some underlying history and theory. And
then tell you exactly what to do about the problem…
Just before Digital Audio took over as a standard, early
DAW manufacturers in the 1990s made a historic mistake in laying out the wrong scales for digital level meters: no warning all the way up until 0dB, so you don’t
know you’re in trouble until it’s too late.
Whereas 0dBVU in analogue used to be a reference
point for solid level, early Digital Audio Workstations
used 0dBFS (FS for "full scale") as the “point of no return”.
Sony and Studer DASH-machines did a lot better, but
they were six-figure digital multitrack-recorders that
were the industry-standard until DAWs took over.
Fast forward – DAWs took over between 1991 and 1998
– Digidesign ProTools being the first and most popular
one, others like Cubase and Logic followed by extending their MIDI-based sequencers with “audio”-features.
What all of them should have done from the beginning,
is declaring levels higher than -18/-15dB as „yellow
zone“, and -8/-5dB as „red zone“ and also make “-18
dB” = 0dBFS, as this is exactly what happens once the
audio leaves the D/A-converter.
By setting the 0dB-reference to Digital “RED”,
DAW designers made a historic mistake that affects the
quality of audio to this day.
Prior to Digital Audio, people were not leaving a lot of
headroom in analogue as it would bring up noise, and
driving a signal hard to tape even had pleasant side-effects for certain types of music (tape compression, softclipping transients, added harmonics).
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Early Digital Audio was at 16bit Resolution (96dB of dynamics in theory), and people still felt it wasn’t clever to
leave a lot of headroom. They were partly right, as the
combined dynamics of 16bit and early converter designs
were barely reaching 90dB, best case.
From the early 2000s on though, the processing of ALL
DAWs was operating at a minimum of 24bit resolution,
which means you have 144dB of dynamic at hand at any
time. 144dB of dynamic at hand, and people complain
about not having enough headroom!
That of course exceeds what analogue circuits are capable of reproducing. Even the best OP-Amps used in
AD/DA-convertors can only go as far as 125dB (e.g. the
LME49720), so we can plan with as much headroom as
we like, it won’t affect the quality of the audio.
To cut a long story short – here’s what you need to start
doing:
GUIDE TO SOLID GAIN STAGING
STAGE 1 – TRACKING
When you’re tracking vocals or instruments, keep average levels around -15dB to -18dB, peaks shouldn’t go
higher than -8dB to -5dB.
STAGE 2 – YOUR INDIVIDUAL TRACKS
most individual tracks people send me for mixing are
too loud – once I start summing them, I’m at least 6dB
into the digital red. I do not compensate for that with
gain reduction in my master or summing bus!
With the channel fader set to unity gain (= 0dB), each
single element needs to have a solid headroom of about
12 – 18 dB. I recommend putting a simple gainer plugin in the second slot of your plug-in chain (always leave
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the first slot free). Producers usually send me their individual tracks pretty hot, with peaks close to 0dB, and
I have made it a habit to put a gainer plugin across every individual track that reduces levels by 15dB (set to
-15dB). That ensures lots of headroom in the master-bus
when mixing/summing the individual signals.

STAGE 3 – YOUR MIX BUS
The Mix Bus is where your Bus Compressor (if you’re
using one) is inserted, and it will have other processors,
from EQs to subtle Tape Compression, etc. (which is not
the topic of this chapter).
You need headroom coming into the mix bus (which
you have assured in „STAGE 2“), but also coming out of
the mix bus.
Take a situation where your mix sounds great and is very
balanced overall, but as you are comparing the mix to
references, it lacks bass. Like, the kind of bass you would
get by cranking up the bass on a HiFi Amp.
That is not really a big deal – I would probably insert a
Linear Phase EQ, and create a very broad boost of the
lower frequencies.
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This looks dramatic (+ 8dB at 30 Hz), but since this is a
linear phase EQ, and the curve is very broad, it’s not: 100
Hz is only boosted 2 dB relative to 400Hz.

This „not so dramatic“ EQ curve adds about 8 dB of level
to the mix bus though!
Added low end needs lots of headroom.
Each processor inserted on the mix-bus gives you – of
course – handles to get some headroom back. I’m talking about the Input and Output-controls of a plug-in.
Back off 1 dB on each of those to get some headroom
back if you’re getting near the trouble-zone.
When all processing is done, and you end up with 3dB
of headroom at the end of your mix bus, you’ve done a
solid job.
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More is nice, but what we’re talking about here is the
headroom you leave for the mastering engineer to do
his thing. In light of 20 years of loudness wars, these
guys are happy if you leave them ANY headroom.
They will love you for 0.3 dB of a ceiling and no brick
wall limiter used!
If you make it a habit to shoot for the -3 to -5 dB range
going OUT of the mix bus, you will have more space for
unexpected events though.
MORE THAN ONE MIX BUS
I make a clear distinction between MIX BUS and MASTER
BUS.
One of the reasons for that is – there are engineers who
work with SEVERAL master busses.
Without going into detail here, this is something pioneered by mix engineer Michael Brauer on SSL consoles that have several stereo mix busses, like the SSL
6000E/G, 8000G and 9000J/K.
Just as a quick example for adding a second mix bus:
your track has punch, solid glue and a great low end –
the bus compressor and EQ work great, BUT UNFORTUNATELY NOT for the lead vocals, they are too affected
and squashed by the bus compressor.
What you do is route them around your main mix bus
and create a second mix bus. The second mix bus (for
the vocals) will of course add some level, has it’s own
signal chain, and the sum of both mix busses feeds the
master bus.
For that to work, you need a healthy gain structure!
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STAGE 4 – YOUR MASTER BUS
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OK, the master bus is where you finally mess up your
audio to compete in the loudness war, right?
Oh well, I’m just joking of course – or not?
I personally always have a brickwall-limiter on my master bus, and thats about it.
I’m limiting maybe half a dB as a standard, not more.
Usually that gives me the average loudness I set out to
achieve.
If I want MORE RMS (average) level, I either reduce some
peaks on individual tracks, or carefully adjust the MIX
BUS, but NEVER do that by slamming the final Brickwall-Limiter.
For the final mix pass, I just take the limiter off, and print
a WAV (to hand on to the mastering engineer).
ITUNES SOUNDCHECK
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The easiest way to find out (in plain numbers) what type
of final loudness you want to achieve, is iTunes.
1.
launch iTunes on your computer
2.
go to the preferences…/Playback, make sure
„Sound Check“ is ON
3.
create a playlist with references of songs in a genre
similar to the song you're mixing. (we’ll talk more
about picking references in another chapter)
4.
“Get Info“ for the song (command + i on the
keyboard), chose the „File“-Tab and check the
„volume“-value
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5.
compare these values to the value that your mix
shows when you play it in iTunes.
If your references sit around – 7,4 dB, and your own mix
is at – 11,3 dB, you’re too loud. If your own mix sits at –
0,4 dB, you need to find about 5dB in volume (maybe the
Limiter is not making up for the headroom gain)
HERE’S WHAT YOU NEED TO KNOW ABOUT
ITUNES SOUNDCHECK AND MASTERING
LEVELS:
•
in iTunes (the app), the „Sound Check“-option that
automatically corrects all songs in your libary to similar subjective playback-levels DOES affect the prelistening in the iTunes Store as well – so no „Loudness-war“ happening on iTunes any more really.
•
ALL Apple-devices have the “Sound Check”-option
•
levels are not as crazy as they were, and differ from
genre to genre
SOME INTERESTING EXAMPLE S OF LEVELS…
The original album-master of “Billy Jean” gets almost no
level-change by iTunes.
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The remastered version of Billie Jean… (picture below)
duh, 8,5 dB louder!!
That screams “Loudness-War”.
The transients are really squashed on this one, and the
snare reverb brought up in a very unpleasant way. I can’t
listen to this!

Not too bad really for an EDM-record - I've seen many
records in this genre at -10 dB and more.

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Big mainstream hit-records don’t need to be squashed!

Another example for a dynamic mainstream No.1-hit.
What I’m proposing here is of course not new – mastering engineer Bob Katz has done incredible work in
educating audio engineers, and his „k-system“ dB-scales
are implemented in a number of plug-ins you should be
using to improve your gain-staging.
The K-system goes beyond levelling, it even includes
standardized listening levels.
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In short, what you can utilise is that iTunes has a very
intelligent algorithm that measures RMS, and adjusts ALL
tracks in a way that regardless of what song you’re listening to, they will all appear roughly in the same perceived loudness.
I think, they did a good job, and I’m planning to release
an album on iTunes with just test-material at different
levels to find out EXACTLY what happens.
The most important thing to know about this is that
when your mix is extremely squashed with high RMS and
little headroom, iTunes will turn it down and it will end
up lower than other tracks.
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FAQ
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When working with acoustic drums, I can find the
tone of the kick with the EQ and Analyser BUT how
would I tune it (the hole kick file) and if I tune it, wont
it be messing with the phase of the tone captured by
the OH and ROOM mics?
Absolutely correct, if working with a live drum set, you
can't tune the kick in the mix - it has to be done before tracking, on the actually drum skin. You can "semitune" acoustic kicks though by setting a resonance and
notches surrounding it.


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CHAPTER 06
IT’S ALL ABOUT THE
VOCALS
CHAPTER 06
IT’S ALL ABOUT THE
VOCALS
O
nce we have created a frame for our mix as described in Chapter 5, it’s time to focus on the vocals - in many cases the most important instrument in
the mix.
The human voice is the most complex instrument we
have to deal with, and you have to understand a number
of important concepts when dealing with vocals.
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A QUICK EXCURSION INTO VOCAL
RECORDING
Let’s have a quick excursion at vocal recording. Mix engineers often have to suffer and repair the results of a
bad vocal recording, and it’s worth pointing out common mistakes:
1. TOO MUCH TREBLE AND ESSES
Vocal sounds are a matter of trends. For years, the Sony
C800G tube microphone has been very popular due to
it’s emphasis and fine resolution on higher frequencies.
Without a doubt, one of the best mics ever made, but at
the same time, depending on the type of vocal, at many
times, there is excessive sibilance on the vocal-tracks.
A mic that emphasizes the highs is not the main source
of trouble here - it’s excessive sibilance pushed into a
tracking compressor that creates the problem.
I don’t really recommend a tracking compressor at all
when trying to achieve a vocal sound with crisp highs.
Along with more organic sounding records, recent years
brought a more organic vocal sound back, with less emphasized treble in the vocals.
2. NO POP SHIELD
If you don’t use a pop shield in front of your vocal mic,
chances are you will literally have dropouts in your audio
signal.
3. RECORDING IN A VOCAL BOOTH
Most small home-made vocal booths are not really
properly absorbing the sound across the entire frequency range. All they do is dampening the highs, leaving you
not only with a boomy sound, but also with comb-filtering that results in resonances across the mids. Vocals
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sound better recorded in a larger room, and it is a lot
easier dealing with a few more reflections from a large
room than removing resonances from comb-filtering.
4. USING A TRACKING COMPRESSOR
Again, if in doubt, don’t use a tracking compressor - it’s
a habit from the analogue tape tracking age. Set your
mic-pre and tracking levels carefully, and make sure you
have plenty of headroom even during the loudest vocal
parts of the song.
TUNING
Tuning vocals is more like fixing the performance than
it is mixing, but since it’s technically possible, it’s part of
our repertoire of tools and makes us and our work look
better when done at the right places.
I you haven’t done it, look into automating the responsetime of your tuning plug-in.
When using tuning plug-ins, avoid having all doubles
of the same vocal-line tuned. Especially on a backing
vocal, if you have two doubled takes, only tune one of
them. If you have more than two, only tune half of them,
and between the tuned ones, fine-tune them against
each other (for example: left BV at -5 cent, right BV at
+5 cent).
BUILDING THE
L E A D V O C A L- C H A I N
All of the following, please do at a low listening level. I
always use my small portable stereo-speakers for that.
Start by leveling the untreated lead vocal so that it sits a
little low in the track, just loud enough so you can understand the lyrics and follow the melody.
I say this now, and I’ll say it a few more times in this book:
keep going back between these building blocks for fine
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tuning. Just as an example, you will need to re-adjust
the sibilance once you’ve added the final „attitude“compressor. That goes for each building block. Go back
and forth between the building blocks and re-adjust.
Also, watch your gain staging - make sure the level that
comes in at each building block is similar to the output.
This chain has 3 dynamics processors with EQs in between, they are kind of „sharing“ the work that has to
be done. None of these building blocks need to do the
„heavy lifting“ on their own, each has their own role, and
less is more.
Bypassing one or more building blocks is an option.
Many times you don’t need all of them.
1.
CONTINUIT Y
Starting with the untreated, „naked“ lead vocal-track, I
want you to first understand the concept of having two
types of volume automation on a vocal: before the processing chain, and after it.
This first one evens out the performance of the singer.
You could say it creates the continuity of levels in the
vocal that you wished the singer had delivered in the
first place. This is something that overlaps and goes back
to mix preparation, and there is a fair chance that this
is already largely fixed. To confirm this, run the mix, go
through the song from start to end and find spots where
phrases or syllabels of the lead vocals drop in level, or
jump out considerably. Correct these phrases in whole
db-values - in other words, don’t go too much into details, don’t draw complex automation curves etc., this
is just one small step to even out the lead vocal across
the song.
2.
SIBIL ANCE
Do you feel the sound of the vocal leans towards too
much sibilance? Time to use a De-Esser, which is a specialized compressor that has an EQ in the sidechain resulting in reducing passages with a lot of treble between
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5-12kHz to be reduced in level. The frequency that you
want to reduce can usually be set, as well as the threshold (minimum level at which it starts working) and the
amount of reduction in dB.
Use an analyzer to determine at what frequencies the
esses are jumping out and reduce them considerably.
Alternatively, reducing the esses pre plug-in chain is a
more precise way to do this. In Logic I usually isolate the
esses to their own regions which I drag to a separate track
but same channel strip (so I can select all esses at the
same time). I reduce the gain on these regions between
6 - 10 dB, depending on how much the vocal gets compressed in the mix. You want to set them low enough to
run below the compressor/s threshold (referring to the
compressors to be added later on in the plug-in chain).
If you can get a De-Esser to achieve the same - use this
technique! You can also completely isolate Esses to another channel strip and EQ/compress them differently.
Same goes for breathing noises.
3.
WARMTH
With the vocal sitting a little low in the track, we now
need to develop a feel for what’s missing. We do that
by adding frequencies using EQs, starting from the low
mids.
A.
Broad boost between 200 - 500 Hz
The Pultec MEQ 5 is usually my first EQ in the vocal
chain, but you can simulate these (broad) curves with
many stock EQs that come with your DAW. I don’t ever
go lower than 200 Hz, and occasionally up to 700Hz.
The effect we want to get here is that the vocal gets
more weight and warmth in the mix. If the vocal is well
tracked, it comes with a lot of that quality in the recording and you may not need to do anything here. This is
why people use Neve 1073s and various tube-based
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equipment (from Tube Mics to EQs/Compressors) during tracking.
However, a lot of modern vocal recordings sound rather
thin, and a nice boost in the low mids fixes it. If you
like the character you’re adding with the boost, you can
even do a little bit too much off it. We’ll counterbalance
it in the next steps. In case the vocal already sounds
overly „muddy“ or „boomy“, add a Linear Phase EQ at
the beginning of the plug-in chain, locate and remove
the frequencies that cause this effect. Watch the interdependence of that - once you’ve removed resonances,
you have more leeway again to use that broad Pultecboost again.
B.
Tube compressor for tone
After the boost, insert a compressor for tone - we are not
after compression for level automation here, but want to
add harmonics on top of the low mid boost and create a
sense of glue. The classic Fairchild 660/670 works great
here, but don’t limit yourself - many compressors can
do the job. Just make sure that it’s not noticeable as
compression.
Light tape saturation can also work.
4.
PRESENCE
In a dense mix, this is were we create the frequencies
that make the vocal cut through the rest of the instruments. We can go to extremes here, but before you start
playing with the mid boost, set up the compressor that
follows it right away. It’s needed to tame the mid boosts
as they can get very harsh.
Often, less or no boosting in the mids is needed in less
populated parts of the song, but when the vocals are up
against a wall of sound, you will need a musically composed texture of „cut through“ frequencies there.
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A. Various boosts between 1k and 8k (SSL EQ)
This is more complicated to get right, compared to creating warmth. Start with an SSL-type EQ and boost the
high shelf at 8k +10dB, then pull back again to 0dB and
find a great setting for it somewhere in the middle. Try
switching between BELL and SHELF characteristics (BELL
will just boost around the set frequency, while shelf also
includes all frequencies above. If 8k is boosting sibilance
too much, go a tiny bit lower.
Continue by using the HMF band to boost at 4k. And
the LMF to boost 2k. Move these around until you find a
good balance - but keep in mind, not boosting anything
is always an option.
The goal is to create a cluster of mid-boosts that really becomes one colourful and musical texture of midboosts between 1 and 8k.
You can achieve good results with stock EQs of your
DAW, but there is a reason why SSL EQs are famous for
their musicality in the mids. API, Neve works as well. Not
a job for a Pultec.
B. Compression. This compressor’s job is to tame the
mid-boosts we just created. We want to see it work hard
and fast, but it needs to have a lot of musicality to reduce the harshness of the mid boosts.
My favorite one is the Gates or Retro Sta-Level, but I
also like the LA-3A. The LA-2A and Summit TL-100A can
work as well - if in doubt, try a few different ones.
The Sta-Level can reduce ridiculous amounts of gain
while you won’t notice any pumping, even when compressing 20dB or more. If in doubt, set the compressor
to less gain reduction. This is one step in the vocal chain
where the quality of the compressor makes a difference
in how much you can go to the edge.
In general, I love to give recommendations that are inexpensive, but in reality there are situations where highend equipment has the edge. This is similar to equipment
for video and photography. On high quality equipment,
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even a random grainy picture all of a sudden has an articistic vintage quality about it.
The goal of using this compressor-stage is to create a
balance of power between the warmth and presence
we created in steps 3 and 4.
5. AT TITUDE
Some sort of 1176 compressor can work at this stage,
especially if you need the vocal to cut through a dense
and/or agressive track. A blue stripe for more harmonics
and distortion, or a black face for smoother tones. There
are tons of interpretations of the classic 1176 compressor - you can try them all.
6. FINAL TONE AND
DYNAMICS CONTROL
A.
EQ
A final EQ can round off highs and mids. I like a Pultec
EQP-1a here, to boost at 20Hz, and Attenuate at 20kHz.
Your vocals will sound more analogue when you roll off
the top end slightly. Both the boost at 20Hz, and the attenuation at 20kHz should not affect the essence of the
tone you have created. The boosts can add little bit of
weight, and the cut removes top end energy that only
hurts at loud volumes.
B.
Brickwall Limiter
When optionally using the 1176 in stage 5, it might add
some sparkle, appearing as a mixture of transients and
added harmonics, so you can use a limiter at the end
of the chain to catch occasional peaks that jump out.
ONLY catch ocassional peaks, and set the release time
higher than 100ms. A modern digital brickwall limiter
works well here.
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IMPORTANT NOTE!
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The plug-in chain outlined above is pretty complex.
There is of course the possibility of processing the vocals too much with a signal chain like this. When you
want a really natural vocal sound, you might just need
3B (the subtle tube compressor) and 6A (Pultec for basic tone control). A natural vocal sound requires a very
good recording though, and we rarely have an influence
on that as mix engineers. On the other hand, the plugin chain is designed to be able to deal with any type of
vocal, and when the vocal needs to cut through a dense
mix, processing is absolutely necessary. However, subtle
settings in each plug-in can still cumulate in a drastic
overall effect, while retaining the maximum amount of
natural tone.
PARALLEL COMPRESSION
Chapter 8 is entirely dedicated to parallel processing,
but it's worth mentioning here - once you really learned
to work the vocal chain above, parallel compression will
bring your vocal skills to another level:
On most songs that have a certain dynamic, and specifically ballads that start soft and end in a big showdown
involving an entire band, orchestra, choir backing vocals, etc., I've made the following experience:
•
the vocal chain for the (light) first verse, that has
only a singer with a piano and (maybe) light drums,
is found very quickly. A natural vocal sound utilizing 3B and 6A in our chain often works very well.
•
as the singer hits higher (and louder) notes, I need
to add more elements of the vocal chain to control the dynamics, and also to tame certain resonances that come with loud and high notes
6
•
on the chorus, especially towards the end, the
vocal has to cut trough a wall of sound, and
several compressors are working in the chain
•
the solution for the dilemma are two separate vocal channels: a natural one, and a processed one!
•
anything between the two is using a mix of those
two channels - by automate both levels you can
create the right mixture of them for each section or
even single notes
•
use this technique when the singer performs across
a wide range of notes, switching into falsetto,
shouting, hitting rough rock notes, etc.
•
just use a separate track for each of those styles, so
you can vary settings and create a balance out of them
•
more details on this are coming in Chapter 8
VOX AGAINST THE WORLD
I have worked on songs where the backing track was
such a big and dense „wall of sound“, that the vocal
would not stick out of the mix, regardless the treatment
on the track.
There are two secret weapons for that:
1.
The Retro StaLevel Gold Edition compressor - while
this might be unuseful info for you, as nobody could
manage to make a plug-in version of it yet - if you
can get your hands on one of these, try the „Triple
Mode“ and a mid to fast time constant. It is my ab-
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solute favorite compressor for lead vocal in a dense
mix. The StaLevel is also useful on a lot of other
sources, sounds extremely clean and precise, and
I’ve seen it compress 30dB without messing with
the integrity of the signal!
2.
Multiband Sidechaining: there are a bunch of plugins that can do this, the most known one is Waves
C6 Sidechain. First check where the lead vocal has
it’s energy in the mids, this could be anywhere between 1k and 5k. Find the right frequency by sweeping through those frequencies with an EQ. Don’t use
that EQ on the vocal. Feed the vocal sidechain into
the multiband Compressor/EQ and set it so that it
reduces exactly this frequency broadly but in a subtle way.
BACKING VOCALS
All Vocals please sing the same song, at the same time!
Backing Vocals are a lot easier to deal with compared to
lead vocals - they usually come in groups of 3 and more,
and they have to sit behind/underneath the lead vocals.
Whereas we strive to create a lot of dynamics, and preserve transients on lead vocals, the backing vocals can
be more heavily treated. You can even merge several
layers of backing vocals into one file for easier handling.
However, there is one thing that you can’t be sloppy
with and that’s timing! I’m really sorry to reveal this to
you now, but you have to zoom into the waveforms, and
match the timing of the backing vocals and lead vocal
doubles to the lead vocal!
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While there’s several software tools that promise to do
this for you, you always have to double check if you
want a flawless result:
•
esses need to sit exactly in sync with where they
sit in the lead vocal - otherwise those messy esses will jump out of the mix left and right like the
snakes on my favourite movie „Snakes on a Plane“;
in addition to that, reduce them in level, they must
not interfere with the sibilance of the lead vocal!
•
breathing noises in backing vocals? pretty useless!
You might enjoy the lead singer’s breathing, but not
20 of them! remove them!
BACKING VOCAL CHAIN
As far as processing, backing vocals are a lot more forgiving. I usually copy and paste the lead vocal chain to
the backing vocals as a starting point, then costumize on
the following parameters:
(the numbers are referring to the plug-in order of the
lead vocal chain, which we use as a starting point)
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6
1.
Continuity
Equally important for the backing vocals - treat
same as lead vocals.
2.
Sibilance
Definitely use a De-Esser - backing vocals should
have a lot less sibilance compared the lead vocals.
3.
Warmth
A. The Pultec-boost
Boost at a different frequency relative to the LVboost. If the lead vocal has a „warmth“-boost at
300Hz, try 200Hz or 500Hz. If you have several
backing vocals or LV doubles, try to boost them
at different points, so that the boosts are spread
across a wide range.
B. Tube compressor for tone
Same as LV, but a bit more gain reduction.
4.
Presence.
Here’s where we go counter to the boosts of
the lead vocal. Reduce the frequencies you have
boosted on the LV. A. Reduce where the LV is boosted (SSL EQ)
B. Compression Same as LV
5.
Attitude = Bypass!
6.
Final Tone and Dynamics Control
A. EQ. Adjust this while running all vocals. Attenuate the treble even more relativ to the lead vocal. Try
to boost at 12kHz or 16kHz to create some „shine“ if
applicable to the genre. B. Brickwall Limiter. You can limit a lot more on
the backing vocal.
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FAQ
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Question on filters and eq etc... The question is general but I'll use a high pass filter as an example. Going on older desks and outboard I've used, there was
always some bleed that came through the filters,
but with advancements and especially in the digital
domain the cut is now sonically precise. I generally keep a little of the original there, especially with
stems because that's what I've become used to. To me
it feels less clinical but also ends up a warmer tone.
(or maybe I am imagining it) I even allow a tiny bit of
bleed into the vocal tracks! So thinking ITB, and with
all the tools there, is there any real danger of being
too clinical and is there a better way of avoiding it?
What you're referring to as bleed is simply a less step filter (12dB/oct was sort of standard in analogue); in general you did the right thing - however, the harder you
compress the signal later on in the plug-in chain, the
more important it is to really get rid of the low rumble
for example in a vocal, and in any recording made with
a mic; you don't want this stuff to come back at you and
start triggering the threshold of the compressor. So, yes
you could indeed end up with a less warm signal when
you cut hard; I personally cut very hard with a 48db/oct
linear phase eq plugin, and I go very high with the cut,
to the the point where I cut into the lowest note of the
vocal. I compensate for that with a Pultec tube EQ emulation as the first stage after that, and boost around the
range of the fundamental note (between 200 - 700 depending on the range of the vocal) to bring that warmth
back in a very controlled way.
When Mixing full album of varied dynamic material
- what’s the methodology of achieving a consistent
vocal level throughout. Would you start the first mix
using the more populated tracks as regards to instrumentation or the more open tracks with just stripped
back instruments and use that vocal level for the rest
of the album tracks?
Start with the more populated tracks, save plugin-chains
of your lead vocal and start with a similar setting on the
next song. Keep the songs you've already worked on at
hand as a reference.
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CHAPTER 07
CONTINUITY
CHAPTER 07
CONTINUITY
B
efore you create dynamics, you have to create control: we have a good foundation and groove at this
point, the central element in our mix - the vocal - is in
control, and we can now start defining the levels of all
the other elements of the mix.
The best place to start is a loud and dense part of the
song. If you deal with a song that has a traditional structure, that would typically be the chorus, so loop that and
try to find one level for each instrument that works best.
There is no particular order in which you build the levels
of the mix, you can just bring them all up at the same
time and then focus on one instrument at a time, going
back and forth.
„Before you create dynamics, you have to create
control"
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ONCE YOU HAVE ONE PART OF THE SONG
WORKING, YOU WILL MOST LIKELY NOTICE
THE FOLLOWING ISSUES:
Problem 1: The level that works in the chorus might be
completely wrong for that same instrument in a different songpart.
Solution: if the same instrument has completely different roles in different song-parts, create duplicates of
that track, and use a separate track for each „role“.
Example: guitar goes from picking in verses to power
chords in the chorus.
Problem 2: the level is not consistent even within one
song-part
Solution: this is what this chapter is about - the goal of
this chapter is to create a natural continuity in all the
sources from start to end. This is similar to the procedure described as „continuity“ on the vocals in chapter 6
- the same will now be done on all elements of the mix.
The continuity of ALL elements in the mix is the foundation for balancing our individual levels, and even further
treating the individual elements.
Important note - we are NOT creating any kind of DRAMATIC automation or dynamics yet - the focus is completely on sections or notes that noticable stick out or
drop in an unnatural way.
We are correcting and automating levels PRE and POST
plug-in chain.
PRE is to create a consistent performance, compensating for volume drops in the performance BEFORE building a plug-in chain, or taming parts that jump out.
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By correcting these issues throughout the track, compressors in the signal path will have to work less hard,
and will be sounding more natural.
The POST plug-in chain automation, which we’re NOT
dealing with in this chapter, is to create your dynamics for the song. A gainer plug-in before the automation
fader gives you access to the relative level of the entire
track - as this is much easier than always correcting the
automation.
Also, instead of creating automation for the verse guitar
(when it’s different ot the chorus guitar), just copy the
channel settings and have an extra channel for different
song-parts.
This chapter is about correcting levels PRE plug-in chain.
Once again, run the mix, go through the song from start
to end, one track/instrument at a time, and find spots
that drop in level, or jump out considerably. Correct
these in whole db-values - and again, don’t go too much
into details, don’t draw complex automation curves etc.,
this is just one first step to even out the performance of
each instrument.
•
•
•
•
as always, start „bottom first“ with bass
next guitars, or synth pads
orchestra-type instruments, strings, brass, etc.
sound FX last
In general, this is a perfect routine to use your little portable speakers for - with one exception:
creating consistency in the level of the bass is the one
place in mixing where a quality headphone is really useful. It has to be one that has a great low-end balance I’m personally using a Beyerdynamic DT990 Pro for that.
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Once you’ve done this on all tracks, use the channel faders to find a rough balance for all instruments that you
like.
CAREFUL LIMITING
On the odd occasion where you have reduced a peaking
level in a track, and your level correction at the source
still sounds like an unnatural jump in level, reduce the
peak less so that it still sounds natural. Try a compressor
at a 2:1 ratio with the threshold set so it catches only that
particular peak. Adjust attack/gain for a natural sound.
NULLING YOUR FADERS
This is the final step of this Chapter. „Nulling your faders“ means that you end up with all faders in „default“position while retaining the balance you’ve created.
We're doing this in preparation of the final mix automation which will follow A LOT later in the process. However, when our channel faders sit at 0dB, overall levels
are a lot more easy to manage, as the fader has a much
higher resolution around its default-position.
Here’s how to do it:
1.
Look at the value of each track’s fader, for example
„- 21dB or + 2dB“
2.
Insert a gain-plugin as the last plug-in in your
chain, and copy the value above.
3.
Set your fader to default position (= 0 dB)
4.
As you progress with your mix, you might have to
go through the "nulling procedure" one more time.
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CHAPTER 08
PARALLEL
PROCESSING
CHAPTER 08
PARALLEL
PROCESSING
H
ave you ever listened to a great mix and wondered
how the vocals were upfront and in your face, had
tons of attitude and at the same time still sounded natural and relatable?
Many times, the secret behind that was Parallel Processing aka Parallel Compression.
Philosophically speaking, you combine a number of
specialised techniques and technologies to achive one
goal - similar to driving a brand-new S 600 Mercedes
that is using an endless list of crazy technologies, but
most of them you never recognize: the car is fast like
a Ferrari, smooth like a Rolls-Royce, comfortable like a
sofa, and safer than Fort Knox. You just keep wondering
how they did it.
„The beauty of Parallel Processing lies in
the simplicity of the concept.“
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AT THE CORE OF IT, HERE’S WHAT
PARALLEL PROCESSING IS ABOUT
let’s take a Lead Vocal-track for example:
01.
Duplicate your original vocal-track, so it
playback on two audio channels.
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02.
Leave the original version untreat ed
(= no plug-ins), but treat the shit out of
the duplicated track, e.g. compress it hard,
add frequencies that give the vocal some
attitude (just as an example – add whatever you’re after). While you’re treating the
duplicate, mute the original, and have no
mercy – go to extremes with the treatment, to the point where you couldn’t use
the signal in the mix on it’s own.
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03.
Turn the duplicated track fully down, then carefully start blending it with the untreated original
by slowly bringing the fader up. Use this the way
you would add sugar and salt when cooking –
it’s about subtly pushing the original signal in a
certain direction without affecting it’s integrity
and natural dynamics.

That’s really easy to do, right?
Congratulations, you’ve just learned Parallel
Processing.
„Duplicate the track you want to compress.
Compress the shit out of A, leave B clean.
Mix.
HOW TO CHECK IF YOUR DAW SOFTWARE CAN DO THIS
The technique of Parallel Processing aka Parallel Compression was first used in the internal circuit of Dolby A
Noise Reduction (introduced in 1965). In 1977 it was first
described by Mike Bevelle in an article in Studio Sound
magazine, and later dubbed “New York Compression”,
around the time when mixing consoles started to feature an extensive routing matrix.
On the SSL 4000 E/G-Series, which was first introduced
in 1979, you simply press any of the channel’s routing
buttons, for example Ch. 1 (like shown below on Channel 9) and get a duplicate of your signal through the
large fader on the channel you just selected (by choosing Subgroup on that Channel as your input source).
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WHY DOES THIS NOT
WORK ON MY DAW?
While duplicating a track is something you can do in every DAW, early DAWs had issues with Parallel Processing.
People still have trouble with it on some platforms, but
it works great in Logic Pro X and most others, and I’ll
guide you through the necessary changes in the audio
preferences.
HERE I S THE PROBLEM:
Every plugin you’re adding to a track adds a tiny delay
to the signal. Time required for the plugin to calculate
the audio.
BUT: your DAW has settings in the Audio Preferences
that will make sure the software is compensating the
exact delay that the plug-ins create – what they do is
starting the audio of a track that has a lot of plug-ins
ahead of the others, so it comes out exactly in time…
If those calculations are not precise, Parallel Processing
won’t work, as you’ll get phase cancellation between the
two similar signals you’re mixing.
Even a tiny delay of one sample will render the technique useless!
Today’s DAWs have a feature called “Plugin Delay Compensation”, and you need to make sure it’s activated:
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To take Logic as an example, we have 3 options for the
“Plug-in Latency Compensation”, as Apple calls it:
•
•
•
Off (no compensation)
Audio and Software Instrument Tracks (compensates Audio and Software Instrument Tracks, but not
Auxes or Output Objects)
All (compensates everything)
I usually use the „Audio and Software Instrument Tracks“Option during Recording & Arranging, and switch to ALL
when I’m mixing.
The software will now fully timealign ALL Audio Tracks,
Software Instruments, Aux Busses and Outputs.
Which means that I can use an Aux to send my Original
Signal to another Channel Strip, and treat that signal like
the duplicated track in the example above. Very similar
to how you would do this on an analogue console.
Of course, the analogue console does all this without
any latency. Which means, you can use all these tricks
even during a live recording. To assure that you’re on the
safe side, you can do a test with your DAW software. Add
a bunch of EQ plug-ins to the duplicate track, but do not
EQ the signal (leave the EQs switched in but flat), and
invert the phase polarity on one of the tracks – in the
picture below I’m using Logic’s “Gain”-plugin:
As you can see, the result of playing two similar tracks
back (one of them with inverted phase) should be silence on the output – when you reverse the polarity on
one of two similar audio tracks, they cancel each other
out.
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“The concept of Parallel Processing is to design two or several signal chains of the same
source that go for different aesthetic goals”
THE PHILOSOPHICAL DIMENSION
Parallel Processing is the solution to a problem that I
remember experiencing with compressors since the first
time I used one in the 80s (a cheap dbx 163X):
1.
on extreme compression settings, the signal has
great punch, tone and attitude, but at the same time
you really can’t use it like that, it just sounds too
manipulated and processed.
2.
without compression, the audio sounds clean and
natural but lacks punch and attitude Parallel Processing takes some weight from the mix engineers
shoulder – by separating the clean and compressed
audio, the audio-source becomes easier to manage.
You start to take more risks during your mix – you
can just experiment on an additional parallel-chain
for fun. If it doesn’t work – throw it away! You never
feel like you’re messing with the original signal.
Be careful though - often, when you have a hammer,
everything looks like a nail.
Parallel Processing is not for all of your signals in the
mix - some instruments work well when transients are
squashed by compression or limiting.
You can’t have too many transients compete with each
other in the mix. Examples would be backing vocals,
doubled lead vocals, keyboard pads, heavy guitars - on
those, transients need to be controlled.
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If you spread a group of guitars or backing vocals over
the stereo panaroma, you can contrast a track with
strong transients with a double that has „squashed“ transients.
The key is to create a healthy balance between excitement and steadyness in your mix. The typical cases that
require both at the same time are lead vocals, and drums.
PARALLEL PROCESSING ON VOCALS
On the first example, at the beginning of this article, I
left the original signal completely untreated. That was of
course an extreme example to illustrate the principle of
parallel processing. While you might not use any compression on your clean (original) signal, you still want to
make sure the audio delivers a consistent performance.
Never use a parallel chain to be a substitute for automation or other audio housekeeping.
Use automation for parts that drop too much in volume,
and never forget to do the necessary EQing / filtering
(like removing low end rumble or doing some DeEssing).
The concept of Parallel Processing is to design two or
several signal chains of the same source that go for different aesthetic goals – here are some ideas:
A–
B–
C–
D–
E–
treat for uncolored, natural, dynamic
treat for extreme, coloured, saturated, or even
distorted. Of course, you can create even more
duplicates of your signal that are differently
treated – some examples:
run through a guitar amp plug-in for distortion
separate breathing noises from the actual
„tones“ in the vocal, and keep as untreated and
natural as possible (use automation to level bits
that jump out)
create a duplicate specifically to send to Echos
and Reverbs
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I like to think of Echos and Reverbs like a blurry shadow
on a photography – hence DON’T send to them from
your „upfront/punchy“ vocal channel. Create something
more cloudy/blurry that has a stark contrast to your direct vocal signal.
PARALLEL PROCESSING ON DRUMS
I gave you plenty of ideas in the sub-chapter on „Mixing
Kicks“. Just a few examples:
•
create a duplicate to bring out the transient
•
separate the duplicates into different frequency
ranges
•
use a duplicate to bring out harmonics (check my
post on that)
Try sending your entire Drum Submix to a Parallel Compression Chain. This can give your drums great stability
in the mix, while transients are unaffected.
My starting point on the SSL-console has 4 parallel compressors set up.
•
1 for the kick
•
1 for the snare
•
a stereo pair for overheads and/or room mics
I send to them from an aux send - in my particular case,
using the small fader on the SSL as an aux send via the
routing matrix.
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CHAPTER 9
COLORS, DIMENSION
+ THE DYNAMIC OF
YOUR MIX
CHAPTER 9
COLORS, DIMENSION
+ THE DYNAMIC OF
YOUR MIX
Let’s look back on the what we have tried to achieve in
the previous chapters:
•
improving room acoustics and listening experience
that allows for an „objective as possible“ judgement
of our mix
•
creating a well organized mix session in our DAW
that assures we can apply our sonic ideas quickly
•
building a strong foundation for our mix by balancing the low end, and saveguard our creativity by always retaining plenty of headroom
•
giving special attention to the lead vocals, assure
they have a round tone, cut through the mix, and
have plenty of attitude
•
creating continuity throughout the mix
•
using parallel compression to add weight and impact on our most important elements in the mix
We now have a very solid foundation, but the mix still
sounds static and one dimensional at this point - which
is what we’re working on in this chapter to finalize our
mix.
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EQ-ING
Here’s something you need to get your head around.
We’re mixing MUSIC.
Try to see the following in every element of your mix:
•
fundamental note/tone
•
harmonics on top of that (often in form of a triad)
•
a noise component added to that, mainly in the high
frequencies
The frequencies in your mix should form a smooth texture where the different instruments add up to a rich
spectrum of colours.
Don’t take this too literally, but think in musical terms
when EQing, and keep that in mind while we go through
the tools available to create this.
Boost frequencies that build a triad, spread wider in the
low register, and you can go more narrow in the higher mids. EQ-ing treble is a matter of asking if you want
highs on this instrument or not? If yes, how much until
they hurt your ears?“
The chart below can help, but always keep in mind, don’t
take this too literal.
http://en.wikipedia.org/wiki/Piano_key_frequencies
Before we have a look at the important types of EQs,
and where to use them, know that you will need to use
a combination of these to cover all your EQ-needs in a
mix. If you are not yet familiar with those classifications,
take some time to explore the possibilities of these on
different sources.
In the context of setting final levels in the mix, which
we get to at the end of this chapter, EQs can be used
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in a very basic way. When you level for example a piano
or guitar in the mix, this is as simple as having the upper range of the instruments, including the noise component (piano hammer noises, guitar picking noises) sit
right in the mix first, and then use a broad EQ between
200 - 400Hz to adjust the lower range of the instrument
by boosting or cutting.
„PULTEC“-T YPE EQS
Probably the EQs I personally use the most in my mixes
- the hardware-versions of these are tube-compressors
that come in two basic types:
1.
the EQP-1 „Program Equalizer“ can boost and reduce
bass at the same time (in steps from 20 to 100Hz),
and has similar controls plus a bandwidth-parameter for treble (switchable boost for 3, 4, 5, 8, 10,
12, 16kHz, Attenuation/Reduction for 5, 10, 20kHz).
2.
the MEQ-1 „Midrange Equalizer“ is an EQ for mids
- it has a boost (here called „Peak“) for selectable
lower mids (200, 300, 500, 700, 1000Hz), a selectable „DIP“ (reduction) for 200, 300, 500, 700Hz, 1,
1.5, 2, 3, 4, 5, 7kHz and another boost for high mids
(switchable between 1.5, 2, 3, 4, 5kHz).
Pultec-type EQs are great for shaping tones in a very
natural way. It is difficult to get a bad-sounding result
out of them. Even when frequencies are fully boosted,
the boost still has a smooth and natural character about
it. The reason for this are the Pultecs broad EQ curves.
Even when you boost below 100Hz, the boost reaches
up to 700Hz. While these EQs have their own character, you can learn from them even if you don’t own one.
Simply try out broader curves (smaller Q-factors) with
the EQ you have at hand.
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The EQ-part of the Pultec consists of „passive“ electronics that reduces gain internally, and the tubes are used
for a 2 stage line amplifier to make up for the gain lost in
the passive EQ-circuit.
There are a couple known variations of these from
known manufacturers, and an almost endless number of
plug-in versions.
While the original hardware-versions are amongst the
most expensive EQs you can buy for money, plug-ins
are of course a way to use these on pretty much every
source in your mix. Note that Pultecs add a very desirable and subtle tube saturation to your signal even when
the EQ is set flat.
CL ASSIC CONSOLE EQS (SSL, NEVE, API)
These are the EQs found on the most popular large format recording consoles from the 1970s until today. You
don’t need to own a recording console as all of these are
available as hardware from the original manufacturers,
for example in the popular 500-series format.
Console EQs were designed to be able to shape all kinds
of signals. They usually have a shelf EQ for the lows and
highs, and 1 or 2 bands or fully parametric EQs for the
mids. These can often reach as high or low as the low
and high shelf EQs.
Today, most people learn about and use them in the
form of plug-ins, some of them developed with SSL,
Neve or API.
I might be a bit simplifying here - but you mainly use
these when you want to boost or shape a sound more
narrow, or more agoressively than what can be achieved
with the Pultec-type EQs.
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LINEAR PHASE EQS
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These are digital EQs, and they were first introduced
as super expensive digital outboard boxes for mastering engineers, who have used them for many years. Like
everything expensive and digital, it is now available in
plug-in form, and for example Logic Pro has a very good
linear phase EQ that comes with the software.
There is a ton of technical info on linear phase EQs on
the web, none of which will help you improve your mix.
One thing they all have in common is adding a significant latency to your signal that needs to be compensated by your DAW. This is a problem when using it on
live instrument, but not in mixing. As shown in the chapter on parallel compression, just make sure your plugin
delay compensation is switched on across all types of
audio tracks, and you’ll be fine using it. The reason for
the added latency is that instead of „post ringing“ which
we see in traditional EQs, the linear phase EQs adds „pre
ringing“, which in turn keeps the phase response linear.
All you need to know is that the linear phase behaviour
makes these sound more neutral and less drastic. They
don’t add harmonics and resonances - their effect is totally isolated to the frequency range you have selected.
They can be used for boosting and attenuating, both
broad and narrow.
FILTER S
To list filters here is somehow redundant. Filters always
come in the package of most EQs, except the Pultectype. You use them to remove frequencies below a certain frequency (HPF = High Pass Filter = high frequencies
are „allowed“ to pass) or above (LPF = Low Pass Filter =
low frequencies are passing).
The most popular application is a HPF on vocals, to remove low rumbling frequency, commonly below 60 120 Hz.
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On many, if not most of my DAW-channels, I’m using
ALL of these EQs at different stages of the plug-in chain.
A DIFFERENT LOOK AT
COMPRESSORS
We all have a basic idea of what a compressor does and
how to use it, right?
I’ve googled „what does a compressor do?“ and the
„top“-results are pretty much all similar but still all wrong.
Something along these lines:
„Compression controls the dynamics of a sound, it raises
low volumes, and lowers high volumes“
If you’ve read a few of my articles, you know that I’m
questioning some things – and if necessary hit you with
raw data to destroy popular myths ;-)
This sub-chapter is about compressors as a tool to
shape the tone of a signal via adding harmonics. There
might still some level correction involved, but as pointed
out in Chapter 7, correcting drops or peaks of level at the
source is preferable to using a compressor for that.
Personally, I think of different models of compressors in
terms of „how they feel“. The choice becomes intuitive,
as a compressor imparts a distinct characteristic on a
sound, pushes it into a sonic direction.
It took me years of practise to develop that feel for certain types of gear. It’s more difficult to develop when
you use only plug-ins, but you can still get to similar
results with both hardware and plug-ins.
The original hardware counterparts are different in that
they show a lot more color and distortion when you
drive them to extremes. Those extremes helped me to
learn the characteristics.
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Since that won’t help you - unless you have access to a
studio with a large analogue outboard collection - this
post is taking an analytic look into popular compressor
plug-ins and their characteristics.
TEST S ETUP & PROCEDURE
Let’s run some popular compressor plug-ins through a
test setup and procedure, then look at what the results
tell us!
(BTW, I’ll give you the test setup as a Logic-Template for
download so you can test your own plug-ins – just subscribe to the newsletter!)

THE TEST OSCILL ATOR IN LOGIC PRO X
FEEDS A COMPRESSOR WITH A TEST-TONE
•
•
•
•
the test tone is a sine-wave (as you know, a sinewave has no added harmonics).
we cycle through 55 Hz, 110 Hz, 220 Hz, 440 Hz
tones
then a sweep from 20 Hz to 20.000 Hz
ending the cycle with a 100 Hz tone
WE CYCLE THROUGH THIS 3 TIMES, WITH
RISING LEVELS
1ST CYCLE
•
•
•
Oscillator hits Compressor with - 18 dB of level
Compressor Threshold is set JUST BEFORE compression
for the compressor NOT to compress (unity gain)
2ND CYCLE
•
Oscillator hits Compressor with - 12dB of level (6 dB
more than on the previous cycle)
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•
Compressor settings stay the same, but of course
compression now kicks in!!
3RD CYCLE
•
•
Oscillator hits Compressor with – 2 dB of level (another 10dB added on top of the previous cycle)
Compressors settings remain the same, but now
hitting compression quite hard!
The upper track you can see in the videos is the automation curve for the Test Oscillator’s frequencies and
levels, the lower track shows a huge analyzer plug-in
after the output of the compressor (using Logic Pro X’s
Channel EQ), and that shows as the frequency spectrum
in realtime.
BTW: – 18 dB in your software is a GREAT average level
for your recordings and signals in ALL situations. It assures clean and pristine sound and compatibility with all
plug-ins. More on this another time when I hit your head
with a bat called GAIN STAGING.
Summary: on the first cycle, the compressor doesn’t
actually change the level of the signal, on the 2nd cycle
there is some compression, and on the 3rd cycle: a lot.
Every cycle ends with a 100 Hz tone – that makes it easy
to read the added harmonics on the analyzer.
2nd harmonic = 200 Hz
3rd harmonic = 300 Hz
4th harmonic = 400 Hz
5th harmonic = 500 Hz
n th harmonic = 100 Hz x n
Just for reference, here’s what the test procedure looks
like with NO COMPRESSOR inserted in the signal path.
https://www.youtube.com/watch?v=fPChmBi3yKU
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As you can see, the analyzer just shows basic sine-tones,
with no added harmonics. Music theory and physics calls
this is the 1st “harmonic” – but don’t be confused, that
is the term for the original frequency of the sine-tone.
FAIRCHILD 670 COMPRESSOR (1959)
– THE ROYAL HARMONICS ORCHESTRA
To give you a proper contrast – here’s what this looks
like with a plug-in clone of the legendary Fairchild 670
compressor, as some of you might know, the most expensive and sought after vintage tube compressor on
the market.
https://www.youtube.com/watch?v=6IR0nRGjOms
I bet you that the designers of this plug-in looked at a
spectrum like that forever, and did endless coding and
testing until the plug-in matched the original hardware
closely.
You can already see some harmonics even when the
compressor doesn’t compress, but they really kick in the
more you compress.
Note: the lower the sine-tone, the more harmonics show
up – I can count 13 added harmonics in the 3rd cycle on
top of the 50 Hz sine tone. When the oscillator reaches
2000 Hz, the Fairchild doesn’t add more harmonics on
top, at least not visible anymore in the spectrum.

If you look at the rich harmonics added by the Fairchild,
you start understanding how it gives a dull bass-sound
or a 808 subsonic kick a richer frequency spectrum.
This is very useful as it helps low-end to subsonic sounds
to translate better on smaller systems (think laptop, tablet, smartphone, kitchen radio).
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At the same time, a Fairchild might not be the ideal compressor to purely control volume, because the more you
compress, the more it changes the sound of the source.
This is not typically what you need if your goal is to level
something that is dynamically uneven.
On the contrary, you want to make sure that your source
is already in control dynamically BEFORE you even hit
the Fairchild.
There are other ways to achieve a consistent loudness
in a performance.
Look at waveforms of your recording and just bring the
lower parts up in level, reduce loud sections, automate,
and a lot of times, a consistent level is what a great performer brings to the table!
If your signal is well-levelled and even, you can alter its
tone by the amount of drive the signal into compression.
I’m almost ready to go into detail on parallel compression at this point – imagine a setup where you’re bring
the same low 808 Kick into two mixer channels.
The first channel kept unprocessed, the second channel pushed hard into a tube compressor like the Fairchild – the added tube channel will make the 808 come
through on smaller systems and adds a nice texture to a
sound thats pretty close to a sine-tone.
On the parallel channel you can even cut off the low
end and just add the harmonics (of course cut off POST
compression) – more about that in Part 2 of this article.
Essentially, what the added harmonics do is adding frequencies to the original sound that weren’t there before.
PUT THE COMPRESSOR IN THE MIDDLE OF
THE SIGNAL-CHAIN
A typical signal chain around a compressor like the Fairchild would look like this:
Source Signal
-> Surgical EQ (to remove unwanted frequencies)
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-> Compressor (adding harmonics)
-> EQ (for color and tonal balance)
PRE COMPRESSION:
SURGICAL EQ REMOVES
UNWANTED FREQUENCIES
It’s very important to put an EQ BEFORE the compressor. Use this EQ to remove unwanted frequencies.
Typical example would be a High Pass Filter that removes rumbling impact noise on vocal recordings.
Imagine how the Compressor would add harmonics to
a rumbling noise at 30 Hz and really bring it out – you
don’t want that.
Same goes for unpleasant room resonances – find them
using a narrow EQ boost then set a small notch to remove them.
This so-called „surgical EQing“ works best with „Linear
Phase EQs“ – many plug-in manufacturers make them,
they don’t add coloring resonances. I like the one that
comes with Logic Pro X a lot.
POST COMPRESSION:
EQ FOR COLOR AND TONAL BAL ANCE
Going back to the example of a low 808 subkick, which
is a sound that is very similar to the sine-tone used in
our test, there would be no point in EQing a pure sinetone, right?
You can’t add frequencies that are not there – in contrast to a tube compressor, an EQ does NOT generate
any frequencies, it can only adjust the tonal balance of
the given frequency-content.
With that, we have once again turned common audioknowledge upside down:
•
Compressors can add color to any frequency
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•
EQs are static, all they do is adjust the tonal volume
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That rule is of course not totally holding its own once
we look at a few more types of compressors. What we’re
interpreting in this article focusses on frequencies and
harmonics, which is just one aspect of compressors. The
other one is the actual ability of a compressor to level,
limit or „grab“ a signal, attributes that all refer to the volume of the sound.
UNIVERSAL AUDIO/UREI L A-3A (1969)
https://www.youtube.com/watch?v=dUoaFsZb4R0
The Waves CLA-3A is a plug-in clone of the original
Universal Audio LA-3A Compressor/Limiter.
In contrast to the Fairchild, it’s a lot better suited for levelling a signal.
The LA-3A adds only one harmonic (the 3rd one). The
Fairchild and the LA-3A can co-exist in a signal chain.
Use the LA-3A to even out levels, then hit the Fairchild.
The LA-3A is typically used as leveller for bass, guitars
and even vocals.
Less suitable for percussive sounds – it’s not following
fast enough to control a drum sound.
TELETRONIX L A-2A (1965)
https://www.youtube.com/watch?v=F5W-fP-MCno
The Waves CLA-2A is a clone of the Teletronix LA-2A.
The design is a few years older than that of the LA-3A.
It adds more harmonics than the LA-3A, but still a lot less
than the Fairchild.
Typically used to control bass, backing vocals or laidback lead vocals. A fairly slow and laid-back tube compressor.
UNIVERSAL AUDIO 1176
REV. A BLUE STRIPE (1967)
https://www.youtube.com/watch?v=YPmKEMYWfS0
The Waves CLA-76 is a clone of the Universal Audio/
UREI 1176.
Various revisions were made, the „blue stripe“-version
being the first one ever created.
Nearly every plug-in manufacturer offers a clone of the
1176. I like the ones made by Waves, and they got their
name because Waves developed them with Mr. CLA aka
Chris Lord-Alge.
The 1176 displays extremely rich harmonics. In comparison to the Fairchild 670, it sounds a lot more agressive
and levels superfast.
That makes the 1176 very flexible – it can be used on
almost any source.
Like the Fairchild, the 1176 is a true studio classic and
it would be worth writing a dedicated chapter about it.
If you have a bunch of those in your rack (or a great
plug-in clone in your collection), you could mix an entire project exclusively with those.
One of the things it works very well for is making vocals
agressive and bring them upfront.
You can drive it hard into a lot of compression, and as a
result will see a lot of harmonics.
It would not be the only compressor in the chain, I’m
usually running another compressor for levelling before
the 1176.
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SOLID STATE LOGIC SSL E/G-SERIES BUS
COMPRESSOR (1977)
https://www.youtube.com/watch?v=vf1daI2SNrg
This is of course one of the most famous compressors
ever build.
Waves teamed up with SSL to create one of the first true
emulations of an original hardware, and this plug-in (as
part of the Waves SSL-bundle) is now a classic, just like
the original SSL 4000E and G-Series consoles.
It does – of course – a great job levelling a signal, and
adds more harmonics the harder you hit it.
The trick with the SSL Bus Compressor is hit compression with your peaks in your finished mix, e.g. the kick
drum.
What happens is that the SSL „grabs“ and reduces the
peaks in a very clever way, while adding harmonics to
them.
The SSL bus compressor controls the dynamics and
makes the bits it compresses more punchy by enriching
it with harmonics (almost like compensating for the lost
level).
This effect has widely been described as mixbus-“glue”
and the reason why everybody loves SSL Bus Compressors.
SOLID STATE LOGIC SSL E-SERIES
CHANNEL COMPRESSOR (1977)
https://www.youtube.com/watch?v=qJdCSCbZNVo
The SSL Channel Compressor, also a part of the Waves
SSL-bundle, adds a healthy portion of bright and agressive harmonics, and is capable of controlling and “grabbing” percussive signals like no other compressor.
Widely used by famous mix engineers on Kick, Snare and
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any type of percussion – the SSL gives drum sounds a
prominent place in the mix, makes drums punchy and
cut through.
SUMMIT AUDIO TL A-100A (1984)
https://www.youtube.com/watch?v=V_TxskKMd3I
The Summit TLA-100A is a very subtle tube compressor.
The original analogue hardware has been used by engineer Al Schmidt as a tracking compressor on many of his
Grammy-winning projects, for example on Diana Krall’s
vocals, to catch some peaks with light compression during vocal recording.
The Summit adds some harmonics when you drive it,
and works well on a wide selection of sources with 3
easy settings for both attack and release-time.
A very subtle leveler for tracking and mixing.
LOGIC PRO X COMPRESSOR
(PL ATINUM MODE, 1996)
https://www.youtube.com/watch?v=JLXcXrclbCs
This is the original compressor plug-in that came with
the first version of Logic, so the design goes back to the
mid-90s.
It’s actually the ONLY plug-in in our test that does not
colour the signal AT ALL. Contrary to all other compressors tested, this is a compressor suitable for applications where you want to iron out the dynamics of a track
without adding harmonics.
The later versions of this plug-ins (like the current one
in Logic Pro X) added a few more modes you can select,
and when you switch from the “Platinum Mode” in any
of the others (like “VCA”), the plug-in starts adding harmonics, trying to mimic some of the compressors I just
introduced.
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U-HE PRESSWERK (2015)
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If you don’t have a huge collection of outboard and/or
plug-in compressors, you can start with just one that
delivers a broad range of application. I personally very
much like u-he PRESSWERK which is currently the only
compressor plug-in on the market, where the amount,
dynamics and shape of the added harmonics can be set
independently from the amount of compression.
If you look at the block diagram, you will see that PRESSWERK unites all the features and topologies found in
classic 1950 – 1970 compressors, while giving the user
full control over these features and clear labels to access
these in detail.
I can see PRESSWERK becoming popular in audio
schools, as there’s no other plug-in that will make it that
easy to educate someone about the details of ALL vintage compressors in one plug-in.
All of the emulations I’ve tested earlier in this chapter,
except one, generated harmonic distortion, and these
would usually increase in level the more gain reduction
(= compression) you’re applying.
With PRESSWERK, you have total control over the
amount of harmonics that are added, and the DYNAMICS-control can seamlessly adjust between „depending
on the volume of the source material“, and „harmonics
always added“.
The basic harmonics added look somewhat similar to
these of a classic UREI 1176 blue stripe, but the various
controls of PRESSWERK’s saturation section can highly
customize them, for example apply them either PRE or
POST compression, statically or following the dynamics of the material, and you can also balance the spectrum of the harmonics slightly using the COLOR-control
which is essentially a „tilt“-type filter/EQ.
When you watch the clip, pay attention at me playing
with the Saturation-section of PRESSWERK starting at
around 1:11min.
https://www.youtube.com/watch?v=HCG_H7_v1e4
You can download a demo of PRESSWERK here:
http://tinyurl.com/kcbjjky
Saturation is of course something that occures - more
or less - in all analogue signal processors, not only
compressors. It’s worth mentioning a few more „standalone“ boxes/plug-ins that can be used to generate harmonics via saturation.
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TAPE SATURATION
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Reel to reel analogue tape machines - remember those?
A magnetic recording „head“ magnetizes a magnetic
tape that can run at different speeds. Analogue tape,
when pushed with extreme levels, doesn’t distort the
same way as a digital converter does. The more you enter the „red zone“ of a tape machine, you get an effect
called „tape saturation“. The sound of tape saturation
varies of course with the type of tape recorder, width of
tape, and tape speed.
The classic analogue studio standards by STUDER or
AMPEX are still loved for their sound, but rarely ever
used these days. Plug-in emulations have come a long
way and do a really good job getting this type of sound
in your DAW.
MIC PRES
Microphone Amplifiers or short „mic pre’s“ come in flavours similar to compressors (minus the gain reduction
circuit) - they can be build with tubes, transformers,
transistors or modern OP-amps, or a mixture of all of
these technologies. Some of them can also be used at
line-levels, and also make a great colour in your palette
- especially if you do vocal recording as well.
The various Neve-models come to mind, they are
amongst the most legendary studio classics. While the
original versions are extremely costly, both AMS Neve
and various other companies build stripped-down alternatives such as 500-series modules.
The classic Neve 1073 is also widely been modelled as
a plug-in.
MAGIC CHAINS
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As you expand your knowledge about processing and
plug-in chains, I recommend that whenever you have
found a combination of settings that works in a mix, save
the entire channel strip as a plug-in chain in your DAW.
Make sure the name you’re using reminds you about
what the application for this preset could be.
Over time you will develope a library you can always go
back to and further refine.
REVERBS AND DEL AYS
Most producers have a very basic set of Reverb Sends
and Returns in their DAW sessions: a standard reverb
(mostly plate or concert hall), one delay for throws, and
maybe a small room simulation with early reflections.
All the tracks in the arrangement are accessing the same
reverb and delay, if needed.
Thats totally fine for arranging and producing, but forget that concept in mixing.
If you want to create an impression of a 3-dimensional
mix, with lots of space and separation, here are some
things you need to think about:
•
•
•
•
you need a huge palette of different reverbs and delays, add to that some modulation FX
the reverb you end up with will be a mixture of several reverbs that differ in many ways, from colour
and density to reverb time
the most important tracks in your mix are accessing completely different reverb and delay sends and
returns
you need to have banks of different reverbs ready in
your DAW template, specialized for different sources like drum room, snare, toms, lead vocals, pads,
lead synth, orchestral instrument
•
•
•
•
group the reverb returns together with the instruments they belong to - if you have a drum subgroup
or VCA group, the reverb returns for the drums belong to that, and exclusively to that
also, you don’t need to create a new send for every
reverb - you can send, for example, to 20 different
snare reverbs from „Send 5“, and then just unmute
and mute them at the aux return, one at a time, until
you find what you’re after; this is a very fast and intuitive process, and gives you the option to use one
or several reverbs at the same time, and level their
balance at the different reverb returns
for subgroups or VCA groups to really work, FX
belong mostly exclusively to their sources - if you
mute the drum group, you mute all drum reverbs
with it, but of course NOT the vocal reverb
it’s essential to stay very organized in that respect
All of the above probably sounds my mixes are drowning
in reverb. Don't be fooled - some of these are very subtle. Also, don't be afraid of long reverb times, but keep
them super low in level. Many of these you only hear
when you switch them off.
Here are some examples for types of reverbs that are
useful to have at hand, all at the same time:
•
most reverbs deliver sounds in categories like plate,
chamber, ambience, rooms, concert halls, church,
etc.
•
most classic reverbs (e.g. Lexicon 224, 480, EMT,
AMS RMX) are great at less defined, „cloudy“ reverbs, regardless at which reverb time
•
modern reverb plug-ins and hardware reverbs (Bricasti M7 is my favourite) are great at super-realistic
room simulation
•
plug-ins using IRs (Impulse Response) can do both IRs are basically just samples or „fingerprints“ of the
original reverbs or even real rooms
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•
What I personally have done over the years was simply try a lot of different reverbs and especially IRs,
and everytime I heard something that I liked, I saved
the channel strip settings, reverb, EQ and some
compression, to a channel strip preset in Logic. Doing this will allow you to keep developing a goto
library, and certain reverb chains you can not live
without after a while, and these make it to your DAW
mix template.
Delays also come in many different flavours:
•
the typical 1/2, 1/4 or 1/8-note delay „throw“, often
used on the last word or sylabel of a vocal-line.
•
short slap-back delay
•
complex delays with polyrhythmic patterns, or even
unpredicted „weird“ stuff happening in the feedback
loop
Reverbs and delays are to your mix, what a shadow is
in a photography, you don’t want them as sharp and
shiney as your main object. Most of my delay returns are
sending to reverbs as I don’t like them to be a totally dry
„sampled“ copy of my original signal.
Look at reverbs (and delays) similar to a shadow in a
photography, you don't need them 100% upfront, shiny
and dynamic, their purpose is to give you a sense of subliminal depth. Compressing them helps that a lot. I also
EQ them contrasting to the direct signal for that reason,
so if you've worked hard to make your lead vocal upfront, direct and "in your face", you don't want to send
your reverb from that direct signal.
BTW, this is one reason why older "vintage" reverbs/delays are still popular - they have a low-res, "grainy" feel
about them (example Lexicon 224 or AMS DMX1580).
Compressing reverbs also makes them more controllable - you can get away with less reverb in the mix, if
you compress the reverbs that you are using.
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CREATING A THREE-DIMENSIONAL SOUND
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There are obviously a lot of small details involved, but
for a start, get your head around the following concept:
1.
2.
important stuff goes to the center: Kick, Bass, Lead
Vocals
try panning everything else hard left or right
Sounds crazy I know. Try panning a guitar hard left, send
it to a small room and pan the room hard right. Now
bring the reverb a tiny bit more to the center until it feels
well-balanced.
The second guitar you might have: pan the dry one hard
right, send it to another small room (or delay or chorus)
and pan the effect hard left.
You get the idea… this works for guitars, keyboards, percussions, noise fx, etc. - I use small rooms/chambers/
spaces from the Bricasti M7 reverb for that. There are
Impulse Responses for the Bricasti out there - for free,
and officially authorized by the guys who designed that
reverb.
MODUL ATION FX
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re: rhythmic gating, the Tremolo (modulates only volume) in Logic Pro is a secret weapon.
I use this all the time on lifeless or over-compressed
tracks - brings some subtle movement into them. From
pads to static synth basses to heavy guitar chords to
backing vocals. You can even make reverbs groove subtly with the beat.
SUBGROUPS, FX BUSSES AND ROUTING
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As the number of different reverbs (delays, etc.) you are
using grows, you need to keep the routing strictly organized. Using the same reverb on several sources quickly
becomes messy. You might need to EQ or compress
your main reverb in a certain way to improve its sound
on the vocals, but as you have used that same reverb on
other instruments you are suddenly affecting the balance and sound of your entire mix.
I know it sounds counter-intuitive, but you’re better off
separating the reverbs you are using for the different instrument groups, which includes the routing and grouping. On my console, I have a pair of channels dedicated
to vocal reverbs, another pair for drum reverbs, and they
are assigned to their respective VCA groups.
On the drums use pre-fader sends, use the drum fx exclusively for drums, and route their returns to the drum
bus as well, and/or put them on the same VCA group.
Same goes for parallel compression busses, the sends to
them are pre-fader, not following automation, but the
returns follow the automation of the source - I do that
via the Drum VCA group - you get the added bonus that
you can treat the FX returns for each group of instruments separately. We want total control in mixing.
STEREO BUS MAGIC?
A very popular question… „What’s on your mix bus? Are
you using PRODUCT X or PRODUCT Z like I do?“
„All in one“ mix bus processors promised to be the one
stop solution since the first TC Finalizer that came out
in the 90s. You all know the various modern plug-in
equivalents of that, and to a certain degree, they work
really well. A quick preset can make a song demo more
presentable. Things that are involved in these proces-
sors are multiband-compression, stereo processors for
more width, psychoacoustic loudness treatment, and of
course EQs, compressors, limiters, etc.
The disappointing truth is that you really need to make
your mix happen before the signals gets to the stereo
bus. If the mix is rotten at the core - and I refer you to all
the things we’ve looked into from the Chapters 3 to this
point, multi-band compression can create more density
and loudness, but never solve problems that were ignored in the first place.
With that, let’s go through my common mix bus signal
chain, and allow me to add that every single processor
used here is doing only extremely subtle things.
1. LOUDNESS METER
From the very start, embedded in my template, the first
plug-in on my mix bus is a loudness meter - it monitors
the level of the signal that comes in. We talked about
Gain Staging in Chapter 5, so you know what to look for.
Also, at this point let me remind you about best A/Bing
practise as discussed in Chapter 3: we can A/B between a treated and untreated version of our mix bus, at
matched levels.
This is essential when working with plug-ins on the mix
bus - you might not need ANYTHING here.
2. BL ANK
I’m not joking. It’s a blank plug-in slot with no plug-in
inserted. Leave it like that because if anything comes up,
you can insert whatever you might need later on.
3. MIDRANGE
The combination of plug-ins I use here is EXACTLY the
same that I’m using on vocal-chains:
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A. Pultec MEQ 5
Remember, this was where on the vocal chain, a Pultec
is boosting when „warmth“ is needed. Test if our mix bus
lacks anything between 200Hz and 700Hz by switching through the frequencies. Don’t boost anything just
because you can. Try the same with the upper band as
well - sometimes a subtle boost between 1.5k and 7k
adds some energy.
On all accounts, I’m talking about a +2 or +3dB boost
here at best - which is still a very subtle amount on the
Pultec.
B. Tube compressor for tone
I use a plug-in version of the classic Fairchild here, and
settings could be called „esoteric“. The gain reductionneedle is hardly ever moving, but when I A/B, it ALWAYS
sounds better with the Fairchild in the chain. The category here is mid-range, as it adds harmonics.
4. FINAL TONE AND DYNAMICS CONTROL
A. EQ
Again the Pultec EQP-1a (don’t confuse it with the MEQ
used in step 3!) does a subtle boost here, usually 2dB at
20Hz, and I attenuate treble by 2dB at 20kHz. Another
very „esoteric“ setting, the Pultecs on my mix bus are
mainly used to add an analogue vibe.
B. Final Dynamics
I happen to use Slate FG-X as a final bus compressor,
and also use the Transient, ITP and Dynamic Perception
Controls to fine tune. I am NOT using FG-X as a brickwall limiter, and my signal is leaving this plug-in with the
same amount of headroom left as it’s entering it. The
„constant gain monitoring“-button is always pressed for
that reason.
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C. Linear Phase EQ
This is my final control for the overall frequency curve
of the mix. I usually add a very broad and subtle boost
on the bass.
And one more time: keep going back between these
building blocks for fine tuning. Watch the gain staging!
AUTOMATION
Let’s go back to a concept that was introduced in chapter 7 - correcting and automating levels PRE and POST
plug-in chain. PRE is to create a consistent natural
sounding performance, which we already dealt with.
The automation we are now working on is the POST
plug-in chain automation that creates your dynamic for
the song.
Of course, even once you start activating the fader automation in your DAW, there will be situations where you
need to change the entire relative volume of the track.
A simple gain plug-in before the automation fader gives
you easy access to the relative level of the entire track
- which is a lot easier than always correcting the entire
automation. The way this is handled differs slightly between DAWs, and console automation might have other
ways to deal with relative levels. Make sure though to
find out how to independently deal with relative levels
and automation, both POST plug-in chain of course.
Another issue we have already dealt with in Chapter 7,
but I’ll repeat it here again - if one instrument has completely different levels or settings in different songparts,
instead of creating automation for that, just copy the
channel settings and have an extra channel for different
song-parts.
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And… another reference to an earlier chapter: automation is best performed or programmed while listening at
low levels on your small portable speaker.
Automation is a great place to emphasize the natural
dynamics of the song. Let’s face it - on a rock-song, the
drummer will hit the drums harder on the last chorus.
Keep in mind that this is not totally depending on levels,
as more intense drumming will reflect in brighter drum
tones.
But still, a subtle push in volume on the final chorus can
create extra excitement.
This is where the final balance and automation of the
mix can be compared to car racing, and you will face
the same dilemma as the race car driver: you want to
drive as fast and risky as possible, without destroying the
car.
Luckily, with a few safety features in place, automating a
mix is still a very enjoyable ride (and much less dangerous compared to car racing).
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These are the safety features:
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• if you have „nulled“ your faders, like described in
Chapter 7, and maintained solid gain staging throughout
adding treatment of the individual channels, you have
a „unity gain“ default position for your faders, which
means there is zero chance to destroying the solid balance you have already created up to this point - you can
always go back to start. When you work your automation around the 0dB position of your fader, you have a
much broader range to do subtle fader movements. 3dB
is still a bit of movement around the 0dB-point - try to
perform a 3dB increase when your level sits at -30dB
default! Impossible. That goes for both physical faders
and visual automation data in your DAW
• oh yeah, you have of course saved your project under a
new version number on a regular base - so that you can
always go back in case your mix gets worse
• the stereo bus treatment we’ve setup in this chapter
will make sure that your dynamics stay within a certain
range. If your bus-compressor has the right setting, it
will subtly keep your dynamics within the frame of good
gain staging - in other words, if you push signals into
the stereo bus with more level, the increased level at the
channels will partly be compensated for by the ratio set
on the bus compressor.
• and one more last time: your portable speakers at low
levels will assure that you’re staying in the right frame
with your automation
With that - go and create excitement in your automation! Go for bold moves!
Definitely start learning to automate with faders. Drawing automation with a mouse will never replace your
hand on a fader while closing your eyes and listening to
the song at low volume.
As a final note on automation, know that you will also
have to go back and forth between setting levels and
EQing. Whenever there is a situation where you feel the
track sits great in the mix, but the bottom is to thick or
thin, you know where to find the handle for that.
The levels of the track are right when you can even hear
very subtle level-changes. As long as you can still move
the levels of an instrument up and down a few dB, and it
makes no difference for the mix, you need to go back to
the start of this chapter and use tools other than level to
make the signal sit right in the mix.
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FAQ
177
Order of operations: first EQ and then compression?
and why?
there is no fix rule for that - before anything else in
the signal chain, you remove problem frequencies, any
boost you apply via EQ will also sit better in the signal
with a tiny bit of compression applied afterwards. Then
there are compressors for tone, adding harmonics - they
can be followed up by an EQ, to emphasise some of the
harmonics that weren't even there before the compressor was in the signal chain.
Do you have ideas how to sort out the midrange on
a busy mix?
vocals: you will have to boost some presence for them
to cut through on a busy mix; we’ll discuss in a later lesson how to build the plug-in chain for lead vocals
Waves C6 Sidechain works great for side-chaining the
mid-band to the vocals; for example when you have
some mid-heavy distorted guitars, whenever the lead
vocal comes in, the presence of the guitars gets pushed
back to make space for the vocals; because only a certain frequency band is affected, the side-chaining is not
noticeable in the mix - the weight of the guitars in the
mix stays the same / you can apply the same to push
back pianos, keyboard pads, string orchestra, etc...
if you need to make space in the lower midrange, you
can use that same side-chain eq to make space for that;
frequency selective side-chaining is only last resort
though; ideally every instrument/vocal "sticks out" at
a different frequency. For example when stacking harmony/backing vocals do NOT boost the same low mid
frequency on all of them, use for example 300Hz for
the lowest, 500Hz for the middle one and 700Hz for the
high backing vocal; that way you create a nice texture
across the mids where all tracks can co-exist.
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CHAPTER 10
STEMS, MASTERING
AND DELIVERY
CHAPTER 10
STEMS, MASTERING
AND DELIVERY
Y
our mix is done. You’re happy with it and ready to
play it to get feedback from the client. I hope you
didn’t prematurely send any half-done rough „outtakes“
up to this point – which would be as unprofessional as
a chef asking you to come to his kitchen while cooking
to check if his soup is „going into the right direction“.
Unless the client is personally listening to the mix in
the studio, the mix will be most commonly delivered
through e-mail, either by sending mp3s, or a downloadlink to an mp3. The first pitfall to avoid is sending the mix
to the wrong people, in the wrong order.
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WHO TO SEND THE MIX, AND WHOSE
FEEDBACK TO CONSIDER
THE CLIENT PAYS
The CLIENT is the person or company/organisation PAYING your mix – treat them with the HIGHEST PRIORITY.
RESPECT THE ARTIST
In case the client paying for your mix is NOT the artist,
always BE RESPECTFUL OF THE ARTIST in your communication, ask the client „Do you want me to send this mix
to the artist for feedback?“
STAY OUT OF CONFLICTS
If the client that pays your mix (e.g. record label, A&R
or producer) does not want to include the artist in the
conversation, respect that. The client might want to sort
all his feedback out with you before getting the artists
approval.
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ARTIST CALL
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However, be prepared for the artist to call or e-mail at
you any time. In case the artist contacts you…
1.
listen to and respect what the artist has to say
2.
politely ask the artist to inform the paying client
about his feedback
3.
wait a few minutes, to give the artist time to consult
with his sponsor
4.
after not more than 15 minutes, inform the client
about the artist feedback
5.
offer to consider the artist’s feedback to the paying
client – ask „Do you want me to do an alternative
version making the changes the artist is asking for?“
If the answer from the client is still a „No“, leave it at that.
When the shit hits the fan, you can earn your degree in
diplomacy. Take no ones side,
STAY OUT OF CONFLICTS.
Be like Switzerland, wait until they have sorted out everything. Just don’t wait to send the updated version to
the paying client.
KNOW THE HIERACHY
If the client is a company (like a record label), be aware
that there is a
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HIERACHY INSIDE EVERY COMPANY.
You could be communicating with a Junior A&R who
commissioned your mix, and when you send your mix
for feedback, suddenly one of her bosses chimes in (e.g.
“A&R director”, “VP of A&R” or the President or CEO, or
all of them at the same time…) via e-mail, or even calls
you directly. In that case, e-mail both the Junior A&R
AND her boss.
Put all of them in the first line of the e-mail (not in Cc
or Bcc) and in the order of internal hierachy. The higher person in the company ALWAYS overrules the rest.
It helps to have good relationships with the people involved, and to have a bit of an idea of everybody’s role
within the company.
MOM IS CALLING
There can be a ton of other people who suddenly come
out of nothing and give you input – from the artists
mom/dad/uncle to various people inside the record
company, even radio stations or radio promoters sometimes call and ask for a change in the mix!
If you follow the rules above, and do great work, you’ll
be pretty safe. Also keep in mind that when an artist
manager is asking you to do a mix, they are representing
the artist who is ultimately paying you. Often the manager might deal with the first round of feedback, but the
artist will very likely have the final say, so expect them to
join at any point.
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E-MAIL DO’S AND DONT’S
For delivering the first completed version of my mix, I
stick to this template most of the times.
10
NO EXCUSES
•
NO EXPLANATIONS OR EXCUSES, don’t send the mix
unless you’re fully convinced it’s the best you can do!
•
Act as if this is the final version – reflect this in
the name of the file you sent, the first version I am
sending to the client is called
„ARTIST – Songtitle (Mixed by Marc Mozart)“
•
After the first revision, addressing client changes,
the name changes to
„ARTIST – Songtitle (Mixed by Marc Mozart) UPDATE”
•
Next one is called UPDATE 2, then UPDATE 3
DON’T BE STUPID
AVOID the following names:
•
bounce_output_1-2.wav
(WTF? And don’t ever e-mail a WAV!)
•
songtitle_mix_version_43
(your internal version # is not the client’s business)
•
songtitle_final_master
(who says it’s final? never call anything a master
unless it IS the master)
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KEEP YOUR ADDRESS BOOK CLEAN
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Before you send the e-mail, add your client with his
full name in your computer’s address book. The name
you’re given the client in your address book will show in
the e-mail your client receives.
You don’t want the e-mail of your client to show up in
their device as “info@artist-domain.com (Amanda hot
huge rack)” or “ar@recordlabel.com (whacky idiot)”.
SERIOUSLY!
After you’ve sent the e-mail, it is OK to send a reminder
after 6-9 hours and it’s also OK to call or text the client the next day if you haven’t heard back from them.
After all, your e-mail could have landed in their spamfolder, while it’s totally possible the client was very busy
and didn’t find the time (yet) to listen to your mix or
write an e-mail for feedback.
It is professional to expect timely feedback, however, if
it takes longer than you like, there is nothing you can or
should do. Just make sure the client has received your
e-mail, and wait for the feedback.
FEEDBACK: WHAT TO EXPECT
Be ready for anything – client feedback comes in many
different shades!
I recommend to acquire a deep understanding of the
different communication styles we find in people. One
of the most widely accepted models around communication types is the „DISC“-model, which categorises four
main types of communication styles by two critieria:
•
•
Is the person outgoing or introverted?
Is the person task-oriented or people-oriented?
Here’s an introduction podcast you can listen to online
from the very respected management consulting firm
Manager Tools. I've been a huge fan of their work for
many years, and I promise it’s worth every second looking into it:
Manager Tools – Improve Your Feedback With DiSC
http://www.manager-tools.com/2006/02/improveyour-feedback
The team from Manager Tools had a huge impact on
my career, as their podcasts helped me to get my head
around dealing with corporate management structures,
which is a field that freelancing artists usually massively
struggle with.
Back to DISC - there is an actual test you can do to determine which style you are, and at the core of all this,
you have to develop the skill to adapt your communication (things you say, how you say them, your body language) to ALL the different types of communicators that
exist in the world, and specifically those we work with.
This is a skill that differentiates the most successful professionals in any field from the ones that struggle – at
least when the job includes dealing with people.
The first thing you do after receiving e-mail feedback is
to carefully read the e-mail a couple of times, and then
reply, thank them for their feedback and confirm you
understood it, or ask questions if you didn’t. Give them
a time-frame of when you’ll be able to get back to them
with an updated mix.
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EXAMPLES FOR CLIENT FEEDBACK
EXPERIENCED PROFESSIONALS
(A&R, PRODUCER)
…will get back to you very quickly and e-mail you a list of
the most important bullet points. They know what they
want and usually, after listening to the mix twice, know
what changes they want.
This type of feedback is very easy to deal with. You go
through the different bullet points and offer a fix or solution to it. Reply to the e-mail in quote-style and explain how you addressed each bullet point. You can stick
mostly to technical terms in your response.
Above: feedback example of a very experience established artist/producer.
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ARTISTS
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Artists, unless they are very experienced and through a
lot of albums, will tend to take a lot more time to consider their feedback. They also tend to listen to the mix
many many times, and often respond in great detail with
how they feel about the mix.
I often get long e-mails with great backround information on how they felt and what they thought when they
wrote the song, and how these feelings relate to the
current version of the mix.
You need to invest time to careful read their feedback
and try to break up their prose style of language into executable bullet points of feedback.
Make sure to fully address their thoughts and how you
translate them into any technical terms of mixing.
You can include technical terms, but don’t forget to talk
about how and why what you’ve done FEELS that way,
“I’ve added a 3D-type of echo with reverb to the Lead
Vox in the chorus, to give it that “spacy astronaut feel”
you’ve mentioned in your feedback.”
EGOMANIC T YPES OF CLIENTS
Egomanic types of characters, guess what, they exist in
the music industry.
“Can you please tilt the lead vocal reverb trapezoidshaped in the depth of the room!”
I remember a mix (and production) I worked on, that
would end up becoming a Top 5 record for a major pop
act.
The mix was long approved by pretty much everybody at
the label, and there were a lot of people involved from
“VP of Marketing” to “Label President”.
But one A&R guy who was (kind of) assigned to the
record, wanted to keep arguing with me – I think he
didnt feel his influence on the mix wasn’t big as his ego
yearned for.
Anyway, beyond approval by all of his bosses, he kept
asking for changes that would not really make a difference (example “move the lead vocal 1 degree left in the
stereo field”) and his feedback culminated in the following request:
“Can you please tilt the lead vocal reverb trapezoidshaped in the depth of the room!”
When he said that on the phone, I had no idea what
he meant, but at the same time I knew that I could not
openly tell him my thoughts about what type of person I thought he was, and where exactly I’d like him to
stick his feedback.
I said: “Oh yeah, I know EXACTLY what you mean. That’s
great feedback. Let me work on that and get back to
you.”
After the phone call, I literally ROFLed for a few minutes.
Obviously I had no idea what he referred to with “trapezoid-shaped reverb” but I had to find a solution for the
dilemma.
The lead (rap) vocal that this song had was sounding
direct, had impact and whatever “reverb” he was after could only mess it up. I played with the “small room”reverb that was already on it, and (perhaps) added a tiny
bit of a pre-delay to it when it suddenly struck me!
Emagic’s Logic was at V4.7 at the time and it had a reverb plugin called “Platinum Verb” which had a GUI of
the plug-in that would show different shapes of rooms.
OH MY GOD! THIS MUST BE IT! THIS IS WHAT THE GUY
HAS SEEN SOMEWHERE!!!
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I didn’t actually change anything and certainly did NOT
use “Platinum Verb” in the mix, but I sent him an update,
called him and said:
“I’ve played around with adding a tiny bit of room sound
on the Lead Vocal, and I found that the Platinum Verb
plug-in sonically does exactly what you’re were after.
Have a listen and let me know!”
5 minutes later I had a reply in my mail saying:
“Spot on – you’ve nailed it! I love Platinum Verb. Mix is
approved!”
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DEADLINES:
HOW TO SURVIVE THEM
When you’re mixing for a client, a delivery deadline is
always part of the process, whether you perceive the
deadline as tight or not.
ASK FOR A DEADLINE
There are cases where a steady client just sends you a
Dropbox-Link with the files and no further comment.
Make sure to confirm the receipt of the files and try to
get an exact deadline from them.
I’m not sure what to think of a client who doesn’t give
you a deadline, or at least a rough time-frame. It’s not
professional. That said, if you have received an advance
payment, there is nothing to complain - the money does
the talking and says: „Get started.“
WHEN YOU HAVE A LOT OF TI ME
There are projects where the deadline is not pressing.
Whether the song is part of an album that is nowhere
near finished, or other factors are involved („we are still
looking for a feature rapper“, „Slash might do a guitar
solo on this“ etc.), there is one thing you can do to take
advantage of the chilled situation.
Do your mix preparation (-> Chapter 3) as soon as you
can! That way, you can leave plenty of time between mix
preparation and doing the actual mix, which, for all the
reasons we discussed in Chapters 3 and 4, will improve
and speed up your mix process.
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IMPOSSIBLE DEADLINES
Every now and then we get offered a job that comes
with what we perceive to be an „impossible“ deadline.
I recommend to look at the following aspects:
A: amount of work involved
B: amount of money paid
C: prestige attached to the credit
D: quality of the song/production
WHEN YOU HAVE NO TIME
OK, here you go. Let’s say, after juggling with A, B, C and
D, you decided to accept the job. Here’s how to survive:
1. Never skip preparation.
Here’s where the top mix engineers in the world are
winning the game early. They have world class assistants
who can start preparing the mix session while the master mixer has a huge plate of spaghetti.
By the time somebody like Chris-Lord Alge arrives at the
studio, the session is ready for him to mix, tracks assigned to faders, computers booted, coffee on the table.
OK, you’re not that guy… don’t skip your mix preparation
though. Yes, go through it a bit quicker than usual, skip
assigning nice little icons to your tracks and other OCDrelated stuff, but do it.
Consider to do more grouping of tracks in the preparation time than usual. Bounce all those backing vocals
to a stereo file, they are not a priority! Same for the 15
shakers. Pan the individual ones hard left or right (you
can always make them more narrow if needed) but don’t
get lost in them!
The less time you have to do the mix, the more premixing you can do during mix preparation.
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10
Same thing during the mix - group more stuff than usual.
You can deal with 8 faders when you got 2 hours to do
the mix, but you’ll go crazy looking at 80!
2. Don’t work through the night
Don’t work through the night, don’t change your sleep
pattern. It will mess up the next few days. If you really
need more time, sleep less, but sleep. Even one or two
hours of sleep aka "power nap" can make a huge difference.
3. Take regular breaks
Taking a quick 10 min walk every 2-3 hours, picking up a
coffee to go, and especially having your lunch and dinner at your usual times is essential when in a tight deadline.
Don’t get sloppy though, a tight deadline is like preparing for the world cup.
4. Keep in touch with the client
The client likes and needs reassurance that you’re on
schedule. A quick message or call helps them to keep
calm.
Even under a tight deadline, don’t ever prematurely send
unfinished mixes.
Keeping in touch during the process will also assure that
when you’re finally ready for client feedback, they can
provide that quickly. Nothing worse than working hard
to meet a deadline, and then waiting around forever to
get client feedback.
193
PRINTING STEMS
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The so-called „STEMS“ (short form or for "STEreo MixeS") that I’m referring to in this chapter are a collection
of audiofiles that, summed together, result in your uncompressed and unmastered Full Mix. Once you have
printed stems of your mix, by using these your mix is
fully recallable for anybody who has a basic DAW.
As a standard, I personally print the following Stems, all
with FX except were noted:
•
•
•
•
•
•
•
•
•
•
•
Kick
Snare
Rest of Drums/Percussion
Bass
Guitars
Keyboard-Pads
Keyboard-Leads
Orchestral Instruments
Backing Vocals incl. LV Doubles
LV (Dry)
LV (FX only)
I print them as WAV-files, in the work sample rate of the
mix project.
•
each of the listed Instruments are recorded as a
stereo-file while all other instruments are muted
•
bypass the compressor on the master bus and
everything that follows it in the signal chain
•
start the recording or bouncing of
file exactly 1 bar before the music
•
leave at least one bar of silence after the end of the
music, make sure to not cut any sound off, including
reverb tails
each
starts
•
This is kind of the minimum setup, as there are certain typical situations for which we will need to access the stems.
EXAMPLES:
•
weeks, months or years after approval of your final
mix, changes are requested.
•
this could go as far as new lyrics, a completely different singer, the song being recorded in a different
language, foul language that needs to be edited out
(clean versions), etc.
•
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•
stems are also essential for creating special versions/edits for radio, TV, medleys, megamixes as
well as syncs for adverts and movies; if the song
you’ve mixed becomes a huge commercial success,
you might receive numerous editing requests on a
regular base.
•
I also print stems when a client takes very long to
get back to me with feedback and I need to free up
the analogue console for another project. This of
course only makes sense when your mix is already
close to a finished state. In cases like that, I might be
doing more stems than stated above, to retain more
flexibility for changes.
•
Don’t forget you can always 100% rebuild parts of
your mix using the stems, while completely remixing other parts. Again, a typical example would be
to record a version of a song in a different language
while keeping the mix balance of the backing track
fully intact.
•
Don’t EVER skip printing stems of your mix. To have
them is a regular life saver for me, not to mention that
most industry contracts for mixers or producers require
you to deliver stems along with mixes and masters.
PRINTING MIXES
I’ve made it a habit to print the final mixes AFTER I’ve
done the stems. Printing stems should be done in realtime to become an additional internal quality control you are listening to the major instrument groups of your
mix in solo, and chances are that you are hearing a few
minor flaws that you want to fix. Some of these might be
things that no one ever notifies when all instruments are
playing, but it’s good to have another look at it.
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Many times I print the final mixes from the stems, even
when the mix was done on the analogue console. There
is a tiny difference in tone and dynamics - the mix
summed from the stems usually sounds a bit harder and
more direct, and is of course fully recallable from your
DAW software.
There are even situations when the producer of the song
wants to have access to the stems of your mix to do the
„final edit of the final version“ of the mix - sounds complicated, but it’s really not.
Often some time has passed since the producer sent
you his session for mixing, and with a fresh mind the
producer changed his mind about some aspects of the
song.
If that happens, just go along with it - it’s part of being a
great team-worker.
Anyway, at one point final mixes will be printed, and here
are some guidelines for that:
•
•
•
use your work samplerate (if that was a high-res
rate like 96kHz, print your mixes at that)
WAV files
don’t include fades at start and end (those are part
of mastering)
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MASTERING
The most important aspect of mastering is very similar
to the separation of production and mix - it’s simply two
different people looking at it from different perspectives.
You are always sending your client a „mastered“ mix for
approval - mastered in terms of normalizing the level
with a brickwall-limiter at the end of your stereo-bus.
If there’s a mastering engineer involved, he will work
with your mix that does NOT have your brickwall-limiter
applied and ideally a few dBs of headroom.
Anything you have had on your stereo-bus chain during
mix is totally fine to stay there (except your final brickwall limiter), but I don’t recommend you take that „extra
step“ of mastering your own mix.
That is likely to go wrong as you have heard your mix too
many times, and the point of mastering is really to have
another fresh engineer look at your work - a mastering
engineer specialised on the task of optimizing finished
stereomixes.
The attempt of most mix engineers to master their own
albums (or even producers, who mix their own album)
mostly goes terribly wrong. If there’s no budget for a
mastering engineer, take your mixes as they are, and use
your final brickwall limiter that you had on the mix, to
correct volumes. Don’t add more than 1 dB of Limiting,
compared what worked for you on the mix. You can of
course, always reduce volume of some of your mixes, by
either reducing the amount of final brickwall limiting, or
just lowering volume.
Selecting a mastering engineer to work with isn’t easy
- I often felt they severely destroyed my mixes. It’s not
always your decision though - the client might have
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a goto person, and there is not much you can do, especially if it’s a known mastering engineer who has a
strong relationship with your client.
It’s a good idea to build relationships with a handful of
mastering engineers.
With the stems in your pocket, you can always take your
mix session to a mastering studio, and get his input on it.
Together you can correct a few things and record your
mix into his system, and he can continue by mastering
your mix while you have a coffee in the lounge. A great
way to do final quality control of your mix, and you are
building a relationship with a mastering engineer at the
same time.
DELIVERY
Deliver to the client ONLY what they really need. Anything that can go wrong, will go wrong.
Before you determine what file-types you are giving to
the client, know exactly what the client is planning to do
with your mixes/masters.
These are things that happen on a regular base:
•
clients converting mp3s to WAVs to to upload them
for digital distribution
•
clients sending mastered WAVs to a mastering
engineer
•
clients sending random rough-mixes to the label
If you know that the digital distributor the client is using is only accepting 16bit WAV-files, ONLY deliver this
file-type. Many independent aggregators including The
Orchard and Believe Digital only accept 16bit WAV-files
for upload, not 24bit WAVs.
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The Orchard also accepts Apple Lossless AACs, and you
could create that from a 24bit WAV-file.
If you upload to iTunes directly using iTunes Producer,
you can of course use 24bit WAV-files, even with high
sample-rates.
FILES
Here’s a complete list of files that I recommend printing
once your mix is approved:
Unmastered versions, all 24bit WAV:
•
for delivery to a mastering engineer
•
bypass brickwall limiter in the 2-bus chain, keep at
the very least 1dB headroom
•
FULL MIX
•
INSTRUMENTAL
•
ACAPELLA
•
optionally: VOCAL UP MIX, VOCAL DOWN MIX
Mastered versions, both 16bit dithered + 24bit:
•
for delivery to a record label or upload to a digital
distributor
•
for red book-compatible audio CDs, 16bit dithered
files are required
•
FULL MIX
•
INSTRUMENTAL
•
ACAPELLA
•
optionally: VOCAL UP MIX, VOCAL DOWN MIX
Stems, printed „pre 2bus“ chain:
•
FULL MIX
•
INSTRUMENTAL
•
ACAPELLA
•
KICK
•
SNARE
•
REST OF DRUMS/PERCUSSION
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•
•
•
•
•
•
•
BASS
GUITARS
KEYBOARD PADS
KEYBOARD LEADS
BACKING VOCALS + LV DOUBLES
LEAD VOCALS (DRY)
LEAD VOCALS (FX ONLY).
Again, I don’t recommend sending all of these to the client. Securly archive these files on several mediums, online and offline, and let the client know you have some
optional files archived if ever required.
ARCHIVING AND BACKUPS
As I just mentioned, consider archiving all the printed
files as a part of your job. While technically, it might not
be your job, you will be an award winning life-saver everytime you can help your client out with the files.
You’d be surprised, how even record labels have sometimes no idea how to get hold of multitrack-files or even
stems of some of their biggest hits.
CLIENT: „THERE’S ONE MORE THING…“
Some of your clients will come back months, sometimes even years after you’ve mixed their song, and will
have all kinds of requests to add or change things to the
mix. Sounds crazy, but it’s just business as usual. A song
needs an edit for a commercial, or film sync, a foreign
language version will be recorded, or some territory’s
censorship requires to edit out words, etc. - it happens
all the time.
If you have followed all of the guidelines in this chapter,
you’ll be fine, and most requests take no more than 15
minutes do deliver.
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THE END
Thanks for taking the time to study this book. This is actually not the end, as I am inviting you to join the online
support group [YOURMIXSUCKS] on Facebook.
https://www.facebook.com/groups/1511612519104171/
On the group you can ask specific questions about the
book, and I will continue to update the book as your
questions are coming in.
The goal is to create a complete software compendium
on mixing that will stand the test of time.
You will receive lifelong updates. Things that are planned
are bonus chapters specific to certain DAWs, and versions in Spanish and German. All free updates for licensed users.
This is Version 1.0, and you can expect a minor update
within a month.
Marc Mozart, January 2015
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