Get your club-bound tracks sounding

bigger and better than ever!

TUTORIALS,
TIPS AND ADVICE
> Mixing in the digital domain
> Maximising low-end impact
> Sidechaining for punch
> How to give the DJ
what they want
MASTERS
OF THE MIX
Pro advice from Camo &
Krooked, Steve Duda,
Marc Mac, TC and more!
 SPECIAL 63 2013
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ENJOY THE ISSUE…

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SPECIAL ISSUE 63

07 THE BEAT
GOES ON
contents 50 LOW END
THEORY
This thing called dance music Once the drums are sorted,
has been with us for a long it’s all about the bass – and
time now, so let’s start our with the help of our
mix-improving odyssey with walkthroughs, there’s no
a journey back through time reason why yours can’t be
tough enough to punch
holes in any dancefloor

58 SIDECHAINING FOR IMPACT

Learn how to use editing, ducking and sidechaining to bring
clarity and separation to your dance music productions

66 MASTERING THE STEREO FIELD

Take full control of the mids and sides and your mixes cant fail
to sound better than ever. We show you the way

72 BACK TO BASICS
The music software market is awash with incredible plugins,
12 KINGS OF THE DANCEFLOOR but you can still make amazing-sounding mixes using nothing
Seven leading lights in the world of dance music production more than the effects built into your DAW
give us their views on the industry and spill the beans on how
they get such great results

80 ANALYSE THIS
Strap a visual analyser on your output and great things can
24 MIXING IN happen. Discover how to see the music with our guide
THE DIGITAL
DOMAIN
Here comes the science bit…
Get your head around the 87 UNIVERSAL TRANSLATION
technical side of mixing digital From laptop speakers to stadium PAs, today’s dance music
audio with our (relatively!) has to sound great on the full array of playback systems.
easy-to-follow guide We’ve got it covered…

31 MIXING AND MASTERING FOR 94 FIX YOUR MIX

CLUB PLAYBACK
Although pretty much anything goes when it comes to
electronic music, there are certain rules that need to be
adhered to if you want your tracks to stand up to the
competition in a club environment
Rarely does everything go to plan, so we close this Special
with some thoughts on what to do when your track just isn’t
coming together as it should

98 CM DOWNLOADS
You can get your hands on all the
38 WHAT DJS WANT tutorial files, samples, software and
Today’s DJ is a busy dancefloor executive with no time to waste videos that accompany this
on sub-par choons. Ensure that yours make it through their filter Special on the disc and at
with our tips and advice vault.computermusic.co.uk
– select ‘Computer Music issue 63’

43 TUNING AND EQING DRUMS

If your drums ain’t happening, you might as well call it a day,
but maximise those frequencies and levels and you’re
already half-way to a great mix

COMPUTER MUSIC SPECIAL / 5

MAG/DVD

Future Music is the mag for the latest gear and how today’s cutting-edge music makers use it.
We’ve been making the future since 1992. Make sure that you’re part of it.
Contemporary club music is certainly
polished, but the quest to create the
perfect dance record is nothing new…
COMPUTER MUSIC SPECIAL / 7
Over the past couple of years, the
unappealing term ‘electronic dance music’
has been coined to help convince new
markets – primarily mainstream America
– that genres like house and techno are fresh
and wholesome, rather than the products of
hedonistic nightclubs and sweaty raves in the
80s. This kind of cultural appropriation is
nothing new, of course: just as house and
techno could only be effectively sold to
middle America by Europeans rather than the
genres’ actual progenitors, blues required
rebranding as rock ’n’ roll before it could
become mainstream.
The idea that dance music is a recent
invention or is somehow unconnected to pop
music is of course nonsense: the roots of
practically all contemporary popular music can
be traced back to blues, the dance music of
America’s ‘juke joints’, where workers would
carouse after a hard day’s graft. Originally the
music would be played by live musicians, but for
practical reasons many social venues would
switch to using a coin-operated phonograph – or
jukebox – to provide music for dancing.
Jukeboxes were so popular that by 1945
three quarters of the records produced in
America went into them. In 1950, the first 45rpm
vinyl record was introduced, leading to the
format becoming the dominant media for music
in the second half of the 20th century. Vinyl
could handle longer recordings and was less
noisy than the shellac discs that preceded it, but
sound quality was down to the recording and
production of the music as well as the format on
which it was reproduced. You might imagine the
so-called ‘loudness war’ to be a phenomenon
associated with digital formats such as CD, but
in reality it’s been going on far longer. Motown’s
Berry Gordy, in particular, was obsessed with
making loud, clean-sounding records; check out
www.collinsaudio.com/Prosound_Workshop/
Motown2.pdf for an insight into how his staff
made his fidelity-based fantasies a reality.
Even The Beatles weren’t immune to the
seductive lure of big, bad bass. During the
Width and depth
It might not do our ears any good, but
millions of us flock to dingy nightclubs to
hear beats blasted out at full volume every
weekend, and mastering engineers are
constantly badgered to make mixes as loud
as possible. So why do people love loud
music so much?
The culprit is a tiny organ in the inner ear
known as the sacculus, which is linked to a
region of the brain associated with pleasure
response. According to research by the
University of Manchester’s Dr Neil Todd, the
sacculus reacts to low-frequency inner
vibrations over 90dB – exactly the kind of big
bass delivered by club sound systems.
Academic and lecturer Dr Barry Blesser
says that “raising the loudness of music, like
a double shot of whisky, elevates the
intensity of the experience”, and that “music
is a stimulant, like caffeine, sugar, alcohol,
anger, vigorous exercise, sexual activity and
8 / COMPUTER MUSIC SPECIAL
recording of Paperback Writer in 1966, John
Lennon felt that the Fab Four’s tracks should
have the same loud bass tones as a particular
Wilson Pickett record. To fulfil this desire,
producer George Martin’s engineers boosted
the sound of Paul McCartney’s bass using a
loudspeaker as a microphone, and the resulting
7-inch single was cut louder than any other
Beatles record up to that point, thanks to a new
piece of EMI mastering equipment known as the
Automatic Transient Overload Control –
essentially the 60s equivalent of something like
iZotope Ozone 5 or Slate Digital FG-X.
Dub plate style
Elsewhere, a new sound was developing that
would have an enormous influence on dance
music. Sound system culture blossomed in
Jamaica in the 50s, with crews competing to
build the biggest and loudest rigs with the best
bass response. Sound system DJs had
progressed from playing American R&B to
cutting their own exclusive acetate records
(called dub plates), creating a Jamaican take on
the sound: ska. This developed into reggae in
the 60s, with producers like King Tubby and Lee
‘Scratch’ Perry taking the genre into more
abstract territory with ‘dub’ versions of tracks
that de-emphasised the vocals, brought the
drums and bass to the forefront, and ran wild
with spring reverb and tape delay effects. This is
where we first see contemporary dance music’s
aesthetic sensibility and approach to production
“Motown’s Berry
Gordy, in particular,
was obsessed with
making loud, clean-
sounding records”
Valve Sound System provide free earplugs to party-
goers – and strongly recommend that they use them
many others. Loud music is a simply a
stronger stimulant than soft music.”
Another aspect of the appeal of loud
music is that it places us in another ‘aural
space’ by masking other sounds and
removing us from our immediate
appear, with beats, bass and atmospherics
taking priority.
A similar evolutionary step was taken in the
States in the 70s. By this point, R&B, soul and
jazz had spawned funk, an uptempo, rhythmic
style that typically revolved around a single
chord and featured tight beats and bass and a
groove constructed from the interplay between
guitar, brass, keys and vocals. The drums were
such an important element that many funk
tracks would feature extended drum solos, or
‘drum breaks’. The style would be an important
influence on Philadelphia soul music, which
would add jazzy vibraphone and sweeping
strings to the mix, ultimately providing the
foundation for disco.
Combining the euphoric elements of the
Philly sound with Latin touches, uptempo four-
to-the-floor rhythms, prominent basslines and
synthesised sounds, disco music was all about
providing a soundtrack to an extended (and
possibly drug-fuelled) dance session. Producer
Tom Moulton had a major influence on the style,
using tape machines to add extended
percussion sections to tracks in an effort to
make them as long as possible and maximise
the enjoyment they delivered.
DJs would also begin to get more hands-on
with the music in an effort to realise the
uninterrupted dance experience. During this
period, the art of the DJ as we know it started to
evolve, with jockeys like Francis Grasso using
beatmatching and slip-cueing techniques to
deliver a seamless sequence of programmed
music, and DJ Kool Herc using two copies of the
same record to create looping, potentially
endless drum breaks.
Electro shock
In the 80s, the advent of relatively affordable
samplers, synths and MIDI sequencers meant
not only that aspiring musicians could forge
their own productions at home, but also that
they had a wealth of creative possibilities open
to them that could scarcely have been imagined
by the previous generation. This led to the
environment, altering our frame of mind. You
can read Dr Blesser’s The Seductive (Yet
Destructive) Appeal of Loud Music at http://
bit.ly/y9Ro4c.
As the title of Dr Blesser’s paper suggests,
loud music can of course have a negative
effect on our hearing, so it makes sense to
take care of your ears when producing and
clubbing. Lemon D, co-owner of the
legendary Valve Sound System, recommends
that punters use the free earplugs provided
at Valve shows to protect their hearing: “If
you go into any bar and there’s loud music,
you’ll need earplugs, but what makes our
system feel louder is the bass and overall
EQing. Bass doesn’t give you tinnitus – high
end does. The reason people wear earplugs is
that some insist on partying all night in front
of the speakers – this is absolutely nuts! It’s
best to move around, then you won’t be
subjecting yourself to loud music [all night].”

Burial’s album Untrue combines skippy dubstep beats with entrancing,
headphone-friendly sound design
Easy listening
Back in the pre-internet days, acquiring dance music wasn’t something
that could be done from the comfort of your own home. One had to
endure the now-unimaginable trauma of leaving the house, finding a
specialist dance music shop and dealing with its cheerless staff. Stores
would have limited stock, so tracking down particular releases could
involve a lot of legwork, and after all that you’d come home with a slab of
vinyl that would have to be recorded onto cassette in order to listen to it
anywhere other than the room that contained your record player!
Nowadays the equivalent process is far more civilised. Thanks to the
internet, we all have access to a truly incredible amount of music, and
people are much more likely to consume dance music at home or via
their smartphone. Though some dance music is still created in a very
raw, unpolished way, designed simply to sound banging on a club
system, most producers will want to ensure that their music is enjoyable
whatever system it’s reproduced on.
In Universal Translation (p87), we look at some ways in which we can
craft mixdowns that work on as wide a variety of systems as possible. In
addition to the mechanics of making a mix sound right on smaller
speakers or earphones, you might find it beneficial to consider the home
listener right from the composition stage. Adding small touches that
give your track more depth and listenability doesn’t mean you have to
detract from how it sounds on a big system.
For example, a good way to make a track more radio-friendly and
enticing for the home listener is to have a brief atmospheric preamble
that precedes the DJ-friendly beats-only intro. Alternatively, simply
including some interesting sounds and textures rather than bare beats
at the start of a track can make it more listenable. To ensure listener
interest throughout, consider varying the pattern or drum fills, or
introducing subtle stereo noises that are audible on headphones but
don’t detract too much from the main elements.

development of new musical styles such as
house and techno, and helped hip-hop come
into its own.
Many early hip-hop records, such as Sugarhill
Gang’s Rapper’s Delight and The Fatback Band’s
King Tim III (Personality Jock), relied on live
funk backing tracks, while others, like West
Street Mob’s Break Dance – Electric Boogie and
The Adventures of Grandmaster Flash on the
Wheels of Steel by Grandmaster Flash were
based around turntable performances. Synths
and drum machines made electro tracks like
Afrikaa Bambaataa’s Planet Rock and Hashim’s
Al-Naafiysh (The Soul) possible. Electro is
arguably the earliest recognisable form of
‘electronic dance music’ – beats and bass-driven
music made almost exclusively of synthesised
sounds or samples.
Hip-hop found a new instrument of choice in
the affordable digital samplers released by Akai
and E-MU in the 80s. These enabled producers
to recycle the drum breaks employed by Kool
Herc and other hip-hop DJs into new musical
compositions, giving a new lease of life to the
work of funk artists – particularly James Brown,
whose productions (including Lyn Collins’ Think
(About It), Bobby Byrd’s Hot Pants – I’m Coming,
Coming, I’m Coming and his own Funky
Drummer) were sampled time and time again.
The vintage breaks were usually supplemented
with lows and highs from drum machines like
the Roland TR-808, generating huge, bassy
kicks and pristine hi-hats. The result was beats
unlike any the world had heard before, with
“The late 90s saw
computers reach
speeds fast enough to
handle multiple high-
quality audio streams”
tracks like Run-D.M.C.’s Run’s House and
Mantronix’s King of the Beats demonstrating
how powerful the combination of sampling and
synthesis could be.
Rave generation
While hip-hop achieved considerable
mainstream success during the 80s thanks to its
comparative accessibility, house and techno
would take a little longer to become big
business. Early house music productions such
as Jesse Saunders’ On and On were rather lo-fi,
with the legendary Trax Records even pressing
its releases on recycled vinyl to save money. The
DIY approach soon caught on in the UK, and the
explosion in popularity of dance music created a
cottage industry of producers pumping out
tracks from their bedrooms and project studios.
The production values of these releases varied
wildly, as many of the producers had limited kit
and technical know-how.
In the early 90s, simple sample-sequencing
programs for home computers called trackers
began to find popularity, and it’s here that we
start to see the first dance records made
partially or entirely ‘in the box’, using a
computer as both sequencer and sound
generator. Seminal rave acts such as DJ Red
Alert and Mike Slammer, Urban Shakedown and
Omni Trio began their careers using Amiga
home computers running tracker software. The
humble Amiga was only capable of rudimentary
8-bit sample playback and trackers didn’t have
any real-time effects processing, but over a few
short years computer-based production
improved dramatically.
The late 90s saw computers reach speeds
fast enough to handle multiple high-quality
audio streams, and programs that had once
been purely for MIDI sequencing, such as
Steinberg’s Cubase, develop to the stage where
they were capable of recording and playing
back audio and hosting virtual instruments and
effects plugins. Now it really was possible to
make an entire track using nothing but a Mac or
PC, and for it to sound on a par with something
produced in a high-end studio.
Shortly after the introduction of plugin
effects, mastering plugins began to appear. This
development meant that artists could take
control of the entire production process, all the
way through to the finished product, and
enabled them to develop an understanding of
the sonic characteristics their mixes needed to
have in order to facilitate a good, loud master. In

COMPUTER MUSIC SPECIAL / 9

Soft options
If you’re new to the world of computer
music, you may well find the sheer
choice of available software
overwhelming. Thankfully, these days
practically all DAW packages are of an
extremely high standard, and it’s possible
to make fantastic-sounding music using
any of them. Some make certain
processes quicker or easier than others,
but as cliché as it sounds, it’s not what
you use, it’s how you use it.
The tutorials in this Special use
three of the most popular packages
(Apple’s Logic Pro X, Steinberg Cubase 7
and Ableton Live 9), but in the vast
majority of cases the techniques that
we’ll be showing you can be applied to
any DAW. If you get stuck following a
tutorial that’s not specifically written for
your DAW, you’ll likely find equivalent
functionality by searching your DAW’s
accompanying documentation.
If you haven’t yet picked a package
with which to begin your music
production odyssey, we recommend
checking out a few demo versions before
you take the plunge. Ableton Live (www.
ableton.com, from €79) is ideal for
making dance music thanks to its flexible
effect routing, easy automation and
fantastic time-stretching capabilities, and
it can even be used to DJ or perform live.
Propellerhead Reason (www.
propellerheads.se, from €120) has an
interface that’s particularly intuitive if
you’re used to working with music-
making hardware, and its unique
selection of included instruments and
effects has won it a legion of fans. Cockos’
Reaper (www.cockos.com, from $60) is
extremely cheap and the demo version
doesn’t have any kind of expiration date
or session time limits, which makes it a
tempting choice for those who want to
take their time and work on a few
projects before they stump up their cash.
Most trial versions of Cubase (www.
steinberg.net, from £82) require a True to its name, Ableton’s Live DAW can
be used for DJing and live performance
hardware eLicenser dongle to run, but as well as music production
the entry-level Cubase Elements 7 demo
doesn’t, so give that a go if you don’t
already have an eLicenser. Currently no
trial version of the Mac-only Apple Logic
Pro X is available, but GarageBand is free
and serves as a great introduction to its
much more powerful big brother.

the 2000s, CD players started to replace vinyl
turntables in clubs, which meant that it was
possible to test these mastered tracks in a club
environment by simply burning them to disc.
Producers began pushing the envelope like
never before, with artists such as Pendulum,
Noisia, Deadmau5 and Skrillex raising the
production bar to previously unimagined levels.
Testing times
The current scenario in which aspiring dance
music producers find themselves is quite
challenging, and not just because standards
have become so high. The availability of cheap,
powerful computers has democratised the once
specialist field of dance music production, which
means there’s now more of the stuff about than
ever before: thousands of new tracks are
released on digital stores like Beatport every
week, and standing out from the dross is tough.
Having a good understanding of the more
technical aspects of music production is vital to
the contemporary dance musician. Throwing
samples together in an ad-hoc manner might
have been an acceptable way to produce a
white-label 12-inch in the 90s, but taking the
same approach these days won’t cut the
mustard. Most small record labels don’t have the
budget to pay for tracks to be mixed down by an
“Having a good
understanding of the
more technical
aspects of music
production is vital”
engineer, and professional mastering can only
enhance a mix so far, so If you want to make
music that catches people’s attention, you need
to build it correctly from the ground up and
know how to push it to its limits.
In this Special, we’ll explain both what you
have to do to achieve a big, contemporary
dance music mix, and why you need to do it. In
order to properly exploit the digital medium in
which we now produce our music, it’s necessary
to understand its capabilities and limitations. In
Mixing in the Digital Domain (p24) we’ll explain
how digital audio works from basic principles,
giving you insight into what’s going on inside
your DAW, while in Mixing and Mastering for
Club Playback (p31) we’ll discuss how we can get
the most out of your tracks using specialised
mastering plugins, analysis tools and general
good practice. Then we’ll launch into a series of
in-depth tutorials, starting with the processes of
selecting, editing, tuning, processing,
sequencing and layering drum sounds, and
creating synthesised bass sounds to sit perfectly
with them, before moving on to maximising the
stereo image, getting the most out of dynamics
and equalisation effects, using sidechain
compression-style volume ducking techniques,
creating mixes that translate well, and using
analyser plugins to ensure your mixdowns are
on point. Finally, we’ll see how to make
arrangements that work well for DJs and look at
strategies for fixing mixes that aren’t working
out. What’s more, we’ve got advice from a line-
up of dance music mavens comprising big-name
mixing and mastering engineers, scene veterans
and fast-rising stars.
As well as all the knowledge and skills you
need to take your productions to the next level,
you’ll find a wealth of amazing free samples in
the Samples folder of your disc or download,
and a selection of exclusive plugin instruments
and effects for Mac and PC in the CM Plugins
folder. So, without further ado, boot up your
DAW and get ready to make your best dance
music mixdowns yet!

10 / COMPUTER MUSIC SPECIAL

KINGS OF THE
DANCEFLOOR
Steve Duda
Legendary producer, mixer
and plugin developer

Beau Thomas
Veteran dance producer
and mastering engineer

Camo & Krooked

DnB legends and Hospital
Records in-patients

Electronic Youth
Rising stars of soulful
deep house

TC
Jump-up DnB DJ and big-
name remixer

Marc Mac
Genre-hopping founder
member of 4hero

Frederic Robinson
Innovative live performer
and electronica prodigy

12 / COMPUTER MUSIC SPECIAL

Steve Duda
Multitalented super-producer Steve Duda has
worked with a diverse range of acts, from
Nine Inch Nails to Deadmau5, and is the
brains behind the Xfer Records range of
plugins. He also recently mixed the new
Britney Spears single Work Bitch. You can
keep up with his eclectic sonic adventures at
www.steveduda.com.

: How does creating a dance track from the

ground up compare to mixing down a pop-
EDM track?
SD: “When I’m creating a track, I pretty much
mix it as I go. There may be some mix tweaking
every time a part is added or a new section
comes along, and maybe a couple of tweaks at
the very end. When I’m mixing someone else’s
music – pop or otherwise – my mindset is that

COMPUTER MUSIC SPECIAL / 13

“To the average listener, there are
two instruments: the vocals, and
everything else”
the producer considered this a finished piece of
music, in the sense that all the pieces are there.
Now it’s up to me to show those pieces in the
best light possible.”
: What do you have on your master output
during the mixing process?
SD: “Everyone has their own opinion – some
people have a rule of no compression or
limiting. Those people wouldn’t like me very
much! I always have two to three compressors
and a limiter on my master. First a gentle
compressor on the way in, just to very subtly
tame peaks, then later in the chain a fast-attack,
medium-release one set with a fairly low
threshold for constant gain reduction, which
also has some of the dry signal in the mix –
parallel compression. Then, at the end, a limiter,
which tends to pull a few dB or more of limiting,
depending on the genre. The more pumping
type of a tune it is, the more limiting I use.”
: How do you ensure the vocals come
through clearly?
SD: “I mix them loud, first of all. This took me a
while to learn, because traditionally I was always
more interested in listening to instruments. As
an instrumental musician, it is important to
remember that the vocals aren’t simply another
instrument to the average listener. To the
14 / COMPUTER MUSIC SPECIAL
average listener, there are two instruments: the
vocals, and everything else. And they don’t want
your stupid ‘everything else’ getting in the way
of enjoying the vocals.”
: How do we keep the groove in dance
music while also getting super-loud mixes?
SD: “It’s all about syncopation and having things
play off each other. For example, you only need
one prominent swung 16th-note in a bar to give
something a swung-16th feel. I think having
sparse music being the norm is quite healthy for
beginners in particular – it’s not about how
many notes you have, so learning to do more
with less is good. With technology, it’s so easy to
layer and layer, and this turns into babble. Years
ago, every electronic demo I would get was
eight stacked loops, all playing 16ths.
“‘Less is more’ is easy to comprehend but
hard to believe. I like to say it like this: imagine
you talk to two strangers at a bar. The first one
tells you their life story, about every pet they
had growing up and a bunch of meaningless
trivia. Then the other walks up to you, says three
words and walks away. Which conversation will
impact you more?
“The same principle in essence applies to the
way I approach mixing. I prefer to think about
what I can ‘get away’ with turning down, rather
than what I want to turn up – I ask myself, ‘What
frequencies are unimportant?’ The first thing I
do when given stems is set the BPM and then
carve out all the silence. Then I start removing
sonically. The whole thing is removing the junk,
so the important stuff is left. Bad noise goes,
good noise stays.”
: Do you have a special technique for
A/B-ing your mixes with reference tracks?
SD: “I never do that. Perhaps I should more! I
think it’s a terrible mistake to expect a song to
sound like another.”
: Do you have any tips for balancing kick
drums and bass?
SD: “The rule of thumb is that you can’t have
both be big, low-frequency sounds content-wise,
so the solution is to first make a decision as to
which of the two is going to ‘own’ the low end,
then make the other ‘sound’ big without actually
being big. For instance, if it’s a dubstep tune and
you decide the bass needs to be big, perhaps
put a big reverb on the kick and remove the lows
a bit. This way the kick ‘feels’ large without being
large. Similarly, if it’s an electro tune with a
buzzy bass and you decide the kick should be
bigger, emphasise the buzz of the bass a bit and
shelf or high-pass it a bit, and maybe add
something to increase the width like a subtle
chorus or short stereo delay.”
Beau Thomas
Industry veteran Beau Thomas has released
dozens of dance records as a member of
various groups including Intense, Babylon
Timewarp and The Whitehouse Crew, and has
a hand in thousands more in his role as
mastering engineer. Find out more about his
services (and even get a quote!) at
www.teneightsevenmastering.co.uk.

: How did you get into mastering?

BT: “When I was about 16 I started writing
hardcore with some friends that were a little bit
older, who worked in a studio as tape ops and
assistant engineers and stuff. I wrote DnB until
about the age of 23, and then realised I needed a
job as the income wasn’t regular enough. So I
got a job at Heathmans Mastering in 1999, just
copying CDs on eight grand a year. That was
quite an established mastering studio, and I just
slowly worked my way up through the ranks.
The best job at the company was vinyl/digital
mastering, so that’s what I aimed for. I’d sit in the
rooms with the cutting engineers, pestering
them, being annoying, wanting to know what
they were doing.”

: Did being an artist help to give you a

good perspective on the mastering process?
BT: “Absolutely! Sometimes as an artist you’d get
a test press of a record back and it just sounded
awful! You’d be like, ‘What happened there?’
Thankfully that didn’t happen too often. Not
everything translates to vinyl perfectly, so if
someone has a song where there’s a shaker or a
sibilant vocal or something that’s a bit too stereo
in the track, it won’t cut very well. Because I’ve
been a producer, I’m able to talk to the artist
about it on a level that they understand. I can tell
them to put a plugin on it and narrow it down.
Some mastering engineers would just say, ‘That
doesn’t work’, and leave it at that!”

: What should producers look out for in

their tracks to avoid causing problems for
mastering engineers?
BT: “Stereo, for one. I love stereo, but you have
to be careful with it! With vinyl, top-end stereo
distorts before mono, so if you have an open
hi-hat – which would be prone to distorting a bit
anyway – don’t make it too wide if you can help
it. As a cutting engineer, it’s generally better to
have something a little duller for vinyl than
something that’s too bright. Pushing the mix too
much is a common issue – people will just slam a
limiter on there, and they don’t have any
metering, so they’re not quite aware of how loud
they’re going. They’ll just squash the life out of a
song, as well as making it too bright. Be careful
of how harsh things are. I always use this kind of
example: if you turn the song up really loud on
your speakers and any part of it hurts your ear
or is a bit sharp, then that won’t cut.”
: A lot of producers now use clipping
plugins on their master channel. Is that
acceptable as long as you supply the
mastering engineer with an unclipped
version too?
BT: “I get asked this a lot by distributors and
record labels – ‘How do you want the files
provided?’ How I see it is if you’re writing
contemporary dance stuff and you want it to
stand up to the industry-standard level, it’s
going to be bloody loud. So, if you write your
whole song with no kind of buss limiting or
compression, when you send it to a mastering
engineer he’ll squash the pants out of it to get it
up to a commercial level, and it could really
change your mix. So people will say, ‘Wow, that
reverb has got really loud’, or, ‘That delay tail has
come right up’, or, ‘I don’t know why my snare is
like this’. So I think it’s a good thing to have on
the end of your chain, so you can understand
how the song is going to be shaped when it
comes to the mastering stage.
: How has mastering changed since you
started in the business?
BT: “The main difference is level – level has just
gone crazy, as everyone knows. A lot of people
blame mastering engineers for that, but we’re
paid to provide something. I’ve submitted
quieter masters in the past and been asked to
turn them up, because they’re not as loud as the
tracks they’re being referenced with. The need
for level has become an overriding force, and
whether or not they admit it, people are
prepared to sacrifice quality for loudness.
“Level wasn’t such an important thing ten
years ago. We’re getting masters now that are 6
or 7dB louder than a commercial release would
have been in 2000. It’s still hitting 0dBFS, so the
RMS is being lowered heavily, and dynamics are
being sacrificed.”
: What advice do you have for producers
who want to make their own master for
playing out?
BT: “First and foremost, really concentrate on
your mix. I say this to a lot of people who ask,
‘How do I get things louder?’ It’s all down to the
mix. Get your mix tight, spend time on it, so
there are no overlaps. Nowadays sidechaining is
the big thing – look at things like that, get your
mix really tight and don’t rush it.
“The package that I think is probably the best
on the market is iZotope’s Ozone 5. It’s a nice
package: it’s got a fairly decent EQ, a very good
maximiser, multiband, a bit of stereo width if you
want it, a bit of excitement. Just push your track
as much as you can without it folding. If you
don’t feel your track can go as loud as one by
NOISIA or someone of that calibre, don’t make it
as loud – back off a little bit! Look at your mix
and assess it; examine where it’s folding and
hear where things aren’t working. So yeah, use
Ozone – just don’t push it too much. That will get
the track loud enough to DJ with. You might
possibly need a little bit of gain on the mixer to
give it more level, but you should be alright!”

COMPUTER MUSIC SPECIAL / 15

Camo & Krooked
Viennese DnB stars Camo & Krooked (aka
Reinhard Rietsch and Markus Wagner) have
just released their latest album Zeitgeist on
Hospital Records. Check out audio and video
clips and get tour dates at http://
camoandkrooked.com.
: How do you ensure that your mixes
sound good in the club?
RR: “After nine years of experience in DJing and
producing you kind of get a feeling of what
sounds strong in the club, and you can always
double-check on weekends when testing out
new material. But for us, it’s way more important
right now that the music not only sounds heavy
in the club, but also sounds good in the ear. It’s
not always about how hard the bass hits – it’s
about the music and its elements.”
: Do you have a limiter or anything else on
your master buss when you’re writing?
RR: “We mostly start without any limiter in the
beginning, but we use one when the time comes
to check that the mix can be made nice and
loud. Once we have the limiter on the master, all
the issues get fixed in the mixdown, as you can
isolate the troublesome elements and fix them
very easily. It’s also easier to adjust transients
and remove resonances while layering drum
elements on top of each other. There is no
EQing, exciting or whatsoever on the master, as
we do everything in the mix, trying to get the
most detailed result. We always play our own
masters in the clubs and as of our last two
albums, we use our own masters as well, to
make sure they sound exactly like we want them
to be: loud and crispy, not pumping at all.”
Electronic Youth
Trevor King and Rustem Rustem are
Electronic Youth, a deep house act who hit it
big with their soulful Angie Stone cover I
Wish in 2012. Since then, they’re received DJ
support from the likes of Pete Tong and
Sander Kleinenberg, and have recorded their
debut album. They’re also the subject of a
Producer Masterclass feature to be published
in a forthcoming issue of . Find out more
16 / COMPUTER MUSIC SPECIAL
: Any tips for mixing on headphones?
RR: “We worked on headphones for Cross the
Line two years ago, and it still sounds quite
strong on speakers and in clubs. You kind of get
an ear for that. Still, for Zeitgeist we got
ourselves proper studio equipment: KH O300
monitors, Prism Sound Lyra soundcards and
Audeze LCD-3 headphones. They did make a
difference, and they helped us to get rid of some
flaws that older productions of ours had.
“Headphones are a very good start,
especially if you don’t have a good room for
speakers or you can’t turn them up as loud as
you want to. But to put the finishing touches on
your mix, you might need a good pair of
monitors to double-check. Headphones can also
be very tricky in the sub-bass area, since you
can only hear the bass, not feel it. The more
reference speakers and headphones you have,
the better, but you have to get to know all of
their characteristics.”
: Do you normally A/B your mixes with
mastered tracks?
RR: “We always compare with our favourite
mixes, but we don’t let ourselves get confused
by them. There is no perfect mix, and different
elements need different mixing. One tune is
louder but has less bass; the other one you can
listen to very loud without grating, but when
you’re listening at a low level it lacks high end.
You have to find the right balance for the
elements you’ve been using. A/B-ing too much
is like copying, and we try to avoid that. We see it
as a nice inspiration for our mixes. Mixdowns are
part of creating a unique sound, so we don’t
want to get too hung up on other mixes.”
about them and hear some of their tracks at
www.electronicyouth.co.uk.
: What advice do you have to offer to
aspiring producers?
TK: “First, get good monitor speakers, like
Adams. If it’s too loud in your bedroom and the
neighbours are complaining, get decent
headphones. You should invest in a proper
soundcard as well – there are a lot of cheap ones
out there, but if you’re going to pay £70 you’re
not going to get the best sound. You need to pay
good money and get something by MOTU or
Apogee. I skinted myself for months to get my
MOTU soundcard!”
RR: “If you haven’t chosen a DAW yet, just pick
one and learn it. I remember [Prodigy producer]
Liam Howlett said he just locked himself in a
room for six months to learn Cubase and his
Roland W30 keyboard inside out!”
TK: “I’ve been on Logic for ten years and I’m still
learning something new to do with it every time
I use it.”
: How do you test your mixes?
TK: “There are a couple of ways. Because we’re
DJs, we test our mixes in the clubs. We sit there
: Do you have any other tips for getting
mixes to sound right in the club?
RR: “It’s important to use the right elements in
your tunes and know about frequencies. The
notes of your sub-bass should be in a certain
range to make sure there’s enough bottom end
in the raver’s belly. Unfortunately, club tunes
should be mastered quite loud to make sure
they can stand the pressure of the tunes before
and after, but make sure the loudness isn’t at the
cost of the sound quality – get the balance right.
A tune that’s made for the dancefloor should
also sound good in mono, so be sure to double-
check your mixes in mono!”
: How about making mixes that sound
good both in the club and at home?
RR: “As we say, mono-compatibility is a very
important part of a track sounding good
everywhere. Often, it’s not a mix issue if a track
doesn’t sound right in the club or at home – it’s
more about finding the right elements, whether
it’s drums or synths, and having tunes in the
right tonal range. Also, just follow your ears, and
check on different systems. Even the most
horrible-sounding mono speakers can help,
because they reveal flaws in your mix that other
speakers don’t.”
: What are your favourite plugins?
RR: “Everything from FabFilter is great – their
products are probably the most used in our
work. Their EQ is the most intuitive one out
there for us. We also dig all the iZotope stuff;
there’s much more to Ozone than just a limiter.
It’s the same with Alloy: the transient shaper is
on a lot of our drum elements.”
and straight away one of us will go, ‘That vocal’s
not loud enough!’, or, ‘The kick drum should
come up.’ It’s annoying because you’re noticing
stuff, but you can’t change it! Another way is to
test it on a standard stereo system. I know a
sound engineer who has a car stereo system in
his studio; so he’ll make the track, master it, then
play it on the car stereo to hear how it sounds.”
RR: “Another thing you can do is turn the
volume of your speakers right down, and if you
can still hear the vocal, you know it’s right. The
vocal should stand out a bit louder than
everything else, so if you can still hear it at a low
level, that’s a good sign.”
: Do you use spectral analysers?
TK: “I know someone who actually mixes on a
spectral analyser and tunes in all the sounds
with it. I can’t do that – I just do it all by ear. I used
to get other people’s tracks and compare them,
but I think that clouds things because you try
and make your music sound like their track. I
think you’ve got to go with what sounds good to
you, what sounds good in your ear. If you’ve just
started mixing, you won’t get it right from the
start. I was using an engineer for years before I
took the plunge and started doing it myself.”
“We always compare
with our favourite mixes,
but we don’t let ourselves
get confused by them.
There is no perfect mix”

COMPUTER MUSIC SPECIAL / 17

TC
Legendary DnB producer TC has created countless
dancefloor anthems and remixed artists as diverse
and Rhianna, Sub Focus and DJ Fresh Vs Diplo. You
can hear his music and get info on his label, Don’t
Play, at http://tcdnb.com.
: How do you ensure from the start that a project
is going to work in a club environment?
TC: “I think you never know, and this is why so many
people are DJ/producers: you just have to test it. When
I was first starting out I wasn’t a DJ, but I got to know
the engineer in a club and I’d go down there and play
my tracks before the DJs would start. That gave me a
lot of insight into what frequencies are going to work
or not work in that environment. There’s a lot of stuff to
do with the chain: when you play your music in a club,
you’re not just playing straight out of the speakers –
there’s usually a mixer, another mixer, then maybe an
EQ and compression. You’ve got to think that your
mixdown’s going to go through a lot of processing, so
maybe a little less in terms of limiting or clipping is a
little bit more for a club mix. Giving it a bit more room
to breathe, a bit more space, can make it sound louder
in the club.
“It’s a balance, but it’s a balance that’s been
achieved a lot of times, by many people. That’s your
guideline. Find all the tracks you love when you hear
them in a club in as high a quality format as you can,
and stick them all into an arrangement. Then look at
them on a spectrum analyser.”
: Do you A/B reference your mixes before
playing them out?
TC: “Not any more. I used to, but by now I’ve done a lot
of records, so I know what’s going to work. Also, I don’t
want to get too influenced, I want to make sure that I’m
doing original stuff. So making sure the mixdowns are
slightly different can give you the edge. I mix into my
final chain, My final chain will be as loud as a mastering
engineer’s chain. I’m not going to say what it is exactly,
but I’ll mix into four or five plugins, and none of them
are doing very much. If I was to take that off, I would be
mixing really quietly. You’d be surprised how quietly
everything goes to my final buss. The very last thing I
do is clip, and I use the Logic Bitcrusher to do that! It
took me a long time to work that out. I’ve tried
everything, every single plugin. I’ve run white noise
through plugins with a spectrum analyser so I can see
what they’re doing in terms of EQ, I’ll run test tones
through to see what kind of harmonics they’re giving
me up the scale. I go really, really deep – If you want to
get that edge, you have to go in deeper than the guy
before you.
“It’s all about balance at the end of the day, because
my mix wouldn’t actually sound that different with the
clipping on or off, it’s just a volume thing. As an
engineer, you need to learn not to be fooled by two
things: top end and volume level. Top-end EQ will make
something sound better even if it’s wrong, so you need
to not be fooled into thinking that just because
something’s brighter, it’s better.
“There’s a story about the mastering engineer Bob
Katz. He bought some new converters; he was trying
them out, and was like, ‘Wow, these sound amazing –
these are the best converters I’ve ever heard!’ Then he
discovered that they were calibrated slightly
incorrectly, and they gave him just half a dB more than
the other converters he was A/B-ing them with. So,
volume is like a drug to us humans, unfortunately!”
: Do you ever release your own DIY masters as
finished tracks?
18 / COMPUTER MUSIC SPECIAL
TC: “Only if it’s a free giveaway. If I’m going to sell a
record, I want everyone to have a piece of the pie. Plus,
having a track mastered gives you someone to blame
if it doesn’t sound good! If you’ve got a really shitty
record and you send it to mastering, it’s going to come
back just as shitty as it was. They’re not miracle
workers, unless you send them something awesome,
in which case they can make it really awesome.
“I don’t have a separate process for mastering,
though – once a track’s done, it’s done. I’m not going to
bounce the file out to another project to do a further
stage of mastering. I did do a track called Tap Ho,
where I gave away the parts and had about 300 or 400
entries for a remix competition, and I did master the
finalist for that. Some of the entries had really funky
stereo imaging that needed to be reigned in a little bit.
For that I’d just use Waves S1, which is an amazing
plugin. Then some of them would be too bright, or the
balance wouldn’t be right. Mostly, I find with young
producers that they forget about the midrange. They
get really excited about bass and top end, but they
forget about the place where all the musical
information is. For example, when you have a
telephone conversation, your voice is squeezed into a
very narrow band, but you can still understand every
word that’s being said. That’s all in the midrange.
There’s always a lot of high mid, but people always
forget about 600Hz down. There’s a lot of stuff going
on there that can add huge amounts of energy and
loudness to a record. The balance of the frequency
spectrum is just as important as the actual loudness of
the sounds you’re putting through to the master buss.”
: Have you made mistakes with your mixes?
TC: “In the old days, I did all the time. In fact, I just did a
remix for DJ Fresh, and I made a massive mistake! My
snare drum, for some reason, had sub-sonic
information that I only saw when I loaded it into Serato
to play it out. I found out that it was actually a plugin
on my final chain, because I’d changed something
slightly. Luckily I had time to make changes and
resubmit the masters.
“I’ll still change stuff now after playing it out. If
you’ve got mates who DJ, if you’ve got any access to
getting your tracks played out, get it done, because it’s
really important for every genre – even pop nowadays.
Your records have got to work in the club.”
: What advice would you give to new producers
who don’t play out themselves?
TC: “Make friends with DJs. Just send them your
tracks! Don’t do it straight away – make sure that you’re
at a point where your tracks are ready – but don’t be
too precious. Don’t just be stuck in your bedroom. Try
and get your tracks out to all the locals to start off with.
In a wider sense of club music, it’s all about building
relationships with DJs.
“I remember very clearly the first time I heard a
track of mine being played in a club environment. DJ
Krust played one of my tracks in the Thekla nightclub
in Bristol. It was one of those moments that I don’t
think I’ll ever forget. It was really cool to see everyone
dancing to my tune. I thought it was amazing – the best
thing ever!”
: How do you make sure a track works well at
home, too?
TC: “When you feel like you’ve finished a track, listen to
it on everything. When I finish a track, I upload it to my
Dropbox, I play it off my iPhone, I play it off a few
laptops, burn it to CD, play it in the car. It’s all about
referencing it in different places.”
“People always forget
about 600Hz down.
There’s a lot of stuff going
on there that can add
huge amounts of energy”

COMPUTER MUSIC SPECIAL / 19

Marc Mac
Reinforced Records co-founder Marc Mac
produces in a mind-boggling array of musical
styles, including hip-hop as Visioneers and
techno as Nu Era, but he’s probably best
known for his work as half of revered dance
act 4hero. Marc has just released an album of
90s-style hardcore rave under his Manix!
pseudonym, which you can listen to at http://
manix-reinforced.com.

: You work in a really wide range of dance

music styles. What would you say is your
overall mixing philosophy?
MM: “It’s horses for courses – the process is
different depending on the record you’re
working on. One of the things you have to try
and eliminate when you’re mixing and
producing or mastering is having a set chain of
processing units. The problem when you have a
set chain is that it pushes you to do everything
in a particular way. I prefer to listen to the track,
think about it, and then decide what it needs in
order to sound right.”

: Do you work with a clipper or limiter on

your master buss from the start of a project?
MM: “You know what? It happens somewhere in
between. Not at the very beginning, not at the
end, somewhere in between. Because in the
digital world, when you put those mastering or
limiting plugins on the end, you have to mix to
them as well. If you wait until the end before you
stick them on, sometimes you’ve got to change
the whole mix. So I find that if I put it on midway
through, I’m EQing and mixing toward what I
want the final thing to sound like, as opposed to
waiting until the end, sticking it on and
changing the characteristic.”

: With the Manix! record, how did you

balance achieving a modern production
sound while maintaining an old school feel?
MM: “If anything, doing the Manix! album was
one of the biggest challenges, because I’d A/B
between an old Manix! track and a new one, and
they’d be completely different sonically. The
thing is, I had to make the decision to make the
album for the new generation. So that led to
more loudness, more limiting, more EQ, making
“There are some great
all the kicks and snares cut through a lot more. I
had to join that loudness fight, which guitarists out there, but you
unfortunately is a fight because everyone’s
trying to be louder than everyone else. You can’t tell because they’re
have to join that without destroying the vibe.”

: It’s sad to say, but house music seems to

be losing certain sonic elements in the
pursuit of loud mixdowns…
MM: “I think the only way it’s going to change is
through knowledge. It’s not going to happen by
accident. It needs to happen by people making
an issue of it, including the DJs – they need to
understand what’s happening, too. It’s gonna be
tough; I don’t know how it’s going to happen.
“The same thing happened to rock music. If I
think about rock music from the 70s – I like
progressive rock – there could be some really
hard guitar parts, but they were played well
enough that you heard every note. Rock music
now, I just don’t like, because you’re not hearing
anything. It’s become so loud and distorted that
you’re not going to hear the great musicians.
fighting against a wall of noise”
There are some great guitarists out there, but
you can’t tell because they’re fighting against a
wall of noise!
“I think that the same thing is happening to
dance music in terms of programming.
Programming is almost becoming like it’s less
needed – producers feel like they don’t need to
put the handclaps in and add the extra
percussion and stuff, because it’s not going to be
heard in the club. That’s a danger, and I don’t
know how to get around it apart from trying our
best not to do it ourselves. Probably the only
good thing about MP3s and Spotify is that
they’re going to force people to make records
that sound great on a pair of headphones as well
as in the club.”
: A lot of people are listening to music on
their laptops nowadays, where there’s
obviously no bass…
MM: “I’ve posted music up on forums and stuff
and said, ‘Don’t listen to this through your
computer; plug your headphones in if you have
to’, because otherwise the music won’t make
any sense. It’s almost like there should be some
kind of disclaimer on the back of the CD!”

20 / COMPUTER MUSIC SPECIAL

Frederic Robinson
Blurring the lines between dance music and
contemporary classical, genre-blending
electronica wunderkind Frederic Robinson
has wowed all and sundry with his debut
album Mixed Signals. Wrap your ears around
his experimental masterpiece at http://
soundcloud.com/fredericrobinson.

: Do you use any visual referencing tools

like spectral analysers?
FR: “I sometimes do. When I started producing, I
read that you’re not supposed to have too many
instruments playing in the same frequency
range. I got used to looking at the spectrum of
all the sounds I used, and made sure they didn’t
overlap too much. To me, it was a great help for
my ears to learn what a clean mix is supposed to
sound like, and it also made it possible for me to
spot mistakes that were maybe not really
audible on my monitor setup. Over the years I
worked in better and better studio conditions,
and my ears got a lot of training. So now I don’t
really need spectral analysers any more,
because my ears can do the job by themselves.
When I work on cheap speakers or in a noisy
environment, I still use them, though, especially
for sub-bass frequency regions.
“From time to time I also use a goniometer
for measuring the broadness of my stereo
image. That isn’t necessary for basic imaging,
like panning a sound to one side, but it’s very
useful for seeing how broad the reverb of
certain samples is, for example. You can easily
miss if a reverb is mono or stereo if you’re not
working under perfect studio conditions, and
keeping an eye on that definitely helps you
create a beautiful, spacious stereo image.”

: What headphones and monitors are you

using at the moment?
FR: “I have a pair of Audio-Technica ATH-M50
headphones and Yamaha HS80M monitors. I got
them around three years ago and even though
they’re not the highest possible quality, I know
them quite well, so I don’t feel like I have to
upgrade any time soon. I know that the Yamahas
have a bit more highs than necessary, and the
headphones too few. When I produce, I regularly
switch from one to the other. It’s actually quite
“I don’t really need spectral
amazing how many things you can completely
miss on one monitoring system and analysers any more,
immediately notice on another. I often use the
headphones to work on moving stereo images, because my ears can do the
because they give you a more detailed picture
of where everything is in the mix.
“Working with monitors is more comfortable,
job by themselves”
though. Your ears take longer to get tired, and I
guess the volume is also a lot healthier. That’s
why I make most of my creative decisions while
working with monitors and save the detail work listening environments. On the other hand, “I started developing a live setup a few years
for later, when the melodies, harmonies and many things that are important in listening after I started producing, and the music I create
structure have been worked out.” music are not necessary in club music, so trying in that environment is quite different. I never
to introduce elements of that in dance music is really considered DJing an option for me,
: Your music takes influences from not really worth the effort. because I wanted to have as much control as
electronica and contemporary classical “I guess it all comes down to where you possible over the music I create on stage. I
music. Does this make it harder to get it experience most of the music you listen to. I currently play my live sets with a selection of
sounding right in a club environment? always mainly listened to music at home, so my acoustic sound sources, like electric violin,
FR: “I don’t actively try to make my music work main goal was to make my music interesting to lamellophone and a bunch of MIDI controllers. I
in clubs. It definitely is tricky, and to me it seems listen to and not to dance to. It’s just a better imitate some elements of my studio tracks when
like most attempts at doing that fail. There are a template for me to work in because I like the on stage, but it’s mixed with so much new
few things in music which work very well in feeling of not being restricted by traditional material that I can hardly say I have ‘played a
clubs, but just aren’t that interesting in other dance music structures. track out’. It sounds different at every gig.”

COMPUTER MUSIC SPECIAL / 21

 If you want to achieve truly slamming mixdowns, an
understanding of how digital audio works is essential. Let’s
find out exactly what’s going on ‘inside the box’

24 / COMPUTER MUSIC SPECIAL

 mixing in the digital domain <

It’s entirely possible to make music on a

computer without having the faintest idea of Fig 1: Quantisation
how digital audio works, but a little
knowledge can go a long way when it comes 15
to pushing the medium to its limits.
For those unfamiliar with the basics of how 14
sound works, let’s start at square one. Some of
the following explanations might seem a little 13
abstract out of the context of actual mixing, but
introducing these ideas now is important, as
12
they’ll be referenced when we get more hands- 11
on later in the issue.
Sound is the oscillation (that is, waves) of 10
pressure through a medium – air, for example.
When these waves hit our tympanic membranes 9
(eardrums), the air vibration is converted to a
vibration in the fluid that fills the channels of the
8
inner ear. These channels also host cells with 7
microscopic ‘hairs’ (called stereocilia) that
release chemical neurotransmitters when 6
pushed hard enough by the vibrations in the
fluid. It’s these neurotransmitters that tell our 5
brain what we’re listening to. 4
The process of recording audio works by
converting the pressure waves in the air into an 3
electrical signal. For example, when we (used to)
record using a microphone feeding into a tape 2
recorder, the transducer in the mic converts the
pressure oscillation in the air into an electrical 1
signal, which is fed to the tape head, which 0
polarises the magnetic particles on the tape
running over it in direct proportion to the signal. The smooth sine curve is the original analogue waveform; the squared version is the quantised digital waveform
The movement of sound through air – and,
indeed, the signal recorded to tape – is what 20,000 oscillations (or cycles) per second. This 3000Hz), we’ll instead get 19,050Hz (22,050Hz
we’d call an analogue signal. That means that it’s is where we encounter the Nyquist–Shannon minus 3000Hz). As frequencies above 20kHz or
continuous – it moves smoothly from one ‘value’ sampling theorem, which deals with how often a so can’t be perceived by the human ear, they’re
to the next without ‘stepping’, even under the sample of a signal needs to be taken for it to be not massively useful anyway.
most microscopic of scrutiny reconstructed accurately. The theorem states In ADC, what’s known as an anti-aliasing filter
that the sampling rate needs to be double the is used to band-limit the sound, stopping any
By the numbers highest frequency, lest what’s known as aliasing frequency higher than half the sampling rate
Computers are, for the most part, digital, which occurs (see Fig 2). Aliasing causes unmusical from getting through. So, that 44.1kHz rate gives
means they read and write information as artefacts in the sound, as frequencies higher us double our 20kHz, plus a bit of extra room for
discrete values – that is, a string of numbers. So, than the sample rate will be “reflected” around the low-pass filter to filter out the unwanted
how do computers turn a smooth waveform of the Nyquist frequency, which is half the sample high-end signal. 44.1kHz was the de facto digital
infinite resolution into a series of numbers that rate. So at 44.1kHz, Nyquist is 22,050Hz, and if playback standard all the way through the CD
they can understand? we try to sample, say, 25,050Hz (22,050Hz plus era, and seems to be maintaining its ubiquity.
The answer is a method called pulse code
modulation (PCM), which is the main system
employed when working with digital audio
information. It starts with the process of
analogue-to-digital conversion (ADC), which
Fig 2: Aliasing
involves measuring the value of the continuous
signal at regular intervals and creating a
facsimile of the original waveform. The higher
the frequency at which the signal is referenced –
or ‘sampled’ – and the greater the precision of
the value that’s recorded, the closer to the A
original waveform the digital recording will be
(see Fig 1).
The number of samples taken per second is
called the sampling rate. CD-quality audio is at a
sampling rate of 44.1kHz, meaning that the
audio signal is sampled a whopping 44,100
times per second! This gives us a pretty smooth
representation of even high frequencies – the
higher the frequency we want to measure B
accurately, the higher the sampling rate
required. That figure – 44.1kHz – is pretty
specific, and there’s a reason for that: the
highest frequency that can be represented by
PCM audio is exactly half of the sample rate, and The grey waveform is the original; the purple is the digital version. The vertical lines represent samples taken from
the human ear can hear frequencies from the waveform. If the sample rate isn’t at least twice the frequency of the highest frequency present in the signal, the
around 20Hz to 20kHz, so at the top end that’s waveform won’t be accurately recreated

COMPUTER MUSIC SPECIAL / 25

> mixing in the digital domain
Fig 3: Bit depths
1-bit (two possible states)
2-bit (four possible states)
4-bit (16 possible states)
8-bit (256 possible states)
26 / COMPUTER MUSIC SPECIAL
Once our sample has been taken, it needs to
be stored as a number. You probably already
know that CD quality audio is 16-bit – but what,
exactly, is a bit? A bit (short for binary digit) is
the most fundamental unit of information in
computing, and can have one of two values –
usually represented as 1 or 0. If we add another
bit, we double the number of available states:
for two bits, the number of available states
would be four (two to the power of two) – that is,
00, 01, 10 and 11. Four bits would allow for 16
states (two to the power of four), eight bits
would allow for 256 states (two to the power of
eight), and 16 bits a massive 65,536 possible
states (two to the power of 16). So, we’ve gone
from a primitive ‘on or off’ state at 1-bit to having
quite a fine resolution at 16-bit. See Fig 3 for a
visual representation of this.
A bit random
When an analogue-to-digital converter samples
an analogue signal and rounds it up or down to
the nearest available level, as dictated by the
number of available bits, we get ‘quantisation
error’ or ‘quantisation distortion’. This is the
difference between the original analogue signal
and its digital facsimile, and can be considered
an additional stochastic signal known as
‘quantisation noise’. A stochastic signal is non-
deterministic, exhibiting different behaviour
“A stochastic signal is
non-deterministic,
exhibiting different
behaviour every time
it’s observed”
every time it’s observed – a random signal, in
other words.
So, in a digital recording, we get our ‘noise
floor’ (the smallest measurement we can take
with any certainty) from the number of bits
we’re using. This is known as ‘word size’ or ‘word
length’. For every bit we add to the word length,
we get twice the number of possible states for
each sample, doubling the dynamic range with
another 6dB or so of headroom over the noise
floor. For example, a 16-bit digital recording at
CD quality gives us a hypothetical 96dB of
dynamic range. It might surprise you to learn
that this is actually higher than the dynamic
range of vinyl, which is more like 80dB. If you’re
unfamiliar with decibels and dynamic range,
don’t worry – we’ll come back to them shortly.
As you can see by looking at the same sample
at different bit depths, changing bit depth
doesn’t affect how loud a sample can go; it just
affects the volume resolution of the waveform.
A common misconception is that increasing bit
depth only improves the resolution at lower
volume levels, but this clearly illustrates that the
increase in resolution improves the accuracy of
the signal across the entire amplitude.
If you’re new to music production and this all
seems very abstract, perhaps a practical
example will bring things into focus. Install the
free audio editor Audacity (download it at http://
 mixing in the digital domain <

audacity.sourceforge.net), launch it and set

the input to your computer’s built-in
microphone. Click the record button, and
repeatedly speak the word “test” into the
microphone, quietly at first, then getting louder
each time. You should see something like Fig 4:
a series of similar waveforms that increase in
size on the vertical axis. If your computer
doesn’t have a built-in mic and you’re unable to
connect a microphone to it, you can find a pre-
recorded audio file (Test.wav) in the Tutorial
Files/Mixing in the digital domain folder.
Fig 4: As the recording of the voice gets louder, the waveform increases in amplitude
Testing, testing
Let’s take a look at our recording up close. Click
the first “test”, then hold the Cmd key on Mac or
Ctrl on PC and press 1 to zoom in on the
waveform. The waveform will look continuous,
but once you’ve zoomed in close enough, you’ll
see discrete points representing the levels of
individual samples – see Fig 5.
Hold Cmd/Ctrl again and press 3 to zoom out
and see the entire waveform. The increase in
vertical size represents the words increasing in
volume, or amplitude. Now click the loudest
“test” and zoom in on it. At the loudest points,
the waveform will approach or even reach the
+1.0/-1.0 points of the vertical axis. No matter Fig 5: By zooming into the waveform horizontally in Audacity, you can see the positions of individual samples
what bit depth you use, +/-1 is the loudest level
your recorded signal can be.

“Having a broad
dynamic range is
important in film
soundtracks, as well
as classical music”
However, the greater the bit depth, the more
accurate the waveform, and this is particularly
important when dealing with signals that
contain both loud and quiet elements –
percussive sounds, for example. This is because
more bits give us more dynamic range. Dynamic
range is the ratio of the largest and smallest
possible values of a variable such as amplitude,
and it can be expressed in terms of decibels
(dB), which we’ll get into in more detail later. In Fig 6: The top waveform is the original drum beat. The second is the compressed version with less dynamic range,
simple terms, it’s the difference in dB between but an overall louder level. The final waveform has been compressed too much, and sounds weak
the loudest and softest parts of the signal.
Here’s an example of dynamic range in action
that most of us will be familiar with: you’re
watching a movie late at night, trying to keep
the volume at a level that isn’t going to disturb
anyone. The trouble is, the scenes in which the
characters talk to each other are too quiet, while
the scenes with guns and explosions are too
loud! Wouldn’t it be better if the movie’s
soundtrack was at a consistent volume level?
Well, for late-night home viewing, maybe; but
in a cinema, it’s the difference in amplitude
between the characters’ dialogue and the
explosions that gives the latter their impact. As
such, having a broad dynamic range is
important in film soundtracks, as well as
classical music, where you’ll get much quieter
sections than you ever would in a dance track. Fig 7: The top waveform is the original version. The lower one has been reduced in amplitude by about 6dB, and
For example, take a listen to the fourth peaks at half the amplitude of the original

COMPUTER MUSIC SPECIAL / 27

> mixing in the digital domain
Fig 8: Inter-sample peaks
0dBFS
-1nf
movement of Dvořák’s New World Symphony,
which features very delicate, restrained sections
and intense crescendos. Dynamic range is
important in club music too, though, because it
helps keep the percussive sounds punchy.
Range finder
The importance of dynamic range becomes
very apparent when compressing drum sounds.
For the uninitiated, compression is a process
whereby the dynamic range of a signal is
reduced to get a sound with a louder average
level. Once the level of the signal exceeds a
certain volume, the amount it goes over that
threshold is reduced by a ratio set by the user.
To compensate for the overall drop in volume
that this introduces, the compressor’s output is
then boosted so that it peaks at the same level
as the uncompressed signal. As the average
level is now higher, thanks to the reduction in
dynamic range, it sounds louder – see Fig 6.
However, too much compression can suck the
life out of a sound, making it feel flabby and
weak. You can hear some examples of this in the
Tutorial files folder: Original beat.wav,
Compressed beat.wav and Over-compressed
beat.wav. Getting the right balance between
sounds with punch, but also maintaining high
average volume, is one of the most important
aspects of mixing for club play.
Generally speaking – and up to a point – the
louder music gets, the better it sounds. Sadly, as
we’ve seen, our waveform can only get so loud:
once all the bits have been set to 1 and we’ve
reached the 65,536th step, there’s nowhere else
for the audio information to go! So, what
happens if we’re recording a signal that’s too
‘hot’ (that is, too loud) for our system? In this
situation we get what’s known as digital
clipping. When a signal exceeds the maximum
capacity of the system, the top of the waveform
is simply cut off, or ‘clipped’. This loss of
information misrepresents the amplitude of the
28 / COMPUTER MUSIC SPECIAL
clipped waveform and produces erroneous
higher frequencies. Although clipping can have
its uses – which we’ll look at later in the issue –
it’s generally undesirable and is the reason why
we can’t just keep boosting the level of our
mixes to get them as loud as we want.
As a music connoisseur, you’ve probably
already noticed that some mixes actually sound
a lot louder than others. Sometimes this is
because their waveforms simply peak at a
higher level, but not always – you might load up
two tracks in your DAW and see that they look
similar in terms of amplitude, but hear that one
sounds a lot louder than the other. This is where
perceived volume comes into play. The human
“Decibels represent
the ratio between two
different values, so we
use them to describe
amplitude relatively”
mind’s perception of loudness is based not just
on the amplitude level at which a waveform
peaks, but also on how long it remains that loud.
Consequently, a better way to quantify the
perceived volume of a sound is by measuring
not its peak volume, but its RMS level.
RMS stands for ‘root mean square’, and is a
method of calculating the average amplitude of
a waveform over time. Say, for example, we try
to calculate the mean amplitude of a sine wave.
The result is going to be zero, no matter what its
peak amplitude, because a sine wave rises and
falls symmetrically around its centre origin.
Clearly, then, this isn’t a useful way to measure
Inter-sample peak
average amplitude. RMS (the square root of the
mean of the squares of the original values)
works better because it always gives a positive
value by negating any negatives – it’s a measure
of the magnitude of a set of numbers. So, a
signal that has been compressed may have the
same peak level as the uncompressed version,
but its RMS level will be higher, making RMS a
better indicator of perceived volume.
Talkin’ loud
When we talk about signal levels, we use a
logarithmic unit known as the decibel, or dB for
short. Decibels represent the ratio between two
different values, so we use them to describe
amplitude relatively. For example, in digital
audio, a signal at 0dBFS (0 decibels full scale)
reaches the maximum possible peak level (+/-
1.0) on the vertical axis. Putting it simply, +6dB
doubles the amplitude of a signal, and -6dB
halves it. If you want to get technical, it’s actually
approximately +/-6.02dB (derived from 20 log
2). So, If we had a sine wave that peaked at
0dBFS, and amplified it by -6.02dB, it would
peak at about +/-0.5. Each time we amplify the
signal by -6.02dB, the peak level will fall by
approximately half again. In terms of RMS,
amplifying a signal by +/-6.02dB will give us four
times the power or a quarter of the power
respectively (see Fig 7).
We calculate a period’s crest factor (the
extremity of peaks in a waveform) by dividing its
peak value by its RMS value. The resulting dB
value will give us some idea of how dynamic a
signal is. You can measure peak, RMS and crest
factor dB using metering software such as Blue
Cat Audio DP Meter Pro (www.bluecataudio.
com). Simply put it on your master channel, and
as your track plays it’ll update the peak and RMS
averages (the time settings of which can be
adjusted) and resulting crest factor in real time.
So, part of the reason why two tracks can
have such different perceived volume levels

when they peak at the same level is that they
likely have a higher RMS value. That’s not the
whole story, though. There are many
psychoacoustic factors that affect how loud
something sounds to our ears, including its
frequency content, how long it lasts and,
crucially, its context in a track.
In the mix
This is where we actually get into the realms of
arrangement and – finally! – mixing. At the most
fundamental level, mixing two waveforms
together is a really simple process: all your CPU
has to do is add the level of each sample state
together to calculate the total. This happens in
exactly the same way in all DAWs, and if
someone tells you that a particular DAW’s
summing engine has a certain ‘sound’ to it,
they’re sadly misinformed. In fact, there’s a very
simple test you can do to see for yourself how all
DAWs sum signals in exactly the same way:
simply load two or more audio tracks into the
DAWs you want to test and bounce out a
mixdown. As long as you use the same settings
(including any automatic fades, panning law and
dithering) and haven’t done anything to alter
the audio (such as apply timestretching), you’ll
find that each DAW produces an identical file.
Then load the exported files into a DAW and
apply some polarity (often erroneously called
phase) inversion – as long as all the files have
been imported correctly and no effects have
been applied by the DAW, they will cancel each
other out perfectly, leaving nothing but silence.
Another thing worth bearing in mind about
DAW mixers is that they use 32- or even 64-bit
“Compressed audio
formats such as MP3
and M4A have the
advantage of much
smaller file sizes”
‘floating point’ summing to calculate what the
mix should sound like before it’s either delivered
to your ears via your audio interface and
speakers or bounced to an audio file. Floating
point means that some of the bits are devoted to
describing the exponent, which in layman’s
terms means that it can be used to store a much
wider array of values. Whereas fixed-point 16-bit
has a range of 96dB, 32-bit floating point has a
range of more than 1500dB! This is clearly more
headroom than we’d ever have any practical use
for, and is the reason why pushing the level
faders extremely high or low in your DAW’s
mixer doesn’t really have a negative effect on
the sound unless it’s passing through a plugin
that doesn’t cope well with extremely high or
low values, or is clipping the master channel. In
fact, you could run every channel on your mixer
well into the red, but as long as the master
channel isn’t clipping, no distortion of the signal
would occur: 32-bit float can handle practically
anything you can throw at it.
Running your DAW’s master channel into the
red is a bad idea, because even if it’s not
mixing in the digital domain <
Fig 9: Visually represented dithering
8-BIT GRADIENT 8-BIT GRADIENT
DITHERED
producing any audible distortion, you won’t
really be able to trust what you’re hearing any
more; this is due to the fact that different digital-
to-analogue converters handle signals
exceeding 0dBFS differently. Distortion can
even be caused by signals that don’t clip! This is
known as inter-sample modulation, and it
doesn’t occur within your DAW – it only happens
once the signal has been converted from digital
to analogue by your audio interface, specifically
the digital-to-analogue converter’s
reconstruction filter. It’s what turns a stepped
digital signal into a smooth analogue one, and
as Fig 8 shows, this process can cause problems
when used with especially loud digital signals.
Whenever audio leaves your DAW – be it as
sound or an audio file export – it’s converted
from a 32- or 64-bit signal to a 16- or 24-bit one.
When this happens, resolution is lost as less bits
are used. This is where an oft-misunderstood
process known as dithering comes into play,
whereby a very low level of noise is added to the
signal to increase its perceived dynamic range.
It works a little like this: imagine you want to
paint a grey picture, but only have white and
black paint. By painting very small squares of
alternating white and black, you can create a
third colour: grey. Dithering uses the same
principle of rapidly alternating a bit’s state to
create a seemingly wider range of values than
would otherwise be available. See Fig 9 for a
visual example of dithering in action.
Dithering should be applied when lowering
the bit rate, so it’s sensible to apply dither to
your 32- or 64-bit mix when exporting it as a
16-bit WAV file for burning to CD to play in a club,
or when supplying a 24-bit WAV file to a
mastering engineer.
WAV and AIFF are uncompressed audio
formats, which means that they’re suitable for
using to transfer audio data between DAWs and
24-BIT GRADIENT
audio editing applications. Compressed audio
formats such as MP3 and M4A have the
advantage of much smaller file sizes, but use
psychoacoustic tricks to remove data from the
signal that the human ear finds hard to perceive.
As such, these ‘lossy’ formats still sound great
under normal consumer listening conditions,
but they don’t behave exactly like their
uncompressed counterparts when subjected to
processing. While using an MP3 file as a source
for, say, a CD master isn’t necessarily going to
sound bad, the results will really depend on the
material, the context it’s used in, and the
encoding algorithm that’s been used to
compress the sound. To be sure that the finished
product sounds as good as possible, it pays to
ensure that you use the highest-quality source
material you can.
Moving on
The human ear and mind are surprisingly
sensitive listening devices, and as a result of this
it’s easy for someone who knows nothing about
digital audio or production to identify a bad or
weak mix – it will simply sound less ‘good’ than
other tracks. This is especially important in a
club environment, where the music is usually
delivered in a non-stop stream, mixed together
by a DJ. When the DJ uses the crossfader to
quickly cut between channels on the mixer, it’s
just like A/B-ing tracks in a studio environment:
any deficiencies in the mixdown of either track
compared to the other will be glaringly obvious!
To get your mixes sounding as good as the
competition it’s necessary to push your digital
audio to its very limits, and it’s only through
understanding what those limits are that you
can achieve this. In the following tutorials, we’ll
take the digital medium to the edges of what it’s
capable of, and show you the negative
consequences of exceeding those limits.
COMPUTER MUSIC SPECIAL / 29
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 Traditionally, mixing and mastering have been
performed by specialists. Nowadays you might find
you have to deal with them yourself, so how do you
ensure the finished product sounds professional?

COMPUTER MUSIC SPECIAL / 31

 Before the advent of the internet, record
labels had no trouble at all selling physical
product, and there was a lot of money in the
music business. Labels could afford to pay
songwriters, artists, backing singers, session
musicians, arrangers, engineers, producers,
mix engineers, mastering engineers – even
quality assurance people! – in the quest to
create potential hits. Nowadays, particularly
in electronic music, musicians usually have to
do everything apart from mastering
themselves. Indeed, some digital labels even
release their artists’ own masters without
batting an eyelid!
Mastering is the process of creating a version
of a track suitable for duplication on its intended
final medium. Mastering engineers typically
work in purpose-built rooms with lots of
acoustic treatment, tons of specialised hardware
and software, and ultra-high-end monitors the
sound and character of which they know inside
out. A mastering engineer’s job is to know how
any given piece of music should sound on those
monitors, and what it takes to get it there.
Historically speaking, mastering has been
particularly important when it comes to making
vinyl records, because vinyl has certain
limitations that must be adhered to in order for
the final product to sound any good – or even
function properly. For example, extreme high-
frequency signals can cause havoc with the
cutting lathe, and excess stereo information in
the low end can actually cause the needle to
jump out of the groove on playback!
Naturally, then, it makes sense to employ a
specialist audio engineer who can anticipate
and correct these problems, creating a version
of the recording that will sound great when
pressed to vinyl. Digital formats such as CD and
MP3 are a lot more forgiving because there are
no cutting lathes or needles to worry about, but
The loudness war
Attempting to one-up the last release with
one that sounds louder and thus bigger is
something that’s been going on since the
days of the 7-inch single. Reducing dynamic
range might make music instantly more
gratifying, but many would argue that it
also reduces its impact and makes it more
fatiguing on the ear. It’s something that
provokes strong opinions from audio
industry professionals and fans alike –
indeed, after the release of Metallica’s 2008
album Death Magnetic, fans noticed that the
version of the album used in musical
videogame Guitar Hero actually sounded
much better than the squashed CD release
(http://bit.ly/bHxvpv) – visual evidence of
this can be seen at http://bit.ly/TjWizz. A
petition was organised to release a less
heavily compressed version of the album
on CD, though it didn’t amount to anything.
Although the same ratcheting up of
volume levels has occured in dance music, it
doesn’t seem to have had such an adverse
effect on the genre – certainly not in such
32 / COMPUTER MUSIC SPECIAL
they do still have their own considerations –
consider, for example, the inter-sample peaks
discussed on p29. Mastering engineers will use
equalisation, dynamics processing, stereo
balance adjustment and even saturation, among
other techniques, to get tracks sounding as
good as they possibly can prior to release. If the
engineer is working on a multitrack project, like
a two-tracker, EP or album, they’ll also need to
ensure that the tracks have a consistent feel and
volume level. Getting a sensible balance
between volume level and dynamic range is
another responsibility of the mastering
engineer, and increasingly over the years,
they’ve been encouraged to make tracks louder
and louder – see The Loudness War, below.
“It’s easy for an
amateur to do more
harm than good to
their music with
overzealous limiting”
Master minds
In an ideal scenario, then, once an artist or
producer has finished a piece of music, they’ll
send a version that hasn’t been subjected to
master buss processing of any kind (limiting,
most pertinently) and still retains decent
dynamic range to a mastering house. This
version will usually sound rather quiet
compared to a commercial release, but the
engineer (who ideally specialises in mastering
immediately tangible terms, anyway.
By its very nature, electronic music
responds much better to a reduction
in dynamic range than the more
organic feel of rock, allowing dance
producers and mastering engineers
to process their productions more
severely without such an obvious
reduction in sound quality.
On the other hand, the loudness
war has actually had a far more
dramatic effect on dance music in
that it’s changed the way it’s written
from the ground up. As legendary
producer and remixer StoneBridge
noted in Special 61: “Sequences
are doing the job that hi-hats used
to: you have little bubbly sequences that
sort of play the hi-hat pattern. It’s a musical
element, so it will carry the tune, and
producers have discovered that they can
make a cleaner production by avoiding all
those treble elements like hats and snares –
if you remove that shit, the mix will be a lot
the particular genre in question) will do an
amazing job of creating a slick, loud yet punchy
version of the track for every intended medium.
Of course, this isn’t always possible. You
might not have the budget to recruit the
services of a reputable mastering house, or you
might need to make a loud version of your track
to test out in a club environment at short notice.
In these situations, you’ll have to resort to
creating your own DIY master. This isn’t the ideal
scenario for many reasons: pro mastering
engineers will have superior monitoring, tons of
specialised equipment and, most importantly,
experience and skill in this area. It’s easy for an
amateur to do more harm than good to their
music with overzealous limiting or excitement,
especially if their monitoring setup isn’t ideal.
That said, this state of affairs isn’t without its
upside. In the past, its been possible for
producers who don’t really know what they’re
doing to deliver less than stellar mixdowns,
which have then been knocked into shape by
mastering engineers. This has resulted in
producers who never take their tunes to the
next level, because they know the mastering
engineer will be there to make their efforts
sound acceptable. Now, however, when every
part of the process is performed by a single
musician, they experience each stage of the
transformation from unmastered mix to
mastered track. They get to really interact with
the production and hopefully more fully
understand how changes to the mixdown affect
how the mastered version will sound.
The most effective way to make club music is
to build it properly, from the ground up. These
days, the lines between sound design,
composition, mixing and mastering are fuzzy, to
say the least, and learning how they work
together is essential if you’re going to make
tracks that sound good wherever they’re played.
cleaner, you can master it louder, and it will
sound more devastating in the club. So we
don’t have any personality left in the
records.” In other words, it is possible to
make very loud mixes, but be warned – it
can come at the expense of making
interesting music.
Level best
One criticism sometimes leveled
at the graphical metering in DAW
software is that it doesn’t really
tell the whole story: most meters
only show peak rather than peak
and RMS levels, don’t show inter-
sample peak level, don’t have
adjustable ballistics (the way the
meter responds to the audio
input), and don’t have great
labelling. Thankfully, this
situation can be rectified
somewhat using plugins: there
are plenty of metering plugins on
the market, including some
excellent freebies.
Starting at the cheapest end of
the scale, Sonalksis’ FreeG (www.
sonalksis.com) is a must for
those after improved metering,
featuring peak and RMS
readouts, multiple metering
ballistic types, and a huge gain
fader that’s great for making fine
adjustments to a track’s level. It
also includes a polarity inversion
switch that could come in handy
for comparing tracks and more
advanced tracking.
Another essential freebie,
Voxengo’s SPAN is a spectral
analyser at heart, but also offers
level metering, stereo correlation
metering and clipping detection
amongst its many features. It can
be downloaded from www.
voxengo.com.
If you’re after something a
little more retro, PSP Audioware’s
2Meters bundle could be right up
your alley. It consists of two
meters with old-fashioned
VU-style displays and overload
indicators, and boasts adjustable
parameters such as integration
times (set by default to emulate
standard hardware VU meter
ballistics) and reference levels.
2Meters costs a very reasonable
$15 and can be bought at www.
pspaudioware.com.
If you’re really serious about
your metering, Blue Cat Audio’s
DP Meter Pro (€59, www.
bluecataudio.com) offers
Metering plugins
practically every kind of
like Sonalksis
metering possible including FreeG, Voxengo
customisable peak, RMS and Span and Blue
crest factor, five different scales Cat Audio DP
including Bob Katz’s K-System Meter Pro can
(plus the ability to make your be a useful
alternative to
own), graphs that show the the lacklustre
development of audio envelopes metering found
over time, a mid/side switch for in most DAWs
checking mono compatibility,
and tons more.

What many novice producers fail to grasp is
that different sizes of sound delivery system
work in completely different ways. For example,
it’s possible to create a dubstep track packed
with big, full-frequency noises that sounds
impressive when played through a laptop’s built-
in speakers, but if the low frequencies aren’t
present and the beats don’t have punch –
characteristics that are much harder to discern
or appraise on laptop speakers – it’ll sound
terrible on a big club rig.
On the down low
Without getting too sidetracked by the science
of how sound works, it’s the lowest frequencies
that give a sound its overall pitch and weight
characteristics. Typically, the lowest frequencies
in a piece of dance music will be the
fundamental frequencies of the kick drum,
snare drum and bassline. We can’t overstate
how important this is: in the same way that any
individual sound’s pitch is determined by its
fundamental frequency, so is the fundamental
frequency of a combination of sounds. The
lowest frequencies of a track define its overall
pitch, and a track’s rhythm is defined by the
sounds with the widest dynamic range – again,
that’s typically the drums. So, these elements of
the music are what give it its weight, solidity,
overall pitch and rhythm.
As these low-frequency drum sounds are
so important, it makes sense to start your
writing process with them. This might not be
how you’re most comfortable working, and it’s
certainly not mandatory, but getting the drums
sounding right is half the battle when it comes
to making an effective club track.
Understanding how drum sounds work makes
this easier, and is something we look at in Low
End Theory on p50.
Other instruments will be subjected to high-
pass filtering so that their low frequencies don’t
clash with these fundamental elements. This is
important for two reasons. The first is avoidance
of the phenomenon known as frequency
masking, which occurs when sounds occupy the
same space in the frequency spectrum. For
example, say the fundamental frequency of
your synth pad conflicts too much with the
harmonics of your bass – by high-passing the
pad, you’ll reveal the bass harmonics that share
those frequencies. The second reason is that it
helps to free up headroom. You can only put so
much energy, or amplitude, into each part of the
frequency spectrum – if you just keep adding
low-frequency sounds to the mix you’ll end up
clipping the master output, forcing you to turn
everything down. By keeping a good balance of
sounds through the frequency spectrum,
everything can be kept relatively loud, giving
you a more impressive-sounding mix.
This means that being able to identify the
frequency range that a sound occupies is
extremely beneficial to the producer. Our ears
are sensitive listening devices, but only those
with extremely well-trained ears can pick out
specific frequency peaks or accurately judge the
frequency content of a sound. Thankfully, there
are a couple of things that can help with this, the
first of which involves using parametric EQ to
boost specific frequencies in a signal in order to
exaggerate them, or remove them so that the
other parts of the signal can be heard more
clearly. The second method is to use a spectral
analyser such as Voxengo SPAN (free from
www.voxengo.com) to visualise the frequency
content of the audio signal.
Both of these methods are particularly useful
when working in a listening environment that’s
less than perfect. While low- or mid-range
monitors are in a totally different league to
laptop speakers for referencing mixes, they still
colour the sound to a certain extent, depending
on their build quality. The sound will also be
coloured by the environment in which they’re
placed, which is why high-end studios are
usually purpose-built and acoustically treated.
For example, if your room’s acoustic properties
mean that the frequencies around 100Hz
appear louder or those around 80Hz appear
quieter than they actually are, you’ll naturally
‘correct’ these frequencies without realising that
it’s the room causing the problem, not the mix
itself. When you play your mix in another
environment, these ‘corrections’ will result in a
mix with an uneven frequency response – the
100Hz area will likely sound too quiet, while the
80Hz region will be too loud!
In the club
Another important goal with any mix is that it
should translate well, and this is particularly

COMPUTER MUSIC SPECIAL / 33

important when mixing for club play – a loud
club system will emphasise poor mixing, and
piercing frequencies could become particularly
unpleasant, and even dangerous! A good way to
avoid this sort of problem it to check your mixes
on a variety of systems – friends’ hi-fis, your car
stereo, your laptop speakers and anywhere else
you can think of. It’s also worth referencing your
mixes on headphones as well as monitors.
Headphones won’t give you as accurate a
representation of dynamics as monitors, but
they can be useful for getting a better idea of
bass level (which can be particularly tricky to
get right when working in small, untreated
rooms) and fine details.
A highly recommended way of establishing
the sort of mix you should be aiming for is to A/B
reference your work with that of other
producers. The first stage of this process is to
find several pieces of music that you think are
well produced and sound good on a variety of
systems. Try to go for tracks that fit closely with
the style of music you’re making: you’re never
going to get your delicate, electric piano laden
hip-hop grooves to sound as loud as a sawtooth-
Don’t (always) believe your ears
The human ear is an extremely sensitive
listening tool, but it has its limits and can’t
always be completely trusted. If you’ve been
producing for a while, you’ll almost certainly
have noticed the phenomenon of music you’ve
made late at night sounding dreadful the next
day once you’ve listened to it with fresh ears!
This can be down to several factors, including
what’s known as ‘ear fatigue’. This is what
happens when you’ve been listening to loud
sounds for an extended period and your ears
become less sensitive to subtle changes in tone
and volume, and it’s just one of the reasons why
it makes sense not to send off your tracks the
second you finish making them.
Another important phenomena to be aware
of is how the ear responds to frequencies at
different levels. The human ear doesn’t have a
‘flat’ response of perceived volume (the
scientific unit of which is known as a ‘phon’) at a
particular sound pressure level (or dB SPL). The
first research on this topic was conducted by
Bell Labs engineers Fletcher and Munson, who
created the first equal-loudness curve. This
visual aid show us how the ear perceives
frequencies at various volume levels over the
range of hearing, though it has since been
superseded by more accurate curves. You can
check out some annoatated examples of equal-
loudness curves at http://bit.ly/16IJZ0Z, but the
main thing you need to know is that at lower
levels you need proportionally more bass and
treble to get a sounds that’s perveices as ‘flat’ –
hence the ‘loudness’ EQ curve modes on some
hi-fis that are designed to boost these
frequencies. If you’re mixing at really quiet
levels, you might find yourself artificially
boosting the lows and highs, which will result in
an unbalanced mix when played on a loud
system. If it’s not possible to monitor at a
reasonable level, use headphones, spectral
analysis plugins and A/B comparison to check
your work.
34 / COMPUTER MUSIC SPECIAL
laden trap banger. Also, aim for a fairly wide
range of mixdowns – for example, a track with
glossy high-end sheen, a track that really
punches satisfyingly in the lows, and so forth.
Get the highest-quality versions of these tracks
that you can – you can buy music in
uncompressed WAV format from Beatport, or
rip WAVs from audio CDs.
Next, add some extra audio tracks to your
project, route them directly to your monitors so
that they bypass any processing on your master
buss, then mute them. Now you can place a
reference track on each one. When you play
your project back you can quickly swap
between your mix and the reference tracks by
soloing them. This should quickly give you a
very clear idea of how your music compares to
the competition. It can also really help to use
spectral analysis on the tracks you’re A/B-ing –
you’ll almost certainly notice that each part of
the frequency spectrum peaks at a similar
volume level, and this is generally something
you’ll want to emulate in your own mixes.
If you’re a new producer, the first thing you’ll
probably discover is that the commercial mixes
130
120
110
100
Sound pressure level (dB SPL)
90
80
70
60
50
40
30
20
10
0
-10
10 100
Equal-loudness contours
Fletcher-Munson curves
are much, much louder than your work in
progress! This is thanks to the magic of
mastering and, naturally, makes it extremely
tricky to make a comparison: realistically, it’s
very hard to make an accurate judgement on
what you need to do to make your tracks sound
better when your references are so much
louder. The obvious solution is to turn the tracks
you’re using to A/B down so that they’re in the
same ballpark level-wise as your track.
To the limit
Another technique favoured by dance music
producers is using processing on the master
buss during the composition stage to get an
idea of how hard a mix can be pushed.
Let’s look at an example of how this might
work. Say you’ve got a mix that doesn’t have any
processing on the master channel and peaks at
about -6dB. This is a very sensible level for a
mixdown that ensures plenty of headroom,
giving the mastering engineer plenty of
dynamic range to play with. So far, so good – but
unless you’re a very experienced producer
you’re probably going to find it hard to judge
(Estimated)
100phon
80
60
40
20
(Threshold)
1000 10k 100k
 Labels and mastering
When prospectively sending
music to record labels, a DIY-
mastered 320kbps MP3 should
be high enough quality – it’s
practically impossible to hear
the difference between that and
a 44.1kHz/24-bit WAV under
normal listening circumstances.
If a label decides to release
your music, what happens from
there is down to them. Ideally,
they’ll ask you for a WAV version
with no compression or clipping
on the master channel, peaking
at around -6 to -4dB so that they
can give it to a mastering
engineer who will create
masters for digital and/or vinyl
distribution. Sometimes, small
labels (often staffed by just one
person running the whole thing
in their spare time) that only
release digitally will create
masters themselves. Try to find
out how the release will be
mastered, and whether it’s
going to be possible for you to
review the master before the
release date – you might find
(especially with label self-
masters) that you’re unhappy
with the job they’ve done or
have suggestions to make. If
this is the case, try to be as
tactful as possible!
An increasingly common
trend is for labels to simply
request a WAV version of your
DIY master, which will be given
directly to the digital
distributor. Obviously this isn’t
an ideal situation, but for low- Thanks to software like iZotope’s Ozone mastering suite, you really can make your
income labels it makes financial own high-quality masters these days – as long as you know what you’re doing!
sense. When this is the case,
there’s not a lot you can do aside using clipping on the master is almost always a better
from either doing the best job channel of the mixer, rather solution than applying, say,
on your DIY master as possible, than working from a single corrective EQ or compression
or paying to have it mastered rendered audio file, is that you after the fact, and will at least
yourself. The advantage of can make changes to the sound give you total control over the
creating your own DIY masters of the master as you see fit. This final sound of the release.

how loud it can go when it’s limited, or what it
will sound like. The solution it to have a crack at
reducing the dynamic range yourself. You can
do this simply by putting a limiter plugin on the
master channel and listening to how it sounds
when you boost the track’s gain.
The definition of ‘limiting’ is broader than you
might think. Compression of more than a certain
ratio – let’s say 10:1, with the output signal only
rising by 1dB for every 10dB by which the input
exceeds the threshold – could certainly be
considered limiting, but with what’s known as
‘brickwall’ limiting, the output signal never
exceeds the threshold – useful in situations
where you’re up against a hard limit like 0dBFS.
Then there’s clipping, which we discussed on
p28. Clipping is useful because, rather than
taking time to get the volume level down to the
desired level as compression does, the top of
the waveform is simply lopped off. The upside of
this is that it can make the signal very loud and
is better at preserving transients (the
characteristic attack stages at the start of drum
sounds) than limiting. The downside is that
prolonged periods of clipping result in an
unpleasantly distorted signal
So, by using a clipper on the master channel
and increasing the gain level, we can get an
extremely loud signal with well-preserved
transients. When we hear the signal begin to
distort, we can back the gain off a little. If the
signal isn’t loud enough for our liking, we can
listen to which parts are distorting (say, the sub-
bass when the kick plays) to work out which of
the elements in the mix needs to be changed so
that we can turn the gain up a little more. Of
course, there’s always going to be a limit to how
loud a track can go, but this method gives us the
best perspective on creating a mix that can be
pushed as hard as possible without making the
listener’s ears bleed.
“If the dynamic range
has been reduced too
much, the mix will lack
punch even though
it’s loud”
Once the mix is finished, you can export a
loud, clipped and dithered version for your own
use, or get rid of the clipper and gain boost and
export an undithered version that peaks the
master channel at -6dB or so for submission to a
mastering engineer. This is a solution that
satisfies everyone: mastering engineers get a
version with plenty of dynamic range that they
can work with, and the artist can create a loud
DIY master for DJing.
With this method, it’s up to the artist to
ensure that their clipped version isn’t too
audibly distorted and that it still retains a decent
level of dynamic range. If the dynamic range has
been reduced too much, the mix will lack punch
even though it’s loud. When played in a club the
mix will just sound loud in an unpleasant way, or
the DJ will end up turning it down to a level
where its perceived volume is the same as that
of the preceding track, reducing the punch even
further! Clearly, this isn’t how you want your
track to sound on its dancefloor debut.
Peak practice
Another digital recording phenomenon to be
aware of is inter-sample peaking, which we also
looked at on p29. As discussed, signals can
generate inter-sample peaks, and how these
sound depends on the kit used to play them
back, potentially affecting the mix when it’s
converted to a compressed audio format.
Severe limiting and clipping can make these
even more likely to occur, but there is a way
around the problem.
Some limiters, such as FabFilter Pro-L and
iZotope Ozone 5, have inter-sample peak
detection algorithms built in, but if yours
doesn’t, download SSL’s freeware X-ISM inter-
sample peak detector (www.solid-state-logic.
com). This clever metering software simulates a
digital-to-analogue converter with
oversampling, and has indicators that light up
when the audio signal appears likely to cause
problems. So, if you want to be sure that your
clipped signal isn’t in danger of containing inter-
sample peaks, place it after the clipper. If the
signal appears to be at risk, turn down the pre-
clipper gain or the clipper’s output itself to
hopefully remedy the situation.
That said, it’s not worth getting too worked
up over inter-sample peaks, as lots of
commercially released music appears to contain
them when run through X-ISM anyway. In
summary, if you’re absolutely hell-bent on
maximising the playback integrity of your music
then they’re to be avoided, which can be
achieved by exporting with a ceiling of -0.3dB
rather than 0dBFS. However, if you’ve got
decent monitoring and aren’t getting any
audible distortion when you convert your music
to MP3 or another compressed format, you
should be fine.
Hey, Mister DJ
If you’re interested in producing mixdowns that
work well in a club environment, it’s likely that
you’ll either want to play your tracks out
yourself or send them to DJs in the hope that
they might be dropped into a set. There’s more

COMPUTER MUSIC SPECIAL / 35

Life in mono
Practically all modern music is
delivered as a stereo mix, which
means we’re working with two
channels of information rather
than just one. This makes our
lives a fair bit more complicated
than they’d have been back in the
days of mono, but it also gives us
the opportunity to get a lot more
sound into our mixes.
If you listen to a mix from the
dawn of the stereo era, you’ll
hear some rather extreme
panning of the sort that doesn’t
happen too often today: drum
kits panned hard right, vocals
panned hard left and so on. This
was fashionable at the time
because it really showed off the
capabilities of the new
technology, but obviously it
doesn’t give you the most
powerful sound possible.
The current trend in mixing is
to have all the solid, weighty
elements of the mix (that is, the
ones with the most bass)
narrowly occupying the central
mid channel of the mix, with the
‘floatier’ elements (pads, reverb
and so on) largely occupying the
sides signal. This works well in a
club environment, where often
the signal from the DJ’s mixing
desk is summed to mono before
it reaches the speakers. While the
stereo information is lost in this
situation, the main beef of the
track – such as the kick, snare and
bass – will remain intact. Use Voxengo’s free MSED plugin to convert stereo elements to mono – simply
You can make stereo elements insert it into their mixer channel and turn down the sides mix
mono by inserting a mid/side
processor such as Voxengo’s flanger or reverb, or by delaying turning down the side signal.
excellent free MSED (www. either the left or right channel While it’s OK for less important
voxengo.com) on their channel, slightly – the Haas technique. elements to dip in volume or
and turning down the sides mix, This last effect sounds great in change timbre, you don’t really
leaving you with just the mid a stereo mix, but can cause issues want anything to disappear. If it
signal. Making mono parts stereo when the signal is summed to does, you should consider
can be done by adding stereo mono. You can check the mono adjusting the stereo processing
effects such as chorus, phaser, compatibility of your music by on the affected parts.

to prospectively sending music to labels and DJs
than you might think, though, so let’s run
through a few rules of thumb for those who are
new to the game.
First of all, you need to make sure that the
music you intend to send is of a high enough
standard to be played out. Determining this can
be really tricky if you’re a newcomer to
production – some people underestimate the
quality of their work, while others drastically
overestimate it. What’s more, feedback from
friends and members of online communities can
be misleading. It’s helpful to find people who
offer useful and impartial feedback, but
ultimately, being able to objectively judge the
quality of your own output is one of the most
important skills you can learn as a producer.
Compare your work with a wide range of
releases in the same genre by A/B-ing with them
and mixing them together in a DJing application.
Ideally, you should get to know a local
promoter or member of staff at your local club –
you might be able to persuade them to let you
test your tracks on the club’s system before they
open, which can be extremely informative! Bear
in mind that if you listen to your music on the
monitor in a DJ booth, it’s likely to have a limited
frequency response in the low end. And, again,
take some commercial tracks along to A/B your
tunes with in a club environment.
Remember that sending DJs poor-quality
music is only going to put them off listening to
your material in the future, so even if you’re an
experienced producer, try to avoid sending
music you’ve just rendered out: your ears will
probably be fatigued, so give yourself a few
days to forget about the project, then go take
another listen and make sure that it sounds as
good as it did when you finished it!
A good way to ensure your music is
DJ-friendly is to follow the conventions of the
genre. If everyone else in your genre is making
tracks that are eight minutes long with an easy-
to-mix intro, it’s sensible to go with the trend lest
your tracks give the DJs a headache. You can
innovate and break the rules when you’re
established! See What DJs Want on p38 for a
guide to making dance music that’s easy and
fun to mix.
Get connected
Once you feel your music is of a high enough
standard, you should get the contact details for
DJs who play out the particular type of music
you make. Realistically, unless the DJ is already
turned on to the kind of music you produce,
they’re not going to have time to give your track
a proper listen anyway, so don’t waste anyone’s
time by sending them music they won’t have
any interest in. The best DJs to target aren’t
necessarily the big names, but smaller ones with
a following and a particular sound, and who are
ideally making music that you respect yourself.
If you already have a connection on a musical
level with a DJ, you’re much more likely to pique
their interest with your tracks.
Many DJs will have a SoundCloud or other
social networking page that will specify how
they like to receive new music. The last thing
you want to do is antagonise your prospective
audience, so follow these rules carefully.
Behaviour that upsets DJs – for example,
emailing them uncompressed WAV files – is
likely to prejudice them against your tunes!
Usually DJs will want a loud version of your
track to play out, so send them your DIY master,
making sure there aren’t any long stretches of
silence or other garbage at the start and/or end
of the file. Label it with your contact email
address, the name of the track, the date you
finished it, and information on whether it’s
signed to a label or not – this latter point is
particularly useful for DJs looking for material to
sign to their own label, and can save a time-
consuming conversation when they contact you
and attempt to sign a tune you’ve already
promised to someone else. So, for example, a
sensible file name for a tune would be:
djwicked@wicked.com-My_Wicked_Tune-29-10-
13-unsigned.mp3. This instantly gives the DJ all
the required information in an easy-to-digest
format, and if they need more they’ve got your
contact details.
Social networking
Most DJs like to receive tracks as maximum
quality (320kbps) MP3 files, so if you can’t find
out specifically how a DJ likes to receive music,
get a Dropbox account, upload your 320kbps
MP3s to a folder, and send a polite, personalised
message to their email or social network
mailbox. Tell them you think that your music is
of a style they might be interested in and that
you’d appreciate any feedback if they have time.
Some DJs are approachable via Twitter, so if
you can’t find a way to send them music, fire
them off a polite tweet to test the water. Other
DJs will have an AIM (www.aim.com) account
logged on constantly in order to receive new
music, so look out for AIM buddy names on their
social media pages, or try to find them out
through word of mouth. Above all, be polite and
respectful! Developing a thick skin is sensible –
you’ll have to learn to deal with rejection and
blunt criticism, and bug people that you don’t
know – but this really is the best way to get your
foot in the door of most dance music genres,
especially if there isn’t a scene in your area.
As well as being a route to getting exposure
for your music, sending music to DJs is also an
important way to get feedback on your
productions. It’s reasonable to send a polite
request for feedback with your music, but don’t
be offended if you don’t hear back, or if what
you hear is negative. If you think the criticisms
are valid, try to address them in your next
productions – you’ll get more useful feedback
from DJs if your productions are of a half-decent
standard to begin with.

36 / COMPUTER MUSIC SPECIAL

 Stunna can be heard every Wednesday at
www.bassdrive.com. Check out his
music at http://soundcloud.com/stunna

Tony Blitz’s Local Talk EP is out now. Hear

it at http://soundcloud.com/djtonyblitz
DJs today have more music to pick from than
ever before. Thousands – yes, thousands! – of
new tracks are released every week,
thousands more are given away for free on
blogs and via SoundCloud, and popular DJs
will receive dozens or even hundreds via
their email and instant messaging accounts
alone. Consequently, your music has to stand
out if it’s going to be noticed amongst the
competition – but creating an innovative,
unpredictable arrangement isn’t the way to a
DJ’s heart.
For better or worse, the reality is that the vast
majority of dance music genres very strictly
adhere to the conventions that have developed
over the years in terms of how they should be
structured. Subverting these conventions in
inventive ways will impress much more than
simply disregarding them out of hand, so it
makes sense to understand what they are and
why they’re important.
If you don’t have any experience of DJing
yourself, it’ll certainly be harder to appreciate
the things that make a track DJ-friendly or a
chore to incorporate into a set – but not
impossible. Let’s begin, then, by examining how
DJs actually go about mixing a track.
If you want to get your track played in the
clubs, it has to be DJ-friendy. Let’s look at the
unwritten rules of arranging a dance
track to give it decks appeal
The traditional way
to ‘beat-match’ two tracks is to get them playing
at exactly the same tempo, synchronised so that
the first beat of the bar on each lands at exactly
the same point. If this is done correctly (and
depending on the source material, as we’ll see
later), the DJ can raise the volume of one track
while lowering the volume of the other, creating
a seamless transition between the two pieces,
facilitating a continuous stream of music and
allowing the audience to keep dancing
indefinitely without a break in the music.
Give the DJ a break
Although most modern DJing software can
automatically identify the tempo of a track and
locate the first downbeat – enabling the DJ to
automatically synchronise tracks – DJing with
vinyl and turntables requires the DJ to locate the
downbeat and correct the tempo themselves.
This is why much dance music begins with
what’s often described as a ‘DJ-friendly’ intro,
with obvious clues as to the tempo of the track
and the location of the downbeat.
While it’s possible to do this subtly using
implied rhythm, that isn’t much use to a DJ
because they need to be able to accurately hear
on their cue channel that the track they’re going
to drop into the mix is indeed synchronised
correctly with the track that’s currently playing.
This is much easier with shorter, percussive
sounds like drums than it is with pads. An intro
comprising just a drumbeat, therefore, is always
going to be a gift for a DJ because it makes the
tempo of the track extremely easy to discern
and enables them to bring the track in without
having to worry too much about clashing
musical elements. Even if the two tracks are in
the same key or sympathetic keys, there can
always be musical parts that just don’t sound
good when played together. By bringing these
in later, you give the DJ time to create a
smoother segue between the two pieces of
music – this is part of the reason why dance
tracks don’t start on the ‘drop’, the point where
the full-on beats and bassline kick in.
The DJ needs to be able to identify the
downbeat, too, and while this will be obvious if
the track begins with drums, it helps to drop
clues throughout the track, in case the DJ starts
from a point in the track other than the
beginning. This is part of the reason why so
much dance music has a cymbal crash every
four, eight or 16 bars – not only do these
‘episodic markers’ sound good and keep the
listener grounded in the progression of the
track, they also help the DJ identify the
beginning of each section. See Deep Structure,
overleaf, for more on this.
Swing time
As well as the issue of tracks being musically
incompatible – not being in compatible keys, for
example – it’s also possible for tracks that are
synchronised correctly and musically
compatible to sound bad when mixed. This is
usually down to the two tracks having subtle
timing differences.
Known as shuffle, swing or groove, having
every other eighth- or 16th-note play slightly
after the beat gives a track a funky, human feel.
This is an integral part of practically all pop
music, from blues all the way up to house and
hip-hop. Swing only becomes a problem when
DJs try to mix two tracks with different grooves
that sound bad together. Sync buttons won’t
help in this scenario – the tracks can be mixed
perfectly, but thanks to their clashing grooves
they’ll always sound messy together. The
solution to this is to lead in with elements that
give the DJ the feel of the swing without fully
committing to it – bringing in swung 16th-note
hi-hats during the intro, for example. This allows
the DJ to find out in advance if the track isn’t
going to work when beat-matched with the
currently playing tune.
It can also be useful to feature a sound that
occurs later in the track in the intro – an easily
identifiable sound effect, melodic hook or vocal
sample, say. This doesn’t actually help the DJ
with the technicalities of mixing, but it does let
COMPUTER MUSIC SPECIAL / 39
Keep them
guessing
There’s a fine line between an
inventive arrangement that keeps
the audience’s interest and a by-the-
numbers production that comes
across like an exercise in box-
ticking. Ultimately, we want to stick
to the fundamental conventions of
our chosen genre, but we also need
to give our music some originality
and personality, so that it’s exciting
to listen to.
The most effective way to fulfill
both of these requirements is to
play with the rules of arrangement.
A common trick in bass music
genres is to use what’s sometimes
called a ‘switch-up’, where at the
end of each, say, 16-bar section the
main musical idea is changed in
some way, typically by altering the
rhythm and melody of the bassline,
or introducing a new hook sound.
This will often happen at the same
time as a rhythmic change in the
beats, and such changes make it all
the more satisfying when the
original version of the beat and
bassline come back in. A good
example of this can be heard in Bad
Company’s Planet Dust (http://bit.
ly/NwX3Q5).
A variation on this trick is to use a
false breakdown or switch-up at the
start of a section rather than the
end. This gives the listener the
impression that the music has
moved on to a different section,
then subverts their expectations
when the familiar pattern returns.
This technique is used to good
effect in Rusko’s Cockney Thug
(http://bit.ly/3XXKpt), where the
switch-ups occur at the start rather
than end of sections, and more
radically in John B’s Rinse It Out
Propa (http://bit.ly/19IoPlV), in
which the drop itself plays for a
mere two bars before changing up
to a different bass and beat!
Another way to surprise the
listener is to do the opposite of what
the cues in the music suggest is
going to happen. For example,
Camo & Krooked’s All Night (http://
bit.ly/19kxksA) features the typical
breakdown and build-up you’d
expect from a DnB track, but the
drop isn’t the usual cue for all hell to
break loose – instead, it’s actually a
rather minimal affair that radically
changes the feel of the track.
them get more creative with the mix by ‘teasing’
your track, cutting in brief previews as another
track plays. This adds atmosphere to the mix
and makes the listener feel like the DJ is taking
them on a musical journey.
The drop
The drop is of the utmost importance in the vast
majority of contemporary dance music styles.
Most will feature a section in which the track
switches from being in its intro or build-up
phase to firing on most if not all cylinders. In
many styles this is preceded by a breakdown – a
section where the beats drop out, often to make
way for a ‘musical’ section and also providing a
contrast to when the beats and bass kick in.
The breakdown/drop structure is common in
many contemporary styles of dance music,
particularly house, dubstep, DnB and pretty
much all forms of bass music. A track in any of
these genres will feature at least one
breakdown, with two being pretty much the
norm. These give the audience a breather from
the generally uptempo vibe, and the producer
an opportunity to make a second drop that
potentially varies from the first. It’s also
common practice to vary the progression of the
track by adding and removing elements, or
switching into ‘middle eight’-type sections
where an alternative set of musical ideas is
temporarily introduced.
DJ-friendly outros work much like DJ-friendly
intros in reverse. Keeping a solid beat going
helps the DJ bring in the next track gradually if
they want to, and the exiting one can be wound
down by removing elements gradually.
Deep structure
Most dance music tracks are divided up into
sections that are eight, 16 or 32 bars long.
This might sound formulaic, but the human
brain finds this regularity satisfying. It also
makes it much easier for the DJ to make
tracks work together. An extremely basic
example of this would be that if the DJ starts
the beginning of a track’s 32-bar intro at the
same time as another track’s 32-bar outro,
the first track will drop as the second ends. As
such, it makes sense to stick to the template,
because it allows the DJ to use your music
more effectively as a dynamic part of a larger
40 / COMPUTER MUSIC SPECIAL
Sometimes a bassline can continue until the
track has finished (DJs can fade this out using
EQ to make room for the low end of the next
track), but many tracks feature a purely beats-
only outro that’s designed to make mixing out
as easy as possible.
Experience preferred
Of course, dance music would be pretty boring if
it all stuck to exactly the same structure, which
is why you need to put thought and effort into
your arrangements rather than recycling the
same old formula. While not all dance music
“The drop is of the
utmost importance in
the vast majority of
contemporary dance
music styles”
follows these rules absolutely, the vast majority
of it is tailored so that even if it’s not designed
just to be easy to mix – an epic intro, say – it’ll
feature subtler tempo cues such as hi-hats, and
sections that are particularly conducive to
blending into or out of another track.
“If the arrangement is ‘bad’ or difficult for a DJ
to follow when trying to mix the track with other
tunes, either the DJ will try and figure out where
work. Admittedly, there are some examples
of (usually older) dance music that add in the
occasional extra bar or even extra beat, but
these are definitely the exception to the rule.
“There’s nothing worse than having a
great track that kicks in off-beat and messes
up the mix,” says Tony Blitz from Welsh
garage heros C.R.S.T. “We’ve all been there,
mixing a track for the first time and it
dropping too early while you’re mixing, then
cringing whilst frantically trying to sort it out,
hoping that nobody noticed! I like tracks that
have a long intro with plenty of percussion,
the track is actually ‘mixable’ at specific points,
or they’ll completely bypass playing the tune,”
says Chicago DJ and producer Stunna. “I always
like the challenge when someone tells me a
track is ‘unmixable’, but there are definitely
some tunes that need to go back to the drawing
board and be worked on further. I structure my
tracks by keeping in mind the ‘formula’ so that
other DJs will be able to mix them with other
tunes if desired. Some curveballs in
arrangement are often thrown in for variety and
originality, but as long as a DJ is able to
recognise where major sections in the track
begin and end, my job is mostly done.”
You can get a better understanding of how
track structure works in the context of a mix by
listening to DJ sets and hearing where and how
they use EQ and volume manipulation to swap
between tracks, as well as by studying unmixed
tracks in the styles of music you produce to hear
how they’re structured.
To fully appreciate how the subtleties of a
track’s structure can work for or against the DJ,
it’s necessary to gain some experience DJing
yourself. Getting into DJing can be a hefty
investment in terms of both time and money,
but if you just want to get a taste, download the
free DJing application Mixxx from www.mixxx.
org. This open-source freeware DJing tool is
compatible with Windows, Mac and Linux, and
works along the same lines as professional
software like Native Instruments’ Traktor. Even if
you’re just mixing a relatively small selection of
tracks in the style in which you produce, you’ll
quickly gain valuable insight into what makes
music easy or difficult to work with.
and a nice long break halfway in to allow the
DJ to get the next track ready to mix. If I’m
sent a demo and the intro is a bit too ‘out
there’, and I find it the slightest bit difficult to
identify which part of the bar a track is at, I’ll
most probably not risk mixing it out. I spend
most of my time making music, not listening
to demos, so it needs to grab me straight
away, and it needs to be DJ-friendly.
Occasionally I’ll be sent something that’s a bit
all over the place, but I’ll play it because it’s
too good to ignore. But those times are few
and far between.”
A classic arrangement: Double 99, Ripgroove
3 4 5
1 9
6
7 12
Double 99’s 1997 track Ripgroove is a bass
music classic that helped define the speed
garage template: uptempo swung house
beats, looped R&B vocal samples (from Tina
Moore’s Never Let You Go), timestretched
reggae soundbites (from Top Cat’s Request
the Style) and ‘reversed’ bass sounds inspired
by jungle classics such as Dead Dred’s Dred
Bass. The arrangement serves as a perfect
example of a simple but effective DJ-friendly
structure that makes the most of all the
elements in the track and, more importantly
for our purposes, facilitates easy, fun mixing.
The track begins with a DJ’s delight: a
straightforward, solid, drums-only intro in
which it’s easy to hear the shuffle in the
hi-hats 1 . From the very start, there’s a vocal
sample or fill every eight bars, giving the DJ
something with which to tease the track right
from the off 2 . After the first 32-bar section, a
rhythmic, one-note stab sample is brought in
3 , introducing the key of the track and
increasing the tension before the breakdown
at bar 49.
The 16-bar breakdown can be thought of
as two eight-bar sections: an atmospheric
8 10
11
part, where we hear the pad and the R&B
vocal begins to fade in 4 , and a build-up,
where the vocal continues to fade in, along
with a kick drum roll and a reggae vocal 5 .
Just before the drop, everything apart from
the R&B vocal cuts out and a spinback sample
plays 6 , clearing the low end for a moment to
give the beats and bass extra impact when
they drop 7 .
Each eight-bar section after the drop
changes things up slightly, adding more
reggae soundbites, for example 8 , and
there’s a switch-up at the end of each 16-bar
section 9 . In fact, the only remotely
unpredictable thing is that this drop and roll-
out don’t last for a multiple of 16 bars,
arriving at another breakdown after just 40
bars 10 . This actually makes it fairly difficult
for the DJ to mix out of the track before the
second breakdown, but – on the plus side – it
makes it more likely that the DJ will end up
playing both drops!
The second breakdown and drop offer a
minor variation on the first – the bass is
teased in the second breakdown 11 – and we
get another 32 bars of drop and roll-out 12
13 14
before we reach the outro 13 , which consists
of another 16 bars of beats and stabs,
followed by 24 bars of just beats and a final
soundbite at the end of the track to round
things off 14 .
What’s perhaps surprising about
Ripgroove’s structure is that the intro and
outro add up to a total of a whopping 88 bars,
compared to only 72 bars of the beats and
bassline playing! This demonstrates that the
track comes from an era in which tunes were
aimed first and foremost at DJs – we have to
wait a minute and a half until the breakdown,
and we only get to hear the bassline after a
total of two minutes. By way of comparison,
most contemporary tear-out dubstep tracks
would drop after 32 bars (about 55 seconds at
140bpm). So, while Ripgroove’s extended
intro and outro mean that it isn’t perhaps the
most immediately YouTube-friendly track,
it’s worth remembering that thanks to its
elegant production and structure, it’s still a
firm favourite with DJs more than 15 years
after it was released.
Web http://bit.ly/15ten5E
COMPUTER MUSIC SPECIAL / 41
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 Correctly tuning and tightening your drums will
get your club-bound mix off to a flying start
As we saw in Mixing and Mastering for samples so that they sit together perfectly and
Club Playback on p32, the kick and snare provide a solid jumping-off point for your
drums in a track not only help provide its project. We’ll pay particular attention to the
fundamental frequency and weight but are frequency content of our drum sounds, as it’s
typically the most dynamic sounds in the mix, the movement of the kick and snare through the
and therefore define its rhythm too. Along frequency spectrum that gives them their
with the bassline, they’re the most important impact. Tuning drum samples is a process that’s
sounds in the mix by far, and are usually the questioned by some producers, who regard it as
base for everything else in the track. This a waste of time. We believe that nothing could
makes getting them right from the very start be further from the truth, however – by tuning
extremely important – do a sub-par job here our drum sounds we can quickly and easily
and you’ll waste a huge amount of time make them sit correctly together, and avoid
making tweaks and adjustments to solve hours spent searching for the magical
problems that could have been avoided. combination of samples that ‘just work’.
In this tutorial we’ll show you how to select, We’ll also show you how to adjust amplitude
tune, edit and process your kick and snare envelope settings to get just the right balance or
loudness and dynamic range without having to
resorting to dynamics processing, and prove
that something as simple as adjusting the start
and end points of your samples can dramatically
alter their sounds.
Finally, we’ll show you that if your beats have
been sensibly prepared as we’ve suggested, it’s
possible to boost them up to the level of a
current, commercial dance track with ease.
Accompanying this tutorial is a library of
versatile one-shot drum samples (in the
Samples folder), created by none other than
Welsh future garage legends CRST. Combined
with the guidance in this tutorial, this should
constitute enough material for dozens of
banging beats!
COMPUTER MUSIC SPECIAL / 43
> Step by step Selecting and tightening up a kick drum
TUTORIAL
FILES
1 2
Start by copying the CRST Drum Hits folder from the Samples
folder onto your hard drive. We’re going to use Ableton Live for
this tutorial because its straightforward built-in sampler and browser
make it quick and easy for this kind of beat-building, but you can of
course follow along with any DAW or sampler you like.
3 4
Choosing the right drum sounds is vitally important to creating a
successful dance music mix, and if you pick inappropriate ones,
you’re going to make life difficult for yourself from the off – see Natural
Selection, p48, for more. Kick Basix.wav has a very natural-sounding
balance of attack and body, and a fairly neutral character.
5 6
Put Voxengo SPAN (a free download from www.voxengo.com) on
your master buss Now when we play the kick back, we can see
where its low end peaks. This gives us an accurate impression of
what’s going on in the low end, which is especially helpful when
working with less than ideal monitoring. Use the horizontal scrollbar to
zoom in on the low end of the frequency spectrum.
44 / COMPUTER MUSIC SPECIAL
Load up Live. Leave the tempo at the default 120bpm and draw in
a standard four-to-the-floor kick drum pattern. In Live’s browser,
open the CRST Drum Hits/CRST Kicks folder, and click the first
sample, Kick 2 Real.wav, to audition it. To cycle through the rest of the
kicks, press the down arrow.
This makes an ideal kick for creating a standard dance beat that
can be customised to a specific style later. Select a MIDI track and
drag Kick Basix.wav onto its device chain. A Simpler instrument will
automatically be created. We know how this kick sounds already, but
let’s take a closer look at it.
If you move your cursor over the spectral analyser’s graphical
display, you’ll see that the frequency in Hz and key at its position
are displayed along top edge of the window. Play the kick back on C3
(its original pitch) and move the cursor over where it peaks – this is
round about E2, or 82Hz. This information will help us make informed
mixing decisions as we go forward.
7 8
Create a one-bar MIDI part on the MIDI track and sequence an
eighth-note C3 on each beat. The length of the note is extremely
important, as we’ll soon see. Set Live to loop around the MIDI part, and
play it back with SPAN’s interface visible. Click the MIDI track again so
you can see Simpler’s interface too. (Audio: Original kick loop.wav)
9 10
Because we’re now missing part of the high-frequency content at
the start of the kick, we have a different sound – still solid, but with
a less aggressive character. Move the start point to the start of the next
cycle, as shown above, and you’ll get a more muted sound still. (Audio:
Muted kick loop.wav)
11
You’ll notice that the sample still peaks at around 80Hz until the
end point reaches almost the very start of the sample. Practically
all the energy and character of the kick is created by the first few
cycles, with the rest of the sample providing the weight and body. We
can get more detailed control over the tail using Simpler’s ADSR
volume envelope.
A kick drum sound moves from high to low frequency quickly, so
we can dramatically affect the frequency content of the sound
simply by changing the sample start or end points. Zoom in on the
start of the kick drum waveform and drag the start point over to the
mid-point of the first cycle, as shown. (Audio: Shifted kick loop.wav)
A drum’s solidity is dependent on a consistent volume level as it
moves from high pitch to low. EQ can detract from this, so moving
sample start and end points to alter the sound is a much more elegant
solution. Return the start point to the beginning of the sample and
drag the end point to the left as you play back the loop.
12
Move the end point back to the end of the sample and set the
Sustain to -124dB. Turn the Decay knob down and the kick will
fade out rather than simply stopping dead. This gives us a smoother
sound. Set the Decay to 1.3s for a snappier feel. We can demonstrate
the power of the volume envelope even more clearly with other drum
and percussion sounds – such as a snare, which we’ll look at in the next
walkthrough. (Audio: Decay kick.wav)
COMPUTER MUSIC SPECIAL / 45
 > Step by step 3. EQing beats with spectral analysis in Logic

1 2
When you’re dealing with a sound that has extreme frequency
spikes – as some drum loops and breaks do – spectral analysis can
help you locate their positions. Create an audio track in Logic and drag
Funky jungle.wav onto it. Set Logic’s tempo to 140bpm.
Click the Audio FX button on the audio track in the Inspector, and
select EQ»Channel EQ»Stereo. Click the Analyser button on the
left-hand side of the Channel EQ’s interface, and when you play the
project back you’ll see the audio signal represented visually behind
the EQ curve.

3 4
Channel EQ’s analyser isn’t particularly configurable, so let’s use
something more suited to the task. Add Voxengo SPAN (free
download from www.voxengo.com) after the Channel EQ, and use the
horizontal bar at the bottom of its frequency display to zoom in on the
high end of the frequency spectrum.
Using SPAN, we can see that the peaks occur at 9.4kHz and
12.2kHz. Return to Channel EQ and set two band EQs with Qs of
2.00 to take off 10dB at 9400Hz and 12,200Hz. This tames the peaks,
giving you more headroom to turn the channel up and make it louder
in the mix. (Audio: EQed beat.wav)

Four exceptional equalisers

DDMF IIEQPRO $39
The commercial version of DDMF’s
IIEQPro includes all the great features
of the CM Plugins version (such as
serial and parallel routing options and
19 filter modes, including several
Butterworth filters) plus more bands
and a built-in spectral analyser. It’s
also very reasonably priced.
www.ddmf.eu
VOXENGO GLISSEQ $100
From the maker of SPAN comes this
incredible equaliser that not only has
the same powerful spectral analysis
capabilities as SPAN, but also boasts
program-dependent band response.
This dynamic approach to EQ makes
GlissEQ ideal for boosting highs and
lows without making them sound
harsh or muddy.
www.voxengo.com
FABFILTER PRO-Q £124
One of the best all-round EQs on the
market, Pro-Q is overflowing with
awesome features. With up to 24 filter
bands, zero-latency and linear phase
modes, intelligent solo mode for
auditioning single bands, and undo/
redo and A/B comparison functions,
it’s a real equalisation powerhouse.
www.fabfilter.com
WAVES REDD $200
Based on the EQ sections of Abbey
Road Studios’ legendary mixing
consoles, Redd delivers that classic
60s sound. As well as the simple EQ
controls themselves, you also get
Drive and Analogue knobs for dialling
in the perfect level of vintage warmth.
www.waves.com

COMPUTER MUSIC SPECIAL / 75

> Step by step 4. Parallel processing in Live
1 2
Parallel processing is a technique whereby a processed (for
example, compressed) signal is mixed with the unprocessed
version of itself. In the case of compression, this would give us a
balance of loud ‘wet’ signal and punchy ‘dry’ signal. Start by dragging
Funky jungle.wav onto an audio track in Ableton Live.
3 4
Set the Glue Compressor’s Threshold to -27dB, Makeup to 7dB,
and Attack to 0.3. Add a Saturator after it, with its mode set to
Sinoid Fold and its Drive turned up to 11.2dB. The processing we’ve
added has boosted the peak level by a few dB, which we can check by
bypassing the effect.
5 6
This puts both effects in an Audio Effects Rack. Click the Show/
Hide Chain List button and a list of chains in the Audio Effects
Rack appears. Currently, there’s only one chain in the Rack. To add
another, right-click below it and select Create Chain.
76 / COMPUTER MUSIC SPECIAL
Drag Glue Compressor onto the track. The effect has a Dry/Wet
knob, which means that we can mix some of the dry signal back in
direct from the plugin itself. However, by setting up a parallel
processing chain we can use multiple effects, which can then be mixed
with the dry signal at the end.
To compensate for this, turn the Saturator’s Output down to -2dB.
Now the signal peaks at around the same level as the unprocessed
signal did, although the dynamic range has been significantly reduced,
so it hits the peak level much more often. Now let’s create our parallel
routing. Select both effects, right-click and select Group.
Now we’ve got our ‘wet’ chain with effects on it, and a new ‘dry’
chain for the unprocessed signal. As we’ve got two versions of the
signal, the output of the track will be very loud, so turn both chains
down to -6dB to halve their volume. You can use the Chain Volume
parameters to balance the signals, and toggle the Audio Effects Rack’s
Device Activator button to compare the parallel processed signal
with the unprocessed version. (Audio: Parallel processed beat.wav)
> Step by step 5. Tightening drum loops with Live’s Slice to MIDI function

1 2
Using drum loops or breaks is a convenient way to give your beats
extra rhythm and character, but if they’re too loose or have too
much reverb, they’ll need tightening up. Doing this manually can be
fiddly, so some DAWs have their own beat-slicing capabilities to make
it easier. Drag Dub beat.wav onto an audio track in Live.
Live will automatically set the project tempo to 140bpm. Select
the clip and press Cmd/Ctrl+L to set loop points around it. This
loop is drenched in dubby reverb, but we can quickly get rid of that
and tweak the loop in various interesting ways by slicing it to MIDI.

3 4
Right-click the clip and select Slice to New MIDI Track. A window
appears, asking how you want it to slice the loop and which preset
to use for the resulting instrument. The default settings of slice by
Transient and Built-in slicing preset will be fine for our purposes. Click
OK to slice the loop.
A new MIDI track with an Instrument Rack loaded appears below
the audio track. Mute the audio track. Now we can tighten up the
new version of the beat on the MIDI track by turning its Sustain down
to -infdB. Turn the Decay down to 362ms for a super-tight version of
the beat. (Audio: Tight dub.wav)

Make your own slicing presets

Live’s built-in slicing preset is convenient
because it enables quick access to the ADSR
envelope controls of each slice, giving you an
easy way to control a beat’s dynamics. The
problem with it, though, is that it doesn’t
have macro controls linked to each slice’s
Transpose and Detune parameters, meaning
that if you want to retune the loop you have
to go through and adjust every slice
manually! Clearly, this isn’t a very convenient
solution, but thankfully it’s possible to create
your own slicing presets with which you can
set up eight macros as you see fit. That’s
enough to control the ADSR and both tuning
parameters, with two left over for other
functions – volume and velocity sensitivity,
for example.
To make your own preset, create a new
MIDI track and put a Drum Rack on it. Put a
Simpler in the C1 slot, click the Show/Hide
Macro Controls button, then the Map button
on the device title bar to enter mapping
mode. You can now assign macros by clicking
a parameter, then clicking the Map button on
a macro. Note that when you do this, the
value on the parameter will reset to 0. When
you’ve finished, exit map mode by clicking
the Map button on the device title bar again,
and set each macro to the default value you
want it to have.
Finally, drag the Drum Rack into the User
Library/Defaults/Slicing folder in Live’s
browser. This creates a preset, which will be
highlighted, prompting you to enter a name
for it. Type in a name, press Enter, and boom –
you’re done! Now, when you use Live’s Slice
to New MIDI Track function, you’ll find the
preset that you just created in the list of
Slicing Presets. Our Tuning Slicer preset can
be found in the Tutorial Files folder – drop it
into the folder in Live’s browser to add it to
the Library.

COMPUTER MUSIC SPECIAL / 77

> Step by step 6. Cutting or enhancing reverb with Logic’s Enveloper
1 2
Apple’s Logic Pro includes a unique and really quite powerful
dynamics processing tool in the form of Enveloper. This can be
used not only to tighten up the reverb on drum loops, but also to make
them sound louder and boomier – ideal if you’re after a big, ‘live’ drum
sound. Start by dragging Dub beat.wav onto an audio track and
setting the project tempo to 140bpm.
3 4
This immediately gets rid of the reverb and gives us a really tight,
dead drum sound. You can get different curve shapes by using the
Time knob to the left of the Gain fader. Set the Time to 60.00ms and
the Gain to -74% for a less severe reverb reduction effect. (Audio:
Release curve.wav)
5 6
We can use Enveloper to treat the attack stage of each hit as well
as the release stage. First, return the Gain fader on the right-hand
side of the interface to 0%, then gradually turn the Gain fader on the
left up as you play the beat back. Once you reach the 60% level you’ll
really start to hear each hit become more pronounced. (Audio:
Transient enhancement.wav)
78 / COMPUTER MUSIC SPECIAL
In the Inspector, click the Audio FX slot on the audio track and
select Dynamics»Enveloper»Stereo. The plugin very effectively
detects the transients in the loop, and enables you to control the tail of
each hit using the Gain fader on the right-hand side of the interface.
For starters, turn it down to -100%. (Audio: Tight dub.wav)
Let’s use the effect to enhance the reverb now. Set the Time to
2000.00ms and turn the Gain up to 100%. As you can hear, this
has a really drastic effect on the sound. The boominess is so loud that
it’s going to clip the master, in fact, so set the Out Level to -9dB. (Audio:
Boomy reverb.wav)
It’s possible to dull the attack stage of the sound as well as enhance
it. Turn the Gain fader down to 100% and it’ll sound like you’ve
turned up the attack time of each slice. This is potentially useful when
you want to layer one-shot drums with a loop to give them character
but the transients of the loops clash with those of the one-shots.
(Audio: Dulled loop.wav)
> Step by step 7. High-shelving drum loops with parallel EQ
1 2
Using a pair of high-shelving EQs can be a very useful way to
shape the high and/or low end of drum loops. To demonstrate this,
we’re going to use DDMF’s IIEQPro CM, which you can find in the CM
Plugins folder. Create a new audio track in your DAW and drag Dub
beat.wav onto it.
3 4
Now click the Band 1 button and select Band 2. Set this band to
HighShelf as well, and this time move the node to -7dB at 100Hz.
The white line displays the overall effect of the two bands, revealing
that by using two high-shelf EQs we’ve created a curve that we couldn’t
have made using just one. (Audio: Serial high-shelves.wav)
5 6
As well as using the Frequency and Gain parameters, we can also
use Q to control the shape of the EQ – and IIEQPro CM’s Q can get
pretty extreme. For example, by tuning Band 2’s Q up to 3.00 we can
remove the low end of the reverb from the beat while simultaneously
boosting the kick. (Audio: Boosted Q.wav)
Insert IIEQPro CM onto the audio track. Click the Off button on the
bottom row to pick an EQ type. Select HighShelf and an orange
node appears at the left-hand side of the graphical display above. Drag
this to the +5dB position at 2000Hz.
Click the Serial button on the right to switch to IIEQPro CM’s
unusual Parallel mode. In this mode, each band works on a dry
version of the signal rather than one that’s already been processed.
This affects the overall sound of the EQ, illustrated by the change in the
shape of the white line. (Audio: Parallel high-shelves.wav)
Finally, let’s try out IIEQ Pro CM’s all-pass filter mode. In serial
mode this doesn’t do a whole lot, but in parallel mode it has a
quite dramatic effect. Select Band 3 and set it to APF. Now, when
you sweep its frequency up and down, you’ll see that the overall EQ
warps around its position – this is ideal for experimenting with
different frequency responses.
COMPUTER MUSIC SPECIAL / 79
 Analyser plugins are regarded with
Sometimes even the suspicion by some producers. “Mix with
your ears, not with your eyes!” they’ll advise.
best producers need Of course, visual aids are indeed no
substitute for your lugholes, but they can be
a helping hand, and helpful in many situations. Let’s not pretend
that our ears are perfect: what we hear
depends to a great extent on the
that’s where analyser characteristics of the room we’re in and the
monitors we’re using, and we’re not all born
plugins come in. Here with the innate ability to detect where that
rogue spike sits in the frequency spectrum.
are four ways in which Analysis tools are becoming a standard
feature in music production software across the
board. DAW mixers have always had level
they can make your meters, but now spectral analysers are finding
their place as a common feature of EQs, and a
life easier… few dynamics processors now include
histogram displays. These aren’t just bells and
whistles for the sake of it: the more production
takes place away from dedicated studio
environments, the more these features are
80 / COMPUTER MUSIC SPECIAL
necessary to ensure that we’re getting the best
mixdowns possible. The worse the studio
environment you’re working in, the more useful
these tools become.
If you’re working in a less-than-perfect
environment, it’s sensible to use as many
possible methods as you can to ensure your mix
sounds right. Along with referencing on
headphones, checking your mixes on as many
systems as possible and A/B-ing with other
tracks, analysis plugins ensure that you get an
accurate idea of what’s going on in your mix.
In these walkthroughs we’ll show you how to
use multiple inputs to reference two signals at
once on the same spectral analyser, use an
oscilloscope plugin to monitor your mixes’
dynamics, use spectral analysis and EQ to tweak
the harmonics of individual sounds, and
compare the peak, crest and RMS values of your
mix with those of a mastered track. Once you’ve
got to grips with these techniques, you should
find it a lot easier to create the perfect mix.
> Step by step 1. A/Bing dynamic properties with Blue Cat Audio DP Meter Pro in Cubase
TUTORIAL
FILES
1 2
In Mixing in the Digital Domain on p24, we looked at how peak,
RMS and crest values relate to each other. Let’s see how we can
use these values practically by comparing an unmastered track with a
commercial master. Load a new project in Cubase and create two
audio tracks. Name one ‘Project’ and the other ‘AB’.
3 4
Mute the AB track so that you can only hear the WIP track. Add
Ozone 5 (download the demo from www.izotope.com) to the
master buss. Activate the Maximizer and turn the Character fader
down to 0. We can now boost the level of the track by bringing down
the Threshold fader. Set it to -3dB for the moment.
5 6
Close the VST Connections window and select the AB track. Set its
output routing to Stereo Out 2. Now we can monitor the AB track
without it being processed through Ozone 5 on the master buss. To
solo a track, hold Cmd/Ctrl and click the track you want to listen to.
The other track will automatically be muted.
Drag Work in progress.wav onto the WIP track, and Mastered
track.wav onto the AB track. Work in progress.wav represents an
unfinished project. It hasn’t been limited at all, so it’s really very quiet
compared to Mastered track.wav, which represents a mastered track
that we want to compare our project with.
Now let’s examine the AB track. Open Devices»VST Connections
and click the Outputs tab, then hit the Add Bus button. In the Add
Output Bus window that appears, select a stereo buss and click Add
Bus. In the Outputs panel, make sure the new buss is routed to the
same physical outs as the main buss. The new buss should be named
Stereo Out 2 by default.
Now, add Blue Cat Audio DP Meter Pro (download the demo from
www.bluecataudio.com) to both the master buss after Ozone 5
and Stereo Out 2. You can now compare the Peak, RMS and Crest
values on your Project and AB tracks as you bring down Ozone 5’s
Threshold to bring up WIP’s volume. Remember not to reduce the
Crest value too much, or you’ll get a squashed, undynamic mix.
COMPUTER MUSIC SPECIAL / 81
> Step by step 2. Dual track monitoring with Voxengo SPAN in Ableton Live
1 2
Voxengo’s SPAN analyser (free from www.voxengo.com) gives
you the ability to view two audio inputs simultaneously, which can
be useful when attempting to mix two sounds in the same sort of
frequency range. We’re going to use Ableton Live to demonstrate this
because its audio routing is quick and easy, but you can use any DAW
you like. Start by creating three audio tracks.
3 4
We’re going to use Live’s return tracks to route the Kick and Bass
tracks to the plugin. We don’t need to use the track’s Audio From
routing, so set it to No Input – this ensures that your audio interface’s
physical inputs aren’t routed to the track.
5 6
SPAN has the same routing architecture as Voxengo’s other
plugins, which is pretty complicated, as you can see. Thankfully, we
don’t need to do anything too scary to get our two inputs up and
running. In the Input Routing section, set IN3 to C, IN4 to D, IN5 to E,
and IN6 to F.
82 / COMPUTER MUSIC SPECIAL
Name the first track ‘Kick’ and the second track ‘Bass’. Drag Kick.
wav onto the Kick track and Bass.wav onto the Bass track. As we
need to use SPAN’s multiple inputs capability, it’s easiest to put it on its
own track. Name the third track Analyser and put SPAN on it.
By default, both Live’s A and B return tracks have effects on them –
we don’t need these, so delete them. Next, we need to set SPAN up
to receive audio on its extra inputs. Open SPAN and click the Routing
button to invoke the plugin’s fearsome Channel Routing window.
Take a gander at the Group Assignments row and you’ll see that C
and D are routed to group 2, while E and F are routed to group 3.
This upshot of this is that the plugins’ 3/4 stereo input will be sent to
Group 2, and the 5/6 input to group 3. We’re done with the Channel
Routing window now, so close it by clicking the cross at the top right-
hand corner.
 7 8
Now we need to get the audio from our Kick and Bass tracks into
SPAN’s inputs. We’re using Return tracks to achieve this, so that the
routing doesn’t interfere with the tracks’ audio output, which we want
to keep routed directly to the master. Turn the level of the Kick track’s
Send A parameter up to 0dB.
Select the Return A track, and set its Output Type to the Analyser
track, and Output Channel to 3/4 – Voxengo Span. Now turn the
Bass track’s Send B level up to 0dB, and set the Return B Output Type
and Channel to Analyser and 5/6 – Voxengo SPAN.

9 10
We’ve just routed the Kick and Bass tracks to SPAN’s 3/4 and 5/6
inputs via the Return A and B tracks. Now, if you bring up SPAN’s
interface and click the Group 2 button, you should see the output of
the Kick track displayed visually.
Currently, Group 2 is set to the default colour. Click the colour
button above the spectral display and change it to light blue. To
see the Bass track, click the Group 3 button. Next, click the Underlay
button and select Group 2. Now we can view both sets of input at the
same time.

Four awesome analysers

MELDAPRODUCTION
MMULTIANALYZER €50
The commercial version of MAnalyzer
offers the same visually appealing
spectral analysis as the freeware
version, plus mid/side functionality,
8-channel input and an online preset
exchange. It’s available as part of
Melda Production’s MMixingBundle
and MTotalBundle packages, so check
them out if you’re after a bargain.
www.medlaproduction.com
BLUE CAT ANALYSIS PACK €229
Blue Cat Audio’s suite of analysis tools
includes DP Meter Pro and a wealth of
other goodies, including the
StereoScope plugins series,
FreqAnalyst Pro and FreqAnalyst
Multi. Be sure to give the
-exclusive FreqAnalyst CM a go,
the installers of which are located in
the CM Plugins folder.
www.bluecataudio.com
PHOTOSOUNDER SPIRAL $99
A revolutionary spectral analysis tool,
Photosounder’s Spiral uses an
innovative circular display to show
the octave and harmonic
relationships of the including audio
signal. This makes it ideal for
analysing and creating synth sounds.
You can find the exclusive version,
Spiral CM, in the CM Plugins folder.
www.photosounder.com
VOXENGO SPAN PLUS $50
We’ve used Voxengo’s stunning free
spectral analyser SPAN extensively
throughout this Special, and if
you’re smitten with its crystal-clear
graphical display and copious
options, this enhanced version might
be for you. With static spectrum
display and real-time spectrum
import and export, SPAN Plus really
puts the anal back in analysis.
www.voxengo.com

COMPUTER MUSIC SPECIAL / 83

> Step by step 3. Visually monitoring dynamics with WaveShaperCM in Logic Pro
1 2
WaveShaperCM, which can be found in the CM Plugins folder, is an
amazing, flexible distortion effect that can also be employed as an
oscilloscope. Let’s see how we can use it to check the dynamic
properties of an audio signal. Create a new audio track in Logic Pro
and drag Kick.wav onto it.
3 4
This gives us gentle distortion, as the title of the preset suggests.
We don’t actually want to change the sound of the incoming signal
at all, and the easiest way to make sure that no waveshaping is applied
to the audio is to click the New button in the top left-hand corner.
5 6
To change the resolution, click 1 bar (4/4) and select 1/4 from the
list. This sets the resolution to a single beat. If you prefer an old-
school unsynced display, click Beat and select Fixed instead. You can
now set the rate of the oscilloscope with the Hz parameter.
84 / COMPUTER MUSIC SPECIAL
Press X to bring up Logic’s mixer, and locate the Audio FX slot on
the Output track. Select Audio Units»CableGuys»WaveShaperC
M»Stereo. You’ll notice that the waveshaping function represented by
the XY panel on the left of the interface is slightly curved.
This gives us a perfectly linear waveshaping function, which
means that the sound passes through the plugin unaffected. Now
we can focus on the oscilloscope display to the right-hand side of the
interface. At the moment the tempo resolution is set to 1 bar.
Finally, click the Audio FX slot on the audio track and select
Dynamics»Compressor»Stereo. Any changes you now make to
the kick’s dynamic properties can be seen clearly in WaveShaperCM’s
oscilloscope. Analysing audio like this is particularly useful when
limiting your entire mix, helping you to ensure that it’s not becoming
too squashed.
> Step by step 4. Fattening up resampled synths with MAnalyzer
1 2
When working with resampled one-shot synth sounds, we can
perform surgical equalisation on particular harmonics that
wouldn’t be practical on parts playing more than one note. Using a
spectral analyser and EQ, we can tweak the level of individual
harmonics and resample the sound in order to play it chromatically.
Start by loading Raw synth.wav onto an audio track in your DAW.
3 4
Now place DDMF IIEQPro CM (in the CM Plugins folder) before
MAnalyzer in the effects chain. With this arrangement, we can EQ
the sound and the results will be reflected by the spectral analyser.
5 6
Set the first band to +3.8dB, the second to +5.7dB and the third to
+8.5dB. This clearly gives the sound a boost in the low end, but we
need to be careful that the processing we’re applying is actually
having a positive effect on the sound. If you bypass the EQ you’ll see
that the track peaks at a lower level when the plugin is deactivated.
We can use any spectral analyser for this, but MeldaProduction’s
MAnalyzer (free from www.meldaproduction.com) is particularly
convenient because it displays the frequency values of the input
signal’s first few harmonics. Add the plugin to the audio track and
bring up its interface. We can immediately see that the second, third
and fourth harmonics sit at 65Hz, 131Hz, 196Hz and 261Hz.
Set the first three bands of IIEQPro to DigitPeak mode, turn their Q
up to 6.50, and set them at 65Hz, 131Hz and 196Hz. We can now
use these bands to adjust the levels of the first, second and third
harmonics, shaping the sound in a manner akin to additive synthesis.
To properly gauge the effect the equalisation is having, we need to
hear how it sounds when it peaks at the same level as the
unprocessed signal. Set the Volume knob at the top right-hand corner
of the interface to -2dB. Now the channel peaks at the same level,
whether the plugin is active or not, and we can assess whether our EQ
is having a beneficial or detrimental effect on the sound. (Audio:
Boosted lows.wav)
COMPUTER MUSIC SPECIAL / 85
 INTRODUCING A NEW SERIES OF APPS
CELEBRATING THE ICONS OF ELECTRO

KRAFTWERK
EDITION
OUT NOW

The Electro Legends Series

Ralf Hütter interview / Autobahn revisited / Video tutorials
Analysis by OMD & John Foxx / Live review
 UNIVERSAL
TRANSLATION
The best mixdowns will sound great on any
playback system, big or small. Here’s how to
ensure your tracks do the same
A trap that it’s all too easy for the aspiring
dance music producer to fall into is that of
creating mixdowns that only work well on
one kind of playback system. It’s an
infuriating problem, as we tend to assume
that once we’ve got our productions up to a
standard where they sound great on a club
system, they’ll also sound good on
headphones, in the car, or anywhere else.
Sadly, this isn’t the case, and we often find
ourselves becoming so focused on the
requirements of a particular type of system that
we fail to consider how a project will sound in
other contexts. This leads to inevitable
disappointment when we test our music out
elsewhere. While there’s no substitute for
experience in this type of scenario, the good
news is that there are various strategies that we
can employ to ensure our music sounds great
over as wide a range of reproduction equipment
as possible.
They key to transferability is in ensuring that
you’re not leaving any ‘gaps’ in your mix, in
terms of both frequency spectrum and stereo
panorama. Having big, beefy bass, solid kicks
and slamming snares might be enough to make
your track bang on a club system, but unless
you’ve got the right energy in the mids it’ll
sound tinny and unimpressive on laptop
speakers, which will be unable to reproduce the
lows. Likewise, your mix might sound fine on a
big rig and a laptop, but have a listen to it on
headphones and you might discover that its
stereo image is lacking.
Working in bedroom or project studio
environments can make getting that elusive
perfect mixdown much trickier, because it’s
harder to trust what you hear in such a
restricted environment. But don’t worry,
because in the following walkthroughs we’ll
reveal some essential tips and tricks that make
use of analysis plugins to ensure that you’re
getting a full-sounding mix no matter how
compromised your monitoring environment.
We’ll also look at how to make your bass-driven
tracks more frequency-rich using layering and
processing techniques that will help them sound
better on smaller systems.
COMPUTER MUSIC SPECIAL / 87
> Step by step 1. Using spectral analysis to balance musical elements with beats and basslines
1 2
While your ears must have the final say in any mixing decision,
spectral analysis can be helpful for balancing elements in a mix,
especially if you’re working on headphones or in a less than ideal
studio environment. Launch your DAW, set the project tempo to
130bpm and create two audio tracks.
3 4
The mids don’t peak that high, so we know that we have some
room in the mix there to accommodate other sounds. Unsolo the
first track. The piano is quite loud, so turn its track down to -12dB. Now,
watching SPAN’s display as you do so, gradually turn the level of the
piano channel up.
5 6
The signal isn’t peaking as high in the 600Hz-4kHz region, which
means we’ve got some leeway to boost the piano’s mids and highs.
Add an EQ plugin to the piano track, and use a high-shelf EQ to boost
by 6dB at 500Hz. We could also do with making the piano’s
fundamental slightly quieter, so use a bell-shaped EQ band to take off
1.5dB at 350Hz.
88 / COMPUTER MUSIC SPECIAL
Drag Groove.wav onto the first track and Chorus piano.wav onto
the second, then add Voxengo SPAN (free download from www.
voxengo.com) to your master buss. Solo the first track so that you can
monitor the drums and bass. In SPAN you’ll see that the highs and lows
peak at between the -39dB and -42dB regions.
As the channel reaches the -4.5dB mark, you can see that the
piano’s fundamental (that is, lowest) frequency is reaching a
similar sort of level to the low and high peaks of the beats and bass.
However, you can also see that the piano sound is relatively dull, as the
fundamental frequency is quite a bit louder than its harmonics.
Now we have a mix with a solid mid-range. You can take a listen to
any specific frequency range to hear how full it sounds by holding
Ctrl/Cmd and dragging over SPAN’s analyser panel. This is handy for
making sure that your mix has an appropriate energy level over the
entire frequency spectrum.
> Step by step 2. Mixing vocals using spectral analysis
1 2
As we saw in the previous tutorial, generally you’ll want all your
musical elements to peak at about the same level on a spectrum
analyser for an even-sounding mix. The one exception to this rule is
vocals, which need to be heard as clearly as possible. Create a new
project in your DAW, and set the project tempo to 130bpm.
3 4
Turn the Vocals track down to -20dB, and unmute it. Gradually
turn its volume fader up. By the time you get to about -12dB, the
vocal will be at the right kind of level to be intelligible over the backing
tracks. You’ll see in SPAN that at this level it peaks the highest of all the
elements, even though it doesn’t sound overly loud in the mix.
5 6
Another way to make the vocal sound clearer is to duck the
volume of the Music track when it plays. To do this, add a
Compressor effect to the Music track, activate its sidechain and set the
sidechain source to the Vocal track.
Create three audio tracks, and name them ‘Groove’, ‘Music’ and
‘Vocals’. Drag Groove.wav onto Groove, Music.wav onto Music,
and Vocals.wav onto Vocals. Mute the Vocals track and put Voxengo
SPAN on the master. Bring up SPAN’s interface and you’ll see that the
instrumental tracks really fill out the frequency spectrum.
We can make the vocal stand out in the mix more by EQing the
music track. Add an EQ to the Music track and use a bell-shaped
band to take out about 6dB at 1kHz. You can use the Q parameter to
establish the width of this band. The wider the band, the more the
music is weakened, but the stronger the vocal will sound.
Set the Compressor’s Threshold to -10dB and Ratio to 2.00:1.
Now the Music track will drop in volume slightly when the vocal
plays. This helps maintain a consistent level of energy in the mids, and
the reduction in volume isn’t obvious because the vocal focuses our
attention. (Audio: Balanced music and vocal.wav)
COMPUTER MUSIC SPECIAL / 89
> Step by step 3. Using the Haas technique in Ableton Live

1 2
In Mastering the Stereo Field (p66), we talked about how some
effects use the Haas technique to create a wide stereo signal. The
downside of this is that it can have a less desirable effect on the mono
signal. Let’s see how we can create this effect manually, then check it
for mono compatibility. Start by creating a new project in Live and
putting Synth chord stabs.wav on an audio track.
Add Live’s Simple Delay effect to the audio track. At the default
settings this creates a basic stereo delay effect with no feedback.
Set the Dry/Wet level to 100% so that we just hear the delayed signal.
Now click the left and right channels’ Delay Mode buttons to switch
them from tempo-synced to time-based.

3 4
Turn both Ms Delay values down to 1.00ms. Now we can control
the delay on each channel using specific millisecond values.
Gradually turn up the right channel’s Ms Delay amount, and listen to
how the signal goes from mono to stereo.
The bigger the gap in the delay between the left and right
channels, the wider the stereo image becomes. Set the right
channel’s Ms Delay to 26.7ms, and add Live’s Utility effect to the
master buss. To check how the signal sounds in mono, bring Utility’s
Width down to 0.0%. The signal still sounds good in mono, so we
know the Haas technique isn’t having too much of a detrimental effect.

Push it to the limit

Throughout this Special we’ve
demonstrated plenty of ways to make your
music sound louder, wider and weightier. The
quest for this kind of ostensible improvement
in sound quality has been ongoing since the
dawn of recorded music, but it seems that,
now more than ever, the pursuit of the
loudest possible mixdown is dictating the
actual content of the music (see
StoneBridge’s remarks in Mixing and
Mastering for Club Playback, p31) as well as its
sound. While it’s obviously futile to even
attempt to reverse this trend, it’s probably
worth questioning whether or not you,
personally, need to follow it.
While many EDM tracks might benefit from
huge amounts of sidechain compression and
gigantic, frequency-filling unison-detuned
saw patches, these techniques aren’t
necessarily applicable to every track or every
genre. Be realistic when you compare or A/B
your music with that of other producers –
your soulful deep house groove is unlikely to
ever fill the frequency spectrum in the same
way that a full-on, sawtooth-powered bro-
step banger does, but it definitely needs to
stand up to the current production values of
its own genre.
Once you’ve followed and understood the
techniques presented in these tutorials, the
next thing you need to develop is an ability to
judge how far you should take them on a
track-by-track basis. This skill can only be
developed through experience, so try to
actually finish as many tracks as you can to as
high a standard as possible. Eventually you’ll
understand which tricks work well in your Not all dance tracks need to sound as though they were
music, and how far you can take them. created by the King of Wobs himself

90 / COMPUTER MUSIC SPECIAL

> Step by step 4. Doubling a bassline with a mid-range part
1 2
The ironic thing about the rise of bass music is that it uses a lot of
full-frequency-range sounds. The genres that bass takes its
inspiration from (such as house, jungle and garage) would often have
the simplest of subs, but that type of bass doesn’t reproduce very well
on smaller speakers. Create a new project in your DAW, set the tempo
to 130bpm, and create one audio track and one MIDI track.
3 4
If we want the bassline to be audible on smaller speakers, one
solution is to layer it with a mid-range element that can be more
easily reproduced. Drag Bass.mid onto the MIDI track. This is the
sequence used to create the bassline, and by using it to play another
instrument we can double the bass part.
5 6
Now we need to make a suitable sound for Dune CM to play that
sits with the sub-bass. Set Osc 1 and Osc 2 to sine waves by clicking
the buttons next to the sine waveforms in each one. This gives us a
simple sine tone, but by turning up the Ring Mod knob in the Osc
Common panel, we can increase the number of harmonics present in
the signal and get a more interesting timbre. (Audio: Ring mod.wav)
Drag Groove.wav onto the audio track. If you’re using laptop
speakers or earbuds rather than serious monitors or decent
headphones to follow these tutorials (which isn’t recommended!), you
might not be able to hear the bassline at all, but by putting a spectral
analyser on the master output you’ll see that it’s definitely there.
Load Dune CM (in the CM Plugins folder) onto the MIDI track. Click
the Bank B button to initialise the patch. Currently the MIDI part is
in the octave of C0, which is fine for a sub-bass but too low for a mid
part. Open the MIDI part, select all of the notes, and transpose them up
a couple octaves so that the part starts on G2.
Set Ring Mod to 100%. We can’t hear Osc 2, but changing it still
affects the synth’s output because it’s ring modulating Osc 1. Turn
Osc 2’s Semi (semitone transpose) up to +12 to create a classic house
organ sound. In the Amp Envelope section, set the Attack time to 9%
and the Release time to 19% so that the start and end of the note don’t
click. (Audio: Mid layer.wav)
COMPUTER MUSIC SPECIAL / 91
> Step by step 5. Exciting sounds with TAL-Bitcrusher
1 2
Another way to help sounds fill out the frequency spectrum is to
use to use so-called ‘lo-fi’ effects that work by reducing sample rate
and bit depth. Let’s look at how we can give a sine-based sub-bass
some high end using the awesome Togu Audio Line TAL-Bitcrusher, a
free download from http://kunz.corrupt.ch. Create a new audio track
in your DAW and drag Sine sub.wav onto it.
3 4
Let’s set up an initialised patch so that we can investigate what
TAL-Bitcrusher is capable of. Set the Bitdepth to 32, Samplerate to
/0, Noise to minimum, Low Shelf and High Shelf to 12 o’clock, Dry/
Wet to 100% Wet, and Volume to 12 o’clock, as shown above.
5 6
Clearly, this is a dramatic way to affect timbre. A more subtle
method is to use sample rate reduction. Turn the Bitdepth all the
way back up to 32 and bring the Samplerate knob up. The higher it
goes, the lower in frequency the aliasing artefacts are, and the more
the signal becomes square in shape.
92 / COMPUTER MUSIC SPECIAL
Put a spectral analyser on your master, and add TAL-Bitcrusher to
the audio track as an insert effect. When you play the bassline
back, you can see from the spectral analyser that TAL-Bitcrusher’s
default settings have added a lot of high-end information to the signal.
Try bringing the Bitdepth down. You’ll hear that the sound doesn’t
change much until we reach 24 bits, at which point some subtle
fuzz appears. This rapidly increases as the Bitdepth parameter falls,
and by deploying WaveShaperCM’s oscilloscope (in the CM Plugins
folder) we can see that by the time we hit a Bitdepth of 2 the signal has
become a pulse wave!
Return the Samplerate to /0 and bring the Noise parameter up.
This adds a crusty high end to the sound, and as the setting rises,
the original signal reduces in volume. By using a little of each of the
lo-fi effects we’ve just described, we can get a heavy bass sound that
nonetheless has a bit of crunchy character in the high end.
 11
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 FIX YOUR MIX
Even experienced producers can have trouble with
some projects. Here are our suggestions for what to
do when you just can’t get that mix right
So, you understand the theory, you’ve got the
hang of the your production tools, and you
have a clear idea of what you want your mix
to sound like. How come, then, certain tracks
just feel impossible to mix down properly?
Being able to determine exactly what the
problem with a particular project is and
whether it’s actually fixable is an enormously
important skill for any producer to master.
Not just because it can save you many hours
of tweaking, but because struggling with a
track that sounds bad can be a truly
demoralising experience! There’s no single
solution to what to do when a project isn’t
working out, but rather several strategies
that you can use to systematically solve your
studio nightmares.
The initial step is to identify the problem. First
of all, is there a particular environment in which
you’re experiencing dissatisfaction with the
track? If the project sounds good in your studio,
but not when played back on different systems,
you’ve got a specific issue with mix translation –
see Lost in Translation, opposite. If your track
doesn’t sound great even in the place where you
94 / COMPUTER MUSIC SPECIAL
produced it, there are more fundamental issues
with the mix that need to be addressed.
It’s actually pretty common for producers to
feel disappointment when reviewing their work,
and it can boil down to a number of factors.
Many of them come about because of staying
up late working on a project. Listening to music
at even a modest volume for an extended period
of time can result in ‘ear fatigue’, which makes it
hard to properly discern subtle changes in a mix.
When your ears are tired, it’s easy to
inadvertently compensate for this by boosting
“It’s actually pretty
common for
producers to feel
disappointment when
reviewing their work”
particular frequencies, or pushing compressors
harder than they should perhaps be pushed. In
the morning, your ears will have had a chance to
rest, and you’ll be able to hear the negative
consequences of your late-night tweaking.
Rest your ears
Another consequence of making music late at
night is that if you have to turn your monitors
down in order not to disturb those in the
proximity of your studio, you might end up
mistakenly compensating for the fact that the
human ear doesn’t have a ‘flat’ response to
different volume levels at different frequencies
(see Don’t (Always) Believe Your Ears on p34).
Also, having your volume level too quiet can
simply make your mix more difficult to hear –
especially if it’s competing against background
noise – and this will make it even more likely that
your nocturnal work won’t be up to scratch. If
you can feel that your ears are becoming
numbed while working on a project’s mix, you’re
better off taking a break to give them a rest
rather than carrying on, as any changes you
make while your ears are tired aren’t likely to
Lost in translation

IK Multimedia’s ARC
System 2 helps you tweak
your monitoring to better
match your room

As we touched on in Mixing and
Mastering for Club Playback on
p32, translation problems can be
caused by the particular acoustic
properties of an untreated studio
environment affecting the way
you mix your tracks. In an ideal
world, we’d all be able to
acoustically treat the rooms in
which we make our music, but
sadly this isn’t possible for the
average bedroom producer.
When working in less than
ideal mixing environments, using
headphones can give us more
clues as to how the mix actually
sounds, but just like our monitors
and room, they colour the sound
and don’t give a 100% accurate
representation. The unfortunate
reality is that no monitoring
system can ever give us a truly
perfect representation of what’s
happening within of our DAW – all
we can get are clues to varying
degrees of accuracy.
Nonetheless, by hearing how
your productions translate to a
variety of other systems, you can
gradually learn the sound of your
studio and how to compensate
for it. A/B-ing and visual analysis
can help with this, and there are
also dedicated correction tools
for dealing with imperfect
monitoring environments. One
such tool is IK Multimedia’s ARC
System 2 (www.ikmultimedia.
com), which uses a specially
calibrated microphone to record
pitch sweeps that tell the
software how your monitors and
room are colouring the output
from your DAW. This information
is then used to create a corrective
EQ curve that can be applied to
your DAW’s master buss using an
included plugin.

improve the mix and will only require correcting
at a later stage.
If you’re having trouble working on a mix
even in optimal conditions and are struggling to
get it sounding as defined as you’d like, the
culprit could well be over-enthusiastic buss or
master processing. Excessive dynamic range
reduction can make it impossible to get the right
sound, because your transients will be smeared,
your bass will be distorted, and the overall mix
won’t have room to breathe. Thus, it’s essential
to check both processed and unprocessed
versions of your mix if you work with a limiter or
clipper on your master. If your mix sounds too
rough or feels overcooked, try bypassing all
buss and master plugins that affect dynamic
range and listening to how the sound of each
element changes. Excess corrective EQ on
busses can also cause problems too, so try
taking it off and getting each element sounding
right before it hits the buss.
If easing off the processing doesn’t cure your
track’s ills, the next stage is to use spectral
analysis to ensure that you’ve got an even
distribution of energy throughout the frequency
spectrum, and that there aren’t any frequencies
that are peaking too low. Not having enough
energy in particular frequency ranges can result
in a track sounding weak or cheap, and the fix is
usually to adjust the balances on your mix
rather than simply apply additive EQ. If the mix
is looking really unbalanced, you might find it
easiest to take the level of all the tracks down
and rebalance everything from scratch, bringing
in the kick, snare, hats, bass and then others,
until each part of the frequency spectrum peaks
at roughly the same level. You might find it
helpful to examine tracks in the genre that you
make using a spectral analyser, and compare
them to your own – see the walkthrough on the
following page for step-by-step instructions on
how to do this.
Less is more
When starting out in production, it’s tempting to
try to cram more and more sounds into a mix to
fill it out and make it sound impressive. However,
more often than not this tactic has the opposite
effect to the one intended. The more clutter
that’s added to the mix, the harder it is for the
important elements to be heard. Dance music is
all about punch and dynamics – if the drums and
bass are having trouble cutting through the mix,
or other sounds are detracting from them too
much, you’re going to have a hard time getting a
groove going. Sidechain compression can help
with this (see Sidechaining for Impact on p58),
but you might find that the mix sounds better
when some parts are simply removed
altogether. Try going through and muting every
part track by track – if the mix sounds better
with a particular part muted, get rid of it!
On a similar note, reverb and delay effects
can really clog up a mix if they’re used
carelessly. If you suspect that the problems
you’re experiencing are down to excess reverb
and delay, try turning the levels of these effects
down, using EQ to attenuate the low end of their
return channels or compressing them with a
sidechain input fed from the kick and snare.
If all else fails, perhaps there’s something
fundamentally wrong with the track, such as the
drums or musical elements not being tuned
properly, the sounds themselves not being up to
scratch, or the track being in a key that makes it

COMPUTER MUSIC SPECIAL / 95

> Step by step 1. Using Voxengo SPAN to visually A/B your tracks
1 2
Voxengo SPAN is one of the best spectral analyser plugins around,
and it can be downloaded for free from www.voxengo.com. Install
it and load a work in progress in your DAW. We’re going to use SPAN to
A/B compare the frequency response of your project with a
commercial track. Create a new audio track and name it ‘AB1’.
3 4
In Ableton Live, you do this by setting the output routing to Ext
Out, and selecting outputs 1/2. We don’t want to play the A/B track
at the same time as the project – rather, we want to swap between
them. To facilitate this, mute the AB1 track. Now you can swap between
hearing your project and the comparison track by clicking the AB1
track’s Solo button.
5 6
Click the Edit button to bring up SPAN’s analysis parameters. Play
your project back and sweep the Avg Time (average time) knob to
see the effect that it has on the spectral analyser display. A setting of
about 300ms gives us a quick response without being too fast to
decipher. Copy this instance of SPAN onto the AB1 track – now you can
compare the frequency response of your track with the other one.
96 / COMPUTER MUSIC SPECIAL
Import a tune that you think sounds well produced onto the track.
We don’t want to route the track through any processing you
might have on your master buss, so send it directly to your audio
interface’s main physical output rather than going via the master.
This is a useful technique in itself, but unless you’re working in a
well-treated studio environment, it’s likely that your room is going
to colour the sound quite a bit. This means that we can’t completely
trust our ears, which is where SPAN comes in handy. Insert SPAN on
your master buss and bring up its interface. Now when you play your
track back, you’ll be able to see as well as hear it!
It’s a good idea to use several comparison tracks so that you can
ensure your project sounds right. You want to make sure that each
part of the frequency spectrum peaks at roughly the same level on the
display. You can band-pass filter the audio by holding Ctrl/Cmd and
dragging on the spectral view. This is helpful if you want to hear what
sounds are sitting in specific areas of the frequency range.
Hearing is believing
The human ear may be amazing, but
it wasn’t designed as a bespoke tool
for mixing music, and thus it needs
to be trained for this purpose. If
you’re a novice producer, you might
have a good idea of what you want
to hear in your music in terms of
ideas and content, but its hard to
gauge whether what you’re hearing
in your studio is punchy enough, or
whether the frequency balance is
right. This is where A/B-ing
becomes invaluable. Sure, simply
listening to dance music of a similar
genre on your studio setup is a great
idea, but the ear has a short
memory. To really compare your
work effectively, you need to have
as short a break between listening
to your project and your
comparison tracks as possible,
which is why the procedure
suggested in the previous
walkthrough, allowing you to
instantaneously switch between
tracks, is so useful.
Another helpful practice for
dance music producers is to have a
DJing session, mixing up a bounced
version of the project in progress
with other tracks of the same genre.
If you don’t have any DJing kit
yourself, don’t worry – it can be
done in your DAW or via dedicated Mixxx is a great freeware DJing tool that offers many of the same features as its commercial rivals
DJ software such as the freeware
Mixxx. This can be downloaded in
PC, Mac and Linux formats from
www.mixxx.org. Putting your
tracks in the mix will help you
quickly identify a range of different
problems, including mixdown and
arrangement issues.

difficult for the sounds you’ve used to sit
together well. In such cases, you might find that
correcting the error quickly improves the sound
of the track, though sometimes changing one
element will cause a domino effect, where you
end up having to adjust one part after another
to make them all fit – essentially sending you
back to the starting point of tuning and
balancing the drums. This is why it’s important
to get this stuff right from the beginning: it’s a
lot easier to work on the drums solo rather than
within the context of a fully constructed mix.
Waste not, want not
If you can get the beats sounding good but the
musical ideas on top of them aren’t working,
you might want to consider ditching the latter.
The drums are always the most important part
of a dance track, so it’s worth salvaging them if
you can, and often it’s quicker to come up with a
new musical idea than it is to fix one that’s not
working out.
The worst-case scenario is that you feel
you’re unable to salvage anything from a track,
and the project should be completely
“Often it’s quicker to
come up with a new
musical idea than it is
to fix one that’s not
working out”
abandoned – but even in this situation, there are
still positives you can take away. If the project
failed because of lacklustre beats, learn from
your mistakes and make sure you don’t allow
yourself to half-bake such an important element
of the track next time. If the project was an
interesting idea that you simply couldn’t make
work, take heart in the fact that you gave it your
best shot and tried something a little different,
rather than just repeating the same formula
over and over. Many successful producers have
folders full of unfinished tunes that they realised
it simply wasn’t worth their time taking to
completion, because the ideas weren’t strong
enough. By abandoning a track that’s never
going to work, you free yourself up to spend
time on something that might.
The final positive side to binning a project is
that you won’t be tempted to send it out to DJs
and other producers. Sure, it’s possible that a DJ
or label could have taken a shine to it, but do
you really want anything less than your best
work out there? Once a track has been released
into the world, it’s practically impossible to get
rid of all evidence it existed, especially if a label
puts it out. You might wish you’d kept less-than-
stellar tracks to yourself when they’re taking the
shine off your Beatport artist page!
Even worse, if a track is really bad, it could put
off whoever you sent it to from listening to any
more of your work. You don’t want to get
yourself into a situation where you never send
out anything, but it makes sense to leave tracks
for a few days before reviewing them, so that
you can assess them in the cold light of day. This
will give you the best chance of making an
objective judgement as to their quality.

COMPUTER MUSIC SPECIAL / 97

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98 / COMPUTER MUSIC SPECIAL
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