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Fri 19 Jun 2015
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July 2015 In This Issue
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2015
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Reviews
App Works
Recording and producing your own music is always

a challenge especially if, like Faith No More,
your previous albums have been done by the best
in the business!
Funkhaus Berlin Studio Tour

Video Feature
Making Music On The Move
Built in the '50s as the broadcast headquarters for

the GDRs state radio, this complex is home to
some of the world's most breathtaking recording
studios.
Akai iMPC Pro iOS sampling workstation and the

Klevgrand Enkl Synth App.
Bereich03 Audio Sideonly

Stereoimaging Processor
Is this humble 500-series processor the start of a

great modular system?
Big Fish Audio The Rock Collective

Multiformat
Library of radiofriendly rock instruments!
Inside Track: Alabama Shakes Sound &

Color
Secrets Of The Mix Engineers: Shawn Everett
In the making of Alabama Shakes Sound & Color,

producer Blake Mills and engineer Shawn Everett
had almost unheardof licence to experiment
and took full advantage.
Oasis & Britannia Row Live At Knebworth

Super Sonics
Oasiss 1996 gig at Knebworth marked the end of

an era for pointsource PA. We asked the people
who made it happen what has changed since.
Boz Digital Labs +10dB Channel Strip

Formats: Mac & PC VST, AAX & RTAS; Mac AU
The ADR F760X-R Compex compressor/expander

was a favourite dynamics processor during the 70s
and 80s.
Choosing & Using Porous Absorbers

Soft Machines
The choice of materials for acoustic treatment is

paramount. We find out what different types of
absorber can and cant achieve.
Cloud 44-A
Active Ribbon Microphone
Scoring I Survived A Zombie Apocalypse

Andrew Barnabas & Paul Arnold
How do you write music for a TV show you havent

seen yet? It helps if you can draw on years of
experience composing for video games...
Sounding Off: Tom Flint

When Does Technology Make Things Too Easy?
When does technology make things too easy?
The Long Game
This classically styled mic incorporates Clouds own Leader

preamp design to negate some of the pitfalls
traditionally associated with ribbon microphones.
Guitar synthesizers have been around since the

mid-1970s but have never really captured the
imaginations of the majority of guitar players.
CPU Performance Tested

The Need For Speed
Our benchmark tests show you which CPUs offer

the best audio performance, whether youre on a
budget or moneys no object.
Dbx AFS2
Voodoo Ray by A Guy Called Gerald

Classic Tracks
Hailed as the first British acid house single, A Guy

Called Geralds sublime Voodoo Ray has since
become a classic in its own right.
Feedback Suppressor
Is this Dbxs most userfriendly feedback

suppressor to date?
Drawmer 1973
Threeband FET Compressor
Te c h n i q u e
Another Level
Logic Tips & Techniques
Take control with Logic 10.1s new mix tools: VCA

faders, region automation, and Relative and Trim
modes.
Its not easy to create an analogue multiband

compressor that competes on quality and price, but Downsizing DP
Digital Performer Tips & Techniques
Drawmer show us it can be done.
When screen space is at a premium, its time to
customise Digital Performers user interface.
Eiosis Air EQ Premium
Equaliser Plug-in For Mac OS & Windows
Nine years after its original launch, Eiosis have

rebooted their Air EQ plug-in with some unique new Fair Shares
Pro Tools Tips & Techniques
features.
Before you deliver your Pro Tools project to a
collaborator, its vital to make sure it has everything
IK Multimedia iRig Pads
theyll need!
MIDI Controller
IK Multimedias iRig Pads brings portable control to

your computer or iOS device.
Filter Fodder
Studio One Tips & Techniques
The right reverb treatment can turn almost any

sound into source material for filter mayhem in
Studio One.
Mackie Mix 12FX

Compact Mixer
Mackies mixers have earned a reputation for

offering excellent value for money. We check out
their most affordable range to date.
Nemphasis X7
Tube Overdrive Pedal
Filter Fodder | Media

Audio files to accompany the article.
The right reverb treatment can turn almost any sound into
source material for filter mayhem in Studio One.
Media Content - July 2015
While an overdrive pedal might not seem like an

Audio files to accompany the relevant articles.
obvious studio accessory Ive often found that using
This page contains a directory of all media pages
an analogue overdrive pedal before a software
that hold the associated media for July 2015.
modelling amp/speaker produces more musical
results.
WIN Great Prizes in SOS
Competitions!
Win Sontronics Aria
Valve Mic
Win Dynaudio & TC
Bundles
Win MunroSonic Egg100
Monitoring System
Neumann U47 FET

Capacitor Microphone
Nearly 30 years after production of the original U47

FET ceased, Neumann have revived this iconic
large-diaphragm capacitor mic.
Orchid Electronics Mic Mute Lite

Microphone Mute Switch
It is often desirable to mute a microphone easily and

this unit might be just what you're looking for!
PreSonus AudioBox iOne & iTwo

USB 2 Audio Interfaces For Mac, PC & iOS
Microphone Types: Good Vibrations

Spotlight
What are the pros and cons of the different types of

microphone, and when would you want to use
them?
Mix Rescue: Spektakulatius

Our Experts Transform Your Tracks
Our engineer picks up where last months Session

Notes recording feature left off, with details of how
he approached the mixing side of things on this
ambitious project...
Mix Rescue: Spektakulatius | Media

PreSonus have expanded their AudioBox range with Audio files to accompany the article.
two compact interfaces that will work with your Mac,
Mix Tricks
PC or iOS device.
Reason Tips & Techniques
PSP MasterQ 2
We look at advanced mixing techniques in Reason.
Whilst PSPs basic design remains the same, with a

graphic display of the overall EQ curve and controls
Plug-in Pruning
for up to five parametric EQ bands. Almost every
aspect has been enhanced in some way or another. Cubase Tips & Techniques
Take some of the work out of your workflow, with
Radial Headload V8
Cubase 8 Pros Plug-in Manager.
Guitar-amp Power Soak & DI
A combination power soak, DI box, speaker

simulator and phase alignment tool, Radials
Headload has a role in almost any guitar rig.
Roland JDXi
Synthesizer
Rolands JDXi crams an awful lot of synth into a

very small box.
System Analysis
Ableton Live Tips & Techniques
We show you how to analyse a song using Lives

built-in tools.
Top Brass
Arranging For Brass: Part Three
Sample Shop | Media

Audio files to accompany the article.
These files accompany the Sample Shop articles in the July

2015 issue of SOS
Soniccouture The Hammersmith

Kontakt Instrument
A simple, catchy tune with a funky rhythm may be

all you need to create a highly commercial horn
hook but harmony is also an essential ingredient
in brass arrangements.
Vocal Point
Sonar Tips & Techniques
How to polish your vocal recordings in Sonar.
With The Hammersmith, a conventionally played

acoustic grand piano, Soniccouture have released
one of their most mainstream and potentially
important libraries yet.
Music Business
Soundizers StereoMonoizer
Notes From The Deadline
Audio File Utility
TV Music From The Inside
This useful piece of software can help you

streamline your projects and preserve precious
system resources.
Its always vital to give the client what they want.

Even if what they want makes no sense.
Spitfire Audio British Modular Library
Off The Record
Sample Library
Music & Recording Industry News
Spitfire Audios BML range packages a full

orchestral library into 20 individual modules.
Hollywood looks like it may be on shaky ground,

but its not all bad news for audio post houses.
TC-Helicon Ditto Mic Looper
The Pool, London
Microphone Effects Pedal
Studio File
For such a simple-looking box, decked out with one

knob, two buttons and no display beyond a single
LED, the TC-Helicon Ditto Mic Looper is a very
sophisticated piece of kit.
The Sound Of Poland B11X

Kontakt Instrument
In over 30 years in business, Miloco has developed

from a small independent studio to a venture that is
operating worldwide.
Sound Advice
Q Can you explain Reapers Playback

Resample mode?

Competitions!
Win Sontronics Aria
Valve Mic
Win Dynaudio & TC
Bundles
Monitoring System
TSOPs first product is a Kontakt instrument called

B11X, which was created by sampling every key
and sound generator of a rare Polish synthesizer
made in the 1980s.
UAD Sound Machine Wood Works

Formats: Mac & PC UAD2
Resample mode?
I cant find good explanations about the Playback
Resample Mode options available in Cockos
Reaper, specifically the difference between the
Good (64pt Sinc) and Better (192pt Sinc Slow)
modes.
Q How can I export a Pro Tools project for

mixing in Reaper?
Acoustic guitar modelling Plug-in.
My band tracked some songs at a local studio using their Pro

Tools rig and I want to mix the tracks at home in
Reaper.
Ueberschall Flugelhorn & Trumpet 2

Elastik
You get two for the price of one with this review, with
two closely related sample libraries from
Q How do I record from my guitar amps
Ueberschall.
headphone out?
Im seeking some guidance on why Im unable to

record from my guitar amp straight to my audio
interface.
Waves Dbx 160

A Waves plug-in based on the original hardware

Dbx 160 designed by engineer David Blackmer in
the early 1970s.
Q Whats the best way to clip my drums?

Ive read that I can get a good hiphop kick or
snare sound by clipping it. But when I try this, it
sounds horrible. What am I doing wrong?
Waves/Digico Digigrid
Networked Audio Infrastructure
Two heavy hitters in the world of digital audio have

come up with an intriguing new synthesis of
hardware and software.
Xfer Records Serum

Software Synthesizer
Serums slick interface belies an extremely flexible

wavetable synthesizer.
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Fri 19 Jun 2015
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In this article:
Akai iMPC Pro
Klevgrnd Enkl
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Making Music On The Move
Reviews : Software: ALL
Paul Nagle
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Published in SOS July 2015
Printer-friendly version
The main iMPC Pro screen with its array of performance tools.
Akai iMPC Pro
Sampling Workstation App For iOS
n 2013 Akais legendary MPC workflow reached the iOS platform in the form of iMPC. Two years later, its adopters could
be slightly miffed to learn theres a brand-new Pro version an entirely separate app. Properly tooled-up and loaded with
more of everything, iMPC Pro offers 64 tracks, a built-in piano roll editor, superior effects implementation and a choice of
friendly sample-fixing tools.
The app is divided into five pages: Main, Program, Mixer, Timeline and Song, and from the outset behaves as much like a
traditional MPC as an iPads screen and operating system permits. Given that iOS is a general-purpose operating system and
that tapping your finger on glass will never trump the experience of hardware, my iPad Air responded fairly consistently to
iMPC Pro, and better than some apps.
Each program consists of four pad banks, which equates to a maximum of 64 sounds. Akais ambition is clear when you
realise there are up to 64 tracks to call upon, one program on each. Keeping with the 64 theme, this is the number of
sequences that you can create and either arrange into a song structure or play them manually as the mood takes you. A newly
selected sequence will play on completion of the current theres no option to switch immediately as on other MPCs. If you
work in songs, these can be exported as WAV files to iTunes, Soundcloud or the Audiocopy clipboard.
Many favourite MPC features made the transition and amongst these are the universally handy note repeat, plus 16
Levels. Engage the latter and the pads switch between 16 different levels for the current voice, a quick and controllable way
to add dynamics. Joining these are several iPad-specific features, such as the 3D Perform button, which enables a number of
performance effects to be controlled by the iPads motion detectors. I cant decide if its novelty value or genuinely useful to tiltsweep the pitch or filter, but if youre prone to seasickness a regular Kaoss-style pad introduces effects such as tape-stop or
glitching. Thanks to these few simple finger gestures a bunch of performance tricks are at your disposal and if you hit on
something good, hit record and capture the actions into a sequence for posterity along with your pad hits.
In performance, youll be glad of Pad Mute to silence individual pads within a program. A muted pad is indicated by a (very)
thin red diagonal line, which doesnt show up well against the grey background. At least thats my excuse for puzzling for ages
over a mysterious inaudible sample. The Track Mute button performs the same trick for entire tracks.
PC
TV
SOFTWARE PLUG-INS that

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Delivered with over 1400 individual sounds (just under 700MB)

and 25 factory programs, iMPC Pro is a large and well-equipped
app. This being iOS, it draws from its own private pool of
samples and is blind even to those shipped with the original app.
The quality is excellent throughout and if the emphasis is on
percussion, thats hardly a bad thing. I was initially blown away
by the reverb in the demo tunes, until I realised this wasnt the
product of a finely tuned iPad, but of serious production polish in
the raw samples. If the long and magical reverb tails swell the
library a bit, Im not complaining and was more than happy with
the supplied drums, ethnic percussion, synths, effects and so on.
Individual pads within a program can be extensively customised
so the samples should theoretically have a long shelf-life. Having
tuned, panned and filtered each one, it can be processed by
either an overdrive or ring modulation effect. Building a library of
user programs from a combination of the factories and original
samples is a job that will reap rewards later.
iMPC Pros mixer, complete with Turbo Duck.
Sampling can be as fast as hitting record and yelling at the microphone, or as in-depth as ripping audio from your iTunes
music collection using a virtual turntable. Puzzlingly, my music collection consisted of a U2 album, but this stubbornly refused
to be sampled thanks to copy protection. Pasting audio from other apps worked smoothly, though, and very soon I was
exploring the graphical sample editor and its pinch-and-expand method of zooming. Samples may be intuitively trimmed,
amplified, reversed, faded, normalised and audiocopied, all to a surprising degree of precision. Its certainly sufficient to make
tracks.
Keeping with MPC tradition, a sample can be divided into up to 64 chunks and distributed across the four pad groups that
make a program. However, I must report a bad experience in my first long session with iMPC Pro. It started well enough: I
made a bunch of fairly elaborate programs on half a dozen tracks then grabbed a loop from another app and sliced it. Perhaps
because iMPC Pro is so seductive, it never occurred to me to save anything, so when the crash came (I was fine-tuning some
individual slices of my loop) it took all my sequences with it. Later, I discovered the quick save option and, later still, that the
various programs and sounds Id made werent lost, only my sequences. Incidentally, the timeline note editor doesnt need any
explanation other than to mention it exists and to point out you can select a note or notes and perform a high degree of
individual tweakage, if scultpting every note is your thing.
Moving quickly on, the mixer bears little resemblance to that of iMPC. Each track has a three-band EQ, effect sends and a
master channel complete with a compressor and a duck. Its no ordinary duck this is a Turbo Duck wearing a red baseball
cap. Its probably the simplest implementation of pumping side-chain compression imaginable and works swimmingly. In
addition to the reverb, delay and chorus sends, you can add an Inter-app effect, which appears on the mixer as a fourth send.
Theres not much to add except that tactile control is offered if you purchase the MPC Element hardware, a slimline unit
blessed with the same type of coloured rubber pads as the recent MPX16. It can be ordered from within iMPC itself, but I
didnt have one to test for the review. Generally, other than the occasional crash, the only real issues were those of iOS in
general, particularly the ease by which you get large samples in and out. At only $6 more than the earlier version, iMPC Pro is
a much more complete product whether for live performance or for developing songs from loop-based material. Paul Nagle
$12.99
www.akaipro.com
Klevgrnd Enkl
Synth App For iPad
levgrnds Enkl is a monophonic synthesizer for iOS, also available as a VST/AU plug-in. Named after an
approximation of the Swedish word for easy or simple, Enkls one-screen approach is visually simplistic in its
subdued greys and blues, but it still manages to host many of the features expected of a monophonic synth.
The two oscillators are primed with triangle, square, sawtooth and noise waveforms with a series of horizontal sliders for
pitch control (octave, semitone and fine-tune), plus a smaller one for start phase. In character the oscillators have a gnarly,
biting quality, but they lack refinements such as PWM or oscillator sync. Unusually, theres an LFO per oscillator which can be
aimed at either the pitch or amplitude. The LFOs have a range from 0.1 to 20 Hz and, pushing beyond unusual into the
positively rare category, each is shaped by its own ADSR envelope. With this neat convenience you can effortlessly achieve
delayed vibrato, tremolo that disappears when the note is released and other advanced modulation shenanigans.
The oscillators interact in one of three ways: addition, subtraction or multiplication. Of these, subtraction introduces phaseshifting effects and multiplication leads to buzzy, ring-mod-like results, especially when the oscillators are detuned. If you enjoy
the sound of broken 80s computer games, multiplication is the ideal place to start. Whichever way you combine the
oscillators, the output is eventually processed by a single amplitude envelope. Unfortunately this commits the cardinal sin (for
monophonic synths), in that it always starts its journey from zero rather than from the envelopes current level. Patches with
fast attacks arent compromised, but Enkls slow attacks suck. Theres also a chiptune-style arpeggiator, with a dedicated
speed control (but no hold option).
After the luxury of an LFO/envelope pair for each oscillator, its
a pity there werent any left over for the filter. The lack of
envelope is especially limiting because the low- and high-pass
filters together produce a good range of cutting, usable tones
that can stretch to formant-like characteristics if you manipulate
the resonance and two cutoffs manually. To help in this, external
MIDI control is provided via a series of fixed CCs, but hopefully a
future version might include a MIDI Learn function, allowing you
to pick your own.
I mentioned there isnt a filter envelope, but there is a single attack control that sets the time taken to reach the high-pass
cutoff. However, the effect of this was far from spectacular. The
GLOSSARY: technical terms

explained
Competitions!
Win Sontronics Aria
Valve Mic
Win Dynaudio & TC
Bundles
Monitoring System
final port of call in the signals journey to the outside world is a

stereo delay and three-band EQ. If you adjust delay time during
playback it goes briefly silent, but other than that its a fine
addition.
In use, many of the controls are quite small and fiddly, which is
odd given the amount of unused screen space. Even though
theres room for more keys, youre offered a single-octave
keyboard thats sure to have you reaching for a MIDI controller. If
you dont have one, or prefer to work entirely in the box, a
second keyboard is available at the push of a button. However,
this Extended Keyboard grabs half of the screen and removes
access to the controls. Not ideal, but at least it gives you a
realistic chance of hitting the right notes and offers basic scale
correction too, with four scales available. Better still, you can
gain extra performance control by dragging your fingers vertically
No skeuomorphism here: Enkls main page.
or horizontally after playing a note. This proves to be superb for
opening the filter, adding modulation or performing Clangers-style pitch-bending.
Enkl ships with 93 lead, bass, percussive, FX and lo-fi presets, sorted into friendly categories. Though a creditable enough
first synth, its priced slightly beyond an impulse buy and difficult to imagine as an app youd return to a lot. Still, its
monophonic simplicity could appeal to beginners, the synth-phobic or anyone who needs a fresh batch of fun sounds. Paul
Nagle
$5.99
www.klevgrand.se
.
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The contents of this article are subject to worldwide copyright protection and reproduction in whole or part, whether mechanical or electronic, is expressly forbidden without the prior written consent of the
Publishers. Great care has been taken to ensure accuracy in the preparation of this article but neither Sound On Sound Limited nor the publishers can be held responsible for its contents.
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Fri 19 Jun 2015
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In this article:
Overview
Operation
Verdict
Alternatives
News
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Information
WebExtras
Bereich03 Audio Sideonly

Stereoimaging Processor
Reviews : Processor
Buy PDF
Bereich03 Audio
Sideonly 400
Pros
Easy to use with a spacious
layout and switched
controls.
Feels robust.
Sonically very clean.
Decent price for the quality
on offer.
Cons
No Mid level control and no
Mid-signal processing
options.
Summary
This useful device does
what it claims to do to a high
quality, and it does so
without fuss and at a
reasonable price.
information
400 Euros (about $440
when going to press) plus
shipping.
info@bereich03audio.de
www.bereich03-audio.de
Is this humble 500-series processor the start of a great modular system?

Matt Houghton
his humble little 500series device does precisely what it claims to do in almost precisely the way I like, not that this
came as any great surprise, as Ive been impressed by Bereich03s previous products, the Density (reviewed in SOS
June 2012: http://sosm.ag/jun12density) and the LCHC (SOS May 2013: http://sosm.ag/may13lchc). Its designer,
Michael Heilrath, says he intended it primarily for mastering duties. The Sideonly, Density and LCHC are the first few
modules of a planned 500series modular mastering system, which has the potential to be very impressive but Ive found it
just as useful for manipulating stereo sources and bus signals while mixing.
Overview
Essentially, the Sideonly comprises a Mid/Sides (M/S) encoder and decoder, with a few useful facilities for manipulating the
stereo image sandwiched between them. The analogue audio inputs, accepted in the usual way via a host 500series
chassis, are balanced, as are the outputs. You feed a conventional L/R stereo signal to the device, which encodes this to M/S
to allow its controls to operate on the Sides signal, before converting to L/R again for the output.
While theres an option to solo only the Sides signal or only the Mid, the processing is performed solely on the Sides, the
Mid being left untouched hence the devices name. This might raise an eyebrow or two: if the processing facilities exist,
wouldnt it be a good idea to allow them to be switched for use on the Mid signal? That would certainly have made it a more
versatile device, but Heilrath says this was a deliberate decision, partly to keep costs (and price) down, and partly because the
application for which he designed it didnt require Mid-signal processing.
The build quality is good and the control layout spacious. There are no rotary pots
involved; everythings governed by toggles or rotary switches. Inside the unit are
three circuit boards populated with highquality components such as Wima
capacitors, sealed relays and THAT Corporation ICs, the last nestling in DIL sockets
that will make repair or replacement easy. The edge connectors that interface with
the host chassis/power supply are goldplated, all the soldering is neat, and all the
boards are fastened securely in place.
Operation
The general idea is that you can control stereo width of the signal in a way that
causes no problems for mono playback. But you can also roll off the low end from
the Sides signal, and there are several reasons why you might wish to do so: first,
removing low frequencies in the Sides signal enables you to avoid any problematic
phasecoherency issues in the low frequencies, a way of bringing more focus or
tightness to the bottom end; second, ensuring all the low frequencies are in the
centre of the stereo field ensures more equal distribution of power across the left
and right channels and thus both speakers in a stereo system; and third, some
people like to keep low bass information in the centre for other reasons, such as
when preparing material thats destined for vinyl.
The simplest tweaks are performed using the lower half of the doublewidth front
panel. Here, you can choose to listen to the whole stereo signal, just the Mid or just the Sides, and can boost or attenuate the
Sides signal by 5dB, courtesy of an 11stage switched control on the right. This both allows for manipulation of the perceived
stereo width, and to compensate for the inherent level changes in the Sides signal when EQing or filtering boosting widens
the perceived stereo width, attenuating does the opposite.

Toggle switches operate the on/off (hard bypass), which has a red status LED, and the solo facility. Two separate toggles
are employed for the latter, one to switch to solo mode a green LED blinks once selected and the other (no LED, but its
setting is clearly visible) to determine whether its the Mid or Sides signal thats heard. I found this dualtoggle arrangement
useful, because while a single threeway switch could have taken care of this solo facility (Mid/Sides/off), it would have made
A/B comparison slightly more fiddly.
A horizontal black line runs above this master section and the controls located above this govern the processing. These
come in the form of a 12dB/octave highpass filter (called LowCut) and a Baxandall EQ. The filter, switched into the circuit
using a toggle switch with a flush, green status LED, offers six turnoverfrequency options for selection via a rotary switch: 55,
80, 115, 170, 250 and 375 Hz. The EQ, which also offers six frequencies, adds up to 10dB of gain/attenuation in switched
2dB steps, with unity gain at the 12 oclock position. The frequency options, again selected using a rotary switch, are 330Hz,
750Hz, 1.5kHz, 3kHz, 7kHz and 14kHz, and the curves are very broad, as youd wish for the devices likely applications.
Another toggle switch (green LED) allows the EQ section to be switched in or out of the signal path.
Verdict
Pegamento
para
Calzado
Accin rpida y
fcil aplicacin.
Mxima e!cacia.
Pida presupuesto!
When testing the Sideonly, I used it on a range of sources including a stereo drums bus, the mix bus and, while I make no
claims of being a mastering engineer, on some completed stereo mixes for a DIY finalising session. It was very intuitive in
use. Its a nicely engineered nononsense device with a refreshingly spacious layout, and it just gets on with its intended
function without imposing any sonic character of its own. The switched settings make both recall and A/B comparison easy,
and I never felt that there were any useful frequencies missing the choice of filter and EQ settings seems to me to have
been well judged. Furthermore, the switches all felt reassuringly solid, which meant I didnt feel I had to be overly delicate with
them when comparing settings.
In fact, the only thing that I ever really felt missing, just occasionally, was the ability to boost or attenuate the Mid signal. But
while that might have been useful, its no deal breaker, and the same result can easily be achieved by attenuating the Sides
and adding clean gain via the next device in the chain.
All in all then, the Bereich03 Audio Sideonly performs a very useful job with ease, and to a high standard. All of which
makes it very easy for me to recommend it to anyone in the market for such a device.
.

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There are various companies making Mid/Side devices Avenson Audio, for example and others make EQs (such as
the TK Audio TKlizer) or dynamics processors with built-in M/S switching to allow independent control over the Mid and
Sides elements of a signal. But those are all more costly than this, and I cant think of another device thats dedicated
solely to tweaking the Sides. Certainly not in this format, anyway.
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Fri 19 Jun 2015
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In this article:
Curtains For You?
Depth Matters
Ahead Of The Curve
Mind The Gap
Animal, Vegetable Or
Mineral Wool?
Cutting Shapes
Homemade Solutions
Choosing The Right
Absorber
The Absorbers On Test
Caveat Emptor
How They Work
Summary
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WebExtras
Choosing & Using Porous Absorbers

Soft Machines
Reviews : Miscellaneous
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The large reverberation chamber at the University of Salford, with EQ Acoustics ColourPanel 60 arranged for testing.
Photo: Trevor Cox
The choice of materials for acoustic treatment is paramount. We find out what
different types of absorber can and cant achieve.
Trevor Cox
o achieve a greatsounding studio, absorbing foam and mineral wool is often used to reduce the coloration that the
room would otherwise create. Both in control and live rooms, these porous absorbents can be used to reduce excessive
reverberance and attenuate reflections that would otherwise change the timbre of the sound at mid and high
frequencies. But how well do they perform? How should they be applied? And do household materials such as duvets offer an
effective alternative, or is this an urban myth? Sound On Sound gathered up nine samples and headed to the Acoustics
Laboratories at the University of Salford to find out.
Curtains For You?

Porous absorbents can be made from many different materials. Products designed for the purpose are often constructed from
fabricwrapped glass fibre or acoustic foams, but there are also soft furnishings that accidentally create absorption such as
curtains and carpets. Figure 1 shows how the absorption varies with frequency for all the products tested at Salford. The
absorption is quantified in terms of the absorption coefficient following an International Standard (for more details see the
Caveat Emptor box). An absorption coefficient of 0 indicates that no sound is absorbed, while a coefficient of 1 means all
sound energy is removed.
The lefthand side of the graph measures absorption at a
frequency of 100Hz in musical terms, the note G2 towards the
bottom of the cellos range and its clear that none of the
products produce much absorption in this region. In general,
porous absorbers are not good for treating bass effects such as
the ringing of roommode resonances. When a room acoustic
needs controlling at low frequency, this is best done with
resonant absorption using a device such as a membrane
absorber (see our February 2015 guide to controlroom design:
www.soundonsound.com/sos/feb15/articles/controlrooms.htm).
Although you can buy foam bass absorbers, the idea that
placing these in the corners of the room is the most efficient and
effective way of controlling the bass defies the physics.
Figure 1: A graph of absorption coefficient against frequency

for all products tested.
Fbrica de
Pegamento
Industria de
calzado y
tapicera. Rpido
y fcil de usar.
Consulte!
As Figure 1 shows, for porous absorption, the general trend is for the absorption to increase with frequency, before
flattening out somewhere close to 1. A curious feature of the measured absorption coefficients is that they can exceed 1. This
is odd, because the absorption coefficient is defined as the ratio of absorbed to incident energy, which means a coefficient
greater than 1 implies a treatment that somehow absorbs more sound energy than is incident upon it. This arises because of a
wellknown flaw in the measurement standard. Over the decades, acoustic engineers have gotten used to reading these
charts with seemingly impossible numbers on them!
What do these results say, then, about what makes a good porous absorbent?
Depth Matters
The shallower devices tend to be at the bottom of the graph, and the deeper ones at the top, so the shallower items absorb
less. Take the absorption coefficient at 500Hz as an example. Figure 2 plots this midfrequency absorption coefficient as a
function of the average depth of the treatments tested, as measured with a ruler. More depth means more absorption until a
depth of 5cm is reached, whereupon no improvement in performance is gained by making the treatment deeper.
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The performance of many porous absorbers can be explained by

comparing the size of the sound wave to the depth of the treatment. The
size of the wave is quantified using the wavelength, which is the distance
between adjacent crests of the wave. The wavelength varies greatly with
frequency: at 50Hz the wavelength is 6.8m, at 500Hz it is 0.68m and at
5000Hz it is 0.068mm. A simple rule of thumb is that for a porous absorber
to be effective it needs to at least a 10th of a wavelength thick. At 500Hz,
this equates to a depth of 6.8cm, which roughly tallies with the 5cm found
in Figure 2. The very large wavelengths of lowfrequency sounds are the
reason why bare acoustic foam makes an inefficient bass absorber unless
the material is extremely thick. For a 52Hz sound, for example, a layer of
porous absorbent 65cm thick is needed to provide absorption and this
needs to cover a large surface area, not just a few pieces in the room
corners.
The products tested at the University of Salford had lots of different
shapes: wedges, flat and wavy. By just looking at flat samples, the effect of
the absorbent depth becomes clearer, as Figure 3 shows. As the foam
gets deeper, the biggest effect is to push the bandwidth of useful
absorption to a lower frequency. The 13mm foam is effective from about
2000Hz up; double the depth to 25mm and an extra octave of high
absorption is obtained, with good performance now from 1000Hz and
above.

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Figure 2: The absorption coefficient at 500Hz

plotted against average depth for the products
tested.
Ahead Of The Curve

The absorption curves in Figure 1 split into two types: (1) a rough
straight line rising steadily across the whole frequency range
measured, and (2) a response that first rises and then flattens out at
and above 500Hz. The latter shape, the one that looks a bit like the
response of a highpass filter, is the most useful for room acoustic
treatment. One use of absorption is to reduce reverberation. In most
rooms there will normally be some absorption at high frequencies due
to carpets and curtains, and where additional treatment is most
needed is at low and middle frequencies. Consequently, the best
response for a porous absorbent is similar to a highpass filter, with
the challenge being to get the useful range starting at as low a
frequency as possible. Another use for porous absorbent is to
attenuate specific reflections from the walls, ceiling and floor of a
room. What is needed here is the broadest possible frequency range
to remove all parts of the reflection. Thinner materials might have a
use in treating specific highfrequency defects such as flutter echoes,
but otherwise a deeper absorber is better.
Mind The Gap
Figure 3: The performance of a porous absorbent is

strongly influenced by the thickness of the treatment
(data from MJ Cocker, Handbook Of Acoustics, John
Wiley & Sons, 1998).
However, all is not lost for the thinner products, because their
performance can be greatly improved by giving them some air. When a piece of acoustic foam or fibreglass is placed onto a
wall, most of the absorption is done by the part of the material that is furthest from the wall (see the How They Work box).
Consequently, getting better performance does not necessarily require making the material thicker: mounting it away from a
wall can be equally effective. Figure 4 shows the absorption coefficient
for two fibreglass samples. The absorption provided by the 51mm
sample is identical to that provided by a sample half the thickness, but
set 25mm from the wall. The only problem with this solution is that it isnt
always easy to mount the absorbents away from the wall. Sometimes it
is easier just to buy thicker material for ease of attachment.
The only product in our tests that explicitly exploits the idea of getting
the absorbent away from the wall was the Profoam. This has a wavy
pattern that naturally places an air gap behind some of the material. This
makes it perform slightly better than the wedges that had a similar depth,
but still there is not enough of the absorbent far enough from the wall to
create sufficient midfrequency absorption.
This need to get porous absorption away from walls also influences
how curtains should be hung if they are to be used for acoustic
treatment. Figure 5 shows the absorption coefficients for two different
hangings of a curtain. As the curtain is hung with deeper folds, more of
the material is further from the wall, and hence more absorption is
gained.
Figure 4: The same absorption can be achieved by
Figure 4: The same absorption can be achieved by

hanging a thinner piece of absorbent away from the
Animal, Vegetable Or Mineral Wool?

Figure 1 shows that of the materials we tested, the two mineralwool
samples perform similarly, and are comparable with Block100, the most
effective of the acoustic foam samples. The Block100 foam is, however,
almost twice the thickness of the two mineral-wool samples. This
illustrates that a variety of materials can be used to make
effective porous absorbents. This is to be expected, because
what counts is the size, shape and tortuous connectivity between
the air pores and not the material of the matrix. I have heard at
least one person suggest that melamine foam is intrinsically
better than polyurethane. In terms of acoustic performance, the
only melamine foam tested was the Profoam, and this did not
show additional absorption compared with the polyurethane
samples. If they are designed properly, all these materials can
absorb sound.
wall, creating an air gap behind (data from CM Harris

(ed.), Handbook Of Noise Control, 2nd ed. McGraw
Hill, 1991).
One difference between the fibreglass and foam samples that

Figure 5: By hanging curtains with big folds to move the
is lost in the standard measurement result is the effect of the
material away from the wall, the absorption can be greatly
edges of the samples. The two mineral-wool treatments had hard increased (after TJ Cox and P DAntonio, Acoustic Absorbers
& Diffusers, CRC Press, 2009).
frames at the edges, one made from wood and the other from
plastic, whereas the foam samples had the absorbent exposed
at the edges. This means that if the foam samples are used in a studio with their edges exposed, you will get a little more
absorption than is shown in Figure 1. It depends on the thickness of the samples, but for the 1012squaremetre samples
we tested, leaving the edges exposed can increase the absorption by between 3 and 10 percent. For this reason, spacing
foam tiles out with gaps between can increase absorption. But if the gaps are too large, there is a risk of getting a strong
reflection from the exposed wall.
Another difference between materials is that the mineral wool needs covering to protect it from damage. If this is done with
a thin cloth that readily allows sound waves to enter into the absorbent, then there is little difference in performance, as Figure
6 shows. A cloth covering tends to provide a small boost in
lowerfrequency absorption, but at the cost of highfrequency
performance. For many applications, this is a good tradeoff to
make, because it is midfrequency absorption that is hard to
achieve. Foam does not need covering, but it is important that it
passes appropriate fire safety standards.
Cutting Shapes
Acoustic foam comes in lots of different-shaped patterns. Some
companies seem to favour foam that echoes the shape of egg
cartons, others copy the shape of acoustic diffusers, and two of
Figure 6: Adding a cloth covering to mineralwool absorbent
the samples tested at Salford were made from wedges. The
boosts lowfrequency absorption but reduces highfrequency
results in Figure 1 show that forming the foam into wedges
performance (after TJ Cox and P DAntonio, Acoustic
degrades the acoustic performance. What is needed is to get as
Absorbers & Diffusers, CRC Press, 2009).
much of the porous absorbent as far away from the wall as
possible. For that reason, I suspect the wedge samples would perform better mounted with the flat side facing the room,
although it would be difficult to attach the wedges to the wall! Auralex make a large number of differentshaped treatments
from the same acoustic foam. Examining their table of published absorption coefficients for 50mmdeep samples, one thing
that leaps out is that all the profiled shapes perform worse than the simple 50mm flat tile.
Homemade Solutions
Is it possible to use soft furnishings to treat a room and save money on expensive acoustic treatments? As Figure 5 shows, it
is possible to gain some absorption from curtains, but the acoustic treatments we tested performed better. To get more
comparable absorption from curtains, the drapes need to be thick and heavy. Special acoustic curtains might be made from
wool serge or velvet velour and weigh about 0.5kg per square metre. A pair of thin drapes from your local home furnishing
shop is not going to be sufficient. One advantage that curtains have over the treatments Salford tested, however, is that they
can be drawn back to vary the acoustic, a feature useful in live rooms.
While hard flooring has become very popular over the last few decades, using carpet in a studio will gain useful absorption
and reduce noise from people walking about and moving chairs. Being a porous absorbent, carpet produces mid and high
frequency absorption. To maximise the absorption, the carpet should have an open back with an opencell underlay. The
absorption is very dependent on the type of carpet and how it is made. Figure 7 gives the performance of the best and worst
carpets from data in various textbooks. Focusing on one
frequency, for example 1000Hz, the absorption coefficient can
vary between 0.1 and 0.8 for different carpets. Data from books
cannot be relied upon when picking a carpet: measurement data
for the particular product you want to buy is needed.
An old belief that refuses to die is the idea that egg boxes
make good absorbers, which perhaps stems from their visual
resemblance to some kinds of legitimate acoustic treatment. As
Figure 8 shows they can provide some absorption, but it is very
uneven across the frequency range, with a resonant peak at
about 700Hz. Presumably this is caused by the cardboard of the
crates resonating using the trapped air behind each bump as a
spring. To get a flatter response, boxes of different sizes would
be needed maybe by using crates made for both hens and
ostrich eggs...
Figure 7: The absorption of carpets varies greatly. The graph

shows the least and most absorbing carpets from tables in
acoustic engineering textbooks (after TJ Cox and P
DAntonio, Acoustic Absorbers & Diffusers, CRC Press,
2009).
A better homemade solution is to use old duvets. The ones

we tested were borrowed from colleagues at Salford University and had a mixture of tog values. As Figure 1 shows, the old
duvets tested had similar performance to the two flat acoustic foam samples of similar thickness. The foams did better from
500Hz and upwards, but then are considerably more expensive.
Choosing The Right Absorber
How would an acoustic engineer use the results from the product test to choose the right absorber? Take the example of
treating a smallish control room that is 4m wide by 4.5m long and 2.5m high. The amount of absorption has to be chosen so
the room is not too dead or too live: to put it in more scientific terms, it has
to have the right reverberation time. The reverberation time measures how
long it takes sound to decay by 60dB. A welltreated control room would
typically be expected to have a reverberation time between about 0.2 and
0.3 seconds across the frequency bandwidth that porous absorbers are
effective. Figure 9 shows a calculation of the reverberation time in the
example control room with different amounts of treatment. The target range
for the reverberation time is shaded in yellow.
It has been assumed that a carpet is being used. If a floor covering has to
be bought, purchasing a carpet and underlay that provides some absorption
will save money on other absorbents. The only way to be sure that the
carpet you buy is going to be effective is to find a manufacturer who
publishes absorption coefficients. The small amount of absorption from 10
square metres of diffusers in the room is also included in the calculation,
using the absorption coefficients published by a manufacturer.
When the room just contains the carpet and diffusers, the reverberation is
a long way above the target range, as the orange line in the graph shows.
Which porous absorber treatment should be used to fix this? The room is
only a little above the target frequency range at the highest frequency, so
sticking any of the three thinnest treatments in our tests to the wall
Profoam, AFW305 and Mercury Wedge 600 would not be the best
choice. If enough of these treatments were added to the room to reach the
target value at a midfrequency like 500Hz, then the
reverberation time would be below target at higher frequencies,
and the room would probably sound too dry in the treble. There
are solutions to this, such as spacing the thinner foam from the
wall, but a simpler approach is to choose a sample that creates a
flatter frequency response above 500Hz and can be attached
straight to the wall.
Any of the other samples tested could be used, and at this
point it might be worth considering the cost, appearance and
robustness of the various treatments. If the other products in the
manufacturers catalogues were also considered, then there is a
bewildering choice of colours, finishes and shapes. I would
always be guided by the absorption coefficients, but the testing
revealed some simple rules of thumb. Choosing a deeper
product increases the bandwidth of absorption to a lower
frequency, and although shaping a porous absorbent may
improve the appearance, if that alteration removes foam from the
part of the absorber furthest from the wall, it risks decreasing the
performance.
Lets suppose the mineralwool treatment with the greatest
absorption per square metre is used to take our hypothetical
controlroom design further. Adding 18 square metres of this
absorber brings the reverberation time within the target criteria
for 400Hz and above. That means 30 percent of the walls and
ceiling surfaces would be covered with the mineral-wool
absorbent. This would normally be distributed on the walls and
ceiling where the sound first reflects when going from a
loudspeaker to the sweet spot (once again, see Februarys guide
to controlroom design). This is because the porous absorption
has to both control the reverberation and reduce the coloration
caused by early arriving reflections.
Figure 8: Egg crates might be a traditional

acoustic treatment, but their absorption
coefficient is too uneven to make them a good
choice for studios (data from
www.acousticsfirst.com/docs/egg.pdf).
Figure 9: Adding treatment to a room to get within the desired

reverberation time range (shaded in yellow).
The structure of an opencell metal foam seen through a

scanning electron microscope.
Photo: SecretDisc
None of the treatments tested can get the reverberation time

within the design limits below 400Hz. For that, an absorber for lower frequencies is needed. Many companies sell blocks of
foam to treat low frequencies, but a better partner to the mineral-wool absorber is a membrane design that can create bass
absorption without large amounts of highfrequency attenuation. You could draw an analogy with a twoway loudspeaker: the
resonant absorber is the woofer treating the low frequencies and the porous absorber is the tweeter treating high
frequencies. As an example, 11 square metres of a commercial product was added to the design to reach the final curve
shown in green.
Trevor Cox is Professor of Acoustic Engineering at the University of Salford and author of Sonic Wonderland (ISBN
9781847922106). .
The Absorbers On Test

A number of manufacturers and distributors kindly loaned us samples from their product ranges for testing. It should be
pointed out that in most cases these companies offer a range of products, so for instance you would often have the option
of buying a thicker panel than the one tested, if necessary. Prices vary considerably according to the material, thickness
and construction: the cheapest product tested retails in the UK at 18 per square metre, while the most costly are over
60.
Profoam by RPG Europe
The melamine foam forms a wavy shape that ingeniously moves some of the absorbent away from the wall to create
additional absorption. However, even with that, the absorption is too low at midfrequencies, and so spacing further from
the wall or using multiple layers stacked on top of each other is needed. The absorption graph on the manufacturers web
site did not match our measurements because the product tested at Salford University was made from a different type of
foam.
www.rpgeurope.com
ColourPanel 60 Acoustic Foam Tile by EQ Acoustics
A 50mmthick acoustic foam absorbent with bevelled edges and a smart nylon fabric on the front to improve the
appearance. It took quite a search to find the absorption coefficients on the companys web site, but once found it was
reassuring to see a close match with the results we got.
The Auralex Sonoflat 22 performed a little better from 250
to 400 Hz.
www.eqacoustics.com
AFW305 by Pro Acoustic
A polyurethane acoustic foam made in the UK by
Comfortex Acoustics in Oldham. The absorption
coefficients were easy to find on the products web page
and matched the measurements made at the University of
Salford. The sample is not very deep, and the addition of
wedges means not much of the absorbent is furthest from
the wall. To get good performance youd need to space this
sample from the wall, or buy one of the companys deeper
products instead (see below).
www.acoustic-foam.co.uk
Block100 by Pro Acoustic
The thickest sample measured, almost twice as thick as
any other, and this is one reason it is amongst the best
three products tested. A polyurethane foam with acoustic
absorption coefficient data easy to find online. The profile
shape breaks up the panel so a treated wall does not
become a block of grey foam.
www.acoustic-foam.co.uk
RPG Absorbor
This is made from mineral wool that is edged with plastic
and fabric-wrapped in a range of finishes. The acoustic
data for this 50mm sample was easy to find online and
matched Salfords measurements. It was slightly better
than the other mineral-wool absorbers tested between 160
to 315 Hz.
www.rpgeurope.com
Universal Acoustics Mercury Wedge 600
A polyester foam with better performance from 500 to 1600
Hz than the similarly shaped AFW305. For most situations,
however, it will need spacing from the wall to give good
performance. Unfortunately, the web site just gives a single
NRC value as a measure of performance rather than the
full absorption coefficient.
www.universal-acoustics.com
Auralex Sonoflat 22
Makes a feature of being melaminefree. The foam can be
purchased in many different shapes, but Auralexs own
data shows that the simple flat design has the best acoustic
performance for a 50mm depth.
www.dolphinmusic.co.uk
www.auralex.com
GIK Acoustics Spot Panels
The Spot Panel is constructed with a solid wood frame and
fibrous material in the middle and is fabricwrapped. They
were among the best absorbers tested, although the RPG
Absorbor had a slight edge for 160 to 315 Hz.
gikacoustics.co.uk
Caveat Emptor
When buying an acoustic absorber, you should expect the seller to provide data for the absorption coefficient measured
using the international standard ISO 354:2003 (or the US equivalent ASTM C423). Measurements are done in a large
reverberation chamber the one at University of Salford is 220 square metres and has a lively acoustic like a large
church. When absorbing material is brought into this room, it deadens the sound. The measurement standard uses this
change in the room acoustic to quantify the absorbent via the reverberation time. Extremely loud noise is blasted into the
room, and then suddenly switched off. The reverberation time is then the time it takes sound to decay by 60dB. The
reverberation time is measured with the room empty, and then with 1012 square metres of absorbent present. The
absorption coefficient is calculated from the change in the reverberation time.
The most reliable data comes from an independent accredited laboratory like the University of Salford. Even if you dont
fully understand what is being measured, seeing a test certificate at least gives some assurance that the firm selling the
acoustic absorption is not just making up the numbers. Salford University has been testing sound absorption since 1965,
and for many years has had UKAS (United Kingdom Accreditation Service) accreditation for ISO 354 tests. This means
that UKAS carry out regular inspections to ensure the acoustic laboratories have the correct qualityassurance procedures
in place to ensure accurate measurement.
One of the problems with ISO 354 is the variation in results between different laboratories. Absorption coefficient values
for the same sample can vary by up to 0.2 between laboratories. What makes this Sound On Sound test unusual is that
the different samples have all been measured in the same reverberation chamber, making the results directly comparable.
Absorption coefficients greater than 1 are not unusual for porous absorbents. This happens because placing a large area
of absorbent in the reverberation chamber changes how sound propagates around the room. While the method has been
around for over five decades, no one has yet come up with a solution to this problem that is satisfactory to everyone.
I would only buy acoustic treatments from firms that provide absorption coefficient data. One site I looked at has a graph
of Relative Absorbance, whatever that might be, and just looking at the graph it is clear these are not measurements
from a reverberation chamber. Another thing I am wary of is companies just giving single-figure values such as the NRC
(Noise Reduction Coefficient). This is an average over the frequency ranges 250 to 2000 Hz. What you need to know for a
porous absorbent is the frequency on the graph at which the absorption flattens out. A final word of warning: many sites
boast testing of their products to an international standard, but many times the standard quoted is the one for fire
resistance and not for acoustic performance. Look out for ISO 354:2003 or ASTM C423, as these are the acoustic
standards.
How They Work

Porous absorbers generally remove sound energy in a similar way. Most of the absorption comes from the way the sound
waves squeeze through the labyrinth of tiny interconnected pores within the material. As the sound moves through the
pores, it causes air molecules to drag along the interior surfaces of the material. Air is a viscous fluid a bit like treacle,
only a lot less viscous! It takes effort to drag the air molecules across the interior surfaces, and this turns the vibrations of
the air molecules that make up the sound wave into heat that is then dissipated in the material. The vibration of the
material itself usually makes very little contribution to the absorption.
As absorption is caused by the oscillation of air within the pores, porous absorbers are not very effective when stuck on
a wall. The absorption is driven by the velocity at which the air molecules are vibrating back and forth, and close to a wall
the air molecules have nearly zero velocity. This is why hanging porous absorbent away from a wall will increase
absorption, because it puts the material in a place where the velocity of the air molecules is greater.
The electron microscope picture below shows the pores for an opencell foam. Good absorbents are about 95 percent
air, giving a large internal surface area to gain high absorption. The pores also need to be less than a millimetre in
diameter to get enough absorption. Normal foams that you might have in your sofa at home are closedcell (think of layers
of bubblewrap): the air pockets are not connected together and consequently they are relatively poor acoustic
absorbents.
Not all mineral wools are good acoustic absorbents, and density alone is not enough information to be sure of their
performance. A mineral wool such as glass fibre is made from woolly filaments that are glued together. Too much binder or
the wrong sized filaments and the absorption can be compromised. Mineral wools should only be bought if test data for the
absorption coefficient is known (see the Caveat Emptor box). There have been some concerns about the longterm
health effects of mineral-wool fibres, but the current evidence is that they are just irritants to the skin and not carcinogenic.
Summary
Porous absorbers for music studios are used to treat mid and highfrequency room-acoustic problems.
They are typically made from acoustic foam or mineral wool; either material can be effective, but for a given depth,
mineral wool performs a little better.
Treatments need to be at least 5cm thick, with thinner treatments needing spacing from the wall.
Shaped foam might look more attractive, but cutting away material generally reduces performance.
Only buy products where absorption coefficients measured using the standard ISO 354:2003 or ASTM C423 are given;
the gold standard is performance data measured and certified by an independent laboratory.
Foam and mineral wool on their own are ineffective for soundproofing, ie. stopping sound getting into or escaping from
studios.
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In this article:
Boz Digital Labs +10dB
Channel Strip
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Boz Digital Labs +10dB Channel Strip

Reviews : VST plug-ins
Hugh Robjohns

The ADR F760X-R Compex compressor/expander was a favourite dynamics processor during my
formative pro-audio career in the early 80s, and was unique and innovative in that it could limit, compress and expand all at
the same time, using a single FET gain-reduction element. Its gating technology was ahead of its time, too, and I loved its
fantastic versatility and the broad range of effects it could produce, while its quirky user interface intrigued rather than fazed
me.
Not surprisingly, the Compex became a mainstay of music production throughout the 70s and 80s, and is largely
responsible for many of the drum sounds on classic tracks of that era. Original units change hands for big money today, but a
new hardware reissue was released last year with a mechanical redesign to overcome the reliability issues associated with
original models (www.soundonsound.com/sos/feb14/articles/adr-compex-f760x-rs.htm). As far as I know, however, there
hasnt been a plug-in version of this amazingly versatile compressor until now.
The Boz Digital Labs +10dB compressor is available on its own, emulating the
F760X Compex, or with a four-band EQ module to exactly replicate the functionality
of the ADR F769X-R Vocal Stresser variant. The plug-in algorithms have been
modelled very closely on the hardware, and the project was aided by producer and
mixer David Bendeth who is an enthusiastic owner of several ADR processors.
Boz Digital Labs 10dB Channel Strip
Like most plug-ins, the +10dB compressor comes with a variety of factory presets, the ability to save more, and a very
handy comparison feature where the current settings are stored automatically into temporary A or B memories. Boz Digital
have made some minor changes from the original, adding a very useful mix control to permit parallel compression within the
plug-in itself, but removing the peak limiters pre-emphasis mode, which was often used as a very effective de-esser.
Other changes are subtle but potentially confusing. The order of the compressor and expander attack-time toggle positions
is different (but arguably more logical), and the colour-coding of the knobs has also been changed. ADR always used black for
Level, green for Release time-constants, yellow for Threshold, blue for Ratio and red for Range. Boz have used black, red,
green, blue and blue, respectively, and while this is obviously irrelevant for novices, it befuddled me!
An output level metering option wasnt provided on the original hardware, but is most welcome here. However, the meter
retains a right-hand-zero action, so loud peaks kick the meter down to the left just like the GR mode, which is quite odd!
0dBFS corresponds to 20 on the scale, while -10dBFS is around 8 and -16dBFS is about 4.
When purchased as a bundle with the EQ plug-in, the compressor can be placed before or after the equaliser, or the EQ
can be moved entirely into the dynamics side-chain for some very powerful effects (including versatile de-essing!), all exactly
like the ADR Vocal Stresser. Its worth noting here, too, that the EQ plug-in is unusual in having its four EQ bands arranged in
parallel rather than serial, giving it a different kind of band-interaction and a distinctive sound character. I found the +10dB
compressor worked just like the hardware version, both as a compressor and expander/gate, and emulated precisely the
same kind of attitude and character that made the F760X/F769X so popular in the first place. Its an impressively versatile and
very welcome plug-in, especially in the full bundle version. Highly recommended! Hugh Robjohns
+10dB Compressor $99; +10dB Equaliser $65; +10dB Bundle $199.
www.bozdigitallabs.com
.
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Big Fish Audio The Rock Collective

Multiformat
Reviews : Sample Shop
John Walden

Like most labels that can be hung on to musical styles, rock will mean different things to different
people. In the case of Big Fish Audios The Rock Collective, what you are getting is very much a slice of radiofriendly rock
that spans the stylistic territories from Train or Coldplay (although perhaps a bit upbeat for the latter) through to Paramore or
All Time Low.
The library can be purchased in either a combiWAV/REX/Apple Loops format or as a Kontaktready package. Either way,
what you get are 10 very well-stocked construction kits, each featuring multiple song sections intro, verse, chorus, outro,
etc so you can easily build a full arrangement. The instrumentation is pretty much what you would expect with stereo drum
loops, electric bass, acoustic guitars (including some 12 strings) and lots of electric guitar loops... Oh, and the occasional
piano, mandolin, tambourine and shaker thrown in for good
measure. These are uniformly well played and recorded. They
are also very well organised and labelled so, for example, within
each kit, sample names include a number (ie. to distinguish
guitar 1 from guitar 2) and a letter (A, B, C...) to identify which
song section they belong to. This makes it very easy to piece the
appropriate loops together.
In addition to the stereo drum mixes within the main construction
kits, you also get a further folder of multitrack drum loops, with
both processed and raw (as recorded) audio files. These include
kick, snare (top and bottom), hihat, tom, overhead and room
mics (room close and room far); if you want to create your own
drum mix then that is perfectly possible. Its great to have this
level of control, but it did leave me feeling greedy for DIed
versions of the various guitar parts as well. While whats here is
supplied with some excellent choices of guitar tones, being able
to dial in your own sound using your favourite guitar amp sim
plugin would obviously give you even more options.
The only other comment Id make is the usual one about having
to work within the confines of the construction-kit format;
somebody might build a chart hit from one of these kits before you do. That said, there really isnt a duff tune amongst this lot
and, with some suitably produced vocal parts, these kits could easily stand alongside most rock tracks on an MTV or Kerrang
playlist. The Rock Collective might not break too many boundaries in terms of musical styles, but it is a very polished and very
efficient option for busy music producers or media composers needing a slice of modern, radiofriendly rock. John Walden
$99.95
www.bigfishaudio.com
.
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In this article:
Overview
Upgraded Technology
Voicing
Opinions
In Use
Alternatives
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Cloud 44-A
Active Ribbon Microphone
Reviews : Microphone
Buy PDF
Cloud
44-A $2099
pros
Big ribbon sound.
Classic styling cues.
Music/Voice filter tempers
pronounced proximity effect.
Smooth and natural top-end.
Takes HF EQ nicely if
needed.
Ships with high-quality
shockmount.
Hand-built in the USA.
cons
The high-pass filter switch
looks like a DIY
modification.
No self-noise figure given.
High cost.
summary
A classic take on the original
long-ribbon design, but
updated with Neodymium
magnets and integral highquality gain stage.
information
$2099.
Cloud Microphones +1
888 365 3278
www.cloudmicrophones.comThis
classically styled mic incorporates Clouds own preamp design to negate some
of the pitfalls traditionally associated with ribbon microphones.
Hugh Robjohns
he Cloud Microphones story is an interesting one, the potted version being that it was set up in Tucson, Arizona by
Stephen Sank. His father, Jon Sank, worked at RCA with Harry Olson one of the leading lights in ribbon mic design
from the 1930s and 40s and subsequently went on to take over as the head of acoustical research there. Stephen
Sank set up a workshop repairing and refurbishing vintage ribbon mics with the knowledge and techniques taught by his
father. Next door was the project studio of Rodger J Cloud, for whom Sank refurbished a lot of vintage RCA mics, and
eventually the two decided to collaborate on the design of a brand-new microphone, hand-built in America, that improved
upon the classic RCA designs. SOS Editor In Chief Paul White reviewed the companys first model, the JRS 34P (and its
active sibling, the JRS 34), back in December 2011 (/sos/dec11/articles/cloud-mics.htm), commenting very favourably upon
the sound character.
The subject of this review is another closely related design, launched at the AES Exhibition in 2013 and serving as the
current flagship model. In clear homage to the original RCA microphone, the Cloud 44-A looks superficially the same as the
two preceding JRS 34 models, but there are some important differences...
Overview
Presented in a large and elegant hardwood box with a sculpted foam interior, the mic is protected from wind blast and stray
metal filings within a soft microfibre storage bag, and it is supplied with a slightly adapted Rycote InVision Studio shockmount
(the customisation being Cloud badge inserts and blue grips on the screw clamps). There is also an Allen key to adjust the
pivot friction on the mics built-in stand adapter, and I suppose there would normally be a user-guide too. Unfortunately, the
review model didnt have a user guide, and none is available for download from the manufacturers web site, but the
specifications and other pertinent details are available on the mics product page.
Measuring roughly 75mm wide, 45mm deep and almost 210mm high, the 44-A is quite a sizeable microphone, broadly
similar in silhouette to the RCA 44, yet much thinner and weighing only 400g. This feels quite wrong; big traditional-looking
ribbon mics are normally quite heavy! The aluminium and steel body of the 44-A is painted black, with the Cloud logo
identifying the nominal front surface although this mic has the traditional, symmetrical fig-8 polar pattern.
The ribbon element is protected behind a dual-layer windscreen with angled front and rear grilles, which dominate the
appearance of the mic with their bright silver finish. The microphones model number is engraved on one body side panel,
while its 48V phantom power requirement is engraved on the other as a reminder, adjacent to the off-centre XLR output
socket. A swivelling (90 degrees) stand-mounting bracket with a 5/8-inch thread is permanently attached to the centre of the
mic bodys baseplate. A silver toggle switch occupies the space on the opposite side of the stand adaptor from the XLR,
marked with the engraved letters M and V (of which more in a moment).
The microphones construction appears solid and reliable with, inevitably, more than a hint of classic retro styling, which is
quite attractive. However, some aspects of the assembly reflect its hand-built nature rather more than I feel is entirely
appropriate in a microphone costing as much as the 44-A does. In particular, the toggle switch and its visible shake-proof
washer make this feature look DIY modified rather than manufacturer designed.
Upgraded Technology
Apparently, the ribbon unit itself is handcrafted to the exact 1930 specifications of Harry F Olsons original RCA 44 design, but
with 21st century materials including a powerful neodymium magnet (which partially explains the low weight). This is a long
ribbon design, measuring 60mm in length (2.35 inches), 4mm wide (0.2 inches), and 1.8 microns thick, and the whole
assembly is suspended from an internal shockmount. Cloud Microphones provide a two-year warranty on the ribbon itself, and
a limited-lifetime warranty on all other parts of the microphone when it is registered with the company.
Another significant technological evolution in the 44-A is also
shared with the JRS 34 active ribbon microphone, and that is the
inclusion of an internal gain stage. The electronics comprise a
discrete J-FET amplifier derived directly from Clouds patented
Cloudlifter cascade preamp design, and bestows the 44-A with a
generous output level of around 18mV/Pa comparable with a
high-output capacitor mic. The specifications claim the maximum
SPL capability is more than 138dB SPL above 1kHz.
Voicing
The frequency response of the microphone in normal

mode...
The mysterious toggle switch marked V and M controls a highpass filter which dramatically reduces the mics low-end proximity
effect. The M position is the default (filter-bypassed) condition,
with M signifying Music. In this mode the frequency response
notionally extends between 20Hz and 20kHz, although it actually
starts to roll off above about 8kHz and is a good 10dB down by
20kHz a fairly classic vintage ribbon high-end characteristic.
...and in V mode, in which the mics low-frequency response
From about 1kHz downwards, the sensitivity just keeps on rising
is significantly reduced.
smoothly, reaching about +5dB by 20Hz. There is quite a
formidable proximity effect here, with considerable sensitivity to very low frequencies, too, so the inclusion of a decent
shockmount is a very welcome one.
With the mode switch in the V (for Voice) position, the top-end response stays the same, but the high-pass filter tames the
low-end proximity effect considerably, gently rolling off below about 250Hz at around 3dB/oct. The actual tonal balance is still
affected by the distance between mic and source, of course, but this switch does help to control the mics proximity effect,
which can be quite overpowering when used close to the source. Cloud Microphones claim the 44-A is, the first and only
active ribbon microphone with switchable Voice/Music response curves although there have been a few passive ribbons
over the years with similar facilities.
Of course, the V position is not restricted only to voice applications; its handy whenever a less powerful low-end output
would be beneficial, perhaps because of the tonal balance of the sound source itself, or because of placement proximity, or
just because of unwanted ambient subsonic energy.
Opinions
Although the built-in gain-stage requires phantom power, of course, it also protects the delicate ribbon generator from
phantom power accidents (which will give many users considerable peace of mind), and usefully minimises the gain, noise
and impedance requirements of the external preamp. The Cloudlifter cascade gain stage is a very elegant and effective
design, which is used to good effect here.
Tonally, the 44-A is a little darker-sounding than many more modern designs, which take more advantage of modern
technologies deliberately to extend the ribbons high-frequency response, but this is part of its charm and contributes to its
very natural character. The ribbon microphone market has expanded enormously over the last decade or so, and there is a
large number of very affordable and well-performing ribbon mics around now, but the 44-A combines classic design both
visually and sonically with high-tech developments, and is well worth auditioning.
In Use
The Cloud 44-A is a lovely, traditional-sounding ribbon microphone with a full-bodied low end and a clear, detailed mid-range.
The latter tends towards a useful mild forwardness when the filter is switched in, but entirely because of the psychoacoustic
effect of the reduced low-end, rather than any actual change of mid-range character. The treble response is smooth, open and
very natural-sounding. There is none of the edgy high-end emphasis or resonance so typical of large-diaphragm capacitor
mics, yet transients are portrayed with plenty of detail, with all the speed and clarity you would need, particularly on acoustic
guitars, pianos and percussion. And should you decide that some more air is required in the context of a busy mix, the 44-A
accepts high-shelf EQ lifts very nicely, without revealing any audible nasties. .
Alternatives
The UK retail price of the Cloud 44-A buys some very highly regarded ribbon mics, the most classic option being a
matched pair of Coles 4038s (with enough change to buy the elastic hangers, too!). Royers R122 and AEAs A840 active
ribbons are also both a little less expensive. If its the classic RCA styling that appeals, AEAs accurate R44C and R44CX
passive models cost more than twice as much, and the companys A440 active version nearly three times more, while
Cloud Microphones own JRS 34 model is around $400 less.
MICROPHONES For Sale in

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Audio-Technica
AT4047 MP
Multi-pattern
Condenser
Microphone
AudioTechnica
have added
multiple
polar patterns to one of
their already successful
designs, bringing
increased versatility in
the studio.
Audio-Technica
AT4047 MP |
Media
Multi-pattern
Condenser
Microphone
Audio files to
accompany the article.
Audio-Technica
AT4050 ST
Stereo Condenser
Microphone
There's
more to this
variation on
AudioTechnica's flagship
microphone than the
simple addition of a
second capsule...
Peavey Studio
Pro M2
Condenser
Microphone
Paul White
explores the
capabilities
of the
understated-yetpowerful Studio Pro M2.
Schoeps VSR5
Microphone Preamp
Schoeps
make some
of the most
revered mics
on the planet, so when
they release a
commercial version of
the mic preamp they
use for testing, you have
to take it seriously...
Schoeps VSR5
Mic Preamp
Test Measurements
The
following
charts, made
using an
Audio Precision
Analyser, accompany
our review of the
Schoeps VSR5
microphone
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In this article:
CPU Basics
Test Methodology
Intel U & M Series
Early Core i7 & AMD
Newer Core i7s
Turbo Power!
More Than Four Cores
Clock Speed Versus
Core Count
Decisions, Decisions...
Intel & AMD
CPUBenchmark
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CPU Performance Tested

The Need For Speed
Reviews : Computer
Buy PDF
Our benchmark tests show you which CPUs offer the best audio performance,
whether youre on a budget or moneys no object.
Pete Gardner
very few years, as resource-hungry software takes advantage of newer, more powerful computers, the same questions
arise about the computer that lies at the heart of your studio: do I need to upgrade this machine, or should I replace it
entirely? And what kind of improvement can I get for my money?
In this article, Ill help you to answer those questions by taking you through the results of some CPU performance tests I
carried out recently. I included a range of chips from the more cost-conscious to the fastest chips money can buy, and have
looked at both mobile and desktop versions, as well as new chips and some that are now three or four years old. The chart
makes clear which CPUs offer the best performance, but nothing about cost and certain other important considerations. So Ill
also explain in these pages a bit more about what makes a good CPU, and the implications of your CPU choice for the rest of
the system (in terms of heat, cooling and noise, for example).
Download the full resolution chart here
CPU Basics
The CPU (Central Processing Unit) is just one factor that influences your computers audio performance. There are others, not
covered here, including the type of storage used and the amount and speed of RAM, but the CPU plays a hugely important
part.
The days of being able to identify the best CPU by looking at the headline clock speed are long gone. Despite the clock
speeds not having increased dramatically in recent years, the processor manufacturers have continued to improve their chips
in other ways. (See the Intel & AMD box.) This means that when comparing different generations of similarly named chips, the
clock speed and core count may appear similar, but the real-world audio performance will vary significantly. When moving on
from a product thats three, four or perhaps even more years old to a new setup, its therefore very difficult to figure out what
sort of performance improvement your money will buy and its particularly difficult to establish at what point spending more
money takes you past the point of diminishing returns.
Test Methodology
I spent quite some time comparing a wide range of CPUs commonly used in audio machines, using a standard test, the freely
available DAWBench DSP Universal 2014 test. This reveals a lot about a CPUs suitability for audio production: although it
wont give the full picture in regards to ASIO performance, which can vary according to other system factors, this test is
designed specifically to stress test a computers CPU. DAWBench has been discussed many times in SOS and theres plenty
of information at www.dawbench.com but, essentially, it loads instances of a plug-in in a DAW, measures the performance and
generates a score. By restricting the tests to a standard plug-in (Cockoss ReaXComp multiband compressor, in this case),
DAWBench enables you to make meaningful comparisons in performance.
Its important to keep other factors as consistent as possible, so I used the same USB audio interface (Native Instruments
Komplete Audio 6) for all tests. Some expensive, better-performing interfaces may result in better performances overall, but
the important thing is to establish a stable baseline. (Its also worth mentioning that, while better interfaces are available, I
have established via group testing that this particular model offers a great performance/price ratio for new users wanting to
make music.) For the same reason, all tests were performed in an identical Windows 7 OS installation, and the same DAW
software (Reaper) was used: we can therefore be sure that the differences in performance shown on the chart arent due to
any differences in software.
Raien
Ingeniera
Ingeniera y
Sistemas. Lideres
En Tecnologa e
Ingeniera
I could have tested just the current generation of CPUs, but chose also to include legacy models because most users will
keep a dedicated audio computer running for three to five years before upgrading I hope the chart will give some indication
of how the performance has progressed over the last several years, and how newer but lower-spec CPUs compare with older
top-of-the-line ones.
As well see, another important take away from this test is that not all CPUs with the same nomenclature are created equal,
even if theyre of the same generation and have the same naming system this is particularly important to bear in mind when
moving from a mobile to a desktop setup, or vice versa.
Finally, these tests are inevitably a snapshot: it wont be too many months before a new bigger, badder, faster model comes
along, so perhaps well revisit this question when the next generation of chips becomes available. With all that in mind, lets
run through the results, from the bottom of the chart upwards...
Intel U & M Series
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First up, we have the Intel Core i3 4010U. Its not surprising to
see the U-series chips at the bottom of the chart: these chips
were initially conceived to power the ultrabook class of superthin, lightweight laptops, were designed to offer longer battery
life, and are closer in performance terms to the old Atom range
than the desktop CPUs that share their Core name.
Nonetheless, theyre also found in many regular low- and midrange full-size laptops as well as high-end tablets, including the
Microsoft Surface Pro 3. For simple multitrack recording or
editing of audio youve recorded on location they offer all you
need in a portable and inexpensive machine. If you wish to use
virtual instruments, effects chains or other processing, though, their performance will soon prove frustrating, and youd be
better off considering one of the more fully featured Intel models discussed later.
The next (small) step up is another popular mobile chip, the basic M-series. This includes the dual-core models found in the
current entry-level MacBooks. They outperform the U-class chips, but by a small margin: the performance levels are around
25 percent greater and so theyre not much better for audio work. Id suggest that anyone wanting an Apple laptop for
anything approaching serious studio work should find the funds for a MacBook Pro a refurbished or secondhand model
from a generation or two ago will probably outperform the current standard-issue MacBooks, and will also offer greater
connectivity.
Early Core i7 & AMD

Id hoped to test chips as far back as the Intel Core2 Quad Q6600. This offered the best bang-per-buck of its generation and
was regarded as the CPU of choice for studio use. Unfortunately, the OS build I used refused point blank to work with this
chipset, and to keep the tests consistent this model was excluded. Still, some useful data can be found on the
CPUBenchmark web site (see box), which suggests that it offers roughly 40 percent lower performance than the Core i7 930,
and only 20 percent higher than our bottom-of-chart CPU.
The Intel Core i7 930 and its family of processors ran on the X58-type motherboard, which saw a number of changes and
significant improvements to the chipset. This is the generation of advancement that saw Intel move the memory controller onto
the CPU die itself, as well as introduce the QPI (quick path interconnect) board design that emulated AMDs HyperTransport
bus design. By shortening the distance that the data had to travel before being processed, this yielded some significant
performance gains this was the first generation that saw Intel stride ahead of AMD in the performance race.
The next cluster of entries make for interesting comparison. The Core i7 4710MQ is possibly the most popular current
option for production laptops, with good reason. It offers the best price-to-performance ratio in Intels mobile CPU range:
theres a handful of better-performing models but they roughly double in price for each five-to-eight percent increase in
performance. Similar performance levels are offered by the AMD FX9370 and FX9590 (AMDs current flagship models) and
Intels Core i7 2600K, their consumer-level flagship in 2012.
As you can see, even today the i7 2600K is no slouch. Similarly, the Core i7 4710MQ currently powers many a laptop for
on-the-road producers and performers. For those wanting to use an audio machine on-the-hoof, these processors are not bad
by any means and should still hold appeal, but if youre thinking of a long-term investment in a studio machine, note how
startlingly different the benchmark score is between these and the highest-performing desktop chips!
The AMD chips cost less, as do related components, because the motherboards go through fewer socket changes over the
generations, which means a computer can be upgraded over a longer period of time. Theyre therefore a tempting proposition
if youre putting together your first dedicated setup on a budget. Their downside is that theyre rather power hungry, largely
because AMD have found themselves squeezing hard to get every last drop of performance out of a platform thats overdue a
redesign. For example, the FX9590 requires almost twice as much power than the current crop of high-end Intel chips. You
dont have to be an eco-warrior for this to be relevant: more power means more heat, which means more effective cooling
and, ultimately, a noisier system. If you can isolate your computer and ensure that any generated noise levels are taken care
of, the AMD models are well worth consideration but the limited budget that makes them attractive probably also means a
smaller space in which to record and mix, in which case the extra noise will be a most unwelcome guest!
The Core i7 3770K is another consumer-level flagship chip from a couple of generations ago. Above this we have the Core
i3 4370. You may be surprised to see an i3 chip included in the tests, as these are positioned as Intels budget-friendly lowend, low-cost CPUs and compete with AMD in the home-office market. Its place here is warranted, though, as the first of the
current generation: it shows us what to expect of the i3 range when lined up against other currently available products in the
range. Although perfect for the sort of role for which theyre intended, they tend to lack raw processing power when it comes to
much more than basic multitracking and editing duties in the studio. Also, AMD offer better-performing options in the same
price bracket.
The Core i5 4960 is a capable chip and a good entry point for those who are working on a budget and building a basic
studio computer. Again, at this price the AMD products enjoy a better performance-to-price ratio but at the cost of more
heat/noise; the i5 4960s power draw is around a third of the high-end AMD chips in the same price bracket.
Newer Core i7s

Moving up the chart, we come to one of the most popular current options for the home studio, the Core i7 4790. Ive included
both its low-power S and full-performance K series revisions. The S version of the chip has a total TDP (Thermal Design
Power, the maximum heat generated by the CPU that the cooling system is typically required to dissipate) of 65W. By
contrast, the TDP of AMDs flagship chip is 220W! This shows just how efficient Intel have made their CPUs over the last few
DAW Techniques
100s of great SOS articles!
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Digital Performer
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generations. Another model, below the K and S versions in performance terms, is the T edition. This has an even lower
TDP of 45W and is particularly well suited to smaller formfactor computers and in builds where passive cooling is desired (for
instance, to create a truly silent PC.)
The K edition is the highest-clocked version; it gains another 30 percent performance and still manages a respectable 88W
TDP its easy to build a low-noise machine around one of these. This, along with the competitive price, gives it amongst the
best price-to-performance ratio and its probably the most popular current choice for those wishing to do everything in the box
in a home-studio. It also means its a reasonable upgrade for users of the i7 930 (or previous) generation of CPU although
looking at the scores of the i7 2600K and above, it may not prove such good value as an upgrade path in the longer term.
Turbo Power!
Another key difference between the K chip and the standard one lies in how they Turbo the CPU. The Turbo feature on the
Intel CPUs allows them to run above the standard clock speed automatically when the system determines that theres the
overhead to do so without overheating or otherwise causing the CPU to become unstable. The Core i5 4690, for instance, is
advertised with a 3.5GHz base clock speed and a 3.9GHz Turbo clock speed, and these quoted figures are the same for both
the standard and K-series chips. The reason for a $15 difference in price is a different implementation of the Turbo feature.
The standard editions stagger the overclock across the four physical cores: in this instance, Turbo will overclock core 1 to
3.9GHz, core 2 to 3.8GHz, core 3 to 3.7GHz and Core 4 to only 3.6GHz. The K version, on the other hand, will overclock all
cores to the advertised limit, so all the cores can work at 3.9GHz resorting to traditional overclocking.
If we then compare the i7 4790K with its standard-edition counterpart, we see a 3.6GHz base clock with a 4GHz Turbo,
versus a 4GHz base clock and 4.4GHz Turbo the K version represents much better value for audio users given the small
increase in price. And thats before we even begin to consider the benefits of a gentle overclock to 4.6GHz or above, which
this chip is easily capable of.
More Than Four Cores

Our next group of chips includes Intels so-called Enthusiast CPUs and a couple of entries from its Xeon server class. The
Core i7 4960X was the flagship chip of the previous generation and is more powerful than the current consumer-class CPUs.
Being a six-core design (the mainstream ones all have four cores) this didnt come as a surprise. The same generations Core
i7 5820K has a similar specification and offers a small increase in performance but is unlikely to prove worthwhile to owners of
the previous generations six-core chips.
Intels current flagship desktop CPU is the eight-core Core i7
5960X. At its regular clock speed its more than a solid performer
but, like previous flagship chips, it doesnt offer great value for
money compared with other processors in its class. It performs
very well when overclocked, though, and it is, without a shadow
of doubt, the best pick if CPU performance is the only criterion
youre interested in.
Along with the extra cores and overall performance, this
chipset can support twice as many RAM DIMMs as previous
ones, making it a perfect choice for those creating large, detailed
arrangements using resource-intensive sample-based
instruments such as Vienna Symphonic Library. In fact, it is users
of such libraries (along with those who also do video work) who
are usually pushing the limits of what systems can do.
For sheer performance, the latest generation of Intel Core i7
This brings us nicely back to the Xeon server-class chips,

chips cannot be beaten.
which seem, at first glance, to hold so much promise. In
principle, the differences between Xeon chips and their i7-series siblings are actually few and far between, the chief one being
that Xeons benefit from an additional QPI link something which enables data sharing between the two or more processors
found in Xeon chips. Another is the sheer number of processors: there are already Xeon chips available with as many as 18
cores, although these are seriously expensive! The downside of the QPI link is that the chips remain locked down in other
words, overclocking isnt possible and theres extremely little scope to tweak for performance enhancement. The reason for
this is that these CPUs are designed for server applications, in which it is of critical importance that data sharing between the
cores remains stable. The Xeon chips with a high core count are also optimised for parallelisation.
Clock Speed Versus Core Count

Unfortunately, in tests of Xeon CPUs, my colleagues and I have tended to find that the higher the core count, the slower each
individual core becomes. With load balancing being handled by both the OS and your sequencer, each trying to squeeze the
most out of the available performance, some headroom is inevitably lost. In fact, fewer cores with higher clock speeds will
almost always yield better performance overall than a greater number of cores running more slowly. This goes some way to
explaining the strong showing by the higher speed-per-core overclocked i7 5960X over the slower-clocked but higher corecount Xeons in the test results.
But its actually slightly more complex than that, because Ive tested Xeon CPUs with a number of sequencers, and its clear
that the Xeon architecture has yet to be fully exploited by any DAW manufacturer. Multi-core support varies from one DAW to
another, and from one version of a DAW to another: some support a maximum of 16 cores, while a few can handle up to 40.
This upper limit includes so-called hyper-threaded cores, which mean in Intels case that there are twice as many virtual
cores as there are physical processors.
By way of example, a hyper-threaded 18-core Xeon CPU makes
36 virtual cores available to your OS, and installing two such CPUs
would give you a whopping 72 cores. Alas, as even the most
multicore-friendly DAW can access only the first 40 cores, the other
32 would simply be released for other purposes, or remain idle. On
the chart we can see this theory played out in the real world: the
Xeon E5 2620 v3, a six-core 2.4GHz (3.2GHz Turbo) model, is
comparable with the Xeon E5 2670 v3, a 12-core 2.3GHz (3.1GHz
Turbo) CPU. When testing each of these, I used a pair of chips, so
the E5 2670 v3s gave us a total of 48 threads (12 per chip makes
24, and hyperthreading doubles this to 48), and I witnessed the 40thread wall in practice. While this pair of chips is capable of better
performance than the chart shows, it requires a software developer
to take advantage of it before you can reap the rewards, in practice.
Of course, having eight spare threads for Windows and other

processes to use allows the DAW unimpeded access to the
processing cores that are assigned to it. Yet this system sits pretty
close to the top-of-consumer-range stock-clocked single-CPU eightcore offering... and the dual-Xeon solution costs roughly three times
the 5960X! In fact, you could quite easily build a complete system
around a Core i7 5960X for less than the cost of two of the Xeon
CPUs alone, without any other system components. Its safe to say,
then, that while some hugely impressive Xeon CPUs are available,
most users in a music-studio environment would get superior
performance for less money with one of the high-end single-CPU
setups. And if thats really not enough for you, youd probably be
better off adding another machine and using software like Vienna
Ensemble Pro to share the processing load across the two
computers.
Until very recently, the Xeon platform offered one significant
advantage over the X99/Core i7 systems: it could support a far
larger amount of memory than any other chipset. Windows 8 Pro 64The Xeon server-class chips offer as many as 36 virtual
bit can handle up to 512GB of RAM in theory, but the limit is lower in
cores per CPU, but DAW software has yet to be
practice simply due to the motherboards ability to host that much
designed to take advantage. For now, they remain an
RAM in the formats currently available. There are Xeon
expensive luxury, probably only of benefit to composers
who have reached the RAM ceiling of the current Core i7
motherboards that can handle up to 192GB of RAM, whereas even
systems but that itself is something that looks set to
the best enthusiast motherboards could at the time of testing only
change!
accommodate a maximum of 64GB. Just as we were going to press,
though, Corsair announced the availability of a (admittedly very
expensive) 128GB RAM package. The only users really likely to require this much RAM are those who are scoring for film
using the huge libraries made by firms like Vienna Symphonic Library, where some users templates can break the 64GB RAM
limit of most consumer-level hardware. The Xeon solutions can, for those who require the very best performance and are
willing to pay, offer a wonderful solution, but it looks very much like the Core i7 systems will soon be capable of similar
performance.
Decisions, Decisions...
So, after all this, what processor should you have in your next studio PC? If your current machine was a mid-range model
three or four years ago or a high-end one five years ago, or if youre looking to move up from a mobile setup to a dedicated
desktop machine, then most of the current desktop offerings from Intel will offer a very noticeable improvement. Looking at the
upper mid-range Intel solutions will, without a doubt, give you a solid increase in performance. Anyone working below the sixor eight-core Intel enthusiast models will currently also see a great performance increase by moving up to a system based
around these chips. On the other hand, owners of the previous-generation six-core i7s or older Xeon systems may find it hard
to see value in an upgrade below the current eight-core i7s or the high-end Xeon products, if memory expansion is right up
at the top of your list of requirements. .
Intel & AMD

For many years Intel and AMD were locked in a CPU arms race: newer, more powerful designs were brought to market
rapidly, and every five or six years a complete architecture redesign would propel one company ahead. Recent years have
seen a less aggressive release schedule, but theres still a constant stream of improvements and refinements that can
benefit studio users.
Intels last generational shift brought the first fully
redesigned and optimised i-series CPUs (Core i5, i7 and so
on). This family of chips has been with us since the turn of
the decade, but while the headline figures show only
modest improvements, the chips are becoming ever more
efficient and thus offer better performance. Intels so-called
Tick-Tock release and update schedule benefits studio
users. The Tick stage refers to the CPU range receiving a
die shrink, which allows for increases in performance and
energy efficiency through design refinement. The Tock
brings a new micro-architecture, with the introduction of
new features and functions. The full cycle completes once
every 12-18 months and means that we see significant
performance gains only every few years, but lower powerdraw editions in between, which give us quieter cooling.
AMDs base design predates Intels by a few years, and
they dont currently offer significant competition in the highperformance market. However, theyve carved out niches
with their well-established hardware and continue to hold
market share, mostly through reasonably performing chips
that compete strongly on price. Much of their recent sales
growth is due to their provision of chips for games
consoles, but theyve sensibly been using these contracts
to fund development of their next generation of hardware,
and making a lot of noise about the planned release of new platforms and chipsets in 2016. Can they re-establish
themselves as a provider of high-performance chips? It promises to be a very interesting year!
CPUBenchmark
CPUBenchmark is a web site that hosts a database of CPU test results. Results are updated daily, which means that even
without specific DAW benchmarking it should still be possible for you to get some idea of the degree of performance
improvement a proposed upgrade could deliver even for CPUs not featured in my own tests.
www.cpubenchmark.net
Fri 19 Jun 2015
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Dbx AFS2
Feedback Suppressor
Reviews : Processor
Buy PDF
HYBRID
STUDIOS
Dbx
AFS2 $300
pros
Good setup wizard for
inexperienced and
occasional users no
manual necessary.
Good live filter
discrimination between
music and feedback signal
components.
Very narrow filter widths
available.
Fast and reliable in
operation.
Flexible operating modes.
Great filter parameter
display screen.
cons
ORANGE COUNTY, CA
Is this Dbxs most userfriendly feedback suppressor to date?

Mike Crofts
remember when automatic feedback suppressors first appeared and were hailed (by some, including me) as a wonder of
their age. Imagine having an invisible, unpaid extra pair of ears and hands constantly monitoring the PA for feedback
If youre convinced you dont

problems and instantly applying selective EQ corrections as necessary, all without interrupting the performance or affecting
like using feedback
the overall sound quality... It all sounded too good to be true at the time, but it actually worked pretty well in many applications,
processors then you
and the detection algorithms have now evolved and become increasingly sophisticated
probably wont like this
either.
summary
The AFS2 is easy to use,
especially with the setup
wizard, and buys you
considerably more level
before feedback sets in,
without drastically altering
your sound.
information
$299.99.
Harman International +1
203 328 3500
www.harmanpro.com
www.dbxpro.com
The new Dbx AFS2 updates and replaces the wellestablished and successful AFS224, which has been around for a few
years now. As a user of the previous model myself I was interested to have a look at its replacement in the product line.
I like to have a feedback processor available for every job I do, in the same way as I like to have extra sets of batteries and
every kind of mains power adaptor I may not use them every time, or even all that often, but every so often a situation will
arise where that extra capability can make all the difference between a horrendous, stressfilled experience and a successful
result. Much has been, and will continue to be, written about the reasons for and against using feedback processors, but I
have always regarded them simply as automatic outboard EQ designed with a particular task the control of feedback in
mind. One of the main criticisms of such devices has been the fact that they alter the sound, and of course this is true
thats precisely what theyre for but they are designed to alter the sound so as to eliminate feedback issues, and as such
this has surely got to be a good thing.
With the AFS2 the team at Dbx have essentially taken the existing functionality of the AFS224 processor and developed
new detection and control algorithms, along with an enhanced degree of user control and the ability to save and recall
settings.
What It Does
In brief, the AFS2 has the ability to detect when feedback begins to occur at any particular frequency, and will insert a sharp
notch filter into the signal path at that frequency. As with most devices of this type, it works by assigning either fixed filters,
which are determined and applied during the settingup process and remain in place until manually reset, or live filters, which
are applied automatically during the performance if feedback is detected, without any action necessary on the users part. You
could think about the two types of filters as addressing different aspects of a live show, in that fixed filters are addressing
major issues relating to the venue and stage layout, and live filters will cope with anything that happens postsoundcheck,
such as increasing monitor levels, or physical changes like singers moving around close to monitors, or even taking the mic
out front so that the audience can assist with the choruses!
The huge advantages that the AFS2 offers over manually applied EQ including traditional graphic equalisers is that
the centre frequencies of the filters are calculated to be at exactly the required frequency. Also, the filter width can be very
narrow so as to apply attenuation at precisely those frequencies and not wipe out the good programme material either side.
Consider what a traditional thirdoctave filter actually covers in terms of a musical scale! The AFS2 can apply filters in three
width settings, according to what the audio material contains and/or the severity of the feedback problems being addressed.
The three types are described as speech (the widest setting), music (much narrower) or music/speech, which is of course
somewhere in between and offers a good compromise. Although the ability to apply very narrow filters is desirable in more
critical applications, its worth experimenting with the different settings as it may, for example, be better to apply one slightly
wider filter than use up several narrow ones close together. It all depends on the application, and its always worth finding out
what works best. The AFS2 does allow different filter types to be set for different filters if you change the setting, it will only
apply to new filters without changing any already set.
Its worth noting two other clever things about the AFS2 filter types: the fixed filters will automatically adjust their centre
frequencies and width during setup, and will only permanently fix their settings when you move on to live mode, and the live
filters have the ability to remove themselves from the signal path when no longer needed, thus freeing up filter slots for new
filters and restoring the original fidelity of the material. If all live filter slots have been used, and new feedback is detected, then
the AFS2 starts to reuse the live slots starting with the oldest, so its always going to address the current feedback issues no
matter what has gone before thats neatness for you. These parameters and more (such as the virtual high-pass function,
which leaves lower frequencies alone, if you wish) can be accessed via the AFS2 option menu, which allows control of most
aspects of the AFS2s operation.
Lets have a tour of the unit itself before getting on to using it in anger...
Racking Up
First off, the AFS2 is a wellbuilt unit, with a neat and sturdy steel single-rack case, and the front and back layout is practical,
simple and neat. There are cooling vents on top (above the power-supply area) and at both sides, but no fan. Although I didnt
mount the AFS2 in a rack, all I can say is that it didnt exhibit any signs of getting warm after several hours of use. Its quite a
shallow unit so Id want to mount it at least one unit below the top of a rack in order to leave some air space around it and be
able to easily access the connectors.
Feedback processors (suppressors, eliminators, destroyers whatever the makers prefer to call them) are functionally
fairly straightforward and have but one goal in life, which is to detect and control unwanted audio feedback. The connectivity,
display and control capability is therefore fairly uncomplicated and the AFS2 has a simple, focused appeal, which like the
very best managers provides everything you need and nothing you dont.
The back panel has left/right balanced input and output connectors, presented in both XLR and TRS jack formats, and
theres a switch for each input channel to select input sensitivity between +4dBu (pro) and 10dBV (consumer) levels. The
channel 1 inputs and outputs are grouped together, as are the channel 2 connectors, which is actually useful in itself theres
much less chance of accidentally swapping the channels (for example in a left/right main mix) when making connections in
low light or where access is restricted. Even if the connections have to be made completely blind (surely weve all found
ourselves groping around in the back of a rack at some point!) then all you have to do is start at the connector nearest one
end of the panel and work along in a sequence of in, out, in, out, and everything will be correctly channelled. Theres a
explained
standard IEC mains inlet at the other end from the audio I/O, but no on/off switch, so the unit will power up (and start passing
audio) as soon as its plugged in to a live supply.
A new feature of the AFS2, compared to its predecessor the
AFS224, is the provision of a USB socket for applying firmware
updates generally a comforting sign that the model will be
staying in the current product range for a good long time, as new
features and upgrades can be applied without replacing the
hardware.

Competitions!
The AFS2 has two inputs and two outputs, which can be
used independently or in linked stereo mode.
The front panel is neat, simple and effective, the traditional

rows of filter LEDs and dedicated button array complemented with a small LED display screen and rotary selector, similar in
size and style to those found on the 1U DriveRack models. A dual inputlevel meter indicates five levels from signal present
up to digital zero (max), with a clip segment above that. The two horizontal rows of red filter indication LEDs are large and
bright, and easily visible from several meters away. On this new model there is only a single bypass/clear button for each
channel, with all other parameters and preferences being accessed via the data wheel and screen. Bringing the AFS2 in line
with the DriveRack family is the inclusion of a new automated setup routine, and theres a big, inviting illuminated button
smack in the middle of the panel bearing the legend wizard, which might as well say press me, you know you want to.
Wave Your Wand

The setup wizard first asks if the unit is to be operated in linked mode, where the same settings and filters are applied to both
channels, or as two completely independent processors (for example when used on bus or channel inserts), in which case you
have to go through the wizard twice. Once that little formality is out of the way you can get on with the actual setup, which
guides you through every step and is very easy to follow. I like this feature as I know of several potential users of this unit who
would appreciate this level of handholding.
A quick gallop through the wizard process gives an insight into what the AFS2 is doing and how easy it is to use, so here
goes: after deciding to stereo link the two channels (or not), the next screen directs the user to perform soundcheck and set
up rough mix for all mics, which is obvious and sensible, and of course youd assume that you leave all the performers in
place if possible as this will affect the potential direct and reflected feedback paths. The wizard then reminds you to bypass
any active noise gates, and fully lower the main mixer outputs. The next step is to choose how many fixed and how many live
filters will be used, and to make things simple the wizard asks if it should set the default number of 12 fixed filters. Choosing
no allows you to dial in the required number so is only one extra step, and then the filter width is chosen to suit the
programme material and/or the severity of the potential feedback issues.
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Now its time to gradually raise the mixer output level, and the AFS2 catches the offending frequencies and assigns filters to
knock out the ringing as it starts to happen, and will assign up to the number of fixed filters you chose a moment ago, or you
can tell it to stop when youve achieved a reasonable level, even if all the fixed filters havent been used. When the fixed filters
have been set and confirmed as all done, the AFS2 switches on the live filters, which will swing into action when feedback is
detected from that point on. The live filters behave in a different way to the fixed set, and are designed to differentiate between
sustained musical notes and unwanted feedback. To demonstrate this, you can play a recorded track through the PA whilst the
fixed filters are in setup mode, and even at low levels with no open mics the AFS2 will start assigning fixed filters as it hears
the music programme. If you try this when the unit is in live filter mode, it wont mistake wanted musical content for unwanted
feedback (at least I couldnt make this happen) because it is using a much more sophisticated detection algorithm. Finally,
when youve completed the wizard, everything can be saved in the user preset library and recalled for future use.
Live Test
I connected the AFS2 into our rehearsal studio PA and went through the setup wizard with four open mics to see both how
effective the feedback reduction was, and what effect it had on the overall sound. I used the default setting of 12 fixed filters
and, as expected, I could set a significantly higher level in the room with the AFS2 than without it. I tried connecting the unit as
an insert and in line between mixer and powered speakers, and I found no noticeable difference in performance between the
two setups; the AFS2 seemed to be tolerant of varying input levels, although the manual sensibly recommends using inserts
as these will deliver a suitable and faderindependent drive so long as you have a couple of input-meter segments lit up it
should work happily. The increase in available volume was impressive, and even with all 12 fixed filters set the resultant sound
was fine. Using the wider speech filter option, playing recorded music through the system and switching in and out of bypass
mode will reveal some tonal differences, but these are slight and would be insignificant in a live situation where youre using
feedbackapproaching levels. What you have to remember about feedback suppressors is that, yes, they will alter the sound,
but thats exactly what you want them to do, and boy do they sound better than feedback! The other thing I noticed about
using the AFS2 was that even at volumes well below the feedback threshold, using it made the PA sound better, as all the
worst excesses of the room response had been filtered and the result was a cleaner, clearer output.
Our rehearsal hall for my 21piece band is a bit of a nightmare for live sound, especially with no audience in place, so I tried
the AFS2 on our vocal mics and was very impressed with the results. I used one channel of the AFS2 on mono mains and the
other on a single floor monitor just in front of the two singers, and not only was it possible to achieve much louder levels, but
by pushing the volume during setup to force the system to ring out the worst frequencies, the whole sound became more
Win Sontronics Aria

Valve Mic
Win Dynaudio & TC
Bundles
Monitoring System
pleasant, cleaner and less fatiguing. Its good to know that not only have the essential filters been set and will remain set, but
the AFS2 is ready to deal with anything that might crop up during rehearsal or performance.
A neat feature is the ability to look at the main parameters of all the filters in place a single press of the data wheel
displays a representation of where they all are across the whole frequency range, and their individual information is displayed
as each is selected with the wheel, so you can see exactly what the AFS2 is doing for you!
Overall Impression
Over the years I have tended to make much use of automatic feedback processors, as I find them a convenient and effective
solution in the varied nature of live work I take on, and Ive installed and specified them for a number of clients, who have
always been delighted with the results. Ive used the older Dbx AFS224 units on monitor buses and always found them to do a
great job, and my DriveRack 260 incorporates feedback suppression, which I dont hesitate to use if the situation requires it.
Im in the process of building up a new rig for larger indoor functions, which has to work for bands, conferences and A/V, and I
will seriously consider putting three AFS2 units in there for monitors the one thing my current digital mixer doesnt have is
feedback suppression! I know that not everyone swears by it, but its been my getoutofjailfree option on many occasions,
and I like the flexible way these new Dbx units work, and I certainly like the new user interface. Given that the Dbx AFS2 so
obviously does exactly what its supposed to automatically detect and suppress feedback without knocking a big hole in
your audio whats not to like? .
Alternatives
Various stand-alone products with similar overall functionality are produced by manufacturers including Sabine, Shure,
Peavey and Behringer.
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Fri 19 Jun 2015
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In this article:
First Impressions
On Test With Laptop &
iPad
Hidden Depths
Conclusion
Alternatives
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Information
WebExtras
IK Multimedia iRig Pads

MIDI Controller
Reviews : MIDI Controller
Buy PDF
IK Multimedia
iRig Pads $149
pros
Solid.
Full-size pad grid.
Works with iOS, Mac OS
and Windows.
Programmable.
Low bus-power
requirements.
cons
Relatively expensive.
No quick way to transpose
pad note range.
Vulnerable micro-USB
connector.
No Android support.
summary
Rugged, iOS-friendly MIDI
pads that are equally at
home with your laptop or
home studio.
information
$149.99
IK Multimedia +1 954
846 9101.
www.ikmultimedia.com
IK Multimedias iRig Pads brings portable control to your computer or iOS device.
Simon Sherbourne
or ages Ive had a craving for a small pad controller that I could use with my iPad, iPhone or laptop, so I was first in line
at the SOS review cupboard for IK Multimedias iRig Pads. The remit was simple: it should be small and light enough to
slip into a shoulder bag with an iPad or laptop, but still feel like a solid set of pads, not a toy or cheap gadget. The iRig
Pads measure up well against these key requirements.
First Impressions
IK have made the right decision by not miniaturising the pads. The 4x4 grid is in fact larger than that on my Maschine, but as
the pads take up most of the surface the overall unit is only a touch wider than my iPad 2. As well as the pad grid there are
Scene and Edit buttons, a further two programmable buttons and knobs, a slider and a rotary encoder with push-button. On
the back are two ports, one for connecting to iOS devices, and the other for connecting to a computer. Cables are provided for
both eventualities, although the iOS cable is only for newer lightning connectors, which was a blow as I have an iPad 2. An
optional cable can be ordered from IK for older iPhones and iPads, but this will set you back $24.99. The computer cable is
micro-USB, making it a bit easy to pop out or bend in fact I broke mine. Luckily this is a standard cable and cheaper to
replace.
On Test With Laptop & iPad

I tried the pads first with my Mac. The unit powered up with just a USB connection to a basic hub, which none of my other
controllers can do and is really useful on a Mac laptop! No driver installation was required, and the pads showed up straight
away in both Reason and Live. I was perfectly happy with the way the pads felt to play, so that was the most important thing
ticked off the list. All was great playing Reasons Kong drum machine, but in Live I quickly encountered a stumbling block:
transposing the grid to different note ranges is slow and fiddly. Unlike Kong, Lives Drum Rack instrument is not constrained to
16 sounds, and if you want to play non-drum instruments, an octave-and-a-bit range doesnt cut it. I hoped the two spare
buttons on iRig would be Octave Up/Down switches, but in fact to shift requires you to hold two buttons to go into Edit Mode,
use Pads 11 or 12 to shift up and down, then press another pad to confirm. This one feature significantly hampers playability,
and for me stood in the way of iRig being something you could use with your laptop as your sole controller in a mobile
composing rig.
As with the laptop, the iRig Pads were simple to hook up to my
iPhone 5S and worked straight away with CoreMIDI-compliant
apps. I tried with GarageBand and Xenon Groove Synthesizer.
IK Multimedia have a few apps of their own, so I installed
SampleTank Free and GrooveMaker Free. While the
SampleTank instruments for Mac and Windows (of which the SE
version is included with iRig Pads) are fairly traditional sampler
plug-ins, the iOS versions are more like mini grooveboxes or
workstations, with built-in multi-part sequencing. The iRig Pads
become an intelligently integrated controller for the apps, with
much more functionality than a generic MIDI controller.
Hidden Depths
Perhaps the most surprising thing about iRig Pads is how
programmable it is. While Id imagined the product to be mainly
designed for simple out-of-the-box fun with an iPad, the level of
customisable MIDI functionality shows IKs pro roots, and make
iRig Pads a powerful live performance controller and a generally
versatile studio tool. Sixteen recallable Scenes store complete
setups of the pads and other controls and all scenes are
completely programmable directly on the unit. Scenes 1 and 2
are pre-loaded with General MIDI Drum and Chromatic Note sets
and another four of the 16 scenes mimic Akai MPC banks and
work with the iMPC Pro app. As well as regular Note/Velocity
mode (on individually set MIDI channels), the pads can toggle
notes on and off, transmit MIDI CC commands, either
momentarily or toggled, and with fixed values or velocity-based
values. You can also switch mode to send a Program change
IKs SampleTank instrument under iRig Pads control.
from the pads in any scene. Furthermore, the push-buttons can
trigger notes as well as CCs and the Rotary encoder can work in
relative mode with user-adjustable sensitivity. Finally, the LED status and colour of each pad can be controlled from external
MIDI messages. By giving the iRig Pads all this functionality IK are providing a framework to enable detailed profiles to be
created for any DAW or app, and making it much more useful than a generic controller.
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explained
Just as I was finishing up my time with the iRig Pads, IK

released a software editor app for Mac and Windows, making it
very easy to edit all control assignments in your 16 scenes. The
app allows you to back up, load and store full configurations in
seconds, and copy between scenes. An option to set the default
octave for each scene was helpful, mitigating my complaint
about octave shifting somewhat, as Scene changing is relatively
fast and simple.

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Conclusion
There are quite a few alternatives to the iRig Pads, many of
which are significantly cheaper. However, with the iRig you get a
highly programmable full-size pad grid in the traditional 4x4
layout, with little compromise in feel and sensitivity, which could
easily do double duty in both the studio and in your laptop bag.
.
The dedicated iRig Pads editor software.
Alternatives
In the mobile pad controller market there are inexpensive offerings from Akai and Korg. You might also look at Akais MPC
Element and Fly, if youre looking for a mobile beat station. Arturias Beatstep also costs less than the iRig, and doubles as
a stand-alone sequencer. If you like the iRigs full-size pads, the most comparable competitor might be Akais MPD18, but
thats stretching the definition of portable a bit.
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In this article:
The Four Elements
Ring The Bell
Workflow &
Controllability
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Eiosis Air EQ Premium

Equaliser Plug-in For Mac OS & Windows
Buy PDF
Eiosis
Air EQ Premium $149
pros
Nine bands in total, with
dedicated shelf and filter
sections.
Unique Character and
Strength controls extend
flexibility dramatically.
Bespoke phase response of
the Earth shelving equaliser
is unique.
Frequency-finder mode is
particularly helpful and easy
to use.
Excellent FFT-style
analyser.
Configurable metering
modes, including RMS
difference to assess overall
loudness changes.
M/S and 5.1 modes with
individual bands selectable
to specific channels.
Both variable-Q and
constant-Q bell responses
available.
cons
You cant (yet) drag EQ
curves and parameters
within the graphical display.
Potentially confusing
boost/cut gain control
legend values displayed.
summary
Nine years after its original launch, Eiosis have rebooted their Air EQ plug-in with
some unique new features.
Hugh Robjohns
A nine-band equaliser with

he software company Eiosis is the brainchild of Fabrice Gabriel, who is also the co-founder and lead algorithm designer
an unusual degree of
of Slate Digital. Fabrice is a talented software designer with some very interesting ideas, and he isnt afraid to take a
configurability and versatility,
Air EQ Premium is
sideways look at things to come up with novel and powerful solutions which are always musical-sounding and easy to
particularly easy to use, with use this last aspect being equally as important as the sound to me.
some clever operational
aids and facilities, and is
Eiosis currently offer only one product, the Air EQ Premium plug-in, which is an unusually sophisticated nine-band equaliser
equipped with an excellent
available in both stereo and 5.1 configurations and primarily intended for mastering and mix-bus applications. If the name
FFT-style analyser and
rings a faint bell, theres a reason; it started out in life under the Eliosound brand, and we reviewed it in 2006:
adaptable metering.
information
$149
www.eiosis.com
Test Spec
Air EQ versions 1.0.22.6
and 1.106 (beta).
PC with 3.1GHz Intel i74770 CPU and 8GB RAM,
running Windows 7 64-bit
Home Premium.
www.soundonsound.com/sos/oct06/articles/plugin_1006.htm. It was something of an ugly duckling back then, but its grown
and matured into a beautiful swan now!
A 30-day, fully functional trial of Air EQ Premium is available, and love it or hate it it is licensed to a second-generation
iLok key, with two activations included in the download. All major native plug-in formats are supported, including AAX 64-bit,
and usefully, Air EQ is a zero-delay processor with a surprisingly low CPU overhead, making it very fast and responsive, with
minimal penalties when running multiple instances.
The plug-in interface is remarkably configurable, and while most of it is instantly familiar and intuitive, there are several
unusual elements here, too. By default, the top half of the plug-ins GUI window features an excellent graphical display which
can be set up to include input and output meters, an FFT-style frequency analyser, and the overall EQ frequency response
and individual EQ band curves (see Display Options box). The only minor drawback is that you cant manipulate the graphical
EQ curves directly: EQ parameters can only be controlled with the knobs, or by typing values into their adjacent numeric
fields. However, I gather curve interaction is planned for a future update. If youd rather just use your ears the entire display
section can be switched off, too, reducing screen real-estate considerably.
The lower GUI section carries all the control knobs, with input and output trim controls, five bands of parametric bell EQ
controls, high- and low-pass filters, and high and low shelving sections, all styled to resemble a 1970s hi-fi unit with a brushed
aluminium front panel.
The Four Elements

Unique to the new version of Air EQ is the patented Character control, an interesting feature which adds considerably to its
versatility. The Character sliders central Neutral position provides EQ responses typical of conventional equalisers. When
raised towards the Fire position, all active bands morph towards increasing gain and focus but without extending the timedomain (impulse) response. Usually, higher Q values result in more time-domain ringing, but not here. If the Character is
lowered towards Water, by contrast, the EQ bands morph towards a much broader, flat-topped shape which alters defined
portions of the spectrum without over-emphasising the area

around the centre frequency. In this mode the impulse responses
become much shorter, which Fabrice claims makes it sound
more transparent and natural.
By default, this Character parameter is applied uniformly
across every equaliser band but, if preferred, individual bands
can be isolated from the global control, and subsequently
adjusted independently for Character by shift-clicking on the Q
control.
A second slider labelled Strength is basically a wet/dry balance
control, allowing the employment of the EQ as a parallel
processing device. In essence, it allows the amount of applied
EQ to be tempered down to all dry, or extended up to 150
percent of the control settings. Again, although this is normally
active on a global basis, individual bands can be excluded. The
Character and Strength controls permit the overall effect of the
EQ to be shaped to an unusual degree, and I found them very
useful.
With two of the four basic elements already mentioned, youre
probably wondering where Earth and Air fit in. The answer is,
fairly logically, that these are the names given to a pair of
dedicated, fixed-frequency, low and high shelving equaliser
sections equipped with On-Off buttons and 18dB gain controls.
The Air control was present on the original Eliosound Air EQ, but
Earth is a new feature. Interestingly, these shelf curves arent
symmetrical in boost and cut, and the actual shelf shape
changes to some extent with the gain setting, too thanks,
Gabriel says, to many hours of critical listening during
development. For example, the Air EQ boosts from a slightly
higher frequency than it cuts, and when cutting the low end, the
Earth EQ introduces a slight boost in the region above the shelf
slope.
Fabrice claims the phase response of the Earth EQ section is
unique. It was apparently inspired by the characteristic lowfrequency phase-lag behaviour of audio transformers, and the
practical benefit is that the Earth shelving EQ tends to sound
tighter and less boomy than a traditional equaliser for the same
amplitude response. Certainly, when you compare the Earth EQ
(at its maximum boost) to other, more conventional equalisers,
its phase response is markedly different, remaining close to 0
degrees over most of the spectrum before increasing the phase
lag at low frequencies. Combining maximum Earth boost with the
12dB/octave high-pass filter allows you to apply a more
moderate amount of low-frequency boost while retaining these
desirable phase characteristics.
This Audio Precision test plot illustrates the EQ response

curves for the Neutral (green), Water (blue) and Fire (red)
Characters, with gain control settings of 1, 2, 3, 6, 12 and 18
dB, with a Q of 1. The enhanced gain and slightly tighter Q of
the Fire mode can be seen in comparison to the Neutral
mode, as can the wider and flat-topped response of the
Water setting. Note that the actual peak amplitude of each
curve rarely tallies with the displayed gain-control setting!
explained
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The upper AP plot compares the phase response with the

Earth EQ set to maximum boost (lower purple line), with the
phase response of a conventional equaliser providing a very
similar frequency-amplitude response. Maintaining the phase
close to zero, before tailing off in a heavy phase-lag at very
low frequencies is rare in modern equalisers, but is
characteristic of transformers and some inductor-based EQs.
The lower plot compares the phase response of the Earth
shelving EQ at different boost levels of 3, 6, 9, 12, 15, and 18
dB. The general shape and phase characteristics at very low
frequencies remain, but the phase lag is less pronounced at
lower gain settings, and there is some phase lead in the 20150 Hz region.
Ring The Bell

All of the seven remaining bands have their own on/off buttons,
and theres also a global in/out button to allow easy assessment
of the applied EQ. Input and output trim controls allow precise
level matching and gain-structure optimisation, and a polarity
reversal button can be configured to swap the polarity of
individual channels, too.
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A pair of conventional high- and low-cut filters is supplemented

with five bell-response parametric bands, the first and last of which
can also be converted to shelving equalisers. The filter slopes are
switchable in five pairs, starting with 6-12dB/octave, then 12-18, 1224, 18-24 and 24-36dB/oct. Both filter sections have their own Q or
bandwidth controls (ranging between 0.35 and 4) such that turning
the Q controls anticlockwise from the centre default reduces the
filter slope, while clockwise introduces a progressively more
resonant peak at the turnover frequency. The turnover frequencies
of both filters can be adjusted over an enormous range, with the
low-cut spanning 5Hz to 10kHz, and the high-cut 20Hz to 30kHz.
The five bell-section centre-frequencies can be placed anywhere
from 10Hz to 30kHz, with a gain range of 18dB, which can be
applied continuously or configured in 0.25, 0.5, 1 or 2 dB
increments. In its normal bell mode, the Q can be adjusted from a
very broad 0.1 to a narrow 7, and the resulting curve is a
conventional variable-Q shape, meaning that the skirt gets broader
as the gain (or attenuation) is increased. By right-clicking, however,
This compilation of impulse responses illustrates how the
each bell section can be switched independently into a steep bell
Neutral settings IR has a similar duration to a
mode where the nominal Q values range from about 7 up to
conventional equaliser (in this case the UAD Oxford EQ
plug-in). Usually, a higher Q would result in a longer IR,
something unspecified but incredibly narrow, and useful for the
but this is clearly not the case for the Air EQs Fire mode.
surgical removal of unwanted resonances or other noises. In this
The Water mode results in an IR which is roughly half the
mode, the curves follow a constant-Q response, where the skirt
duration of the Neutral mode, and the manufacturer
claims this shorter IR is responsible for this modes clarity
width remains constant regardless of cut/boost, which is a more
and transparency.
useful arrangement when applying surgical correction. Confusingly,
perhaps, the Q control legends still indicate the standard-mode
number range rather than the much higher Q values actually being applied, but Im told this is because changing them causes
significant complications with automation.

Another unusual aspect of Air EQ is that the law of the gain control is quite different from more conventional designs.
Fabrice has arranged it to provide a linear change in loudness rather than peak amplitude when adjusted on its own, and also
to maintain constant loudness when the Q is adjusted. This latter behaviour felt rather odd to me at first but, after using it for a
while, I think this arrangement does make it easier to hear the effect of a changing EQ bandwidth without being fooled by the
normal change in loudness. The inherent complication with this approach, though, is that although the graphical display
reveals the actual frequency-amplitude response outcome, the decibel values displayed below the gain controls are usually
considerably adrift from the reality.
For example, a bell boost centred at 500Hz with a Q of 1 and a gain of +3dB actually raised the signal peak amplitude by
6dB, and a boost of 6dB produced a peak 9dB higher (see image). The BBC engineer in me found these blatant discrepancies
disturbing at first, but the solution was simply to look at the graphical display or rely upon my ears, and ignore the gain
numbers completely!
As I mentioned earlier, the first and last bell sections can be converted into shelving equalisers, and in this mode the Q
range runs from 0.35 to 3. The lower end of the range produces a traditional shelf, but higher settings introduce an S-shaped
response where, for example, a low-shelf boost includes a modest cut just above the turnover frequency. This S-shaped
response is characteristic of classic designs like the Pultec, and helps to give better definition and precision in the EQ effect.
Unlike the fixed-frequency Earth and Air shelves, the turnover frequencies of these switchable bell-shelf equalisers can be
adjusted anywhere from 10Hz to 30kHz.
Workflow & Controllability

Air EQ is not only versatile in terms of the responses of its EQ bands: the same attention to detail is present in the user
interface, and this is a major strength of the product. For example, holding down the Ctrl (Windows) or Command (Mac) key
when dragging any parameter activates a fine-adjust mode, making critical tuning much easier. Similarly, holding down the
Shift key when adjusting the frequency parameter automatically engages the frequency finder mode which boosts the gain
and Q temporarily to help track down a problematic resonance something that always seems quicker and easier to do on
hardware EQs than plug-ins, until now! An alternative configuration setting causes this Shift function to solo the band currently
being adjusted, instead, making the efficacy of the EQ setting even more obvious.
Double-clicking or Alt/Option-clicking resets controls to the default value, and the whole EQ can be reset in an instant, too.
Or you can just reset the gain controls, or just the active (or inactive) bands... Another handy facility is a button to rearrange
the bands control values into ascending frequency order, which is very useful should you end up employing the right-hand
bands for low-end EQ duties and vice versa! However, the potential trap here is that reordering swaps the control values
around the band controls, and that will mess up any automation associated with the specific controls!
I love the ability to switch the frequency control legends from Hertz to musical note values, and Q numbers to octave
bandwidths. Its even possible to have all the control values displayed permanently, or only when adjusting a control, or never!
And the graphical display part of the GUI can also be toggled on and off by Ctrl/Command-clicking on the control panel
background.
Extending the Air EQs capabilities even further, the stereo version can be switched between left/right or Mid/Sides modes,
and while the EQ bands normally affect both channels simultaneously, each can be switched independently to affect only one
channel, if required a particularly useful facility when using the Mid/Sides mode, of course. The 5.1 version allows similarly
independent selections of the front left-right channels, the left and right surrounds, the centre channel, and the LFE
although the surrounds, centre or LFE channels can also be individually linked to follow the front left-right EQ settings, if
required.
Im not a fan of presets on processors because I think people rely on them without understanding, and so routinely fail to
optimise the settings appropriately. However, I have to admit that the 37 presets supplied here are actually quite well thought
out, particularly in taking advantage of the ability to name each EQ band. In this way, the different presets effectively detail
what each bands controls will do (the frequency and Q settings are preset with appropriate values, but the gains are left at
zero for the user to adjust). For example, four of the kick drum presets bands are labelled Sub, Impact, Attack and Click,
giving a good idea how they will affect the sound, while the mastering correction preset hilariously offers EQ bands to unSub,
unBoom, unMud, unAcid, unSharp and unThin!
Impressions
Air EQ is more configurable and versatile than most other software equalisers Ive come across, and is remarkably quick and
easy to use. These attributes, along with the low CPU overhead and instant reaction to control inputs, have made it one of my
preferred tools, not just for mix-bus and mastering roles, but for individual source EQ applications too. The sound quality is
particularly transparent and sweet-sounding to my ears, although a little familiarity is required to get the most out of Air EQs
capabilities, especially in learning how the gain and Q controls interact, and how to temper the full Earth boost with the highpass filter to take advantage of the useful phase-lag characteristic.
I really liked the frequency finder mode, which made Air EQ a joy to use, and the Water character was very useful too,
especially in mastering roles. In contrast, the Fire position seemed just to make each band a bit louder, which was not
something I found much need for on a global basis. The ability to operate in M/S mode and to apply different bands to alter
different channels is another real bonus (as are the similar channel-selection facilities in the 5.1 version).
The choice of equalisers can be quite a personal thing, as much for the way they respond to control inputs as for their
sound character. With Air EQ, I found it very easy to achieve the tonal modifications I sought, and the considerable versatility
and flexibility of its EQ parameters ensured that I could achieve everything I wanted in the one unit whether surgically
correcting resonances, sculpting a mix-bus, or gently polishing the spectrum while mastering. The FFT analyser is very
informative, and the EQ display is clear and precise (and more accurate than the controls legends!). I was also impressed
with the responsiveness of Fabrice and his team in developing the product. For example, the original review version (1.0.22.6)
only provided 6 and 12 dB/octave filter slopes, which I thought inadequate. After discussing this with Fabrice, an updated beta
version (1.106, as described here) was quickly supplied, and this should have evolved into a downloadable version by the
time you read this review. The download price is very reasonable for a plug-in of this quality and unusual versatility, and I
thoroughly recommend taking advantage of the 30-day free trial but do read the handbook first to appreciate how to use Air
EQ to its fullest extent.
.
Alternatives
There are countless EQ plug-ins, but none with Air EQs unique Character morphing or Earth shelf-EQ phase response,
and few that can match its configurability, extreme flexibility and excellent usability.
Display Options
Most of the elements of Air EQs graphical display can be customised. For example, an overall EQ response can be
shown overlaid upon the coloured curves identifying the influence of individual EQ bands. The amplitude scale of the
response graph can also be adjusted anywhere between 1dB and 24dB, making it easy to see whats going on even
when making very subtle mastering-style adjustments. The input/output meters can be switched on or off, and scaled in
dBFS or any of the three K-metering alignments. Moreover, the input meter can show an RMS differential which displays
the difference between the average volume of the input and output signals. This makes it very easy to adjust the input or
output gain trim controls to compensate for any loudness difference introduced by applying the EQ although it begs the
question: if the meters know what the level difference is between input and output, why not include an automatic gaincompensation option?
The superimposed real-time analyser employs a customised variation of the typical FFT display which is claimed to be
more musically relevant and easier to interpret. The analysed signal can be the raw input, the effect of the current EQ on
the input, or the actual output signal spectrum (the latter revealing the effect of the input and output level trim controls).
Furthermore, the FFT response can be switched between fast or slow analysis, and with infinite or 10-second averaging.
The whole analyser display can even be dragged up or down in the screen to aid visibility of the EQ response curves.
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Drawmer 1973
Threeband FET Compressor
Reviews : Processor
Buy PDF
Drawmer
1973 $1649
pros
Analogue multiband
compression is rare, but
very well executed here.
Considerable versatility,
thanks to wide control
ranges.
Very effective programme
dependent recovery options.
Useful hot level mode on
VU meters.
Wet/dry mix control makes
parallel processing easy.
cons
I couldnt find any!
summary
This is a costeffective
multiband analogue FET
compressor, with wide
control ranges and superb
audio performance, making
it a worthy, yet more
affordable, successor to
Drawmers S3 valve multiband compressor.
information
Its not easy to create an analogue multiband compressor that competes on quality
and price, but Drawmer show us it can be done.
Hugh Robjohns
rawmer have been making audio processors for over 30 years, yet their first product, the DS201 noise gate, set a
standard that others still try to match today. Alongside an impressive current list of around 30 hardware and four
software products, the Drawmer web site lists 18 discontinued designs, including the DC2476 digital mastering
processor (http://sosm.ag/drawmer-masterflow), the S3 multiband stereo tube compressor and the ThreeSum bandsplitter
(http://sosm.ag/drawmer-threesum). I singled these particular products out for special mention, since theyre the conceptual
ancestors of the subject of this review, the ingenious 1973 threeband stereo FET compressor.
$1649.
TransAudio Group +1
702 365 5155
Multiband Heritage
sales@transaudiogroup.com
For
mastering,
I often use the fabulous DC2476 Masterflow in preference to a stack of plugins. Being able to target different
www.transaudiogroup.com
dynamics processing on precisely defined spectral elements within a mix is an incredibly powerful tool, and Id be quite lost
without it! The S3 compressor translated the same multiband dynamics concept into the analogue domain with a no
compromise design that combined three complete optical compressors in a large 3U rackmount box, housing 10 valves and
an electronic oven to maintain the three lightdependent resistors at the optimum temperature for calibration stability. The
ThreeSum repackaged the S3s bandsplitting and recombining electronics into a compact 1U device to allow users to
integrate any combination of their own external dynamics processors.
The idea of multiband processing remains popular in the digital world but, expensive mastering processors aside, it seems
less popular in the analogue domain, probably in part due to the cost of production. The S3 and ThreeSum were
discontinued some time ago, but Drawmers new 1973 is, essentially, a more costeffective version of the S3 (which Drawmer
tell me was an unanticipated hit with European dancemusic producers!). As well as clearly representing the ideas and
practical benefits of the aforementioned products, the 1973 derives some operational aspects and styling cues from still
popular classic Drawmer compressors, such as the 1960 and 1968.
Design & Construction

As the name makes clear, the 1973 employs FETs for the gainadjusting elements its a completely new gain-cell design
not used on any of Ivor Drawmers previous compressors. The active crossover filters and gain stages are built around
NE5532 and TL072 opamps. Despite the lack of tubes, the overall signal path design and operational controls are very
similar indeed to that of the S3.
Housed in a 2U rackmounting steel case extending about 250mm behind the rack ears, and with an aluminium front panel,
the 1973 is instantly recognisable as a Drawmer product, thanks mostly to its whiteonblack control legends and graphics
combined with bright yellow pointers on the smart fluted Sifam knobs. A pair of large, illuminated VU meters dominates the
righthand side above the output controls, while the three compressor sections are laid out in a wonderfully ergonomic and
intuitive way across the rest of the panel.
The rear panel features four XLRs for the electronically balanced stereo line inputs and outputs, along with the ubiquitous
IEC mains inlet and a voltage selector switch (115/250 V AC). Internally, a compact toroidal transformer feeds a linear power
supply section, hosted on the righthand side of a large printed circuit board that covers the entire chassis floor. Two daughter
PCBs carry the front-panel controls and gainreduction meter electronics, with all the circuitry being constructed using
conventionally sized components. Sealed relays are used for bypass and bandmuting functions, and I counted no fewer than
twenty separate trimmers, too, for calibrating the various gain stages and side-chains!
Operating the 1973 is straightforward, and aided considerably by the clear and logical layout of the front-panel controls.
Each of the three independent compressor sections is dominated by its large centrally placed threshold knob, which spans a
wide signallevel range of +26 to 32 dBu. Immediately below the threshold control is a trio of smaller rotary controls to set
the attack, release and makeup gain. A threeway toggleswitch is also provided to mute or bypass each individual
compressor section (orange and red LEDs indicate the mute and bypass status, respectively). This facility makes it easy to
check what each section is contributing to the overall dynamic control of the input signal.
The attack and release controls are both sixway rotary switches, with the attack range spanning a very fast 0.2 to 50 ms,
and the release offers three fixed timeconstants of 80ms, 300ms or 1s. There are also three programmedependent
(automatic) releasetime options, labelled F (fast), M (medium) and S (slow), and these can range between 0.10.5 seconds,
0.32 seconds, or 0.55 seconds, respectively. The makeup-gain control is another conventional rotary pot, this time
covering a gain range of 10 to +20 dB, and sitting immediately above these three smaller rotary controls is a horizontal bar
graph meter of red LEDs. This indicates the sections current level of gain reduction, and the meter can display up to 20dB
gain reduction.
All of the rotary controls described so far have black caps, but
sitting between the three dynamics sections, and positioned in
line with the threshold controls, are two greycapped rotary
controls. These are made distinct from the others because they
adjust the crossover frequencies of the multiband filters. The
panel graphics neatly separate the three compressor sections,
with these crossover filter controls sitting obviously between the
bands. The crossover filters have gentle 6dB/octave slopes, with
the low/mid crossover being adjustable between 60Hz and
1.4kHz, and the mid/high crossover spanning 1.4 to 14 kHz.
Stereo in, stereo out: all the bandsplitting and band
combining shenanigans are taken care of internally.

explained
Moving now to the righthand side of the front panel, just
below the meters is a master output level control (with a 12dB range) and a wet/dry mix control to cater for parallel
processing effects. A pair of toggleswitches to the left of the output controls configures the VU meters between normal or fast
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transient peakreading modes, and pad the 0VU level by 10dB to make it easier to view the levels when working with very hot
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signals (so 0VU equates to +14dBu instead of +4dBu). A second pair of toggleswitches on the right-hand side provide a full
Valve Mic
relay hardbypass of the entire unit (with red status LED), and turn the mains power on and off (also with a red status LED).
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Two more toggleswitches, one each in the low and highband compressor sections, are located to the left of the
respective bands threshold controls. The lowband switch is marked Big, and this engages a 100Hz, 6dB/octave high-pass
filter in the side-chain which reduces the band compressors sensitivity to very low frequencies. This makes it less enthusiastic
about clamping down on very lowfrequency components (kickdrum hits, for example), with the result that the very low end
becomes subjectively bigger and more powerful while the upper bass is still being dynamically constrained. Its exactly the
same kind of sidechain filtering facility thats found on many conventional fullband compressors, and it works in the same
way here. The equivalent highband compressor switch is labelled Air, but instead of employing a sidechain filter, this
reintroduces some of the original signals HF back to the output. The idea is to restore some of the air and transient clarity
thats often lost during compression. The wet/dry control has a similar effect, of course, but that affects all bands, whereas the
Air switch just affects signals in the highband section.
In Use
Multiband compression is one of those techniques that many people love, and just as many seem to hate! Its inherently a
more complicated way of controlling dynamics than conventional fullband compression, of course, and that probably puts
some people off. On the other hand, the multiband approach allows a very precise and targeted form of dynamic control, with
almost complete freedom from many of the more common sideeffects of conventional fullband compression.
Some conventional fullband compressors offer several
different sidechain filter and EQ responses to help focus which
parts of the spectrum the compressor is required to react to the
most (or the least), and many compressors have external side
chain access points to allow elaborate external equalisation to be
employed. However, while sidechain EQ can emphasise or de
emphasise certain frequency regions and thus influence the
compressor to apply more or less gain reduction to signals in
those different regions, when gain reduction is applied it inherently alters the level of the entire signal spectrum.
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In contrast, a multiband compressor allows separate parts of the audio spectrum to be processed entirely independently.
This allows different attack and release times, different ratios and amounts of compression, and different makeup gains, the
last of which can serve as a form of relative EQ, too. This extra level of versatility and finesse can make a surprising
difference, particularly when processing a complete stereo mix or stereo loop. For example, highfrequency transients can be
controlled precisely with a high ratio and fast attack and release characteristics, while lowfrequency signals can have slower
release times to prevent the compressor pumping or trying to track the LF waveform, and the midrange can be polished with
a gentler ratio and a slower attack to let snare transients really punch through.
Checking out whether the crossover points are dividing the spectrum in the most useful way, what each compressor section
is doing, and whether they are correctly optimised is easy with the ability to mute or bypass each band. Similarly, the three
gainreduction meters make it clear whats going on moment by moment, and the output meters and level overall control allow
proper gain structuring. The +10dB mode is very handy when using the 1973 as an external processor in conjunction with a
DAW, when the mix levels could be close to the digital peak.
To that end, my Audio Precision bench tests revealed that the
1973 can cope with +26dBu on its inputs, while returning up to
+25dBu at the outputs, so its perfectly suited to working with
highend converters calibrated for 0dBFS = +24dBu. My
measurements broadly matched the published specifications,
with channel crosstalk around 88dB at 10kHz and THD+N at
+4dBu of 0.06 percent, increasing to around 0.45 percent at
+20dBu. The signaltonoise ratio (ref +4dBu) was 88dB (A
weighted, 22Hz22kHz), while the frequency response was
3dB at 7Hz and somewhere in excess of 80kHz.The flatness of
the overall response is critically dependent on the gain make-up
settings of the individual sections.
Verdict
The 1973 threeband stereo FET compressor is a welldesigned
product, which combines three excellent FET compressors with
An Audio Precision plot showing the overall frequency

response (black line), and the contributions of each of the
three separate compressor bands with the crossover filters
set to 200Hz and 3kHz. The very mild midrange lift evident
in the plot is due to the midband makeup gain control
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versatile control ranges and effective crossover filtering. It allows
a level of precision and flexibility in programme dynamic control,
which is unobtainable with fullband compressors, and very costeffectively. If youre looking for something a little different in
outboard compression, the 1973 warrants a serious audition. .
Alternatives
While the likes of Tube-Tech and Maselec make highend mastering multi-band compressors, these have a high price tag,
and more modestly priced multi-band compressors are few and far between. SM Pro made one for the 500series format,
but since that companys acquisition by Harman Group its not clear that they have plans to continue production. If you
search hard, you might also be able to find a secondhand bargain like Focusrites Platinum MixMaster, which featured a
good threeband stereo compressor or indeed a secondhand Drawmer S3.
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Mackie Mix 12FX

Compact Mixer
Reviews : Mixer
Buy PDF
Mackie
Mix 12FX $160
pros
Elegant Mackie ergonomics.
Robust all-metal
construction.
Small footprint.
Internal digital effects
engine.
Remarkably low cost.
cons
Internal DC-DC power unit
gets surprisingly hot.
No facility to change effects
parameters.
External line-lump mains
transformer.
summary
Mackies new Mix 12FX
mixer offers streamlined but
useful facilities, with a
simple but effective digital
effects engine, in a
remarkably affordable and
robust package.
information
$159.99.
Mackie +1 425 892 6500
www.mackie.com
Mackies mixers have earned a reputation for offering excellent value for money. We
check out their most affordable range to date.
Hugh Robjohns
ackie have been dominant in the compact and cost-effective mixer market for 25 years now, and while the companys
small-budget mixers may not offer the very last word in technical excellence, they are always well-designed, with a
good feature set, and very usable. Mackie say that their brand-new Mix series, launched at the end of 2014, is the
most affordable mixer range the company have ever produced, and the Mix 12FX reviewed here is the largest in this new
family. Its smaller siblings are the Mix 5 and Mix 8, with the Mix 5 featuring a single mic/line input plus two stereo line inputs
(ie. five inputs in total), while the Mix 8 expands on that to offer two mic/line, two stereo line, and a stereo tape input. However,
the Mix 12FX ups the ante considerably, with four mono mic/line and four stereo line inputs, a stereo tape input, and an
internal digital effects engine.
The Mix 12FX has a very low profile, measuring 297 x 244 x 53 mm (WHD) and weighing in at just 1.7kg, the weight being
largely due to its all-metal chassis. The only rear panel connector is a mini-DIN which accepts a dual 9V AC supply from a
line-lump transformer unit. Everything else is connected on the mixer top panel, where you can see it, with XLRs for the four
microphone inputs, and TRS sockets for almost everything else. All of the line inputs can be used with balanced or
unbalanced sources, of course, and similarly all of the outputs (apart from the stereo headphone socket and tape outputs) are
wired as impedance-balanced sources, which means they can be used to feed either balanced or unbalanced destinations
without level changes. Phantom power is available for all four mic inputs via a global switch near the console meters.
Although only supplied with a very simple Quick Start Guide for the whole Mix series, rather than a model-specific full
manual, I think the guide provides more than enough information to get most users up and running. However, a full manual is
available as a download, with extra operational details, descriptions of the effects programs, block diagrams and a more
complete set of specifications.
Signal Path
Working through the Mix 12FXs signal path, there isnt much to confuse anyone. The four mono mic/line inputs each have a
rotary gain control providing 0 to 51 dB for mic inputs or -15 to +32 dB for line inputs. The maximum mic input level is +19dBu,
which is plenty, and +21dBu on the line inputs (which are padded down and routed through the mic preamp for the mono
mic/line channels, as is normal for mixers of this kind). Perversely, the unity mark for the line-input mode actually provided
+9dB of gain (with the channel and main faders at their unity marks).
There are no channel insert facilities, so the next item in the signal path is a push button below the EQ section which
introduces a very useful third-order (18dB/oct) high-pass filter with a 75Hz turnover frequency. A three-band EQ is provided on
all four mic/line channels, comprising high (12kHz) and low (80Hz) shelf equalisers, plus a fixed-frequency (2.5kHz) bell
section, all with 15dB ranges. There is no EQ bypass button, but all three gain knobs have centre detents.
Routing to the outputs is taken care of via the last three knobs in each channel strip. The first is a post-fader effect send,
which feeds both the internal effects engine and a mono physical output a very useful feature enabling the use of an
external effects processor. The effect send control has a unity-gain position marked at the centre detent, and 14dB of extra
gain available when fully clockwise. There is no effects master level control, so whatever level is sent to the effects bus is what
appears at the effects send output. A pan knob is also provided, with a centre detent, and an adjacent overload LED
illuminates when the channel signal is within 3dB of clipping. The channel fader is another rotary control, and although the
centre position is marked for unity gain there is no centre detent. The Mix 12FX is not equipped with channel mute or PFL
buttons.
The four stereo line input channels are much simpler, with no
EQ or gain controls, but they do have push buttons to change
the nominal input level between +4dBu and -10dBV (which is
roughly -8dBu). With the channel and main faders at their unitygain marks, and the sensitivity switch in the +4dBu position, the
output level is the same as the input level, and switching to the
-10dBV mode introduces the correct 11.8dB of additional gain.
The stereo channels two input sockets are cross-linked such
that plugging into the left socket only routes the signal to both left
and right outputs. In place of the mono channels pan controls,
the stereo line channels have balance knobs, and a stereo rotary
fader is provided. A mono sum is used for the post-fader effects
sends.
Outputs
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Here you can see the effects of all the EQ gain knobs at their
maximum and minimum settings, as well as the response of
the 75Hz high-pass filter.
On the output side, the stereo mix bus passes through the main
mix fader a real short-throw (60mm) fader, this time and on to the main output TRS sockets, the RCA/phono tape output
sockets, the meters, and the monitor section. The meter bargraphs comprise just four LEDs which illuminate at output levels
of -20, 0, +6 and overload the first two being green, with orange for +6dBu and red for overload (at +18dBu). The mixer
outputs clip at +21dBu, so the red overload lights warn when there is less than 3dB headroom left. I was surprised to find that
the tape outputs provided the same output level as the main outputs, rather than a lower level for better compatibility with
-10dBV standards. In practice this wont be a problem provided care is taken setting the external units input level.
The monitor section receives the main mix by default, but a push-button switches to the tape input RCA/phono sockets, if
required. A rotary control adjusts the monitor level feeding both the physical TRS output sockets (labelled CR Out), and the
stereo headphone socket. A second push button in this section routes the tape inputs to the Main Mix, but there is no level
control.
Setting the Mix 12FX apart from its siblings is, of course, the internal digital effects engine, which automatically receives any
channel signals routed to the mono effect send bus. A 12-way rotary switch selects the effect type and a rotary control adjusts
the (wet) output level feeding into the main mix. A red/green LED indicates signal presence and overload. The 12 effects
options start with eight different reverbs (small stage, small room, large room, warm hall, bright hall, classic plate, bright plate
and vocal plate), but there are also chorus, flanger, slap-back and space echo effects. These are all preset effects with no
facilities to change any parameters at all, but they are all generally quite usable and well chosen and if you really want
something very specific you can always turn down the internal engine level and hook up an external effects unit instead.
The first three reverbs mostly add ambience and early reflections, rather than overt reverberation, while the two halls are
more obviously reverberant, and the vocal plate has a rather longer decay time than the classic and bright plates. The space
echo mode has a longer delay time than the slap-back echo, and includes regenerative feedback.
Hands On
Bench tests broadly confirmed the published specifications. I measured the microphone amplifiers EIN figure at -121dB,
which is about 10dB noisier than the theoretical limit, but perfectly adequate for the kind of applications for which this compact
mixer is intended. The THD+N figure was fractionally higher than published, but not enough to worry about. Phantom power
measured on the low side, just within specification at 44.6V, but with plenty of current available and no sign of sagging when
all four channels were powering mics.
The Mix 12FX is well built with a robust and solid feel to everything, so it should prove reliable. However, I noticed that the
internal DC-DC power converter (located in the top right-hand corner of the unit) generates quite a lot of heat. You can feel
warmth around the headphone socket, but the bottom panel gets positively hot enough to make me almost drop the unit
when I reached underneath to pick it up! Although there are ventilation slots underneath and holes in the side panels, they all
appear to be covered over.
Mackies new Mix 12FX might not be the most comprehensively equipped or specified mixer, but the facilities that are
provided are very well thought out, appropriate for the intended applications, and it sounds good. Given the feature set, the
robust construction, the compact footprint, and the extraordinarily low asking price, the Mix 12FX is a very attractive problemsolving product ideally suited to small PA jobs and band rehearsals, or desktop mixing for home studio setups.
.
Alternatives
There are countless compact budget mixers on the market, but the Mackie brand has always represented reliable quality
and design elegance, and the new Mix Series are the most affordable compact mixers ever offered by the company.
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Nemphasis X7
Tube Overdrive Pedal
Reviews : Effects
Paul White

While an overdrive pedal might not seem like an obvious studio accessory (other than for conventional
guitar recording using an amplifier), Ive often found that using an analogue overdrive pedal before a software modelling
amp/speaker produces more musical results than relying on digitally modelled overdrive. Overdrive is actually quite difficult to
model accurately, as new frequencies are generated that can fall outside the Nyquist limit for the converters and that can, in
turn, cause unpleasant aliasing. The better designs get around this by increasing the sample rate within the plug-in, but at the
end of the day theyre still chasing the analogue sound so why not just use the real thing?
Italian in origin, the Nemphasis X7 Tube Overdrive is an analogue pedal that uses a valve as the distortion-generating
element, and in this case its an EEC83/12AX7 dual-triode valve. The pedals folded metal case is fitted with simple gain, tone
and level knobs, and the footswitch operates a clunk-free hard-bypass mode. A red LED shows when the pedal is active and
connection is via jacks on the sides of the case. Power comes from an external, Boss-compatible 9V adaptor (350mA
maximum current draw) but this isnt supplied with the pedal as standard. Presumably, the company would like you to consider
one of their own pedalboard power supplies, which are amongst the most comprehensive Ive seen; the top model in the
range offers various output voltage, isolation between outputs and even adjustable battery sag.
In use, the tone control goes from dark to quite bright, so it
doesnt sound like just a simple top-cut affair. Drive takes us from
a bluesy edge all the way to classic rock and beyond, but what I
like about this pedal is that the valve biasing has been finely
judged so that the characteristic asymmetry of triode distortion
isnt overplayed to the point that offensive low-frequency
intermodulation distortion becomes evident. In this respect, I
found that the overdrive had something of a Marshall DSL quality
to it.
Theres enough output level on offer to overdrive the front end
of an amplifier if desired, but its important to appreciate that,
when using it with a DAW, overdriving the input stage of the
audio interface is unlikely to produce particularly musical results.
In that scenario, you should set the level to leave enough
headroom to keep you well away from clipping. Other than that,
you can plug the pedal into a line or instrument input, though
note that if you bypass the pedal the clean sound will only be
correct if you go into an instrument input, as this will have the necessary high impedance.
Best of all is that, despite its strong performance, the X7 costs no more than many non-esoteric analogue pedals. It works
brilliantly for giving a clean amp what amounts to a separate overdrive channel, and studio hermits will be pleased that it plays
nicely with DAWs. Paul White
USA price TBA (about $120, based on exchange rates when going to press).
www.nemphasis.com
.
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The K47/49 Capsule
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Neumann U47 FET

Capacitor Microphone
Reviews : Microphone
Buy PDF
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STUDIOS
Neumann
U47 FET $3999
pros
ORANGE COUNTY, CA
Accurate in every detail.
cons
Expensive but likely to
maintain its value alongside
originals.
summary
A genuinely precise
reproduction of the classic
U47 FET, built from the
original plans, and available
as a limitedrun Collectors
Edition.
information
$3999.
Neumann USA +1 860
434 9190
www.neumann.com
Nearly 30 years after production of the original U47 FET ceased, Neumann have
revived this iconic large-diaphragm capacitor mic.
Hugh Robjohns
he Neumann company have always been known for pouring their considerable resources into advancing microphone
technology. Each new Neumann model has brought some worthwhile technical improvement, incrementing the industry
forward. For example, the original U47 was the first largediaphragm mic with a switchable polar pattern (1949), the
KM84 was the first ever phantompowered microphone (1966), and the TLM170 was the first massmarket transformerless
microphone (1993). Neumann also introduced the first digital microphone, the SolutionD, in 2003. Of course, not everyone
appreciates these technical advances as evidenced by the great KM84 vs KM184 debate but thats a different matter!
However, for only the second time to my knowledge (the other case being a limited edition U67 reissue in 1993), Neumann
have uncharacteristically decided to revisit their past by recommencing the manufacture of a 46-yearold design which
ceased production some 28 years ago. As a result we can now for a short time, anyway purchase a brand-new
Collectors Edition U47 FET.
After 17 years of continuous production, the last U47 FET was built in 1986, by which time the U87 had evolved into the
U87Ai as the default studio microphone of choice, and the TLM170i had become the companys flagship model. So why
break with tradition and reintroduce the U47 FET? Well, Neumanns press release for this reissued mic reflects on the fact that
it played a significant role in shaping the sound of popular music in the 70s and beyond, and that it is still a very common
choice in todays digital audio world. Perhaps it also has something to do with the burgeoning market for classic microphones
and the remarkable prices people are willing to pay! The Collectors Edition tag certainly suggests that, but whatever the
underlying reason, Neumann are now building the U47 FET again, which is surely something to celebrate.
Importantly, the Collectors Edition U47 FET is manufactured directly from the original production documents and
schematics, so this is not some kind of faux reissue with redesigned and modernised internals; Neumann genuinely have
resumed production of the microphone as it was being manufactured from 1980. This is an important caveat because the
original U47 FET went through a number of manufacturing variations over its production lifetime (as all models tend to do).
Consequently, there are a few very minor variances to accommodate modern production requirements, and Ill detail some of
those in a moment, but rest assured that in every way that matters, this new U47 FET is exactly the same as the old one.
History
The U47 FET was conceived in 1969, the driving impetus being that Telefunken had stopped production of the VF14 valve
that was employed as the impedance converter in the original U47 microphone. With the rapid advances being made in the
late 1960s in the quality and capability of silicon transistors, Neumann decided to reengineer the U47 as a solidstate,
single-pattern microphone, using the recently perfected fieldeffect transistor (FET) to provide the highimpedance interface
with the capsule. The resulting microphone, which entered mainstream production in early 1972, was called the U47 FET,
although it is often referred to simply as the fet47. The original model employed a Tuchel output connector, but that was
changed to an XLR and the model rebranded the U47 FET i. The internal construction also changed slightly at the same time,
with a plastic inner chassis instead of the traditional brass-plate-and-pillars assembly.
A whole generation of solidstate microphones using this new FET technology was introduced under the family name of the
FET80 range. This included such revered microphones as the KM84 (and its siblings, the KM83 and KM85), the KM86, the
U89, the nowlegendary U87, the stereo USM69 and, of course, the U47 FET. Interestingly, the U47 FETs impedance
converter circuitry was far more complex than any of the other mics in the range, and this complexity was only shared with the
U89 (and some KMR shotgun models). In most FET80 microphones the FET impedance converter feeds the output
transformer directly, but in the U47 FET (and U89), the FET buffer is followed by a further five bipolarjunction transistors,
which essentially form a discrete opamp to drive the output transformer. A sixth transistor serves as a voltage regulator to
ensure a stable 43V power rail from the phantom supply.
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Although the FET model incorporated exactly the same K47/49 capsule (see box) as the original U47, the revised body and
grille metalwork, and especially the radically reinvented internal electronics, all contributed to a different, rather less 3D and
organic sound character. Nevertheless, the U47 FET became a highly regarded and immensely popular mic in its own right:
different from an original 47, of course, but a useful and reliable workhorse all the same.
One of its particular strengths was an ability to cope with extremely high sound-pressure levels up to 147dB SPL with the
10dB pad switched in (for comparison, the original U47 was getting pretty crunchy by 120dB SPL). The FET 47s high SPLhandling capability quickly made it a firm favourite for capturing kick drums, bass instruments, guitar amps, brass and many
other demanding instrumental applications. Although never as popular as a vocal mic as the original U47 it sounds a little
more closedin and flat than the original 47 (which has a much broader 5dB presence boost) the FET 47 was still quite
usable in this role too, particularly for male vocals and speech. Its modest 2dB of presence boost between about 2 and 5 kHz
is enough to aid clarity and intelligibility in a mix, while the gentle proximity boost adds a degree of body and warmth.
Collectors Edition
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Supplied in a traditional dark wood case with a sculpted foam interior and a brass model badge on the top surface, this new
Collectors Edition microphone ships with its operating manual inside a cardboard wallet, alongside an individual serial
numbered certificate signed by Wolfgang Fraissinet, President of Neumann GmbH. The review model was serial number
10083, but I believe standard production units are all over 10100.
For anyone not familiar with the U47 FET, I think it is fairly described as a chunky microphone, measuring 160mm high by
63mm in diameter, and weighing a substantial 710 grams. The new microphone is only available with the traditional satin
nickel finish, whereas the original could be obtained with either matte black or nickel finishes (although Ive never seen a black
one!).
An integral standmount is included, comprising a simple arm permanently attached and pivoted from the side of the mic
body. The standadaptor thread is 5/8inch, but the mic ships with an adaptor for both 3/8inch and 1/2inch standard mic
stand threads. Usefully, the base of the stand adaptor also incorporates a grip for mic cables up to 5mm in diameter (secured
with a thumbwheel), which helps to isolate mechanical vibrations travelling along the cable.
Neumann describe the U47 FET as a fixedpattern cardioid microphone, although it actually has a distinctly hypercardioid
response for all frequencies above about 500Hz. The nominal sensitivity is a very modest 8mV/Pa, and this can be further
reduced in two ways. The gain of the internal electronics can be attenuated by 10dB through a recessed slideswitch on the
rear of the mic. Internally, this increases the amount of negative feedback around the FET, thus reducing the risk of frontend
overload. In addition, the output transformer can be reconfigured with a slideswitch at the base of the mic, rearranging the
transformer secondary windings in parallel rather than series to decrease the output level by 6dB. The aim is to reduce the
risk of overloading the subsequent console or preamp, but this option was removed from the original FET 47 for the last
couple of years of its production run.
Selfnoise is given as 18dBA, which, combined with the normal maximum SPL rating of 137dBA, gives an impressive
potential dynamic range (Aweighted) of 119dB, rising to 129dB with the 10dB attenuation switched in. The microphone
requires standard 48V phantom power, but has an extremely low current requirement of just 0.5mA. A switchable highpass
filter is also provided, via another rearpanel slideswitch, to help control the mics proximity effect. This raises the low
frequency rolloff point from 40 to 140 Hz.
Differences
To the casual eye, the Collectors Edition and original (post-1980) U47 FET microphone are absolutely identical as they
should be given that Neumann have recommenced production rather than created a new version! The body is the exact
same size, shape and weight, with the same purple Neumann lozenge and the same threelayer (coarse, medium, fine) mesh
grille.
However, in comparison to early FET 47s, one immediately obvious indication is that the milled polarpattern symbol just
below the grille is inverted. The new Collectors Edition model has the pattern null facing downwards (as it did in the post1980
models), whereas it faces upwards on the earlier originals. The mounting arms threaded standadaptor section is slightly
smaller than early original models, too, again being the same revised design as employed on the post1980 mics (and the
TLM170, in fact).
Inside the mic every part with the exception of a new and improved version the of sealed slideswitch remains directly
interchangeable with the original model. Even the handwired rats nest of electronic components located around the
switches under the top circuit board is made in exactly the same way as the original was (in the precise form that it was being
built around 1972 apparently later editions employed a PCB for these components!).
However, there are a number of almost infinitesimally subtle constructional differences which would make a great U47 FET
nerd quiz! Examples include the knurled cable clamp wheel being plastic instead of metal, the detail of the output
transformers construction, the inner mesh layers weave density, and the off position of the new slide-switches. The last item
is because the new switches seem to have a shorter slide travel, resulting in the off position being noticeably closer to the
centre of the microphone cases switch slots than the bottom. Thankfully the on position is still unambiguously at the end of
the slot next to the legend, though. Another minor detail change from the early FET 47s is that the capsule has gained a
Audio-Technica
AT4047 MP
Multi-pattern
Condenser
Microphone
AudioTechnica
have added
multiple
polar patterns to one of
their already successful
designs, bringing
increased versatility in
the studio.
Audio-Technica
AT4047 MP |
Media
Multi-pattern
Condenser
Microphone
Audio files to
accompany the article.
Audio-Technica
AT4050 ST
Stereo Condenser
Microphone
There's
more to this
variation on
AudioTechnica's flagship
microphone than the
simple addition of a
second capsule...
Peavey Studio
Pro M2
Condenser
Microphone
Paul White
explores the
capabilities
of the
understated-yetpowerful Studio Pro M2.
Schoeps VSR5
Microphone Preamp
Schoeps
make some
of the most
revered mics
on the planet, so when
they release a
commercial version of
the mic preamp they
use for testing, you have
to take it seriously...
Schoeps VSR5
Mic Preamp
Test Measurements
The
following
charts, made
using an
Audio Precision
Analyser, accompany
our review of the
Schoeps VSR5
microphone
protective nylon collar across the top, providing some protection to the diaphragms in case the mic gets dropped and the
flexibly mounted capsule accidentally strikes the grille housing.
preamplifier.
The capsule assembly plugs into a goldplated sevenpin socket mounted on a PCB with the 2N3819 FET mounted all
alone on the top surface. A number of components are soldered directly to the underside of the socket and to the two slide
switches, which are supported on a metal frame. The remaining electronic components are accommodated on a circuit board
positioned below the output transformer, with the power regulator transistor and the four other audio transistors, as well as the
output level slideswitch. Again, this board is identical to the post1980 version of the U47 FET.
Handheld Condenser
Microphone
Designed as
a hand-held
live vocal
mic, this mic
has a cardioid pickup
pattern, and seems very
robustly engineered.
Conclusions
In every sensible interpretation of the term, Neumann really have recommenced production of the U47 FET, but it should be
noted that this new version isnt actually identical to any specific original model, because it contains the best elements of
several iterations drawn from across the 17year production run. The new model should, therefore, actually be better than any
original model and not least because of the far tighter manufacturing tolerances of current K47/49 capsules.
To my ears, the new U47 FET retains all the character (or lack of it, some might argue!) of the original, with that same
slightly twodimensional sound, contrasting with the pronounced threedimensionality of the progenitor U47. Personally, I
think Neumann made a mistake in sticking with the U47 moniker; it should have been christened the U57 or something to help
break the sonic link with the U47, in much the same way that the transistorised U67 became the U87.
That said, the U47 FET remains a very useful and soughtafter microphone and the new version is every bit as
accomplished as the original, with the benefit that it is available brand new for similar or less than a mintcondition original
currently changes hands (especially in the US). Given the total authenticity of the new models construction, that surely makes
it something of a bargain, even at its eyewatering price. And a small comfort is that its future value is likely to compare
directly with the original models if the reissued U67 valuations are anything to go by. Neumann have not given any clear
information on how many Collectors Edition mics it intends to produce, but its rumoured to be around 100 so get your order
in quickly!
.
Alternatives
There are several microphone homages based upon the Neumann U47 FET, including the Bees Neez Tribute 3 FET, the
Bock Audio iFet, the Wunder Audio CM7, and the Lawson L47FET. However, for considerably less money, the Audio
Technica AT4047 is designed to deliver the U47 FET character too, albeit with a more modern styling.
Blue Encore 300
Cartec EQP1A
Mono Valve Equaliser
British
'boutique'
outboard
manufacturers seem to
be rather thin on the
ground these days, but
if this Pultec clone is
anything to go by,
newcomers Cartec look
set to make a big
impression.
Prodipe TT1
Dynamic Microphone
Prodipe say
they wanted
to offer a
high-quality,
live-sound, cardioidpattern dynamic mic at a
very affordable price.
Sontronics
Saturn
The K47/49 Capsule

The double capsule number is a source of confusion to many. The K47 capsule design dates from 1958 and it was
intended as a replacement for the M7 capsule employed in the earliest U47 mics (amongst others). It is a centre
terminated dualdiaphragm capsule built around a shared central backplate electrode, with screweddown diaphragm
tensioning rings (the M7 diaphragms were glued on).
A major problem with this type of shared backplate construction is ensuring that both diaphragms have identical
performance, and thats why Neumanns later dualdiaphragm capsule designs were built in two identical halves, each
with their own backplate electrode, and then screwed together. This approach made construction much easier, and
allowed considerably better consistency in matching and performance between capsule halves. The first example of this
approach was the K67 capsule built for the U67 mic (using a Telefunken EF86 valve), which was itself intended as a direct
replacement for the U47 in 1960.
However, because of the matching consistency issues with the K47 design, Neumann started employing a second
model number to identify the very bestmatched capsules, which was K49. Consequently, these K49 capsules were
specified for use in switchable pattern mics like the U47, U48, M49 and so on, where front and back capsule matching was
critical to the overall performance.
Although it still uses a dualdiaphragm or BraunmhlWeber capsule, the U47 FET has a fixed cardioid pattern, so
front/back capsule matching was far less critical for this application. Neumann therefore denoted capsules with one perfect
side and one less wellmatched side as K47 FET capsules, specifically because they were only suitable for use in that
model.
Today Neumanns production tolerances for the K47 capsule which is still used in several current production mics
are so tight that the company no longer makes performance distinctions. Either the capsule is up to standard or it isnt
(there is just a single grade of capsule called the K47/49), and thats what is employed today in the M147 Tube, M149
Tube, TLM49, and now, once again, the U47 FET.
Multi-pattern
Condenser
Microphone
Sontronics
mics usually
sound as
distinctive as
they look - and this one
looks more distinctive
than most!
MXL Revelation
Multi-pattern Valve
Microphone
Hot on the
heels of the
impressive
Genesis
cardioid valve mic, MXL
have unveiled their
flagship multi-pattern
model, the Revelation.
Does it live up to its
name?
MXL Revelation
| Audio
Examples
Multi-pattern Valve
Microphone
These audio files
accompany the SOS
September 2010 review
of the MXL Revelation
microphone.
Samson Go Mic
USB Microphone
USB mics
are nothing
new, but the
Samson Go
Mic is probably the
smallest and cutest I've
seen to date. This
metal-bodied mic,...
Shure X2UUSB
Microphone
Preamplifier
AKG Perception
820
Valve Microphone
Does AKGs Chinesemade Perception 820
maintain the Austrian
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PSP MasterQ 2
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PSP MasterQ 2
Martin Walker

HYBRID
STUDIOS
ORANGE COUNTY, CA
When I reviewed PSPs original seven-band MasterQ equaliser plug-in way back in SOS June 2004, I was impressed with its
sweet oversampled sound and its incredible versatility. It has continued to find its way into my songs over the years, thanks
also to its array of limiting and soft-clipping algorithms, which prevent harsh digital clipping in favour of a selection of gentle
compression characteristics.
Tests have shown if you can exactly match their frequency response curves, many digital EQ plug-ins sound vanishingly
similar, so you might as well choose one with a graphic interface you find pleasing or most easy to use. The exceptions are
those that model some sort of analogue behaviour, such as the aforementioned soft clipping or low-level harmonic distortion
which brings us neatly to MasterQ 2. PSPs basic design remains the same, with a graphic display of the overall EQ curve
in the upper portion, and controls for up to five parametric EQ bands plus high-pass and low-pass filters beneath. However,
almost every aspect has been enhanced in some way or another.
For starters, apart from the usual 12 and 24 dB/octave options, its high-pass and low-pass filters now offer an extra-steep
36dB/octave slope, which is particularly handy for removing cone-flapping subsonic excursions or high-frequency aliasing.
The outer two parametric bands still offer a shelving option as well as the normal peak response, but all five peak bands also
now have a soft peak mode that flattens the tip of the EQ curve for a sweeter sound when you have to dial in severe
equalisation settings. Clicking on the plus symbol next to the Q label switches the EQ skirt to a different shape, more closely
matching that of some analogue EQs.
MasterQ 2 also features the frequency hunter mode first seen on PSPs flagship Neon EQ, which isolates the audible effect
of a chosen band to make it easier to home in on unwanted frequencies in your material, and the linked filters mode that
moves all band frequencies simultaneously, making it easy to tune them individually to different harmonics of the first
perfect for notching out hum-based interference, for instance. The EQ can be applied to the full stereo signal or just the left,
right, Mid or Sides signals, which, together with the new stereo width and balance controls, gives you much greater control
over the final stereo image.
You still get the same generous selection of seven limiting/saturation options, from the transparent VintLim to the more
obvious soft and hard-knee saturation, but the addition of the new Ceiling knob, gain-reduction meters, and automatic makeup gain option extends the usefulness of this section from clip-avoidance to many more creative applications, such as
overdrive and heavy pumping effects.
However, for me the most welcome feature is the new Analog section, which models the subtle harmonic contributions of
components such as driven transformers, the effects of which are felt most at lower frequencies. Using its Character knob,
you can dial in only odd harmonics or a combo of odd and even. You can also adjust the overall level of these harmonics, and
with the Analog button enabled, each active peaking band contributes its own mojo to the cumulative effect, so you can add
warmth even when the EQ is set totally flat.
By now, youve probably realised that Im a fan of PSPs MasterQ 2. I liked the previous version, but this one offers much
more, being both an extremely flexible mastering EQ with surgical precision when required, and a very flexible design tool for
more extreme purposes. The latency is still around 1100 samples (26ms at 44.kHz), so it might not be the ideal tracking tool,
but for mixing, sound design and mastering I love it to bits!
Estudio
Jurdico
$149
www.pspaudioware.com
Asesora y
Soluciones
Jurdicas.
Consulte por
Asesora Legal!
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Fri 19 Jun 2015
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In this article:
Case Study
Installation
Studio Tour
In The Field
Alternatives
Specifications
Compared
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PreSonus AudioBox iOne & iTwo

USB 2 Audio Interfaces For Mac, PC & iOS
Reviews : Audio Interface
Buy PDF
PreSonus
AudioBox iOne & iTwo
pros
Easy to use, with solid
construction and clean
signal paths.
Zero-latency software
monitoring.
Useful iPad integration: no
camera kit required and
easy transfer from Capture
app.
Capable entry-level Studio
One Artist DAW included.
Two simultaneous inputs
available even on the iOne
(mic plus instrument).
cons
Gradual drain of the iPad
battery when powered
through the interface.
Full Capture app is a paid
extra.
Headphone output could be
slightly louder.
A stereo option for direct
monitoring would have been
useful on the iTwo.
summary
Attractive, affordable
packages for recording on
computer and iPad. Neat
iOS integration and useful
bundled software are
definite plus points, but
those requiring higherquality preamps and more
flexible direct monitoring
may want to explore other
options.
information
Audiobox iOne $129.95,
Audiobox iTwo $159.95.
PreSonus Audio
Electronics +1 225 216
7887.
www.presonus.com
PreSonus have expanded their AudioBox range with two compact interfaces that will
work with your Mac, PC or iOS device.
Barry Watson
he AudioBox iOne and iTwo are the most recent additions to the popular line of USB interfaces from PreSonus, sporting
a familiar blue paint job and adding iOS compatibility to the specification sheets. The larger of the devices features two
simultaneous mic, line and instrument inputs, an input-versus-playback mix control for direct monitoring, plus MIDI I/O.
Meanwhile, the iOne has a single mic preamp, one instrument input and is devoid of line inputs or MIDI ports (a simple on/off
toggle switch is included for direct monitoring here). Unlike many competing iOS-ready interfaces, no additional cables or
adapters need to be purchased to start recording, and the package is augmented by the inclusion of an Artist version of the
manufacturers own Studio One DAW software, plus a handy way of moving work in progress from the iPad to the computer.
Case Study
The build quality of both boxes is very respectable on the whole. The sturdy aluminium cases have an even-colour finish and
chunky, industrial-looking bolts on corners. There are, however, some sharp edges where holes have been cut for the input
sockets, and even the edges of the case could be a little smoother to the touch. Unlikely to cause injury perhaps, but this does
detract from the overall quality. The buttons feel a little bit hollow when pressed, but gain and headphone level pots are more
solid, with gentle clicks to assist with setting and matching levels. The chunky main volume dial is pretty sturdy too, and
doesnt wobble about like the knobs of some competing devices. The various controls are laid out sensibly on the front panel,
with a signal activity/clipping LED included on each input to act as a simple level meter.
When connected to a computer, the iOne and iTwo are bus powered through the supplied USB cable. For use with an iPad,
the interfaces are first connected to a charger using the USB cable (ie. the charger plug supplied with the iPad) and then
connected to the iPad using its own 30-pin or Lightning cable. Unusually, the charging icon isnt shown on the display of the
iPad in this configuration, although Presonus maintain that a trickle charge will reach the iPad to avoid a drain in power. A
support article on the web site suggests that apps running in the background should be manually killed and flight mode
selected to avoid battery drain. During testing with a first-generation iPad Mini, the battery did in fact slowly drain when using
the Capture Duo app, losing charge more rapidly on more intensive apps such as GarageBand. I consider this not an
inconsiderable shortcoming as it renders extended sessions impossible without having to stop recording to recharge. The
associated battery range anxiety detracts from the would-be relaxed atmosphere of recording on the iPad very frustrating!
Installation
The driver installation package for Windows is reached via a download. On the Mac side, no driver is required as the
interfaces are class compliant. However, the hardware does require Mac OS 10.8 or later (despite 10.7.5 being listed as the
minimum specification on the packaging). Commendably, PreSonus supply a user-friendly operation manual as an additional
download also including tutorials for recording a range of instruments, processing and mixing tips, and even a recipe!
Following product registration, the Studio One Artist software is downloaded. The installation packages are rather large:
around 3 to 5 GB for the main application with the software instrument soundsets. The download runs into many more
gigabytes still when users select options for third-party content (eg. the free version of Native Instruments Komplete player).
The Capture Duo app for iPad is available for free from the
App Store. It offers an incredibly simple way to record sketches
as lossless audio files without the bells and whistles of virtual
instruments or loops. You record tracks on the app and then,
cleverly, the app sends the project over to Studio One on the
computer via the Wi-Fi network. The feature is extremely helpful,
considering that alternative apps such as GarageBand fail to
offer a method for moving recording projects across to the
computer for more detailed editing and mixing. The downside,
though, is that the free version is limited to only two tracks; you have to pay $9.99 to get the full 32-track recorder, so this extra
cost needs to be taken into account when purchasing one of the devices.
Studio Tour
The entry-level Studio One Artist is an incredibly worthy inclusion, rather than being bundled software you might never use
due to limitations. Unlike starter packages from some other manufacturers, there is no audio track limit and it has a wide
range of useful plug-ins. Notably, these include a surprisingly capable amp simulator with additional stomp-box effects, a
passable digital reverb, an impressive range of monitoring utilities, including real-time frequency analysers/phase checking,
and a full seven-band equaliser. These features alone make Studio One Artist a very strong contender for audio recording and
mixing when compared to its competitors.
Added to the capable audio facilities, there is also a multitude
of software instrument patches. Among these are some perfectly
decent synthesizer, drum and pad sounds that wouldnt be out of
place in a finished production but, as you might expect, samplebased presets such as pianos are only really suitable for demo
material. Nevertheless, theres enough to get you started before
wanting to add any additional third-party software instruments.
In The Field
All the software is straightforward to operate. On the iPad, the
Capture Duo app offers simple level and pan controls plus basic
editing features, allowing the user to concentrate on recording
decent takes, and I found it fairly straightforward to get a project
underway in Studio One. The simple approach to direct
monitoring works fairly well on both interfaces: a 50/50 mix is fine
in almost all applications and helps to avoid any latency that
would be experienced when monitoring through software
(particularly with iOS). Personally though, Id prefer to have the
option to pan the input signals hard left and right in the monitor
mix to help separation, and the headphone amplifier could be a
little louder for some applications.
When monitoring real-time effects through the software on the
iPad (eg. using an amp simulator in GarageBand) the resulting
latency is bearable on all but the fastest of musical passages. In
Studio One, virtual instruments can be triggered with no
discernible latency; in testing, the interfaces handled low buffer
settings without any nasty side effects.
In terms of audio quality, I found the iOne and iTwo to be just
fine. No obvious noise is apparent on inputs and the preamps
are perfectly functional, but you shouldnt expect miracles in the
quality of the converters. I took the time to compare the playback
quality of uncompressed audio between interfaces to the onboard sound from my MacBook Pro. At similar listening levels,
the iOne and iTwo gave slightly clearer reverb tails on the
reference tracks and an very slightly fuller low-frequency
response. Whilst these are only very small differences that dont
really wow the listener, the interfaces are well up to the task for
recording in a home studio.
All in all, the AudioBox iOne and iTwo offer a useful feature
set, good sound and build quality and a refreshingly useful
software bundle. For the home studio, these are well worth a
look, but its unfortunate that the iPad integration isnt quite as
refined as it could be. Recommended.
.
The interfaces feature two balanced line outputs and a

Kensington security slot. The USB 2 socket connects to
either the computer or iPad charger plug via the supplied
USB cable, whilst the iPad is connected to the Device port
using its own power cable. The iTwo adds MIDI I/O.

explained
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The Duo Capture iPad app. Whilst there are only very basic
mixing controls (volume and pan per track), projects can be
sent via Wi-Fi to the Studio One DAW software on your
computer. A paid upgrade is required to access the full 32track version.
Mixing with Studio One Artist. The entry-level DAW offers a

range of capable audio plug-ins including a full seven-band
equaliser with real-time spectrum analysis and digital room
reverb.
Alternatives
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Although it costs more, the Apogee One works with Mac and iOS (but not Windows) and provides up to two simultaneous
inputs including a useful internal condenser mic. Theres no MIDI I/O, but direct monitoring is taken care of by an app,
which includes the option to vary pan and volume for each input in the monitor mix. The Roland UA22 Duo Capture EX is
a closely matched contender to the iTwo in terms of specification, and its solid build quality and respected preamps make
this well worth your consideration. However, it does require the Apple Camera Connection Kit to make it work with an
iPad, and sample rates are limited to 48kHz. The Alesis iO Hub is an entry-level, two-input audio interface for Windows,
Mac and iOS. It too requires the Apple Camera Connection Kit as an additional purchase, and a 9V battery is also
required to supply phantom power to condenser mics when used with an iOS device so Id say this is more readily
suited to those who record using dynamic mics.
Specifications Compared
iTwo
USB 2.
24-bit with sample rates of up to 96kHz.
Two simultaneous mic/line/instrument inputs on combination connectors.
Two TRS jack line outputs.
One headphone output.
MIDI In/Out.
Direct monitoring input/playback mix control.
Requires Mac OS 10.8.5 or 10.9.4/Windows 7 or later (tested using Mac OS 10.9.4).
Works with all iPads except first generation (iOS 7.0.3 or higher required).
Supplied with USB cable and Studio One Artist DAW (via download).
iOne
As iTwo except:
Two simultaneous inputs comprising XLR mic input and one instrument input (no line inputs present).
Simpler direct monitoring on/off button only.
No MIDI In/Out.
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Orchid Electronics Mic Mute Lite

Microphone Mute Switch
Reviews : Accessory
Hugh Robjohns

It is often desirable to mute a microphone easily for example, a musician might need to mute the
mic clipped onto an instrument when its not being used on stage, or a producer might want to control an extension talkback
microphone. There are myriad applications when you start thinking about it.
From a technical perspective, the best way is to mute the line-level output of the mics preamp with a relay or electronic
switch, but thats not always very convenient sometimes the switch needs to be close to the mic and wired directly into the
mic cable. The traditional solution in such cases is to short together the hot and cold sides of the mics balanced output with a
switch, and there are countless passive switch boxes of this form on the market. However, while this approach works
adequately with dynamic mics, it usually produces rather nasty clicks if phantom power is present on the cable, so its not a
great solution when working with capacitor mics.
The clicking can be reduced with the addition of a few passive
components, but a really quiet system requires a rather more sophisticated
approach, and thats precisely what Orchid Electronics have done in the
phantom-powered Microphone Mute and the Mic Mute Lite products.
Needless to say, the Lite version is the simpler of the two, with just a pair
of XLR sockets to connect the unit into the cable run between microphone
and preamp, and an electronically latched footswitch on the top panel to
activate the mute indicated by a bright red LED. The Mic Mute Lite is
housed in a small 300g aluminium die-cast box measuring 120 x 95 x 35
mm, with a non-slip foam pad on the bottom.
The more elaborate version is designed primarily for use with miked-up
instruments on-stage, but it is a versatile unit with many applications. It
supplements the basic facilities of the Lite model with two actively
buffered, unbalanced, quarter-inch instrument-level outputs. One follows
the muting condition and is intended to feed the mic signal to a stage
amplifier, while the other provides an always-on output for a tuner (or
other device). The top panel again features an electronically latched
footswitch with associated LED, and the sturdy die-cast case is a little
larger at 145 x 120 x 45 mm.
The miniscule power needed for the internal switching circuitry (and
buffers) comes from normal phantom power, supplied by the mics preamp or mixer, and this is also passed on to the
microphone, of course. When first connected to a phantom supply both models power up in the muted state, with the red LED
illuminated to indicate that everything is working and ready for use.
Internally, the Mic Mute Lite is very nicely engineered with a lot of attention to detail to ensure a long and reliable life. The
silent mute functionality is achieved with a FET-based relay specifically designed for low-level analogue signals, and it
effectively shorts together pins two and three of the output XLR in a controlled way that minimises clicks. The larger model
also incorporates a small bridging transformer to split off the microphone signal, which is then amplified to feed the two
unbalanced instrument-level outputs, one of which is has its own mute circuit which follows the microphone signals muting.
My Audio Precision bench tests of a Mic Mute Lite pedal revealed a small residual attenuation of the mic signal when
unmuted, amounting to about 2.5dB. This is not significant in normal applications. With the muting circuit engaged, the
attenuation at 1kHz reached a tad over -60dB (but was slightly dependent on the source and destination impedances),
decreasing by 10dB at 20Hz and increasing by the same amount at 20kHz. In practice, this amount of attenuation is quite
sufficient, and the mic can be considered to be switched off for all normal intents and purposes. Theres no significant increase
in noise or distortion, and no headroom limitation, imposed upon the microphone through signal.
As Ive come to expect from Orchid Electronics, the Mic Mute Lite worked just as claimed. The review model imposed a tiny
low-level thud when muting, and a tiny click when unmuting, both being somewhat dependent on the audio present at the time
of the switch activation. However, theres no audible degradation to the microphone signal at all (other than a barely
noticeable drop in level), and no audible breakthrough when muted. Its a simple device which works well, and you quickly
wonder how you ever managed without it because its so handy to have around!
I gather that the more elaborate Mic Mute model is very popular with folk fiddle players and country music players in the
USA, Canada and elsewhere around the world, and I can certainly see the attraction. It has also found applications with
vocalists and instrumentalists for looping purposes, splitting the signal to feed a looping effects pedal (with either a mutable
or permanent signal, as appropriate).
I am unaware of any comparable-quality alternative which is anywhere near as cost-effective as these Orchid products. The
design and build quality is impressive, and both of these units are highly recommended for anyone in need of simple
microphone muting. Hugh Robjohns
Balanced Microphone Mute 68, Balanced Mic Mute-Lite 56. (About $89 and $107 when going to press). Prices exclude
shipping from the UK.
www.orchid-electronics.co.uk
.
HYBRID
STUDIOS
ORANGE COUNTY, CA
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Radial Headload V8
Guitar-amp Power Soak & DI
Reviews : Guitar Amplification
Buy PDF
Radial
Headload $900
pros
Built to a professional
standard.
Feels great to play through.
Easy to get a great sound.
Good value for money.
cons
You have to pay for this kind
of quality!
summary
Combining an amplifier
attenuator, loudspeaker
emulator, DI, phase
alignment tool and a range
of interconnection options,
the Radial Engineering
Headload V8 is one of the
most comprehensively
equipped units in its class.
The units premium price is
more than justified by its
build quality and
performance, and it is
definitely one to audition for
anyone seeking to record or
gig with up to 120 Watts of
guitar or bass valve
amplification without
deafening the
neighbourhood.
information
$899.99
Radial Engineering +1
604 942 1001
info@radialeng.com
www.radialeng.com
A combination power soak, DI box, speaker simulator and phase alignment tool,
Radials Headload has a role in almost any guitar rig.
Bob Thomas
cant be certain that Radial Engineerings founder Peter Janis maintains a network of spies who lurk around studios and
stages looking for problems for his company to solve, but its hard to imagine another explanation! Amongst the 40-plus DI
boxes, switchers, splitters, processors and interfaces that form a significant part of the Radial product line, there are plenty
of boxes that have, at one time or another, saved my proverbial bacon.
Radials reputation sits solidly on their design innovation and a build quality that revolves around solid steel casings, heavyduty circuit boards and the liberal use of transformers when appropriate. Their latest product, the Headload V8 Guitar Amp
Loadbox and DI (lets just call it the Headload), follows that paradigm, and its all-discrete circuitry packs resistor-based volume
attenuation together with Radials existing JDX Reactor DI and Phazer analogue phase alignment tool into a chunky all-steel
case, with the sort of price tag you might expect given all thats on offer.
Overview
The blue-enamelled steel case is vented high up on all four sides and at the back edge of the top plate, which implies that
heat is going to be generated in the Headloads upper reaches. Whipping off the top cover reveals a phalanx of cementencased resistors that can attenuate a maximum of 120W RMS of valve amplification. Achieving this means dissipating up to
a peak of 180 Watts as heat, and a temperature-controlled fan is strategically positioned to drive airflow across the resistor
board.
The front-panel controls are divided into four functional areas, the leftmost of which contains the five-position attenuation
level selector switch, which reduces the amplifier output level leaving the Headload from 100 percent down to zero in
increments of 20. At the 20 percent position, a trim control (which has the feel of a wire-wound variable resistor) comes into
play, allowing you to reduce the amplifier output anywhere from 20 down to one percent.
Associated with the attenuation level selector are the separate Hi and Lo Resonance on/off switches. These act in much the
same way as the loudness control on a stereo system, by compensating for the vagaries of the human hearing at low volume
levels, boosting 6.5kHz and 60Hz respectively by +3dB at 20 percent reduction, rising in +3dB steps to a +12dB boost at 80
percent reduction.
The central area carries the controls for the JDX Reactor outputs. The JDX input is taken pre-attenuation and passes
straight to a transformer, which effectively sits in parallel with the loudspeaker. The transformer and loudspeaker act together
to form a reactive load which is not only designed to capture the amplifiers output and the effect of the back-EMF (electromotive force) generated by the loudspeakers voice coil as it moves within the loudspeaker magnets magnetic field, but also
to provide a realistic playing experience for the guitarist. From there, the signal passes into an active, multi-stage band-pass
filter, which is designed to replicate the sound youd get from a Shure SM57 sitting a couple of inches away from one of the
loudspeakers in a Marshall 4x12 cabinet.
At that point, the signal splits in two, one part passing via a balancing transformer and polarity-invert switch to the pre-EQ
XLR output, and the other through a voicing section that adds five cabinet emulations and offers a two-band EQ for further
tonal fine-tuning. From here, the signal is split to feed three outputs: the front-panel headphone jack, an unbalanced, linelevel, post-EQ, post-JDX jack and, via an active balanced driver and polarity-invert switch, a second, post-EQ balanced XLR
output.
Logout
The unbalanced line-level output shares a volume control with

another unbalanced jack output that carries an unfiltered, preattenuation, line-level version of the amplifier output taken preJDX and pre-Phazer.
The Phazer enables you to align the phase of the Headload
output with the signal from, for example, a microphone set up a
couple of feet back from the amplifier cabinet or a digital
processor acting on one of the outputs. If you think about it, your
amplifiers output signal will travel down the cable from the
Headload to your mixing console or DAW at the speed of light
while the sound from your speaker cab will have to travel across
a foot or two of air at the speed of sound (which will take a
couple of milliseconds) to reach the microphone before getting
up to light speed. Similarly, if youve got a digital processor on
one of the Headloads unbalanced outputs, there will be a few
milliseconds of delay introduced by the A-D and D-A conversion
processes.
Different impedance versions are available, and depending

on which you buy, youll find one of the inputs blanked off.
Youll hear these time differences as phase effects when you combine the signals at the console. In action, the Phazer
delays the signal feeding the Headloads back-panel line-level outputs by a slight amount in order to produce a phase shift of
up to 180 degrees. The amount of phase shift is controlled by a rotary potentiometer and a further switch that brings in an
additional 180 degrees of shift, giving a total of 360 degrees to play with. If using the Headload on stage, youll find that
adjusting the phase between the sound of your amplifier coming through the PA via the Headload and the room sound from
your speaker cab can often pay big dividends.
Finally, Radial earn themselves a chunk of kudos for the full-size, four-pin, locking XLR connector that carries the voltages
from the Headloads external power supply.
In Use
The Headload is available in 16, 8 and 4 versions. Depending on the version, one of the two From Amplifier inputs will be
blanked off for the 8 review model, the 4 jack was blanked off. You need to think a bit about your impedance selection if
youre using a two-cab setup, as the outputs run in parallel. This means that youd need two 16 cabs to work with this 8
version, or, if you were using two 8 cabs, youd need to purchase the 4 version.
The whole point of using a valve amp with an attenuator/speaker emulator is to enable you to experience the sound of your
amplifier running in its (inevitably loud) sweet spot and the Headload is (to my mind) one of the best at providing this that Ive
played through. The playing experience felt right at all attenuated levels even right down at the one percent setting. The Hi
and Lo resonance switches let me tune the speaker output to keep a pretty accurate representation of the unattenuated sound
coming from the cabinet in use. Obviously it would be unreasonable to expect to get an absolutely exact facsimile of the
original amp/cab tone at high attenuation levels, but what I did get sounded (and felt) close enough across the whole
attenuation range to keep me more than happy.
The JDX DI section performs equally impressively. Position A on the speaker cab Voicing control bypasses the voicing
section so that the classic JDX Shure SM57/Marshall 4x12 sound appears at all the pre-EQ and post-EQ outputs.
In a live situation, the intended use of the pre-EQ XLR output is as the feed for the PA FOH, while the post-EQ XLR is
designated as a source for IEM or stage monitors. With the growing prevalence of IEM systems, guitarists (and bassists) are
becoming more dependent on attenuated or emulated guitar sounds and the Headload fits into this niche by offering an
additional five voicing choices, all of which can be further refined by the associated two-band EQ.
Of course, these differing voicings also come into play in a
studio setting, and I have to say that I was very impressed by the
choices on offer. When I want to get loud and distorted guitar
sounds, I usually run my amps into a 1x10 isolation cab, and Im
pretty happy with the results that I get or at least I was happy
until the Headload turned up! Its not that the Headload is
intrinsically a better solution but, out in the real world, the results
were as good as any from my isolation cab.
The Phazer is an interesting additional facility. It does what it
says on the tin and allows you to pull a cab/room mic into phase
with the DI output, or to introduce a phase anomaly between
them for creative effect. Im used to carrying out these processes
in the digital domain, so Im not sure that Id personally use this
facility much when recording, but I think that Id find the Phazers
facility useful when engineering live sound.
Finally, there are the additional options offered by the pre- and post-JDX unbalanced line-level outputs to bear in mind.
These offer the possibility of connecting (for example) another cabinet emulator via the pre-JDX jack and/or an effects unit
through the post-JDX output.
Conclusion
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The Headload V8 certainly isnt the first analogue loadbox/DI combination that Ive seen, but it is one of the most
comprehensively equipped examples in terms of control and output connections that Ive come across. Playing through it feels
and (with a bit of tweaking) sounds very close to the source amplifier/speaker combination, even at high attenuation settings.
Radial Engineerings Headload V8 is a professional tool and thus it doesnt come cheap. While this means its not going to
be an impulse buy for the typical home recordist, its build quality and performance more than justify its premium price. The
Headload would certainly pay its way for any guitarist, bassist or studio engineer who needs a high-quality, fully featured
amplifier attenuator, dummy load, loudspeaker emulator, DI and phase alignment tool. And having all of these in one box with
a variety of connection capabilities is appealing. Im certainly more than happy to record and gig my valve amps with a
Headload V8, and Im not at all sure that Ill be letting this one leave me any time soon.
.

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Alternatives
At this price level, the only units that I can think of that directly compete with the Headload V8 are the Two Notes Torpedo
Studio and Torpedo Live Digital Loadboxes. The major difference between these and the Headload is that the Torpedo
units process the attenuated amplifier output digitally, to emulate a wider range of mic/cabinet combinations than you get
with the all-analogue emulation of the Headload V8. Both approaches have their merits and both are capable of extremely
impressive results, depending on your application and personal taste. Coming down the price scale, youll find a wide
variety of choices from the likes of Rivera, Palmer and SPL, and theres also the UK-designed and -built Sequis
Motherload Elemental to consider.
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Fri 19 Jun 2015
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In this article:
Toy Story
Crossover Palaver
Great, Smashing,
Super
Different Drummer
Analogue Retention
Pattern Sequencer
Arpeggiator
Effects
Conclusion
Alternatives
The Voice
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WebExtras
Roland JDXi
Synthesizer
Reviews : Synthesizer
Buy PDF
Roland
JDXi $499
pros
A compact, high-quality
sketchpad or jamming
companion at a tempting
price.
Generous polyphony.
Deep, lush and rewarding
digital synthesis and
superbsounding drums.
Wellimplemented effects.
Sequencer offers three
classic recording methods.
cons
Several important functions
demand the sequencer
stops.
The user interface isnt
suited to a synth engine this
complex. Would be more
appealing with a software
editor.
Tiny keys and general
flimsiness.
No battery option.
summary
A small, lightweight synth
with an impressive sound
palette, monster drums and
a whiff of real analogue.
Despite mini keys and
lengthy menus, the JDXis
sequencer, effects and
signal processing options
add up to an impressive
instrument for the money.
information
$499
Roland Corporation US
+1 323 890 3700.
www.rolandus.com
Rolands JDXi crams an awful lot of synth into a very small box.
Paul Nagle
ince last years launch of the Aira range, Rolands musical legacy has been widely celebrated for its influence on
popular culture. Not content to rest on past glories, the Japanese giant continues to fight for its rightful place in a
market increasingly awash with analogue. The JDXi is a step in a new direction, a sub$500 combination of
technologies, rejoicing in the description interactive analog/digital crossover synthesizer. Apart from that being quite a
mouthful, its immediately obvious there isnt a whole lot of crossing over going on at this stage. The analogue component is
heavily outnumbered: one voice against 128 of the shiny Supernatural variety. These resources are shared across four parts:
two digital synths, drums and the (hybrid) analogue monosynth.
An onboard sequencer provides classic step and realtime recording and helps define this diminutive keyboard as a
credible musical sketchpad and live accompaniment machine. Present, too, is a vocoder and vocal transformer, while better
thanaverage effects complete a sonic toolkit scoped to leave rival small synths and grooveboxes in the lurch. It adds up to a
serious package of high-end technology for the money, but how does it all hang together?
Toy Story
The JDXi is a portable, plastic performance synth in the Korg Microkorg or Novation Mininova mould. Whether these
instruments succeeded because of their tiny keyboards or in spite of them well never know for sure, but the compact (575 x
245 x 85 mm) JDXi follows in their footsteps by adopting mini keys 37 of them. These are velocity sensitive and have a
configurable response, but no aftertouch. Such a keyboard may well be adequate for sequencer recording and interaction, but
its always going to be unwelcome to players happy with the traditional size. The teensy pitch and mod wheels wont win many
converts either, but the wellspaced knobs providing muchneeded instant access to filter, LFO and effects fare better.
Less appealing are the buttons, especially the eight rubber imps camouflaged in black beneath the display. They really
should have been bigger and bolder because not only do they select programs, they are essential to navigate the long (and
mostly linear) menu system. Condemned to endless buttonprodding, your lot is improved slightly by shortcuts and shift
operations that generate larger or faster increments, but what is sorely lacking is a data entry encoder.
Keeping to the Microkorg brief, a snakelike microphone bursts from the front panel to unleash yet more singing robots
upon the world. Actually, its a good deal more flexible than that and, as well as the microphone socket, theres a second input
on the rear for line/guitar effect processing. When this is used, it overrides the microphone.
I welcomed seeing MIDI In/Out ports of industry standard dimensions and shape so much so that I could almost forgive
the lack of a dedicated Thru (soft Thru is available). Theres nothing miniature about the stereo outputs either, nor does the
headphone jack conjure unpleasant thoughts of earbuds. Sadly theres no sustain pedal input, but for recording piano parts
or similar you might get by, at a push, with the panels Key Hold button.
In common with pretty much everything Roland makes these days, the JDXi can function as an audio interface. A high
speed USB 2 connector provides this connectivity but doesnt supply enough power to run the JDXi youll need the supplied
9V adapter for that. Battery operation isnt an option either. The USB implementation is rather neat in that it can switch
between a generic (class-compliant) MIDI mode and Vendor, which uses Rolands proprietary (two in, two out) Audio and
MIDI driver.
Surveying the interface, youre struck by several incongruities. Firstly, you notice that the dull red text on black background
is difficult to discern in low light. Next, although most of the knobs are selfexplanatory, you might puzzle over the single
control marked Envelope. Experimentation reveals that it provides a continuous series of amplitude envelope overrides,
progressing through short and snappy through to organlike shapes, arriving eventually at a slow-attack, long-release ambient
mush. However, the filter envelope isnt tweaked in parallel, so the effect is often compromised, forcing you to grapple with
menus. Lastly, even those who arent hung up on aesthetics might find the twoline amber LCD drab, the construction light
and insubstantial, and the shiny plastic panel somewhat toylike. Fortunately, looks arent everything.
Crossover Palaver
Whilst pondering truth, beauty and the JDXis crossover nature, I started to wonder how far 128 digital voices would go. The
number doesnt represent absolute polyphony because it is divided up by Rolands familiar tone system, in which multiple
waveforms are layered. Acknowledging that some sounds will consume more voices than others, I never once experienced a
polyphony issue. This, despite playing large pads, dense drum patterns and twofisted electric piano.
There are 256 preset programs (banks AD) and the same amount of user locations (EH). Having made a selection, you
can freely modify all constituent parts then resave to another location without impacting other programs. Unhelpfully, the
synth always powers up at the first preset and, equally unhelpfully, the program name is only shown when you hold the shift
key. Fortunately the Favorite button is on hand to access 16 banks of 16 chosen programs directly a much faster method
than pressing Value buttons. The top line of LCD realestate is reserved for the patch/bank number, tempo and the bar
counter, whether the sequencer is active or not.
A programs four parts each have a dedicated sequencer track and a path through the effects. A vertical column of buttons
affords instant Part selection or, when the shift key is brought into play, track muting.
Great, Smashing, Super

Before exploring the two digital synths that head the column, I
should mention that only one at once can use the vocoder,
On the JDXis back panel we find a power input and switch, a
otherwise they are functionally identical. For an introduction to
USB port and MIDI I/O sockets, a line/guitar-level input
the JDXis wealth of factory sounds, select a category using the
switch and, all on quarterinch jacks, audio in, stereo out and
large, slightly stiff, rotary switch. The offerings are: Vocoder/Auto
headphone sockets.
Pitch, Brass, Seq, FX/Other, Keyboard, Bass, Lead and
Strings/Pad, although some contain unexpected inclusions. For example, Keyboard is where youll find guitars, harps and
sitars, while FX/Other houses orchestral hits alongside weird noises.
There are a total of 256 preset tones for the digital synths, but its no hardship to audition so many because the quality is
consistently high. The stunning renditions of orchestral strings, synth basses and choirs caught my ear at once; while there
isnt a regular acoustic piano, the punchy JD Piano and Electric Grand are both more than serviceable, the various electric
pianos are fab too. Still, no matter how playable these all are, they only represent the tip of the iceberg.
Theres nothing wrong with selecting tones and making basic tweaks with the panel knobs, but sooner or later youll head
for the Menu/Write button to see how deep the rabbit hole goes. With 10 toplevel entry points (11 if you count version info),
the menu system covers every major area of the JDXi, including global or system functions such as MIDI sync and
metronome behaviour. Some pages contain programspecific data (eg. the tempo, part levels and effects routings), but by far
the most heavily populated menu is Tone Edit. Once in there, the squishy Cursor and Value buttons can expect a thorough
workout, especially since a part consists of up to three layered tones, all which can be edited independently.
Rolands Supernatural synthesis is the basis for all the
expected virtual analogue waves, plus Supersaw and PCM
based sources. Each tone has an envelope for pitch, filter and
amplifier (one stage per page) and perhaps due to limited screen
space, the 160 PCM waves arent shown as names, only as
numbers. For convenience, it could be worth printing a few
sheets from the downloadable Parameter Guide, which shows
the full list.
Online
Private
Network
Access Secure
Virtual Private
Network From
Anywhere W/ Free
App!
Having chosen a sound generator, your source may be

Rolands designers have crammed a lot of controls into the
JDXis 57 x 24 cm front panel.
processed by seven types of filter including four lowpass filters
with a choice of 12dB or 24dB modes. Backing up the primary
choice theres always a high-pass filter on hand to thin things out. Without getting too sidetracked by all the options, suffice it
to say that this is a superiorsounding synth engine with qualities that almost transcend the cumbersome menus. Ive never
liked adjusting envelopes one stage at a time, but at least these have some useful extras in the form of loop points and a
tempo-sync function.
Different Drummer
The menu trawl continues into the Drum part. Indeed, there are so many parameters that kit production could become an
obsession, not that theres any urgency because the 33 factory kits are, without exception, wonderful. Its obvious Roland
have been listening closely to their old machines because represented in all their glory are the TR606, TR626, TR707, TR727
and CR78 alongside their more famous siblings, the TR808 and TR909. You wont go far wrong with the selection of genre
specific kits either.
A kit consists of 26 voices and, if the lighting is right, you might even spot their names printed in faint grey above the keys.
Each voice can be set to single or multi play, with the latter denoting that multiple hits are allowed; a useful feature for cymbals
and other instruments suited to an accumulating decay.
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Every voice has up to four wave generators, whose output can be independently panned to create a complex stereo
instrument. For the more experimentally minded, individual waves can be randomly thrown around the stereo field, or
switched from left to right on alternate hits. Supplementing this extravagant panning are tools to define the waves velocity
range and to specify a desired behaviour for velocities outside the range. By now it wont come as a surprise to learn that
each wave can be extensively transposed, filtered, envelopeshaped and tossed into the effects processors.
When considering drum kits and the effects, theres more to play with than first impressions suggest. If you want a load of
reverb on your clap, a bit less, plus delay, on the hihat, distortion on the first kick but more distortion plus flanging on the
second, all are achievable on the JDXi. As a further example of the level of detail within a kit, no fewer than 31 mute groups
permit sophisticated interaction between voices: far more than the usual open/closed hihat or conga muting.
Roland GR55
Guitar Synthesizer
Roland have
put elements
of their two
very different
approaches to guitar
synthesis in a single
box. Could this be the
best guitar synth ever?
Moog Minimoog
Voyager XL
Analogue Synthesizer
Theres no
more
revered
name in the
history of synthesis than
Moog, and the Voyager
XL aims to cement their
reputation for topflight
instruments. Is this
the Rolls Royce of the
synthesizer world?
Dewanatron
Swarmatron
This is
a synth like
no other,
eschewing
conventional controls,
nomenclature and even
an ordinary on/off
switch. Is it destined to
become a cult classic?
XILS Lab
PolyKB II
The original
was
a diamond in
the rough
so is PolyKB II a highly
polished gem?
Spectrasonics
Omnisphere 1.5
Spectrasonics bring yet

more goodies to the
Omnisphere party,
aiming to make their
highly acclaimed synth
even better.
M-Audio Venom
Synthesizer
M-Audio's
debut synth
may have a
pristine
white exterior, but it
hides a sample-based
synthesis engine
capable of getting down
Analogue Retention
Which brings us to the analogue synth, or more correctly the analogue oscillator, sub-oscillator and lowpass filter set in an
ocean of software. I was unable to uncover whether were talking VCO or DCO (my money would be on the latter), but its
output involves familiar waveforms triangle, sawtooth and variable square. A petite Monotronlike knob is on hand for direct
access to the pulsewidth, but the range on the review model didnt extend to really thin pulses, rendering it impossible to
create the fine, reedy tones we know and love. And while pulse-width adjustment is as accessible as the word analogue
deserves, it takes numerous menu hops before you can introduce pulsewidth modulation.
The sub-oscillator is unusual in responding to any PW or PWM tweak. Its equally unusual in being at a fixed volume, either
on or off. Heres where an optional thinner pulse might have helped because sometimes on was a bit much. The sub is
available at one or two octaves below the main waveform and theres no doubt it adds extra beef to an oscillator thats sweet
enough, if not quite so full and inspiring as, for example, a Roland SH101s.
No information was forthcoming about the analogue LPF, but a quick fondle of the cutoff and resonance suggested this to
be a 24dB filter, and a decentsounding one at that. Its also slightly steppy and possesses a dominant resonance on a par
with the System 1s ACB filter. Roland inform me that the review model is a late-stage prototype and that the stepping is one
of several issues already resolved. Personally, Id love to have heard the filter overdriven, or frequencymodulated by the
oscillator, but such fripperies are beyond the simple bleep and squelch remit and maybe beyond DIY hacking too.
Ultimately, the JDXis capacity for real analogue bass lines and solos will satisfy many users.
Pattern Sequencer
I hope the term Pattern Sequencer doesnt bring false hopes of a free pool of patterns to fool around with. Its strictly one four
track pattern per program. All tracks share scale settings (ie. timing resolution) and are of equal length (between one and four
bars). If you switch programs during playback, the next programs pattern takes over smoothly at the end, with delay repeats
continuing without interruption, as it should be.
With a maximum of four bars to play with, your scope for recording elaborate keyboard riffs isnt huge, but if you opt for a
scale setting of 32nd-note divisions then halve the tempo, its possible to imagine there are eight bars. The sequencer
capacity isnt stated, but if you record too much data, warning messages demand you hit Exit before you can proceed.
Apart from its step buttons and flashing tap-tempo, the sequencer isnt the most visually stimulating. The current beat and
number of measures take pride of place on the display, but theres no obvious way to make the bars scroll automatically for
editing. More seriously for an interactive synth, several important commands cannot be executed while the sequencer is
running. In particular, a program cant be saved nor can an iffy chord be fully cleared from a step without first stopping.
Adapting to these limitations is essential if youre going to enjoy the JDXi experience, but in its defence Id argue there are
plenty of reasons to make the effort, not least if you relish having three classic Roland recording methodologies in a single
box.
Of all the recording tools on the shelf, TRRec is universally familiar thanks to certain legendary drum machines. The
strength of this implementation is in its simplicity: select a voice and then enable steps with the 16 backlit rubber keys. In
performance, your speed is hampered slightly by having to locate the shift key in order to select the bar to edit a bit of a
stretch for onehanded operation.
Realtime recording is ideal for capturing drums or regular keyboard parts, subject to the current pattern resolution.
Regardless of the inherent quantisation, recording this way is a blast and a major boost to the JDXis performance credentials.
You can also capture the movements of panel knobs, but here the manual warns the pattern might fail to keep up if you record
extreme knob movements. This is especially likely if you get carried away with envelope tweaks or filter sweeps for individual
drum voices.
Put it down to my lingering adolescence if you like, but I found it far too easy to stray into extreme knob movements and
often hit the Pattern Full! or Rec Overflow! warnings. If youre not afraid to really stress the poor things processor, you can
activate the transmission (and therefore recording) of all menubased edits. Despite the resulting glitches and warnings,
recording tweaks can bring a pattern to life, so its to be hoped the issues can be addressed, or at least improved. My last
observation concerns the recording of effect adjustments. Since these arent linked to any specific track, they arent cleared
when you erase the pattern data. The suggested workaround is to rerecord, but Id like to think a better solution for wiping
them can be devised.
Step time is the third means of producing sequences and its almost exactly as it should be: fast, intuitive and fun. Notes are
entered as per the Roland JX3P many moons ago, with chords, ties and rests added to each step. Surpassing that venerable
synth, velocity can be captured too, and since you can effortlessly slip into realtime recording and add filter sweeps or LFO
wobbles, the sequencer is ideal for lovers of dripping acidic patterns. Its main drawback in this respect relates to the fixing of
all tracks to equal lengths, which cruelly rules out the oddlengthed, polyrhythmic patterns so typical of the JX3P and SH101.
Theres one final area in which the JDXi falls short of its groovebox aspirations: theres no convenient way I could see to
mute/unmute individual drum voices. One way to achieve it is by selecting a voice and quickly grabbing the level knob, but
thats far from ideal.
Arpeggiator
After enjoying the sequencer, I continued to reminisce fondly over SH101 functions, turning next to the arpeggiator. This is
complete with a physical hold button even though the rest of its options are menu-based. Arpeggiators represent another
form of automated jamming, here restricted to a single part at once. The usual up, down and random directions are offered,
but there are 128 preset patterns built in too, full of slidey bass lines, sequences and the like. There arent any drumspecific
versions but, nevertheless, a ton of variation and fill mileage is at your fingertips by simply muting the drum track and letting
rip with arpeggiated noodles.
Effects
In choosing an effects implementation, the JDXi went for quality over quantity with just four simultaneous effects. Every part
and each individual voice in a drum kit can be processed, with a fair amount of finesse, through the two insert and two send
effects, even if not every permutation is possible.
The first effect is generally intended to bigup or nastify your signal. I might ordinarily skip past its distortion, fuzz,
compression and bitcrushing, except that I got hung up on the distortion for quite some time. Unlike most synth distortion
algorithms, this one doesnt set your teeth on edge and it revealed further gritty appeal once I checked out the scope for
customisation. It transpires that all the effects are wellstocked with programming options under the hood; for the distortion
alone, there are six different flavours, plus adjustable drive amount and presence (highend response).
and dirty...
Waldorf PPG
Wave 3.V
PPG's Wave
series were
sadly
beyond the
budget of most of us,
but, through the miracle
of software, the powers
of these innovative
synths may now be
within our grasp...
Novation
Ultranova
Synthesizer
The
Ultranova
may be
a return to
Novation's roots, but it's
still a very forwardlooking synthesizer...
Yamaha Motif
XF7
Workstation
Synthesizer
Yamaha's
long-lived
Motif range
continues to
go from strength to
strength. Could the
latest model be the best
Motif yet?
Mode Machines
Xoxbox
Everybody,
as Fatboy
Slim so
wisely notes,
needs a 303. However,
with originals becoming
ever more scarce and
expensive, the dream of
universal 303 ownership
was starting to look
unlikely until now...
Vermona Mono
Lancet
The
peculiarly
named
Mono Lancet
is an analogue synth of
the old school, boasting
two oscillators, a filter
with a debilitating debt
to Moog, and knobs
galore!
Tom Oberheim
SEM
Tom
Oberheim
has returned
to the
analogue synth fold with
a revised and updated
version of his classic
70s monosynth, the
celebrated Synthesizer
Expander Module.
Korg Monotron
Its their first
analogue
synth in 25
years, but is
Korgs Monotron a toy or
a tool?
Roland Gaia
SH01
Analogue Modelling
Synthesizer
Next along you find a choice of flanger, phaser, ring mod or slicer, of which only the ring mod was disappointing. The
phaser is a warm, rich bath of swooshiness and the slicer chops up the signal rhythmically, ready for your next euro dance hit.
Anyone hoping for a classic Roland chorus shouldnt abandon all hope either the flanger becomes quite chorusey with its
feedback set to zero.
The final two effects are traditional send effects. Both delay and reverb are clean, professionalsounding and finetunable,
to an extent. Its a shame the main panel sports just a single depth control for each, but when you consider the rewards its not
a massive inconvenience to rely on menus for setting the reverb type, the delays feedback and so on.
Conclusion
For many years and despite countless pleas, Roland have been resolutely not getting back into analogue, which is why this
little crossover synth has been such a pleasant encounter. It shouldnt be a surprise that the digital components take all the
honours; the JDXis strength lies in its Supernatural synth engine and in its highly malleable drums. But such complexity
comes at a price, and I found programming via this particular menu system about as enjoyable as eating soup with a fork.
Roland dont have an editor in the pipeline right now, but are considering the idea a potential gamechanger in my opinion.
The massive voice count makes a real difference to pads and typical workstation keyboard parts, all of which the JDXi
handles like a dream. Although only fourpart multitimbral, this doesnt feel limiting, but it is rather sad that the analogue synth
is not bolder and more fully featured. If that earns only a lukewarm reception, the keyboard leaves me positively cold, but
admittedly it keeps the overall size to a minimum and its certainly up to the job of building sequencer patterns and tapping in
drums. True, having to stop the sequencer before saving imposes limits on spontaneous song development, but otherwise the
sequencer has bags of potential thanks to its varied recording methods.
Summing up, the JDXi is an affordable, wellrealised blend of synthesis, drums and sequencer. Its not the most sturdy
instrument ever made, but its a highly portable groovebox full of sounds youll never be embarrassed to perform or record
with.
.
Alternatives
I cant think of much competition in the same price bracket but thanks to its sequencer, polyphony, multifaceted synthesis
and effects, the JDXi feels more like a padless alternative to Korgs new Electribe than it does to physically comparable
keyboards such as the Korg Microkorg XL and Novation Mininova.
The Voice
Text on the front panel suggests you should see the owners manual about Mic, but Im sure youve worked out this refers
to the included microphone. Previously I was captivated by the VT3 Vocal Processor, so its exciting to see some of its
vocoding muscle lodged within the JDXi. The vocoder must be fairly resourceheavy because its use disables the
analogue synth, but as vocoders go, it strikes a good balance between ease of use and clarity. Favourite presets include
the choirtastic VP330 and Robot, for those allimportant Man Machine moments.
There are alternate uses for the snaky appendage too, in the form of Auto Pitch. Primed with a set of major and minor
scales, it improves on typical pitch-correction by throwing in temporary gender reassignment too, courtesy of octave and
formant shifts.
Lastly, and for sheer novelty value, press the Auto Note button and notes will be generated by microphone input with no
need to touch the keyboard. My vocalisations tended towards the wild and erratic at first, but as the process even works
for drums and the analogue synth, I had a ball anyway.
If you dont
like
programming synths via

obscure two-line
displays and arcane
menu systems, the
Roland Gaia SH01
could be just what youre
looking for...
Moog Taurus 3
Bass Synthesizer
The
resurrection
of Moogs
stellar bass
synth has caused a
considerable stir. Can
the Taurus 3 live up to
the venerable reputation
of its ancestor?
Doepfer Dark
Energy
Synthesizer
The latest
product of
Doepfers
modular
know-how is the Dark
Energy: a compact,
powerful and hands-on
desktop analogue
synthesizer.
Cwejman
Synthesis
Modules
Modular Synth
Wowa
Cwejman is
already in
possession
of a fine reputation for
esoteric synth modules,
but he hasn't run out of
ideas yet. Join us as we
take a tour of his latest
creations...
SMS Planet 7
System
Modular Analogue
Rack Synthesizer
Synthetic
Music
Systems
have a
unique approach to
designing modular
synths that are both
high in quality and, wait
for it, low in price. Let's
investigate...
Analogue
Systems
Synthesis
Modules
RS420 Octave
Controller RS100
MkII Low-pass Filter
RS370 Poly Harmonic
Generator
Analogue
Systems'
modules
continue to
develop and evolve. We
take a look at a
selection of the latest
designs.
Cwejman
Synthesis
Modules
DLFO Dual LFO
RM2S Stereo Ring
Modulator VCEQ3
We conclude our threepart exploration of
Wowa Cwjeman's new
range of exclusive
analogue synth
Fri 19 Jun 2015
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Kontakt Instrument
Reviews : Sample / Sound / Song Library
Robin Bigwood
With The Hammersmith, a conventionally played acoustic grand piano, Soniccouture have released
one of their most mainstream and potentially important libraries yet.
The source piano is intriguing, a Steinway D with a motorised, MIDIdriven action that Soniccouture say assisted greatly with
the consistency of the sampling. Twenty one velocity layers were captured, with sustain pedal both up and down, using an
abundance of expensive mics from close, mid and room perspectives. The location was the nicesounding British Grove
studios in Hammersmith.
In Kontakt (or the free Kontakt Player), those multiple mic feeds are represented by a threechannel mixer with switchable mic
arrays. Each channel also has mute, solo and phase buttons, a foldout channel strip with fourband EQ, and (rather
unconventionally) synthstyle attack and decay parameters. They can optionally drive separate outputs too, allowing for
complex DAW mixing setups.
An Options panel exposes parameters for key, pedal and room noise, roundrobin sample switching, and one of the most
flexible velocity mapping schemes youll come across. Theres good support for temperament and microtuning too, with preset
saving and loading, and adjustment available on a notebynote basis.
A useful range of effects is built in: stereo width, a compressor, velocity-controlled HPF
and LPF filters, a convolution reverb (with naturalistic as well as whacky impulse
responses), and a master EQ.
All these features raise The Hammersmith up alongside the Synthogys and Garritans of the virtual piano world in terms of
flexibility, but its the sound quality and playing experience that may be more telling. Personally, I found it thrilling, immersive
and addictive. This Steinway has clearly got plenty of miles on the clock, and theres something of a wartsandall quality to
the sampling that lends authority, believability and lots of vibe. Not all the mic perspectives immediately sound beautiful
(especially the AKG D19 shoved in a soundhole), and thats the joy of it. Theres everything here from opulence and intimacy
to otherwordly and even ragged tones, all egged on enthusiastically by the EQ and effects. The resonant pedaldown sound
is fantastic, and depressing the pedal with notes already sounding does what youd hope. The sense of scale and harmonic
complexity increases thanks to a flawless behindthescenes crossfade. Nice programming.
Drawbacks? A bass F has its unisons tuned a smidgeon too far apart for my liking, and I noticed that presets utilising the
room-noise feature dont load with it sounding. You have to play a note first to trigger it. But this is piffling stuff. The
Hammersmith is an absolutely top-class virtual piano. For rock, pop and jazz it seems endlessly inspiring. It would not be my
first choice for classical styles, but thats not to say it couldnt be used successfully in that context. Its available at two prices:
the 19GB 99 Standard Edition with two mic perspectives and no mic model switching, or the 52GB 199 Professional Edition
described here. Robin Bigwood
$129/$229
www.soniccouture.com
.
Equipos de
Cmputo
Venta de
Computadoras,
Notebooks y
Tablets. Delivery
sin costo!
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Soundizers StereoMonoizer
Audio File Utility
Reviews : Software: ALL

This useful piece of software can help you streamline your projects and preserve
precious system resources.
Matt Houghton
How many times has someone sent you the multitracks for a project and youve found that every single file is in stereo
even the mono sounds? Usually its because the default bounce settings of a DAW are set to render things through the stereo
bus, irrespective of the mono/stereo nature of the source track. But when sounds are mono this is incredibly wasteful in terms
of storage, upload/download time when sharing projects via the web, computer processing power and hard-drive efficiency
(youre having to stream twice the data from disk that you should).
US company Soundizers have developed a natty little utility to address this issue. StereoMonoizer scans any audio files you
drag and drop into a box on its interface and identifies those stereo files which contain only mono content (ie. where the left
and the right channels contain identical information). It will also reveal any superfluous blank files (which might result from
careless rendering inside a DAW or recording channels with no input source), and detect split stereo files and allow you to
interleave them. It can scan files located in subfolders too, which means you can drag an entire projects folder structure
straight into the application.
On the left of the GUI, you have a number of options to manipulate the files the ability to convert stereo files to mono, to
compensate for stereo pan law settings, and to normalise the files, for example. There seems to be no undo facility, but you
are given the option to overwrite the original stereo versions of the files (in which case you get to create a backup if desired),
or to write processed versions to a new folder of your choice, leaving the originals untouched.
On the right is a table, whose columns offer information about each audio file. Beneath this is displayed a waveform of the
currently selected file and a play button, allowing you to audition any file before committing to processing. Speaking of which,
there are two steps: analysis and processing. Dropping the audio file folder structure from a large mix (88 tracks) revealed two
files that didnt need to be stereo it took about half a second to analyse each file on my 2013 MacBook Pro. Next to each of
the files, the Analyzed status was displayed, and the proposed stereo/mono conversion process No Conversion for most
and Convert Mono Left for the others, which presumably means extracting the left channel from the stereo file (there being no
difference between left and right, its hard to prove this by listening to the result!). However, if you click on that field, youre
presented with other options, such as Convert Mono Right, No Conversion, and Convert Split Stereo. You can click on the
column headers to sort the files by the source file format, proposed process and so on, which makes it easy to work your way
through a large number of files systematically.
In fact, the only thing that threw me initially is that the default GUI needs to be stretched to reveal all the information that it
contains. Making the GUI wider reveals two more columns, one which tells you the format of the source file (Mono File, Stereo
File or Two Mono Channels), and another which allows you to tick/untick a box for each individual file that instructs
StereoMonoizer to normalise the files.
This is an extremely well thought-out piece of software. While its always going to be possible to suggest potential additions
to the functionality of an application such as this, it not only does what it set out to do but does more than Id anticipated. Its
incredibly easy to use, too. It may not be the sexiest thing youll add to your studio, but at $49 for a single license, or $79 for a
two-installation license, if time is money or your computer resources are under pressure its well worth the price of entry. I
tested the Mac version, but as we were going to press a Windows 64-bit version was made available. Matt Houghton
Single license $49; dual license $79.
www.soundizers.com
.
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In this article:
Overview
Sable Strings Vols 1 &
2
Sable Strings Vol 3
Sable Strings Vol 4 &
Sable Ensembles
Mural Symphonic
Strings
Flute Family
Reeds
Low Reeds & Low
Winds
Horns & Trumpets
Trombones & Low
Brass
Conclusion
Alternatives
Efficiency Drive
Pricing
BML Volumes
News
Articles
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Spitfire Audio British Modular Library

Sample Library
Reviews : Sample / Sound / Song Library
Buy PDF
Spitfire Audio
British Modular Library
pros
Top players, expensive
instruments, opulent hall
acoustic.
Recorded from seven
different microphone
positions.
Contains some great
performances, including
excellent interval legatos.
cons
Has some tuning issues.
The articulation menu is not
fully comprehensive.
Currently has no solo
trombone.
summary
Its big, it sounds very posh
and its an established votewinner. No, were not talking
about Boris Johnson: this is
Spitfire Audios 20-volume
British Modular Library,
featuring a superior
collection of strings,
woodwind and brass
sections and solo
instruments recorded in AIR
Lyndhursts hall. For those
considering investing in an
orchestral sample library,
BML offers a buy-as-you-go
solution: beginners can dip
their toe in the water and
buy single volumes, while
committed pros will be
tempted to dive in and
acquire the whole collection.
information
See Pricing box.
www.spitfireaudio.com
Bass clarinet, one of the more exotic instruments in BML Low Winds Vol 1.
Spitfire Audios BML range packages a full orchestral library into 20 individual
modules.
Dave Stewart
ince January 2013 Spitfire Audio have been steadily developing their largest sampling project to date: a
comprehensive, modular collection of orchestral strings, brass and woodwinds, recorded in Londons AIR Studios from
multiple listening perspectives. Titled the British Modular Library (BML for short), this ongoing series currently
comprises 19 individual volumes containing solo instruments and sections of various sizes (see the BML Volumes box below
for a complete listing).
Upholding the companys proud-to-be-British traditions, BML features the best players our green and intermittently pleasant
land has to offer. In this case, rather than yet another example of the tedious, faux-patriotic nonsense currently disfiguring UK
politics, the British bias is justified: its an undisputable fact that UK players and studios are among the best in the world, and
the musical talent, lavish hall sound and excellent engineering featured in this collection have graced many acclaimed
recordings and film scores.
The BML range runs exclusively on the full version of NI Kontakt version 4.2.4 or 5, and will not work with the free Kontakt
Player. All volumes are available as downloads from Spitfire Audios site; the company also offers a hard-drive delivery service
for larger libraries and bundles.
Overview
In common with Spitfires other orchestral offerings, the samples were recorded in AIR Studios Lyndhurst Hall with players
seated in their correct concert positions. Consequently, BML instruments fit together in a balanced stereo picture with no need
for panning. Taking advantage of the halls hexagonal shape and high galleries, Spitfire recorded the samples from seven
perspectives: four of these positions (close, Decca Tree, outriggers and ambient) are included in each library as standard, with
the remainder due to be issued as free updates. The updates also include three additional mixes created by engineer Jake
Jackson, which include a symphonic presentation optimised for film soundtracks and a closer, more detailed mix for pop
production.
Refined over a number of years, the Kontakt GUI used in BML includes a microphone mixer and controls for legato interval
transition speed, round-robin behaviour, articulation loading/unloading and sample purging. The Ostinatum sequencer, a
quick and enjoyable way of creating repeated ostinato rhythm patterns, is also included, as is the ingenious Punch Cog
facility, which allows you to remove and/or tweak the tuning of any sample used in a patch.
Sable Strings Vols 1 & 2

The first BML title to see the light of day was Sable Strings Volume 1, released in early 2013. In contrast with the large
sections in their other string libraries (more on which later), Spitfire went small with Sable and sampled a chamber-sized group
optimised for nuance, definition and detail. Reflecting this desire for pointillistic focus, the Sable library takes its name from a
type of artists brush designed for fine, detailed work.
Sable Vol 1 (86.9GB) features four first violins and a separate,
three-player cello section playing essential articulations such as
long notes, short spiccatos, staccatos and legatos. I was pleased
to find the long notes have fast attacks, rendering them usable
for quick rhythmic playing as well as slower-moving pads. Con
sordino (muted) samples are a nice extra, and provided an
unexpected highlight: the cellos sordino sustains, which sound
majestic and imposing in AIRs lush hall acoustic.
The librarys excellent legato samples offer a choice between
bowed (featuring a bow attack on every note) and fingered,
which changes note without re-bowing. Legato portamentos
provide fantastic Bollywood-style slides, a perennially eargrabbing and charismatic delivery which cries out to be used in
pop arrangements. In a lighter vein, the flautando articulation is
an interesting, breathy and tender texture, while the high-pitched,
ghostly harmonics (which sound almost flute-like on the cellos)
are strikingly atmospheric.
The BMI Kontakt GUI contains a microphone mixer,

performance controls and the articulation stanza. To load or
unload articulations, simply click on the small rectangle
positioned underneath each note icon. Keyswitches (marked
in pink) allow on-the-fly changes between the different
playing styles.

Competitions!
While the short spiccato performances are agreeably light, brisk and precise, a more aggressive short-note style which
caught my ear was the violins short staccato dig, featuring strong, dramatic, fiercely propulsive bow attacks. I also
appreciated the lively tremolos and tone/semitone trills, which are played with great spirit and animation. Percussive Bartok
pizzicato and col legno hits (the latter produced by hitting the string with the back of the bow) are also enjoyably raucous and
robust the cello col legnos are particularly strong.
Despite the small player numbers and limited instrumentation, Sable Strings Vol 1 can produce surprisingly grand, fullsounding string arrangements spanning the vital C2 to C7 pitch range. However, for a full chamber strings line-up, prospective
BML buyers need to add Sable Vol 2. This 93.8GB companion library contains three second violins (different players from the
first violins), three violas and three double basses playing the same articulations (with a few omissions) as the instruments in
Vol 1. Though slightly less bright-sounding, expansive and assertive than the first violins, the second violins musical quality is
in no way inferior; in fact, their downplayed, more supportive approach provides the ideal foil for the first violins, and helps the
two sections to blend together.
I was impressed by the strength and emotional depth of the violas long notes, legatos and con sordinos, which sound like
they were played on very expensive instruments! My only minor criticism of this fine section is that it lacks an aggressive
delivery to match the first violins short staccato dig. The double basses play this confrontational style with great vigour, along
with some fabulous long notes and earth-shaking tremolos. On the downside, some of the basses pizzicato samples suffer
from unclean attacks and extraneous noises, but theyre weighty enough to pass muster in a mix.
Sable Strings Vol 3

Having created a full chamber strings orchestra with a basic range of playing styles, Spitfire pushed the boat out in Sable Vol
3 (70.5GB) and introduced a host of extended articulations for all five sections. Measured tremolos (fast, repeated 16th notes
played at 150bpm and a hurtling 180bpm) are an exciting new style, the perfect accompaniment for frantic on-screen action.
Kontakt 5 owners can activate the sync to tempo function and lock these energetic rhythmic performances to their song
tempo.
Sounding thin, metallic and a little scary, the sul ponticello
articulation has great horror-film and sound-design potential.
When combined with tremolo bowing (as in the first violins and
cellos tremolo sul pont patches), it adds a harsh, menacing
edge to a classic tension-building style. Use with care.
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Somewhat disappointingly, multi-octave runs and a small

selection of meandering mini-phrases come in a very limited
selection of keys. I was surprised to hear the second violins
ending up a semitone apart on the final note of one of their
ascending three-octave runs, but the basses make amends by
grunting out two-octave runs with commendable vigour and tight
timing. A set of lively single-octave ascending runs (which have
no descending version) are supplied in all 12 keys, but
unfortunately the second violins have been programmed a
semitone too low. These runs and phrases have no temposyncing facility.

explained
BMLs cimbasso duo. The cimbasso is a floor-standing,

contrabass brass instrument with a trombone-like sound.
A comprehensive effects section comprises moody tense longs (a slow, wheezy crescendo which builds into a hushed,
expectant, dynamically undulating sustain), a nerve-jangling collection of high-pitched, avant-garde noises, and some terrific
glissandos and slides. (In theory you can alter the slides direction with the mod wheel, but that didnt work in my review copy.)
Also included are entertaining disco falls (short notes with a quick descending slide on the end), evoking the golden age of
Saturday Night Fever, excessively tight white trousers and unfeasibly high, falsetto male vocals (presumably the last two have
a causal link?).
The first violins and cellos benefit from three new legato styles: the brilliant fast legato is optimised for rapid, leaping
phrases, legato runs work a treat for super-quick scales of intervals up to a minor third, and legato tremolo puts a smooth
sheen on tremolo melody lines. One caveat: these patches require the legato long-note samples from Vol 1 to work properly.
The two sections are further expanded by romantic, impassioned molto vibrato performances, some great, biting sforzando
marcato attacks and trills on minor 3rd and 4th intervals, which have a variable sample start point depending on how hard you
hit the key.
Sable Strings Vol 4 & Sable Ensembles

By June 2014 the Sable chamber strings series had grown to five volumes, making it the largest and most evolved part of the
Win Sontronics Aria

Valve Mic
Win Dynaudio & TC
Bundles
Monitoring System
BML library. Sable Vol 4 (9.7GB) provides an eclectic collection of extra articulations, including the expressive sul tasto
(bowed over the fingerboard). Whereas the second violins and cellos perform this romantic, dreamy style as an intimate
whisper (Spitfires words), the first violins and violas version is considerably more assertive.
An auxiliary sul G (aka sul C) legato articulation consists of legato intervals played exclusively on the instruments bottom
string. Of necessity this style has a restricted pitch range, but its sound is satisfyingly full and fruity, with a slight hint of
portamento. A straight sustained version is also included, featuring some rather sour tuning by the second violins. Muted sul
ponticello patches are a fairly extreme texture, thin and wiry like a hurdy gurdy; the tremolo version of this style is the ultimate
in edgy creepiness, but if you cant handle the tension, the regular muted tremolo bowing is far more mellow and musicalsounding. Though the basses dont play all of these new styles, Sable Vol 4 belatedly fills some gaps in their ranks by
providing flautandos, trills and regular sul ponticello samples.
Composers like myself who like to work with full strings patches will enjoy Sable Ensembles (26.8GB), which combines
Sables violins, violas, cellos and basses into eminently playable, great-sounding ensemble patches. Mapped and blended
over six octaves according to range, the sections play a set of basic articulations an excellent compositional resource.
Mural Symphonic Strings

With a little simple programming, its possible to make Sable Strings sound like a bigger section. However, if thats the sound
you generally hanker after, its simpler to use BML Mural Symphonic Strings, which sound that way straight out of the box.
Featuring 60 players (16 first violins, 14 second violins, 12 violas, 10 cellos and eight basses), the Mural sections immediately
create a full symphonic string sound. The only problem (if you can call it that) is that they can sound almost too big, and need
to be handled with care so as not to overpower other sounds in the track!
Mural Vol 1 (33.5GB) contains all five sections playing
essential articulations; Mural Vol 2 (30.6GB) adds important
playing styles, as well as filling gaps in the second violins and
violas articulations. The recently added 20.9GB Mural
Ensembles blends the sections into full-range patches, featuring
13 essential articulations. You can read my review of Mural Vols
1 & 2 in the October 2014 edition of SOS
(www.soundonsound.com/sos/oct14/articles/spitfire-audio.htm).
Flute Family
Spitfire Audios expert panel offers further control over

various expression functions, release trigger duration and
round-robin behaviour. The options list includes a button for
syncing the tempo of rhythmic performances such as BMLs
measured tremolos to your song tempo.
The first woodwind library in the BML catalogue, Flute Consort

Vol 1 (13.9GB), contains a solo flute and a flute duo playing essential articulations. You can combine their patches to emulate
the classic three-flute ensembles found in the first wave of orchestral sample libraries. This flexible ensemble-building
approach occurs throughout the BML range.
If you share my view that woodwinds are under-used in contemporary media composition, you can help redress the balance
by employing this librarys great solo flute. Its delivery is lyrical, elegant and expressive, with beautiful tuning and a sweet
vibrato: map a MIDI controller to the on-screen vibrato fader and you can start a note with no vibrato, then fade it up at will.
This is an easy, subtle and natural-sounding way of adding expression to flute parts.
The solo flutes legato samples do a fine job of smoothing out melody lines: the legato articulations patch automatically
switches between slurred and progressive legato styles according to your playing speed, with the progressive version
handling fast note transitions. For more rhythmic work, you cant do better than the flutes admirably precise and incisive short
staccatissimos.
Alternative deliveries include hollow long notes and upper-register harmonics (both of which have a primitive ethnic
flavour), and overblown samples which recall the jazzy flute stylings of 60s film scores. The multi-tongue articulation (two,
three or four quick repeated notes culminating in an end note) is very cool: the mod wheel controls the number of initial
repeats, while the length of the final note is governed by note velocity. Similarly, flute trills play their natural start on quiet
velocities, but cut straight into the trill when you play hard. The flute duo duplicates the solo instruments articulation menu
with no diminution in musical quality. One niggle: it seems the solo flute and flute duo short marcato sfz patches use the
same samples.
BML Additional Flutes (8.6GB) extends the flute family by adding alto flute and piccolo, which between them play most of
the articulations found in Flute Consort Vol 1. The alto flute sounds lovely, maintaining a sumptuous tone from its low register
up into the high range the hallmark of an exceptional player. Long notes are played with a delayed vibrato, while the legato
performances (which I could play all day) feature a more immediate vibrato style. The perky, piping piccolo has a nice, clear
and bright sound, handles melody lines with aplomb and can even do a good impersonation of an Irish whistle when you play
its legato patch quietly.
A final family member, the bass flute, can be found in BML Low
Winds Vol 1 (see below).
Reeds
The BML Reeds Vol 1 library (11.5GB) is a simple, straightforward presentation of oboe and Bb clarinet, both of which have
solo and a2 (classical-speak for two players) versions. Articulations are limited to long notes, legatos and short staccatos. I
like this solo oboe a lot. Its deliveries are elegant, lyrical and romantic, warm-toned but retaining that hint of angularity which
makes the oboe such a characterful and distinctive instrument. The solo instruments legato articulation is another treat: internote transitions are very smooth, and you can perform rapid twiddles, runs and grace notes with no break in the timbral
continuity.
A pair of out-of-tune oboes wouldnt be much fun, but thankfully that doesnt occur here; the two players manage to
precisely match their intonation, creating a strong, controlled unison timbre that works well both for short staccatos and legato
long notes. In contrast to the solo instruments expressive, slightly delayed vibrato, the two oboes use no vibrato at all. This
gives their long notes a slightly stern, serious atmosphere which would be very effective in solemn liturgical music.
The solo clarinet resonates beautifully in the AIR Lyndhurst hall, but its pure sweetness of tone is offset by some dodgy
tuning in its long notes, particularly noticeable on the F and F# above Middle C; move the mod wheel while holding a note,
and youll hear a tuning discrepancy as one dynamic layer crossfades with another. This problem (which occurs to a lesser extent on
a few other pitches) diminishes the usefulness of what would
otherwise be a very good instrument, so one hopes Spitfire will
address it before long. Thankfully, the clarinet duos short staccato
and long-note performances are nicely in tune. In keeping with
orchestral tradition, the clarinets perform long notes with no vibrato,
but a subtle (real) vibrato can be added with the on-screen fader.
Low Reeds & Low Winds

An essential volume for serious composers and arrangers, BML Low
Reeds Vol 1 (15.6GB) completes the basic orchestral woodwind lineup with cor anglais, solo bassoon, two bassoons and a
contrabassoon. BMLs cor anglais is another wonderful legato
instrument: it can play Charlie Parker bebop solos (and indeed, any
other fleet-footed melodic passages) with ease, and its plaintive,
slightly melancholy tone is an inspiration for melodic writing. The
bassoon also has very nice legatos, played with a subtle, restrained
and dignified vibrato, which can handle lines of any speed.
Accessible from the Kontakt GUI, the Advanced

Articulations feature lets you determine how
articulations are triggered. For example, you can use
MIDI CC values rather than keyswitches to select
articulations, a method favoured by Spitfire Audio
themselves.
The solo bassoonist obviously took the short part of short

staccatos to heart, as that particular patchs note durations are so
brief as to sound almost clipped. However, that can be remedied by
using the TM Shorts patch, which allows you to globally stretch
staccato note lengths. Alternatively, the bassoon duos brisk, urgent, well-played short staccatos avoid over-brevity and match
other BML instruments versions of that articulation.
Composers looking to extend their woodwind low register

would be well advised to check out the instruments in BML Low
Winds Vol 1 (10.6GB). With its languid, unhurried long notes,
fruity staccatos and fabulous legato performances (some of the
best in the BML range), the bass clarinet is a star turn. Standing
10 feet tall, the librarys contrabass clarinet doesnt do anything
in a hurry. Its short marcatos are the closest it gets to a fast
attack, while trying to play a bebop solo on its legato patch would
be a thankless task. But for big, fat stately pedal notes, this ultralow-pitched wind instrument is unbeatable.
The third instrument in Low Winds Vol 1 is bass flute. This
exotic instrument adds a breathy, velvety lower octave to the
flute familys range, enabling you to add bass notes to flute
chords and contrapuntal bass lines to your flute chorales!
BML Bone Phalanx features an ensemble of six trombones

(five players depicted here).
Horns & Trumpets

Moving on to BMLs brass, Horn Section Vol 1 (26.5GB) adopts the woodwinds format of combining a solo instrument and a
duo. The French horn duo gets the lions share of articulations, and delivers them with great confidence: sustains, slurred and
detached legatos are strong, steady and well-tuned, brassy cuivr performances sound impressively noble and the loud,
blasting bells up patch is positively regal. Effects (consisting of exuberant, whooping rips and uninhibited falls) are a welcome
bonus.
Though it has only five patches, the solo horn is highly effective and performs its simple deliveries immaculately. For a
bigger (dare I say epic?) sound, Spitfire offers Horn Phalanx (8.4GB), a classic name for a six-horn section. Its not known
whether the players marched around the studio in tight military formation, but the size, majesty and force of some articulations
certainly evokes the might of ancient Rome (or should that be Greece?).
The 12.3GB Trumpet Corps Vol 1 also contains a solo instrument and a duo. The soloist turns in a serviceable handful of
articulations, including brilliant-toned long notes and somewhat more reflective legatos. The duos collective tone is equally
bright, clear and assertive, a call to arms at loud dynamics and a calming presence when played softly. Quiet notes contain a
hint of vibrato which all but disappears in the louder dynamic layers; by contrast, the vibrato in the braying Mariachi delivery is
so strong you could practically trim a hedge with it. If you want galloping, rhythmic figures, the multi-tongue patch
automatically generates initial fast note repeats, or you can perform them yourself with the Fanfare Sculptor patch.
Six-player Trumpet Phalanx and Bone Phalanx volumes are also in the works, but at the time of writing the musicians had
not yet arrived from ancient Athens (or should that be Carthage?).
Trombones & Low Brass

Dem bones, dem bones, dem dry bones... well hardly, nothing recorded in AIR Lyndhursts reverb-enriched hall ever sounded
dry. BMLs trombone library, Bones Vol 1 (11.3GB), provides a tenor trombone duo and a separate section of two bass
trombones. (Solo tenor and bass trombones are due to appear in Vol 2.) I love the sound of the tenors. Their timbre ranges
from warm and mellow to big and grandiose, a rich, stately, spectacular and dynamic sound which is ideal for chord pads and
melody lines. Two minor issues: sustain pedal is disabled on long-note patches (easily fixed in Kontakts instrument options
window), and imperfect tuning takes the shine off some high long notes using the Punch Cog to remove the offending
round-robin samples improved matters.
The bass trombones duplicate the tenors articulations, reinforcing

the low register with their powerful, raspy deliveries. In the effects
department, both sections perform a set of lusty rips and falls, the
speed of which can be controlled by the on-screen speed slider.
Amusing effects glissandi are also included.
Adding a further low octave to the trombone range, the contrabass
trombone in BML Low Brass (27.1GB) contributes an impressive array
Best used with short-note articulations, the Ostinatum
of performances: its stupendously powerful, unlooped cuivr style
sequencer can be used to create repeated rhythms,
ostinato patterns and arpeggiated sequences.
reminds me of the glass-shattering, sub-bass blast of the mothership in
Close Encounters of the Third Kind. The librarys solo and duo
cimbassos (a floor-standing, valved contrabass brass instrument) make an interesting alternative to the bass trombone: their
timbre ranges from big, warm and comforting to a spiky, brassy shout (as heard in the fabulous zippy short staccatos).
Underpinning the whole shooting patch is a wonderfully rotund tuba, providing a great, solid foundation to BMLs brass
instruments.
Conclusion
Spitfire Audio certainly release a lot of material I sometimes wonder if these guys ever find time to eat and sleep, let alone
write music. If we factor in the companys initial license-only releases, one could say theyve come close to sampling an entire
orchestra twice, an achievement akin to building a scale model of both Houses of Parliament out of grains of rice.
With a comprehensive and flexible instrumentation, a brilliant team of players maintaining a high standard of performance
throughout and a great hall acoustic, BML is a force to be reckoned with. Admittedly it faces competition from overseas, but
without wishing to sound like a UKIP supporter, why go abroad when you can enjoy the best of British?
.
Alternatives
Most orchestral sample companies sell their wares in large, one-size-fits-all volumes, which can be overkill for anyone
looking to buy a limited selection of instruments. If youre in the market for a library that can be purchased in smaller,
modular chunks, an obvious alternative to BML is the Vienna Symphonic Library. Like the BML volumes, VSLs
instruments are available in themed bundles (the Vienna company also sell some individual instruments), and both
libraries benefit from interval legato sampling, great sound and superior musicianship. In contrast to Spitfires multi-miked,
reverberant hall approach, VSL favours a dryer studio acoustic and a stereo-only configuration, but weighed against that is
VSLs large and complete range of performance styles across all instruments.
Efficiency Drive
To save on disk space, Native Instruments lossless NCW compression is employed throughout the BML library: this
reduces the samples to half their original size (though the sound quality is identical). The GB figures in this review refer to
libraries installed sizes. At the time of writing the entire BML range uses 500GB of disk space, but with new releases in
the pipeline, anyone interested in buying the whole collection should invest in a 1TB drive.
Pricing
Strings
Sable Strings Vol 1 399
Sable Ensembles 249
Mural Strings Vol 1 399
Mural Strings Vol 2 399
Mural Ensembles 249
Woodwinds
Flute Consort Vol 1 199
Additional Flutes 169
Low Reeds Vol 1 169
Low Winds Vol 1 169
Brass
Horn Section Vol 1 169
Horn Phalanx 149
Trumpet Corps Vol 1 249
Bones Vol 1 279
Low Brass 245
Trumpet Phalanx TBC
Bone Phalanx TBC
BML Volumes
Catalogue Contents Number
Woodwinds
Flute Consort Vol 1 BML 101 Solo flute, two flutes.
Additional Flutes BML 102 Piccolo, alto flute.
Low Winds Vol 1 BML 103 Bass flute, bass clarinet, contrabass clarinet.
Low Reeds Vol 1 BML 104 Cor anglais, solo bassoon, two bassoons, contrabassoon.
Reeds Vol 1 BML 105 Solo oboe, two oboes, solo clarinet, two clarinets.
Brass
Horn Section Vol 1 BML 201 Solo French horn, two French horns.
Trumpet Corps Vol 1 BML 202 Solo trumpet, two trumpets.
Low Brass BML 203 Tuba, solo cimbasso, two cimbassos, contrabass trombone.
Bones Vol 1 BML 204 Two tenor trombones, two bass trombones.
Trumpet Phalanx BML 209 Six trumpets.
Bone Phalanx BML 210 Six trombones (three tenor, two bass, one contrabass).
Horn Phalanx BML 211 Six French horns.
Strings
Sable Vol 1 BML 301 Four first violins, three cellos (essential articulations).
Sable Vol 2 BML 302 Three second violins, three violas, three double basses (essential articulations).
Sable Vol 3 BML 303 All five sections (extended articulations).
Sable Vol 4 BML 304 All five sections (additional extended articulations).
Sable Ensembles BML 305 Blended ensembles.
Mural Vol 1 BML 308 Sixteen 1st violins, 14 2nd violins, 12 violas, 10 cellos, eight double basses.
Mural Vol 2 BML 309 As above (additional articulations).
Mural Ensembles BML 310 Blended ensembles.
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TC-Helicon Ditto Mic Looper

Microphone Effects Pedal
Reviews : Effects
Paul White

For such a simple-looking box, decked out with one knob, two buttons and no display beyond a single
LED, the TC-Helicon Ditto Mic Looper is a very sophisticated piece of kit. Designed specifically for looping vocals or miked
instruments, the pedal has an integral mic preamp with phantom power, while compliance with TCs Mic Control protocol
allows looping functions to be controlled directly from the switch on a compatible microphone. Conversion is 24-bit and power
comes from an included universal mains 12V DC power adaptor. The mic-level analogue inputs and outputs are both via
balanced XLR connectors and theres also a micro-USB connector for firmware updates. Its worth pointing out that the
hardware is also very robust, with a metal case, confidence-inspiring switches and a nice, grippy rubber base.
An automatic mic-gain function means theres no need to set up gain trims
manually. Operationally, the paradigm couldnt be easier: press the Loop switch to
record, then press again to replay the loop. You hold it down to undo or redo the
last overdub, and the knob sets the level at which loops are played back. The
Loop-switch actions can also be configured by the user to be Record, Play,
Overdub (the default mode) or Record, Overdub, Play. The maximum loop
storage time is five minutes, and your loop is recorded and stored when the unit is
powered off. Switching modes is done using a power up while holding buttons
down sequence. Stop is used to start and stop playback, or you can hold it down
to both stop playback and perform a silent erase (erasing takes three seconds).
As loop recording is essentially a sound-on-sound process, you can build up as
many layers as you want. The LED colour shows whether you are recording (red)
or playing back (green). Having the option to undo the last take is very useful, and
has the potential to be used as a performance tool as well as a corrective one.
The Ditto Mic Looper is a breeze to use and, aside from its obvious humanbeatbox applications, can be used to loop anything you can pick up via a mic. And
if thats too limiting, you can always use a DI box if you want to capture a linelevel or instrument source. The audio sounds clean, even when youve added
multiple layers, and theres no audible glitching as the loop comes around. The
auto level setting also works very nicely in maintaining a healthy level while
avoiding clipping, so you really can just plug and play. While there are more sophisticated loopers on the market, I like the way
the Ditto Mic Looper sticks to the essentials youre so much less likely to get into a mess when using it! .
$150
www.tc-helicon.com
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Ueberschall Flugelhorn & Trumpet 2

Elastik
John Walden

You get two for the price of one with this review, with two closely related sample libraries from
Ueberschall: Flugelhorn and Trumpet 2. Unlike the majority of the Elastikbased titles from Ueberschall, these two releases
dont follow the songbased construction-kit format but, instead, are part of their Instrument Series. What you get in each
library, therefore, are a series of solo instrument phrases/licks that you can piece together to form a complete performance.
You are not left completely without guidance, however, as within each library, the various phrases are organised into some
sensibly structured folders based upon tempo and, in the case of the slightly larger Trumpet 2 library, also on the basis of the
different mutes bucket, cup, plunger, wah-wah, etc used with the trumpet.
Flugelhorn provides just under 1GB of samples spread across
700 loops/phrases, while Trumpet 2 runs to 1.4GB and over
1100 individual samples. In terms of musical style, think anything
from classic 1930s jazz at tempos that go from a sensual 65bpm
up to a more frenetic (and perhaps a little more modern?)
150bpm, although Elastik really does make it possible to
stretch/compress the material over some considerable tempos
ranges.
Whatever tempo you explore, the playing is fabulous. Both
instruments are played by Gary Winters and, yes, he really can
play rather well. The sound is wonderful both in terms of the
actual tone captured and the recording quality.
The Flugelhorn is somewhat darker and mellower than the
Trumpet; close your eyes, listen to the sound and you could be in
some smokefilled, New Orleans jazz club. There is perhaps a
touch more variety within Trumpet 2; you still get that 30s vibe in
places but you can also come right up to the modern day.
In use, both libraries actually make it pretty easy to chain a few
of the phrases from a particular subfolder together to construct
a complete solo performance. Indeed, the samples are good enough to stand on their own as a solo performance, but add in a
suitable drum and upright bass backing (just make sure they dont get in the way) and something rather wonderful is almost
inevitable.
Both of these libraries might cover something of a specialist musical genre but, within that genre, they really nail the sound
and performance. I could imagine busy media composers finding Flugelhorn and Trumpet 2 to be excellent resources when a
little bit of atmospheric and utterly convincing solo brass is needed in an instant. Classy playing, classy sound; listen out
for them in a movie soundtrack near you.
.
Flugelhorn 49, Trumpet 2 99.
www.ueberschall.com
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In this article:
Sound Machine Wood
Works
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UAD Sound Machine Wood Works

Formats: Mac & PC UAD2
Paul White

The latest v8 update to Universal Audios UAD platform brings with it full Yosemite OS support for Mac,
a raft of new plug-ins and a new version of Console allowing multiple Apollo devices to be combined to behave as a single
device. The new plug-ins, which run on UAD2, Satellite or Apollo devices, include the Distortion Essentials Plug-In Bundle, the
Friedman Amplifiers Plug-In Collection and Wood Works. Distortion Essentials models three classic overdrive pedals including
the Ibanez Tube Screamer, while the Friedman Amplifiers Plug-In models a couple of British-voiced amplifiers that have a
distinctly Milton Keynes flavour about them. With Apollo hardware these can be used in real time with no latency penalty, as
can Wood Works, which was created by UA development partners Sound Machine.
The nearest hardware equivalent to Wood Works is probably the Fishman Aura pedal, and both have a specific aim: to
improve the sound of a DId acoustic guitar fitted with a pickup system by remodelling the body resonances that are
suppressed when under-saddle pickups or magnetic soundhole pickups are used.
Behind all the GUIs fancy retro wood and brass, Wood Works
has relatively few controls, making it easy to set up. A rotary
switch selects from 16 modelled guitar body types, four
designated as Studio (small-bodied guitars), eight as
Dreadnoughts and four as Jumbos. The other clever part of the
plug-in is that it further processes the signal to split it into what
you might expect to hear from a mic over the neck and another
over the body, with separate level and pan controls for each. You
can choose whether the Neck and Body level controls are reset
to factory default positions when you change guitar types, or left
in their last positions. Either way youre still free to make
subsequent manual adjustments. To the right of the plug-in are
input and output level controls plus an output level meter with a
bypass switch. And really thats all there is to it.
Theres no explanation of the underlying technology other than
that it uses a form of modelling, but whatever is going on, it isnt
just simple EQ but more like the complexity youd hear from a
fingerprint EQ that has hundreds of EQ bands. In any event,
Wood Works does a pretty good job of injecting believable
character into the sound. Not only can you completely change the character of the guitar, but Wood Works also lets you
rebalance the virtual neck and body mics as you mix with the added benefit that, if inserted on a stereo track, these can be
panned apart to create a sense of stereo spread.
In practice, the best way to use Wood Works is simply to spin the dial to find the guitar type that sounds best for the song
you are working on, then balance the neck and body levels to taste. You can always add further EQ with another plug-in if you
need to, just as you might when miking a guitar, and a little added artificial room ambience also helps give the end result a
sense of existing in a real space. In fact the only control I felt might be usefully added is one to morph between your original
guitar sound and the modelled sound for those occasions where the modelled sound might sound a little overdone. Having
said that, I used the plug-in to revisit some old songs Id recorded with a band I play with, and in all cases, the DId acoustic
guitar sound worked better in the mix after processing.
A side-benefit of the modelling process is that in addition to creating more of a miked-up tone, the brittle attack of a typical
piezo pickup is somehow softened but without losing any of the essential sonic sparkle. Some of the smaller Dreadnought
models also do a passable job of turning a clean, DId electric guitar sound into a pseudo-acoustic, and if the guitar is fitted
with piezo saddles, then the illusion gets even closer. Overall then Wood Works is a genuinely useful tool with a learning curve
thats so flat you could cycle up it in top gear.
.
$299
www.uaudio.com
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The Sound Of Poland B11X

Kontakt Instrument
Tom Flint

TSOPs first product is a Kontakt instrument called B11X, which was created by sampling every key
and sound generator of a rare Polish synthesizer made in the 1980s. Apparently the source instruments had been designed
solely by Polish engineers and assembled from Polish electronic components, plastic and copper!
TSOPs emulation is a faithful copy in many regards, but the designers have taken the liberty of making some improvements,
such as adding extra effects, as well as MIDI control over the parameters.
At B11Xs core is the Instrument page which has a strange mix of sound options and effects. Firstly there is a Solo panel with
Trombone, Trompette, Clarinet, Violin and String options, all with switches, volume controls and tune knobs. None of the
options sound much like the acoustic instruments they are named after, but they are all full of character nonetheless.
Smoother, but no more realistic, sounds can be created using the Flute section, which offers five tuneable channels/registers.
There is also a Percussion section for adding a bit of bite to the sound, and
there are effects such as Tremolo, WhaWha and Vibrato. A kind of mixer
panel makes it possible to adjust the relative levels of the Flute, Solo, Tremolo
and Percussion sections, and altering their balance in this way can have a
dramatic affect on the overall sound.
Further effects are found on a page headed by a Filter Matrix section that
includes a panel of switches for assigning the Flute, Solo and Percussion
channels to either one of two filters. Below the Matrix panel are Phaser,
Chorus, Delay and Amp processors, collectively providing plenty of sound
expanding and thickening options.
The most interesting page, though, is found on the Applications tab. Here there
are a set of userdefinable keyswitch memories, plus something called a Block
Chord Harmonizer, which allows the user to select a chord type, be it a minor,
major, seventh, ninth, quartal or octave, and then shift the octave and level of
the voices used in the chord.
Finally, clicking on the Presets tab brings up a rather lovely menu page where it
is possible to select a musical genre, such as Classic Pop, Movie Soundtrack,
or HipHop, to name just three of the 10 options, and then pick a type of
sound, the options being Pad, Lead, Organ, Piano, Mallet, Bass, Stab, Noise
and Soundscape. Sounds that appeal most can be saved to a 10slot Favourites menu, and there is another panel for storing
usercreated presets.
Frustratingly, the various presets and favourites menus cant be accessed when the other tabs are selected.
B11X is unlike any synth Ive come across recently, which cant be a bad thing. Its controls are clear and simple, but
altogether make a rather complex tool, capable of generating a reasonably wide range of interesting sounds. Although the
modelling is based on an 80s synth, its sound reminded me of something older. I eventually decided it was the domineering
Dudley Simpson score for the 1971 Doctor Who serial The Mind of Evil, memorable for its aggressive detuned and modulated
synth lines. Similarly, one Piano-type sound evoked memories of an early Tomita record! Synth enthusiasts should definitely
take a look. .
$99
www.tsop.pl
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Waves Dbx 160
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Waves Dbx 160

Paul White

Designed by engineer David Blackmer in the early 1970s, the Dbx 160 series of compressors have
enjoyed continued popularity, especially when working with drums, because their tight and assertive character helps make
things sound big and punchy. The original Dbx 160 was a hard-knee compressor; later variants included soft-knee versions,
but Waves emulation is modelled on the original hard-knee design. Blackmers design used a VCA built from discrete
components as its gain-control element. This VCA was later produced as an integrated circuit by THAT Corporation, and has
since been widely used in compressors and VCA-controlled automated consoles. Blackmers innovations also include the
RMS side-chain detection circuitry used to drive the feed-forward gain control configuration of the Dbx 160.
HYBRID
STUDIOS
There are already digital emulations of the Dbx 160, not least on
Universal Audios UAD platform, but Waves have collaborated with Dbx to
extend its capabilities with a few up-to-date additions that can be
accessed by means of a separate drop-down panel. These features
include a Mid/Sides matrix, a wet/dry mix control for setting up parallel
compression, noise-floor control and a choice of mono or stereo
operation. All the common Windows and Mac OS plug-in platforms are
supported, and authorisation can be to a specific computer or any
suitable USB memory stick.
The adoption of auto attack and release times that respond to the
envelope of the input signal makes the Dbx 160 very simple to set up, as
other than adjusting the output level, theres really only the Threshold and
Compression controls to think about. The compression ratio can be set
from from 1:1 through to hard limiting, with a maximum gain-reduction
capability of over 60dB, meaning that the Dbx 160 can manage anything
from gentle audio glue compression to seriously slamming drums or
pumping dance mixes.
In addition to having the standard Waves plug-in features relating to
saving, loading and comparing, the interface draws on the cosmetics of
the original with its wooden side cheeks, large VU meter and radio-style
buttons for switching the meter to read input, output or gain reduction levels. There are just three control knobs, labelled
Threshold, Compression in essence a ratio control and Output Gain. An external side-chain feed can be selected where
appropriate, while a Collapse button toggles the visibility of a lower panel containing the additional parameters. These include
input level adjustment, wet/dry balance and a knob that simulates the circuit noise of the original, should you feel it is needed
for the sake of authenticity.
Waves have also added switchable side-chain low-cut filters that are useful in situations where the bass end of the sound is
having too much say in how much gain reduction is being applied. Three further radio buttons select mono, stereo or Mid/Side
operation with four more to set the monitor mode, which can be set to Left, Right, Mono or Stereo.
Despite the minimal set of controls, the Waves team have managed to create numerous factory presets. These make good
starting points, though youll invariably need to adjust the Threshold control to suit your recorded material. At the polite end of
the scale with compression ratios of 1.5:1 or less, a low Threshold setting produces a subtle glueing of the mix elements
without making any really obvious change. As the ratio is increased, the style of compression becomes more assertive just
as youd expect from this particular compressor. What impresses is the tight, punchy character of the compression, and at the
same time, the high-frequency elements in the material never seem to suffer as they do with some other compressors. The
Dbx 160 is often talked about as a drum compressor, and I have to admit that drums is what it does best, though its also
worth trying on bass parts, electric guitar and even vocals. There are plenty of worthy compressors out there capable of acting
as mix glue, but if you need to add attitude to a drum part or even to a whole mix, the Dbx 160 really comes into its own.
None of Waves new controls gets in the way if all you want is old-school authenticity, but in a real-life mixing situation, they
prove very welcome in setting up parallel compression or applying M/S processing to stereo mixes where centre-panned
signals need a different compression treatment from side-panned material. Waves are to be applauded for getting so close to
the character of the real Dbx 160 while not shying away from adding the extra controls to make the compressor more relevant
to the way mixes are made today. If youre a fan of the hardware version, I dont think youll be disappointed by this plug-in.
.
$150
www.waves.com
ORANGE COUNTY, CA
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Black Boxes
Off Grid
Getting Started
Silly Wizard
Second That eMotion
Racks & Racks
Who Benefits?
Up Scale
Conclusion
Alternatives
Multirack
Digigrid & HDX
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Waves/Digico Digigrid
Networked Audio Infrastructure
Reviews : Computer Recording System
Buy PDF
Waves/Digico
Digigrid
pros
A scalable system with
something to offer everyone
from serious home studios
to large multiroom
complexes.
SoundGrid Studio offers
flexible cue mixing and the
ability to monitor and record
through Waves plugins at
minimal latency.
Can be used simultaneously
by multiple engineers, even
if theyre running different
DAWs.
StudioRack provides some
neat plugin chaining
facilities, and the ability to
run in native mode means
projects can be worked on
away from the host system.
Plugin chains can be
transferred seamlessly
between studio and
compatible livesound
setups.
cons
The complexities of the
Digigrid system are present
at all levels, but some of its
advantages will only be
apparent in large setups.
There are gaps in the initial
hardware product range,
most notably for a monitor
controller.
No thirdparty plugins
available yet, and no
surround support at present.
Though it works well on
Windows, theres no MME
or WDM driver to support
nonASIOcompliant
applications or the Windows
OS itself.
summary
Waves and Digico have
drawn on their shared
expertise to create a
versatile and well thought
out modular product range.
The scalability and
networking features arent
relevant to everyone, and
come at a cost in terms of
simplicity, but they mean
youll never outgrow a
Digigrid system.
information
See Pricing box.
Waves +1 865 909 9200
ext. 2
sales@waves.com
www.waves.com
The five rackmounting members of the Digigrid hardware interface range are, from top, the DLI and DLS, which are
specifically designed to connect with Pro Tools HDX systems; the versatile IOX which, with its plentiful analogue I/O, is
intended to live in a live room; the IOC, which offers mainly linelevel and digital I/O; and the IOS, which includes a builtin
SoundGrid server and a broad palette of connection options.
Two heavy hitters in the world of digital audio have come up with an intriguing new
synthesis of hardware and software.
Sam Inglis
rom string libraries to sealing wax, Sound On Sound has evaluated a great variety of products over the years. But as far
as I can remember, weve never before reviewed a scalable network infrastructure. In fact, had anyone asked me what
a scalable network infrastructure was, Id have guessed it was something from The Thick Of It.
This particular scalable network infrastructure is a collaboration between two big names in digital audio: Waves, who have
created the software component, and Digico, who supply the hardware. And although the three words in question make for a
slightly indigestible mouthful, they do in fact describe the Digigrid system well.
A Digigrid system is made up of three types of physical element: computers running Mac OS or Windows; audio interfaces;
and DSP servers. These are connected physically by Ethernet cables, and functionally by Waves SoundGrid Studio software,
which handles routing, monitoring and audio processing. Its an infrastructure because its designed to be permanently
installed at the heart of a studio. Its a network in the sense that its features can be shared and used by any connected
computer (with the limitation that individual DSP servers cant split their processing between more than one user). And its
scalable because you can add more of the physical components at any time. If, for instance, you find yourself running out of
audio inputs or DSP horsepower, you can simply buy or rent another Digigrid unit, power it up and attach it to the network. If
more users want access to your Digigrid network, then as long as they have the software installed and licensed on their
computers, they too can simply plug in a Cat 5e or Cat 6 cable.
Black Boxes
To start with the physical side of things, the current Digigrid range includes a total of seven Digico audio interfaces. Of these,
four have no analogue I/O at all: the MGO and MGB are portable devices designed primarily to enable a laptop to hook into
the MADI output of a live-sound rig or similar, while the DLS and DLI are intended to allow a Digigrid network to interface with
Pro Tools HD, HDX and HD Native systems (see box).
Assuming you run a nonHDequipped studio, that leaves a palette of three multipurpose interfaces to choose from. With
12 mic/line inputs on combi XLR connectors, six line outs on quarterinch jacks and four independent stereo headphone outs,
the 1U rackmounting Digigrid IOX is designed to live in your live room, piping miked signals into the network and cue mixes
back to the musicians. The IOC, by contrast, is intended for connection to other manufacturers preamps or AD converters: it
has two frontpanel headphone sockets and two mic/line inputs, but the bulk of its I/O is on DSub connectors, providing 16
channels of AES3 digital and eight analogue line inputs and outputs, along with word clock and a single optical ADAT in and
out.
Finally, the Swiss Army knife of the range is the IOS, a 2U device that offers all you need to set up a smallscale Digigrid
system. It features eight mic/line inputs, eight line outs that are mirrored on DSub and quarterinch jacks and a pair of front
panel headphone outputs plus MIDI, word clock and AES3 digital I/O. Like the Digigrid DLS, but unlike any of the other
models, the IOS has a SoundGrid DSP server built in, so you can set up a complete Digigrid system with just an IOS, a
www.digigrid.net
computer and an Ethernet cable.

If you have sufficient I/O already but want to add further DSP
grunt, you have the option of three Waves hardware products:
the portable SoundGrid Impact Server, the standard Server One
and the more powerful Extreme Server. All three are, basically,
embedded PCs. Live-sound engineers looking to connect digital
The Digigrid IOS offers eight combi mic/line inputs and eight
consoles from Digico, Yamaha and Allen & Heath can also
line outs, mirrored on jacks and a DSub connector.
choose from a range of appropriate expansion cards for those
Connections with the SoundGrid network are made using its
four RJ45 Ethernet sockets.
desks. A Digigrid network can consist of up to 512 channels at
48kHz and below, or 256 at 88.1 and 96 kHz. These can be
divided between inputs and outputs in any way you like, and controlled from any attached computer, but the ASIO and Core
Audio drivers are currently restricted to 128 channels, so you cant at present have a combined input and output count greater
than this on any one computer.
Off Grid
Turning to the software dimension, whats not included is any sort of recording or playback program. Instead, Digigrid is
designed to be compatible with any DAW that supports ASIO or Core Audio, as well as with Pro Tools HD, HDX and HD
Native. Its software functionality is split between two main components: a standalone application called SoundGrid Studio,
and a plugin called StudioRack. Users will also want to buy some SoundGridenabled Waves processing plugins in order to
take advantage of the Digigrid DSP resources.
The StudioRack plugin does not process audio, beyond allowing you to trim input and output levels. However, it is itself a
plugin host or chainer, with eight sequential insert slots that can accommodate any SoundGridenabled Waves plugins
installed in your system. The clever part is that this chain of insert processors can run either natively, using your computers
CPU resources, or remotely on the Digigrid DSP server. A simple dropdown at the top of the StudioRack window lets you
switch just the instance thats open, or all instances at once. If you open a project that uses StudioRack on a computer that
isnt attached to a Digigrid server, all instances will automatically be switched to native mode. So, for instance, you can track
or mix a project at the studio with processing offloaded to the server, then take it away on a laptop for further work, then bring
it back again, with minimal interruption.
Any StudioRacks that are in DSP mode also show up within the SoundGrid application, which is centred around a mixing
utility called eMotion ST. The key advantage of this is that StudioRacks in DSP mode can be monitored directly, without the
additional latency incurred by your DAWs buffer settings. So if your singer needs to hear her voice compressed, equalised
and reverberated, or youre a guitarist playing through Waves GTR amp simulator, you can hear the processed sound without
any distracting delays. Whats more, with a little bit of configuration, its possible to have input monitoring switched
automatically when you engage Record in your DAW.
Getting Started
For this review, Waves and Digico loaned me a Digigrid IOS along with temporary licences both for the SoundGrid software
and a selection of Waves plugins. Costing $3760 without factoring in the plugins, computer or DAW software, this is pretty
much the smallest and simplest DSPbased Digigrid system with I/O that you can buy: the scalability from this point is all
oneway!
Connecting everything up is supremely simple, and even if you dont need the networking side of things, the advantages of
Ethernet as a connection protocol are obvious: cables are cheap and plentiful, and cable runs can be pretty much as long as
you like. The review IOS had a rather annoying cooling fan, but production units will be fitted with a fan controller which should
improve matters; Digico sent me one to retrofit at the end of the review period, and it did indeed eliminate most of the noise. In
any case, its feasible to hide the IOS away in a machine room, as all of its features are controlled remotely from the
SoundGrid Studio utility. Hardware metering is minimal, so the only thing youd really miss are its headphone outputs (which
are, incidentally, bloody loud!).
Digico describe the IOSs mic preamps as awardwinning, but although I like the fact that phantom power can be switched
individually and gain set to within half a dB from SoundGrid Studio, they are more functional than flashy. The gain range is
restricted to the usual 060dB found on most budget interface preamps, there are no pads or highpass filters and, sonically,
they are transparent rather than characterful. Theyre perfectly usable, but Id have though most serious studios would want to
augment them with something a bit more sexy.
Its worth pointing out that in this sort of configuration, the IOS works perfectly well as a standard ASIO interface. All of its
inputs and outputs were visible and accessible in my DAW software, and buffer sizes right down to 32 samples are supported,
so if you want to keep things simple, its possible to run a recording session quite happily without using the Mixer or
StudioRacks features of SoundGrid Studio at all. However, Windows users should note that there is no MME or WDM driver
for SoundGrid or the IOS. It thus doesnt show up as an option in the Playback or Recording pages of the Windows Control
Panel, and you cant route the output of nonASIO programs such as iTunes into it.
Silly Wizard
When you first install the SoundGrid software, it runs a Wizard which attempts to configure the system and routing to your
needs. I suspect I must have done something wrong at this point, because the DSP server within the IOS did not get added to
my system, and I spent really quite a long time trying to figure out why I couldnt use many of the features of SoundGrid
Studio. Once Id visited the Setup page and added it as a Server in the System Inventorys Device Racks, it sprang to life and
I had no further problems.
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Although SoundGrid Studio boasts a commendably large,

clear and friendly graphical interface, however, configuring a
SoundGrid system manually is not for the fainthearted. This is
especially true if you need to venture into the Patch page, where
youll find four separate routing matrices that interact in ways
which are not always obvious. It would be easier if the large,
clear and friendly graphical interface could be resized I dont
like to think about the amount of scrolling youd have to do in a
large Digigrid patching setup with multiple I/O units and servers.
Fortunately, this should be the sort of thing you can forget, once
set, and the preset arrangement worked fine for me.
The System Inventory page lets you see what devices are
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Second That eMotion
The System Inventory page lets you see what devices are
connected to a Digigrid network. In this case the Digigrid IOS
is supplying both my Server and my Hardware I/O Device.
Most of your time with SoundGrid Studio is spent in the eMotion

ST Mixer page. As well as utility areas for configuring the preamps and setting channel input and output routings (which are
mirrored in the Patch page), eMotion ST provides two mixer layers. Typically, inputs that you are recording into your DAW
project would be accessed via the StudioRacks layer, which well come to in a minute, while everything else is visible in the
main Mixer layer.
As well as the Preamp and Route pages mentioned above, this has two main modes. Rack mode gives you access to eight
insert slots per channel, each of which can be populated by an individual SoundGridenabled plugin. This opens up a
number of possibilities. For instance, plugins applied to input channels process the sound prior to recording, so if for instance
you want to record guitar through an amp simulator or bass through a compressor, you could do so here. Alternatively, if for
instance you wanted to use a metering or monitor calibration plugin across your outputs without affecting the mixdown within
your DAW, youd insert it on an output channel.
The other main mixer mode is Sends, which is fairly self
explanatory, and makes it easy to create up to eight different cue
mixes routed to different destinations. A nice touch is that
selecting an output channel on the left side of the mixer screen
flips the faders so that they control the aux send level to that
destination rather than the main output. The whole mixer turns
blue as a friendly reminder that youve done this.
Racks & Racks

The eMotion ST mixers StudioRacks button is, by default,
greyed out. To make it active, you need to have open a DAW
project containing one or more StudioRack plugins that are set
to run on the SoundGrid DSP server.
The eMotion ST window has two layers. In the main Mixer

layer, shown here, you can use the Sends page to create up
to eight separate cue mixes, as shown; alternatively, the
Preamp, Rack and Route buttons select other views.
The StudioRack plugin window consists of several

functionally independent sections. At the bottom left are input and output faders; these have metering superimposed, and
there is also a third meter labelled Dyn which can report the gain reduction being applied by any of the dynamics plugins
inserted within that instance of StudioRack. Plugins themselves are inserted within the column of eight slots to the right of
this area. A miniature version of the plugin interface appears within the slot, and a further area to the right can display the full
GUI for whichever of the plugins is currently selected. Above the faders are visible four assignable rotary controls, each of
which can connect a knob on an attached MIDI controller to the plugin parameter of your choice.
Perhaps the most important group of controls, however, are those found at the top left of the StudioRack window. The
Processing popup lets you switch the current instance, or all instances in your DAW project, between native and DSP
operation. As Ive already mentioned, StudioRacks running in native mode dont show up in eMotion ST, and within the plugin
window, the eMotion Monitoring controls are greyed out unless in DSP mode.
To set up lowlatency input monitoring, you first need to tell the StudioRack plugin which physical input to your DAW its
controlling, and give it an appropriate name. For example, supposing you were recording a stereo keyboard track into Pro
Tools via line inputs 5/6 on the interface, youd need to make sure to select that input pair within the StudioRack plugin on the
keyboard track, and enter a suitable name within the plugin. Setting all this up for a large session could be a bit of a faff, but
with DAW templates and so on it shouldnt be something you need to do often.
Once youve done this, each StudioRack plugin can be switched between Playback and Input modes. In the former, it
takes as its input the recorded signal from the DAW track; in the latter, its signal is taken directly from the selected audio
interface input, bypassing the DAWs input buffer. This makes it possible to monitor the input through any processing plugins
applied within the StudioRack plugin but with minimal latency. Pressing the Grp button adds the open instance to a group
that can be switched in one go. This is handy for situations where you are working with a lot of inputs, but not nearly as handy
as the Auto option. Making this work requires adding Waves SoundGrid StudioRack as an HUI controller, but this is a matter
of moments to set up, and once its done, StudioRacks in Auto are automatically switched to Playback during playback, and
Input in stop or record. Neat.
Finally, hitting the Sends button on the StudioRack plugin displays all eight eMotion ST aux sends for that channel, so you
can adjust its contribution to each cue mix without leaving your DAW. For the most part, in fact, the choice of whether to use
eMotion ST or to work entirely in your DAW depends on your preferences. If you want to add or modify plugins in a
StudioRack, youll need to do it in your DAW; if you want an ataglance overview of all active StudioRacks and their
contribution to your cue mixes, eMotion ST is the place to do it. Otherwise, most of the actions youd need to take during
recording can be performed equally well in both.
From a mixing point of view, the plugin chaining functionality
in StudioRack has some neat advantages. Entire StudioRack
configurations can be saved and reloaded as presets that are
independent of your DAW, so in a studio where multiple users
who favour different DAWs collaborate, this would make it much
easier to transfer settings and projects. Its also possible to
automate individual plugin parameters, and each instance of
StudioRack has a sidechain input that can be shared among
any or all of the plugins it hosts. On the down side, if you want
to perform an offline bounce within your DAW, you need to
switch all StudioRacks to native mode. Otherwise, as I found to
my cost, youll get silence.
Who Benefits?
The other main eMotion ST view is StudioRacks. In this

mode, any StudioRack plugins within your DAW project that
are currently set to DSP mode are visible as mixer channels,
allowing you to create lowlatency cue mixes from inputs.
As Ive already mentioned, the review system supplied by Waves

and Digico is pretty much the simplest complete SoundGrid configuration available that incorporates a DSP server. Its
interesting to contrast this system with something like Universal Audios Apollo, which is similarly priced and likewise offers
highquality DSPbased plugin processing, but is much more narrowly targeted at the home and projectstudio market. In
comparison with eMotion ST, the UA Console utility has several obvious limitations: plugin processing can only be switched
globally between the record and monitor paths, and theres no real integration into your DAWs mixer. Consolebased cue
mixing cant be done from within the DAW, and if you want to employ the same plugin configurations at the mix stage, they
need to be manually reloaded into your DAW. Yet for this market, UAs approach also has some major advantages. Because
it doesnt offer the same scalability or flexibility, its far simpler to learn and use; and the hardware elements of the Apollo
systems ooze glamour in a way that Digicos utilitarian black boxes dont.
If equipping a slightly more upmarket singleroom studio, you
might ask yourself what a Digigrid system would offer as an
alternative to a Pro Tools HDX rig based around, say, an HDX
Core PCIe card with an HD Omni interface. In terms of bang for
the buck, at least, the Waves/Digico solution looks very tempting,
delivering more I/O and quite possibly more DSP plugin
resources for less than half the price, with its scalability opening
up more varied prospects for future expansion. On the down side,
setting up lowlatency monitoring and cue mixing using
StudioRacks and eMotion ST is significantly more complex and
slower than doing it all within the Pro Tools HD mixer. The HD
Omni also builds in a powerful and sophisticated monitor
controller, something which is currently absent from the Digigrid
range. (You do, of course, have the option of combining Digigrid
with something like an HD Native system, for the best of both
worlds.)
One further way in which SoundGrid differs from the Avid and
UA systems should be highlighted. Both of the latter run their
plugins on dedicated DSP chips, and the plugins themselves
The StudioRack plugin running in DSP mode in Pro Tools.
are coded in such a way that their DSP load is always known in
As Auto monitoring mode is active and the transport is
advance. This means theres a fixed limit to the number of plug
stopped, it is currently processing the signal arriving at its
input, bypassing the DAWs input and output buffers. The
ins that you can run on an HDX card or Satellite, but that within
Sends panel is open, and its settings mirror those for the
that limit, performance is guaranteed. The same is not true of the
relevant channel in eMotion STs StudioRacks page.
SoundGrid DSP servers, which are basically embedded PCs that
allocate their CPU resources dynamically. This is a more flexible
approach in some ways, but cant guarantee a particular level of performance; the manual warns that using over 80 percent of
the resources of an individual server risks causing dropouts and other problems.
Up Scale
The Digigrid system does have its attractions for a small, singleuser studio, then, and I particularly like the option of being
able to add additional I/O simply by hooking up extra IOX units. On balance, though, Im not sure it would be my own first
choice in such a context. When you consider the power of modern computers and the latency figures that can be achieved
using interfaces such as RMEs Fireface 802 or Apogees Ensemble Thunderbolt, the idea of introducing the extra complexity
of a SoundGrid server in order to run Waves plugins outside your DAW seems less compelling. If hooking up a multichannel
interface to a native DAW gives you plentiful plugin power and low enough latency, theres not much reason to add further
complication. SoundGrid is a deep product that takes time to get your head around, and in the heat of a session, Id feel
slowed down by the whole StudioRack and eMotion ST apparatus.
However, the advantages of the Digigrid system become more apparent when you consider applications that exploit its
scalability and networking features. For example, an increasing number of studios now follow a costeffective model where
multiple mix and programming suites are built around a smaller number of shared live rooms. Place a couple of IOX interfaces
in the live areas, run a few Cat 5e cables about the place and
shove a server in the machine room, and you make it possible to
run sessions in any live room from any of the connected control
rooms. Users would be able to share not only hardware facilities
but also StudioRack processing chains, no matter whether
theyre Mac Logic acolytes or PC Cubase devotees. Likewise, its The Digigrid IOX looks a tempting option for many studio
not hard to see the advantages that Digigrid and SoundGrid
applications, not least because of its four builtin headphone
sockets.
could bring to larger educational institutions, broadcasters or
postproduction and voiceover facilities. And theres the
advantage of close integration with a variety of popular livesound desks.
There are, of course, plenty of other networked audio systems around, but the StudioRack element marks out Digigrid as
being clearly different from these rivals, giving users the ability to employ almost any plugin from Waves vast catalogue as a
live signal processor for frontofhouse and monitor mixing, as an insert to print to disk, in the monitor path and at mixdown. I
have to say, though, that its appeal in a multiroom context would be greater if the Digigrid hardware range included a
dedicated controlroom unit with talkback and monitor control. As things stand, its not immediately clear how best these
features might be implemented. I mentioned this to Digico, and apparently Im not the first user to have pointed out the need
for such a unit, so hopefully future developments will yield fruit in this area.
In its present state, there are also limitations to the software side of Digigrid which will limit its appeal in some quarters.
Perhaps the most obvious is the lack of any support for surround sound. And although one of the selling points of the
SoundGrid system is that it is open to thirdparty plugin manufacturers, at the time of writing, the first thirdparty offerings
from Brainworx, SPL and Eventide were not yet commercially available. It remains to be seen how much takeup there will be
for SoundGrid among other developers.
Conclusion
Its obvious that both Waves and Digico have invested considerable development resources into the Digigrid range, and for a
product thats still in its commercial infancy, it struck me as impressively mature. The hardware might not be sexy, but its
reassuringly solid, and the ASIO driver proved as robust in practice as any Ive encountered. I didnt run into a single bug,
crash or other oddity during the entire review period, and the consistent, clear and userfriendly design of the SoundGrid
software goes a long way to mitigate for its complexity. For simple home and projectstudio environments with just one
control room and one live room, Im not sure that the benefits of the Digigrid system outweigh this complexity. However, if you
need to equip a multiroom facility, integrate a modern digital live-sound rig, cater for the differing tastes of several engineers,
or even if you just want to leave open the possibility of future expansion, Digigrid deserves serious consideration. .
Alternatives
A number of audiooverEthernet protocols have been developed, and manufacturers such as Focusrite and MOTU have
taken advantage of these to develop scalable systems that have applications in the studio. However, the way in which the
Digigrid system runs Waves plugins on DSP servers at minimal latency is, as far as I know, unique.
Multirack
The Digigrid concept has close connections with Waves Multirack system, which is designed to allow livesound
engineers to use Waves plugins on their mixes. At its simplest, Multirack is a plugin chainer and host which can be run
natively on a Mac or PC, with audio piped in and out either using an appropriate multichannel audio interface, or the
builtin Firewire or USB connections on digital desks such as the Behringer X32, PreSonus StudioLive and so on.
However, theres also a SoundGridenabled version, where the plugins themselves run on a SoundGrid server, with the
Mac or PC acting mainly as a controller. This provides guaranteed lowlatency performance, which is of course hugely
important in a livesound context, especially with complicated multispeaker PA systems.
In both cases, chains of Waves plugins created using Multirack can be opened in the StudioRack plugin, and vice
versa. This means that a recording engineer working on a multitrack recording of a live show that was mixed using
Multirack can access all the effects and processors that were used by the frontofhouse engineer; and, conversely, that
the studio mix engineer can make his or her plugin chains available as starting points to use in a live show. The potential
benefits of this integration are obvious, especially when you consider the challenge of bringing to the stage a heavily
produced album with lots of spot effects.
Incidentally, those with long memories might recall that the Digigrid system isnt the first to run Waves plugins on a
remote DSP server connected via Ethernet. Almost 10 years ago now we reviewed their APA 32: just like the current
generation of DSP servers, this was basically a PC embedded in a rackmounting box and fitted with a very noisy cooling
fan. It also had some basic networking features. Unlike Digigrid, however, the APA processors were purely intended to
boost DSP power at mixdown, in the manner of TCs Powercore or UAs Satellite, and did not feature any audio I/O or
lowlatency input monitoring.
www.soundonsound.com/sos/sep05/articles/wavesapa32.htm
Digigrid & HDX

The way in which you use StudioRack plugins is similar in all native DAWs, but a different approach is needed to get
SoundGrid working with Pro Tools HD, HD Native or HDX. Pro Tools accesses Avids HD hardware and its associated
DSP processing using its own driver. This, unsurprisingly, cannot talk directly to a SoundGrid network.
Integration between Digigrid and HDX is achieved by attaching one or more Digigrid DLS or DLI units to a spare Digilink
expansion socket or to an HD or HDX card, in the same way as you might attach a thirdparty AD converter unit. Once
this is done, plugins running on the SoundGrid DSP server are treated as if they were external hardware effects units
with their own plugin controllers: a Pro Tools Hardware Insert plugin handles routing to and from the server, whilst the
StudioRack plugin operates in I/O Insert mode, configuring the processing that is applied. In this sort of system, the only
I/O that is handled by Digigrid consists of connections between Pro Tools and the SoundGrid server, so there is no low
latency input monitoring that is handled by Pro Tools, as usual.
I did not have the opportunity to test a system that used Digigrid and HDX hardware, so cant tell you how well this
works in practice.
Pricing
Because Digigrid is a modular system, it can be configured in many different ways. The SoundGrid software is included
with all hardware units, but Waves plugins will need to be purchased separately. Pricing for the Digigrid interfaces is as
follows:
Digigrid MGB and MGO: $2000 each.
Digigrid DLI: $1280.
Digigrid DLS: $3120.
Digigrid IOX: $2640.
Digigrid IOC: $tbc.
Digigrid IOS: $3760.
Fri 19 Jun 2015
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Xfer Records Serum

Reviews : Software: Instruments
Buy PDF
Xfer Records
Serum $189
pros
Clean, clear and detailed
sound.
Fast and intuitive interface.
Superb and comprehensive
wavetable implementation.
Quality filters and effects.
cons
Can be processor-needy.
Those precious modulation
connections are soon used
up.
summary
Wavetable synthesis is
known for its rich harmonic
sweeps, glitchy transitions
and sonic complexity. To
this, Serum adds new
dimensions of clean, high
fidelity and accessibility.
information
$189
www.xferrecords.com
Test Spec
Recommended spec:
CPU with SSE2.
Windows XP/Vista/7/8 or
Mac OS 10.6 or greater.
VST2.4 , AU or AAX
compatible host software.
Review system:
Mac Pro running OS X
(10.8.5), 16GB RAM,
2x2.66GHz Quad-Core Intel
Xeon running in 64-bit
mode.
Logic 10.0.7 in 64-bit mode.
Serum v1.01.
Serums slick interface belies an extremely flexible wavetable synthesizer.

Paul Nagle
erum is the first synthesizer from Xfer Records, creators of the enduringly useful LFO Tool. Its aims are simple: to be a
dream synth, which in this case translates to a wavetable synthesizer producing high-quality sound from a workfloworiented interface.
Wavetables were first developed by Wolfgang Palm of PPG, the concept later taken up by Waldorf and Access (amongst
others). The distinctive sound is derived from groups of digital waveforms, known collectively as wavetables. Movement and
tonal complexity are introduced by scanning the table, either manually or by modulators such as LFOs and envelopes.
Perhaps because of the potential for complexity, its a synthesis type well-suited to the graphical world of VST-land, hence the
many examples that exist, trumping the older hardware in fidelity and in the number of wavetables.
Available in VST, AAX and AU formats (both 32- and 64-bit), Serum is much deeper than its unencumbered panel implies. It
ships with a large vault of prepared wavetables and an extensive toolkit to roll and shape your own. As dreams go, its a good
start...
Plain & Simple

After a brief and pain-free installation, youre presented with an interface of unparalleled directness and welcoming clarity.
Given the number of messy, confusing and inconsistent synth panels I find myself juggling each day, this one is a delight. I
couldnt even find fault with the grey background as it perfectly highlights the subtly coloured wavetables, filters, envelopes
and LFOs. I doubt anyone will seek the manual with any great urgency.
Aided by a crack team of patch designers, Serum ships with approximately 450 factory sounds. A quick perusal reveals a
wealth of sweet pads and bright, fuzzy strings, as youd probably expect. Equally impressive, though, are the hard, deep and
cutting basses, plus more snappy plucks than a harpist on speed. Strolling through the collection, youre left with an
impression of shimmering uppers and rock-solid bottoms, peppered with razor-sharp leads. I did find it odd that the obvious
capacity for atmospherics and cinematic sound design wasnt more fully explored, whilst simultaneously relishing doing so
myself.
Tables In Motion
The four possible sound sources consist of two independent wavetable oscillators, a sub-oscillator and a deluxe noise
generator. Its an architecture as familiar as it is logical, which means theres nothing to stop you plunging in right away.
Selecting a single oscillator unprocessed by filter or effects, I began auditioning the factory wavetables. Theyre selected
from categories such as Analog, Digital, Spectral, User and Vowel, and while the names give a good idea what to expect, its
the pervasive clarity that hits you. With over 140 tables to choose from, I could have happily spent days selecting wavetables
and manually hurtling through them using the WT Pos knob. Instead, I set an LFO to automatically update the position,
achieving the task by simply dragging the header tile of a chosen LFO to the knob in question. This is one of several methods
of assigning modulation and took around a second from thought to execution. The modulation range is shown by a blue arc,
which in this case I extended around the knob in order to scan the whole table. You can see how many destinations each
modulator has by numbers on the header tiles.
A long-time fan of wavetable synthesis, I was eager to make an original table. In Serum, a wavetable consists of up to 256
single-cycle waveforms, known as frames. Although its straightforward to edit an existing table, I already felt confident enough
to start from scratch. A click of the pencil icon in the graphical window of either oscillator opens the wavetable editor. This is a
fully stocked carpenters shop offering table construction by means of imported audio, drawn waveforms or mathematical
formulae. The drawing of waveforms is one of those arts that seems incredibly exciting until you try it. Fortunately, there are
far superior ways to populate a range of frames. Serum can dynamically interpolate (ie. fill in the gaps) between waves using
explained
either crossfading or spectral morphing. Its therefore entirely under your control whether the final table has smooth, buzzy or
glitchy transitions.
If you opt to import a WAV, either into the current frame or to fill the whole table, you shouldnt expect regular sample
playback, although it is possible for the resynthesis process to create material thats recognisable speech and drum loops
are great fun to experiment with. Anyone whos spent unhealthy hours in near-darkness probing Waldorfs 19-20 wavetable
will testify that a lot can be achieved with a minimal assortment of syllables, and Serum can cram in whole conversations! The
maximum table size weighs in at around 2MB but most are much smaller. If you import data from large WAVs, it is truncated
when the table is built.
The manual features tutorials aimed at creating high-quality wavetables from the output of other soft synths. However you
choose to do it, your creations (or favourites from the factory tables) can be exported for use externally. Initially the supported
formats were 8-bit (aimed at certain hardware modulars, eg. Wiard) or 32-bit WAV. Cheekily, I requested another option and
was quietly stunned to see an update posted on Xfers forum soon afterwards. Serum now includes 16-bit WAV export too,
which by happy coincidence is perfect to feed the PCM oscillators of my Roland V-Synth. My blagging wasnt entirely selfish
because plenty of other instruments can benefit from exotic digital waveforms too.
Remaining within Serum, theres a huge number of ways that
waveforms can be manipulated. This powerful magic is stored in
the lengthy Warp menu. Warps modes and parameters are
unique to each oscillator and the results can sound like a megasophisticated tone control one moment, a psychedelic waveform
twister, wrapper and bender the next. The less adventurous may
prefer to start with a familiar process such as PWM, but even this
isnt fixated on squarewaves itll squeeze anything you throw
at it. At the other end of the adventure spectrum, usercustomisable wave remaps are probably the pinnacle of
advanced waveform squishing. Whatever Warp mode you
choose, each is a great candidate for modulation, from sources
located a few pixels away.
Continuing to cherry-pick through the Warp menu, there are
various types of sweet, clean-sounding sync and a quantise
option that turns everything decidedly nasty and rate-reduced.
After that its extremes all the way, courtesy of AM, FM and RM
(Ring Mod). All these modes force the otherwise pristine
oscillators beyond their usual comfort zone.
The effects rack can be dynamically re-ordered and most of

its parameters modulated.
With wide-spread waveform mashing taking place, the best way to grasp the effect of each Warp mode is to switch to the
2D view. With a single click, the scrolling wavetable becomes an invaluable real-time waveform display.
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Last but not least, voices may be stacked in a powerful unison mode. The extra voices can be detuned for extra lushness,
but the freakiest stuff comes with the introduction of Warp or wavetable position offsets. This makes each unison voice unique,
although with a corresponding CPU cost.
The sub oscillator is basic (and no less useful for being so),
but the noise generator is a bit special. Its a stereo sample
player loaded, as youd hope, with noise of every colour. But it
doesnt stop there. Joining the noise are sampled ambient
whooshes, windchimes, vinyl crackles, dripping water and
atmospherics galore. Adding samples of your own is as easy as
dropping them into Serums Noises folder, remembering that
stereo samples consume twice as many voices. This could be an
issue if youre prone to extravagant CPU-taxing chords.
The Filter
Peering into another nested menu, I see four categories that
ultimately hold around 90 single and combined filters. Tick boxes
are provided to determine which sound sources are processed
and a mix control sets the filters overall effect, from none to
maximum. Even though Serum lacks the serial/parallel routing
seen elsewhere, its pretty comprehensive and covers analoguestyle multimode filters through to formant filters dripping with
vowel-like enunciation. The quality is high throughout. Notable
All your modulation connections in one convenient location.
amongst the throng is the terrific zero delay German LP filter
and the distorted, dirty French LP (with added boeuf). Its not just regular filters either, there are comb filters, flangers and

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phasers, all of which are relatively predictable, followed by ring modulation and a sample and hold filter, which are much less
so. Actually, the latter reminded me of one of the weirder filters from Waldorfs Microwave 2, which is no bad thing. Finally,
youre able to place any filter across the whole synths output as the filter collection is mirrored in the effects rack for
convenience.
Rack Em Up
The effects rack contains 10 modules, their order reassignable with a swift mouse drag. There are no duffers here and my
personal favourites were the warm and spacious plate reverb and dense, classy chorus. Most parameters can be modulated
and by the same sources as the rest of the synth. Thus, if the mood takes you, you can tie the phaser speed and delay time
together, or simply put both under external MIDI control.
Hyper/Dimension is a dual effect that pairs a simulated (but
CPU-friendly) unison with four subtly modulated delay lines. I
mention it only because it has so far found its way into every pad
Ive made.
Before moving on, distortion also deserves a quick mention.
Not usually an effect I use extensively, this one has over a dozen
modes and at times is so full of presence its almost alive. Best
of all is its highly customisable waveshaper mode, complete with
dual waveform editors that facilitate an almost ludicrous level of
fine-tuning.
Conclusion
Serum is an oddly named synth built around refreshingly clean,
low-aliasing wavetables. Even when forced to be filthy, it
somehow retains a refined and polished character, which is no
mean feat. With its zappy envelopes, highly malleable LFOs and
extensive Warp tools, Serum is not just a source of shifting pads
and atmospherics: its a very capable and versatile synthesizer. It
also happens to possess one of the most elegant interfaces a
complex synth was ever given.
The wavetable editor where you can work on an entire table

or individual frames.
I dont have many wishes for future versions. An arpeggiator or a step sequencer would be welcome, but more slots in the
modulation matrix would be even better. If it were possible for patches to load just a fraction quicker that would be lovely, but
there really isnt much to find fault with. I expect some will demand lower-quality or draft modes for those ultra-clean but
needy wavetable oscillators. Personally, even though my system occasionally creaked under the strain, Ive got hooked on the
quality and would rather reduce polyphony than give it up. If you have any interest at all in wavetable synthesis, grab the
demo today.
.
The Matrix
Whichever method you prefer for adding modulation, Serums 16-slot modulation matrix shows every source and
destination connection. You arent confined to a linear correlation between them either: the response curve can be
individually shaped for each modulator. Furthermore, auxiliary modulation sources can be added to each slot, which is
ideal when you need to multiply the actions of two sources. One obvious example of this being the control of vibrato depth
via the mod wheel. If Ive one complaint (and I do), its that those 16 slots fill up way too quickly!
With three envelopes, four LFOs, external modulators, random numbers and two sources of chaos, you shouldnt run
out of modulation sources in a hurry. Four macros shown on the front panel serve as handy performance controls, even
more so when assigned to favourite knobs on a MIDI controller.
There isnt much to say about the envelopes, except that they are fast, responsive and behave impeccably. Lovers of
snappy sequencer parts will relish the ability to tweak the curves of each envelope stage. I certainly did!
The LFOs, with their variable resolution grid, should look familiar to LFO Tool users. The grid is invaluable for drawing
LFOs that are intended for rhythmic or chromatic purposes. At top speed, the LFOs max out at a lively 100Hz or they can
be optionally synced to song tempo. The slowest synced rate is a staggering 32 bars, which presents all kinds of
opportunities for building evolving patches. Ive used a number of LFO designers in the past but this one ticks all the
boxes. Those of you whove noticed theres no arpeggiator or sequencer can take some comfort from the LFOs.
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In this article:
No Obligations
All Kinds Of
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Drums Down
Added Creepiness
Know What Works
Too Close For Comfort
Electric Koolarrow
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Faith No More
Bill Gould: Recording Sol Invictus
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People + Opinion : Artists / Engineers / Producers / Programmers
Bill Gould at Faith No Mores Estudios Koolarrow, where Sol Invictus was recorded. The grey cylinders are an ASC Attack
Wall that goes around his SSL Matrix console and Focal Twin monitors.
Recording and producing your own music is always a challenge especially if, like
Faith No More, your previous albums have been done by the best in the business!
Matt Frost
ans of genre-twisting cult rock five-piece Faith No More could have been forgiven for thinking that 1997s Album Of The
Year would be the last long player the band would ever cut. Less than a year after that record hit the shops, the group
split acrimoniously.
Eventually, however, Faith No More returned as a live band in 2009, and they spent parts of the next three years touring the
globe. Then, last year, the band debuted two new songs during their Black Sabbath support slot at Hyde Park, and in
September, bassist and producer Bill Gould confirmed that they were working on a new album.
After more than two years of off-and-on writing, recording and mixing, Sol Invictus finally hit the shops last month to a flurry
of critical acclaim. Although Bill Gould is a successful record label owner and producer in his own right, and co-produced
Album Of The Year alongside Roli Mosimann, Sol Invictus was the first Faith No More album to be entirely produced by him.
Whats more, Bill also engineered the album, which was largely recorded at the groups rehearsal space-cum-studio, Estudios
Koolarrow, in California. Singer Mike Pattons vocals were overdubbed at his own home studio, echoing the way hes favoured
operating on all musical projects over the last decade, while Matt Wallace was called in to mix the record.
No Obligations
When Faith No More first got together for what would be the
album sessions at Estudios Koolarrow, they didnt even know
whether any of the material would ever see the light of day. We
just started working, and nobody talked about albums, explains
Bill. It was just kind of a way of being creative and just trying to
see what we came up with, but it flowed very easily and very
quickly. I mean, we didnt really want to make this thing a big
obligation where it was like, Were making an album now, we
need your commitment! We just spent a while working amongst
ourselves, and we tried to keep it fun rather than a career thing,
more like, Lets just see what happens. We know each other
really well so lets make some sounds. If we dont like what we
do, the world will never know! I think the whole way it came
together was really the best way it could have happened.
The decision for Gould to engineer, and to record in the
Faith No More (and friend).
The decision for Gould to engineer, and to record in the

rehearsal facility, followed naturally from this mindset. Bill already
had a load of studio gear set up at Estudios Koolarrow, and hed
also cut one album there for experimental Chilean collective Como Asesinar a Felipes.
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At the beginning, I remember saying, Ive got all this stuff here so, since weve got this resource, lets just start miking the
drums and lets just start having some fun and making some sounds, recalls Gould. And it sounded pretty good! That kind of
kick-started a process and we just decided to continue in the rehearsal room. Initially, I just thought we were miking things up
to help with ideas and to help us be creative, but it came to the point where nobody was really in any hurry to bring any
outside people in. We were comfortable with each other already and the performance is always a really good one when its
just amongst ourselves. Everyone said, Well, why dont we just produce it ourselves? Why dont you produce it? and I was
just like, Oh god, man, thats a responsibility! I mean, my stuffs going to get stacked up next to Andy Wallaces stuff and Matt
Wallaces stuff! [Andy Wallace produced 1995s King For A Day Fool For A Lifetime, while Matt Wallace co-produced Faith No
Mores first four albums] I dont know if I can handle that kind of pressure! but, you know, it was really cool actually. It was a
real vote of confidence by them and it really motivated me to really work hard on it. I have no problem producing other bands
but I didnt ever expect to be producing this band!
All Kinds Of Everything

When it comes to writing songs, Faith No More have never kept to a single tried-and-tested formula. This is a unique band,
says Bill. Theres all kinds of processes and we try to keep our processes open to basically anything, just like we try to keep
our styles open in just about any way. I think the thing that we try to do is keep ourselves out of boxes so that were not always
using the same formula to approach songs. We get things from old ideas and new ideas. Sometimes, somebody starts
something and that starts the process, and sometimes somebodys written a whole song. It comes from wherever we can get
it, really. The most important point is that it all fits together at the end as a whole, and that actually takes more doing than the
writing. Coming up with the material is a lot easier than making it all fit together.
Going back to us in the early days when we worked with some great producers and we were on a major label, we always
still had this do-it-yourself thing. We always tried to have all arrangements done before we went in the studio so that the
producer always kind of acted more like an engineer, and didnt get much involved with sounds and arrangements. We did all
that on our own. Weve always operated that way, and that process goes a long way towards how things sound at the end and
how things fit together. All that probably takes the most amount of time. You can have a musical idea but how the actual
instruments fit together like Lego blocks is really the core of the whole thing. Its one thing for people to have abstract ideas,
but then you have to actually make that idea fit with everybody in the way they approach things, and that idea also has to
sound right and feel right. That takes quite a bit of back and forth and experimenting and listening and stuff like that.
Even though the band collaborate on these decisions,
balancing the roles of bass player, co-songwriter, recording
engineer and producer was still a challenge for Bill Gould. The
production parts not so bad. Like I said, we were always into
arranging our music before we went in the studio so I was kind of
always doing that. I think the production and the writing come
hand in hand for how the songs are going to be put together from
their very core, but it was real challenging because, you know, I
also like to be part of this band as a bass player, just working on
my part rather than thinking about the big picture all the time.
That is a bit of a challenge and thats one of the things that
maybe takes a little more time, digesting everything but then also
being able to walk away from it. Taking off one hat and putting on
another hat. You know, its not just a matter of putting the hat on,
you do have to live in each hat a little bit.
Drums Down
While Faith No Mores approach to arranging their material is
unpredictable, Bill Goulds chosen order for tracking at Estudios
Koolarrow followed a more regular pattern. On Sol Invictus,
drums and bass were laid down first, followed by guitar and
piano overdubs. The last things to be completed were singer Mike Pattons lyrics and the recording of his vocals, which were
cut at his own home studio where he has his own unique gear setup.
It all started with drums and bass, as far as physical tracking goes, explains Gould. We were in a room that was kind of
like a corrugated shed, and its not a sterile recording room. It has a lot of instruments stacked in it so theres a lot of diffusion
there, and it kind of has its own vibe and sound. The first thing we did was we got the overheads up and [the whole process]
really started with the overhead mics. They really kind of captured the vibe, which helped us get creative in the first place.
Thats the way I look at it. It just sounded really good in that room and so I started supplementing the overheads with kick mics
and snare mics and everything else. Once we were recording, we always got the drums down first and sometimes I played
bass with Mike [Bordin, drummer] or wed have Jon [Hudson] playing guitar with him just to the point where it was feeling
really solid.
It was our space, so I had time to fool around, and I did a lot of stuff on my own when the other guys werent there. I
thought it was really important to have everything together before we started recording. I dont want, say, Mike the drummer
sitting around waiting for me to try a bunch of different drum mics out and try all these different things out, so I do stuff on my
own and, that way, when he is there, he can just start playing and we can just talk about the music, not about different
microphones.
Starting with the kick drum, I used an AKG D30, but then I
sometimes used the Shure Beta 52A and I used a [Yamaha
SKRM-100] Subkick too at times. On the snare drum, I had a
[Shure] SM57 or sometimes the Telefunken M80. Its a little more
of an open kind of mic. With the toms, I used a [Shure] Beta 52A
on the floor tom, which I really liked. Its really got a lot of bottom
and some top but its got a thing where it kind of glues things
together with the rest of the kit when you bring it in. It worked
really well with the overhead mics. I also used a [Shure] SM7 on
one tom. For overhead mics, of all things, I used these [MXA]
MCA SP1s. Theyre these $39 microphones, and I had them
modified by Jim Williams from Audio Upgrades. I changed the
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Much of Mike Bordins drum sound on Sol Invictus was down
to Bill Goulds choice of overheads a pair of cheap largekit a kind of dirty, grainy darkish sound that gave it a real vibe. I
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scientific, surgical sound. Theres almost a visual to them. I used the U67 head on a Korby KAT microphone for a mono drum
mic, and then I used these JZ Microphones Vintage V11s as stereo room mics. They were really cool.
The single-room space at Estudios Koolarrow did present Gould with certain issues during the recording process, especially
when it came to tracking the drums. It is a real big challenge that theres no control room, he admits. All the gear and
everything was all in the same room and Mike [Bordin] is a very loud hard-hitting drummer. When I was listening on the
monitors, things sounded really good and I thought everything was fine. It wasnt until a day or two later sometimes that Id
hear some snare honks or some distortion on the floor tom that I hadnt actually heard at the time, so wed have to come back
and do a couple of things again. Not having any isolation there was a real challenge but there was a good side to it in that I
was just sitting in the room with him when we were recording. There was absolutely great energy in the way the performances
always came out really natural with that setup.
Added Creepiness
As Faith No Mores bassist since their inception back in 1981, Bill
Gould knows exactly what he needs to nail his bass sound in the
studio. My live amp is an Aguilar Tone Hammer 500 or a pair
of them, I should say into two [Ampeg] SVT cabinets. But, for
the recording, I used a Fender Bassman Pro 300, which has
great tone but is very unreliable, travel-wise. Ive done
recordings where Ive done a single pass but then when its been
time to do overdubs or fixes five minutes later the tone has
already changed dramatically! I sometimes used a Royer R121
ribbon and I would sometimes use a [Shure] Beta 52. I actually
also used [Yamaha SKRM-100] Subkick on all the bass, which
was sometimes used in the mixing, but I always like to have
bottom end because its always good to capture it. You can
always not use it, but if you dont have it and you need it, its very
hard to bring in later. It never quite sounds the same if you have
to approximate that sub stuff.
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Estudios Koolarrow is arranged as a single large space, with

Bill Goulds production and recording tools set up at one end.
Chief among these are his SSL Matrix mixer and ASC Attack
Wall configurable acoustic treatment; also prominent in this
shot is the Kemper Profiling Amp used for some guitar
sounds on the album.
For Jon Hudsons guitar tones, meanwhile, We basically used Jons [Marshall] JCM800 and a Royer R121 and [Shure
SM]57 combination for the most part. I also had a Kemper [Profiling Amp] that we used so we would capture Jons tones, and
then Jon would take the Kemper home and work on things further. For the most part, we used [the Kemper] for blending with
Jons original tones later when it came to mix time. You know, theres always one character that an amp is missing that you
can complement with another sound, and the Kemper was really good for that.
One of the most characteristic stand-out sounds on Sol Invictus is that of Roddy Bottums piano, despite the fact that the
Steinway theyd acquired was not completely in tune. Roddy came over from New York to record his keyboard parts and we
had an old Steinway piano in the studio, which was actually Mike Bordens grandmothers! laughs Bill. It was a small
Steinway, smaller than an upright. We tuned it as best we could before we started but it just wasnt quite completely in tune. It
did seem to fit with the character of the room, though, sharing that kind of imperfect personality and it seemed to work. We
spent a couple of weeks with Roddy just with piano parts and miking the pianos and getting everything where it sonically fitted
with what we were getting with the drums and bass and guitars. That was the key. We used those JZ [Microphones] Vintage
11s for the most part because they have a lot of bottom end in them, which I think added to the kind of creepiness of the
piano. We did move the mics around too. We used them behind the soundboard and in front of the soundboard. It varied from
song to song.
Know What Works

The final overdubs on the album were Mike Pattons vocals, which were laid down at his own home studio. We made superrough demo versions of some songs and sent them to Mike to work on, which is the way weve always worked in the past, but
generally speaking, vocals were always the last thing to track. The big arrangements and the musical parts go first, and Mike
kind of fits himself into that. Hes got a studio at his house and hes got this little system that he likes to record with. Hes been
doing things that way for the past 12 or 13 years, and its kind of interesting actually, but its very unorthodox. Hes just found a
way where it works for him. He has an old Orban spring reverb that he really likes that he can drive, which he almost uses like
a guitar pedal. And he has these compression settings where he just knows where it feels good for him. He gets a good
performance but then he also knows where the sweet spots are and how he can drive it. He also has some kind of funky old
delay which he distorts and he brings that in too and records it on a separate track. You can blend that to taste, if you want it. I
dont even know what it is but its really funky. Mike has a [Neumann] U87. Thats his mic but we actually did use a [JZ
Microphones] Vintage 11 on a couple of things. On some songs, it was great for his voice but on others not so great. It did kind
of complement his U87 that hes always used in the past and that he knows really well. Between those two, we had it
covered.
Too Close For Comfort

Although Bill Gould was initially planning to mix Sol Invictus himself at Estudios Koolarrow, he eventually chose to get old pal
Matt Wallace in to provide some outside assistance. We did mix the first single Motherfucker in our spot, and I was going to
mix the whole record there, Bill tells us. But the touring was looming and, also, it was the kind of thing where youre so close
to material, you can just end up over-analysing everything. I just really felt I needed to have somebody with me who could tell
me it was OK, because I was hearing every flaw. Ive known Matt since I was 18 years old and he knows us backwards so, at
the last minute, I just called him up to ask him, and he was totally into it. Its actually ended up being fantastic, and its the best
thing we could have done, because he knows the band and he knew what he wanted to do. I was kind of involved in it too, but
only to the point where wed just check in with each other. If he was feeling good about something then I could relax and vice
versa! .
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same place where Faith No More rehearsed, so miking things up was a very convenient thing to do.
[In terms of acoustic treatment] I have an ASC Attack Wall that goes around my console, and I put that around the
drums sometimes. I also have a lot of those 2x4 RealTraps that I put behind the drums, especially if the drums are against
the wall and I dont want the wall bouncing back into the microphone and to different parts of the room. The good thing
about the place is that the ceilings are really high. Theyre about 18 feet high and its about an 1800 square-foot room, so
its kind of like an airplane hangar. For bass frequencies, thats really good we dont get a lot of those standing waves
coming back, because its big enough and we also have trappings in the corners. That parts pretty solid.
I have an SSL Matrix console, which I got about three or four years ago, and Ive really been happy with it. Its really
cool for monitoring with the drums. I had all my preamps hard-patched in already. It actually sounds really good for mixing,
and I like it better than mixing in the box. We mixed the Motherfucker single there and we used the Matrix for that and it
was great. Its also not too big and its just a really great usable thing that actually does what I need.
Ive got a mish-mash of outboard. I have a Pendulum MDP1, which is really cool and is my go-to pre. I use it for most
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Voodoo Ray by A Guy Called Gerald

Classic Tracks
Buy PDF
Gerald Simpson playing live with 808 States Graham Massey at Manchesters Victoria Baths, 1988.
Photo: Alamy
Hailed as the first British acid house single, A Guy Called Geralds sublime Voodoo
Ray has since become a classic in its own right.
Tom Doyle
he first acid house single produced in the UK, Voodoo Ray by A Guy Called Gerald was an eerie and hypnotic dance
record created in Manchester in 1988, at the height of what was known as the Second Summer of Love. Originally
released that year on tiny Merseyside label Rham!, it sold out of its initial run of 500 12inch singles within a day,
requiring a very hasty repressing, before spending 18 weeks in the UK charts, peaking at number 12 in 1989.
Inspired by the house music sounds emanating at the time from Chicago and Detroit, Voodoo Ray creator Gerald Simpson
was a regular visitor to Manchesters then acid house mecca, The Hacienda nightclub. There was a lot of energy there, he
says today. For me, it was a kind of ideas place. There was a melting pot of people, everyone from students to breakdancey
type people. I used to be amazed by the size of the place and what was going on in there, cause Id only been in these little
clubs before. It just became a focal point of Manchester club life. I used to take advantage of going in there and then leaving
from there and going to the studio.
Roots
The story behind Voodoo Ray is one of a synth and drum machineobsessed individual who grew up in Manchesters inner
city area of Moss Side. Born of Jamaican parents, both Simpsons father and mother had a notable effect on his early interest
in music: the former through his collection of ska and bluebeat records, the latter due to her attendance of a local Pentecostal
church where live music featured every Sunday. They used to do a bit of talking about the Bible and then they would go into a
jam, Simpson remembers. That was really great, really enjoyable when youre a kid. Cause it was loud [laughs]. Loud and
with loads of energy.
Louder still were the sound-system house parties that Simpson first experienced in Moss Side in the 70s from around the
age of 10. I grew up in a really diverse place, but it was really safe, he says. It was in the middle of a ghetto, but you could
leave your door open. Someone would be having a party and youd be able to hear it and youd go over. Thered be a sound
system playing and you could just kind of hang out.
The sound systems were pretty powerful and they were all homemade. People used to build their own amplifiers and
speakers. It was interesting. I wanted to be a part of it.
Simpsons first real musical passion was for hip hop, beginning around 1982, when he was still at school. There was a
bloke who would come to school, he had a ghetto blaster and he would sell tapes, he remembers. Then, later on, he started
an electro funk sound system. Anything electronic like that used to make my ears prick up, so I wanted to get one of these
drum machines. Everyone was into getting the records and stuff. I was like, Naw, I want that machine that makes them beats.
I want to do it myself.
The first drum machine that he bought was the comparatively primitive Boss Dr Rhythm DR55, a steptime programming
beatbox with only four sounds bass drum, snare, rim and clap. Tap tap, space, tap tap, he laughs. I started off with that,
so I had this restriction. Then when you heard other stuff youd be like, Wow. You wanted to try and get to that stage. It was
1986 when Simpson progressed to a Roland TR808, buying it secondhand for 150 from the A1 music shop in Manchester.
Going from the Dr Rhythm to an 808, he says, it was like, Oh my God, I can do all this stuff.
At the same time, Simpson began to learn more about sound from listening to jazz fusion records on his cassette Walkman.
Mainly Weather Report, Chick Corea Return To Forever, and some of the experimental Miles Davis stuff like Bitches Brew,
&
he says. I remember just recording stuff to listen to on the Walkman. I was amazed at the space. Cause up until then Id
been hearing stereo, but Id been hearing it through one broken speaker over there and another one over there. Getting a
Walkman was just like, Oh, I can hear everything how it was recorded. Thats amazing. I always used to try to get Japanese
imports because they were digitally mastered.
More than anything, Simpson says these albums taught him about sonic placement. For me, a really important thing is
imaging, just getting the stereo in the right place, he stresses. I see that as a painting. So you can either be a realist and go,
OK, well the bass drum should be here. Or you can go, Let me be really weird with this, like Im in a space capsule [laughs].
Another key part of Simpsons musical development was the fact that as a teenager, he studied dance. It was
contemporary, jazz and classical, he says. It was really interesting for me because it made a connection between movement
and sound. It made it easier for me, cause I was trying to create stuff thats gonna try and make people dance, yknow. Its
almost like synaesthesia where you see colours to music, you can see movement to a sound. Im thinking of movements all
the time.
Quickly, he found himself being drawn to breakdancing more
than his more traditional dance studies and, learning the art of
turntablism, began DJing. My style was more like performance
DJing, he points out, so it was cutting and scratching. From
here, Simpson formed a hip hop crew, the Scratch Beatmasters,
along with Moss Side rapper MC Tunes. We were really
influenced by what was going on with the American music and
some of the English stuff, he says. We were getting everything
from everywhere, from New Order to hip hop and soaking it all
in. But we didnt have an outlet. And then when we found one,
we just exploded. I wasnt forcing myself to be creative. But it
was just like, you can either sit on the high street with a can of
Tennents or do something else.
808 State
Along with his purchase of the Roland TR808, through working
shifts at McDonalds, Simpson managed to get enough money
together to buy a Tascam fourtrack and his first synth, the
Judging by the copy of SOS in the foreground of this photo,
Roland SH101. From 1986 on, he became a regular of the
we can assume it was taken in the summer of 1996. By this
time A Guy Called Gerald was one of the pioneers of drum
Manchester shops which imported dance records from America,
and bass and the Roland gear has largely been replaced by
frequenting Spinning Records to buy electro funk 12inches and
Akai equipment.
Eastern Bloc Records to buy house. It was in the latter that he
first encountered other key, kindred souls who were to prove crucial to his burgeoning music career. Id buy stuff from Eastern
Bloc, he recalls, and I was saying to these guys, Ive got all these machines at home that make this music. And they were
like, Yeah, yeah. So I played them a tape and they were like, What, you do this yourself? Weve got this room downstairs in
the basement you can use.
Simpson began working with Eastern Bloc owner Martin Price and a student sound engineer, Graham Massey, on dance
tracks together at Spirit Studios in Manchester, originally as the Hit Squad, before they mutated into 808 State. Unusually for
the times, where dance producers tended to work individually or in pairs, 808 State were a fully fledged band.
Everything was really organic, Simpson points out. There were no turntables involved, there was no DJing involved. It
was all synthesis and drum machines and no loops. Im not saying loops are a bad thing. But I think whats kind of happened
now is that people are forgetting the actual skills of synthesis and programming, by getting really lazy and just grabbing.
Over an intensive weekend in early 1988, 808 State made their first album, Newbuild, which showcased a loose and playful
homegrown British take on house music. At the same time, Simpson would take his machines back home to Moss Side to
work on his own material, which he felt at the time was too dark and experimental to be presented to the other members of the
band.
At the time, he says, they were new to the whole thing with the synthesizers, so if I gave them the crazy shit that I was
doing, they wouldnt have bought it anyway. I thought it wouldve been a bit too weird for them because it didnt sound like
something from Chicago or Detroit. So I kept that stuff separate. And, yeah, it was a good thing [laughs].
Timing Issues
Voodoo Ray began life as a home recording experiment for Gerald Simpson, when he was trying to multilayer sounds on his
Tascam fourtrack from the monophonic Roland SH101. I mean, everything always starts off cause youre trying to do
something else, he reasons. I was trying to mimic a polyphonic synthesizer using a monophonic synthesizer.
In attempting to overlay a second synth part generated by the SH101s internal sequencer, Simpsons hitandmiss method
involved pressing play at the beginning of the recorded backing track and hoping the sequence would stay in sync. I had an
issue with timing, he smiles. To get the same sync, you had to really hit the thing on. It was really hard to do. But I managed
to get this riff together.
Over these two SH101 parts, Simpson added a sequence from the Roland TB303, the machine originally designed to be a
bass lineplaying partner to Rolands TR606 Drumatix beatbox, but whose squelchy tones were coopted by pioneering acid
house producers. I got the 303 to do a counter riff and I think it kind of covered up the timing, he laughs.
To his growing collection of equipment, Simpson soon added a second SH101. By then, I was hooked... I was a synth
junkie, he admits. I didnt realise it at the time, but its only later when Ive seen friends go down the same path, Im thinking,
Should I tell them theyre hooked? Nah, Ill just leave them to it.
Working with this preMIDI setup, Simpson used two of the
three trigger outputs from the 808 to link to the clock inputs of the
SH101s, while the drum machines DIN sync out connected to
the 303. Around the same time as figuring out how to link and
synchronise these machines, Simpson bought an Akai S900
sampler, which offered just under 12 seconds of recording at its
full sample rate. Being canny, the young producer found ways to
work around this limitation.
You could record something at a really high pitch and then play it down and you would have more time, he says. So I
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Running out of sample time on the S900 was to accidentally

produce the title of Voodoo Ray. Lifting a spoken word sample
from Peter Cook and Dudley Moores Bo Duddley sketch, from
their 1976 comedy album Derek & Clive (Live), where Cook says
voodoo rage, Simpson found that the sample had cut off,
reducing it to what sounded like voodoo ray. Elsewhere in the
track, the cry of later! comes from Moore.
I used to collect spoken word records, says Simpson.
Cause in my cutting and scratching days, they were kind of like
gold dust. I used to try and do these megamix things where
youd cut in bits from obscure old movies and whatever. So if you
could find movie soundtracks on vinyl, youd be like, aw wow. So,
yeah, he said voodoo rage and I kind of chopped it.
Recently Gerald Simpson has rekindled his love affair with

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The Sky Was The Limit

In summer 1988, Simpson took the homeprogrammed parts for Voodoo Ray into Moonraker Studios in Manchester. Using a
SMPTE to DIN sync box to connect his 808driven synth setup to the studios Fostex 16track tape machine, he was able to
layer sequences without timing issues. Suddenly, his sonic world opened up.
By then, the sky was the limit, he remembers. It was just about that time when people started to discover Cubase. But I
didnt really want to go that way because it was really unstable at the time. I just wanted to create a tapestry on tape, because
Id started to do that with the Tascam fourtrack. Id discovered how to record with the Tascam and find my own space for
things. So I wanted to do that... I mean 16 tracks was almost too much. It was just crazy. All this space. So as well as being
able to trick for time in the sampler, youd be able to throw it onto tape and go back and still do things. Given that the 808
offered the luxury of 11 separate outputs, Simpson began to experiment with treating the drum machines sounds with different
effects. The 808 was my main sequencer, but if you really wanted to, the drum could be the bass or it could just be a click in
the background because you could have a cut-off. For EQing, you could really get into it. With the snare, you could put more
snap on it and get this biscuit-tin sound that I was really loving at the time, which worked really nicely with some Lexicon
reverbs. You could create almost like this white noise from the reverb.
You had one snare on the drum machine but you could make that snare sound like nearly anything you wanted. If I wanted
a lower snare, Id pitch the tape machine up a couple of ips and then when Id put it back down to the normal speed it was like
another snare. Then youd put a reverb on it or something. So there were all these things that you could do. I used to love
spending time treating every single instrument. At one point, I created a diagram of a perfect system, which was a grid-like
snare, bass drum, hihat on one side and then on the other side I would have reverb, flange, delay, compression and then I
would put one to the other.
Elsewhere, Simpson experimented in Voodoo Ray with mixing tones from his two SH101s. I was trying to do something
that someone told me later on is called heterodyning, where you use two tones to create an imaginary tone. The same sound
but its two oscillations are creating a third oscillation. Its like an audio illusion, with almost like a metallic sound coming from it
in some places. Its kind of going in and out of itself.
The iconic female vocal sample on Voodoo Ray was added when Simpson brought in session singer Nicola Collier, initially
to work on other tracks. We had a bit of time in the studio so we did four tracks on that day, he recalls. I remember we did
this track called Spend Some Time, which was like a soul and funk track. I had the bass line already down for Voodoo Ray
and I can remember Lee [Monteverde, engineer] said, Lets just get Nicola to sing over the top. There was no words or
anything so we just created a vibe and had her sing over the top of it. Then I just went in and grabbed a sample of what she
was singing and reversed it and then played it backwards over what she was singing, which created this almost Asiany
sound. It was basically the sound reversed back on itself.
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says. He was showing me how to properly record. We gated
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the stuff that was on tape, it didnt have any noise on it. Say
there was a bit of hiss on the snare, that wouldve amplified
when I started to put reverb on there. I wanted everything to be
really clean, and the space around everything to be clean, so
when it came to doing the stereo imaging, everything was clean
and in its own space.
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Given that Voodoo Ray was designed for the dancefloor,

Simpson took the test pressing of the track first to Legends
nightclub in Manchester and then The Hacienda to hear how it
sounded on a club system. The first time I heard it in The
Hacienda, it really blew me away, he says. It was interesting.
Compared to a lot of the stuff, the bass at the time was more
protrusive.
Due to its thumping bass and trancey groove, topped with its
spooky vocal hook, Voodoo Ray was an instant underground
hit. The label put it out, but in the style of someone whos had
stuff out already, says Simpson. They didnt say, This is the
first track, kind of thing. That was intentional apparently. They
sold shedloads of them in a week. I mean, I wasnt really
concentrating on it, I was just getting on with stuff in the studio. Then I heard that it was in the charts.
Daft Punk
Off the back of the success of Voodoo Ray, Rham! Records asked Simpson to make the first A Guy Called Gerald album,
Hot Lemonade, released in 1989. It was like, You need to do an album straight away, he remembers. I mean, I had loads of
material. I was just working on stuff constantly, so I just got back into the studio.
At the same time, Simpson upgraded his home setup, incorporating a Soundtracs Quartz mixing desk. It was a
development borne out of his realisation that, at the end of booked studio time, he didnt ever want the session to end. There
were so many ideas and so many different ways I was learning about doing things, he enthuses. Doing all the creative stuff
and then doing the more kind of technical stuff. I thought if you could have a balance of these things, basically it could be
unlimited.
Ooh Oo Hoo Ah Ha Yeah

Having launched his career with Voodoo Ray, Simpson has since produced 11 albums as A Guy Called Gerald. As a result,
at the age of 48, he remains in constant demand, performing and DJing all over the world. He stresses, however, that he is
firmly against preproduced DJing. You get people using Ableton Live with an entire mix already predone, he says. Then
theyre standing there, pretending that the knobs on the mixer are too hot [laughs]. You kind of think, Is it all fake?
Thats one of the reasons why I got back into using the Roland stuff, just trying to step out of only using the computers. I do
use Reason to do live shows, but everything is serendipity and nothing is reproduced. Its all stuff that I can do live. I wanted to
try and do a gig where, instead of being on a podium like a DJ, Im right in the middle of the dancefloor. So people can come
up and actually see what Im doing if they want. At the same time it gives me a chance to monitor exactly whats going on on
the dancefloor, so Im working with them. I prefer to be on the dancefloor, having a bit of a dance with them and making the
grooves at the same time.
As far as Voodoo Ray is concerned, Simpson doesnt have a big theory as to why the track has remained such a
dancefloor staple. Yeah, I dunno, he grins. Im totally confused. But it was part of an era that people still remember today.
Probably one of the reasons is it was the first acid house tune from the UK. That might be the thing. If you go back to the start,
you go back to there.
Twenty seven years after its release, Simpson admits he is slightly sick of Voodoo Ray. Kinda sick of it, he laughs.
Twenty years ago I was sick of it. But its part of my history. I kind of always had a fear that it would get in the way to my
progressing and doing other things. I mean, today Ive got no worries with it. But at the time, as a young man, I was like, Wow,
if I put all my apples into this one thing, Im never gonna move on, Im not gonna be allowed to move on.
So I always wanted to move forward with what I was doing and try and get to another level. Its nice to accept the
compliments, but youve also got your own personal goals to achieve too. For me, it was trying to create a balance between
the creativity and the technical side of things.
When it comes to the latter, Simpson is very proud to now be an advisor for Roland, flying to Japan to check out prototype
drum machines. For the once techobsessed teenager from Moss Side, this represents success far more than any record
sales or big gigs ever could.
Nowadays, going to Japan and talking to these people and going, The drum machines really nice, but we could do this or
we could do that, that to me is a payoff, he states. Talking to someone at Roland about a drum machine, its like, wow.
Some people have to have Grammys and gold records on the wall. With me, Im happy and satisfied with that.
.
Remixes
With his next A Guy Called Gerald album, Automanikk in 1990, Simpson secured a major deal with CBS. For a 12inch
included with its UK vinyl release, at the Roundhouse Studios in London, he reworked his most famous track as Voodoo
Ray Americas. It was basically just a lot of reprocessed stuff, he explains. I did that version because I was going on an
American tour. By then Id got an [Akai] ASQ10 [sequencer] and all the old-school stuff, I kind of decided I was gonna
leave that at home and just sample everything. So I thought, on the album version, Im gonna do a resampled version.
There were a few things that were changed. There were some new instruments on it. But its just a bit cleaner in a way. It
doesnt sound so different.
As a club classic, Voodoo Ray has been remixed countless times down the years. Simpson says his favourite
reworking of the track remains the one by Chicago house pioneer Frankie Knuckles, operating under the name Paradise
Ballroom, who stretched the track out with bubbling percussion and piano breaks into over eight minutes. He broke it
down so he got this really nice flow, Simpson says. He just made it sound like more of a New York-style thing.
These days, laughs the producer, new unofficial remixes of Voodoo Ray are constantly being uploaded to SoundCloud
and other sites. Theres a few a week, he says. At first I was really angry about it. But I suppose its a way of people
enjoying the tune.
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Fri 19 Jun 2015
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In this article:
The Art Of Production
Full Spectrum
Individual Guitars
Room To Experiment
Take Your Time
Working To Picture
Stem Therapy
The Low Down
Crazy Vocals
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Inside Track: Alabama Shakes Sound & Color

Secrets Of The Mix Engineers: Shawn Everett
Buy PDF
Shawn Everett engineered and mixed Sound & Color.
In the making of Alabama Shakes Sound & Color, producer Blake Mills and engineer
Shawn Everett had almost unheardof licence to experiment and took full
advantage.
Paul Tingen
econd albums are generally considered to be notoriously difficult, but Alabama Shakes have given received wisdom a
good, er, shake with Sound & Color. The album entered at number one in the US charts and was top five in many other
countries around the world, while critical reaction was overwhelmingly positive. One critic aptly described Sound &
Color as a deliberately weird record, but authentically weird; its chaotic yet cohesive, full of sound, colour and unshakable
vision.
A few reviewers were perceptive enough to give some of the credit for the high achievement of Sound & Color to producer
Blake Mills, a 28year old guitar virtuoso from Los Angeles who has released two solo albums to date: Break Mirrors (2010)
and Heigh Ho (2014). As so often, there also was the unsung behindthescenes engineer and mixer, in this case Shawn
Everett, who was to a large degree responsible for the albums thunderous bass, sonic depth and wide array of colours.
According to Mills, the production approach of Sound & Color

finds some of its roots in his first, barely there solo album, Break
Mirrors, which purely through wordofmouth ended up being
namechecked by a few dozen artists as an inspiration. I made
my album with Shawn, and the idea was to just experiment. It
was shocking to me to see something lowkey made at a friends
house turn into this calling card for me working with other artists.
I first met the Alabama Shakes a couple of years ago when they
were doing a show in Los Angeles, and afterwards we talked
about the process of recordmaking, and they asked me
questions about how we did Break Mirrors.
They were in a place thats familiar to most musicians with a
first record under their belt: having a few finished songs that they
want to record, and others that are incomplete and being in need
Blake Mills.
want to record, and others that are incomplete and being in need
Photo: Mike Piscitelli
of some input. So last year we went to Sound Emporium Studios
in Nashville for some toeinthewater sessions, and as things
went really well it gradually turned into us more or less taking over the studio. We treated it like home and were being very
experimental, just like with my first solo record.
The Art Of Production

At times Mills, Everett and the band took such a leftfield approach as to ruffle feathers at the studio, but, said Mills, We took
that as a compliment! Nonetheless, he stressed that weirdness for its own sake was never the name of their game. As a
producer, the first thing I focus on is the quality of the songs. I want them to be strong, lyrically, melodically and harmonically,
and so each song was combed over for that. Some songs were perfect, and for others I offered suggestions as to how to
improve them. That might be as simple as saying to Brittany [Howard, the bands singer and guitarist], Your lyric idea in the
first verse was really good, stick with that! But after that, producing is a bit like being in the restaurant business. If youre
dealing with ingredients that are of a high quality, you can take advantage in the way you prepare and present them, or you
can overwork things and lose what you started out with.
There are so many plugins and ways in which things can be recorded today. Every laptop today comes with some sort of
recording software, and this means that while its easy to quickly capture something, people often undervalue the importance
of how to present performances. Of course, you want to capture things the right way, so the performance is exciting. For
example, this was very much the case with the recording process for Alabama Shakes first album [as described in SOS July
2012: www.soundonsound.com/sos/jul12/articles/alabama-shakes.htm]. But the production was based on a laissez faire
approach of trying to only capture the performances and otherwise just stay out of the way. It was like taking a photograph,
and while this worked well enough, I felt that the end result did not match the energy that the band had. I wanted a record that
really hit me sonically with its energy, not just with the performances but also the way that the performances were captured
and presented. For me recording is much more like painting then taking a photograph. It requires technique in painting just to
get it to feel real and emotive, as opposed to just set the microphone up in the room and get out of the way. It requires a bit
more finesse to get something to connect.
Full Spectrum
Bringing this painting finesse to the project was to a large degree the responsibility of Shawn Everett, a Canadian who
attended the Audio program in Music & Sound at the Banff Centre, in Alberta, Canada, and who continued to work at the
centre from 20015. Following this he travelled to Los Angeles, where he quickly started work at producer Tony Bergs
Zeitgeist Studio, which is located at Bergs house in LA. It was here that Everett recorded Mills Break Mirrors. Everett still
regularly works with Berg, but has also spread out into other projects, in some cases also taking care of production, with acts
like Weezer, Julian Casablancas + The Voidz, Jesca Hoop and the Belle Brigade.
We did four twoweek sessions at Sound Emporium to record maybe 20 songs in total, plus we recorded a final song,
Over My Head, at Ocean Way [now called United Recording Studios] in LA, where we mixed the album, explains Everett.
Blake and I didnt talk a lot about the sound image we wanted for the album before we started, I think we kind of knew what
approach we were going to take. Blake and I both like recordings that are very hifi, but that also have small lofi moments.
He and I talked a lot about frequency range, and wanting a wide sonic range, with the lowest lows and the highest highs,
making sure that all aspects of the rainbow were filled in. There definitely was a lot of crafting after the recordings of each
instrument to get it to fill a particular aspect of the sonic range. But to be honest, there was no real methodology behind it,
other than that we wanted things to sound good, and cool.
Before we started at Sound Emporium, Blake went to visit Alabama Shakes for a few days in their hometown in Alabama,
but other than that there was no preproduction. The band had only in a couple of cases rehearsed the songs before they
came into the studio. Usually the procedure was that at the beginning of each session Brittany played us a couple of demos,
and the band talked about what song they wanted to work on. The reason the final album is so colourful is in part because her
demos were very colourful to begin with. She works in Logic at home in her basement, and often had already mapped out a lot
of the instrumentation, using a drum machine and some synths. Her demos had a lot of emotional intensity and a wide,
cinematic scope, on which we tried to expand. For example, on the song Gemini her demo had a crazy synth sound that she
called lasers, and there was a lot of messing around to create this big, crazy guitar sound that comes in at 238 and 428,
which simulates the sound and feeling of that synth sound.
There also was a lot of experimentation during the basic tracking sessions. Generally there was about an hour and a half of
trying things out before we were really getting into takes. Everyone, including me, was just figuring out what they were going
to do. The band would just keep on playing and at some stage it became obvious that something was happening, and
eventually these rehearsals turned into takes. I was recording the entire time, so I ended up with enormous Pro Tools sessions
several hours long! But this didnt give me a lot of additional work, because we didnt do a lot of combing over what they had
done before the take and comping between stuff. Theyd keep on playing the song, and there would be a point at which they
got it, and that was cool. Generally, we simply used that take, and we only very occasionally went in and tried to grab
something from another take to improve the main take.
I had a stock microphone setup to record the band with, and
once they had decided what song to work on, we readjusted
things in the studio accordingly. I might change or move mics,
they might change amps or work on drum tuning or set up in
different places. We tried to track everyone playing live in the
same room as much as possible. Id say about 90 percent of the
album was recorded in the big live room at Sound Emporium
Studio A [10x13m with a 6mhigh ceiling]. The vocals, guitars
and drums usually were in the main room, with some baffling
here and there, though we sometimes used the separate drum
room if we wanted a smaller drum sound. The bass cabinet
usually was in one of the isolation booths, because when
recording the drums with the bass cabinet next to it, the spill
made it impossible for us to later work on getting the kick to
sound bigger. We had Heath [Fogg]s guitar cabinet in the small
room occasionally as well.
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Mills and Everett recorded Alabama Shakes as a band in one

room, with isolation booths used only for the bass amp. This
shot shows the later sessions at Ocean Way Studio B. Left to
right: Heath Fogg (guitar), Steve Johnson (drums), Zac
Cockrell (bass) and Brittany Howard (guitar and vocals).
My drum setup is pretty standard, and I used it both in Nashville and in LA. I have both the AKG D12 and D112 on the kick
drum, so I can choose from two different tonalities, and in Nashville I often had a second big kick drum in front, which I was
miking as well, for resonance. I had an RCA 77 above the kick drum, and in Ocean Way I had some foam next to it, because I
was compressing it, and the cymbal got too loud. Most of the drum sound at Ocean Way came from that RCA 77, but most of
the drum sound recorded at Nashville came from a Shure SM57 underneath the snare. I also had a 57 snare top mic,

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MD441 on the tom [the Ocean Way pic also shows an unused 441, waiting for another tom that never was set up].
We also sometimes used something Blake and I called the
Zeicrophone, which were headphones we strapped to the snare
drum, loosely sitting there and picking up more snare resonance.
All mics went through the 48channel Neve VRP desk at Sound
Emporium studio A. Id say 70 percent of the bass guitar sound
on the album was recorded with a DI and 30 percent with mics
on the cabinet, probably a D112 and a 57 for some more grain.
They also went through the desk, and I would have had an Urei
1176 on the DI. Most of the bass sound came from the DI,
because the sound had more knuckles.
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Individual Guitars
A lot of work also went into the guitar sounds, as Shawn Everett
explains: The guitar amps were in the main room for the most
part, although if we needed a totally clean drum sound, wed
move the guitar amps to one of the isolation booths. Both
Brittany and Heath usually had two amps going, and Id have a
Royer 121 on one cabinet and an SM57 on the other, so I could
later pick the sound I wanted. I also often had room mics going
to pick up both the guitars and the drums. Theres an acoustic
guitar on This Feeling, which I recorded with some sort of tube
mic, probably a [Neumann] KM56. All mics went through the
A closeup view of the drum miking used at Ocean Way
desk, usually without any compression or EQ. I wanted to record
Studio B.
as neutral as possible. There also were some keyboards, by the
way, like a Vox Continental [organ], and a piano, which I
recorded with a pair of Neumann 67s, and once a 57 as well, and a Hammond B3, on which I had a D12 for the bass and
again two 67s. The vibraphone also was recorded with two 67s.
Mills: Some of the guitars were played through my custommade guitar amp, which is an old Bell & Howell film projector
which had its audio section rebuilt by Austen Hooks. These
projectors have hifi amps with lowgain tubes, and have
characteristics on the guitar that I really like, like a lot of high
end information that does not sound shrill. The first time I went to
Nashville I brought a bunch of my guitars and amps that I
thought might inspire the band if they wanted to experiment with
different tones. But they ended up largely using their own
equipment. I found that the more I meddled with Heaths
equipment, the less he sounded like himself. You can drastically
influence the way people play and sound by giving them different
instruments.
Room To Experiment
Shawn Everetts Ocean Way outboard recreation of his in

thebox mix was subsequently exported as a stem session.
During the companys eight weeks at Sound Emporium Studio A,

there was, said Everett, a lot of room to try everything. This included Everett spending many hours on treating the sounds he
had recorded and doing rough mixes. When youre tracking you try to get something going thats exciting, and that sounds
like a record. I tracked to Pro Tools at 48/24, but during tracking I had things laid out over the desk, and mostly there was a lot
of fiddling around on the desk trying to get cool vibes, without using many plugins. Once we had a cool vibe on the desk, and
everyone knew what the sonic palette was going to be for a song, Blake would say, Lets let Shawn have some time with this,
and I would transfer the mix I had into the box, get equivalent sounds and balances, and after this point I mixed in the box and
started applying plugins.
But in the first instance, a lot of sonic experimentation happened out of the box. Just like we tried to get huge kick drum
sounds by not using samples and instead using the resonant kick drum, I reamped the kick drum, for example through
Blakes Bell & Howell projector amp. We also had these old weird 1990s fourtrack cassette recorders, one of them being a
Korg CR4, that has weird effects on it and speakers. Id run sounds out of Pro Tools through the CR4, record them to its
cassette tape, and then apply these weird 1990s chorus and other effects, and play the sound back via its speakers, and Id
put a mic on them. Or Id apply a weird Portastudio cassette EQ and distort things in that. I also transferred things to multitrack
tape and experimented with how hard we could drive the tape machine, and then recorded that back in.
Then there were the experiments and treatments that were done in the box. For example, there are strings on the song
Dunes, and before they come in theres something very spacey that sounds like strings, and in fact are space sounds created
by NASA, as well as several other sounds, like an orchestra tuning and things like that, which we layered and then I took
that into Melodyne to create polyphonic information from these layers that moved inside of the chords of the song. So we got
these swooping, undulating sounds that sound like strings, but arent. There also were some plugins that I used a lot, for
example the PSP Vintage Warmer and SoundToys Decapitator, to warm and beef things up.
One essential element when I work, whether tracking, rough
mixing or doing the final mix, is that I have a bunch of plugins
over the mastering bus inside of Pro Tools, so that when we
listen to the mix, it sounds like a record. I have developed a
mindset over the years, particularly because of the loudness
wars, of always working with the question in mind of how the mix
is going to sound once its mastered, and I am adjusting what I
am doing for that. I know how loud people want things
nowadays, and it is almost like an artform to see how loud I can
get my mix and at the same time retain all the things I like about
records that arent loud. If everything is balanced in the entire
frequency spectrum, it can naturally feel loud, without having
been limited to death. So a lot of the experimentation I did at
Sound Emporium, running things through cassette recorders,
VHS tapes, reeltoreel tape, amps and so on, and applying all
sorts of different EQs, was a matter of making sure all the
different instruments have their own place, and that it sounds as
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Take Your Time

ATO, Alabama Shakes record company, gave Everett and Blake
four weeks at Ocean Way Studio B to complete the final mixes for Sound & Color, an extraordinarily long time by todays
budgetconscious standards. They were very generous, agrees Everett. So we did more experimentation! It was mostly me
during the day, and then Blake would come in at the end of the day, and wed try stuff out until late into the night. Brittany was
also there a lot of the time. We were working the entire time, 14 hours per day, nonstop. When you have more time, you can
really get it to the point where you want it to be. Sometimes you hand over your project and you feel like people are stealing
your baby, but this time we had the time do things properly and get everything out of each track that we wanted. This was one
of the few times when handing over a project that we all just felt excited.
My biggest challenge while doing the mixes at Ocean Way was that everyone was really happy with my rough mixes, so I
did not want to screw things up! It wasnt like the beginnings of a mixing stage with everyone wondering how it was going to
sound. For the most part what I did was to try to replicate my entire inthebox mixes on the console, replacing as many plug
ins as possible with equivalent outboard gear. The trigger for the idea was that I had used quite a few API 550 plugins, and
Ocean Way B has this crazy wall of real 550A EQs! Id never seen so many. So, first of all, I laid my mix sessions across the
board, trying to replicate all the fairly complicated bussing shenanigans I had in Pro Tools, which allowed me to A/B the mix on
the console with a print I had done of my rough inthebox mix. Then I replaced all the plugin EQs with outboard EQs, and
then I did the same with plugin 1176s, and other plugins. I did this one plugin at a time, each time checking if I was
improving or ruining things.
The biggest challenge was recreating the sounds of plugins for which there is no analogue equivalent, things like the
Decapitator, the Vintage Warmer, and the [Avid] Lofi, and so on. To replace the Decapitator I sent things through Blakes Bell
& Howell projector guitar amp, but this did not always work, and sometimes I had to give up and live with the sound of the
plugins, who each have their unique sound. To replicate the Lofi Id send the signal through a preamp, get it to distort a bit
and then roll off high end with another EQ. I replaced the Waves Jack Joseph Puig 660 with an outboard Fairchild, but
something like the Waves Renaissance Bass I kept. To replace plugin delays I often used the AMS RXM16.
Often the combined effect of steering the mix through Ocean Ways Neve desk and outboard really sounded like an
improvement, but I also tried to measure the changes objectively, for example using the Waves QClone plugin, comparing
the frequency response of an inthebox Pultec versus an outboard Pultec, and insofar as they were not the same, adjust the
settings to make the outboard effect sound as much as possible like the plugin! However, in the case of three songs the reply
came back, Oh, we like the rough more. No matter what I did to try to replicate the same thing in the analogue world, they
just liked the rough! Brittany always preferred the rough mix of Sound & Color, so thats whats on the album. Some things
trigger an emotional response in us, and we dont know why that is.
Working To Picture
The albums first single Dont Wanna Fight was one of the songs in which Shawn Everetts approach of replicating his inthe
box mix in the analogue domain did prove successful. There is not space here, however, to reproduce Everetts Pro Tools Edit
window screenshots, which total 260 tracks. At the top of this session are 15 stem tracks, eight for drums and percussion, one
for bass, three for guitars, one for lead vocal, one for the room mics, and one for backing vocals. Below that is the mastering
track, with a complicated plugin signal chain of UAD Studer A800, Waves Q10 EQ, PSP Vintage Warmer, Waves Linear
Phase Multiband, UAD Precision Multiband EQ, Waves L3 Ultramaximizer and another Vintage Warmer, and below that a mix
print track.
On completing his analogue parallel mix at Ocean Way, Everett created a second stem session, which he took over to his
own studio, where he ran the stems through his API desk, did a final adjustment to the balance and printed to stereo via an
Alan Smart C2.
I really did a number on myself with this song! The fact that the session looks so complicated, and also all the work I did in
trying to replicate plugins in the analogue domain, may suggest that I like getting very technical, but in fact, the opposite is
the case. Equipment as such doesnt really excite me as much as creating a universe and an interesting sound world. There
was a lot of acknowledgement of the space that the band were playing in, and I just love the way sound can create an
environment and a space.
I also like getting inspired by photography and ideas and films and so on. I was often referencing photographs when I was
working on mixes that helped me visualise sonically what we were going for, for example John Conns late70s/early80s
shots of the New York subway. Specifically, I tried to get my mix for Dont Wanna Fight to sound like and give me the same
weird feeling as a photograph taken in the same era by Bruce Davidson of this woman staring at the camera.
Stem Therapy
In general there was a ton of stemming going on. Basically when I liked something I was doing Id make stems of it and Id
keep it. It makes the session look insane but it was really a way of keeping myself sane during the process. You can see from
the stems of the big session that I only used the snare under microphone, which I darkened significantly with the Decapitator.
As I mentioned earlier, sometimes I could not find a good analogue equivalent, in which case I kept the plugin. In some other
cases, when the sound I got from the plugins was exactly what we wanted, I also kept the plugins. I had some parallel
compression on a drum bus with an outboard Fairchild, and also printed that as a stem, plus there was a clean drum bus, so I
could blend the two. I replicated the 550A plugins on the drum room mics with outboard 550As, and I dont think I used the
L1 and L3 plugins. There are some snare Zeicro tracks, recording the snare with headphones, with the Decapitator, which I
did not replace.
Theres a [Waves] RBass on the bass that I kept, and I had an 1176 on the bass. The guitars also have a real 1176, and a
[Pultec] EQP1A, and theres a [Waves] ZNoise on some of the guitar tracks, because we had some kind of buzzing amp. We
kept the humming and buzzing for the most part, but for example the solo was made up for six or seven tracks, and on one of
the amps the buzzing must have been too much. The guitar solo was made up of a bunch of guitars recorded with the Royer.
One of the guitars was pitched with the AMS and I also used the chamber from Sound Emporium on them as well. They were
then all bused together and treated with a Vintage Warmer plug in. Two of the guitars in the solo also had the Decapitator.
On Brittanys vocals I had the [Waves] RVox, the Massey TapeHead and the Waves CLA76 and C4. I kept most of the
plugins on the backing vocals, like the UAD Roland R201 and Dimension D and the Waves S1 and Super Tap plugins, to
make them wider, because they worked really well. The master bus plugins are the same in the big Sound Emporium
session and in the final stem session from which I printed the mix at my place. I did not try to replicate them in the real world,
because theres no realworld equivalent for most of these plugins. I did print the stereo mix through an Alan Smart C2
compressor, though.
The reason we went to my place to print the mix was because we had to make some changes that were requested later.
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The reason we went to my place to print the mix was because we had to make some changes that were requested later.
We had problems matching up the Ocean Way mix print and the one done at my place, and we were unsure whether it was
the difference between the two desks or between Ocean Ways grey Alan Smart and my black Alan Smart. Ocean Way then
lent me their grey Alan Smart and suddenly we could match the mix prints! But its really interesting retracing my steps for this
mix. It definitely was a crazy session!
.
The Low Down

Both Blake Mills and Shawn Everett repeatedly referred to the pursuit of big bass as a major preoccupation during the
making of Sound & Color, and they explained some of their thinking and approaches. The low end is really important in a
record to get right, insists Mills. You can have sounds fighting for their place in almost every other register, and somehow
they will find their place, but elements that are disagreeing in the low end dont only fuck up the bottom end but can mess
with the entire recording. Theres a hierarchy to the low end. Sometimes the kick sits above the bass, sometimes the other
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way round not necessarily in terms of volume, but also in terms of their frequencies. There is this battle, and if
everybody is the lowest, nobody is the lowest.
Mixing the Led

Zeppelin Reunion
In this respect, the parts that are played are of major importance. If theres a lot of room in the kickdrum part, you can
make the kick drum sound massive. For example, in the song Gimme All Your Love Steve [Johnson]s kickdrum part is
so sparse that the decay can be very long, and the bass is also very sparse, and this gives a lot of space for the bottom
end. You cannot do this with every band, but it worked well with Alabama Shakes because of the way they play. You can
compare it to reggae records, which have amazing low end on vinyl. Of course, vinyl cant contain the amount of bottom
end that were used to these days, and so it really is the low midrange you are hearing. But you get a kind of auditory
illusion: because of the arrangement and the fact that theres nothing below the low midrange, the low end sounds huge!
So we spent time with Alabama Shakes getting the arrangements right, and constructing things to fill up the low end. For
example, I brought my old, 32inch Gretsch concert drum to the sessions, and it has a massive sound. If he played a big
downbeat every other bar, it made that kick drum sound massive.
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Everett: Blake and I are both big fans of big bass. In fact, I spent years working out how hiphop records manage to get
so much low end in a record! I was now applying what I learnt to a band like the Shakes. One major thing we did for many
of the sessions was to have a second, halfopen bass drum next to the kick, just for resonance, and I put a mic next to
that as well, for extra low end. There was a lot of talk about the resonance of the kick drum during the making of this
album! There was a lot of messing around in the pursuit of big bass! At the same time, I did not use that resonant kick
during the Ocean Way recording, and Over My Head has one of the biggest kick sounds on the record! But the resonant
kick was the result of our aim not to use samples. We wanted to get the sound of adding an 808 or other sample to the
kick drum without actually adding a sample. In our attempt to get that big hiphop kick that is on a lot of records nowadays
we also often reamped the kick drum in a different room.
Crazy Vocals
Brittany Howards lead vocals were, at times, recorded in outlandish ways, as
Shawn Everett describes: Yes, the way we recorded Brittany often was crazy! She
realised at one point that she liked singing in the control room, with the speakers
going, and in these cases she often recorded her vocals with a handheld mic that
she had found on Ebay. I dont know what mic it was, I just recorded it. On Guess
Who we used the NS10 woofer, which resulted in this strangely muted tonality,
pokey and weird. Its one of the odder songs on the album, and we wanted to have
her vocal sounds a bit differently. I had to EQ it and do a lot of weird jazz to it
afterwards, but we used it on the entire song, including her background vocals on it,
and it sounded cool.
We did use some normal mics, with the Neumann M49 probably the most used,
and sometimes an SM7 or an AKG C414, but often we were just struggling to get
from these what we wanted. So for Gimme All Your Love, we strapped a
headphone to her face which we had reversewired so it would record, and while it
also required a lot of postproduction treatment to make sure the sound didnt rip
your ears off, the end result was cool. There was even one instance when she
wanted to get a very muffled sound, which resulted in her experimenting with putting
cotton wool in her mouth and even Anbesol [an overthecounter anaesthetic gel to
combat toothache] so she could not feel her mouth.
Blake Mills: We recorded Brittany in ways that were a bit dangerous. Brittany is
Three of the experimental vocal
recording techniques that were
revered as a vocalist, I mean people talk with such respect about her singing, and
tried at Sound Emporium: from
all this is deserved, but it might mean that people become very careful when they
left, Brittany Howard with
record her and just put a 67 in front of her. But as soon as we started doing the
headphones taped to her face,
with her mouth stuffed with cotton
crazy things, I think she, maybe for the first time in her recording-studio experience,
wool, and singing into a Yamaha
felt like she could be herself. She could follow the craziest tangent, experimental
Subkick.
ideas she came up with and the people around her wouldnt look totally confused.
So yes, when the idea came up to put the NS10 woofer in front of her, we just did it. Shawn, who is really great at getting
guitar tones he can really get them to explode! also treated her vocals almost like a guitar, giving them colours that
were exciting.
Peter Cobbin &

Kirsty Whalley
Secrets Of The Mix
Engineers: 2012 London
Olympics
Underpinning the
biggest
spectacle of
2012 London
Olympic Games was
probably the largest
multitrack recording ever
made. Just how do you mix
a thousand-track project?
Mike Stevens
Musical Director For The
Queens Diamond Jubilee
Concert
Mike Stevens
has worked with
some of the
worlds biggest
pop acts at countless highprofile live events, including
the Queens recent Diamond
Jubilee concert.
Fri 19 Jun 2015
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In this article:
To The Point
Look Back In Anger
Angle Delight
Mix OClock
Just The Ticket
At The Faders
Loudness War
Scaling Down
Superior Sonics?
Flying High
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Oasis & Britannia Row Live At Knebworth

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Oasiss 1996 gig at Knebworth marked the end of an era for pointsource PA. We
asked the people who made it happen what has changed since.
Gary Cooper
he fight of the century! The match of the year! The gig of the decade! It sells newspapers and it sells tickets whats
not to like? Rarely, of course, do the actual events live up the that sort of billing, but one band and two gigs in particular
stand out as having not only defined their decade but left an indelible impression on their audiences. The band was
Oasis and the two gigs were at Maine Road on 27th April 1996, in their native city of Manchester, and their triumphant open
air appearances at Knebworth on 10th and 11th of August, where a claimed 250,000 (or more) watched them finally settle the
argument about which Brit Pop outfit was boss. Noone put it better than Noel Gallagher, who, with characteristic immodesty
and absolute accuracy said, as he walked onto the stage: This is bloody history. Were all making history tonight.
To The Point
The 1996 Knebworth gig was historical in another way, too. It was possibly the high point in the evolution of traditional
analogue, pointsource PA systems. These were the giant dinosaurs of audio, which saw huge analogue mixers with multiple
racks of outboard processors feed sound via strata of power amplifiers to mountains of speakers, not just at the front of the
stage, but on stage too, as well as to delay towers, so essential at a huge outdoor event like Knebworth to even out the sound
experience for those not lucky enough not to be nosetobouncer at the front.
Just like the dinosaurs, the proponents of the giant pointsource rigs at the end of the 20th century may have been aware of
the small furry technologies scurrying about their feet, but few realised just how quickly line arrays and digital mixers would
change the appearance, sound and, perhaps most crucially of all, the cost, of providing largescale live sound.
Handling audio duties for Oasis at Knebworth were Britannia Row, no strangers to that venue (the companys very first gig
was at Knebworth in 1975, when they provided sound for a 100,000strong audience there to see the companys originators,
Pink Floyd). It was one of a handful of sound companies that pioneered modern PA and is still at the top of its game today
but even when it handles events like Knebworths Sonisphere, or this years Simple Minds European tour, nothing like the
physical size of rig that was built for that 1996 Oasis event is called for. Times have changed.
Back in 1996, SOSs then sister magazine, Sound On Stage, ran an extensive feature on the Oasis Knebworth event. Writer
Mark Cunningham, clearly a fan, went into forensic detail on the giant Turbosound rig assembled for the event, and were
indebted to him today for having provided that resource as we pose the question: whats changed?
And change it most certainly has. Anyone who had just woken up from a trance
induced in 1996 (quite likely given some of the substances on offer at an Oasis gig)
and wandered into a contemporary equivalent might think he had stumbled upon
the setup for a support band. Gone are the gargantuan analogue mixers; if there
are any at all, toy racks will contain just one or two favourite processors; and the
speakers will be one of the ubiquitous line array systems. Pioneered by L
Acoustics Dr Christian Heil, the French particle physicist turned PA designer who
blazed the trail for them, line arays have more or less wiped pointsource PA off the
map today certainly for most major concerts worldwide. The modern approach
offers many advantages, but is better sound quality among them? Thats another
question we wanted to ask.
This being the music industry, and 1996 being quite a long time ago, tracking down the people who helped make the Oasis
gig happen wasnt easy, but Mike Lowe, Britannia Rows Director, is still very much in the thick of things, so we began by
quizzing him on the event and how differently he would handle it today.
Fundicin
metales Piura
Encargue bienes
industriales de la
mejor
calidad.Solicite
presupuesto!
Look Back In Anger

But first, it might be an idea just to look back at what actually had been assembled
in that Hertfordshire field in the attempt to make the decades most inyourface
band actually sound that way, when you were sat a quarter of a mile away in a field
full of very noisy, excited people...
The Sound On Stage article is exhaustive, revealing that Brit Row had provided
the biggest ever set of Turbosound pointsource Flashlight PA yet assembled. FOH
duties saw no fewer than 34 TFS 780 Flashlight narrows used per side, with 96
lows in four columns at each side, plus 46degree wide dispersion boxes, to
provide extra penetration in the field. The aim was to cover 120 degrees, which
meant, as Mark Cunningham revealed, no fewer than 168 Turbosound cabinets
72 lows, 72 narrows, 16 underhung, and eight wides were spread between 11
delay towers in three zones: at 75, 150, and 225 metres from the stage, providing
80degree delay coverage.
Driving that mass of speakers were BSS EPC760 and EPC780 power amps:
17 amp racks, containing two of each amplifier type, drove the delay towers. Eight
identical racks drove each side of the PA and a further nine racks powered the
monitor system, contributing to a total of 168 BSS power converters.
The frontofhouse setup was mainly

comprised of Turbosound Flashlight
speakers and set up without the aid of
computers or software!
At the time, Mike Lowe said that, Noone else has ever
installed 11 delay towers at a Knebworth show, but Oasis are
very committed to getting good quality sound in peoples faces...
everywhere!
To decide on their positions, we marked out the actual
dispersions that we were looking for, on the ground with white
lines. We went through a whole rigmarole of staking out the
ground with red stakes to indicate the delays and blue ones to
mark angles. We used a big protractorlike board which was
staked to the front of the stage and had that as the point from
which we measured out. The first thing was to stake out the
distances between the three delay zone arcs. That was OK on
paper, but after about an hour and a half, it really didnt look
good, and we came to the conclusion that the stage had been
set up at a different orientation to the plans wed been given. We
checked this out with the staging guys, and it turned out that our
hunch was correct, so we rearranged things at our end.
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The insert and effects racks (left and right, respectively).
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The stick and string approach reminds us of another sea change in the way sound is handled today: the adoption of software
and accurate measurement and prediction...
Today, Mike Lowe looks back on 96 and the technical challenges and says that not everything has changed. The structure
people would probably still mark out on the site with stakes, but the tools used to create the sound design have come on a
long way. We would be provided with far more advanced AutoCad drawings of the site by Production. System prediction
software would be used by our system technicians, to position every loudspeaker enclosure, together with its height and the
vertical and horizontal angle of each array. Laser measurements would now replace tape measures. The accuracy in getting it
right first time has improved immensely.
The delay rigs in 1996 were at 75m, 150m and 225m from the stage. We would use almost the same distances today.
However, due to the throw capability of todays linearray systems, the consistency of volume down the field is greater.
A serious consideration back in 96 was the local authoritys concern about noise pollution. Oasis were famously loud and
Brit Row had to work hard to keep the sound in the venue, as far as was possible. Today, limits are enforced even more
fiercely but, paradoxically, the power the company would use on a comparable gig today would be greater, he says: The
amplification used in 1996 would have been capable of delivering around 210,000 Watts into 4. If we were to put a system
into Knebworth today, the amplifiers would be capable of delivering something like 850,000 Watts. The enormous difference in
the firepower of the system and delays between now and then is because of the ever increasing expectations of audiences,
and the artists and promoters desire and/or need to meet those expectations and, of course, the technology has
developed.
The increase in the capability of the PA has less to do with
volume and more about better coverage and audio quality.
Having said that, the Flashlight system used in 1996 was truly
great for its time.
Mix OClock
If the speakers have changed, so, dramatically, has been the
way the sound is mixed, as Mike Lowe says: The 1996 system
was mainly analogue. The improvements in system performance
today by advanced digital signal processors are also another
giant step forward in potential audio quality. By using digital
consoles today, Huw Richards (FOH engineer) and Gareth
Williams (Monitor engineer) would have much of the mix stored
in memory to step through during the show, so that they would
One of the many Midas analogue consoles that the show was
mixed on.
have far more time to work on the finer details. The same would
go for the support band engineers. Much of the outboard effects
and dynamics would no longer be in evidence as they are now in the digital console systems.
Obviously, anything that somebody felt very passionate about maybe valve equipment would be in an outboard rack,
and very possibly if they felt they couldnt quite get the reverb they wanted with the onboard stuff, maybe something like a
TCM6000 might be used, but were talking about less than 10 or 12 units of outboard gear, if anything, as opposed to three or
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Engineers: Mike
Crossey
'Chocolate'
The 1975's
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Years In Showbiz
From elaborate
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arrangements to
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Goldfrapp
Tales Of Us
Will Gregory
took the
unconventional
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FOH Engineer
Backstage at
a major festival
in France, we
caught up with
four 20U racks, which would have been quite common then.
One area where change has not been so dynamic (pun intended) is in miking. The microphones we use would be much the
same, or updated but largely unchanged models. The odd esoteric microphone might now be thrown in. Microphone
technology has been good for a long time. If one listens to the Nat King Cole or Frank Sinatra recordings from the 1950s from
the Capitol Studios in LA (remember that multitrack machines were almost a thing of the future then), the quality is amazing.
Just The Ticket

Which brings us to another great driving factor that seldom gets a look in amid all the talk of technological advances and
artists demands for better and better quality sound money. A gigantic rig takes a lot of people to truck, assemble, run,
disassemble and then take home again. It also represents a huge capital investment that has to be paid for. In as much as the
evolution of modern PA has been driven by considerations of improved sound quality and ease of operation, so it has also
been goaded by the sharp prong of profit. And that has coincided, curiously enough, with a huge rise in ticket prices. If you are
sensing a disconnect there, sorry. Thats just how it is.
As Mike Lowe says: One other observation about the gig was that it grossed 5,625,000. For 250,00 people the average
ticket price would be 22.50. My guess is that the average ticket price for Ed Sheeran at Wembley Stadium this summer will
be sixfold and the ticket price for the 1975 Pink Floyd Knebworth concert was 2.75 in advance, or 3.50 on the door!
The $64,000 question for audiophiles, of course, is not whether its easier to truck, set up and tear down, or if it costs the
band less money, but is the sound as good? You wouldnt expect Mike Lowe to answer no, but his answer is interesting.
Yes, I believe it is. The first thing is that, because of prediction software, you can get the whole thing far more accurately
arrayed. Secondly, the thing with point source and it has its advantages is that at the end of the day youre talking about
fairly narrow dispersion boxes, probably 25 degrees. Side by side it can be got very clean, but the laws of physics dictate that
youve got a lot of boxes interacting with a lot of other boxes, so there is a limit. Also, with any touring box, consistency is an
issue. You get a 12 or 15inch speaker that may be just slightly not performing to the degree that the next one is, and youve
got a problem. With line array, you can cover a lot better, especially in the vertical plane from the front of the stage to the
top gallery, you can get much more accurate arraying and more even coverage. And that applies just as much as an outdoor
gig.
That said, at a dance event, for example, Ive heard a really
well setup Funktion One pointsource system sound absolutely
superb, but the time involved in getting it there compared to a
line array is something else that were all conscious of.
At The Faders
In charge of the audience end of assembled might was the
bands resident FOH engineer, Huw Richards, with fellow sound
man Rick Pope working alongside him. Both have remained very
active in the business, Rick Pope having spent the past 21 years
working for Jay Kay and Jamiroquai, managing Jay Kays studio
as well as handling the bands FOH duties on tour, though he
also handles FOH for other artists, most recently Will Young.
What were Ricks memories of Knebworth?
Gareth Williams, who mixed the monitors for the Knebworth
That was probably the last of the great pointsource systems.
show.
Thinking back, Huw and I were probably doing Peter Gabriels
tour around then and on that tour we were one of the first to try
an LAcoustics linearray system. I remember we were playing Paris when we first met them. Wed come from the days when
you needed loads of boxes, so that was a revelation. So yes, Oasis at Knebworth must have been one of the last gigs where
that huge amount of gear would have been used.
One big difference was the software that we didnt have back then. Putting the PA up then was a handson art that you
learnt by doing it, and if it didnt look right or sound right, youd bring it back down, change your angles, and fly it up again.
There was no computer programme to tell you what to do. You had to be good to make it work. Of course, you still have to be
good, but today youre relying on a computer programme to tell you how high youre going to have to fly the boxes and so on.
Another thing that software allows you to do is put little delays in between your stacks, which back then you couldnt do
because it was so cost prohibitive so bass coverage is a lot more even today than it was.
Its also a lot quicker to put a system up today and then take it down again, so theres a lot less manpower required. Im
pretty sure Knebworth was 10 wide and six deep... Thats an awful lot of boxes. And there would have been another 10 boxes
as well and they used to have dollies, four to a dolly, so think of the manpower to move all that, just to get it in and out of
trucks. You do a gig today and youd probably get it in one truck, rather than two and a half. From a cost point of view, its
been a great change.
At least one aspect of the sound has changed though, Rick believes. The stereo image that you used to get back in the
days of the multibox rigs was much wider than you ever get today, but personally I use line arrays. One thing you find is that
the sound coming from the back of the boxes doesnt affect the stage sound half as much as it used to with, say, a Flashlight
system. It used to be horrendous for the poor monitor engineers.
But the sound at Knebworth at that gig was fantastic. I remember during the soundcheck saying how loud the monitors
were on stage and the guy said, Oh, thats Liams monitor rig, but we havent turned it round yet. At the moment its facing up
stage. I said Really? Youre kidding me! I think hes calmed it down now. We did a gig with him recently, a double headliner
thing with Jamiroquai and Prodigy, and hes nowhere near as loud as he was back in those days.
Loudness War
Back to the show as Rick says, Oasis simply smashed it that day. It was a fantastic gig. Their volcanic energy was always
tangible on stage notably in the explosive relationship between the two Gallagher brothers. That must have been a
minefield for monitor engineer, Gareth Williams, surely? These days, Gareth holds down what you might imagine to be a
somewhat more sedate gig, promoting the Cropredy Festival for Fairport Convention, though he still works as a sound
engineer, too.

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Hey man,
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I started with Oasis at Maine Road, he recalls. I put the system together for
them there and there were so many speakers that it was completely uncontrollable,
but it wasnt my business to say so. When we got to the rehearsals for Knebworth, it
required a great deal of thought because they wanted to see a huge system but
there was no way I could have had it all on at the same time. I remember the
sidefills you could probably have done Hammersmith Odeon with them alone as
FOH. They were flying, I think, eight Turbosound Flashlights a side, with
underhangs, and on the ground I had either nine or a dozen 21inch subs per side
with three or four Flashlights. It was absolutely huge. There wasnt just a left and
right, I had about six mixes going on in the sidefills. Flashlight is very directional and
you could zone it, which was what I did. You couldnt leave that on all the time.
Liam wanted his vocal blisteringly loud, and Noel hadnt really found the guitar
sound he wanted he was experimenting. I remember at Knebworth he brought
loads of different amplifiers and he had a switcher box, the idea being that he could
use different amplifiers for different songs, which was fine in theory, except what he
did was leave them all on all night. He was nine feet behind the vocal mic, so for
him to hear his vocals meant the monitors were constantly at 106dB. I remember
Huw Richards (FOH engineer) saying that the best guitar sound he got from that rig,
which was all out of phase, producing this monstrous noise, was when Noel moved
his head out of the way of the vocal mic to look down at his pedals and start a guitar
solo that was the clearest guitar sound all night. It was a ludicrous amount of
noise. Someone would get within 10 feet of a microphone and you could hear their
breath.
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Huw Richards was Oasiss frontof

house engineer for the gig.
It wasnt just huge, it was also very analogue. I think I had two Midas XL3s and then a third one which was a 24channel
quad mixer theyd built for the Pink Floyd Division Bell tour. In fact, it might have been three with the quad mixer as the fourth,
I really cant remember. Obviously, I didnt use the quad function, I just needed it for extra channels to feed into the master
XL3.
Why all the extra channels? Well this was the Oasis gig that more or less crowned them as heavyweight champions, and
like all champions, there was an entourage helping reinforce their status, as Gareth Williams remembers.
Anybody who knew Oasis at the time got a gig. Thats how it was in those days. With something on that scale it had to be
special, but it was a bit too special and they did cut it back for the next tour, the Be Here Now tour. They still had the string
section and a brass section for a couple of the songs, but it was four and four rather than eight and eight. At Knebworth it was
a case of, We had a Hammond organ on the album, so we need an organist anyone know one? Then it was, We need a
piano player. Someone would ask Cant the organist do that? and the answer was, No. We need two keyboard players. We
need one to roll on to the start of Dont Look Back In Anger and roll off again for Bonehead. Then we need a harmonica
player. I wouldnt say it was nave, because it was a bloody good show, but it was chaos. They were making a statement. For
Brit Pop to be playing Knebworth was quite something, and demand for the tickets was amazing they estimated that one in
20 people in the country had applied for tickets.
Scaling Down
So how would Gareth Williams approach the same gig today? If it was a guitar band, something like the Foo Fighters, there
would be a lot less trucks for PA, thats for sure! At the Cropredy Festival that I organise we use an LAcoustics K1 system.
You look at it and think theres no way that is going to fill this field because its a tiny little stack but, of course, it does.
Thats a big difference; you just would not have so many rigging points, so much PA its all grown up. Then you go on
stage, and of course youd still have sidefills, but the vast majority of bands these days are using inear monitors. Youll have
a bloke there with a desk and he might have a few favoured compressors because he doesnt like the plug-ins, but thats all,
because its a digital desk. Its so different. There was a drain on the National Grid when I poweredup the monitor system at
Knebworth! They were exciting times and it was a colossal system that took forever to build and forever to take down.
Superior Sonics?
So does Gareth Williams think, if he were handling the same gig today, that the sound would actually be better?
Well, youve probably lost a lot of complaints. The sound wasnt as controlled as it is today. If you want an oldfashioned
sound, and many bands do, then you can take an old bin-and-horn system, and thats absolutely fine. Sometimes bands do
that, too a few years ago AC/DC took out an old Midas Pro 40 desk because that was the sound they wanted, and thats
absolutely fine. I still to this day love TMS3s as sidefills, the old Turbosound big boxes. I really preferred those to the later
Floodlights and Flashhlights for sidefills. But I think it is much better now, yes far less dangerous, for one thing. At some of
those Oasis shows, Id walk out front of house during a soundcheck, somebody would hit a tom and youd feel your ribcage
cave in. People would laugh when I told them that I EQd Liams monitors in two minutes, but it was true, because after two
minutes youd have to go away for an hour and then come back. It was really very loud.
At the time, there was talk of an early inear system being
used on the gig but not, perhaps predictably, by the band.
Apparently, the string players (no doubt struggling to keep pitch
during I Am The Walrus) used inear monitors, but they were
the only performers who did. Impressively, though, Gareth
reveals that he did manage to make a breakthrough with the
band themselves, albeit later.
I finally did get them on inears. On the tour after Knebworth,
the Be Here Now tour, I think we were going to Australia or
South America and I said, Look lads, I have absolutely no idea
whats waiting for us over there but its probably not going to
work and its probably not going to be loud enough. As an
emergency measure, you will hear yourself clearly perfectly with
Mike Lowe, Director of Britannia Row.
inears. Noel said, OK, Ill come with you and get sorted, and
Liam said No fucker in my bands wearing hearing aids! The
Beatles never had them so were not having them! I said the Beatles never had monitors at all and he argued and argued...
He wouldnt do it. Noel did use them in the end and finally, on the last tour I did with them, Id had enough and said hed have
Peter Franco & Mick

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four years and
over a million
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of choice for
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Douglas
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to go on inears and, I cant believe it, he just said OK then. Ill give it a go. He did and hes been on inears ever since.
Inears started a reverse arms race in the band, too. When Liam was using inears he still had the wedges, and on the
first show I turned them down to about 25 percent, and turned his sidefills down as well. Noel tuned round to me and said
What the bloody hells happened? I said Ive turned him down. Hes got the inears now, so he doesnt need that volume. He
said All I can hear is me guitar now, so Ive turned it down. And he did in fact it brought the whole sound down to a very
sensible level. Finally Noel came over to me and said, It sounds fucking great. Turn him off!
According to the people who were driving the Knebworth Brit Row system, live sound today is better, cheaper, easier to
provide, more controlled and prevents premature baldness (I lied about at least one of those). Now all that remains is to
decide whether there is a band today that can generate the excitement of Oasis performing to a quarter of a million people
and at the height of their powers. Perhaps thats a question for another time...
.
Mixing the Led

Zeppelin Reunion
Alan Moulder | Secrets Of
The Mix Engineers
The film of Led
Zeppelins
reunion concert
was five years in
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Moulder had only three
weeks to mix the entire
soundtrack!
Peter Cobbin &

Kirsty Whalley
Flying High
Reflecting on just how fast things have changed, Gareth remembers another Brit Row gig he handled, this for the VJ Day
celebrations in 1995. I endedup on the roof at the apex at the front of Buckingham Palace, putting some speakers up,
and there was a chap from the Daily Express there. There were 63amp threephase sockets, and also a telephone
socket, and I said Whats that for? He said: Im going to take a picture down the Mall of the flypast dropping the poppies
on the Mall, and within five minutes that picture will be on my editors desk. I thought Youre joking, but look how far
weve come since then with digital cameras!
Secrets Of The Mix

Engineers: 2012 London
Olympics
Underpinning the
biggest
spectacle of
2012 London
Olympic Games was
probably the largest
multitrack recording ever
made. Just how do you mix
a thousand-track project?
Mike Stevens
Musical Director For The
Queens Diamond Jubilee
Concert
Mike Stevens
has worked with
some of the
worlds biggest
pop acts at countless highprofile live events, including
the Queens recent Diamond
Jubilee concert.
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Fri 19 Jun 2015
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In this article:
Back To Front
Under Score
Sharp Shooting
The Gaming Approach
Adaptable Cues
Doing Yourself Out Of
a Job
News
Articles
Forum
SOS TV
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Information
WebExtras
Scoring I Survived A Zombie Apocalypse

Andrew Barnabas & Paul Arnold
Buy PDF
How do you write music for a TV show you havent seen yet? It helps if you can
draw on years of experience composing for video games...
ear reader, please let me introduce myself. My name is Andrew Barnabas and Im one half of composing duo Andrew
Barnabas and Paul Arnold. Some folks know us from our earlier incarnation as Bob & Barn.
I started writing music for computer games back in 1990 on the Commodore Amiga A500, with a game called SWIV. I was
17 at the time, studying for my A levels and spending all of my free time writing music under the auspicious moniker
NightShade of Crusaders for a Norwegian Amiga demo scene group. Many games composers were still using MIDI at this
time, but the demo scene composers, as we were known, were using trackers: the Soundtracker format, invented by
Karsten Obarski, was a fourchannel, eightbit sample sequencer using a piano roll to enter note and parameter data, and
gave us a bit of an advantage in the realism front. Id already begun to do the rounds at trade shows meeting developers, so
when Sales Curve rang in late 1990 and asked me to come in and play some music, they gave me my first gig on the spot!
I carried on writing scores for games whilst at university and

moved to Cambridge in 1995 to become the first inhouse
composer for a developer called Millennium Interactive, largely
because theyd built the best recording studio Id ever seen
they had a 32track Pro Tools rig in 1995! I met Paul Bob
Arnold in 1996 when he was employed as the second inhouse
composer. Sony bought the studio in 1997, and between us we
ran the sound and music department up until 2001, when we left
and started an independent company, called Bob & Barn Ltd
after our amusing nicknames.
We focused on being games composers for most of the
noughties, but often wed come into contact with composers who
suggested we look into scoring traditional media. At the same
time, kids whod grown up listening to our game scores in the
90s got in touch in the late 00s and asked whether wed be
interested in scoring their TV and film projects. Surprised?
Definitely! Flattered, absolutely.
Andrew Barnabas (left) and Paul Arnold in the control room

during sessions with the City Of Prague Philharmonic.
Until a few years ago, we actually found it to be a stigma to mention games to those in film and TV who werent familiar with
the medium. However, Bob attended a networking event for TV folks where he met a successful producer who loved games,
and knew a lot about scores that wed written. He got us in as the curveball option to pitch on a TV show called The King Is
Dead, and we got the gig. We stayed in touch, and by last Summer he was a senior exec at independent TV production
company Tiger Aspect. Bob and I trundled down to London and sat in a meeting room as both he and another exec told us all
about an idea they were developing for BBC3...
Made on a big budget, I Survived A Zombie Apocalypse was to be a unique blend of two very different genres: horror and
reality TV. Ordinary people would try to survive in a postapocalyptic world plagued by the undead.
Back To Front
Theres a certain irony in the way composers are chosen in a pitching environment which is, in many ways, backwards. In
order to win the pitch to score a TV show, youre asked to provide demos of what the title tune would sound like. Considering
that the titles have to encapsulate the entire ethos of a show in a small amount of music, theyre the hardest thing to get right.
When working in house, wed invariably write the title tune last. This was partly due to the way game development worked, but
was ideal for us as composers. Whilst composing and developing the score, you begin to establish themes, instrumentation,
motifs and so forth, and you get to know the project inside and out so if you can write the title theme last, youre already
armed with all of this knowledge and the tune pretty much writes itself.
In this case, there was absolutely nothing to go on, and what made things even harder was the fact that there had never
been a show like this made before. So, we did our homework. We listened to a lot of reality and horror music, and
documented our thoughts, trying to identify common factors. Research showed that realityshow music is generally
contemporary, upbeat, propulsive, light and occasionally a little fluffy. Horror music, on the other hand, is mostly dark, serious,
atonal and textural. Meaning that the ideal title track for this particular show would be light, contemporary and a little fluffy, yet
also dark, atonal and serious. Hmmm. Add to that a request from the producers that it needs to appeal to a younger audience
though, the show being fairly gory, its still 15+.
The most important thing for us was to identify the tone of the show. Was it going to be tongueincheek, or were we taking
this seriously? Music has that wonderful ability to convey the mood of a show very quickly, so it was imperative that this first
impression hit the nail on the head. With this in mind, we pitched five unique title -theme demos, and Seb, the shows
executive producer, gave detailed feedback as to what he thought worked and, just as importantly, what didnt. This all paid
off, since he then offered us the gig. Hooray!
The tone we were guided towards was serious; any humour would come from the situation itself, not from music. We were
then shown a script telling the story of how the outbreak happened over the title sequence, using faux news footage they were
going to film. There was a lot of exposition, meaning that a voiceover would run over the whole sequence and the music thus
couldnt be too thematic (when we read the first draft, it ran to over five minutes, so we knew thered also be a lot of editing to
be done!).
Under Score
We threw all of these requirements and feedback into a musical pot and started to devise our approach. The main hook came
from an orchestral string motif, giving us the serious angle. We knew from the script that the reason for the outbreak was a
little scifi new 5G on mobile phones so we got a bit clever and sampled the interference patterns that phones used to
cause near speakers, made it into a rhythmic track and subsequently into a piano motif (piano is used frequently in horror
music, so that ticked that box too!). To propel the narrative throughout the sequence we gently upped the tempo, which
increased momentum. With the target audience in mind, we chose to use glitchy sounds for rhythm. The hardest challenge
was to keep it simple and ride the narrative, but still give it momentum, with no more than four main musical elements at any
point, to leave enough space for heavy sound design and a voiceover running throughout. (Fortunately, the title sequence
wasnt finalised until late in the day, so we had the opportunity to tweak the music to fit.)
We have a long relationship with the City Of Prague Philharmonic going back to 2002, so we drafted in Nic Raine to
orchestrate and conduct, James Fitzpatrick to produce, Jan Holzner to engineer and Gareth Williams as music editor and
mixer. Once wed completed the title track, we got into producing the music for the show itself.
Biophilia!
Musical Director
Bjrk's stage
show is bizarre
and beautiful,
and it takes
a team of dedicated
to make it happen.
An Orchestra of
Pianos
Maxime Le Guil:
Recording Vincent
Delerm's Les Amants
Parallles
Under the
guidance of
engineer and
producer
and hammered...
Secrets Of The Mix

Engineers: Mike
Crossey
Sharp Shooting
This project was unusual for several reasons, but above all, for
the schedule. The production team needed all of the music
delivered a week after filming finished, to coincide with the start
of the edit. We knew immediately that they wouldnt have time to
get any footage to us beforehand, as there was too much
footage to sift through (the data wrangler told me theyd shot
36,000 hours of footage across 40 cameras, many of which
running 24/7!). They were on such a tight timeframe for
delivering edits to the BBC that they needed to assign a different
editor to each episode, and in order to hit the ground running,
they needed music right from the start. This meant that for the
first time in our careers we had to produce all of the music blind
without seeing a thing.
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'Chocolate'
The 1975's
chart-topping
album is just one
of a string of hit
Crossey.
Sparks
The City Of Prague Philharmonic recording their contributions

to the soundtrack.
So, we needed to think laterally. First off, we asked whether

we could attend the shoot. This bit isnt unusual for us. Ever
since we got invited on set a few years ago we made a pact that wed make the effort to come and see the action unfold in
front of us whenever we got the chance and I highly recommend it. If you get the chance, do it! Not only do you learn an
enormous amount about the process, but the buzz of being on set is great. As composers, most of our work comes to us in
post, when the shoot has already happened. Being on set gives you a chance to chat with the actors and crew. Bob and I
came off a film set once and found wed both written themes for the characters in our heads just from chatting to the actors
between takes! It helps connect you with the project in a tangible way.
Bob couldnt join me this time, as hed just become a father for the first time, so I went up on my own for five days. The first
thing that struck me was the sheer scale of the show. The producers had found an abandoned shopping mall between
Edinburgh and Glasgow which had opened nine years ago, shut less than a year later and had been slowly reclaimed by
nature. Unlike normal British shopping malls where everythings enclosed, this was more American in style long, open
boulevards with boutique shops both sides and even a bandstand to boot! It was pretty much tailormade for a zombie
apocalypse. There were around 60 fulltime crew, and most amusingly, a coachload of 5075 zombies would turn up every
day. There was even a sign reading Do not park here. Zombie coach parking only!
The Gaming Approach

I was set up in the gallery, which served as mission control. In front of me were three huge screens, each showing the views
from four cameras. During each mission I would frantically take copious notes in real time and email them to Bob. During
missions where some of the survivors were in jeopardy wed all be huddled around the screens as it got more and more tense,
on occasion shouting and swearing at the screens. We all really got into it.

Years In Showbiz
From elaborate
band
arrangements to
their pioneering
Goldfrapp
Tales Of Us
Will Gregory
took the
unconventional
decision to base
Kevin Lemoine:
FOH Engineer
Backstage at
a major festival
in France, we
caught up with

14 years.
Id been sat there for days and one day an assistant

suggested I take a break and go to the food tent for dinner. Being
my first time, and not realising there was crew catering, I instead
sat in zombie catering. Watching a large fully madeup, blinded
zombie with his eyes completely covered up attempting to eat a
peach cheesecake with a wooden fork is an image Ill never
forget!
As the show was unscripted, even seeing the action live only
gave us a framework, as we didnt know how it would be edited.
A chase sequence through a corridor filled with zombies that
lasted five minutes in real life might take only 30 seconds of
screen time. Watching the action gave me a guideline as to what
key emotions the music needed to convey during missions, since
Id documented how I felt when they happened, but the overall
look was still unknown. The producers were in the same boat as
us, so ultimately, we took it upon ourselves to come up with our
own brief in the week or so we had remaining after I flew back to
get it all done.
Secrets Of The Mix

Engineers: Mikey
Donaldson
With no chance of viewing edited footage before the music

had to be completed, Andrew chose instead to watch the
action live in the gallery and make extensive notes.
Jonathan Wilson:
Fanfare
Sound
For Jonathan
Wilson, the
quality of
recorded music
same heights.
Our solution was to turn to the methodologies wed been using for years when
delivering interactive scores for games, since the criteria were surprisingly similar.
A game composer has no idea how long the player will take to complete a given
task, be it tentatively creeping around a forest looking for something or running
away from a big bad boss. When you come across that boss, an interactive score
can be governed by instructions as simple as start boss fight music and stop
playing boss fight music when boss is dead, but it can be much more
sophisticated: play boss fight start jingle, then seamlessly crossfade to boss fight
intensity 1, play random musical stabs as overdubs to punctuate action, when
boss health is down to 50 percent, crossfade to boss fight intensity 2 + more
musical stabs, if this lasts more than 32 bars, crossfade to boss fight intensity 3
which is quicker and modulated a minor third up, and so on. This weve done
many times before.
Tony Maserati
The Mix Engineers
A simple song
and an
outrageous
video turned
Maserati.
Adaptable Cues
To give the editors as much help as possible, we had to give enough control over
the music to enable them to elongate or shorten the cues themselves, since we
knew that they wouldnt have time to ask us to rework a cue to add four seconds
here, lose two there, and so on. Not only that, but we also wanted to enable them
to layer stems of the music together to up the intensity should a
scene develop from exploration leading to suspense and chase.
And to add a bit of colour and variety, we gave them overdubs that
they could sprinkle over the top. In sequencer timeline terms, we
had to think horizontally as well as vertically.
Inside Track: Tamar

Love And War
was not only
a remarkable
comeback for
Mikey Donaldson.
The eventual title theme, in manuscript

(above) and Logic arrange page (below).
We made a list of the main areas wed want to underscore. These

were:
Tension bed: drones to slowly add tension to a scene where
there might be danger around the corner.
Rhythmic bed: a subtle heartbeat rhythm that ups the tension to
the next intensity, indicating that zombies are definitely close by.
Tension swell: a short 1020 second overdub that ramps up the
tension and fades. These add variety and could even be used as
musical false alarms.
Low action: a timer of sorts, adding a ticking clock sound to a
scene for that Mission Impossible effect.
High action: fullon chase music for times where the survivors
are in mortal peril.
These all had gentle endings but were, in essence, loopable. We
composed 10 beds, swells and action cues. We tested as many
combinations as we could think of, to ensure that each tension bed could be
layered with each rhythmic bed. The swells were written in related keys so
each would fit and the low and high action cues were written at tempos either
half or double that of the rhythmic bed, so again, they could be layered and still
sound natural.
This was all well and good if the editors wanted to quickly jump in with a
musical bang when things kicked off on screen, but as with games, there are
times where you want sections to come in and out more musically and not jar.
To that end we also produced risers and shock stings. The riser would act as
a crescendo which could crossfade smoothly into a loopable bed. The shock
stings could be used for a sudden shock moment on screen or as a way to end
a looping rhythm section with a bang; it would sound natural but actually be
completely fake!
Whilst on set, I noticed that theyd hired a camera drone for establishing
shots. So we produced 10 establishing stings for those beautiful overhead
panning shots.
Planning music for scenes where the survivors were in base was more
difficult (the food hall in the shopping mall had been turned into a makeshift
camp, complete with sleeping area, sofas and cooking area, a bit like the Big Brother house). There were moments of
comedy, sadness, a blossoming romance, preparing for missions but invariably those sequences would come together in
the edit, so we produced 22 additional cues to cover how we imagined they might be depicted, if they decided to use them in
the show at all, bearing in mind there was a lot more time spent on base than on missions.
lafur Arnalds
Composer & Producer
Many classically
trained
musicians have
ended up playing
direction...
Pioneer Of Digital
Synthesis
Erkki Kurenniemi
Years before the
Minimoog
appeared,
a Finnish
brainwaves.
Inside Track:
Jamie Cullum's
Momentum album
Secrets Of The Mix
Jamie Cullum's
sixth studio
album,
Momentum,
stylistic palette.
J. Cole & Juro

'Mez' Davis
Hey man,
nobody ever
asks me about
this stuff. I love
Caro Emerald
In total, we produced 99 underscore cues and a title track, with a total running time of 52 minutes. These broke down as
follows:
Seven base cues (comms room, mood music, preparation for missions)
Seven comedy stings
Five romantic beds
3 sadness beds
10 establishing stings
Five Low-action loops
Five high-action loops
11 rhythmic bed loops
13 risers (crescendos)
10 shock stings (decrescendos)
10 tension bed loops
10 tension swells
Three success stings
production team
chose to make
music they
loved. The result
Inside Track:
Black Sabbath 13
Secrets Of The Mix
Engineers: Andrew
Scheps
Under the
guidance of Rick
Rubin, Black
Sabbath
charts worldwide.
This gave them the variety they needed, thanks to the number of different permutations that were possible. To make sure it all
worked, we tried lots and lots of different combinations: riser #6 into tension bed #4, followed by rhythmic bed #7, peppered
with three tension swells, and ending with shock sting #8, and so on (if you want to hear what it sounded like, download a
3MB MP3 file from www.bobandbarn.com/isaza.mp3). We then put together a detailed document for the editors suggesting
how this musical toolkit could be used, thus putting control in their hands.
Bob had already been working on cues whilst I was on set,
and seven days after I returned, we delivered all 99 cues, on
time and on budget. Phew! By that point we were very much in
the zone and couldve produced many more. However, the
mandate to appeal to a younger audience meant that they also
had access to licensed contemporary tracks to lift the mood, and
this helped lighten the load.
Daft Punk
Peter Franco & Mick
Guzauski: Recording
Daft Punk spent
four years and
over a million
dollars on their
the way.
Inside Track:
Paramore
Our final task in January was to score hilarious spoof

infomercials theyd made for the show, mirroring those Open
University programmes from the 70s. Weve got an example in
the news section of our web site, along with the title sequence
that is in the TV section.
Doing Yourself Out Of a Job

So, what conclusions did we draw from the experience? Giving editors control over a virtual interactive toolkit of music made
their job more involved, but also made it quicker, since they werent waiting around for composers to make changes. It was
more work for them than having a composer create a bespoke score for every scene, but it meant that they controlled how
each scene played out and could experiment with different edits and change the music in an instant. This system isnt
revolutionary weve been using it in games for years but applying this methodology to a bespoke library of music cues
for a TV programme is unusual, if not unique.
Of course the best results are still obtained by writing bespoke cues and tailoring them to picture, but if that option isnt
available, wed do it again in a heartbeat. Applying knowledge acquired in games to a TV show how times are changing!
Someone asked me the other day whether I Survived A Zombie Apocalypse had been commissioned for a second series. I
have no idea, but thought about it afterwards: if they do, weve kind of shot ourselves in the foot. Since weve already given
them hours of musical permutations, they wont need much in the way of new material. Oops! Maybe they should set season
two in space or underwater, so that we could easily justify a whole new musical toolkit...
www.bobandbarn.com
.
Secrets Of The Mix

Ken Andrews
won a blind
shoot-out
against some of
plum job of mixing
Paramores acclaimed
comeback album.
Nitin Sawhney:
One Zero
Vinyl is still the
listening format
of choice for
Inside Track:
Recording
Aerosmith
Secrets Of The Mix
Douglas
Their latest
album saw
Aerosmith return
to their roots,
wasnt all retro...
Beyond The Grave

And Now
Signed to
Harvest, Janus
made one album
and hated the
properly...
Shahid Naughty
Boy Khan
Shahid Khan has
gone from pizza
delivery man to
in-demand
from Noel Edmonds.
Inside Track:
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In this article:
VCA Faders
Regions Versus Tracks
Relativism
News
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Another Level
Logic Tips & Techniques
Technique : Logic Notes
Buy PDF
Screen 1: You can assign a collection of channels to a VCA group by first selecting them and then Ctrl-clicking on one of their
channel strips over the VCA drop-down menu. From here choose the Create New VCA for Selected Channel Strips option.
You can now use the newly created VCA fader to control their level.
Take control with Logic 10.1s new mix tools: VCA faders, region automation, and
Relative and Trim modes.
Geoff Smith
n this months Logic workshop we continue looking at the new features in Logic 10.1, including VCAs to control the level of
tracks with post-fader sends, Region and Track automation, and Trim and Relative automation modes.
VCA Faders
If you are used to using live-sound consoles you will no doubt be familiar with mixing using VCAs or DCAs, and know how
useful they are for simplifying large mixes down to a few faders. It has taken the world of DAWs a while to catch up and
include them, and now with the 10.1 update Logic has joined the party. In Logic a VCA fader is one that can be assigned to
control the relative level of other faders. Like an audio sub-bus, it is possible to assign any number of channel and bus faders
to it but, unlike a sub-bus, it has no audio path.
One of the key advantages of VCAs over conventional sub-bus grouping occurs when using tracks with post-fader sends to
effects such as reverb. For example, say you had 10 backing vocal tracks and on each one you used Bus 1 to add a bit of
reverb. If you route their track outputs to a sub-bus and then turn down the bus fader, you end up changing the ratio of dry
sound to reverb. The tracks running through the sub-bus would be attenuated but the reverb level would remain unchanged.
However, if you use a VCA-style group, reducing the level with the VCA fader reduces the individual level of each channel
and, therefore, the signal being sent through each post-fader send is also proportionally reduced. Essentially, grouping by
VCAs lets you preserve your post-fader effect balance.
Lets have a look at how to use VCAs with an existing Logic project. I am going to use a project I have been working on that
has a lot of backing vocal tracks. Find a similar project and follow along.
First, we have to make sure that VCAs are visible in the Mixer channel strip. Open the Mixer, then from the View dropdown menu, navigate to Channel Strip Components and ensure the VCA box is ticked.
To assign a group of channels to a VCA fader, select the channels you want to assign in the mixer. In my example, Im using
16 backing vocal tracks. Now Ctrl-click on the VCA field on one of the selected channels, and from the pop-up menu choose
Create New VCA for Selected Channel Strips (see Screen 1). Logic fills in the VCA row of the selected channels telling you
they are assigned to VCA 1 and adds a VCA fader to the far right-hand side of the Mixer. You can now use the VCA fader to
control the level of all the backing vocals assigned to that VCA group. If your backing vocals use post-fader effect sends to a
reverb plug-in, notice how the level sent to the reverb reduces as you turn down the VCA group fader. Another advantage of
VCA-style grouping is that it requires no changes to your existing channel outputs, and a single VCA group can control a
series of channels that are all directed to different outputs.
Having VCAs also gives you room for experimentation with automation. Route all your backing vocals to a conventional bus
and set their channel-strip output to Bus 1. Leaving the VCA in place, now put a compressor across Bus 1 so its compressing
all of the backing vocals together. We can now use the VCA group to control the level of our backing vocals flowing into that
compressor. This allows us to control how hard the backing vocals are hitting that compressor. This is useful to automate in
&
cases where there is a loud word or phrase that is causing the compressor to react in an unpleasant way. To record VCA rides
as automation, go to the right-hand end of the mixer and Ctrl-click on the VCA channel strip. From the pop-up menu choose
Create Track. Logic puts a track lane onto the Arrange page for our VCA fader, allowing you to draw or record automation into
that lane in the same way as a normal track. As a shortcut, setting a VCA faders Automation mode to Write or Touch will
achieve the same outcome.
Regions Versus Tracks

Logic 10.1 makes using automation with Regions or Tracks a lot
easier than in previous versions. In a new Logic project create a
software instrument track and from the Library load the
Synthesizer / EDM Bass / Dub Crush Bass preset. Record a
simple two-bar phrase and loop it so that there are four
Screen 2: With the track header set to show region
repetitions. Now, at the top of the Arrange page to the right of the
automation, drawing automation into a looped region
View menu, click the Show/Hide Automation button to show the
produces looped automation.
automation lane (by default, the shortcut key for this is A). In the
Track header there is a yellow button with Track written on it. This is a toggle button and it switches the Automation display to
show either Track or Region automation. Switch it to Region by clicking it, and from the Automation parameter combo box
navigate to 4 Averb / Mix. Now use the pencil tool to draw in some automation to alter the Wet/Dry balance of the Averb plugin (see Screen 2). Notice that as you are drawing in automation, it appears in each of the looped regions as well.
Now, switch the Automation display to Track and from the
Automation parameter combo box choose Main / Pan / Absolute.
Draw a slow pan sweep from left to right thats eight bars long
(see Screen 3). The result is a reverb automation that loops with
our region and Track automation running across the entire eightbar length. On a personal level I like to use Region automation
for any repetitive automation, and Track-based automation for
broader brush strokes that work across different regions.
Relativism
DAW Techniques
Cubase
Digital Performer
Live
Logic
Pro Tools
Reaper
Reason
Sonar
Studio One
Screen 3: Changing the track header to show Track

automation allows you to draw automation across the looped
Another excellent addition to Logics automation facilities is the

regions. Here I have used pan automation to move the
synthesizer part from the left to the right speaker over an
ability to input automation in Trim or Relative mode. These
eight-bar period.
modes address the problem of massaging the overall level of a
vocal track which already contains detailed rides, allowing you to
easily turn it up in certain sections and down in others, whilst still
preserving the existing automation. Load up a project with a lead
vocal part that already has automated level rides written (see
Screen 4). On the lead vocal move across to the channel strip
and set the Automation mode combo box to Touch, and from the
same menu make sure it is set to Trim. Now hit Play and move
the fader, then hit Stop. Observe that the automation you just
recorded has completely replaced all your careful level rides.
Undo that! With the Automation mode still set to Touch, switch
the behaviour from Trim to Relative. Once again, hit Play and
move the Fader to add some overall volume rides. See how your
Screen 4: If you have a lead vocal part with detailed fader
rides that needs some further balancing, then you can use
previous detailed volume rides remain unscathed and you have
Relative automation mode to nudge the vocal up and
a new +/- volume automation lane that works in tandem with your the
down in volume whilst preserving your existing rides.
existing automation. Lastly, Ctrl-click on the automation and
choose the Consolidate Relative and Absolute Volume Automation option to merge your volume automation together (see
Screen 5). Continue this refinement process, recording more relative automation, and merging it with the existing automation
until you are satisfied you have the perfect level throughout your whole track.
Relative and Trim modes are very welcome additions to Logic
10.1. Along with the other automation improvements, they give
you much greater control over your mixes.
.
Screen 5: Once you have recorded further volume

automation in Relative mode, you can merge that with your
existing volume automation by Ctrl-clicking on the automation
lane and choosing the Consolidate Relative and Absolute
Volume Automation option.
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The Long Game

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People + Opinion : Miscellaneous
Paul White
uitar synthesizers have been around since the mid-1970s but have never really captured the
imaginations of the majority of guitar players. Is it because guitar players are not interested in making un-guitar-like
sounds? Is it because of the hassle of fitting a hex pickup to the guitar? Or is it because they never really track what
youre playing in a natural way? Some would say that its all three, but I cant agree that guitar players arent interested in new
sounds why else would the effects-pedal market be so healthy?
Ive been using guitar synths on and off since 1976, and from
my own experiences Id have to say that the main frustration is
that a typical pitch-tracking guitar synth is too easily fooled by
playing pinched harmonics, fast strums or untidy fingering. My
second frustration is that triggering sampled sounds, which is
how most guitar synths work, doesnt provide the same feeling
you get when playing normal guitar, where the way you attack a
note changes its timbre in a very organic manner. Some players
also cite tracking delay as a major issue, though I find that
modern systems track adequately quickly.
In fact the guitar synth that was most fun to play was the
Roland GR300, simply because, instead of generating new
sounds using oscillators or samples, it actually used the
waveforms coming off the guitar strings, albeit in some quite
creative ways. It was limited to stringy and brassy sounds with
varying degrees of filter sweep, but it played like a real
instrument.
Paul White
Early rumblings of a new direction for guitar synthesis came

from Electro-Harmonix with their various pitch-meets-filter-meets-envelope pedals and later their B-series organ emulators,
all of which work polyphonically using a standard electric guitar pickup. Then, at Musikmesse 2015, Rolands guitar arm, Boss,
unveiled the SY300, a stomp-style processor finished in a very similar blue paint job to the old GR300. Sure enough it turned
out to be a new guitar synth that applies polyphonic processing to the signal coming from a standard guitar pickup. In many
ways it can be considered as the modern incarnation of the GR300, except that the GR300 needed a special guitar fitted with
a hex pickup and the SY300 is capable of a much wider range of analogue synth-style sounds. Id been expecting something
along these lines for some time now as DSP technology has finally started to get to grips with unpicking the different notes
from a polyphonic guitar signal. In fact, TCs polyphonic guitar tuner was probably the first clue that something new was
happening in that area.
Its been a long time coming, but now that a standard guitar can be used, without having to change your playing technique,
to deliver a wide range of new sounds that respond to the way the player picks the notes, I think the guitar synth might finally
stand a chance of making it into the mainstream. Needless to say Ill be reviewing the SY300 the moment I can get my hands
on one. .
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Sounding Off: Tom Flint

When Does Technology Make Things Too Easy?
People + Opinion : Sounding Off
Tom Flint
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few years ago, I interviewed a photographer who had spent the majority of his career using film
cameras. He said lots of profound things about photography, but there was one thing that interested me a great deal.
He said, I have to say, I am not a big fan of digital photography, even now. It feels too simple, it feels like cheating
somehow.
Its the sort of statement that makes sharp-suited marketing executives choke on their cappuccinos and start muttering
insults under their breath. However, not being a marketing executive I thought it was interesting that something designed to
make a process easier could be problematic. Admittedly my interviewee had other gripes with the digital format, but this was
the one that captured my imagination.
These days, as I try to help my two children develop some useful skills, I find myself thinking that technology has made
many things too easy. My two (who are now seven and 10) are so savvy with the iPad that they have no difficulty knocking up
a few tunes using some of the apps Ive downloaded for review.
In some ways its great that they can have such fun with apps,
and theres no doubt that they learn a lot from using them, but
the speed with which they can get a result sets up the
expectation that all musical achievements can be that quick.
When I try to teach my son to play a few chords on the guitar I
have to explain that only by practicing again and again over a
long period of time will he master the instrument. He
understands this, as does my daughter, who has gravitated
towards the keyboard a little more, but when a whole
composition can be pieced together in a matter of minutes using
tempo-synced samples and a bunch of exciting effects and drum
loops, it makes learning to play an instrument seem like a
endless trudge.
Im certainly not the only person to notice the problem. Earlier
this year I saw an article on the BBC web site in which ballerina
Tamara Rojo claimed that modern children lack discipline.
Specifically she was quoted as saying We live in a society that
rewards fast success based on little talent or commitment, which
is transient and a dangerous place to be. She continued by
saying Do we want to promote instant success and instant failure, or do we want to promote self-esteem and hard work?
She attributed her own success to persistence and hard work. I suspect that she was thinking about the influence of TV talent
shows rather than modern music technology, but the point is that both things promote a quick route to success.
Some people might suggest that Tamara and the photographer are stuck in the past and cannot see beyond their own
experiences, and that modern technology and fast routes to success have enabled more people to exploit their talents.
Indeed, there are many compelling examples of artists who have made fantastic uses of technology and might never have
been discovered if it werent for certain media channels.
But there is more behind the opinions of my two examples than stuffiness and dogma. A lot of creative ideas are the result
of taking a wrong turning or making a mistake, and it is said that you learn from your mistakes. Learning to play something
that is challenging is a constant process of making mistakes and discovery. Amid that toil a person begins to find their voice
and style, and gains self-confidence through their hard-earned achievements. Instead of instant gratification they get
something long-lasting and rather more satisfying.
In terms of music technology, I wouldnt propose that designers make their software deliberately difficult to learn or operate.
It is, after all, easy to pick up a guitar or sit at a piano and make a noise. And I am not sure how easy it is to design something
that presents character-building challenges.
Ultimately it is the responsibility of us all to try to gain some mastery over difficult things. It is through that process that we
discover ways to enrich our music making. And those of us who have children need to ensure that they enjoy what
sophisticated apps, synths and sound generators have to offer, but still experience processes like learning to play an
instrument, or even creating samples from scratch.
.
About The Author

Tom Flint is a composer, filmmaker and regular SOS contributor.
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In this article:
Power Windows
Slimming Down
Pad Rock
Minimal MIDI
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Downsizing DP
Digital Performer Tips & Techniques
Technique : Digital Performer Notes
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Right-clicking on DPs Control Panel lets you dock it, shrink its height and configure which sections it displays. Really
focused, streamlined versions can be concocted for use with smaller screens.
When screen space is at a premium, its time to customise Digital Performers user
interface.
Robin Bigwood
n these days of affordable multiple monitor setups and 5k retina iMacs, were often spoilt for screen area and resolution.
But at the same time, laptop screens are getting smaller, and if you work with one, things are quite different. Heavyweight
apps like Digital Performer can feel awkward to deal with, especially if youre used to spreading out, so to speak, and
workflow can suffer amidst all the mouse-clicking and window-shuffling.
It doesnt have to be like this, though. DPs flexible window management can help customise your workspace and keep
everything humming, with several other features assisting along the way. And when you get fluent with it all, it helps on the
multi-monitor setup too. Everyones a winner.
Power Windows
For many years now, the key to DPs user-interface flexibility has been the Consolidated Window: a single window containing
separate cells in a central body area and left/right sidebars. The fully stuffed Consolidated Window looks impressive, but is
more of a big-screen thing. On a small screen you need something much more simple and focused, and DP can deliver just
that.
With a new or existing project open, try manually closing cells using their x buttons to leave just a single central body cell
and one sidebar. If your Control Panel transport strip is set up as a floating window, consider docking it into the Consolidated
Window (by right-clicking an empty part of it and choosing Dock in Consolidated Window. Once its docked, you can then
switch to a space-saving version by right-clicking again and choosing Compact Height. Further right-clicks let you choose
what elements get included in the now narrow strip, so you can pare it down to really important stuff like the time counter and
audio performance meters. Personally, I rarely use the mouse pointer for operating transport controls or memory/playback
modes, so am happy to lose those.
Then, a visit to the Preferences and Settings window is in order; look for it in the Digital Performer menu (OS X) or Edit
menu (Windows). In its Consolidated Window pane youll find value fields for the maximum number of rows DP will add to the
central body area and sidebars. If Im using a small laptop I might enter 1 and 2 respectively. You can still exceed those
numbers by setting up the window manually, dragging its dividers, but the preference
prevents DP from automatically opening additional space-consuming editing views as
you work. Instead, itll switch tabs, which is much more efficient.
For example, lets say youre working in the Tracks Overview, and double-click on a
MIDI phrase. Instead of opening an additional MIDI Graphic Editor, DP simply
switches to one. Double-click a Soundbite or some automation data, and youll get into
the Sequence Editor. And when you need to go back to the Tracks Overview, the
easily remembered Shift+T shortcut will oblige.
Indeed, a clutch of other intuitive keystrokes come in incredibly useful in
manipulating your editing environment. Its Shift+S for the Sequence Editor and
Shift+D for the Drum Editor; substitute G for the MIDI Graphic editor, Q for Quickscribe
and M for the Mixing Board. Beyond this, most windows and info views such as event
lists, Markers and Soundbites open in sidebars. Easy to deal with really, given that
those sidebars are adjustable in width. And hitting Shift plus the left or right square
bracket keys will toggle sidebars open and closed respectively. If Im working on a
small screen, I usually set up a single sidebar as I like it, and toggle its visibility as and
when necessary.
Slimming Down
An interface element that is rather extravagantly wide is the track settings part of the
Tracks Overview. The divider between it and the actual track lanes can be dragged
By setting the numerical values low in

the Consolidated Window preferences
pane, you prevent DP adding spaceconsuming cells to the user interface,
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left, but that only obscures the settings columns. Try double-clicking a column heading
instead: this calls up a Preferences pane in which you choose which columns are
visible. I can easily live without Loop, Lock, Patch and Default Patch, Controllers
Display and Comments. Disabling those makes for a much more compact
appearance.
and force it to switch tabs instead.

Check also the options here relating
to the Mixing Board: Scroll the Mixing
Board... is great for the sidebar
channel strip approach.
One part of DP that can appear both in the body cells or

sidebar is the Mixing Board. This makes perfect sense:
sometimes you need a big mixer, with lots of channels
simultaneously visible, and at other times just a single channel
fits the bill, to tweak the track you happen to be working on. For
a small-screen mobile approach, try opening the Mixing Board in
a sidebar wide enough to show just one channel. Then dive into
Preferences and Settings once more, and in the Consolidated
Window pane, enable Scroll the Mixing Board to the Selection.
That makes it into what is, in effect, a context-sensitive channel
strip.
Pad Rock
Streamlining the Tracks Lists columns makes for a really

clear, space-saving view.
DPs shortcuts make extensive use of numeric keypads. Default

keystrokes assigned to the keypad are all the important Play,
Stop and Record, as well as useful playback mode functions such as Memory
Cycle. Pressing the full-stop/period key is also the way into the infamous dot
trick. A press selects the first field in the main time counter. Further presses
cycle through the other fields. At any time, new values (for measures, beats,
minutes, seconds or whatever) can be entered with the keypad number keys,
before a tap of the Enter key relocates the playback wiper. Its a very effective,
one-handed, quick-as-a-flash location system. However, most laptops (and
even now some desktop keyboards) dont have a numeric keypad. If youre
working with such a keyboard, there are a few ways to restore the keypads
functionality:
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Some non-Apple, Windows-oriented laptops do still have embedded

keypads overlaid on a group of conventional letter and number keys, and
accessed with a specific modifier key. MacBooks used to have these too,
but Apple did away with the feature some years ago. If you can get into the
swing of using the embedded keystrokes, its a great way to go.
You could add a real hardware keypad, connecting via USB or Bluetooth.
They generally arent too expensive, but are a hassle on planes or trains.
Not all work well with Apple laptops.
Heres a really tiny Consolidated Window with
Your iPhone can be used as a numeric keypad thanks to the app NumPad.
single, narrow, left sidebar, just wide enough
Its notably music-oriented (having specific Finale/Sibelius functionality) and a
for one Mixing Board channel the classic
works with Mac, Windows and Linux. The down side is that both it and your
channel strip approach. Thanks to the
DP computer need to connect to the same Wi-Fi network, which mightnt be Preferences settings, it automatically scrolls to
the currently selected track.
possible or desirable on the road. It also requires the computer to run a
helper app. It could be good for home installations, though.
Finally, how about just programming your own DP keystrokes? Play/Stop is of course available via the space bar anyway.
The number row is ripe for assigning to fast/slow forward/reverse and other commands. I rather like the simple R for
record. And an equivalent to the dot is there already: Command+T (OS X) or Ctrl+T (Windows). To program or reprogram
these, choose Commands from the Setup menu, and youll find them listed way down the largely alphabetical list, in the
Transport section. By entering new keystrokes in the Key 2 column youll retain the default ones too. .
Minimal MIDI
MIDI Keys is a DP feature beloved of users everywhere crammed into train or plane seats, simply travelling light, or
otherwise bereft of a controller keyboard. The idea is simple: call it up from the Studio menu, and your computer keys
generate various kinds of MIDI message. Everything is prompted from the little window. So the central two rows of most
keyboards generate note data (with a momentary sustain
on the Tab and \ keys). The letter keys on the lowest row
select different velocity values, with comma and period
keys providing a fine 1 adjustment. The number row
generates momentary pitch-bend and modulation data, and
switches octave. Just as with any other MIDI source, youll
need to record-enable any MIDI tracks you want MIDI Keys
to drive. And dont forget to specifically select it as an input
for MIDI tracks when youre using the Multi Record option.
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Media Management
Things Fall Apart
Back In Time
Stem The Tide
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Fair Shares
Pro Tools Tips & Techniques
Technique : Pro Tools Notes
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This screen capture from Pro Tools Import Audio dialogue shows you the typical folder structure of a Pro Tools session.
Note, however, that Add rather than Copy is selected, meaning that the audio file Im importing wont be copied to the Audio
Files folder of the destination session.
Before you deliver your Pro Tools project to a collaborator, its vital to make sure it
has everything theyll need!
Mike Thornton
nline collaboration is increasingly common in the world of music production, and we are promised built-in tools for
real-time collaboration in Pro Tools 12. In this workshop, however, Im going to look at what needs to be done to share
projects in a more traditional, non-real-time way, such as if youre sending a track to someone else to mix. Get things
wrong, and the person you are sending the project to might encounter the dreaded Missing Fields window, indicating that they
have an incomplete session.
Media Management
Before we dive in with some techniques, I want to start by showing you how Pro Tools organises the media in your session.
When you create a new project in Pro Tools, it creates a folder with the name that youve given to the session. Inside this
folder, it creates a number of items. Firstly it creates a session file, with the same name as the folder. This is just a list of
instructions of what to play and when to play it: this is sometimes called an EDL or Edit Decision List, and does not contain
any audio data in itself.
Pro Tools will also create an Audio Files folder, which is where it will normally put all the audio related to that project.
However, audio used in a session doesnt have to be in this folder, and it is quite easy to end up with a session where some
audio is played from other locations.
Which other items end up inside the project folder largely depends on what version of Pro Tools you have. In versions of Pro
Tools prior to 10, all fades were rendered as separate audio files, which were kept in a Fade Files folder inside the project
folder. You wont see this for projects created with Pro Tools 10 or later, as fades are now calculated in real time. With Pro
Tools 11 projects, however, you might see a new folder called Bounced Files. When you carry out a bounce, Pro Tools will
offer you the option to put the files in a dedicated folder, called Bounced Files. That folder isnt created until it is needed, so
you wont find a Bounced Files folder in your session folder at the start.
Other optional elements include a Session File Backups folder, where Pro Tools saves the backup session copies if you
have Auto Backup on in the Preferences, and a Video Files folder, which is created if you import a video into a session.
Things Fall Apart

Why am I telling you all this? Well, as Pro Tools tends to keep most of the material related to a session in the Session folder,
people often share sessions simply by duplicating the session folder and sending that to a collaborator. However, as has
already been mentioned, media doesnt have to be inside the Session folder for Pro Tools to use it, and that is where things
can start to go wrong when collaborating, especially if you dont understand how Pro Tools normally handles media files and
what else can happen.
When you use the Import Audio dialogue to import audio into a session, the default option is Add rather than Copy; and
although the audio file will be added to the session and everything will play fine, the audio you imported is still in its original
location, which could be anywhere, and isnt inside the Audio Files folder in the session folder. My advice when importing
audio into a session is never to use Add unless you are absolutely sure that is want you want to do. Instead, select Copy, as
this will make a copy of the audio file and save it in the sessions audio file folder.
If you try to share your working session by simply duplicating
the folder there is always the risk that you will share an
incomplete session, so my advice for people collaborating is
always to use the Save Session Copy In option from the File
menu.This will collate all the media used in the session,
irrespective of where it is on your system, and create a fresh
session folder with everything in it. Then you can share this new
folder with your collaborator. So lets take a closer look at the
Save Session Copy window and how it can be configured to
make collaboration easy.
Back In Time
At the top of the window, you can set which session format will
be used in your duplicate project. It is important to set this
correctly if your collaborator has an older version of Pro Tools
than you do, as they will not be able to open a session saved in
a later version. In Pro Tools 11 and 12, there are now three
The Save Session Copy dialogue makes it easy to
different session format options. Latest refers to the session
consolidate everything a collaborator will need from your
project, even if theyre running an older version of Pro Tools.
format used in Pro Tools 10 to Pro Tools 12, with the file
extension .ptx, while Pro Tools 7 to 9 sessions used file
extension .ptf; its also still possible to save a version that will be compatible with Pro Tools versions from v5.1 to v6.9, with
the file extension .pts. However, although Pro Tools 11 and 12 can still open Pro Tools 4 sessions, they can no longer save in
the v4 format. Pro Tools 10 supports formats all the way back to v3.
Lets assume that your collaborator has a copy of Pro Tools 8 LE; in this case, youd select Pro Tools 7 to 9. Note that the
Audio Files option lower down has been automatically ticked. This is because PT10 and later can support mixed audio file
formats within a session but earlier versions cannot, so its necessary to force all audio files to be saved in the same format.
I would always leave the format as BWF (.WAV), as this is the most universal audio file format and it supports metadata too.
Likewise, unless you have a reason to change the sample rate and the bit depth, its best to leave them alone. Under Items To
Copy, it is essential to tick Audio Files even if Pro Tools doesnt do it for you.
You can choose to only include specific tracks with the Selected Tracks Only option. This can be very useful to create a
version of the session with only the tracks your collaborator needs, so reducing the size of the folder you will share with them.
You can also select to just include the Main Playlist, which is the Playlist on each track that is currently visible and in use on
your session. Again this can help to keep the session you share as simple and clean as possible, in situations where your
collaborator wont need material from the other Playlists. If you are working to picture, ticking the Movie/Video Files option
adds your video file to the Session Copy folder. However, it usually isnt necessary to tick either of the two Plug-in Settings
options unless you need to share plug-in settings that currently are not used in the session, as active plug-in settings are
saved within the session file.
Finally, I would not recommend using the Preserve Folder Hierarchy option, as it recreates the folder structure of the
location of your existing session and means your collaborator would have to dig down through loads of folders to get to the
session.
Once you are happy, click on the OK button and you will get a
dialogue box allowing you to decide where the Session Copy
folder will be saved. By default, Pro Tools adds Copy of to the
session name. I usually remove this and amend the session
name to make it useful and relevant for your collaborator.
When you save to an earlier session format, a dialogue box
will pop up warning you that certain features that were added
since PT10 will be lost.If you are happy, click the Yes button and
Pro Tools will start to create the Copy Session folder, copying all
the session files and all the audio files needed for the session
irrespective of where they are located on your system. The result
will be a completely fresh and self-contained copy of your project
that you can then share with a collaborator.
Stem The Tide

In some circumstances, though, its either unnecessary or
inappropriate to send a full Pro Tools session. For instance, your
When saving in an older project format, some Pro Tools
collaborator might only need to add a couple of parts, or might
features will be unsupported.
be working in a different DAW altogether. In this case, an option
is to create stem and submixes using offline bounce, and share the audio files (or a simpler session containing only the tracks
with the stems on them). If youre going to be sending just the audio files, make sure they all start at the same point and that
this is clearly identifiable (for instance, the session start). That way itll be easy to line up your collaborators contributions
when they come in.
One final tip: always, always, always label and name everything clearly. You may know your way round your session and
your odd or non-existent labelling scheme may be OK for you but for someone else it might be a complete nightmare!
.
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When Bad Reverbs
Come Good
Filter Tips
Into The Groove
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Filter Fodder
Studio One Tips & Techniques
Technique : Studio One Notes
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Screen 1: A short impulse response stretched to be much longer, followed by two filters with cutoff frequencies spread a full
two octaves apart. Listen to Audio Example 1 to hear this. Notice that the decay time (Length) is set much longer than the
length of the impulse response that is used, meaning that the impulse response is stretched.
The right reverb treatment can turn almost any sound into source material for filter
mayhem in Studio One.
Larry the O
he effectiveness of tempo-synced filter effects depends on the choice of source material put through them as much as
on the sound of the filter and the way in which it is manipulated. Spectrally complex signals often provide interesting
results for filter effects, and sometimes a recording is so engaging it makes good source material exactly as it was
recorded. Other raw material, though, can be massaged to make it better filter fodder.
Under the premise that the only thing more fun than completely abusing a plug-in is abusing two plug-ins, applying plug-in
processing to source material can make it sonically richer as a fruitful first step to feeding a nice filter effect. This month I will
show you how I like to use reverb, itself a complex filter effect, as a messer-upper processor to enrich whatever raw source
material you choose to process.
When Bad Reverbs Come Good

For the purposes of increasing spectral complexity, the worst-sounding reverbs often make the best choices. Metallic reverbs,
reverbs with audible shimmering in the tails, and other flawed sounds can turn out to make the coolest source material. Lets
be bad.
Convolution reverbs make excellent tools for this purpose, and PreSonuss OpenAir plug-in fits the bill well. Pass by the
standard rooms and halls for the moment, and look for factory presets with names like AtomSmasher. Thats the stuff.
Set up a loop of some source material you can play around with. To make the examples for this column, I grabbed a loop
from The Caribbean Experience folder in PreSonuss Studio One Loops Vol 4 collection, because percussion always works
nicely for synchronised filter effects. The tempo of the loop was 125bpm, but to make my first example, SloCalypso1, I
slowed it down by about half, to 67bpm. Screen 1 shows the OpenAir setup I created for it, and you can hear it online as
Audio Example 1.
Predelay, Length and the parameters in OpenAirs Processing section are all worthwhile parameters to try varying, while
two other settings that sometimes make an interesting difference are the Shorten with Stretch and Stretch with Pitch settings.
These have to do with how reverb is handled when the length of the impulse response differs from that set by the Length
control.
Dont be afraid to save lots of interim presets; they can always
be ditched later if you dont want them. However, note that
changing almost any control on OpenAir causes it to mute
momentarily, so I recommend finding a useful setup on OpenAir,
then basically leaving it alone and concentrating on using the
filter for real-time manipulation of the sound. One exception is
OpenAirs Mix control, which can generate some cool sounds if
changed on the fly. Since you are using the reverb almost as a
sound generator, you will frequently find yourself setting the Mix
control pretty high, for a very wet mix. With a drum loop like I
used for this months examples, though, I frequently like hearing
a bit of the dry sound mixed back in just to recover some
transients and propulsiveness.
Screen 2: Here the decay time is set to the same length as

that of the impulse response. Listen to Audio Example 2 to
hear this.
Studio Ones Room Reverb can also be tortured into yielding

some interesting source processing Audio Example 4 was created using Room Reverb but I find convolution reverb
generally better suited, not least because of the option of using your own WAV file for the impulse response. Any old WAV file
with appropriate sample rate and bit depth can be used, though it takes a decent amount of trial and error to get a feel for
what may work well. Vocals can produce unearthly vocoder-ish textures, and complex sounds that naturally decay after a few
seconds, such as tapping a glass or striking a bell, tend to produce interesting results. Long impulse responses, such as an
entire sung line from a song, can produce more of a surging texture than a rhythmic bed, though shortening the Length results
in a choppier, more rhythmic feeling. Screen 3 shows OpenAir using a vocal sample as an impulse response.In this example,
the decay time is set much shorter than the length of the impulse response.
Filter Tips
Once armed with spectrally rich source material, I have two
favourite PreSonus plug-ins I use for filter effects: Autofilter and
Groove Delay. I followed OpenAir with one or both of those in
creating this columns examples; obviously, you could add even
more cars onto this train.
The first thing to do in Autofilter is click the Sync button in the
LFO section. There are lots of variables to play with here, so Ill
give you some idea of my preferences. The first major sonic
choice is the types of filters used and whether they are
configured in series (chained) or parallel. I like the spatial feel of
two different filter types in parallel. In the LFO section, the 16
Step setting is intended for sync filter effects, so I make use of it
often. The Analog and Digital SVF filters can be varied
continuously from low-pass response, through band-pass to
high-pass. The various responses can create very different
sounds.
Screen 3: A complete sung vocal line is used as the impulse

response in this example, but the decay time is set much
shorter than the length of the impulse response. Listen to
Audio Example 3 to hear this.
Screen 1 above shows the Autofilter setup used for the first example. Note that the filter cutoffs are two octaves apart. For
this kind of effect, I tend to either have the filter cutoffs far apart or quite close; I dont often use, say, one octave difference.
Autofilter responds much better to real-time manipulation than OpenAir, and resetting step levels during playback is
perfectly feasible. I try to make changes to steps immediately after the play cursor goes by a spot, rather than risk a glitch by
trying to change something right in front of it. LFO Speed is another good parameter to change on the fly. Changing FLT
Spread (filter spread) on the fly interrupts the audio, but sometimes the interruption sounds cool. The range of filter modulation
is largely determined by the range in the step sequencer, the filter cutoff setting, and the amount of modulation being applied
to the cutoff through the cutoff modulation sliders. I get the most pleasing results when one of these three might have a wide
range, but the other two really dont.
While high values of resonance produce a classic sound, if used indiscriminately all source material starts to sound the
same. Its worth the time to try listening with very low resonance for a few minutes and see what you hear.
Into The Groove

Groove Delay is a more complex story, having four filters, each with its own delay. Screen 4 shows the Groove Delay setup
used in Audio Example 2.
Groove Delay not only allows filter cutoff and resonance to be
controlled, but its filters are state-variable and can be
continuously varied between the available response types: lowpass, high-pass, band-pass and peaking. Nearly all of the
parameters can be changed without glitching, but changes to
delay times certainly produce a major disruption in the sound. I
particularly like the major shift created by switching the
Beatlength parameter. The Cutoff Mod control is also rateoriented, so changing it creates some interesting rhythms.
Setting Cutoff Mod to a dotted note value generates good
rhythmic complexity.
If you have a decent control surface available, Control Link
Screen 4: Groove Delay offers tempo-synced delays with
can be used to hook controllers up to some of these parameters
filtering. Each tap has a filter of a different response type and
for real-time manipulation, after which, as it is said, hilarity
centre frequency.
ensues. Imagine having the delay tap levels on faders, with
buttons that momentarily jump the filter cutoff or response type to a different value. That alone could keep you busy for a
while. The variations offered by Groove Delay are so broad I really recommend spending some time fooling around with
different parameters while in record and capturing the results to a new track.
.
DAW Techniques
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Audio Examples
Surf to www.soundonsound.com/sos/jul15/articles/media-0715.htm to hear the audio examples showcasing this months
Studio One techniques.
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Fri 19 Jun 2015
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In this article:
Poles Apart
Dynamic Designs
Capacitor Mics
Ribbons
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Microphone Types: Good Vibrations

Spotlight
Technique : Miking Techniques
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What are the pros and cons of the different types of microphone, and when would
you want to use them?
Chris Korff
ere often asked at SOS what the best of something is, and microphones are no exception. Unfortunately theres no
short answer mics can seldom be said to be simply good or bad; rather, a mic can be suitable or unsuitable,
depending on how youre using it. Confused? Read on...
Poles Apart
Microphones can be divided up in various ways, but the two most important design characteristics are the pickup pattern and
the operating principle. The pickup pattern, or polar pattern, describes how a mic responds to sound arriving from different
directions. A so-called omnidirectional mic or omni should, in theory, pick up sound equally well from every angle. At the other
extreme, a mic with a figure-8 polar pattern will be sensitive to sound arriving at the front and back, but deaf to sound arriving
from the sides. By combining these characteristics, a subsidiary polar pattern called cardioid (because it looks heart-shaped!)
can be created, which is most sensitive at the front and rejects sound to the rear.
Omnidirectional mics are often thought of as having the most natural reproduction of sound, and one reason for this is that
their frequency response is not affected by distance from the source. By contrast, figure-8 and other directional mics exhibit a
phenomenon known as proximity effect or bass tip-up, which means that when placed close to a sound source, they capture
disproportionately more low-frequency sound. This effect can be useful in helping to beef up a thin-sounding source, but can
also cause significant problems.
Theres far more to microphone directionality than Ive got space to discuss here, but if you want to learn more, this is an
excellent place to start: http://sosm.ag/mar07-polarpatterns.
Dynamic Designs
At heart, most mics on sale today employ one of two basic
operating principles. Moving-coil dynamic mics employ a
diaphragm (usually made of some kind of light plastic) with a
metal coil attached to it, suspended such that the coil sits inside
the field of a magnet. When the diaphragm moves in response to
sound, the coil also moves and an electrical signal is generated
in the coil. Moving-coil dynamics are relatively cheap to make,
and most are very robust.
The down side (for there is always one!) is that, because the
coil is usually relatively heavy, moving-coil mics tend to have a
sluggish response that inhibits their ability to react to high
frequencies and fast transients. As such, theyre not usually
peoples first choice for delicate acoustic recordings but they
The ability of dynamic mics to be able to withstand high SPLs
very much are the go-to type for things like kick and snare
makes them ideal for loud sources, such as drums.
drums, guitar amps, brass instruments and live vocals, where an
extended high-frequency response isnt necessarily desirable,
and where their ruggedness (which includes an extreme tolerance of high SPLs and being gargled by heavy-metal singers)
are a real boon.
Fundicion
metales Piura
Realizamos
trabajos
personalizados.
Solicite presupuesto
ahora mismo!
Moving-coil dynamic mics are nearly always designed to have a cardioid or similar polar pattern, though omnidirectional
models are available for specialised purposes such as speech reporting. Most dynamic mics are also passive devices,
meaning that they dont require electrical power to operate.
Capacitor Mics
The other main type of microphone design in widespread use is the capacitor mic. These too employ a
diaphragm made of plastic or other material, but in this case the diaphragm is coated with an
extremely thin layer of gold or other metal, and is placed in front of a charged backplate to form a
variable capacitor. As it moves in reaction to sound waves, the capacitance of the capsule varies and
an electrical current is generated. As the impedance of the capsule is very high and its output level
very low, capacitor mics always incorporate active circuitry to amplify the signal and present it at an
impedance suitable for connecting to other devices. This means that they require external power;
some have their own power supplies, but most work with the 48V phantom power that is standard on
mixing consoles and preamps. Generally speaking, capacitor mics show the opposite traits to movingcoil dynamics: they tend to have excellent high-frequency resolution and are usually quite hi-fi, but
can be less robust.
It used to be the case that capacitor mics were much more expensive than dynamics, but that
playing field has levelled to a large extent, especially in the sub-$500 range you can get extremely
serviceable capacitor mics for that kind of sum, and many premium dynamic models exceed it. Ive
already listed some typical situations that call for a dynamic, but if youre looking for drum overheads,
or need to mic up any stringed instruments (acoustic guitar, violin, harp and so on), then a capacitor
mic will often be the best choice. Ditto if you want to record multiple instruments in a room with one
mic, or perhaps a stereo pair.
Capacitor mics can broadly be split into two camps: those with a large diaphragm typically
around an inch across and those with a small one, of around half this diameter. Many largediaphragm capacitor designs employ a single, circular polarised backplate with diaphragms on either
side and ports which allow some sound to enter behind the diaphragm. This, in effect, gives both the
front and back diaphragms a cardioid pickup pattern. By combining the signals from the front and back
diaphragms in different ways, approximations to omni and figure-8 polar patterns can be created, and
thus many large-diaphragm mics offer switchable pickup patterns.
DAW Techniques
Inside a capacitor
microphone. The
capsule (the round
gold bit at the top) is
a capacitor, the front
plate of which
functions as the
mics diaphragm.
Small-diaphragm capsules use a slightly different design which usually has a fixed rather than a variable polar pattern.
However, some manufacturers have developed modular systems, where a cardioid
capsule assembly can be unscrewed and replaced with an omni or other capsule if
required. Sonically, small-diaphragm mics are often thought to be more accurate to
the source, and in directional models, any sound that is picked up from the sides or
rear often retains a reasonably natural quality. Large-diaphragm mics have different
strengths, including lower noise and a tendency to sound slightly larger than life
thats often employed to good effect on close-miked vocals and other instruments.
Ribbons
Finally, its worth pointing out that not all dynamic mics use the moving-coil
principle. Ribbon mics use a thin strip of metal foil, suspended between the poles of
a magnet. The ribbon flaps about in accordance with any acoustic flapping going on
about the place, and a commensurate current is induced in the ribbon. The ribbon is
extremely thin, however, so this kind of mic tends to be very fragile, as the ribbon
will easily tear if subjected to knocks and bumps, or strong gusts of air. One thing
that distinguishes ribbon mics from other dynamics and capacitor mics is that they
have a very precise native figure-8 polar pattern that can achieve impressive
rejection of unwanted sound from the sides. Sonically, their high-frequency
response is often curtailed, but where that trait might sound dull in a moving-coil
mic, in a ribbon its often better described as smooth: perfect for taming sibilant
vocalists and scratchy instruments. .
Inside a ribbon mic. The crinkled metal

you can see is the ribbon, suspended in
the middle of a chunky magnet.

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Fri 19 Jun 2015
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In this article:
Panning & Blend
Overall Balance & Mix
Tonality
Additional EQ &
Filtering
Dynamics Spotfixes
Mix Flattery: Effects &
Masterbus Processing
A Stitch In Time
Mix Rescue: Audio
Files
Mixing A Whole Album
Editing Recordings
With Spill
Video: Watch Mike
Build A Mix
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Mix Rescue: Spektakulatius

Technique : Mixing
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HYBRID
STUDIOS
ORANGE COUNTY, CA
Although the band normally played together in the main recording room (below), all the vocals were recorded in an adjacent
DIY vocal booth (above). As a result, no ambient sound from the singer spilled onto any of the other mics in the setup, so the
voice, therefore, didnt blend naturally with the rest of the band in the mix.
Photo: Daniel Plappert
Spektakulatius: Our engineer picks up where last months Session Notes recording
feature left off, with details of how he approached the mixing side of things on this
ambitious project...
Mike Senior
ast month, I explained how I managed a highspeed onlocation tracking session for the band Spektakulatius
(www.spektakulatius.de), recording 28 songs spanning various musical styles in less than five days. My brief had been
to record multitracks that would require as little mixing as possible, the plan being that the band would tackle the post
production. So, I recorded the musicians as an ensemble where possible, made assertive sonic commitments while recording,
rather than leave such decisions to the mixing stage, and used spill between mics to reduce the amount of sonic
enhancement and blending work required at mixdown. (For full details of the recording setup, see the SOS June 2015
Session Notes column at http://sosm.ag/jun15-session-notes and its associated resources page at www.cambridgemt.com/rs-ch10-case1.htm.) In the event, though, the band liked the session rough mixes enough that they asked me to mix
both records too and that gave me a golden opportunity to find out just how effective our tracking approach had been in
lightening the mixing workload!
Panning & Blend

Well, the main good news was that the fundamental sound of each song was already in place the moment I pulled the faders
up. I made a point of rechecking the phaserelationships between the channels, though, which occasionally yielded slightly
more appealing results than before. There are no right or wrong settings when theres so much spill flying around, so all I did
was check which combination of polarity switches I liked the sound of most. On some mixes I also take the extra time to
experiment with more refined phasemanipulation tools such as sampledelays and phaserotators, but normally this is more
timeconsuming than its worth, to be honest, unless Im clearly not getting results Im happy with from the polarity switches
alone.
I also spent a little more care with stereo panning than I had while tracking. For example, I sometimes narrowed the panning
of the stereo tom close mics to match the tom positioning in the overhead mics, and usually tried to pan any solo instruments
close mic(s) to the same side as its respective spill (as heard though all the other mics in the recording room). However, there
was an aesthetic element to the panning too, because many stereo decisions are largely a question of individual taste for
most engineers me included! I prefer, for instance, to keep kick, bass and snare fairly central at all times, even though the
snare here was clearly offcentre in the spill image. There were also occasions when I set the image width of rhythm
instruments Id recorded in stereo (such as the piano and acoustic guitar) to avoid a lopsided stereo image, which is
something Im not crazy about.
That done, there remained three essential mixdown jobs.

Firstly, any parts that had been acoustically isolated or DId
during the ensemble takes, or had been overdubbed separately,
needed room ambience added to them to blend them with the
main ensemble backing tracks (which blended naturally by virtue
of the spill between all the different mics in the room). For a few
overdubs I was able to plan for this by recording the drum
overhead mics as ambience mics alongside whatever close mics
I was using, but for the most part I had to rely on artificial reverb I
designed to roughly emulate the character of the recording room
itself.
Im no tweakhead when it comes to reverb parameters,
though, so the thought of designing such a patch algorithmically from first principles thrilled me about as much as the prospect
of mediaeval dentistry! Instead, I just surfed through a few dozen reverb impulse responses in search of something promising,
but with the convolution engines envelope parameters set to isolate just the reverb onset you dont need any real decay
tail for blending dry sounds into a smallroom recording. Once Id found a shortlist of three impulse responses I liked, I
sculpted each tonally using EQ and fine predelay settings, before deciding on a favourite within the context of my first full
mix. I was then able to reuse this setting across all the mixes, although of course the send levels to the effect had to be
adjusted to suit each specific arrangement.
Overall Balance & Mix Tonality

The second essential task was to finalise every musicians balance thoughout the course of each songs arrangement. The
fact that the ensemble takes included lots of musical interaction made life a lot easier in this respect, but there was still plenty I
could do to enhance their natural performance dynamics. For example, I made the balance of some instruments more
consistent using compression. The bass was typically compressed 26dB with Fabfilters ProC, using a softknee 2:1 ratio
and fairly long 400ms release, and similar settings were used for solo woodwind parts too clarinet, in particular, can have a
very wide dynamic range. Similarly, in some of the pop and rockflavoured numbers, I applied higherratio look-ahead
compression to maintain more dependable kick and snare levels hittohit, carefully adjusting the release time setting in each
scenario to avoid tampering unduly with the respective instruments decay contour.
However, the bulk of the balancing was actually carried out

with level automation, because anything more than
comparatively polite compression usually sounds odd within
primarily acoustic instrument textures like these. The vocals and
instrument solos understandably received the lions share of the
attention here, in order to maximise the intelligibility of the lyrics
and the audibility of expressive performance nuances, but fader
rides also played a role in drawing out entertaining background
details in other parts, as well as in solidifying the bass
instruments levels to form a firm foundation for the rest of the
ensemble.
The final fundamental mixing task was to reference the
balance against some other commercial productions and
massage the overall mix tonality as necessary. I always try to
check this to some extent while tracking, but I knew I couldnt
Here you can see the ambience reverb Mike applied to the
realistically expect myself to nail it down completely when
vocals to make them sit better with the naturally cohesive
liveroom band sound. Notice how the amplitude-envelope
working on location with unfamiliar monitors, unfamiliar
controls within the convolution processor have been used to
acoustics, and scant session time. In retrospect, I reckon the
dramatically shorten the selected impulse responses decay
monitoring in our location control room (an unfurnished 4 x 5
tail.
metre spare room) was basslight, because both the kick drum
and bass were distinctly porky at the low end although nothing that some lowfrequency EQ cuts couldnt easily rectify.
Additional EQ & Filtering

Rarely did it take any more work than that to achieve a reasonable illusion of the whole band playing in a room together, so
our recording strategy had clearly paid off. Mind you, nobody wants just reasonable at mixdown, and mixing is also just as
much about presenting musical events in the most flattering light. As such, I made some gentle EQ refinements on all the
songs now that I could hear everything within a more controlled listening environment. Mostly, this amounted to the usual cuts
to combat frequency masking, for example shaving a touch of upper mid-range from drums or piano to maintain vocal
presence, or rolling off some low end from the piano and guitars to clarify the bass tone. I also did a little work to bring the kick
drum and upright bass through more clearly on smaller speaker systems, usually by boosting the 700Hz1.4kHz octave to
some extent especially for the bands more pop/rockoriented arrangements.
Equalisation did play an important role in adapting the vocal recordings for each song, though, particularly in those numbers
where some voices were alternating between lead and backing roles. A vocal tone that sounds great when a singer is
performing on their own will usually be too fullsounding as part of three or fourpart vocal harmonies, for instance.
Sometimes I multed the vocal parts to make it simpler to set up separate frequency contours for different song sections,
sometimes I just automated the EQ on a single track, but the thinking behind each approach was identical. By the same token,
there were some situations where I automated a band instruments EQ as well, most commonly to add a little more presence
to solo sections.
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I also used filtering to tackle some sonic niggles that wed

been unable to address within our supertight tracking schedule.
On the one hand, these filters tackled spillrelated issues. For
example, highpass filtering often tightened up the kickdrum
sound by reducing its lowfrequency spill on the piano, drum
overhead and snare mics; whereas lowpass filtering
occasionally helped smooth cymbal spill coming through the
saxophone microphone. On the other hand, though, filters also
compensated for two sideeffects of closemiking: low cuts
counteracted proximity effect bassboost (especially on the
acoustic guitar and vocal mics), while high cuts took the edge off
things like stick noises, guitar picking and woodwind key clicks,
all of which are typically overemphasised by miking close up
particularly when youre using condenser mics, which normally
exhibit a sizable onaxis high-frequency boost.
The thing with filters, though, is that theyre not choosy: they
cut out everything, whether you want them to or not. As a result,
there were occasions on which I had to turn to more
sophisticated processing instead. For example, the pad switches
Id engaged on the drum overhead and tom mics during tracking
resulted in recordings that were a little noisy on the quietest
songs, but you cant just filter out this hiss while cymbals are
playing. This is where specialist noisereduction software really
comes into its own, and iZotopes RX2 Denoiser plugin made
short work of the problem, even in its lowCPU realtime mode.
Similarly, I sometimes also used multiband limiting or dedicated
transient processing in place of lowpass filtering to tame ride
cymbal stick noise, guitar picking spikes, and even hard vocal
consonants, again to avoid a general loss of highfrequency
detail on those sources.
Dynamics Spotfixes
Although some compression was used on individual

channels, particularly on the bass and woodwind parts, the
majority of the detailed balancing work for these mixes was
carried out using level automation, as you can see in this
screenshot, for instance.
The control room on the tracking session, which was an

unfurnished spare room without any acoustic treatment, led
to some lowend misjudgments that had to be addressed
when mixing.
In addition to the basic compression mentioned above, more specialised dynamics techniques were used for a few other
spotfixes. For example, I mentioned in the Session Notes column that I made the mistake of over-compressing the vocals
while recording, so that left me with a certain amount of rueful deessing work to do. I often use Toneboosters TB_Deesser
plugin for this these days, because its got a nicely controllable splitband mode that helps minimise lisping. It worked well
here, not only processing the dry vocal, but also ironing esses out of the vocal effect sends. With heavier sibilance processing,
though, you cant afford to set and forget, and I had to automate of the deessing Amount parameter in some instances to
ensure consistent results.
There was a sporadic instrument resonance around 80Hz on the upright bass (by which I mean, it was something that
manifested itself in both the mic and DI signals, so it wasnt just the result of a roommode effect at the miking position), so I
trained a single band of dynamic EQ on that to make its musical lines a little more even for a few of the songs. A couple of the
singers also moved around quite a lot while singing, which made the degree of proximity effect rather variable, and I
occasionally used the lowfrequency band of a multi-band compressor to address this although most of the time I just
automated a lowfrequency EQ shelf in tandem with my general vocalchannel fader rides.
A handful of the ensemble lineups required some extra work

to finetune the mixs frontback perspective. The main offender
here, as I already mentioned last month, was the saxophone,
which generated a lot of spill on our snaredrum and upright
bass mics. While this caused no problems when the instrument
was playing an accompanying role, the room sound prevented
my pulling the players solo parts sufficiently upfront. A solution
was to use my DAWs automation system to tweak the mix just
for those solo sections, bringing in a limitedrange gate on the
snare track and rebalancing the bass sound more in favour of its
DI.
Not only were the lead-vocal levels automated in detail for

every mix, but the vocal deessers Amount control was often
also automated to avoid lisping artifacts on certain specific
sibilants.
Likewise, a few numbers featured louder snare hitting, which gave quite a roomy snare sound in the mix (on account of spill
through the piano and bass mics). Because the tone of the snare ambience was actually pretty nice, this didnt worry me in
most cases, but on a few occasions I chose to dry things up a little with a limitedrange ducker on the piano channels,
triggered from the snare mics. About 34dB of gainreduction was all it ever needed.
Mix Flattery: Effects & Masterbus Processing

Id used a single ambience reverb for basic blending purposes, so there was plenty of scope to polish the mixes with send
effects. My first move was to expand the perceived space beyond the inevitable smallroom acoustic signature of the raw
recordings by using an additional short early-reflections patch. I selected this in much the same way as I had the vocal
blending reverb, but here I deliberately sought out impulse responses which suggested slightly larger rooms. A little of that
reverb across the board opened out the sound significantly.
Another family of reverbs I found useful were longertail patches with less obvious early reflections (often based around
platereverb impulse responses), which I sometimes used to enhance the sustain of the piano, acousticguitar and vocal
parts, especially for the more downtempo arrangements. (This wasnt the only way I increased sustain, however, because
both temposynced delays and parallel compression occasionally played a

role as well, wherever I felt that too much reverb was unduly clouding the mix.)
Given the genrehopping that the band were doing between songs, some of
my send effects were also inevitably more stylespecific. So, for instance, the
poppier arrangements tended to feature more audible vocal echoes; the 80s
influenced tracks brought in a bit of audible chorusing; and slapback delay
effects cropped up more frequently on retro rock/blues tunes.
Finally, there was always some element of masterbus processing involved.
Given the entirely solidstate, digital recording setup wed used, I decided to
run all the mixes through some modelled analogue processing to thicken them
up a touch. Mostly I used Softubes TubeTech CL1B emulation for this, but
occasionally switched to their modelled Summit TLA100 instead if the Tube
Techs flavour didnt seem to suit any specific song. In neither case did I use
their compression, though just the modelled valve saturation. Toneboosters
TB_Ferox tapeemulation plugin also came in handy on a few occasions to
smooth the overall high end.
Some gentle masterbus compression from Cytomics The Glue helped the
mixes cohere slightly better too: I typically used a slowattack, fast release
setting with 23dB gain reduction to steer clear of obvious gainpumping
sideeffects. I usually dialled in a few decibels of air boost as well, using the
18kHz shelf of Variety Of Sounds excellent BaxterEQ freeware plugin,
simply because most commercial productions these days are a little brighter
than reality, and I prefer to make final mix decisions with that in
mind.
A sporadic resonance of the upright bass

instrument required some more specialised
mix processing, namely a band of dynamic EQ
from Meldas MDynamicEQ plug-in operating
at 78Hz.
A Stitch In Time
Many of the band recordings featured in this column take days of
painstaking mix work to transform into a creditable end product.
But this months session demonstrates that it really doesnt have
to be that way despite budget gear, domestic acoustics, and
general time pressure while tracking Spektakulatius, the
mixdown stage still averaged out at only three to four hours per
song. So next time youre recording a band, try to frontload the
production process as much as you can by making sonic
decisions early on, because that makes it much more likely your
mix will look after itself. .
Mix Rescue: Audio Files

The audio files available on this page accompany my Mix Rescue article in SOS July 2015 about mixing two albums I
recorded for the band Spektakulatius. The filenames are fairly selfexplanatory, but here are some additional notes to
describe exactly what youre hearing in each case. For complete details about the setup of the Wayfaring Stranger
recording mentioned below, see www.cambridgemt.com/rsch10case1.htm, and there are also a selection of full
multitracks from this session available for download at www.cambridgemt.com/msmtk.htm#Spektakulatius if you fancy
having a go at mixing them yourself. A threepart video series showing all the basic processing I used for many of the
Spektakulatius mixes can be found at www.cambridgemt.com/rsch10case1.htm#Mix.
BassResonance01
One bit of necessary mix troubleshooting resulted from a sporadic lowfrequency resonance within the upright bass
instrument itself, which you can hear in this audio example. Its most noticeable in terms of the unmusical emphasis it
gives to the fundamental frequency of the E note at 0:04 and 0:06, but it also affects the fundamentals of adjacent pitches
a little too.
BassResonance02_DynEQ
To tame the sporadic resonance in the bass part, I trained a single band of dynamic EQ on it using Melda Productions
MDynamicEQ plugin. The band was set to a frequency of 78Hz, with a moderately narrow Q value of 2, and it attenuated
up to about 5dB at its maximum gain reduction, using an attack time of 37ms and a release time of 432ms. The effect is
still quite subtle, though, if you compare this example with the BassResonance01 file directly.
Enhance01_DryMix
The following eight examples demonstrate how I used
effects and bus compression to enhance many of the
Spektakulatius recordings at mixdown. This first audio file
comes from a song called Wayfaring Stranger and so far
Ive just put together a dry balance of it in the Cockos
Reaper DAW. The only processing Ive used so far is six
highpass filters, four bands of EQ, two compressors, a
single transient processor, and some level automation.
Although the balance is now fairly creditable, the vocal part
sounds a little dislocated from the mix. This is because the
singer recorded her part in a DIY isolation booth, so she
generates no ambient spill on the rest of the bands mics.
Enhance02_BlendVerbSolo
To draw the singer back into the mix and blend her with the
rest of the band, Im going to add a short ambience reverb,
roughly designed to emulate the early reflections of the
room the band recorded in. You can hear this effect in
isolation in this audio example. To hear it in the mix, listen
to the Enhance03_BlendVerbMix file.
The proximity of the piano to the drums meant that some of

the songs which had louder snare hits on them generated a
lot of spill on the piano mics. Although this often enhanced
the snare sound in the mix, there were a few occasions
where Mike side-chainducked the piano channels to reduce
the apparent spill levels and bring the drum more forward in
the mix.
Enhance03_BlendVerbMix
If you compare this audio example with
Enhance01_DryMix, youll hear how the singer now sits
more comfortably as part of the band, on account of the
added reverb showcased in the Enhance02_BlendVerbSolo
file. Notice, though, that the effect is deliberately quite
subtle, to avoid it sounding like an obviously artificial
addition. (To hear the differences more clearly, import the
audio files into your DAW so you can switch
instantaneously between them on the fly.
Enhance04_SpaceVerbSolo
The effect that Ive soloed in this file is designed to expand
the perceived dimensions of the recording room slightly, so
that the smallroom signature of the raw recordings
themselves is less apparent. The reverb is still fairly short,
but in this case its added to all the instruments in the
ensemble, not just the vocal. To hear this reverb in context,
check out the Enhance05_SpaceVerbMix file.
Enhance05_SpaceVerbMix
Mixing in the reverb you heard in the Enhance04_SpaceVerbSolo audio file subtly expands the apparent acoustic space.
Again, though, its not designed to be a massive change, because the aim is to enhance the natural musical event, rather
than making it sound like weve added something artificial.
Enhance06_SustainVerbSolo
Another reverb effect that can flatter acoustic music is a longerdecay patch without appreciable early reflections, of the
type you can hear in this audio example. This kind of reverb helps add warmth and sustain without supplying too much
conflicting spatial information. Notice that Ive deliberately biased the effects send levels to emphasise the piano in the
reverb sound, because I feel that this instrument will benefit most from the enhancement. Listen to the
Enhance07_SustainVerbMix file to hear how this reverb sounds within the full mix.
Enhance07_SustainVerbMix
Now you can hear the mix with the sustainenhancement reverb (isolated in the
Enhance06_SustainVerbSolo audio file) added to it. Although this reverb is slightly more
audible in its own right, its still subtle enough not to distract from the nuances of the
musical performance, especially given that the end listener wont be comparing this mix
with the dry version as we can here.
Enhance08_BussComp
Some masterbus compression from Cytomics The Glue plugin provides a further gentle
cohesion, using a slowattack (30ms), fastrelease (100ms) setting to trigger 23dB of
gain reduction at a 4:1 ratio. A 75Hz highpass filter was engaged in the compressors
sidechain to make it less sensitive to lowfrequency information from the upright bass
and kick drum.
Sustain01_AllOut
Although the Enhance07_SustainVerbMix file has already demonstrated the kind of subtle
reverb processing I used to enhance warmth and sustain on many of the Spektakulatius
mixes, some of the songs demanded more assistance of this kind. The following audio
examples demonstrate some of the other strategies I used for this, in order to avoid
washing the mix out with too much reverb. This audio example contains a section of the
song Forever Young without any artificial sustain enhancements just a subtle
Hammond organ pad I suggested the bands keyboard player add during the tracking
session.
Sustain02_ParaComp
For this example Ive added in a pair of parallel compression channels fed from the
acoustic guitar and piano parts respectively. Both of these channels use Stillwell Audios
fastacting The Rocket compressor plugin to duck attack transients, but the two are then
EQed differently: the guitar has a highpass filter rolling off the low end below 300Hz, and
a further shelving cut focusing the remaining energy into the sub3kHz band; whereas the
piano is highpass filtered below about 200Hz to emphasise the sustain of upper
spectrum frequencies. Compare this example to the Sustain01_AllOut file to more clearly
hear how this affected the mix as a whole.
Sustain03_ParaCompDelays
For this audio example, Ive added in temposynced
feedback delay effects on the lead vocal, backing vocals,
and acoustic guitar. To prevent the echoes drawing too
much attention to themselves Ive reduced the HF levels of
all three effects, as well as deessing the send to the vocal
delay and transientprocessing the send to the guitar delay
to duck the instruments picking noises. Line this file up
against the Sustain02_ParaComp file to hear a before/after
comparison.
A selection of different
masterbus processors
was used for the different
Spektakulatius mixes,
including: Cytomics The
Glue for (typically quite
gentle) compression;
Variety Of Sounds Baxter
EQ for upperspectrum air
EQ boost; Softubes Tube
Tech CL1B and Summit
TLA100A hardware
emulations for valve
thickening; and
Toneboosters TB_Ferox
tape emulator for high
frequency smoothing.
Sustain04_ParaCompDelaysReverbs
Now Ive added in my sustainenhancing reverb, although
in this case its a combination of two patches: a long plate
emulation on the lead vocal, and a slightly shorter hall
reverb primarily for the vocals, piano and guitar.
TrackingRoughs_Overview
This audio file contains a selection of the rough balances I
put together during the Spektakulatius tracking sessions. Not only does this indicate the wide variety of instrument line
ups and musical styles the band covered, but also demonstrates how much of the sound was already in place before any
real mixing occurred these balances were put together with negligible processing within the Roland VS2480
multitracker we used for recording purposes. Compare this file with the FinalMixes_Overview audio example, where you
can hear my final mixes of all these song sections.

FinalMixes_Overview
Heres a some snippets of my final mixes from the
Spektakulatius session, all mixed within the Cockos Reaper
DAW. On average, each of the 28 songs we did on this
session took about three to five hours to record and edit,
and a further three to five hours to mix.
Mixing A Whole Album

When youre mixing a whole album, its human nature to look for some economies of scale by copying settings from one
song to the next. This tends to work best with plug-ins aimed squarely at technical troubleshooting tasks such as reducing
picknoise, hiss, sibilance, or undesirable spill. It can also make a lot of sense to duplicate generic reverbs/delays (like the
vocalblending, roomenlargement and sustainenhancement reverbs mentioned above) and masterbus processing
across all the songs on an album, as this can help give the different songs something of a family sound.
But the copying approach can also be taken too far, in my opinion. Things like channel polarity, compression and EQ
settings tend to be very arrangementdependent, for instance, so I definitely wouldnt copy, say, my kickdrum chain from
one song to another without carefully reevaluating the appropriateness of every plugin within its new context. Another big
reason why I dont like copying too many settings between songs is that rebuilding my processing from scratch
encourages me to experiment with new approaches to similar problems, and frequently yields better solutions and even
if it doesnt the exercise will still improve your mixing chops for your future work.
Finally, its important to realise that mixing an album is usually something of an iterative process, if only because some
lateintheday mix decisions are better made by comparing all the different mixes. This is why I always set up a separate
DAW project containing all my mixinprogress bouncedowns, so I can switch between them to check for balance and
overalltonality inconsistencies that would be extremely tough to spot in any other way. In this specific project, leadvocal
and solo levels were tweaked a fair bit at the final moment, as were the relative levels of the kick drum and bass. Some
might say that this kind of work is best left to the mastering engineer but I disagree, for two reasons. Firstly, youve got
more scope to fix problems at the mixing stage than at the mastering stage. And, secondly, I think the aim (however
unattainable) of any mix engineer should be to create sonics that require as little mastering work as possible. Besides, the
less remedial work you leave for mastering, the more time the engineer can spend concentrating on niceties.
Editing Recordings With Spill

I think a lot of people worry too much about recorded spill limiting their posttracking editing capabilities. In reality, theres
still plenty you can do at the editing stage to improve the end result. The most obvious thing, of course, is to do several
takes and then stitch the best ones together, as we did for several of the Spektakulatius songs. But that same kind of
editing approach also gives you masses of scope to repair smaller performance blunders. There was one occasion here,
for instance, where the bassplayer played the wrong part for three beats of the master fullband take! It was no problem
to patch that up by extracting those beats from an alternate take. Similarly, I could usually replace the odd mishit snare
drum or sloppily timed ensemble stab with a snippet copied from a similar musical section elsewhere in the same take.
The main thing you have to bear in mind is that its always best to edit across the whole multitrack, rather than on
individual tracks: that way, the phaserelationships between direct sound and spill remain consistent. And, of course, that
does mean you have to keep your ears open while tracking to be sure that you have enough material to edit from if
some important fill is fluffed on all your takes, then you wont have anything suitable to patch it with.
Video: Watch Mike Build A Mix

In this article, Ive given an overview of the range of techniques I used to mix all 28 songs I recorded for Spektakulatius. To
provide a more concrete example of my workflow, however, Ive put together a short video series in which I take a section
of one of the songs and demonstrate how little processing is actually required to build up a polishedsounding mix. You
can find these videos on my web site, accompanied by download links for both my Cockos Reaper mix project and the raw
multitrack recordings.
www.cambridge-mt.com/rs-ch10-case1.htm#Mix
Listen For Yourself!

A variety of supporting audio examples are available on this articles associated media page, including snippets of a dozen
of the songs (in both tracking rough mix and final mixdown versions) and demonstrations of several of the mixpolishing
effects Mike used. http://www.soundonsound.com/sos/jul15/articles/mix-rescue-media-0715.htm
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On Buses
Parallel Lines
Key Chain
Group Dynamics
Source Splitting
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Reason Tips & Techniques
Technique : Reason Notes
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We look at advanced mixing techniques in Reason.

Simon Sherbourne
n the last Reason column we covered all the typical ways of using Insert and Send effects in Reasons Main Mixer. This
time were going to look at some more advanced mixing techniques that are handled rather elegantly in Reason.
On Buses
A standard mixing strategy in any DAW or mixer is to create submix

groups. By routing several tracks to a bus which then feeds another
mix channel, you split off part of your mix into a more easily
manageable chunk, and make it possible to process that section as a
single entity. These stems can also be bounced very simply to create
remix-ready deliverables or archives. Traditional submix food groups
include drums, backing vocals, guitars and so on, but can be any
sources that need lumping together, like that ridiculous fourlayer
dubstep bass you made.
By default all tracks are routed directly to the Master Section of the
Main Mixer; in other words the final stereo mix bus that you control
with the Master Fader. Each track has a panel under the fader,
handily labelled Output, that indicates where the track is routed. This
panel will be grey and blank on any track that routes straight to the
Master. Click on the panel and youll get the option to reroute to a
new Output Bus. So to combine two or more tracks to a new bus,
select them all in the mixer, then click the output panel on any of the
tracks and choose New Output Bus. The keyboard shortcut for this is
1 & 2: Creating a submix group in the Reason Mixer.
CmdG/CtrlG. A new mix channel will appear with a red fader, and
the new bus name will appear in the Output panel on all the tracks. The panel is also colourcoded the same as the new bus
fader, making it easy to see the group. After a new Output Bus has been created it will appear in the list when clicking any
tracks Output panel, allowing you to add other tracks to this group.
An Output Bus channel has all the same features as a regular mixer channel: EQ, Dynamics, Insert point, Sends and
Panner. There are of course many applications for this structure, such as filtering a multilayered sound as a whole, as well as
EQing within the individual parts or adding consistent reverb to multitracked backing vocals. A very common application is to
add Bus Compression to drums. This can be achieved by applying the mixer Dynamics on your drum Output Bus. More
commonly, though, you might want to add the MClass Compressor or Limiter as an Insert Effect on the drum bus or add
character with something like Pulveriser or a favourite Rack Extension.
Parallel Lines
You dont need to read many Secrets Of The Mix Engineers articles to see that parallel effects processing is a key technique
in modern music mixing. The trick is to split a source signal and bring it in on more than one Mixer channel, allowing you to
apply different effects and processing to each copy and blend the results. Luckily you dont need to do any manual cabling or
mixer routing to achieve this, as Reason is already equipped for the task. Simply rightclick on a mixer channel and choose
Create Parallel Channel. Reason will create a new Mixer channel next to the original, and set up the routing for you (check the
back of the Rack if youre interested to see the patching). You now have two independent versions of the track.
This technique is probably most commonly used for
processing a drum bus, typically blending a heavily compressed
and/or driven version of the drums with the original clean mix.
This is an incredibly effective trick for achieving punchy drums,
giving a big, loud sound, while maintaining dynamics and
transients. This uses both the functions weve looked at so far:
routing tracks (the drums) to an Output Bus, then creating a
Parallel Channel for the new Bus Master channel. In screen 3
Ive set this up, then on the Parallel Channel Ive inserted an M
Class Maximizer and a Scream Distortion set to Tape Sat mode.
Reason lets you make multiple Parallel Channels for a track,
opening all kinds of fun possibilities. For example you could
create several completely different sound treatments for a track,
then use fader automation to bring in different versions during a
song. This is much easier and more handson than trying to
automate multiple effects devices, and also allows you to morph
different-sounding versions into one another.
Key Chain
If youre already familiar with using sidechains in the Main Mixer, feel free to skip ahead to the next section where well bring
everything together and look at how Output Buses and Parallel Channels can be used together with sidechains. Keep
reading for the sidechain basics...
Sidechaining allows you to trigger, or key, a compressor (or other
dynamics effect) from a different audio source than the one that is being
processed. The classic example is using a kick drum to duck other parts
(typically bass) to makes space for itself. Many Reason devices have
sidechain inputs which you can connect manually, and the Dynamics
section of the Main Mixer is no exception.
Unlike the other tricks weve looked at, theres no automatic way of
patching a sidechain, youll need to do some manual cabling. Each Mix
Channel rack device has a sidechain input on the rear panel. To side
chain the channel dynamics, you need to feed this from another
channel. As weve already seen, each mixer channel has a pair of
Parallel outs that provide a clean, prefader split of the channels input.
Use this to connect the sidechain (screen 4). When the connection is
made, the Key button will automatically engage and youre in business.
Of course, Reasons modular environment means that this mixerbased
setup is only one option. You can freely cable in any audio signal from
the rack as a side-chain, and you can also feed the sidechains of any
dynamics processors in the Insert FX section, not just the builtin
Dynamics (see the box for help with cabling).
Group Dynamics
DAW Techniques
3: Setting up parallel compression is easy in Reason.
As a final flourish lets look at a popular mixing trick that brings all
these concepts together: making a whole mix pump around a
kick drum. In this scenario you use a kick drum to key the
compression on many of the other tracks. It would be a patching
mess to try to feed all the other tracks individually, and in any
case you might be using the dynamics on other channels. You
4: Each Mix Channel device has an input for keying the
also want the compression to be consistent, so it really needs to
Channel Dynamics
happen after all other effects. The answer is to submix all the
sources you wish to pump to a new Output Bus. Some of those sources may already be submixed, but thats no problem as
you can route Output Buses to other Output Buses (sometimes called nesting buses).
In screen 5 Ive created an Output Bus called Pump It! to carry all the tracks that I want to breathe around the kick. Ive left
the vocals and the kick itself going to the main mix, as I want them left untouched. A parallel output from the kick is then
connected to the sidechain of an MClass Compressor, which is inserted on the Pump It! bus. A bit of tweaking of the
compression settings and the whole backing track starts to throb around the kick in classic dance-music fashion. Come on,
just because its cheesy doesnt stop it sounding awesome!
.
5: Using side-chaining and a final submix bus to create the

classic kick pumping sound.
Source Splitting
Many Main Mixer routing functions are performed automatically: for example, bus routing in the Reason Mixer is handled
internally (no cables), and the cabling for creating Parallel Channels is done for you. However, as we saw with keying the
mixer dynamics, sometimes you need to go off piste and get your hands dirty with a bit of cabling. An example is splitting
individual sources from multitimbral instruments. In some of the examples I had a kick drum on its own channel, but in all
cases the Kick was coming from a Kong or Redrum instrument. These devices submix sounds within themselves and
send a stereo mix to the Mixer. In order to separate out the Kick, I created an empty Mix Channel (Create / Utility / Mix
Channel) and cabled a direct output from the Kong or Redrum. At other times you might want to split a signal to route to
more than one place, and maybe there are no Parallel outputs left. In this case Reasons Spider Audio Merger/Splitter
device is your friend, adding a tiny patchbay to the rack where you can cable a signal to multiple destinations. You can
also use the Spider, or either of Reasons smaller Rack mixer devices (Mixer 14:2 and Line Mixer 6:2) to create submixes
completely independently of the Mixer.

Cubase
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Plug-in Pruning
Cubase Tips & Techniques
Technique : Cubase Notes
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When creating a new Collection, you can base it on the entire plug-in list or an existing Collection, or you can start
from scratch by selecting Empty.
Take some of the work out of your workflow, with Cubase 8 Pros Plug-in Manager.
Carsten Kaiser
ere really lucky these days to have such instant and affordable access to so many high-quality plug-ins. But, for
many of us, acquiring new plug-ins becomes an unhealthy habit that gets in the way of actually using them to make
music. How many times have you clicked to insert a plug-in and then spent an age browsing through an endless list
in search of the one effect or instrument that will do something special? Several minutes later you still havent found it and,
worse, youve lost your creative mojo.
While experimentation is obviously important in music production, theres a time and a place for it, and such an unfocused
trial-and-error approach to plug-in selection is seldom productive. While its possible to search for a plug-in by typing its name,
this only works if you can remember that name, and its not going to help you browse a few select alternatives for the same
task a more focused way of browsing or those of us who prefer to use a mouse/trackpad rather than the computer
keyboard.
Thankfully, Cubase 8 Pro has a feature that can help. Its called the Plug-in Manager and if you use it wisely it will save you
lots of time and help you keep your focus on the music. In this months column, Ill suggest some strategies to get the best out
of it.
Basic Setup
The basic idea of the Plug-in Manager is that you can impose your own preferred structure on your plug-in collection by
creating what Steinberg call Collections of plug-ins. Its really easy to use, too. To get started:
1. Go to Devices / Plug-in Manager to open the Plug-in Manager window.
2. Open the Collections submenu and click on the small arrow at the right-hand side of the window.
3. Choose New Collection / Empty from the drop-down list, type in a suitable name (eg. EQs) for your Collection and press
OK.
4. Add your choice of plug-ins to the Collection by dragging and dropping from the left to the right window pane.
5. To organise long lists of plug-ins within your Collection, create and name folders and drag and drop your plug-ins into those.
You can create as many folders and subfolders as you like. And if you feel the need to move plug-ins from one folder to
another, you can choose one or even multiple plug-ins and drag-and-drop them to another folder of the respective Collection.
The same goes for subfolders.
Choosing Plug-ins from your new Collection is really easy. In any Cubase project, go to an insert slot and click, as you
usually would to select a plug-in. Instead of seeing all of the plug-ins that are installed, youll see just one Collection displayed,
which will be the last one you created, or the last one you selected. To select another Collection, click on the little arrow at the
upper right-hand side of this plug-in selection window and browse for the Collection of your choice from the drop-down list,
then select the desired folder, subfolder or plug-in.
Youve not lost all your other plug-ins, by the way! If youd prefer to see Cubases standard overview of all plug-ins, just
select Standard. This feature is always available, whatever Collection is selected.
Divide & Rule

Useful as this exercise is, weve only covered the basic mechanics of the Plug-in Manager, and before you plough ahead, Id
like to offer a word of advice: stop! Do not follow the above steps without first having laid out a carefully chosen plan for these
Collections. Why? Unfortunately, Cubase saves your plug-in Collections in the order you create them thats the order in
which theyll be listed when you go to select which one will appear, and this cannot be changed. Also, while you can move
plug-ins between folders and subfolders within a Collection, you

cannot move them from one Collection to another.
All this means that, before you start creating Collections, you
really need to design a Collections structure that will give you the
flexibility to reorganise things easily in the future. I recommend
setting up Collections for more stable categories, and to use
folders/subfolders for structures that are likely to be under
constant development. Lets look at an example of how you
might organise things.
More Cubase Notes
I like to set up Collections for certain tasks or track types. So I

might have one Mix Task Collection, for instance, which
contains folders that span from Recording to Mastering, with
further folders broken down by the process (widening, saturation
To load a plug-in from a Collection, you click on an insert slot
and so on), as well as another Collection named Track Types,
in the normal way (1) and select the Collection using the
drop-down arrow at the top-right of the plug-in list (2).
this one housing folders with plug-ins I regularly find myself
wanting to use on the drum bus, guitars, lead vocals and what
have you a bit like a slightly more fluid take on channel strips and presets. Note that you can place a plug-in in multiple
folders, so you could make the same compressor appear in your bass, guitar and vocals folders, for example.
As every user has his or her unique workflow, its not possible to give the perfect recommendation for how to organise
things here what Ive described above may or may not work for you. But hopefully youll see how important it is that you ask
yourself what a useful structure will look like to you.
In my opinion, though, you cant go wrong in creating a Collection that sorts your plug-ins into a Collection named FXCategories, with folders for Dynamics, EQ, Modulation and so on.
Alternatively, you might pigeon-hole your plug-ins by software developer/vendor. This is already possible without Plug-in
Manager via Cubases preferences, but now its possible to combine this with other approaches. For example, perhaps you
have a huge bundle of Waves, Melda or UAD plug-ins. Creating a Collection for each brand, you could create subfolders for
each plug-in type in each brand. While this kind of sorting might not always save you many mouse clicks, it does help to keep
you focused it instantly makes your unwieldy plug-in list a whole lot more wieldy!
Empty Space
While it might seem strange, I also propose that you create some
empty something I call blind or dummy categories and
folders, which do not contain any plug-ins. The reason is that
Cubase allows Collection names which consist of punctuation or
even blank spaces, and these may be used to impose a visual
structure on your plug-in list. You could also save blind entries
named in capital letters to get dividing headlines. In the
screenshot you can see six filled Collections (Mix Tasks, Track
Types etc.) and six blind Collections (spacers, hyphens, all
caps).
This little knack provides you with an arsenal of dividing lines,
spacers and headlines and will thus help you to overlook all
other entries. And the best thing about it is, if you open up your
list of Collections via an insert/VSTi slot, all your Collections will
be displayed according to your personal needs. Well, thats what
I call an overview!
In the above screenshot, you can see a handy solution, in
case you are using bundled plug-ins. If you still feel the need to
structure your Collections even further, you could additionally
work with folders that duplicate the main categories you just laid
out for your FX-Categories Collection. You could have the
same structure (Analyzer, Delay, Distortion etc.) as subfolders
for each bundle Collection.
One quick and useful way to organise your plug-ins is to

place them in effects categories. Note that you can assign
one plug-in to multiple categories if you wish.
So, lets have a look at a possible structure of Collections and

subfolders that makes use of all of the above suggestions. As
you can see in the screenshot, I am working with Collections that
represent classic effects categories, mix tasks, track types and
bundles. Subfolders give additional structure to bundles that
contain several or many plug-ins thats particularly useful for
those bundles that install everything as a demo version, even if
you only own a couple of plug-ins (such as the UAD collection).
Blind Collections appear as headlines and dividing
It can be helpful to separate your different Collections using
lines/spacers. By combining all these tricks, you can profit from
spaces and punctuation characters.
all the advantages the new Plug-in Manager has to offer you.
Whether you are in need of the right compressor, the perfect sound for an instrument or that certain plug-in from inside a
bundle, you will find it dead on target. And whenever you are going to select a new VST, Cubase will open up the plug-in
Collection you have recently worked with.
Rundown
It will inevitably take some time to set up such a structure. And yes, staying organised can be quite some work, too
whenever you install a new plug-in, you should make sure to assign them to your relevant Collection(s), or they will only show
up in the Standard view of the plug-in list and your time investment will have been wasted. But the Plug-in Manager already
has the potential to save most of us far more time and head-scratching than we currently spend on pretty much every project
we work on. It would be great if Steinberg were to make a few improvements in terms of flexibility, such as allowing you to
reorder your list of Collections, but until that day arrives, the key to making this system work for you is to grab your pen and
pencil and develop a good plan for the structure of your Collections. .
DAW Techniques
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Technique : Ableton Live Notes
We show you how to analyse a song using Lives built-in tools.

Len Sasso
hen analysing a piece of music for its structure and content, theres no substitute for careful listening away from your
computer and your instrument. But when you need a closer look, Live is a great tool for breaking a song down into
its component parts. This month were going to look at how Live makes it easy to implement key elements of this
process. You can do most of the things described in either Session or Arrangement view, but because of its timeline
orientation, it is usually easier in Arrangement view, and thats the approach Ill take. You can find more information on the
hows and whys of song analysis as it applies to composition in the new book, Making Music, by Ableton Head of
Documentation Dennis DeSantis (makingmusic.ableton.com). It is a tremendous source of ideas for getting started, getting
finished and getting unstuck.
Its About Time

When you import an audio file to an Arrangement view track for analysis, youll probably
want it to play as recorded rather than trying to match Lives tempo. For that, youll need
Lives Record/Warp/Launch preference Auto-Warp Long Samples set to Off. Notice that
when you change Lives tempo the audio clip stays locked to the Time ruler at the bottom
of the arrangement, and the Beat ruler at the top moves relative to the audio. You can
warp the clip to Lives tempo and tie the clip to the Beat ruler by clicking the clips Warp
button in Lives Clip viewer. That will place a Warp marker at the beginning of the clip and
set its warping tempo (Seq. BPM in the Clip view) to Lives current tempo. The clip will still
play as recorded unless you change Lives song tempo. If the song youre analysing is
arhythmic, set Live to a convenient tempo (240bpm will align Lives bar lines to Time ruler
seconds) before warping the clip. If the song has a fairly consistent tempo, it is worth the
effort to figure out what that is and then warp the song at that tempo and set Live to the
same tempo. If the song has significant tempo changes, you can warp each section and
automate Lives tempo to follow.
The quickest way to deduce the tempo of a song or song section is to use Arrangement
views Loop Locators to isolate a short loop and then manually stretch or shrink that to the
correct number of beats. To isolate the loop, ensure Lives Snap to Grid option is turned off
(Command-4/Control-4), start the loop playing and then drag Arrangement views Loop
Start and End Locators until the loop plays correctly. Once youve done that, heres how to
calculate the tempo:
1: A loop in an Arrangement view

audio clip is captured, split and
consolidated, warped and then
shrunk to match its true length of
four bars.
Click the Arrangement view Loop Brace to select the section of the clip within the loop.
Split and consolidate that section (Command-E/Control-E followed by CommandJ/Control-J).
In the Clip viewer, youll now see a Warp marker at the beginning of the loop. Right-click it and select Set 1.1.1 Here from
the drop-down menu.
Double-click at the end of the clip to place a Warp marker there and drag that Warp marker until the loop spans the correct
number of beats. The Seq. BPM reading in Lives Clip view
will now be set to the loops tempo.
When you know the loops tempo, set Lives tempo to match,
replace the clip on the Arrangement view track with the original
audio file, click its Warp button in the Clip view and, if need be,
set its Seq. BPM to match Lives tempo (see screen 1).
When Lives tempo is matched to your songs, song sections
will in most cases fall on or very close to bar lines. You can clickhold in the Scrub area at the top of the Arrangement view to find
the exact start of any section and then right-click in the Scrub
area to add a Live Locator there. You might also want to use
locators to mark other events that are difficult or impossible to
fully analyse on the fly. For performance, that might be a fast
solo or some chords buried in the mix. For composition it might
be layered sounds or the transition from one section to another.
2: To copy a section of audio from Arrangement view to

Session view without disturbing the original, first select the
area to extract and then click-drag on the clips label while
using the Tab key to switch to Session view.
What You Hear

The first step in getting to grips with a mysterious part is to isolate the snippet you want to analyse and move it to a Clip slot in
Session view. To do that, click-drag in the Track display of the audio clip in Arrangement view to select the desired region.
Now click-hold the clips label above the selected region, hit Tab and drag the clip to a Session view clip slot. More than the
selected region will probably have been copied, but the selected region will be enclosed in the clips loop. Right-click on the
clip and choose Crop Sample from the drop-down menu to crop the clip to the loop (see screen 2).
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If need be, use the *2 button in the Warp section of the Clip
view to double the clips length and thereby halve its playback
speed. In Complex Warp mode, thats usually enough to learn
the notes whizzing by. For hard to decipher chords, you might
want to try a third-party utility such as Transcribe!, a free Mac
and PC application from Seventh String Software
(www.seventhstring.com). The easiest way to use a stand-alone
sample processing application like Transcribe! to process Live
clips is to select it as the Sample Editor in Lives File/Finder
preferences. Clicking the Edit button in the Sample section of
Clip view will then launch that application and load the clip (see
screen 3).
Homing In
3: An audio clip comprising a piano chord is opened for
When youre trying to ferret out a specific part from a stereo
analysis in Seventh String Softwares Transcribe! (bottom).
audio clip, you have three parameters to work with: pan,
frequency and level. Although none of those will completely
isolate the part, one or more of them can make it more prominent. Lives Utility, EQ and compression devices are helpful tools
for this. Lets start with pan. A sounds position in the stereo field comes down to how much of it is present in the left channel
and how much in the right. If the sound appears in both channels, you can use the pan control in either Lives mixer or the
Utility device to home in on the position that provides the least interference. If the sound appears in only one channel, use
Utility to reduce or eliminate everything that appears in both channels in order to make your sound more prominent. To do
that, set Utilitys Width to 200 percent and click its Phz-R (phase-invert) button. (Youll find more details on using Utility for
mid/side conversion in the September 2012 Live column: www.soundonsound.com/sos/sep12/articles/live-0912.htm.)
After youve manipulated the pan position as best you can, try
using one of Lives EQs to create a frequency band around the
pitch range of the part youre after. EQ Three is the easiest: just
turn off the GainLow (L) and GainHi (H) bands and play with the
two frequency knobs to narrow and position the GainMid (M)
band. The 48dB/octave mode is typically best for this. When you
need visual feedback and more control over the shape of the
curve try EQ Eight instead. Start with two filter banks, one set to
low cut and the other to high cut, and position them to best
capture the part youre after. Then add one or more bell-shaped
banks between them to see if you can make the part clearer. If
youre still not quite there, you might be able to use compression
or limiting to suppress hot transients from other parts, allowing
you to raise the levels underneath. In many cases, youll only
need one of these options, but if you use more than one, the
order listed is a good place to start.
.
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4: In this clip, piano is on the left, bass is in the middle and

drums are on the right. Utility solos the right channel (drum
and bass), and EQ Eight emphasises the bass frequencies.
Limiter then suppresses the loudest drum transients allowing
you to bring up the bass level.
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In this article:
Twos Company...
Threes A Chord
Miles Ahead
Expressive Techniques
Grace Under Pressure
Pop Horns Are Go
Coming Up
20 Assorted Brass
Arrangements
Instrumentation &
Texture
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Top Brass
Arranging For Brass: Part Three
Technique : Composing / Arranging
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A simple, catchy tune with a funky rhythm may be all you need to create a highly
commercial horn hook but harmony is also an essential ingredient in brass
arrangements.
Dave Stewart
great horn arrangement can galvanise a record, and Beyonc Knowles knows it. When asked in a 2004 MTV interview
what made her Crazy In Love song so lively and infectious, the Texan singer summed it up succinctly: Its the horn
hook. Earlier this year, when the same TV channel sought expert advice on the question Why Is Uptown Funk so
catchy? (a topic which never fails to provoke heated debate in the House of Lords), UK musicologist Joe Bennett opined,
The big release, the most exciting bit of the song, is the brass riff.
The horn hook on Crazy In Love is actually a sample of a lick from the Chi-Lites 1970 song Are You My Woman? (Tell Me
So). Like all good stick-in-your-head riffs, the horn part is deceptively simple a triumphant-sounding unison phrase played
twice, the second time with an altered last note, over a standard C-major to A-minor chord sequence propelled by a driving,
funky beat. Despite initial concerns that this iconic sample might be regarded as old-fashioned, Beyonc has since praised its
crucial contribution to the songs runaway success.
Its unlikely that Mark Ronson had any such qualms when co-producing Uptown Funk with Bruno Mars: since the aim was
to make a classic funk song, its seventies-inspired instrumental horn break comes as no surprise. A jaunty four-note lick parps
out three times in quick succession, followed by an offbeat stab and a short descending run; after a bars rest, the vocal hook
Dont believe me, just watch is answered by a staccato, machine-gun-like burst of six repeated notes (doubled by snare
drum) this short call-and-response is repeated three times at the end of the break. Ive notated the horn part in diagram 1
and, as you can see, its rhythmic mini-phrases are classic funk
figures, animated, catchy and danceable. The horns are voiced
in time-honoured style, with saxes and trombone doubling the
high trumpet part an octave down, underpinned by an alternating
Dm7 to G7 chord sequence.
Twos Company...
The Uptown Funk instrumental break horn line, transcribed

by Dave Stewart. The part focuses on rhythm, with only four
notes used in the melody note how the first F D C D
phrase is played three times in the first bar, with the second
iteration falling on the offbeat between beats two and three.
This passage starts at 2.21.
I hesitate to suggest that the two aforementioned worldwide

smash hits might be lacking in any way, but from a strictly
musical point of view, its true to say that their horn arrangements
rely mainly on rhythmic syncopation, with little or no melodic development. Thats not meant as a criticism these parts are
highly effective, and to elaborate them would only dilute their impact. However, a good brass arranger needs more strings to
his or her bow than funky licks and a catchy tune: at some point, youll also need to consider the vital ingredient of harmony,
that mysterious, beautiful musical element that can transform base metal into gold.
Because the highest part in a horn arrangement defines the melody, catches the ear and gives the music its character, the
convention in brass arranging is to write harmonies underneath a top line. (This isnt a hard and fast rule, but bear in mind that
if you write an upper harmony, youre effectively creating a new melodic top line, with the original melody relegated to a
supporting role!) Applying the harmonies underneath approach to the rising line shown in diagram 2 (written over an
ascending chord progression of D, E-minor, F and G) gives us
the simple thirds harmonisation depicted in diagram 3 the
lower harmony part is marked in red. Chord names are written
above the stave, with note names displayed underneath.
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If youre thinking, Oh no, not another music lesson, please

bear with me Ill get back to the juicy funky stuff before long,
but want to give a few basic pointers in horn chord construction
which will be useful in many different musical situations. Feel
free to skip the music theory if you already know it, but if not, rest
assured that making an effort to get your head round it will be
time well spent far from being a dangerous thing, a little
musical knowledge is actually extremely useful!
A simple rising line written over an ascending chord

progression of D, E-minor, F and G.
Staying with the same chord sequence, we can move the

melody line up to the notes of A, B, C and D (each of which
forms a fifth interval over the root note of its chord). As in the
previous example, this can be simply harmonised by adding
thirds underneath, as shown in diagram 4. Pursuing the same
logic, we can move the top line up further to play the octave of
each chord (ie. a high D, E, F and G), but finding a lower
harmony for that is less straightforward: adding thirds
underneath doesnt work in this case, and a harmonisation of A,
B, C and D (as in diagram 5) gives us a row of fourths, a rather
plain and austere-sounding interval. Unless you particularly want
that effect, a better solution would be to keep the same lower
harmony of F#, G, A and B used in diagram 4, which creates an
interval of a sixth between the harmony and melody notes (see
diagram 6).
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Thirds and sixths enjoy a special musical relationship (which

Diagrams 3-6 show how adding a lower part (marked in red)
to a melody line creates simple harmonisations.
becomes evident when you compare the note names in
diagrams 3 and 6), and share a similar, sweet, easy-on-the-ear
sound. Choosing between them is purely a matter of taste: being the most ubiquitous harmony in popular music, thirds can
sound trite and nave if overused, but they work well in punchy, upbeat rhythmic passages, and their unpretentious simplicity
fits the cheerful DIY ethos of music styles such as ska pop-punk. The wider sixths voicing works well in slower music and
spacey ballads (the horn intro on the Exodus / Kaya Demos version of Bob Marleys Natural Mystic is a good example). But
ultimately, there are no rules you should write what you think sounds best, and feel free to mix and match different harmony
voicings throughout a song.
Threes A Chord
While many bands get by with two horn players, it takes three to complete the basic chordal picture. Diagram 7 shows the
three possible close voicings of a simple three-note D-major
chord. We can expand the voicings by dropping the second note
from the top down an octave, as in diagram 8 this vertical respacing introduces wider fifths and sixths intervals, which has
the effect of breathing air into the chord. Though an unsupported
fifth (such as the A and D in the last chord) can sound stern and
serious, placing the third (F#) in the bass creates a lovely, lyrical
sound.
In jazz circles, this method of expanding chord voicings is
known as drop 2. While its a musically sound technique, I
wouldnt advise that you apply it every time you need a wider
voicing: a more creative, open-minded approach would be to
experiment with different wide intervals on a keyboard until you
find some agreeable note combinations. Whether youre working
on musical details or simply trying to dream up basic ideas, trial
and error is an important part of the creative process. Embrace
your mistakes with a little minor readjustment, they often lead
to exciting new possibilities!
The three possible close voicings of a three-note D-major

chord. Middle C is marked in blue.
You can use the drop 2 method to introduce wider intervals

into chord voicings.
Although you can get a fair bit of mileage out of simple major and minor chords, it would be sad to limit your horn charts to
those basic harmonies. Suspensions such as sus 4 and sus 2 chords can spice up an arrangement, and sound great played
on three trumpets the example in diagram 9 shows those two
chords in the key of A, resolving to a major triad in the third bar.
As an alternative to block chord changes, you can resolve
suspensions in an interesting way by introducing an inner
contrapuntal movement, as illustrated in diagram 10.
Miles Ahead
Another simple three-note chord that sounds good on horns is
the minor seventh, voiced as in diagram 11 with horns playing
the top three notes. This chord voicing has a cool, slightly jazzy
flavour, and shifting it up a tone and back down again over a held
bass note (as shown in diagram 12) creates a nice chordal
movement. A version of this occurs in funkmeister James
Browns 1967 single Cold Sweat. The song has an interesting
lineage Browns colleague and co-writer Pee Wee Ellis
recalls, I was very much influenced by Miles Davis and had
been listening to So What six or seven years earlier, and that
crept into the making of Cold Sweat. You could call it subliminal,
but the horn line is based on Miles Davis So What.
Sus 4 and sus 2 chords sound great played on three

trumpets this example shows those chords in the key of A,
resolving to a major triad in the third bar.
Suspensions can be resolved in an interesting way by means
James Browns Cold Sweat does indeed bear a resemblance

of an inner contrapuntal movement, marked in red.
to Miles So What even the title sounds similar! Although both
are based on the simple two-chord movement shown above, So What famously employs a fourth interval under the right
hand chords (notated in diagram 13), creating a super-cool, sophisticated jazzy sound. The Cold Sweat version is simpler; as
you can see in diagram 14, it uses a higher, triadic inversion of the chords and dispenses with the E to D movement in the
bass, replacing it with a repeated funky bass line in D.

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Both of these horn licks get their rhythmic drive from the characteristic long-short phrasing I mentioned in my previous
article; if you play both chords the same length, the feel evaporates! The Cold Sweat riff is a great, simple pattern which
could be repeated by horns behind a solo, while the fourths-based harmony of Miles Davis chords is worth exploring if you
want to take your brass arrangements into more adventurous territory.
Expressive Techniques
OK, enough on harmony already. Its important, obviously, but its a huge subject, and we need to consider other matters such
as, how you add musical expression to sampled or synth brass instruments.
One simple way is via the use of volume swells, or crescendos as theyre known
in the trade. A regular crescendo note starts quietly and grows in volume, an
exhilarating sound when performed by horns as the volume of the instrument
increases, so its tone opens up from a warm hum to a bright, commanding shout.
The reverse process, a diminuendo from loud to soft, is also very effective. You can
hear both at work in KC & the Sunshine Bands disco classic Thats The Way (I Like
It), starting in the verse section at 0.35; the horns play a crescendo and diminuendo
on alternate bars, a mobile and engaging dynamic effect. So you can see how it
looks in a score, Ive notated this dynamic movement (using a melody line of my
own!) in diagram 15.
If youre looking for something more animated, applying short crescendos to
repeated, fast staccato trumpet notes (as shown in diagram 16) is a great attentiongrabbing device. A good example of it occurs at 2.37 in Jerry Heys terrific horn
arrangement for the Tubes Tip of My Tongue its also worth listening out for the
amazing raucous unpitched glissando horns effect which occurs shortly
beforehand. Although such dramatic techniques may be beyond the remit of straight
pop arranging, they could be just the thing you need to energise a film cue.
A simple three-note voicing of a minorseventh chord.
A nice variant on straight volume swells is the forte-piano

crescendo that may sound like a contradiction in terms, but
what it actually means is a loud, accented long note which goes
quiet immediately after the initial accent, then swells up again in
a crescendo. Combining the rhythmic impact of the accent with
the dynamic expression of the crescendo swell, this style works
particularly well on horns, and has become a fixture in pop brass
writing.
You can simulate brass swells and fades on a synth by
sweeping the low-pass filter cut-off frequency (not to be confused
with filter resonance, which makes the sound thinner) of a brass
patch. Some modern workstations assign a front-panel knob to
this function: Jonah Nilsson of Dirty Loops uses the technique to
play some great, powerful brass-like swells on his Korg X50 in
the trios song Hit Me. It doesnt sound much like real horns, but
in the context of this synth-driven track, it works well using
Shifting the minor-seventh voicing up and down a tone over a
held bass note creates a nice chordal movement.
real players would arguably have been overkill. As with the socalled Minneapolis sound pioneered by Prince in the late 70s,
Nilssons highly rhythmic synth playing features the syncopated stabs,
accents and phrasing of funk horn riffs, demonstrating that in pop an
artificial sound can be OK, providing the musical intent and spirit is
right.
More realistic-sounding crescendos can be achieved by using
samples, but it does involve a bit more legwork. Although some pop
brass sample libraries supply played crescendos of different lengths,
making them fit the timing of your track can be a head-scratcher. The
current convention in sample libraries is for users to create their own
volume swells and fades using special dynamic cross-fade patches;
these give you control of the crescendos timing by using the mod
wheel to fade between the instruments dynamic layers, which can
sound great on brass long notes.
The forte-piano crescendo effect is harder to simulate. While few
pop brass sample libraries contain this particular articulation, most of
them do have straight forte-piano (fp) samples, so you can fake the
style by layering an fp sample with a crescendo sample. I dont own
the library, but it appears that Native Instruments Session Horns Pro
offers a more elegant solution by providing tempo-synced forte-piano
crescendos played over two or four beats, which can be timestretched if you need them to be longer or shorter.
The cool, sophisticated jazzy voicing used in Miles

Davis So What introduces a low fourth interval under
the minor-seventh chords.
Grace Under Pressure

Small musical details can make all the difference to the realism of a sampled horn arrangement. Looking at the musical
examples shown above, youll notice that the first chord of the Cold Sweat lick in diagram 14 has grace notes on the front,
each of which is a tone below its target pitch. I tried playing the top line as written on a synth using a sustained saxophone
patch, but no matter how carefully I phrased it, the transition from the grace note to the main note sounded clunky. It became
clear that a sampled solution was required.
Most pop brass libraries contain grace-note articulations but few offer a choice of interval one notable exception is the
Vienna Symphonic Library, which provides grace notes for every interval up to an octave. Although Ive generally cautioned
against using orchestral libraries for pop brass work, VSLs saxophones are excellent, so I set about programming the top line
for alto sax using the Vienna Instruments Pro player. This entailed creating a two-way keyswitch between a grace note patch
and a sustained note patch, then playing the part as shown in diagram 17, with right hand playing the melody (marked in red)
and left hand operating the keyswitches (marked in blue). The keyswitches (which are silent) triggered the patch change
depicted in diagram 18, creating a smooth musical elision which sounded perfectly natural. Yes, thats a lot of work for one tiny
musical moment but if 100 percent realism is your goal, this is what it takes.
Pop Horns Are Go

Id like at this point to introduce UK trumpet player Tom Walsh:
having earned his spurs as lead trumpet player with the National
Youth Jazz Orchestra, Tom has worked with some of the UKs
top big bands and also played principal trumpet in orchestral
settings with John Wilson and the RTE Concert Orchestra. A
disciple of US horn-arrangement guru Jerry Hey, Tom says,
Thanks to the jazz arranging and composition tutelage I
received at music college, and using Jerrys writing as an
influence, I got started on writing my first horn arrangements on
top of existing pop tunes for practice, and ever since then Ive
been bitten by the bug! Ive included diagrams of two short
extracts from one of Toms horn scores, which illustrate how
some of the musical examples in this article can be put into
practice in a pop arrangement.
The first, diagram 19, shows the abovementioned forte-piano
crescendo dynamic in action. Scored for (from the top down)
three trumpets, tenor sax and trombone, the instruments start on
a unison note of D, with sax and trombone playing an octave
below the trumpets. On the ensuing chord, trumpet one plays a
high E, trumpet two takes the D a tone below that, and the third
trumpet plays the A below the D, making a composite chord of
Asus4. The sax and trombone strengthen the chord by doubling
the lower two trumpet notes an octave down. Having hit an
accent on the eighth-note between beats two and three, the
horns quickly drop down in volume then play a crescendo swell
over six beats, cutting off on the and of beat four, an eighth-note
before the end of the bar. This may seem an unimportant detail,
but leaving that small gap creates a breathing space which
increases the impact of the incoming chorus.
The second of Tom Walshs score extracts occurs in the songs
intro: it illustrates a typical horn arrangement technique, a unison
phrase culminating in a chordal stab. As in the previous
example, the three trumpets start off in unison, then divide to
play the three notes of the final chord; the lower horns double the
trumpet melody an octave down, then reinforce the chord with a
pair of lower notes. By coincidence, the chord in question is the
Miles Davis So What voicing mentioned earlier, played up the
octave in E-minor (though the bass line remains in D throughout
the two bars). You can see this notated in diagram 20. Thanks to
Tom for providing these extracts well be hearing more from
him next month.
A simplified, funky treatment of the minor-seventh chord

movement, as featured in James Browns Cold Sweat.
Horn chords with alternating crescendos and diminuendos

produce a mobile and engaging dynamic effect heres how
the hairpins look in a score!
Applying short crescendos to repeated, fast staccato trumpet

notes is a great dramatic device.
Coming Up
Next month Ill look at all the elements that go into creating a
successful brass arrangement, give a few more tips about how to
get the most out of your brass samples, show you some more
chords and take a peek into the musically rich world of big-band
arrangement. Thanks for reading! .
Using keyswitches to switch between a grace note and a
sustained note. The right hand plays the melody (red) and
the left operates the keyswitches (blue). In this case, the lowC# keyswitch selects a grace-note patch, and the low C a
sustains patch, as in diagram 18.
Diagram 18: The Vienna Symphonic Library provides grace

notes for every interval up to an octave, which came in handy
when we needed to program the saxophone top line in
diagram 14. The gr2 patch name signifies that the grace
note is a two-semitone interval.
Tom Walsh (foreground) with British trumpet legend Derek

Watkins, the sound of James Bond, Tom Jones, Shirley
Bassey, Kenny Wheelers big band, James Last and more.
An extract from trumpet player Tom Walshs pop horn

arrangement, showing the dynamic forte-piano
crescendo style in action. The parts are scored for
(from top down) three trumpets, tenor sax and
trombone
Another Tom Walsh score extract, this one illustrating a

unison melodic phrase culminating in a chordal stab a
typical horn arrangement technique. The hat symbol on the
phrases last note indicates a short, slightly accented
articulation.
The Broadway Big Band sample library contains jazz and bigband horns playing an extensive range of articulations.
Amongst its other tricks, Native Instruments Session Horns

Pro provides tempo-synced forte-piano crescendos which
can be time-stretched to the length you require.
20 Assorted Brass Arrangements

Here are 20 pop, rock and funk brass arrangements dating from the 60s to the present day, with notes on their points of
musical interest. Spanning everything from simple pop-punk to arty, avant-garde atonalism, this eclectic selection shows
how the simple founding principles of 60s arrangements have evolved and diversified down the years. Certain songs are
included merely in order to illustrate a particular arranging technique, but I recommend that would-be horn arrangers give
them all a listen.
1. Strawberry Fields Forever (the Beatles) 1967 George Martins highly effective arrangement for four trumpets
(which starts at 1.10) combines simple triadic chords and urgent rhythm stabs.
2. Are You My Woman? (Tell Me So) (the Chi-Lites) 1970 the original recording of the fabulous horns riff powering
Beyoncs Crazy In Love, also features some great funky syncopated rhythm patterns at 1.51.
3. 25 or 6 to 4 (Chicago) 1970 this enigmatically titled song gains much of its character from a hooky, adventurous
brass arrangement that finally grinds to a halt with five defiantly odd horn chords.
4. Give Me Just A Little More Time (the Chairmen Of The Board) 1970 proves that you dont need more than three
notes to write a catchy horns hook.
5. Got To Get You Into My Life (Earth, Wind & Fire) 1978 a jazzy 12/8 take on the Fab Fours soul-flavoured tune,
featuring a flash, brilliantly played arrangement with some startling shake stabs.
6. Burn This Disco Out (Michael Jackson) 1979 a great arrangement by the legendary Jerry Hey, performed by the
Seawind Horns.
7. I Keep Callin (Al Jarreau) 1984 another precisely played, funky and musically hip Jerry Hey chart. A masterclass in
fitting horn parts round a vocal line.
8. Cant Get Enough (Supergroove) 1994 a supremely efficient and economical horn arrangement helps propel this
super-funky track by the engaging Kiwi band. Warning: contains humorous interjections.
9. City Of Angels (the Generators) 1997 energetic, uptempo ska pop-punk with a horns intro voiced in thirds, as
explained in Twos Company earlier.
10. Museum Of Idiots (They Might Be Giants) 2004 a powerful brass arrangement, written along classical lines and
played with great gusto, turbo-charges this touching and quirky little song.
11. Some Skunk Funk (the Brecker Brothers) 2005 this manically fast, intensely exciting commotion combines mad
funk horn riffs with virtuosic sax and trumpet soloing. Conclusive proof that if you push the boat out far enough, you get
into very deep water.
12. Who (David Byrne & St. Vincent) 2012 idiosyncratic and stylish art-pop which veers into orchestral territory by
blending saxophone funk riffs with beautifully played, warm-sounding French horn swells, courtesy of arranger Tony Finno.
13. Flick Of The Finger (Beady Eye) 2013 Liam Gallaghers erstwhile rock outfit piles on the tension with a
straightforward, steam-rollering brass arrangement featuring an epic, blasting theme, a regal high trumpet counterpoint
and a nicely insinuating trombone semitone lick.
14. My Sad Captains (Elbow) 2014 mournful trumpets, largely voiced in thirds, enhance this songs melancholy mood.
15. Scream (Funk My Life Up) (Paolo Nutini) 2014 a subtle, skillfully written brass arrangement with some deft
rhythmic touches. After a quiet start, the horns come increasingly to the fore over the course of Nutinis soulful ode to the
doubtful charms of contemporary womanhood.
16. Sue (Or In A Season of Crime) (David Bowie) 2014 intriguing, left-field avant-garde jazz-pop with atonal
tendencies, arranged by US composer/arranger/bandleader Maria Schneider. Lovely trombone chords at 3.56 and nice
use of the Harmon mute on trumpet at 4.11, adding subtle, glinting notes in the background.
17. Lingus (Snarky Puppy) 2014 a live recording by this talented New York ensemble. A long-ish instrumental piece
with a great, exuberant and catchy 5/4 horns lead line, in danger of being overshadowed by the blinding keyboard solo
which follows.
18. Roller Coaster (Dirty Loops) 2014 the darling of musos the world over, this gifted Swedish trio wisely called in
Jerry Hey (that name again) to arrange horns on their debut album Loopified. This harmonically mobile slice of fusionfunk-pop (for want of a better word) is given extra sheen by a brilliant Hey arrangement containing some frighteningly high
trumpet notes. The track also features a jaw-dropping performance by bass phenomenon Henrik Linder.
19. Got Me Runnin Round (Nickelback) (2014) tasty, carefully placed brass riffs and swells intensify the feel of this
funky offering by the Canadian rockers.
20. How Big, How Blue, How Beautiful (Florence and the Machine) 2015 a hypnotically repetitive, lyrical ballad
which steadily builds to a climax with Will Gregorys stately brass arrangement.
Instrumentation & Texture

Here are a few suggested instrumentation options for those using sampled pop horns. As mentioned in Part 1 of this
series, a pairing of trumpet and tenor sax is very common, and its a good starting point for simple harmonies like those in
diagrams 3 to 6. Trumpet takes the top line, tenor sax plays the lower part marked in red. If writing two-part harmonies in a
higher register than these examples, you could substitute alto sax for tenor sax, or use two trumpets.
The closely voiced three-note chords in diagrams 7, 11, 12 and 14 sound fine with trumpet on top and two saxes
underneath again, depending on register, you might prefer to substitute alto for tenor sax, or go for the brighter sound of
two high trumpets supported by a sax. However, the classic three-piece horn section sound of (from the top down)
trumpet, tenor sax and trombone also sounds great for mid-range chords, particularly wider voicings such as those in
diagram 8. Bear in mind that the trombone has a lower pitch range than tenor sax, so traditionally plays the lowest part.
For a regal, fanfare-like sound in the higher register (as in diagram 9), you cant beat the sound of three trumpets, and if
you want to maintain that pure, orchestral sound over a wider pitch register, add trombones for lower parts.
Different instrument combinations produce subtle variations of texture, so be prepared to experiment; there are no rules,
and as a wise man once said, the only thing that matters is whether what you write sounds good!
Arranging For Brass Part One
Arranging For Brass Part Two
Fundicin metales Piura

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A Question Of Balance
Department Of
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Vocal Point
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The lower waveform is the original vocal, while the upper one has been tweaked to even out levels.
Polish your vocal recordings in Sonar.

Craig Anderton
ocals are the most important part of a song, because they not only tell a story, they are also the primary means to
connect with your listeners emotions. And you dont even need to have a voice with operatic quality think of all the
singers who had marginal voices (Bob Dylan, anyone?) but became celebrated because they were able to project a
personality that people liked. However, you want to make sure that your voice is always being presented in the best possible
light, and that involves editing as well as mic technique, EQ and dynamics.
A Question Of Balance
Its common to add compression to vocals to even out the sound, so why does it often sound like theres something wrong
after compression has been applied? Simple: you want a consistent volume going into the compressor, otherwise if you set
the compression to even out the lower-level signals, it will over-compress the higher-level ones, which interferes with
dynamics by robbing the peaks of their power. When the incoming level is more consistent, you can use less compression
so, paradoxically, an incoming signal with less dynamic range can sometimes result in a signal coming out of the compressor
that sounds like it has more dynamics.
When its time to mix, I solo the vocal and listen to it phrase by phrase. Sections where individual words or phrases are
lower in volume, and not intentionally so, can be either normalised (to save time Ive set up a keyboard shortcut for this
Process / Apply Effect / Normalize) or have their gains raised. This can also work the other way around, by reducing a word if
it stands out too much. Note that if you select a section to boost thats after any breath intake sounds, then those will sound
softer in comparison to the boosted section.
While youre in editing mode, its a good time to do some other tweaks. Its worth cutting the silent sections where singing
isnt happening, and fading into and out of the wanted vocal recording. Note that the Remove Silence DSP process has a bug
where the attack and decay times are always 0, so you cant use it to add fade-ins or fade-outs. However, if you select
multiple clips and apply a fade to one of them, Sonar will add a short fade to all of them simultaneously. Where youre fading
over a p or b sound, longer fades will reduce popping.
If a doubled phrase doesnt end at the same time (eg. one held note lasts longer than the other one), split just the word with
the held part and use the timing tool (shortcut: Ctrl-click and drag the clip edge) to stretch or compress just that one section.
You may need to crossfade the beginning of the stretched clip with the end of the previous clip, or add short fades, so that
theres no click at the transition.
Listen for mouth clicks. If theyre short enough, you can often cut these out, then slide the two sections on either side of the
click together for a crossfade.
Also, for some reason, Sonar seems very tolerant of selecting snippets of audio within words. For example, part of the word
might poke out while the remainder is lower; if youre careful with your selection, you wont hear any clicks or pops due to the
gain change (although you should always audition the edit to make sure).
Finally, even though compression is the default choice for vocals, a good limiter can sometimes give a more natural sound.
When I need compression the CA-2A seems to flatter my voice the most, while for limiting, the Concrete Limiter does a great
job.
Department Of Corrections
Many otherwise rational people think pitch-correction has taken all the soul out of vocals. Thats not correct: people who dont
know how to use pitch-correction take the soul out of vocals. Actually, pitch-correction has let me put more emotion into my
vocals, because Im not judging myself about the pitch as Im singing. I just sing and dont worry about it, knowing that if
theres an errant note or two, it can be fixed. Used judiciously, pitch-correction can encourage you to sing with greater
abandon and often, thats what a vocal really needs.
Escuela en Lima
Formacin integral de
excelencia.
Profesionales
expertos.Contctenos!
However, I never select a clip and do wholesale pitch-correction. Listen to your vocal carefully; if you hear something that
actually sounds wrong, isolate that phrase and open it in Melodyne. The Melodyne Essential version included with Sonar can
do basic pitch-correction, but as many Sonar users have discovered, the upgrade to Melodyne Studio gives much greater
flexibility.
For example, suppose the vocal runs out of steam towards
the end of a note, where the vibrato gets shaky and the pitch
drops or rises. If you select that note, Melodyne will base its
correction on the average pitch. Splitting the clip at the vibratos
zero crossings changes the average pitch for each segment, so
Melodyne detects the pitch accordingly. Now if you correct pitch,
each segment will fall into line (if not, one segment may go sharp
or flat a semitone, in which case you should move it into line with
the others). Furthermore, if you have Melodyne Studio or above,
you can use the Pitch Modulation tool as needed on each
segment to make the vibrato more consistent, or reduce (or
increase) the vibrato amount.
Tightening Time
DAW Techniques
VocalSync is rapidly becoming one of my favourite new Sonar

features because of how it can tighten up vocals. It does this by
designating one vocal as the guide track, after which you open
The top blob is the original held note, and the middle one has
been split where the vibrato got nasty. Note how Melodyne
up the dub track as a Region FX (like you do with Melodyne).
has detected that the average pitch of each segment has
Turning the VocalSync dial causes the dub waveform to move
changed. The lower blob has been pitch-corrected and also
and stretch so that it matches up with the guide tracks
has reduced vibrato.
waveform. However, to get the best results from VocalSync, its
crucial to understand that the VocalSync dial is not calibrated, and for a reason. This is because turning it up doesnt
necessarily tighten vocals more; instead, theres a sweet spot along the knobs travel where vocals line up best. This can
happen anywhere from being slightly turned up to being turned up all the way.
While youre adjusting VocalSync, the audio is in preview mode. This lets you know how the vocals are syncing up, but the
audio quality isnt that great you need to render it, just as you do with transposition or time-stretching. Of course, you can
always undo it if it doesnt sound as expected.
The longer the clip you want to sync, the harder the algorithm has to work. Splitting a longer phrase into smaller pieces will
often give optimum results.
For the best visual feedback, zoom in and increase the vertical track height. This makes it easier to line up the waveforms
visually, which helps when VocalSync is in Preview mode.
Also note that you can choose different algorithms for the
online preview and the offline rendering. The default algorithm for
rendering is (not surprisingly) Radius Solo Vocal, but for some
voices other algorithms may work better (if youre James Earl
Jones, try Radius Solo Bass). To change algorithms, with the clip
selected and the VocalSync Region FX window open, choose
the Clip tab in the Inspector, open the AudioSnap section, then
choose the desired algorithms for online and offline Render.
When All Else Fails...
The blue is the guide track, and the white, the dub track
(these have been coloured for clarity). The lower waveform
shows the uncorrected version. Note how the dub-track note
outlined in pink ends too soon, the note outlined in orange is
too long, and the notes outlined in red have transients that
are way off compared to the guide track. The upper
waveform shows the post-VocalSync processing.
As you do your vocal editing, you may run into sections that
need to be redone. Although the comping method introduced in
Sonar X3 has sort of taken over as the preferred way to fix these kinds of issues, dont forget about the merits of punching.
Simply drag over the region where you want to punch, then enable the Auto-Punch Toggle button. To remove the area over
which you want to punch, hit Ctrl-X to get rid of it, or click the regions handle and then drag the region off the track to remove
it. If you do a lot of punching, consider inserting a temporary blank track and shift-dragging the selected region to that track, in
case you change your mind and want to revert to the original region.
However, remember that having the Auto-Punch Toggle button enabled has tripped up many a Sonar user who couldnt
figure out why the program wouldnt go into record when trying to record outside of the punch zone.
.

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Notes From The Deadline

TV Music From The Inside
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If you could just split it out into individual words, then I can finetune the sentences as I read...
Its always vital to give the client what they want. Even if what they want makes no
sense.
Paul Farrer
i Paul! Greetings from a dark edit suite in sunny LA! Thanks for the last track you sent over. Amazing. We all love it. It
really captures the emotional emptiness of the scene and reflects beautifully the desolation and hopeless feeling of the
script at that point. I like to work with submixes and stems so I can balance the levels of everything here in the dub so
it doesnt crowd out the dialogue too much. Can you send them over for this piece as well? Many thanks.
Hi Chad. Glad you like it and that its working well. Im not sure what you mean when you ask for submixes. Theres not a lot
going on in this track to mix out. What do you suggest?
Hi Paul. Sorry for not being clear enough. Im known as a bit of a control freak round these parts. I have a background in
music production so for the last 15 years as a programme maker whenever Ive used music I always have to have (along with
the full mix) all the individual track elements as separate files. Then I can play with the mix as we go along and finetune it to
the other sound elements to make sure it all beds in perfectly. I drive the guys nuts here working this way, but it gets the best
results.
Hi Chad. I get that, but theres not much I can give you for this track. Its kind of just a complete recording as it is. And I cant
think of any other way to present it to you. I could send it to you without any reverb, I suppose. Would that help?
Hi Paul. No, I need the submixes of it. I always have to have the submixes. Everyone else has always been able to provide
these, and if we cant mix the individual elements Im afraid we wont be able to use your music in this project at this time.
Hi Chad. Its a solo piano. Would you like me to add some more elements to it, in case you need to make it sound fuller? I
can certainly add some strings. That would give you some options.
Hi Paul. No please dont add anything. I feel youre still not hearing me about needing the submixes of it. I want to be able
balance all the elements at the point of mixing, and at the moment, its just too inflexible for what we need.
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Hi Chad. Would you like me to record a different version of the same thing but with fewer notes in? I could certainly make an
emptier arrangement with more space.
Hi Paul. No. Please dont change any of the music. Its perfect. The notes are exactly where they need to be. Im just
worried about not having any mix options to play with.
Hi Chad. Attached to this email is a link to an online folder with 5GB of WAV files in. Ive recreated the piece in my DAW
software programme using a sampled piano and have mixed each individual note in the place it occurs in the song as a WAV
file. Ive also given you a separate recording of the reverb of each note. Each of these recordings is presented as a stereo file,
and left and right channels individually. Youll see there are 16,480 WAV files in the folder (at both 44.1 and 48 kHz), and if you
start them all at position 00:12:04:08:13, you should be able to pick which notes you want to favour and which ones you dont
like.
Hope that helps.
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Off The Record

Music & Recording Industry News
Hollywood looks like it may be on shaky ground, but its not all bad news for audio
post houses.
Dan Daley
ast year, according to US Bureau of Labor Statistics data, employment in both motion picture and sound industries (the
BLSs rubric) dropped eight percent from 2013 and 19 percent from 2012. This trend was graphically illustrated by the
distress sales, completed earlier this year, of the assets of several iconic audio post facilities, most notably ToddAO,
which ultimately joined Soundelux and POP Sound in liquidation, as well as Sound One in New York City.
There are a few reasons for this seemingly sudden decline.
For starters, the cinema business here made less money last
year, taking in about $10.3 billion in 2014, down 5.2 percent from
the prior year. There were more misses than hits, especially in
the crucial blockbuster category. Now 2015 is shaping up to look
even worse industry tracker Box Office Mojo is showing the
business being off by as much as 25 percent from last year.
Of course, thats just the first quarter of the year, and
historically January and February are Hollywoods leanest
months. People are already talking about flicks like Avengers:
Age of Ultron and the next iteration of the Star Wars franchise,
due out this year, as having the potential to set new boxoffice
records, and both will feature the kind of overthetop
multichannel sound that these kinds of films are famous for. The
problem is that, with profit increasingly staked on a smaller
number of bigbudget releases, Hollywoods volatility index is
going way up. And thats ultimately going to have an effect on
people who mix cinema sound, record production sound,
fabricate Foley and do ADR.
Dan Daley
Blockbustin Beats
Sound familiar? Its the same thing that happened to commercial recording studios 15 years ago. As record labels
consolidated, they also strategically pursued their own version of blockbuster artists, the Katy Perrys of the entertainment
world, so to speak. Not to denigrate Ms Perry specifically, but she is the avatar of a glossily produced, carefully crafted
generation of music products, one that has benefited (and benefited from) a relatively small number of producers, engineers
and studios. The purchase orders that were once the bedrock of commercial studios have thinned out considerably, and lets
face it, some of whats left goes to artists and producers own personal studios, narrowing the revenue stream even further.
Thats whats taking place in audio post now. Film revenues are down and the focus on a small number of bigbudget
productions tends to cluster spending around a smaller number of facilities. And a good share of those are owned by the film
studios themselves, most notably Disney, Fox and Sony, all of which maintain large sound stages in LA and elsewhere. Its put
enormous pressure on the independent commercial post facility sector.
Not that pressure wasnt there before. Twentyfive years ago, Hollywood was sending its scoring work to Vancouver, in
Canada, which had initiated a fairly aggressive campaign to win those jobs based on the currency exchange rate and on
reduced or nonexistent backend royalties. And within a few years of that, national orchestras in places like Prague and
Warsaw were getting in on that same act, offering even lower costs for scores.
In the last two decades, several US states have begun to invest heavily in post production, as part of a larger strategy to
attract more film and TV production. Its the productions themselves that bring in the greatest amounts of money, and that
revenue is scattered around the local economy in the form of hotel stays, restaurants, shortterm employment, transportation
and so on. But more and more of the allure is the states investment in technical infrastructure, to keep the production there
longer. That includes post production Georgia, North Carolina and Louisiana have all subsidised the construction of post
facilities in their states.
Most recently, the challenge is coming from even further away. Chinas been on a tear lately, with new film facilities opening
up and existing ones expanding. The largest is scheduled to open in 2017 Qingdao Oriental Movie Metropolis will cost an
estimated $8.2 billion and offer 20 sound stages. And like much of Chinas media-production infrastructure, it will be available
at low rates compared to US facilities. The studios parent company, the Dalian Wanda Group, is serious: it bought Americas
secondlargest cinema chain, AMC Entertainment Holdings, in a $2.6 billion deal in 2013. When a company make 11digit
investments in production and distribution on a global scale, the cost of also chucking in audio post is barely a line item.
On The Box
The good news is that theres more content being made than ever before films are struggling but when a blockbuster
franchise does hit, like Avengers, the success can help seed smaller productions and the use of a larger number of facilities.
TV is having a banner decade, with zombies, vampires and sociopathic advertising executives driving a creative renaissance
in the medium, most of which is going out in 5.1 surround. (And the nuance of this audio has been in some cases sublime
listen, with a subwoofer, to the constant background noise on Mad Men to hear what New York actually sounds like all the
time.) The small screen has been a boon to smaller post studios, especially programmes from some of the smaller
companies.
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Finally, streaming is the new frontier, with productions funded by Silicon Valley giants like Google and Facebook, assuring a
growing stream of content that will require proper audio to compete with broadcast. All of this new work is subject to the laws
of digital, of course you can do sound design on a laptop just as you can a pop song, so facilities costs and revenues will
be under pressure. But as with music, it also puts a premium on individual expertise: those with the chops to deliver the best
creative work, and do it cost effectively, will be the winners. .
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Twisting By The Pool
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The Pool, London

Studio File
Hannes Bieger
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The Pool was originally conceived as a one-room studio, and can be restored to such a state in a couple of hours.
n over 30 years in business, Miloco has developed from a small independent studio to a venture that is operating
worldwide. Today, dozens of recording facilities belong to the Miloco Studios group, which also specialises in studio builds
and equipment sales. Miloco rooms are spread out all over the globe, from Bali via Iceland and Turkey to the USA, and
almost 30 of them can be found in London. In the case of The Pool, however, the recording facilities form an integral part of
the companys headquarters in the South London disctrict of Bermondsey. The space is shared by the Miloco offices, as well
as a number of other production spaces including their flagship mixing studio The Engine Room, as well as the
writing/production studios The Bridge and The Bunker.
Miloco acquired the building in 2000, and The Pool opened in 2006. Though less than 10 years old, the studio has already
undergone a remarkable metamorphosis as well as building an impressive track record, and today it offers a versatile, flexible
and unique layout along with a vast collection of esoteric equipment.
Wide Open Space

Built to house a Sunday school a long time ago, the building afterwards became a warehouse, and was redeveloped into a
recording studio in 1986. Built by Tom Astor and named Orinoco, the room that would become The Pool was designed to be
used as a TV studio or sound stage. When Miloco moved in they planned to build a recording studio on the premises, but in
light of diminishing budgets in the record industry, the idea was put on hold and the room was used for equipment storage until
producer and songwriter Ben Hillier came in and fell in love with the space. Subsequently The Pool was launched as a joint
venture between Hillier and Miloco.
In the early days the studio was configured with an open-plan
layout, with the control-room gear integrated within the live area.
As Hillier puts it, this sort of layout means that communication is
fantastic, and everyones in touch with whats happening.
Recording a full band we all work on headphones, we hear what
the band hear so its very easy to notice anything that might be
putting them off their performance.
However, despite the benefits of such an unusual approach,
some potential clients preferred a more conventional layout, so
the studio was remodelled to its current configuration a couple of
years later. It now provides a 100-square-metre live area and a
separate 30-square-metre control room, as well as a dedicated
drum booth which can be converted acoustically from a fairly dryWhen the studio was remodelled, a separate control room
sounding space to an echo chamber by removing the modular
was added.
acoustic panels. However, the original open-plan architecture
can still be restored within a couple of hours, and the unusually
small footprint of the console Hillier chose as the centrepiece for The Pool certainly helps. Hillier acquired the EMT A100
console with 30 channels and eight buses from a studio in Berlin, not only for its preamps but for the summing, which he
describes as really amazing.
Twisting By The Pool

After the remodelling The Pool wasnt exactly lacking gear, but received a significant upgrade nonetheless when Swedish
musician and producer Johan Ekelund brought in his vast collection of vintage kit, including a rack with nine Neve 1079 and
1272 modules, as well as numerous dynamics and effects units.
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Today, The Pool boasts a rich outboard collection which

includes both classic studio standards and extremely rare items.
In addition to the EMT consoles preamps and the Neve
modules, there is an eight-channel Neve 542 desk, a pair of
Telefunken/TAB V76 preamps, almost a dozen vintage API
modules including the 550, 550A and 560 types, as well as
original Pultec MEQ5 and HLF3C units as well as a pair of Lang
PEQ2A Program Equalizers. The dynamics department is
equally well-equipped, offering not only the ubiquitous Urei units
as silverface and blackface 1176LN limiters and an original
LA3A, but also a super-rare Universal Audio 175B. Other
dynamics processors include a pair of Neve 2254/A modules, the
infamous Gates Level Devil and a pair of vintage API 525s, as
well as three Tweed Audio CL604/1 modules.
The Pools large mic locker doesnt only hold familiar
Neumann classics such as the U47 FET and U87, and the
inevitable AKG and Shure standards, but also more unusual
models from Heil, Lomo, Earthworks, Bruel & Kjaer and Reslo.
And the instrument collection is just as strong: alongside a vast
number of vintage Premier, Slingerland and Ludwig drums and
an endless list of guitar amps and pedals, the synth collection
stands out, featuring gems such as the EMS VCS3, Korg MS20
and SQ10 units, Roland TR808 and Sequential Circuits Pro One.
One of the studios most distinctive features is the EMT A100

mixing console.
Despite having been in business less than a decade, The Pool

has attracted many notable artists, from early clients such as the
Maccabees and the Horrors to the Arctic Monkeys, Depeche
Mode, Franz Ferdinand, Graham Coxon, LCD Soundsystem and
Mumford & Sons. Recent clients include Florence & the
Machine, Plan B, Tom Odell, Bastille and Tvvins. One questions
remains: why the name? Well, with the control room located a
few steps above the live area, and the live-room floor painted
blue, the space looks like a swimming pool...
.

June 2015
May 2015
April 2015
March 2015
February 2015
The Pools outboard setup includes rare equalisers such as a

pair of Lang PEQ2s, an NTI EQ3 and an Altec 9062A.
Classic and more unusual dynamics units: from top, Altec

436C, Universal Audio LA3A, CRL Systems APP and three
Tweed Audio CL604/1 modules.
The infamous Gates M5546 Level Devil compressor, not a

favourite unit of radio stations in the Bible Belt in the 1950s...
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Q Can you explain Reapers Playback Resample

mode?
Sound Advice : Recording
Hugh Robjohns

Setting Cockos Reapers playback resample mode.
I cant find good explanations about the Playback Resample Mode options available in Cockos Reaper, specifically the
difference between the Good (64pt Sinc) and Better (192pt Sinc Slow) modes. Also, the default track mixing bit depth is
64bit float. I guess thats because most CPUs are now 64bit engines, but how does this setting relate to my outboard
converters, which all work at 24-bit? Michael, via email
SOS Technical Editor Hugh Robjohns replies: The Playback Resample Mode options relate to the application of automatic
samplerate conversion when importing an audio file with a different sample rate from that of the current project, for example
when importing a 24bit, 96kHz source file into a 44.1kHz project. When samplerate conversion is applied to a digital signal,
the original audio waveform effectively has to be reconstructed from the existing samples, so that the correct amplitudes at
each of the (new) required sample points can be calculated.
Sinc refers to a mathematical function which is intrinsically involved in reconstructing the original audio waveform from
individual samples. In very simple terms, it describes how each sample contributes to the amplitude of the audio waveform
between the sample points, both before and after each individual sample. There isnt really space to get into the mathematics
of the sampling theorem here, but if you want to know more I recommend Dan Lavrys Sampling Theory white paper
(http://lavryengineering.com/pdfs/lavrysamplingtheory.pdf).
&
The important point to note is that the Sinc function looks like an impulse with decaying ripples, which extend, in theory,
forever, both before and after each sample, but always with zero amplitude at each sample point. Consequently, these ripples
influence the amplitude of the entire reconstructed waveform and need to be taken into account when performing samplerate
conversion.
Calculating the Sinc contributions of every sample for every other sample is not practical in most cases, and so Reapers
samplerate conversion process can be optimised for varying levels of accuracy and speed. Performing the calculations for 64
sample points either side of the current sample gives good results, but extending that out to 192 sample points either side is
more accurate (it achieves lower noise and distortion). However, it takes much longer because it involves significantly more
computation.
Moving on to the internal 64bit float mix engine, this is about maximising the internal dynamic-range capability to
accommodate very loud or very quiet signals without degrading them. When a lot of signals are combined, the result is usually
much louder than any individual source, so the DAW engine needs additional headroom to cope. In a similar way, changing
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the level of digital signals often results in remainders in the calculation. Additional bits are needed to keep track of these
remainders, to avoid degrading the signal while processing.
This additional dynamic-range requirement is achieved in different ways in different systems, and depends on the type of
processing being applied. Youll often see references to double or tripleprecision, for example (where the calculations are
done with 48 or 72bit resolution), and most early DAWs used 32bit floatingpoint maths, which gives a notional internal
dynamic range of something like 1500dB. Modern computer hardware is designed to work in a 64bit operating system
environment, and a lot of DAW software has followed suit for practical convenience. It just means an even greater internal
dynamic-range capability, which makes internal clipping all but impossible and the noise floor of processing distortion
impossibly small.
However, audio signals always have to be auditioned in the human world, and our ears and replay equipment cant
accommodate a dynamic range of more than about 120dB. A 24bit system can (in theory) accommodate a dynamic range of
about 140dB, so the industry has standardised on 24bit interfaces, which are more than sufficient. The implication is that we
have to manage the (potentially) huge dynamic range signals created inside a DAW to make sure that they fit into the 24bit
dynamic range for realworld auditioning. Thats why DAWs have output meters that show clipping if the internal level is too
high; its not the 64bit floating-point signal in the computer thats clipping, but the external 24bit converter. .
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Q How can I export a Pro Tools project for mixing in

Reaper?
Sound Advice : Mixing
Matt Houghton

Moving imported WAV files to their timestamped position in Cockos Reaper.
My band tracked some songs at a local studio using their Pro Tools rig and I want to mix the tracks at home in Reaper. Whats
the best way of getting the session data out of Pro Tools and into my Reaper system?
Dave Matthews via email
SOS Reviews Editor Matt Houghton replies: I feel your pain: we often go through this sort of thing with Mix Rescue projects!
The best method really depends on the nature of the project. If its just a plain multitrack recording session then it should be a
simple case of importing all the audio files to different tracks so that they all start at bar 1, beat 1. Youll see at a glance which
were from the same take as theyll be the same length. Hopefully, the engineer labelled the tracks/files so you can see which
sounds are which.
If youve done edits or punchins, that adds a layer of complexity, but is still fairly simple. First, ask the studio to
consolidate all the clips in Pro Tools, to create files that line up as described above. If thats not an option (or the studio
charges you too much for the privilege!) note that Pro Tools automatically timestamps the WAV files it records and Reaper
can read those time stamps. Drag and drop your files into Reaper, select all, right-click on a clip to bring up the context menu,
and select Item Processing/Move Items To Source Preferred Position (BWF). All files should line up in the same positions as
they were recorded in Pro Tools. Be aware, though, that some DAWs have a default timecode offset (for reasons relating to
audiotopicture applications that you neednt understand for this task). So you might find, as I once did, that all your files
start as if the bar 1 beat 1 position is at +1 hour on Reapers timeline. To remedy this, zoom right out to find the files, select all,
and drag all files to your preferred starting point
If the Pro Tools session includes more information, such as pan settings, clip gain, volume automation and so on, youll
need either to bite the bullet and redo all that work yourself, or use another method to transfer the files. Some of that
information can be transferred as OMF/AAF files, but (a) Ive found that these are unreliable and inconsistently implemented
by different DAWs and (b) Reaper doesnt support them! The best bet in this situation is to invest in Suite Spot Studios AA
Translator software, which reads and writes more project data in a wider range of formats than any other software I know of. I
have a copy here under review and have been most impressed so far. Its Windows only, but with the manufacturers help I
have it running in Winebottler on Mac OS 10.9.5.
The trickiest thing is transferring plug-in effect or instrument data from one DAW to another. Your best bet is to print those
effects as audio. But you should also be able to get the studio to supply the MIDI data from the session to enable you to
rebuild any instrument parts. .
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Q How do I record from my guitar amps headphone

out?
Hugh Robjohns

These are all jack-to-jack cables, but theyre not wired the same. To record from a headphone output to an audio interfaces
input, youll usually need the bottom one, often known as a Ycord or insert cable.
Im seeking some guidance on why Im unable to record from my guitar amp straight to my audio interface. Im working with
two small amps, a Marshall and a Crate, and trying to record a feed from the guitar amps headphone outputs via a Focusrite
Scarlett 2i2 interface. Originally, I thought that the headphone jacks on both amps might be broken, but when I plugged in a
pair of Sony headphones, the sound came through perfectly clearly. Then I thought the issue might be that the headphone
had a quarterinch TRS connector instead of TS, so I just picked one of those up and tried to hook the amp up to the interface
that way. No dice.
Im not getting total silence, though if I turn the amp up loud enough I get very crackly audio that sounds like it has a
highpass filter on it. Your guidance would be greatly appreciated!
SOS Forum post
SOS Technical Editor Hugh Robjohns replies: Although headphone amps arent technically the best output source for a
recording, you should still capture something reasonable if using the correct cables. But the issue here is that the apparently
similar quarterinch sockets in the amp and interface are wired very differently, and therefore carry/expect differently formatted
signals.
The headphone output is unbalanced and is wired to be compatible with stereo headphones, even though the guitar amp
produces a mono signal. That means that the unbalanced amp output signal is wired to both the tip and ring contacts in the
headphone socket, with a common ground on the sleeve. The interface input expects a balanced linelevel signal, which
means that it only responds to the difference between the signals on the tip and ring contacts.
With a TRSTRS cable connecting the amp headphone output to the interface balanced line input, the signals on the
interface tip and ring contracts are identical; there is no difference, and so there will appear to be no signal. All youll hear, as
you describe, is the very small error signal resulting from an imperfectly balanced input amplifier, which is usually a very quiet
hissy, spitty, toppy sound (see Figure 1).
With a TSTS (instrument) cable connecting the amps headphone output to the interface balanced line input, there is no
ring contact, and so the ring output terminal in the headphone socket is shorted directly to ground by the plug sleeve. Inside
the amplifier, I suspect the headphone-socket tip and ring contacts are actually wired directly together (rather than having a
true stereo output amplifier) since there is only a mono source. This means that the act of inserting a mono TS plug will
actually short the entire headphone amp output to ground, leaving no signal to output to the interface at all! (See Figure 2.)
The only workable solution, if you want to record from the headphone output, is to use a Ycord, which comprises a TRS
plug at the amp end, and two TS plugs for the interface end. Its often sold as an insert breakout cable or a stereotodual
mono output splitter cable. When using this kind of breakout cable, the TRS plug provides the headphone socket tip signal on
the tip of one TS plug, and the ring signal on the tip of the other TS plug. If you then plug one (or both) of these TS plugs into
the balanced line input(s) of your interface, the balanced input circuitry looks for the difference between the signals on the tip
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and ring again, but this time the TS sleeve shorts the ring signal to ground, and the wanted headphone output signal is applied
between the tip and ground, so all is well. (See Figure 3.) Hey presto! It will all work as you want and expect... .
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Q Whats the best way to clip my drums?

Matt Houghton

The Clipper section of Vladislav Goncharovs excellent freeware Limiter No6 can be a really useful tool for certain drum
sounds, even if thats not what it was created for. Just be sure to turn off any other sections you dont need for example, the
limiter can soften the edge of drum transients before the sound hits the clipping stage.
Ive read that I can get a good hiphop kick or snare sound by clipping it. But when I try this, it sounds horrible. What am I
doing wrong?
Jake Johnson, via email
SOS Reviews Editor Matt Houghton replies: Lets be clear what we mean by clipping. Digital clipping, whereby the part of the
waveform that exceeds the digital headroom is flattened, is difficult to achieve in your DAW by accident because theres bags
of headroom (as Hugh makes clear in his previous reply) but its possible if, for example, gain is applied to a signal before it
hits an old, poorly designed plug-in. The reason this form of clipping sounds so bad is that the clipped section of the waveform
is essentially a series of square waves, with strong, odd-order harmonics extending up beyond half the sample rate. These
high harmonics arent supposed to exist in the digital domain, so they cause aliasing distortions at the D-A converter.
The nice form of clipping youre referring to was originally done by abusing the analogue stages of an A-D converter. When
you clip in the analogue domain, the artifacts of clipping are all harmonics at frequencies higher than the fundamental: clipping
a 100Hz sine wave would generate odd harmonics at 300Hz, 500Hz and so on. Any harmonics above half the sample rate
were filtered out by the converters anti-alias filters, so there would be no aliasing. To work in this way, you must run out of
analogue headroom before you run out of quantisation levels (digital headroom). Some converters are designed like this and
others arent, but quite a few (including on many audio interfaces) offer a soft-clip facility, to ensure the device behaves in the
desired way when clipping.
Today, you can achieve the same effect by recording your sound without clipping and then using a dedicated clipper plugin
such as the freeware Limiter No6 by VladG. (There are lots of other processors in this plugin, and the key to success is
disabling the sections you dont need.) Such plug-ins usually include anti-alias filtering, to give an analogue-like effect.
Of course, not all sounds will benefit from being clipped, particularly those with a strong pitched element. Snare drums, hihats and cymbals are usually better candidates, as they all feature a strong noise component and little pitch information,
particularly during the attack phase of each hit the brief transient peak, which is really the only bit youre looking to clip. Its
important to understand that the effect wont work for every track, either, as clipping tends to give a sound more bite, which is
not always what a tracks going to require. As with any creative processing t hat changes the tonality of a sound, you must use
your ears.
.

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