Sound On Sound 2015-01

Fri 19 Dec 2014
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January 2015 In This Issue
January
2015
News
Go
To open articles, click title links

Competition
>> Access ALL ISSUES
People + Opinion
Win PreSonus Sceptre S8 monitors
David Wrench
Deadline: 2015-02-09
Mixing Caribou, FKA Twigs & Jungle
Fantastic chance to Win PreSonus Sceptre S8

monitors in January's Issue Competition.
Reviews
Angelbird SSD2go Pocket

USB 3 Solid State Drive
Mix engineer David Wrench has the rare ability to

bring discipline to experimental projects, without
losing their leftfield qualities.
David Wrench | Media

Audio files to accompany the article.
Inside Track: Tom Lord-Alge
If you want more storage, want it faster and want it Secrets Of The Mix Engineers
smaller, Angelbirds portable SSDs might be just the
For Rivers Cuomo and Weezer, going back to
thing.
basics meant a reunion with one of the biggest
names in rock mixing: Tom Lord-Alge.
Appworks
Making Music On The Move
We take a look at how the iPad Air 2 deals with

music software.
Sounding Off
What is the future of mastering?
Russ Hepworh-Sawyer asks, 'will auto-mastering

eventually produce a result thats good enough to
satisfy the would-be clients of the future?'
Arturia Beatstep
MIDI Controller & Step Sequencer
The Thrill Of The New
Arturias Beatstep combines a MIDI controller and a Leader

step sequencer in one extremely reasonably priced
box.
I could swear that we now have at least three

Christmases a year!
Audiowiesel Hammered Acoustic Guitar

Kontakt Instrument
Audiowiesel are a small fledgling company run by

Austrian composers Frank Herrlinger and Daniel
Szwedek.
Chandler Limited TG2500
Te c h n i q u e
Automation Action
Reaper Tips & Techniques
Get to grips with Reapers range of automation

facilities.
Solidstate Microphone Preamplifier
Can Chandler pack all of the EMI/Abbey Road

sound for which theyre so famous into a singlemodule 500series preamp?
Dave Smith Instruments Pro 2

Monophonic Synthesizer
Deep Space
Live Surround-sound Performance
For this ambitious project, John Crossley had a full

live band play through a 16-speaker system, to
create an immersive performance.
The depth and complexity of the Pro 2 could make it

Freaky Filter
Dave Smiths most sophisticated monosynth yet.
Studio One Tips & Techniques
DoReMIR ScoreCloud
If youre in search of novel effects in Studio One, fire up the

Autofilter plug-in!
Music Transcription Software For Mac OS & Windows
DoReMIRs unique program helps to bridge the gap

between performance and musical score and its
free!
Freaky Filter | Media

Dynacord PM502
Powered Mixer
Dynacord apply their typically thoughtful design to a Improve Your Groove

Reason Tips & Techniques
compact desk, which could be just the ticket for
We look at ReGroove and the new quantise
small gigs and installations.
features in Reason 8.
Focal Alpha 65
Active Nearfield Monitors
Focal speakers have long been coveted for their

performance, but have traditionally come with
professional price tags attached until now...
Media Content - January

Audio files to accompany the relevant articles.
This page contains a directory of all media pages related to

the January 2015 articles.
Mozaic Beats AutoTheory
Mix Rescue
Intelligent MIDI Chord & Melody Tool For Mac OS & Windows
Our Experts Tranform Your Tracks
If you want to create convincing keyboard parts but

dont have the playing chops, AutoTheory might
save your bacon.
Max Farrar: we take on a wellcrafted

contemporary pop tune with a massive number of
tracks, and create two different mixes.
Munro Sonic Egg 150
Mix Rescue | Media
Active Monitors
Formerly sold under the SE brand, these Andy

Munrodesigned Egg speakers have been given a
major update. So whats new?
Hi-end
Analog
Mastering
Sontec, Fairman,
Massenburg, Weiss
Mastering Engineer
Misjah
Win PreSonus Sceptre

S8 monitors
Win Munro Sonic Egg
150 (Global Except
Americas)
Win Yamaha White
Christmas bundle
(Americas Only)
Sonar Tips & Techniques
Become a Sonar ninja with these time-saving tips.
Native Instruments Action Strikes

Kontakt Instrument
Like its predecessor Action Strings, the Action

Strikes percussion library is a collaboration between Normal Service
Pro Tools Tips & Techniques
software giants Native Instruments and the
Pro Tools acting funny? Chances are youve just
production company Sonuscore.
changed playback mode by accident.
NI Komplete Kontrol S49

Controller Keyboard
NI turn their hybrid expertise to keyboard design

with an innovative controller for Komplete and
Maschine.
WIN Great Prizes in SOS

Competitions!
Need For Speed
Ocean Audio Signature Three
Perfect Harmony
Logic Tips & Techniques
Learn how to create vocal harmony parts and

doubling effects using Flex Pitch and Logics
vocoder plugin.
Dual-channel Equaliser
The Signature Three offers something no other EQ

does four inductor-based bands per channel.
Perfect Harmony | Media

Session Notes
Panda Audio MidiBeam
Wireless MIDI Interface
Panda Audios MidiBeam wireless MIDI interface

does pretty much what it says on the tin.
The Practical Craft Of Recording
Brushes Held Like Hammers: Our recording

engineer looks at ways to reduce spill without
compromising a fullband tracking session.
Session Notes | Media

Playback
Readers' music reviewed
This month we listen to Easychair, Man Bites

Zombie, Lukas Creswell-Rost and Lunacybot.
Strip It Good
Cubase Tips & Techniques
Set up your own hardwarecontrolled channel strip in

Cubase.
Ploytec PL2
MIDI Powered Duophonic Synthesizer
Can this tiny box deliver a big sound?
Strung Out
Ableton Live Tips & Techniques
Plug-in Folder
This month XILS Lab 'XILS4' and Digital Brain's
'Multiplex Vocoder'.
We get plucky with Ableton Lives Tension

instrument.
Studio SOS
Positive Grid BIAS
Guitar Amp Simulator Plug-in For Mac OS & Windows
With component-level adjustment and amp

matching features, BIAS can target almost any
guitar amp sound real or imagined!
Paul and Hugh head off to The Cave!
Team SOS travel to deepest, darkest Cornwall,

where they encounter some troublesome low
frequencies and tackle a few common mix
quandaries.
The Mix Review

Prodipe Pro 8-3W
Three-way Active Monitors
Three-way speakers tend to reside at the higherpriced end of the market can Prodipe succeed
with a budget-friendly option?
Commercial Productions Analysed
This month's tracks from Clean Bandit, Oliver

Heldens, Meghan Trainor, Barbra Streisand and
our Classic Mix Bill Haley & His Comets Rock
Around The Clock.
Music Business
Sample Shop | Media

Samplecraze Low End

eBook Review
This extensive and in-depth eBook explains so

much more than how to process your kicks and
bass sounds!
Notes From The Deadline

TV Music From The Inside
Its hard to make something new. Its even harder

to make something thats both new and old.
Off The Record

Music & Recording Industry News
Small Signal Audio Design 2nd Edition

Book Review
Despite a recent downturn, the videogame

industry still has much to offer to audio pros.
Douglas Selfs excellent technical reference book for

solid-state audio-gear designers has been updated.
Studio File
RMV Studio, Stockholm
Sonokinetic The Carnival

Kontakt Instrument
A couple of years ago Sonokinetic released

Carousel, a rather splendid Kontakthosted
fairground organ. The Carnival is very much a
sister instrument to Carousel and presents two
smaller-scale barrel organs.
SSL XLDesk
Swedens newest high-end studio is built on very

strong foundations, both literally and
metaphorically.
Sound Advice
Q Are mineral-wool acoustic panels safe?

Im looking to get some acoustic panels of the rigid
fibreglass/Rockwool type for my bedroom studio.
24channel Analogue Mixer
The XL-Desk is the first Solid State Logic mixer that

can host hardware processors in the ubiquitous 500- Q Do surround-sound speakers need to be of

Competitions!
S8 monitors
Win Munro Sonic Egg
150 (Global Except
Americas)
Win Yamaha White
Christmas bundle
(Americas Only)
series format.
the same type?

Im a sound engineer planning to have a surroundsound setup. The problem is, I already have a pair
of Yamaha MSP5s so, can I have Yamaha HS5s
for the centre and rear speakers.
Studiomaster Starlight 12ML

PA, Monitor & Lighting System
Could this crafty three-in-one PA and lighting system

save you both time and money?
TC Electronic PolyTune 2 Mini

Polyphonic Guitar Tuner Pedal
Guitar tuners now come in many guises, from clipon devices, plug-ins and phone apps to pedals and
rackmount offerings. TC...
Q How do I tackle the unwanted noise

coming from analogue modelling plug-ins?
I like to use a lot of analogue modelling plug-ins when Im
mixing, but while I like the sounds Im getting in general, I
always seem to end up with too much noise. Whats the best
way to tackle this problem?
Q Should I buy my own digital mixer for a

band tour?
Telefunken DD4
Dynamic Drum Microphone Set
Renowned microphone brand Telefunken tackle the

drum-mic kit market, with this high-end collection of
dynamic models.
Thermionic Culture Nightingale 2
Im the sound engineer with a group called Flyte,

who are heading out on a mini headline tour of
150-500 capacity venues next month. I have the
chance to acquire a little digital mixer such as a
Behringer X32 rack, but Im not sure if it would be worthwhile.
Channel Strip
With its parallel inputs, Thermionic Cultures valvebased channel strip enables all sorts of creative
recording options. And it sounds great, too!
UA RMX16
Reverb Plugin For UAD2 Platform
UAs emulation of the AMS digital reverb shows

exactly why the original is still regarded as a classic.
XLN Addictive Drums 2

Virtual Drum Instrument
Its been a long time coming, but the follow up to

XLNs Addictive Drums was worth the wait.
ZeroG Electro Cinematic

Multi Format
ZeroGs Electro Cinematic library provides

approximately 2.5GB of sample material.
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Fri 19 Dec 2014
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Competitions!
Win PreSonus Sceptre S8 monitors

S8 monitors
Win Munro Sonic Egg
150 (Global Except
Americas)
Win Yamaha White
Christmas bundle
(Americas Only)
Fantastic chance to Win PreSonus Sceptre S8

monitors worth 1239
In this months competition, we are giving you the chance to win a pair of
fantastic PreSonus Sceptre S8 active CoActual monitor speakers. Very much the
companys pro-level speaker, the Sceptre S8 is notable for employing a coaxial
driver. This means that the low and high frequencies originate from the same
point, rather than from two different drivers spaced apart on a baffle. The
practical benefit of this should be a wider sweet spot and more precise stereo
imaging and this is exactly what Editor In Chief Paul White discovered in his
July 2014 review (http://sosm.ag/jul14sceptre), saying that the the tonality held
up well over a wide listening area, with a good sense of stereo placement and
a stable phantom centre image.
One of the reasons you dont see many coaxial speakers on the market is
because its notoriously difficult to make them work well. However, PreSonus
have overcome the technical challenges with the help of speaker-design expert
Dave Gunness of Fulcrum Acoustics, whose proprietary TQ, or Temporal EQ,
system uses DSP to time-align the HF and LF sections of the driver. The DSP
also powers a user-adjustable EQ section that helpfully lets you tailor the
speakers response to your listening environment.
Driven by a pair of 90W Class-D amplifiers in each cabinet, the Sceptre S8 is
a great sonic performer with a frequency response of 46Hz to 20kHz (-3dB), with
a peak SPL of 116dB at one metre. According to Paul White, the mid-range
clarity is appealing and the sense of a mix that extends in three dimensions is
better than from many competing speakers. Our Editor In Chief also called the
sound of the PreSonus Sceptres revealing and involving, noting that they pull
no punches when it comes to sorting the good sounds from the bad all
enviable traits in a studio monitor, which will help you hone your mixes to
perfection.
If youd like the opportunity to win these serious studio speakers, just fill in and
return the form, or enter via our web site. The closing date for entries is 9th
February 2015. Good luck!
What is the frequency

response of the Sceptre S8? *
46Hz-20kHz (-3dB)
56Hz-20kHz (-3dB)
66Hz-20kHz (-3db)
46Hz-46kHz (-3dB)
What does TQ stand for? *

Temporal EQ
Time Quantisation
Trim Quality
Tiny Quail
Which speaker-design expert

helped with the design of the
Sceptre S8? *
Dave Gunness
Dane Guinness
Maeve Dunnes
Peter L. Jensen
Tie-breaker *
With a coaxial speaker, low and high
frequencies originate from the same
place. Tell us something about the
place from which you originate. Amuse
us in 30 words or fewer, please!
Prizes kindly donated by PreSonus.

Source Distribution +44 (0)20 8962 5080
www.presonus.com
The small print
Only one entry per person is permitted.
Employees of SOS Publications Group, PreSonus, and their immediate families
are ineligible for entry.
No cash alternative is available in lieu of the stated prize.
The competition organisers reserve the right to change the specification of the
prize offered.
The judges' decision is final and legally binding, and no correspondence will be
If you would like to receive more

information about PreSonus products,
please check this box.
Details
Name *
The judges' decision is final and legally binding, and no correspondence will be
entered into.
No other correspondence is to be included with competition entries.
Please ensure that you give your DAYTIME telephone number on your entry
form.
Prize winners must be prepared to make themselves available in the event that
the competition organisers wish to make a personal presentation.
Address *
Country *
- Select Daytime Telephone *
Email *
Difficulty reading the code?

Generate a new one.
What code is in the image? *

Enter the characters shown in the image.
SUBMIT ENTRY
Published in SOS January 2015
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All contents copyright SOS Publications Group and/or its licensors, 1985-2014. All rights reserved.
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. The views expressed
are those of the contributors and not necessarily those of the publishers.
Fri 19 Dec 2014
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In this article:
Bird Spotting
Test Flight
Real World
In Summary
Angelbird
SSD2go Pocket $450
pros
Very fast performance.
Compact and stylish format.
cons
Limited capacity compared
to standard hard drives or
disk arrays.
SSD technology is still
relatively expensive per
gigabyte of storage.
summary
Excellent performance in a
compact format, providing
you can justify the price of
entry.
information
Angelbird SSD2go
Pocket 500GB $449.99,
250GB $249.99, 128GB
$179.99.
info@angelbird.com
www.angelbird.com
Test Spec
Apple iMac, OS 10.9.5,
3.5GHz quad core i7, 32GB
RAM.
Focusrite Scarlett 8i6.
Cubase 7.5.20.
Kontakt 5.3.1.37.
News
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Subscribe
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Angelbird SSD2go Pocket

USB 3 Solid State Drive
Buy PDF
Reviews : Accessory
If you want more storage, want it faster and want it

smaller, Angelbirds portable SSDs might be just the
thing...
Printer-friendly version
John Walden
ike lofts, garages and garden sheds, no matter how big your hard drive is, you always manage to fill it. A little extra
storage is, therefore, always welcome. However, for us media types streaming lots of audio tracks, sample libraries
or video footage that extra storage needs to be large, fast and, preferably, portable. Solid-state drives (SSDs), while
still more expensive than traditional hard drives, can offer all these features, and Austrian company Angelbird have a range of
compact external SSDs aimed at media professionals. So, if you have a computer equipped with a suitable USB 3 port (fast
enough to exploit the performance of the drive itself), is a little bird a solution to your storage needs?
Bird Spotting
The model we received for review was the Angelbird SSD2go Pocket, a 512GB, USB 3 SSD and with an asking price
of $449.99. The company also make less expensive 128 and 256 GB models in the same range and there are other series
available offering even greater storage.
Physically, the SSD2go Pocket measures 90 x 90 x 11 mm and weighs 90g. The aluminium case seems very robust and the
design is intended to be resistant to the usual bumps and vibrations encountered in mobile use; I certainly wouldnt have any
qualms about travelling with it within my standard laptop case. Connected to a suitable USB 3 port, the stated performance is
sustained read speeds of 450MB/s and write speeds of 390MB/s that, if delivered in practice, would certainly offer plenty of
capacity for audio and video work.
The review unit came in a rather fetching metallic blue, but, for this
model, you can also choose between red, silver and black. It is supplied
with a dedicated high-quality USB cable of around 30cm in length. There
is also a five-year limited warranty. Angelbird have obviously got media
professionals in their sights with these units as, alongside a time-limited
trial of Parallels Desktop 10 for Mac, is a further time-limited, but fully
functional, trial version of BitWig Studio music production software for
Windows, OS X and Linux, and a number of audio production video
tutorials from Pure Mix.
Test Flight
The SSD2go Pocket can be hot-plugged like any other USB-based drive
(providing you follow the usual procedure for your particular OS). I did
most of my testing using a USB 3-equipped iMac and had no issues
mounting/unmounting the drive. It instantly became visible to my system
and seemed very nifty when used for standard file transfers.
While Im always a little cautious about the data generated by diskspeed benchmark software, testing the Angelbird via BlackMagic Desk
Speed Test returned read and write speeds that averaged around
420MB/s and 360MB/s respectively. For comparison, the 3T 7200rpm SATA internal hard drive on the test system reported
around 115MB/s in both directions, while a Thunderbolt 1-connected, six-disk RAID array (considerably more expensive and
less portable than the SSD2go, although it does have much greater capacity also) gave figures of approximately 400 (read)
and 520 MB/s (write).
This Thunderbolt drive could probably deliver somewhat faster
performance if configured differently. It is, however, the drive that I
use for all my audio and sample library streaming and its
performance has never caused me any bottlenecks even when
running quite busy projects. That the SSD2go Pocket gets very
close in performance is, therefore, actually pretty impressive.
Incidentally, brief speed tests of the SSD2go with a similarly USB 3equipped Windows PC gave similar results.
Real World
Routine benchmarking is, of course, something different from the
performance in a practical real world situation. To explore this, I
copied a few Kontakt-based sample libraries over to the SSD2go
and then configured a Cubase project to run from the drive using a
combination of both audio and MIDI tracks, with the latter running
various Kontakt patches from the SSD2go.
While I appreciate this is a totally unscientific test, having set up a
project featuring eight audio tracks and 16 MIDI tracks, I then simply
set about duplicating these and offsetting them on the timeline. By
However many pinches of salt you take with your routine
the time I reached around 100 tracks in total, not only was the
project making an unholy racket but I got bored with waiting for the
disk to bottleneck me. It didnt, but I did get the occasional CPU
spike as my iMac tried to keep up. If you need to run a Hans
Zimmer-esque mega-sample-library mock-up arrangement for
your latest film score then perhaps the Angelbird will eventually
run out of steam, but, for the rest of us in the real world, for
typical music production projects, I suspect the performance of
this compact SSD would be more than adequate and outstrip a
standard internal SATA drive by some considerable margin.
disk benchmark tests, the Angelbirds performance is

pretty impressive.
In Summary
If you are old enough to remember the early days of computerbased hard-disk audio recording and what passed for an
adequate track-count performance, todays generations of hard
How busy do you like your DAW projects to be? The
drives can seem like something of a minor miracle. Of course,
Angelbird seemed quite happy streaming all the audio and
we now also place far greater loads on our music production
sample data required for my complex Cubase test project.
computers; more complex recording software, multiple virtual
instruments, complex arrays of audio effects plug-ins and multi-gigabyte, multi-layered sample libraries. Any boost in
performance is therefore welcome, wherever it might sit in the processing chain.
SSD technology, while still carrying something of a premium price per gigabyte of storage, is now getting to be mainstream.
The Angelbird SSD2go Pocket will not be the cheapest 500GB of drive storage that you can currently buy, but it provides that
storage in a very portable, seemingly robust and very stylish format. Whether you can justify the cost is, of course, a personal
decision, but, used with a suitable USB 3-equipped computer, the Anglebird SSD2go Pocket also delivers very impressive
levels of performance. If you need high performance in a mobile drive and can live within the 500GB capacity, this bird really
can fly.
.
GLOSSARY: technical terms

explained
Competitions!
S8 monitors
Win Munro Sonic Egg
150 (Global Except
Americas)
Win Yamaha White
Christmas bundle
(Americas Only)
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In this article:
Apple iPad Air 2
MIDI Comes To
Windows (Again)
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Appworks
Making Music On The Move
Reviews : Software: ALL
Mark Wherry
Buy PDF
Apple iPad Air 2
Tablet Computer
hen Steve Jobs returned to Apple, one the first decisions he made as interim CEO was to simplify Apples product
line. Today, looking at the companys latest iPad line up, you cant help wondering if Apple couldnt benefit from re
enacting the same strategy. In striving to have an iPad at a range of similar prices, the iPad product line now
consists of the iPad mini, iPad mini 2, the new iPad mini 3, the iPad Air and, the new flagship iPad, the somewhat awkwardly
named iPad Air 2. If I was coming to the iPad for the first time as a consumer, I have to say that Id probably be more than a
little confused as to the differences between the various models, other than the price.
To be brief, the iPad mini (reviewed back in the January 2013 issue) is best avoided as its architecturally similar to the iPad
2 (reviewed in June 2011) and the A5 chip powering the device is starting to buckle under the pressure of the most recent iOS
releases. The iPad mini 2 and iPad Air (reviewed in April 2014) are more or less the same technology in two different form
factors and are still worthy of consideration. The latest iPad mini 3, however, disappointingly still has essentially the same guts
as the iPad mini 2, but with the addition of the Touch ID fingerprint sensor.
All of which brings us to the iPad Air 2. Design wise, this latest and
greatest iPad is basically the same as its predecessor, except that Apple
have added Touch ID and once again made the device slimmer. Where the
iPad Air had a depth of 7.5mm (consider that the original iPad was 13mm),
the Air 2 is 1.4mm thinner at just 6.1mm. Now, you might be thinking this isnt
a notable figure, but the impression of svelteness that hits you when you pick
the device up is remarkable, and were clearly getting to the point where
Apple are going to have to remove the headphone jack to flatten the device
any further.
One thing that has been removed, though, is the switch that was
previously next to the volume buttons, which isnt perhaps a big surprise
given that the company has struggled for some time over whether the
function of this switch should mute the sound or lock the displays rotation.
Internally, the iPad features an A8X SoC (SystemonaChip), which is
based on the A8 SoC that debuted in the iPhone 6, but is clocked slightly
faster (1.5 versus 1.4 GHz) and features more powerful graphics to drive the
iPads larger display. While I wrote in the iPhone 6 review that the
performance jump from the A7 used in the iPhone 5S wasnt as significant as
the previous generation, the A8X represents a very significant improvement
over the A7 used in the iPad Air and, for that matter, the A8 used in the iPhone 6. Rerunning Geekbench 3 on my iPad Air,
now with iOS 8.1, I got a multicore score of 2663; but the Air 2 has a multicore score of 4418, making it theoretically, at
least 78 percent faster. (The iPhone 6 scored 2908.) To put this score in perspective, in terms of raw computing, the A8X is
on a par with a late2013 13inch Retina MacBook Pro with a 2.4GHz i5 processor, or a 2006 Mac Pro with dual 2.66GHz
Xeon 5150 processors.
One of the reasons for the dramatic increase in performance is
the fact that the A8X is Apples first triplecore An chip; the A8
and A7 are dual core. So the actual singlecore performance
improvement isnt quite so significant: in Geekbenchs single
core test, the iPad Airs A7 scores 1469, iPhone 6s A8 1622, and
the Air 2s A8X 1747. What this means, since I would be willing to
bet the audio engines of most iOS music applications dont scale
across multiple cores in the way that regular Mac and Windows
applications do, is that youll only get a marginal performance increase when running a single app. Indeed, playing the demo
song that comes with WaveMachine Labs Auria, the Max CPU reading showed 40 percent for the iPad Air and 38 percent for
the Air 2. Playing the song a second time (which isnt something I would recommend), you could observe about a roughly five
percent improvement during playback on the iPad Air 2.
Where having the extra core should come in handy, though, is if youre running more than one app at the same time.
Although its hard to really analyse whats going on in iOS, it should mean that different audio engines for different apps can
run on different cores and thus allow for more processing power. Running Cubasis, I opened up three different synths (Nave,
Sunrizer, and PPG WaveGenerator) via InterApp Audio, each playing a simple pattern at the same time, and using System
Status (an app that is roughly equivalent to OS Xs Activity Monitor) it seemed very much as though the iPad Air 2 was being
taxed less than the original Air. The only slightly weird thing was that when switching out of Cubasis on the Air 2, I would
usually get a brief moment of crackling in the audio, which could be resolved by setting the buffer size to 512 samples (from
256). This didnt happen while actually in Cubasis or any of the synth apps, or indeed on the iPad Air, which was also set to
256 samples.
While the A8Xs performance is impressive, the really great thing about this new SoC is that Apple have finally included
2GB of memory. Geekbench reports 1.94GB of memory (versus 975MB on the Air), and this is a big deal for those musicians
who use AudioBus or InterApp Audio to run an increasing number of apps simultaneously.
In terms of pricing, the iPad Air 2 starts at the same $499 price as its predecessors. However, as with the iPhone 6, the
32GB model that previously sold at $599 has been replaced by the 64GB model for the same price. So unless budget is an
issue, to pay an extra $100 for four times the storage of the base model seems like a pretty good deal. But if even 64GB isnt
going to satisfy you, theres also a 128GB model for $799, and cellular Internet connectivity is available on any model for an
extra $130.
If you want to run music and audio apps on an iOSbased tablet, the iPad Air 2 is without doubt the device for you. If youre
using a preAir iPad and use it as more than just a controller, the Air 2 is a worthy upgrade with a noticeable improvement in
performance. For existing iPad Air users, an upgrade is probably only worthwhile if you find yourself running out of gas when
using multiple apps. Mark Wherry
From $499
www.apple.com
.
MIDI Comes To Windows (Again)

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Earlier in the year at Microsofts Build conference for developers, the company announced the concept of a universal
application one app that can run on phones, tablets, computers and, eventually, the Xbox One. What this means for
music and audio apps remains to be seen, given that up until now the number and quality of music and audio apps in the
Windows Store has been lacking to say the least.
One reason for this has been the absence of a MIDI API (Application Programming Interface) developers could use
when creating apps with the newer Windows Runtime (WinRT) architecture to which Windows Store apps must conform.
And while, considering the competition, Core MIDI wasnt actually implemented until iOS 4.2, the enthusiasm for the
iPhone and iPad led to developers creating all manner of workarounds, custom peripherals and APIs (remember the first
version of Line 6s MIDI Mobilizer?) for iOS. Sadly there hasnt been such developer fervour for Microsofts Surface range
and thirdparty Windows 8 devices.
However, Microsoft clearly understand this and one of the sessions at the Build conference was entitled Sequencers,
Synthesizers, and Software, Oh My!. During this session, a preview was announced and made available of a MIDI API for
WinRT that allows apps to use MIDI hardware finally! This makes it possible for MIDI Windows Store apps to exist that
can run on any device running Windows 8.1, whether the device is based around an Intel chip or an ARMbased SoC
(systemonachip). The only slight difference is that ARMbased devices, like Microsofts own Surface 2 tablet, can only
use classcompliant USB MIDI devices (as with iOS), while Intelbased devices can make use of whatever custom drivers
a manufacturer might supply with the MIDI hardware.

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The reason its a preview is that Microsoft are asking developers for feedback, to see where the API needs to go, and a
big thing I would imagine most people wanting is support for RTP MIDI so you could send MIDI back and forth over a
network. RTP (RealTime Protocol) MIDI is the protocol on which Core MIDIs networking features are based.
Disappointingly, though, at the time of writing in lateNovember, this preview MIDI API has only been downloaded 369
times, so it may still be some time before we see a rush of MIDI apps on the Windows Store.
So as not to end this box in a depressing tone, the session concluded with a demonstration of a simple universal audio
app (a very basic MPClike app) that could run on Windows 8.1 or Windows Phone 8.1, which was kind of neat. Mark
Wherry
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Fri 19 Dec 2014
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In this article:
Beat That!
Identity Parade
Sequencer Mode
Conclusion
Alternatives
System Requirements
Arturia
Beatstep $99
pros
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Arturia Beatstep
MIDI Controller & Step Sequencer
Reviews : Hardware Sequencer
Arturias Beatstep combines a MIDI controller and a step

sequencer in one extremely reasonably priced box.
Buy PDF
Paul Nagle
aving broken free of the virtual world to build analogue synthesizers, Arturia continue to expand and consolidate their
range. The Beatstep is a combined drum pad, MIDI controller and step sequencer eager to communicate with CV/Gate
synths, MIDI hardware, the studio computer or iPad. In or out of the box, on of off stage, this tiny controller promises
much and costs little. It sounds almost too good to be true.
Intriguing combination of
MIDI controller, drum pads
and step sequencer.
Includes CV/Gate
Beat That!
sequencing and MIDI to CV
translation too.
The
Beatstep
is heavier than its tasteful creamcoloured body suggests. Measuring just 320 x 130 x 17 mm, the design is
Solid, compact and portable.
similar to the Spark LE drum machine, right down to the rubber buttons and charcoal grey encoders. The encoders are slightly
The price is right.
cons
No MIDI In port.
Sequencer lacks accent and
some functionality is only
accessible from software.
summary
An ideal DAW companion
whether for its controller
features, those handy drum
pads or the simple step
sequencer. With its
capabilities extending to
MIDI and CV synths, only
the lack of external sync
prevents it being an impulse
buy for hardwareonly users
too.
information
$99
www.arturia.com
notched and feel good, as do the backlit drum pads arranged in two rows of eight. In sporting pads with adjustable velocity
response and polyphonic aftertouch, the Beatstep earns an initial rush of positivity. I spotted only one minor flaw: the grey,
paperlike side coverings protrude slightly over the metal base. This feels cheap and vulnerable to being accidentally torn off.
Rated as a compact, assignable MIDI controller, the Beatstep would score highly, but theres another side to its personality
that of a step sequencer. (The clues are in the name, the transport buttons and the labels printed beneath the pads.)
Fortunately, the two modes coexist peacefully, with overall control toggled by alternate presses of the Cntrl/Seq button. When
in controller mode the button is red, while sequencer mode turns it blue. Other buttons provide external synchronisation, patch
storage and MIDI channel selection, and we mustnt forget the inevitable shift key, which is used to access features such as
scale, sequence direction and speed.
Due to the Beatsteps sylphlike form, regular MIDI
connectivity is not a practical option. Instead, theres a 3.5mm
jack, plus a short MIDI adapter packed in with the USB cable.
Arturia provide only one of these adapters because theres only
one MIDI port, the output. This is a blow to those hoping for an
affordable sequencer to slot into a hardware environment. Bi
directional MIDI is possible, but only via the mini-USB port and
an attached computer. Since its a class-compliant device, the
Beatstep is neatly plug and play. It derives power from USB and
this applies whether connected to a studio computer, laptop or
iPad. Fortunately the draw is light and my iPad Air handled it
without complaint, which is just as well because theres no stand
alone power adapter.
Lastly, and cementing the Beatsteps jack of all trades
credentials, two 3.5mm sockets carry notes and gates to the outside world. Arturia have a nifty line in petite analogues and the
Beatstep is ready to talk to them or to any synth or modular that conforms to the Oct/Volt standard.
Identity Parade
The pads and encoders are evenly distributed, making both finger drumming and knobtwiddling equally comfortable. There
are 16 sequences and 16 sets of controller assignments stored
for instant recall. Without a display to help, I couldnt claim to
remember every encoder assignment, but two or three complete
configuration memories were sufficient for most of my soft synth
tweaks. I only briefly wished for an oldschool paper overlay to
scribble on.
Lets start by considering the Beatstep as a typical MIDI
control surface. It arrives primed with a default set of controller
and note assignments. Given that most virtual instruments have
a MIDI Learn function, the defaults could be fine for quite a while.
Eventually though, youll want to personalise them. All
assignments are made in the MIDI Control Centre software and
its a fair bet this will be the first download once youve registered
your Beatstep and checked for updates. The MCC software
requires little explanation, being instantly accessible and well
The Beatsteps MIDI Control Centre software.
organised. It permits the definition of CCs and NRPNs for the
encoders, or a choice of notes, CCs, program changes or even MMC commands for every pad.
The software offers a number of acceleration curves to ensure this neednt be one of those tedious encoder experiences
where it takes a dozen spins to cover a single filter sweep. I couldnt manage the full range in one go, but taking the Fast
option I was able to zip from zero to 127 in only two rotations and slow down enough for individual increments.
The pads feel great straight out of the box and although I was quite satisfied with their default action, four velocity curves
are available to address different finger strengths. If I were being picky Id mention that the polyphonic aftertouch wasnt
completely smooth across its 7bit range. However, its hard to remain critical for long. Perched on top of my Yamaha KX8, a
master keyboard low on knobs and entirely devoid of pads, the

Beatstep began to look very appealing.
Sequencer Mode
Flip into sequencer mode and you have a basic but effective
simulation of an oldschool step sequencer. The encoders set
the pitch of each step over a four-octave range, while the cool
blue pads act as step mute buttons. The main limitation is that
the sequencer only sends patterns of notes, all at a fixed velocity.
So there are no patterns of MIDI CCs to amuse the
experimentally minded and lovers of 303type sequences will
have to cope without accents. Get beyond these omissions,
though, and the Beatstep has a number of tasty features with
which to spice up your patterns.
The Beatsteps front panel measures a compact 320 x 130

mm.
During playback, a progress light marches across the pads and theres an easy visual correlation between pads and
encoders. Heres where those subtle notches become really useful because, unlike the fully variable tuning of a classic
analogue sequencer, every notch represents a semitone. You therefore have a reasonable chance of setting intervals by feel
as the sequencer loops. It can get ugly though, especially at speed, so its a good job Arturia included that most important of
noteprogramming aids scales.
Picking one of the eight available scales has no effect on the active sequence. Instead, it restricts the notes dialled up by
the encoders. Assuming your base sequence is in C, selecting a major scale would mean the encoders only ever produce
white notes. The scales on offer are: chromatic (all notes), major, minor, dorian, mixolydian, harmonic minor and blues.
Joining these presets is a single user scale. As youd expect this is defined in the MCC where it appears as a graphical filter
blue notes are included, white notes excluded. Id have preferred all the scales to be userprogrammable, but perhaps
thats because I spent way too long spinning Frank Zappa records in my youth.
The shift key, in conjunction with the large data encoder, is used to transpose the current sequence. Again this is in
semitones down by a maximum of one octave, up by three. Transposition is realtime ie. it cant be prepared in advance
and confirmed at a critical point during playback. Its usually preferable to transpose with a MIDI keyboard, providing your
DAW provides a friendly means of rerouting MIDI inputs and outputs (ie. not Logic). With no visual indication of the current key
or transposition, it becomes natural to work by ear, discovering new patterns by the ageold method of random grabbing and
turning. Whatever you come up with, up to 16 sequences can be stored internally to be archived at your leisure.
In order to make patterns more interesting, there are four playback directions: forwards, backwards, alternating or random.
Step sequencers are all about interaction, so as well as the different directions, patterns can run at one of four different
speeds, from quarter up to 1/32nd notes (triplets arent catered for). Lastly, at any time during playback, a patterns length can
be reduced from its default 16 steps by holding down the Shift and Chan buttons and pressing one of the pads.
If you switch speed, length or direction, its likely youll get out of step with other material in your current song. Depending on
the tempo, it might take a few attempts to get lined up again. This is something live performers need to be aware of and its a
pity you cant simply stop and start the sequencer while synced to your DAW, as youd expect to do in a purely hardware
setup. Some kind of resync function would be much appreciated in a future firmware update.
Not all the available functionality is accessible from hardware. Specifically, swing, legato and gate time (step duration) are
only reachable from the MCC software. It seems an opportunity lost that you cant program these into Beatsteps controller
persona. When legato is enabled, all notes are played as if they overlap and any two steps triggering the same note are
merged. While this can hint at 303type slides, the feature is seriously compromised without the option to set legato on
individual steps only. I guess thats another request.
Patterns can be freely routed to virtual instruments, MIDI hardware or the CV/Gate outputs but its perhaps less obvious that
the Beatstep can serve as a MIDI to CV converter. Fortunately, it requires no more effort than routing a chosen DAW track to
the Beatsteps virtual port, which is then neatly forwarded on to the CV/Gate and/or MIDI output.
Naturally the sequencer continues playing when you switch to controller mode, providing the opportunity to capture long,
gradually evolving patterns and simultaneous parameter tweaks. With the Beatstep synced to Logic and Uhes Diva firing on
all cylinders, I came scarily close to the experience of analogue hardware noodling; with the added advantage of going back
over it later to fix cockups and add individual accents and slides.
Conclusion
The Beatstep is a tasty and compact controller for MIDI tweaking and drum programming. Making it a step sequencer too
even one so basic is a master stroke! Admittedly its patterns are of fixed velocity and the sequencing of MIDI CCs has
been omitted, but within limitations, youve got a ready source of inspiration ripe for further development in your DAW.
Certainly, the technique of spinning encoders and switching scales should lead to patterns in complete contrast to anything
youd play on a keyboard. And having chanced across a sequence that takes your fancy, with the press of a single button, the
same interface is ready to tweak filters, envelopes and effects.
For a modest outlay this creamy little Arturia ticks so many boxes its hard to resist. Its a shame theres no MIDI input, but I
doubt external sync will be an issue for the bulk of users working mainly in a DAW environment. Ultimately, if youre a laptop
performer in the market for drum pads, a flexible control surface and a handson step sequencer, the Beatstep could be
exactly what youre looking for!
.
Alternatives
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For a similar price, Akais MPX8 offers eight pads and no sequencer or programmable encoders, but does add sample
playback. Other controllers from Korg and Akai provide inexpensive drum pads too, but if you want the full Beatstep
package theres currently nothing to beat it.
System Requirements
Windows: Win 7+, 4GB RAM, 2GHz CPU.
Mac: 10.7+, 4GB RAM, 2GHz CPU.
Runs standalone or VST 2.4 (32bit and 64bit), VST 3 (32bit and 64bit), RTAS, AAX (Pro Tools 11), Audio Units
(32bit and 64bit).

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Audiowiesel Hammered Acoustic Guitar

Kontakt Instrument
Reviews : Sample Shop
Tom Flint

Audiowiesel are a small fledgling company run by Austrian composers Frank Herrlinger and Daniel
Szwedek. The pair created Hammered Acoustic Guitar, or HAG, as they refer to it, using an Ibanez AE guitar and beating it
with drum sticks and brushes. The resulting samples are accessed through a series of Kontakt interfaces that include a
number of sound-processing tools and offer seven velocity layers and nine roundrobin variations per patch. Customers have
the option of purchasing HAG Core and HAG FX & Percussion separately, or both together as the HAG Full Bundle.
The Core interface is more sophisticated than the FX & Percussion ones. It
enables up to two brush samples and two stick samples to be mixed together
to form a patch. Balancing the four is done in mixer mode, however, there are
also four settings pages from where the output of each sample channel can
be altered. Controls include a lowpass filter, dynamic range adjustments,
envelope controls and pitch tuning and transpose options. Alternatively, a
Presets mode offers a selection of readymade patches that are good to go.
The FX & Percussion interfaces differ in that they lack the multichannel
mixing abilities of Core, but instead benefit from a number of performance
related keyswitch options which make it easy to do things like select sounds,
transpose the keys by an octave and choose LFO Sync fractions and
multipliers, which reference the hosts tempo.
Although all of the FX & Percussion instruments are based on the same
basic interface, Frank and Daniel have customised some of them so that they
provide a few more programming options. For example, FX instruments have
a MIDI gate density slider as part of their interface, unless they are labelled
Tonal, in which case a randomdetune slider exists instead. The Percussion
interfaces, on the other hand, have a morphing slider that is clean at one end
and distorted at the other.
But it is the Core instrument that is the more flexible. It is capable of sounding clean and simple if one or two channels are
used, or rather complex if all four sources are active and given very different envelope, effect, EQ, pitch and range settings.
For example, it can be made to sound like a spooky cimbalom reminiscent of the one used in The Ipcress File film soundtrack,
which in my book is a very good thing! That said, a few more modulation options would help it compete with the most
sophisticated rival instruments.
Without knowing where they came from it would be nearimpossible to guess that the FX instruments were derived from
hammering guitar string. They range from subharmonic thuds to synthlike tones, and could be used to underpin drums or
for atmospherics effects in scifi sound design.
The Percussion instruments are similarly far removed from their origin. Audiowiesel have assembled a sort of electronic
drum kit of sounds, including approximations of wood block, brushes and bongo. There is also an instrument called Dirty Slam
Hits MW, which offers heavily processed and distorted sounds and has a keyoperated MIDI gate for creating stutter effects.
To sum up, hitting an acoustic guitar has created a reasonably unique virtual instrument that will interest composers looking
to expand their sonic palette. Tom Flint
Audio Example
Audio examples for the Sample Shop articles can be found here.
$99
www.audiowiesel.com
.
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In this article:
Overview
Sound
Chandler
TG2500 $955
pros
Useful preamp colour option
for 500series systems.
The level of distortion can
be balanced easily.
Simple, easy-to-use
controls.
Mic impedance options.
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Chandler Limited TG2500

Solidstate Microphone Preamplifier
Reviews : Preamp
Can Chandler pack all of the EMI/Abbey Road sound for

which theyre so famous into a single-module 500series
preamp?
Buy PDF
Hugh Robjohns
t seems odd that one of the first highquality British professional mixing consoles, which underpins so many famous rock
albums, lives on as hardware only in the products of an American manufacturer. Nevertheless, the fact it does must be
considered A Good Thing, because Chandlers EMI TGbased products recreate this unique British design from the late
cons
1960s and early 70s remarkably well. I reviewed the companys TG Channel back in January 2005
Phantom power not disabled
(www.soundonsound.com/sos/jan05/articles/chandlerchannel.htm), a product which uses the same EMI TG12428 preamp
in line mode.
section as employed in Chandlers rackmounting TG2 preamp/DI, retaining the alldiscrete silicon transistor circuitry and
No DI input option.
(virtually identical) input and output transformers.
summary
A redesign of Chandlers
TG2 preamp/DI to fit a 500
series chassis. The new
product lacks the DI input,
and there are some other
minor design changes, but
the underlying sonic
character remains as strong
and attractive as ever.
information
The latest incarnation of this classic preamp design is Chandlers new TG2500, which, as the model numbering indicates,
is a reformatted version designed for use in an API 500series Lunchbox chassis. Apparently this product was quite a while
in the making before company owner Wade Goeke declared himself happy with the various compromises and practicalities of
the physical remodelling.
Overview
The TG2500 is a single modulewidth unit that employs only the input and output connectors on the host 500series rack, so
it lacks the DI input facility of its rackmount antecedent, and there are a few other changes too. However, the rearpanel input
can be configured for either mic or linelevel signals via a frontpanel toggle switch, allowing the preamp to be used as a line
conditioner if required.
$955.
Chandler Limited +1 319
885 4200.
support@chandlerlimited.com
chandlerlimited.com
With just three rotary controls (all with nice chickenhead knobs), two toggle
switches, and two pushbuttons, the TG2500 is pretty straightforward to use,
although some status indicator LEDs would have been nice, even if not authentic.
The top rotary switch has a red knob and sets the gain in seven 5dB steps, covering a
30dB range. This is rather different from the rackmounting TG2, which has a 15
position switch covering a much broader gain range. The middle (blackknobbed)
potentiometer is also a departure from the TG2 preamp/DI and provides a continuous
gain trim over roughly 10dB to allow fine level adjustment, filling in the gaps of the
coarse gain switch and extending the gain range in both directions. The bottom
potentiometer has a grey knob and provides an outputlevel control, acting as an
output fader, the same as the rackmount unit.
When switched to mic mode the maximum gain, as marked on the coarse gain
scale, is 50dB which is what I measured with the Trim control centred and the
output control at maximum. With the trim control turned up to maximum as well, a
further 10dB of gain is applied, for a total mic gain of 60dB. If switched to line mode
the input sensitivity is reduced by 10dB. This means that with the coarse gain at
minimum, trim centred and output at maximum, a 0dBu line input signal appears at the
output at +5dBu. Turning the output control back to 6.5 on the scale establishes unity
gain.
Increasing the input level results in gently increasing harmonic distortion, with the output stage eventually clipping at around
+27dBu. This is reached with a mic input signal of 20dBu when operating at maximum gain and with the output control fully
clockwise, or +12dBu when at minimum gain. This indicates a very usable headroom margin, plus the ability to back off the
output control and increase the input gain to achieve overdriven effects.
A second toggle switch, below the coarse gain control,

changes the input transformer connections to offer two input
impedances which are marked as either 300 or 1200 . These
derive from obsolete oldschool impedance-matching practices,
but according to my NTI Minirator 2 actually present 640 and
2200 loads to the microphone. Nevertheless, they offer two
slightly different tonal characteristics when used with dynamic
and transformercoupled microphones, which can be useful.
Two Neveesque white pushbuttons at the bottom of the
module activate 48V phantom power and introduce a polarity
inversion respectively. Sadly, and potentially dangerously to
some connected linelevel equipment, the phantom power is not
disabled when the input is switched to Line mode, and without a
status indicator its easy to overlook the fact that phantom power
is switched on.
The rackmount TG2 preamp/DI model runs on internal 24V
supplies, whereas the 500series racks offer only 16V supplies.
These reduced power rails theoretically limit the headroom,
although there was no evidence of any problems during my tests
and usage, as Ive already suggested. The TG2800s
specifications indicate fairly modest power supply demands, too,
drawing 110 and 80 mA from the positive and negative power
rails, respectively.
The inputstage circuitry appears to comprise two
complementary pairs of BC214 (PNP) and BC184 (NPN)
transistors, working in concert with the free gain provided by the
Carnhill input transformer. This is a relatively lownoise design
which works perfectly well in most applications, but might not cut
the mustard for distant, lowoutput ribbons! The output stage
employs two chunky metalcanned output transistors (the type
markings had been scratched off) in a pushpull configuration
driving an Altran 4339 gapped output transformer.
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One for the electronics geeks out there...
Neve 1073LB &

1073LB EQ
Here, you can see how the frequency response changes,

with more harmonic distortion being introduced as the
preamp is driven harder.
500-series Microphone
Preamplifier & Equaliser
Neves venerable
1073 preamp and
equaliser are both
now available in
APIs popular Lunchbox
format. Were they worth
waiting for?
Cloud Microphones
Cloudlifters
In-line Microphone
Preamplifiers
Do these in-line
Sound
mic preamplifiers
mean you can use
This Chandler preamp could not be described as a neutral preamp: it boasts a larger than life sound with a full bottom end, a
a passive ribbon
noticeable richness through the mid-range, and an open, airy top. For comparative purposes, Id suggest the early Neve or
mic
with
any
preamp?
Calrec preamps: all shared similar characteristics due to the similar discrete transistor circuitry and transformers. In short,
everything is slightly exaggerated, but in a musically flattering way. The topend lift is not harsh or aggressive, just airy with a
gentle rise above about 3kHz. This is largely courtesy of the output transformer, reaching a modest 2.5dB peak at 25kHz. The
frequency response overall lies within 1dB between 50Hz and 12kHz (at maximum gain), and the low end rollsoff gently to
3dB at 20Hz. The LF response extends further at lower gain settings, which is not unusual, although theres no highpass
filter option on the preamp.
Usefully, driving the input stage harder than normal, by juggling the coarse gain, trim and output fader, allows the amount of
harmonic distortion to be increased in a musically enhancing and very controllable way. The distortion tends to be mainly odd
harmonics the infamous transistor fizz rather than the rich thickness thats more associated with valve electronics. The
Chandler web site refers to John Lennons vocal on the classic Beatles track Polythene Pam as a good example of the kind of
(heavily overdriven) distortion on offer here.
So, the TG2500 is definitely a flavoursome preamp which injects nice sense of character and body into what might
otherwise appear sterile and boring sound sources. Im not sure Id want to use it for everything, simply because too much of a
good thing can detract, but for vocals and solo instruments, and especially for bass guitars, I found it a very useful option to
have in my Lunchbox.
.
Slate Pro Audio Fox

| Media
Dual-channel Microphone
Preamplifier
Test plots to
accompany the
article.
Cloud Microphones
Cloudlifter
Audio Examples
Audio files to accompany the
article.
Slate Pro Audio Fox
Dual-channel Microphone
Preamplifier
With two channels
and four flavours
on offer, Slates
mic preamp
promises plenty of flexibility.
Does it also deliver on quality?
Neve 4081
Four-channel Microphone
Preamplifier
Neve believe that
theres scope to
bring classic
designs up to date
and thats exactly what
theyve done here, taking their
revered 1081 mic preamplifier
as the starting point.
Radial Tonebone PZ
Pre
Acoustic Instrument Preamp
James Dunkley is
on the case of the
Radial Tonebone
PZ Preamp.
Drawmer HQ
Preamplifier & DA
Converter
Can a preamp and DA
converter successfully
straddle the pro-audio and hifi
Fri 19 Dec 2014
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In this article:
Down The Signal Path
Modulation & Analogue
Control
Sequencing &
Arpeggiation
In Use
Conclusions
Alternatives
Paraphonic Mode
Memories
Round The Back
Special Waves
Abridged Specification
DSI
Pro 2 $2000
pros
It looks and sounds
gorgeous.
It can produce the dirtiest,
angriest sounds Ive ever
heard emanating from an
integrated synth.
Its deep but not
impenetrable.
Paraphonic mode is much
more useful than you might
imagine.
The amount of analogue
connectivity is unusually
high for an integrated synth.
cons
There are issues with the
CV architecture to be
addressed.
The sequencer would
benefit from further
development.
summary
Dave Smith has been asked
on numerous occasions to
release a modern
incarnation of the Pro One,
but the Pro 2 isnt it. Its
much, much more than that.
And, although a superficial
glance suggests that its
similar to one voice of the
Prophet 12, its much, much
more than that too.
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Dave Smith Instruments Pro 2

Monophonic Synthesizer
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Reviews : Synthesizer
The depth and complexity of the Pro 2 could make it Dave

Smiths most sophisticated monosynth yet.
Gordon Reid
hen DSI announced the Pro 2, many considered it to be the longanticipated successor to the esteemed Sequential
Circuits (SCI) Pro One from the early 1980s. Of course, its not; its architecture and the breadth of its capabilities
make it at best a very distant descendent. Nevertheless, the name is well chosen because it suggests that this is
going to be a cut above any previous DSI monosynth. Unfortunately, some people arent going to give it a chance to prove
this; Ive already read comments (many written, I fear, by people who have never played one) that dismiss it because of its
hybrid analogue/digital architecture. So lets be clear: the filters, the audio amplifier and the final output stage are analogue.
The oscillators and Character effects are digital, as are the LFOs, the contour generators, the modulation matrix and the delay
lines. If this bothers you, feel free to read no further. But it would be a shame if you went for a bevvy now, because theres lots
of exciting stuff to be discovered here.
Down The Signal Path

Given its appearance, its inevitable that people are also going to view the Pro 2 as a monophonic version of the Prophet 12.
In fairness, there are many points of similarity, but there are as many significant differences between the two. For example,
whereas the 48 primary oscillators in the polysynth offer 12 Special Waves and three colours of noise, in addition to the four
standard analogue waveforms, the Pro 2s four primary oscillators also offer 13 additional Superwaves. These comprise
multiple instances of the sawtooth and each of the special waves, and you can detune these to create chorusing. Because
you can modulate the amount of detune (and do so using numerous different modulation sources acting upon a different
Superwave in each of the four oscillator slots) you can generate all manner of sounds ranging from the delightfully subtle to
extravagant layers of wibbly mayhem. Add oscillator sync, FM and AM of each oscillator into the equation, and a vast range of
chorused, synced and modulated (and simultaneously chorused, synced and modulated) sounds are possible. Now add
Osc1s sine wave suboscillator, noise modulation, and multiple types of portamento that you can apply to each oscillator
individually, and its clear that the Pro 2 is far more than a modern hybrid pretending to be a traditional analogue monosynth.
Lying between the oscillator mixer and the filters, there are five
waveshaping effects derived from the Prophet 12. Girth and Air
accentuate the low and high frequencies (respectively) of the
sounds, while Hack and Decimation reduce their word length and
sample rate before conversion to the analogue signals that will
be fed to the filters. The fifth effect is Drive, which allows you to
create anything ranging from a gentle overdrive to (with
appropriate Hacking and Decimation) the sounds of alien hordes
invading.
Next, the Prophet 12s powerful filter section has been

replaced by an even more powerful architecture that consists of
a selfoscillating 24dB/oct lowpass filter and a 12dB/oct,
resonant but not selfoscillating, statevariable (LP/BR/HP/BP)
filter that can be patched in series or in parallel. Actually, thats
not quite right. At the two extremes of the routing parameter
youll find pure in series and pure in parallel, but you can dial in
information
any value you fancy between these, blending the signal filtered
$1999.99
by both devices in series with the signals processed by Filter 1
Dave Smith Instruments
alone and by Filter 2 alone. In addition, you can direct just
+1 707 963 7006
oscillators 1 & 2 to Filter 1, and just oscillators 3 & 4 to Filter 2.
mail@davesmithinstruments.com
These are not trivial tricks.
www.davesmithinstruments.com
Each filter has a dedicated contour generator to control its
cutoff frequency. From the control panel, you can treat each as a
bipolar ADSR, but the menus provide access to a delay before
the onset of the Attack (making each contour a fivestage
HADSR), plus velocity sensitivity for the contour amount.
Naturally, the contour rates can be modulated, and theres also a
Repeat mode that loops the HAD stages. In short, theres a huge
amount of flexibility here.
Photo: Richard Ecclestone
The Pro 2s front panel measures 74.2 x 32.5 cm.
Turning to the filters themselves, you could describe these with no little accuracy as Prophety (Filter 1) and Oberheimy
(Filter 2) and, if you send a simple signal through one or the other in isolation, that gives you a reasonable idea of what to
expect. However, the real flexibility lies in the simultaneous use of both. Imagine passing one waveform through Filter 1 with
one combination of cutoff frequency, resonance, keytracking, contour and modulation parameters, and passing a second
through Filter 2 with a different profile and a different set of parameters; you can obtain two very different sounds, and the
possibilities for layering and otherwise combining them are huge. You can even create D50style SAS patches with one
timbre for the attack phase and a different one for the body of the sound. The only obvious limitation is that you cant push
Filter 2 into selfoscillation. Thats a shame, but not a problem.
The next stage in the signal path is an output boost that can drive, overdrive, or even distort the filtered signal before its
passed to the audio amplifier. This also has a dedicated contour generator that, at first, appears to echo that of the Prophet
12. But, while the velocitysensitive, looped HADSR structure is retained, theres a huge difference: where the polysynth has a
Pan parameter that allows you to spread its 12 voices across the stereo soundstage, the monosynth has another distortion
circuit that can add yet another level of overdrive to your sound. Interestingly, the tunable feedback loop following the amplifier
which takes the analogue signal, converts it to digital, and then injects it back into the Character effects section is also
different from before. When ringing, it will track the keyboard for another octave, and it has a cleaner and more defined sound.
But if you add drive, drive and yet more drive... well, if you like to listen to synths scream, this is one way to make them do it.
You can now pass the analogue signal unmolested to the final output, which comprises a fourth overdrive and a master
volume control. Happily, even at maximum distortion and level, the Pro 2 doesnt suffer from the oscillator leakage that made
its way to the outputs of the Prophet 12, and the signal/noise ratio is remarkable given the number of gain/overdrive stages
available. But if youre willing to allow the digital side of the Pro 2 to do its thing, the Delay section which sits between the
amplifier and the output section offers four, syncable delay lines. The legends on the control panel imply that three of these
are digital, whereas the fourth (by use of the name BBD) is analogue. In fact, all are digital, although DSI claims that the fourth
has been programmed to emulate an analogue delay. The manual states that the maximum delay time is 1s, but thats wrong;
the maximum for the BBD effect is 1.5s, and theres an additional Long/Short parameter in this to help you to create
modulation effects such as chorusing and flanging. Inevitably, the results lack the deep whoosh of an analogue stompbox so,
if you want the sound of the Pro 2 going through an Electric Mistress, youll just have to plug it into one. But imagine what you
can do with four delay lines that allow you to modulate all of their delay times, feedback amounts, levels and pans
independently, and which also offer independent LP/HP filtering for each! The potential for spatial effects is mindboggling.
Nonetheless, if for some reason you have access to only a single quarterinch cable, an unusual but very welcome mono
mode defeats the pan and stereo modulation settings so that a sensible monophonic signal emerges from a single output.
Modulation & Analogue Control

In addition to those already mentioned, there are two further contour generators that you can assign to the full set of the Pro
2s modulation destinations within their own programming pages. (Of course, all five contours are also freely assignable within
the modulation matrix, and well get to that in a moment.) There are also four LFOs, each offering control over rate, amount
and waveshape and, in addition to being available as sources within the modulation matrix, each can be directed to the
complete list of destinations from within their own pages. Each LFO is individually syncable to the internal clock or MIDI, and
you have control over slew rate, wave reset at the start of each note, and the phase of the waveform when reset.
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
This then brings us to the modulation matrix itself, which offers

16 freely programmable slots (source, amount and destination)
plus 25 described as fixed, although many of these still allow
you to determine the destination and/or modulation amount as
you choose. So, thats 41 slots offering a total of 51 sources
(many of which can run at audio rates) and 142 destinations,
which is considerably more than the Prophet 12. When you add
the destinations accessible directly from the auxiliary contour generators and the LFOs, the power of the Pro 2s modulation
architecture becomes readily apparent.
The four CV outputs on the rear panel of the synth are four of the destinations in the modulation matrix, and these suggest
all manner of exciting possibilities. But while Ive read eulogies elsewhere regarding the analogue I/O, I have to wonder if the
writers ever tried using it, because this was the one area in which the Pro 2 presented some unexpected difficulties.
I took my Roland SH101 and patched the Pro 2s CV Out1 and Gate Out to the SH101s CV In and Gate In, then directed
the Note Number to CV1 Out in the modulation matrix and played... and the notes I obtained from the SH101 were three
octaves higher than I expected. I measured the voltage produced by the CV output and, sure enough, it was 3V higher than I
had expected. What was going on? I found the answer in a DSI Technical Support document that explained that this wasnt an
error; the company had taken the decision to make middle C generate a pitch CV of 5V. So I used the modulation matrix to
direct a large, negative DC offset to the CV output driving the SH101. Actually, I needed to use two slots to bring the Roland
down to its usual pitch, but then all was fine.
Similarly, when I patched the CV and Gate outputs from the SH101 to the CV In1 and CV In4 on the Pro 2, setting the latter
to be its Gate source, I obtained a pitch from the Pro 2 that was approximately 14 semitones above what I expected. Again, a
suitable DC offset sorted everything out in a few moments, but I would have implemented the Pro 2 the other way around: set
it to generate conventional voltages as standard, and use the DC offset to push it into more esoteric ranges if needed. I
discussed this with Dave Smith and he confided, Weve had a handful of requests to add a Global parameter to shift the CV
ranges, and its a possible future feature. Thats good news.
I spent many days experimenting with the Pro 2s analogue connectivity, and I particularly liked its ability to scale its CV Ins
and Outs to match synths that dont track 1V/oct precisely. But not all was hunkydory. For example, once connected in the
modulation matrix, the CV inputs can affect parameter values even when theyre sitting at 0V. I found that a DC offset of 2
was needed to correct CV In1 and CV In2, whereas an offset of 10 was needed to correct CV In3 and CV In4. I contacted
DSIs technical support team, and they confirmed that this was normal. They told me that, There will be some offset inherent
on the CV inputs as we assume that a device driving the CV input will have some amount of its own offset as well. In this way,
one would likely correct for the overall offset on the external device or in the Pro 2 itself. This boils down to, weve allowed an
offset error within the Pro 2 because there might be one in the external voltage. I cant be alone in thinking that this is
unsatisfactory, can I?
Sequencing & Arpeggiation

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Although the claim that the Pro 2s is the most powerful step sequencer ever designed on a synth is possibly overstating the
case, the facilities provided by its 16, 16step tracks (which can also be configured as eight, 32step tracks) place it in the
premier division of onboard analoguestyle sequencers. Every patch contains a sequence with Track 1 connected to the
pitch of the note, and you can then direct every other track to any of the modulation destinations. All the usual facilities are
provided ties, rests, slew, legato or retriggered notes, variable-length sequences, MIDI sync, and so on and you can
transpose sequences, set them to loop continuously or play once through per trigger, or play one step each time that a trigger
is received. Recording is simple, and sequencing things such as filter sweeps is a doddle: just twist the parameters knob
while the sequence is running. (You can also determine values on a stepbystep basis if you prefer.) Once youve
programmed your patches and recorded their sequences, you can even build songs using Cue to switch between them,
whereby the current patch/sequence is completed before the next is initiated. Unfortunately, I have to point out a serious
oversight: you cant send triggers or gates directly from the sequencer. I think that DSI will have to add this if the Pro 2 is going
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
Software Synthesizer
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
to justify the companys claim that its a natural choice for the centre of your [analogue] studio or live setup.
and dirty...
Finally, we come to the arpeggiator, which offers Up, Down, Up/Down, Note Order, and Random modes that can be driven
over a one-, two- or threeoctave range, with each step playing up to a maximum of four instances per note. Theres also a
Latch, which works together with the Hold function to provide a further selection of useful results, and everything can be
synced to the synths internal clock or MIDI Clock. Just be aware that the arpeggiator doesnt transmit the generated notes via
MIDI.
Waldorf PPG
Wave 3.V
In Use
When I first received the Pro 2, I found that Filter 1 tracked correctly but Filter 2 didnt so, before proceeding further, I ran the
calibration routines to ensure that everything was at factory spec. I also downloaded and installed the latest OS. While I was
waiting, I studied the Pro 2s specification a bit more carefully, and started to speculate about the conversations that might
have led to its creation. I imagined a room full of engineers, each trying to outdo the previous...
Well, its got to have at least four types of filter, someone might have suggested.
Four oscillators and a suboscillator, someone else piped up.
And four LFOs, suggested the next.
And five contour generators, added another.
And scores of modulation destinations and a huge
sequencer!
And at least three feedback loops and four distortion stages.
Lets face it, our customers would be lost without at least three
feedback loops and four distortion stages.
And so it continued late into the night until the design brief
included everything that they felt we might want from a flagship
monosynth. The result was not only a powerful synthesizer, but a
rather beautiful one too. From the sleek walnut case and steel
chassis, to the fabulously precise OLED screen, to the almost
universally wobblefree pots and encoders, everything screams
Dont be put off by the Pro 2s OLED screen, its menu
structure is very shallow.
Program me!. Given that the Pro 2 also offers dual pressure
and positionsensitive ribbon controllers (which means that you
have six degrees of performance control under your left hand in addition to the 44note, semiweighted, velocity and
pressuresensitive keyboard under your right) it also commands, and, while youre at it, play me too! With four velocity
curves and four levels of aftertouch sensitivity to choose from, plus three modes of key assignment as well as single and
multitriggering, I have little choice but to do so.
In truth, there are so many nice touches to the Pro 2 that its impossible to discuss them all here. Nevertheless, its not a
difficult synth to use; if you want to twiddle knobs and press buttons to see what ensues, it allows you to do so. But if you want
to use it to its fullest, youre going to have to put in a bit of work, especially since turning a knob or pressing a button will often
reveal additional onscreen parameters. Happily, the menu structure is very shallow, so nobody but the most hardened
analogue nutter should have a problem with this.
So, what of the sound? With remarkably little aliasing (unless you ask the Character effects to generate it) the Pro 2 can
sound astonishingly analogue. It can also produce all manner of vintage (grainy) and modern (smooth) digital sounds. So
Im not going to delve into the clichs regarding screaming leads and thunderous basses, or all the other hackneyed phrases
used to describe monosynths, because its capable of so much more than that. Indeed, if you experiment with things such as
the modulation matrix, the character effects and the multiple overdrive/distortion and feedback stages, youll soon discover
powerful sounds that youd never obtain from any integrated vintage monosynth. Sure, there are limitations, but I was always
impressed by the versatility of the Pro 2. At times, I was tripping over complex, evolving and involving sounds that could have
come from the BBC Radiophonic Workshop in the 60s and 70s, and thats no small accolade.
So, finally, lets return to that niggling question about the relationship between the Prophet 12 and the Pro 2. Polysynths
almost always have simpler voicing structures than monosynths because hypercomplex sounds rarely work as well
polyphonically as they do monophonically. Consequently, it should be no surprise to find that the voice structure of the Pro 2 is
more complex (and more aggressive) than that of the Prophet 12. Does that mean that the Pro 2 is in some way better and
that you should therefore buy one in preference to a Prophet 12? The correct response to that is dont be daft. The Prophet
12 has facilities that the Pro 2s lacks and, despite their similarities, they were designed to do different jobs, and they remain
distinct in both sound and use.
Conclusions
Many prospective owners will view the Pro 2 as a sound designers dream, while others may find it too detailed and look
elsewhere for something simpler. But its not as complex as it seems, and youll soon be wringing everything from delicate,
PPGesque timbres to monstrous, overwrought sounds and sequences from it. Selling for little more than a wellpreserved
Pro One or Mono/Poly, it blows its ancestors away, and when you consider what it might cost to buy and maintain a vintage
synth and step sequencer, the Pro 2 begins to look like very good value too. In 20 years, I wouldnt be at all surprised to see
some youngster point at a Pro 2 and proclaim that they dont make em like they used to. .
Alternatives
The Moog Sub 37 is probably the closest competition for the Pro 2, and this also builds upon the reputation of a revered
ancestor while stretching its sonic and performance capabilities way beyond what was possible in the 1970s and 1980s.
Both are excellent synths with great strengths, and both will appeal widely. The Sub 37 is more immediate and has
bucketfuls of that classic Moog sound, while the Pro 2 is much more flexible and can be integrated into an analogue studio
in ways that the Moog cant. Both are top-quality instruments, and I recommend that you win the lottery and buy both.
Paraphonic Mode
Although primarily a monosynth, the Pro 2 echoes instruments such as the Korg Mono/Poly which, back in 1981, allowed
you to play up to four singleoscillator notes through its single VCF/VCA signal path. Happily, the Pro 2 is much more
flexible than that, because the amplitude of each note can be shaped individually by the amplifier contour generator, and
the tone of each can be shaped individually using tools such as waveshaping and sync. I programmed all manner of
analogue piano and Clavi-style patches as well as some luscious pads, ensembles, analogue choirs and spacey organs
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
using a combination of dynamic waveshaping, modulation and delay, and I suspect that few (if any) listeners would realise
that they emanated from a paraphonic synth rather than a true poly.
Nonetheless, I have three minor complaints in this area, although all are operational, not sonic. Firstly, I cant
understand why the waveform selector cant adjust the waves for all four oscillators simultaneously in paraphonic mode.
Secondly, theres no All Oscillators Shape Mod parameter in the modulation matrix. Thirdly, theres no way to sequence
fourpart compositions in paraphonic mode. Counterintuitively, you can record and play fourpart compositions in
monophonic mode by turning off Key Follow for oscillators 2, 3 and 4 and then assigning three other tracks to control their
pitches. But when I tried this, the results didnt justify the effort. DSI told me that theyre considering paraphonic
sequencing for a future update, so I think that Ill wait for that.
Although the Pro 2 offers 792 patches, half are factory sounds that can be edited but not overwritten. This means that you
have 396 memories for your own creations. Its still a generous number, but Ive never understood why manufacturers do
this. It also offers four playlists, each with 16 slots. These allow you to place your sounds in a desired order (which is a
boon on stage) without having to reorder them in the patch memory.
Round The Back

In addition to its quarterinch stereo audio outputs and the associated headphone output, the Pro 2 offers an Audio In that
accepts a wide range of signals and injects them into the internal signal path via the oscillator mixer. You can also derive
control signals from the external signal using the associated Gate extractor and Envelope Follower. But if no cable is
plugged into this socket, its Level and Gain controls determine the amount of final output signal thats internally fed back
into the mixer, emulating an old Minimoog trick used to thicken the sound or cause screaming chaos.
Inputs for a footswitch and expression pedal are also provided and, sandwiched between these and the audio I/O, youll
find no fewer than four CV Ins and four CV Outs as well as a Gate Out, all on 3.5mm sockets. I would have preferred
these to be quarterinch too, but DSI appears to be looking toward the Eurorack market here, as evidenced by its own
DSM01 filter module.
Digital control is provided by a bidirectional, classcompliant USB port (MIDI only, not audio) and three fivepin DIN
MIDI sockets labelled In, Out and Thru/Out2, the action of the last of which is defined by a Global parameter. As you would
expect, the Pro 2 offers extensive MIDI capabilities, and the largest section of the manual (nearly 40 pages!) is dedicated
to NRPNs which, as well as mapping all the controls, include such things as the values in each of the steps in each of the
tracks in a sequence.
Finally, theres an IEC socket for the synths internal, universal (50/60Hz, 100240V AC) power supply.
Special Waves
Introduced on the Prophet 12, the Special Waves were added because the engineers at DSI felt that they extended its
range of timbres in interesting ways. Their names such as Boing, Ahhh and Shrill are roughly descriptive (of the
waveforms, not the engineers) but their uses are by no means limited to creating sounds that go boing, voices that go
ahhh, or anything shrill.
However, the real power of the Special Waves lies in your ability to place three of them in a line and morph between
them, either playing the static waveform derived at any given point on the line, or modulating that position in interesting
ways. This can take the Pro 2 far beyond the range of sounds that you might expect from it. I even used the Decimate and
Hack effects to reduce the sample rate and word length, and then programmed the Pro 2 in paraphonic mode as a
filterless, fourvoice, 8bit wavetable polysynth, and imitated the legendary PPG Wavecomputer 360A, a hopelessly
unreliable synthesizer that I nonetheless bitterly regret selling. A synth nightmare or yet another accolade to the Pro 2s
flexibility? You decide.
Abridged Specification
Description
Hybrid analogue/digital monosynth with fourvoice paraphonic capability.
Performance
controls
Semiweighted 44note keyboard with velocity sensitivity and aftertouch.
Memories
396 factory plus 396 user memories, with 4 x 16-slot playlists.
Oscillators
Four, each offering 29 waveshapes and three noise colours, with independent waveshaping per
oscillator, plus crossmod, AM, and hard sync.
Pitchbend and mod wheels, plus dual, latchable, position/pressuresensitive ribbons.
One suboscillator derived from Osc1.
Filters
Low and high boost.

Sample-rate and wordlength reduction.
Saturation.
One resonant (selfoscillating) 24dB/oct LPF.
One resonant 12dB/oct statevariable LP/BR/HP/BP filter.
Feedback
section
Tunable bipolar feedback loop from the output of the amplifier section to the input of the Character
section.
Delay section
Four syncable digital delays with independent modulation for flanging/chorus/ensemble effects.
Output
effects
Analogue distortion.
Contours
Five HADSR envelopes with looping, assignable within the modulation matrix.
LFOs
Four with sync, phase offset and slew.
Mod matrix
Sixteen slots, 51 sources and 142 destinations.
Arpeggiator
MIDI
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
Memories
Character
effects
If you dont
like
Up, Down, Up/Down, Note order, Random.

Threeoctave range.
USB.
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
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
modules.
Cwejman
Synthesis
MIDI
Fivepin DIN In, Out, Thru/Out2.
Modules
32step/eight-track or 16step/16-track sequencer.

Additional
Stereo outputs.
VCO-2RM MMF-1
ADSR-VC2 VCA-2P
Part 2: We continue our
exploration of Wowa
Cwjeman's new range
of exclusive analogue
synth modules.
Headphones.
Cwejman VM1
External Audio In with Envelope Follower and Gate Extractor.
Analogue Voice
Module
Swedish designer
Wowa Cwejman has
built a reputation for
exclusive analogue
synths. Now he's going
modular, starting with
the VM1 Voice module...
Four x CV In (1V/oct).
Four x CV Out (1V/oct).
Four x Gate Out (10V).
Audio
Power
Internal 50/60Hz, 100V 240V PSU.
Weight
8.5kg.
Buchla 200e:
Part 2
Patchable Analogue &
Digital Synthesizer
PART 2: We
conclude our
look at synth
pioneer Don
Buchla's extraordinary
new 200e modular
synth.
Buchla 200e:
Part 1
Patchable Analogue &
Digital Synthesizer
PART 1:
Alongside
Bob Moog,
Don Buchla
is one of the founding
fathers of synthesis, and
yet much less is known
of him and his
instruments. With this
two-part review of
Buchla's latest synth,
and a history of some of
his pioneering work, we
hope to redress the
balance...
Analogue
Systems RS370
& RS375
Polyphonic Harmonic
Generator & Expander
Having built their
reputation on knobheavy modular synths,
British manufacturer
Analogue Systems
surprise everyone by
bringing out a menudriven additive synthesis
module! But don't worry
the optional expander
is covered with things to
tweak and turn...
Cwejman Sound
S1 MkII
Semi-modular
Analogue Synth
Swedish company
Cwejman have
recognised that there is
no way to build a cheap
semi-modular analogue
synth for mass-market
sale these days (if there
ever was). Enter the
premium, but
meticulously crafted S1
MkII...
Doepfer A100
Modular
Synthesis Modules
Since the launch of their
A100 modular synth in
the mid-'90s, Doepfer
have been quietly
adding modules to the
Fri 19 Dec 2014
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In this article:
Head In The Cloud
Clean & Simple
Polishing The Cloud
Get In
High Scores
Alternatives
DoReMIR
ScoreCloud Free
pros
When it comes to
generating readable scores
from freely played keyboard
parts, nothing can touch it.
Editing features and
notation options much
improved over the original
ScoreCleaner.
Manual note entry and audio
part analysis both work well.
Free to download and use!
cons
Drum notation still under
developed.
Wholly cloudbased unless
you get the Platinum
version, so requires an
Internet connection.
For advanced notation
tasks, or creating scores of
publishable quality, youll
need a conventional
notation package too.
summary
If you work with notation,
ScoreCloud will almost
certainly make your life
easier. And its free!
information
Silver version free; Gold
version $4 per month;
Platinum version $7 per
month.
www.scorecloud.com
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WebExtras
DoReMIR ScoreCloud
Music Transcription Software For Mac OS & Windows
Reviews : Software: Scorewriters
DoReMIRs unique program helps to bridge the gap

between performance and musical score and its free!
Buy PDF
Sam Inglis
hen we think of revolutionary developments in music technology, the ones that spring to mind are those that do
previously unheardof things to sound. Witness, for instance, the excitement that was generated by Celemonys
Melodyne DNA, which seemed almost to defy the laws of physics in its ability to perform polyphonic pitch correction.
For my money, though, one of the most impressive technological achievements of recent years is one that has no audible
aspects whatsoever. Any form of musical notation inevitably represents a drastic simplification of a musical performance. No
matter how many expression markings you add to a score, it still leaves a great deal to the interpretative skills of the player.
Over the years, notation packages such as Sibelius and Finale have got better at mimicking the interpretation that might be
applied by human performers, so as to make score playback more realistic. Theyve been less successful, however, in tackling
the reverse challenge.
Most notation packages, and of course all DAWs that support notation, will let you play or record a MIDI performance, and
represent the notes you play on one or more staves. What they wont do, as a rule, is analyse this MIDI data to sift the wheat
of the actual score from the chaff of the players interpretation. In other words, if you incorporate expressive devices such as
rubato into your playing, the chances are youll end up with an unreadable score full of odd note lengths. And if you cant
figure out what time signature your piece is in, and where the bar lines should fall, youre unlikely to get much help from your
software.
Distilling a readable and correct score from an expressive human performance is a formidable
challenge for any computer algorithm and its one that Swedish developers DoReMIR largely
DoReMIR ScoreCloud
solved in their ScoreCleaner software, reviewed in SOS July 2012
(www.soundonsound.com/sos/jul12/articles/score-cleaner.htm). As Robin Bigwood found in his review, ScoreCleaner really
could take freely played MIDI input and spit out a score that would, in the vast majority of cases, be very nearly spoton,
correctly identifying pickup bars, odd time signatures, polyphonic voices and so on.
If there was a down side to ScoreCleaner, it was that this core technological breakthrough was embodied in a software
package that was somewhat limited in other respects. If you wanted to create publishable score documents, extract parts, or
even add devices such as guitar chord symbols and repeats, youd need to export its results into a conventional notation
package to do so.
Head In The Cloud

Many of these omissions are addressed in DoReMIRs new ScoreCloud, which, though it incorporates much of ScoreCleaner,
is marketed as a new product rather than as an update of that program. The new name reflects a change of business model
on the companys part. Whereas ScoreCleaner was a conventional, paidfor program that worked with local files on your hard
drive, ScoreCloud is based on a subscription model; and unless you opt for the forthcoming Platinum version, it offers only
cloud storage, with no option to save your songs locally.
DoReMIR arent the first software company to make this shift, but it must be said that their pricing seems extremely
generous. The ScoreCloud Studio software is a free download, and you can also get a free ScoreCloud Silver account which
will allow you to store up to 50 songs without ever getting your credit card out. If you need more storage, $4 a month gets you
a Gold account which will give you an unlimited number of songs, and at $7 a month, even the Platinum version that supports
offline files wont break the bank. (Those who bought the original ScoreCleaner get automatically upgraded to ScoreCloud
Platinum.) Theres also an iOS app called ScoreCloud Express, which uses DoReMIRs new audio analysis engine to
transcribe monophonic melody parts recorded via your iPhones mic input. This can be synced to the full version of
ScoreCould running on your computer, for overdubbing and further editing.
Clean & Simple

Descriptions such as userfriendly and immediate are not ones that spring to mind with most notation software, but
ScoreClouds user interface is disarmingly straightforward. It consists of a main window displaying whatever score youre
working with, surrounded by three smaller panels. The panel running down the lefthand side is a browser, which displays the
contents of your ScoreCloud online storage. That to the right is contextsensitive: when you select an element of your score,
the upper part displays a graphical menu of possible actions that can be applied to it, while the lower shows help material
related to that element.
Perhaps the most important of the ancillary panes, though, is the one running along the bottom. This is called the Listener,
and its job is to capture MIDI and audio input. Assuming you
have some sort of MIDI device attached to your computer,
theres no need to set anything up or hit a Record button: you
The Listener captures raw MIDI data and groups it into
simply start playing (with ScoreCloud generating a basic piano
Snippets, which are then analysed to turn them into scores.
tone), and notes appear in the Listener. Press the space bar or
leave a long enough silence, and ScoreCloud will treat the notes
youve played as a selfcontained unit called a Snippet. In theory, all you need to do to turn a Snippet into a score is hit the
Analyze button.
Whats more, theory and real life actually coincide in ScoreCloud, and most of the time, what appears in the main score
window will be a recognisable and intelligently notated version of your performance. Theres really no need to employ the
forced, mechanical stricttempo playing youd need to get acceptable results from other notation packages; nor, for the most
part, do you need to tell ScoreCloud anything about the piece before you start. And if you do need to correct something, you
can almost always do so simply by rightclicking on it and choosing from the resulting contextual menu. For instance, I
sometimes found that the algorithm was reluctant to employ compound time signatures, so it would choose to represent what I
thought of as 6/8 as 2/4 with lots of triplets. Changing this is as simple as rightclicking on the time signature. Likewise, on the
rare occasions where ScoreCloud failed to correctly identify a pickup note, shifting the entire score left or right to rebar it is
trivially easy.
Polishing The Cloud

Back when we reviewed ScoreCleaner, its editing facilities sufficed to tidy up what was generated by the transcription
algorithm, but lacked many of the features youd expect to see in even a moderately capable conventional notation package.
Many of those holes have now been addressed, and the version of ScoreCloud I tested had full support for transposing
instruments, repeats (with variant endings if required), system breaks, slurs, hairpins and other dynamic markings, fermata,
mordents and trills, and even basic bowing marks for string parts. You can even notate quarter tones and double sharps/flats.
Theres no fine control over the graphical position of these additions, but in most cases I found that ScoreCloud did a good job
of placing them appropriately. In most cases theyre easy to find and use, too, though there are a few features (such as
inserting empty measures into a score) that for some reason cant be achieved by rightclicking, requiring instead a visit to the
Add More menu.
When it comes to tidying up, there are some wellthoughtout editing features that serve to make most operations relatively
painless. For instance, if you position the mouse pointer near, but not on, a note, and leftclick, a red box will appear
highlighting all the notes that make up that beat. You can then click on the dropdown Rhythm menu in the Actions pane to
choose from various alternate rhythmic phrases that might suit that selection of notes. In general, I found that I was always
able to correct the actual content of the score, but its not always possible to clear up the occasional graphical conflicts that
arise where lyrics, chords, and interpretation marks fight for space above and below the staves. Nor can you make major
changes to the way a score looks.
Get In
As well as realtime MIDI input, ScoreCloud offers new ways of entering score data. A Manual Input mode brings up a palette
of notes and rests where the browser usually lives, and lets you step through the score adding them either from the QWERTY
keyboard, a MIDI input device or by dragging and dropping. This is pretty streamlined and works well for the most part, though
using it for any length of time certainly brings home just what a boon the realtime transcription algorithm is.
ScoreCloud can also accept an audio input, and given a
suitably clear monophonic signal, will notate it as a melodic line
in a score. I tried it on a variety of sources singing, electric
guitar and synthesizer and it was extremely effective. On the
Mac version, it was also perfectly reliable, whereas on Windows
7, I often got an error message telling me my recording was
empty; this may be because although it supports ASIO,
ScoreCloud doesnt give you the option of selecting an audio
input if you have a multiinput soundcard. And oddly,
ScoreClouds audio transcription only works with audio recorded
directly into the program; theres no way to open an existing
audio file, though this is planned for a forthcoming update.
Also new in ScoreCloud is a drum staff, which is nice to see,
but pretty basic in its current implementation. If you use real
time entry to try to record or overdub a drum part, you dont hear
drums but piano; and manual entry can lead to awkward conflicts
between the drum staff and the rest of the score. For example, if
you try adding a drum staff to a score that is swung, its quavers
default to being straight, and theres no obvious way to get them
to conform to the overall triplet feel. And talking of swing, despite
the ease with which ScoreCloud can flip between representing a
swung score in triplets or standard quavers, its not currently
possible to add a note to the tempo indication at the start of the
score to explain to the performer that something should be
played with a swung feel. This, along with numerous other minor
improvements, is on DoReMIRs radar and may well be fixed by
the time you read this.
As well as realtime MIDI input, ScoreCloud can also derive

scores from monophonic audio parts (above) or the kind of
manual, stepbased data entry familiar from conventional
notation packages (below). The Actions and Help panes to
the right reflect the possible changes that can be made to
whatevers selected (in this case the crotchet that is shown in
red).
High Scores
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In terms of the types of scores it can represent, and the detail

with which they can be edited, ScoreCloud still doesnt really
compare with a dedicated package such as Sibelius. But thats
absolutely not the point. ScoreCloud is not supposed to be the
last word in sheet music publishing, allowing you manually to
position every dot and line with meticulous accuracy; nor is it
intended to provide complete control over massive scores and all
their subsidiary parts, or to convincingly play back a symphony
New in ScoreCloud is a basic drum notation facility.
using highquality sampled instruments. If you need those
features, you need a conventional notation program and since
ScoreCloud supports MusicXML export, theres nothing to stop you starting work here and then migrating your projects to
Sibelius or Finale later.
What ScoreCloud excels at is the basic process of transcribing a musical performance. If you need to create musical
notation and you work from the keyboard or indeed any other MIDI instrument theres simply nothing to rival the speed
and accuracy with which ScoreCloud will turn your playing into a readable score. Since we reviewed its earlier incarnation
three years ago, DoReMIR have done a great deal of work to make it more versatile, and I also found it much more stable
than ScoreCleaner. Whether you work in education, composition or any other branch of music where notation is needed, I
think ScoreCloud will save you time and effort. And since its free, theres absolutely nothing to stop you downloading it and
trying it out!
.

explained
Competitions!
S8 monitors
Win Munro Sonic Egg
150 (Global Except
Americas)
Win Yamaha White
Christmas bundle
(Americas Only)
Alternatives
There are several wellestablished notation packages for Mac OS and Windows, including Avids Sibelius, MakeMusics
Finale and PreSonuss Notion, but ScoreCloud is probably best seen as a complement to these rather than as a rival for
them.
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In this article:
Power Trip
Setting Up
In Use
Conclusion
Alternatives
Focal
Alpha 65 $798
pros
Dynamic, detailed, accurate
and balanced sound.
Large sweet spot.
Easy to set up.
Excellent value for money.
cons
Power On/Off switch is on
the back.
summary
A dynamic, detailed and
tonally balanced active
nearfield monitor that
delivers a professional level
of performance at a projectstudio price. Definitely well
worth auditioning.
information
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Focal Alpha 65
Active Nearfield Monitors
Reviews : Monitors
Focal speakers have long been coveted for their

performance, but have traditionally come with
professional price tags attached until now...
Buy PDF
Bob Thomas
here are very few monitor loudspeaker companies around who have the capability to design and manufacture the
drivers that are fitted in their cabinets. One of these rarities is Frances Focal, whose high reputation in the audiophile,
hifi, incar and studio worlds is due not only to their skills in loudspeaker development, but also to their expertise in
driver manufacture. That combination of competencies has enabled Focal to build highquality studio monitors at attractive
prices in their France factory.
Competitive though Focals previous studio monitors may be in terms of price and performance, these products are
positioned more towards the professional end of the market, with their cheapest current 6.5inch monitor (the CMS 65) listing
at 1140 per pair. Fortunately for those of us with more desire than cash, Focal have now released the new Alpha range of
active twoway nearfield monitors, at prices that are more in line with the budgets of home and small project studios.
The Alpha 65, with its 6.5inch bass driver, sits in the middle of the Alpha range, with the fiveinch Alpha 50 (398 a pair)
and eightinch Alpha 80 (558 a pair) on either side. Unlike existing Focal monitors, the Alpha range is assembled in China,
and their use of vinylcovered MDF for the speaker cabinet reflects one area of the value engineering involved in this move.
However, Focal wouldnt be Focal without a neat design touch or two, and the cabinets, which are larger and weightier than I
had expected, feature bolton sculpted side panels that effectively camouflage the simple rectangular box sitting between
them. I cant help thinking that a custom paint job could look fantastic on these panels.
$798 per pair.

Audio Plus Services +1
Twin frontfacing laminar bassreflex ports exit via a moulded section that
800 663 9352
extends back into the cabinet underneath the bass driver. Between these
www.audioplusservices.com ports youll find the Focal logo that is illuminated when an Alpha is active.
www.focal.com
The treble unit sits in the middle of a small waveguide, and the frontpanel
lineup is completed by the 6.5inch bass driver. The bass driver itself
features Focals proprietary rigid, lowmass Polyglass cone, and the treble
comes courtesy of the inverteddome tweeter that is a feature of all of
Focals professional loudspeakers. To help keep costs down in the Alpha
range, the tweeter is made of aluminium rather than the beryllium used in
the higherpriced units.
Power Trip
Amplification in the Alpha 65 comes courtesy of two highcurrent ClassAB
amplifiers: 70W for the bass and 35W for the treble. These amplifiers have
massive power supplies to deliver the transient response and dynamics
required in studio monitors, and also probably account for a good deal of the
weight. The amplifiers automatically enter standby mode (indicated by a
rearpanel LED) after 30 minutes without an input signal and start up again
as soon as an input is detected. Frequency response is quoted at 40Hz
22kHz (3dB) and, should you want to pump up the volume, the Alpha 65 is
specified as being capable of delivering 106dB SPL at one metre at which point you really ought to be wearing hearing
protection.
Balanced XLR and unbalanced RCA phono input connectors sit on the rear panel. Both are active at all times, which means
that you can connect two sources simultaneously to an Alpha 65. A slide switch boosts input sensitivity by +6dB if needed,
and twin shelving EQs with corner frequencies at 300Hz (6dB) and 4.5kHz (3dB) allow you to adjust the monitors
frequency response in order to help correct for any high and/or lowfrequency anomalies caused by either speaker
placement or room acoustics or to tailor the overall sound to your own requirements.
Setting Up
The supplied manual gives a comprehensive guide to setting up the Alpha 65 in equidistant configurations for both stereo and
5.1 applications. Essentially the advice boils down to ensuring that the treble drivers are positioned at ear level (even if that
means turning the Alpha 65 upside down) and decoupling the cabinets from their stands or shelves using the rubber feet
supplied. There is also a useful reminder that, in stereo use, the distance between the monitors determines the width and
coherence of the soundstage and that that distance will also determine the length of the sides of the equilateral triangle
between you and the Alphas which, in my studio setup, is approximately four feet (1.22 metres).
As always, running in is essential in order to allow the monitors to settle in

mechanically and to adapt to their new environmental conditions. Focal
recommend 20 hours of bassheavy music at a moderate level for starters.
The Alphas improved noticeably during this process, with the bass filling out
more and more as time progressed. Personally, I wouldnt go near the
shelving EQs until the Alpha 65 had run in to the point where you cant hear
any more obvious development in their sound, and even then Id tend to use
EQ very sparingly. However, since I could see myself using the Alphas
slightly bulky though they are for mobile recording work, having these
EQs available to help cope with the lessthanideal monitoring
environments that Ive found myself in over the years would be really useful.
The two separate sets of simultaneously active inputs on XLR and RCA
phono turned out to be a useful convenience feature, although, just to cater
for every eventuality, it would have been nice if Focal had fitted an XLR/TRS
combi connector instead of just an XLR. I used the XLR inputs for my
mixers outputs and linked the phono inputs to the Microbook II that I use
with my laptop. This arrangement meant that I could work on that computer
which I use mostly for transcribing music and recording rough ideas
without having to switch on the entire studio in order to hear good audio.
This setup also allowed me to refer to scratchpad ideas on the laptop as I
worked on a track without having to change anything on my mixer, which I
personally find to be extremely useful.
AVI Neutron
Five
The rear panel houses the XLR and RCA phono

inputs, as well as LF and HF shelving controls.
In Use
Monitors Demystified
Monitoring Tips
Monitors vs Hi-Fi?
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The first thing that I noticed when I settled down with the Alphas was the scale and solidity of the stereo image. Every element
of the soundscape had a real sense of clarity and position. An especially wellrecorded grand piano had a sense of reality that
Ive heard only in more expensive monitors. Vocals and other instruments including percussion were reproduced with
similar transparency and precision when recorded well, whilst any shortcomings were ruthlessly exposed by the Alpha 65
as they should be.
Bass frequencies were solid and wellcontrolled and there was no apparent lag or smearing due to the reflex port. Electric
and acoustic bass guitars and double basses were reproduced with great accuracy, as were bass drums, where both initial
attack and body were delivered with power, providing that they had been wellrecorded (which I didnt always feel was the
case).
Across the mid-range, reproduction was extremely detailed, and this gave vocals, acoustic instruments and percussion a
feeling of real presence. The higher mids and treble shared this precision, and the combination helped make lowlevel detail
such as reverb tails easy to make out.
The Alpha 65 also handled sampled, synthesized and electronic material in the same precisely delineated fashion, and
although youd probably want to add a sub to shake things up with really low frequencies, the Alpha 65 can provide more than
enough level and detail in the low bass to let you make mix decisions in the bottom end with confidence.
Conclusion
The Focal Alpha 65 delivers a wonderfully detailed, dynamic and tonally balanced sound within a sizeable sweet spot.
Although the overall performance is, to say the least, extremely impressive, to my ears the Alpha 65s aluminium tweeter
doesnt quite manage to deliver the seemingly effortless, smooth top end that I remember hearing from the more expensive
beryllium tweeter that features in Focals higherpriced monitor ranges.
Having said that, to me the Alpha 65 is a very highquality monitor that, not only, wouldnt sound out of place in any
professional studio, but also that I would have absolutely no hesitation in recording with and mixing on. Given their
performance and price, I think that Focals new Alpha range will give competitors in the home and project studio market
something to think about. In the meantime, if youre in the market for a pair of monitors at around the 500 mark, you really
have to take a listen to the Focal Alpha 65 I think youll be just as impressed as I am.
.
Alternatives
The project-studio active nearfield monitor market is not short of alternatives at the Alpha 65s price, and youll find
competition (although possibly with smaller bass drivers) from the likes of Adam, Eve, Event, Fostex, Genelec and KRK.
2.1 Monitor System

This
interesting
monitor
system uses
the natural roll-off of the
satellite speakers to
provide the crossover
with the subwoofer.
Tannoy Reveal
601A
Studio Nearfield
Reference Monitors
Building to a
price
inevitably
entails
compromises. The art is
in choosing the right
ones...
Quested V3110
Three-way Active
Monitors
Sometimes,
a dose of
oldfashioned
good engineering
delivers something well
worth listening to...
Adam A7X
Active Two-way Studio
Monitors
Their A7
nearfield
monitors
received
many plaudits, not least
in the pages of SOS, but
manufacturer Adam
thought there was room
for improvement.
PMC TB2S AII

Active Nearfield
Monitors
PMC broke
new ground
a decade
ago with
their TB2 monitors, but
the competition have
been catching up. Does
PMCs new activated
design nudge them back
to the front of the pack?
Avantone Active
MixCube
Secondary Reference
Monitors
Thumbnail Avantone
for article: Avantone Act
have added
on-board
amplification
to their contemporary
take on the classic
Horrortone secondary
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In this article:
The Theory Behind
AutoTheory
Theory & Practice
Zoning In
Two-fingered Salute
Conclusion
Alternatives
Mozaic Beats
AutoTheory $50
pros
Allows even non-musicians
to create credible MIDI
chord and melody parts.
At the current price, very
good value.
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Mozaic Beats AutoTheory

Intelligent MIDI Chord & Melody Tool For Mac OS & Windows
Buy PDF
Reviews : Software: Instruments
If you want to create convincing keyboard parts but dont

have the playing chops, AutoTheory might save your
bacon.
John Walden
any of us are not blessed with stellar keyboard skills. I include myself in this category: I can get a half-decent tune out
of a guitar, but when it comes to the piano keyboard, Im far less fluent. As well as having led me to try almost every
MIDI guitar system ever invented (they are always interesting and frustrating in equal measure), it has also meant
that Im a keen student of other tools that can assist me in creating MIDI performances in my DAW/sequencer projects.
A new option on this front is Mozaic Beats AutoTheory software. Available for both Windows and OS X, AutoTheory sits
between your MIDI keyboard and your sequencer and, with a little pre-configuration, can allow you to play complex chord
It might be nice to be able to sequences with a single finger, or constrain your melody playing to notes within a chosen chord or key/scale combination.
cons
define your own scale tones

for melody construction.
Some might say its
cheating!
summary
If you have musical ideas in
your head but lack the
keyboard skills to realise
them, AutoTheory can sit
between your MIDI
keyboard and MIDI
sequencer and help you
create them. A neat little
utility and, at the current
price, very good value.
information
The Theory Behind AutoTheory

Mozaic Beats describe AutoTheory as an open arpeggiation system, and it has two main functions. First, to the left of a MIDI
keyboard split point, you can map a set of chords for the currently selected key and then play any of these full chords simply
by pressing the appropriate single MIDI key. Second, on the other (right-hand/high-pitch) side of the keyboard split, the
software automatically adjusts the notes mapped to your keys so that they are confined to those available either within the
currently triggered chord or an associated scale. Within this zone, the note generated by a particular key changes as you
trigger different chords.
In practice, what this means is that you can leave the fingers of your right hand poised over the same five keys on your MIDI
keyboard and, based on the chord you trigger with a single finger of your left hand, whichever notes you hit with your right will
be harmonically correct given the current chord. No more fluffed chords and no more out-of-key melody notes. And this is, of
course, where the open arpeggiation system concept comes from, as you could play an identical finger pattern with your
right hand but the MIDI notes generated would change in response to the chords your left hand triggers. The end result will be
a perfect arpeggio pattern from your right-hand playing. You can, of course, vary your right-hand note pattern to vary the
nature of the arpeggio.
$49.99.
www.autotheorybymoziac.com And while I can hear a few cries of cheating from the piano-
Test Spec
Apple iMac with 3.5GHz
Intel Core i7 CPU, 32GB
RAM and Focusrite Scarlett
8i6 interface, running Mac
OS 10.9.5.
Tested with Steinberg
Cubase 7.5.20.
playing readership, I can tell you already that, for my guitarshaped fingers, when I apply AutoTheory to my keyboard
playing, it means I can work faster. For many, especially those
working to regular deadlines, that might be reason enough to
look further.
Theory & Practice

AutoTheory is a very small download (the OS X version I used
was about 14MB in size) and activation can be computer-specific
or to a second-generation iLok. I used the former and the
process was suitably painless. Incidentally, there is also a version
of AutoTheory available as a Reason Rack Extension. This is
configured slightly differently to the stand-alone version I used
here but the functionality is similar.
Once installed and authorised, AutoTheory simply runs
alongside your DAW/sequencer software. The only initial
configuration to be dealt with is to ensure that the MIDI
connection with your MIDI controller keyboard is disabled in the
DAW and set as the MIDI input into AutoTheory. In essence,
MIDI data from your keyboard is routed to AutoTheory and, once
remapped by AutoTheorys settings, the resulting MIDI data is
then passed on to the DAW/sequencer via any of the eight virtual
MIDI ports created by AutoTheory.
I had no configuration issues within Cubase on my test system
and there are simple instructions on the Mozaic Beats web site
that illustrate the setup process for a number of the other
common DAW packages. It should, in principle, work will almost
any MIDI sequencer, providing you can control which MIDI inputs
are active.
AutoTheory provides eight virtual MIDI outputs for your

DAW/sequencer to use, as shown here within Cubase.
Zoning In
AutoTheorys interface is split into five zones. The upper strip contains controls to set the key, scale and the mapping. The
first two are fairly straightforward, while the latter produces gradually more complex (flexible?) note mappings in the right-hand
melody zone of your keyboard. These range from restricting the notes to just those within the currently selected chord right
through to all the notes within the chosen key/scale; the choice is therefore yours depending upon the level of control you
want. There is also a preset system in the upper strip where you can save your own configurations.
The two boxes below hold additional settings for the Chord Generator (left hand) and Melody Lock (right hand). The Chord
Generator shows the mapping of the current chords across the keyboard so, if you hit a C note, then you get a I (tonic)
chord in the current key/scale combination, and so on. This pattern of chords is repeated as you progress further to the left of
the keyboard and the chords are generated in the appropriate octave.
You can choose to mute the MIDI output from the Chord
Generator, in which case the chord triggered here simply
controls what happens within the Melody Lock mapping. In
addition, the Root Separate switch allows to you send the
chords root note out to a separate MIDI destination. As
described below, the MIDI output for the root note can be
selected in the Chord Generator Output panel at the bottom of
the window.
The Melody Lock panel shows you how the combination of the
current chord and the Mapping setting places notes from the
scale on the right-hand split of your keyboard. As mentioned
above, the Mapping setting allows you to keep this mapping
either simple (if your keyboard skills and knowledge of musical
theory are a bit scratchy) or make it a bit more complex. The two
additional switches Inversion and Accidentals allow the
mapping to reflect any chord inversion or to include additional
notes if you have specified more harmonically complex chords
within the Chord Editor.
AutoTheory provides a number of global chord settings that

you can use as a starting point for building your own presets.
Speaking of which, the Chord Editor panel allows you to customise the voicing of the chords available in the Chord
Generator. The Global Chord Types setting provides a basic configuration for all chords, and a drop-down menu provides a
number of choices, from simple triads to seventh chords spread across two octaves. Once set, you can then edit any specific
chord simply by selecting it (press the appropriate key on your MIDI keyboard) and tweaking the Step, Accidental and Octave
settings for each note within the chord. Seventh, ninth, 11th and 13th chords can all be constructed and you can apply
accidentals to create more harmonically complex chords as required. Finally, there is a basic strum function where you can
adjust the duration and velocity of the strumming effect.
The two panels at the base of the display configure AutoTheorys MIDI output. You get up to eight virtual MIDI outputs, and
this means plenty of flexibility. For example, you can send the data to multiple virtual instruments to layer sounds, and you can
also apply an octave transpose to any of these MIDI outputs.
Two-fingered Salute
In practice, then, you can use AutoTheory as a means of playing chords, as a means of simplifying melody parts based on
chords, or to generate both at the same time. With the Mapping control set to Chord Tones its essentially impossible to hit a
duff note. While that perhaps means you dont have the freedom to create melodic lines that are more harmonically
interesting, it is still a satisfying experience for those with limited piano skills and for some musical genres, simple melodic
forms work best anyway.
At one level, AutoTheory really is a doddle to set up and use and, if you are suitably challenged by a piano keyboard, when
it comes to just knocking out a few chords and a basic melody line whether via some trance synth or a grand piano it is
remarkable just how far two fingers and a bit of AutoTheory assistance can get you.
However, at another level, you can use the various configuration options to make that process produce something more
harmonically complex. Each of the Scale setting options produces a shift in the set of chords available, while with the Mapping
setting you can reduce the extent to which AutoTheory confines your melody lines.
While Im sure you could find all sorts of ways to put this technology to good use, in my own experiments with AutoTheory I
had a lot of fun with three applications in particular. First, with a simple acoustic piano sound, it was rather too easy to create
some ballad-like chords and a perfectly in tune melody line. Second, with a suitably massive dance-synth patch following the
Chord Generator and a similarly full-on lead sound driven by the Melody Lock, crashing out a few club-ready dance tune ideas
was a breeze. Finally, while it took a little longer to configure, with the different MIDI outputs feeding a set of orchestral string
sounds, and set to suitable octaves for those sounds, I managed to coax a pretty decent full string section performance into
life using just two fingers!
Those with better piano skills than me (thats almost everyone, then) might perhaps yearn for some additional options in
terms of defining custom scales for the Melody Lock section, but AutoTheory does exactly what it sets out to do: it brings
harmonically correct keyboard parts into the reach of almost anyone.
Conclusion
Almost everywhere you look in music technology, there are products that help you create a musical performance that is
somehow better than you might otherwise have been able to, whether this is in terms of audio quality or performance.
AutoTheory is a tool in the same mould as pitch-correction software or automatic drumming plug-ins: it enables non-keyboard
players and even non-musicians to create musical performances that are technically and harmonically correct, and it
does it in a very straightforward and accessible fashion.
Now, whether you think that is a good thing is another matter. I could, and perhaps should, be spending more time
practising my piano skills but, at the time of writing, AutoTheory is currently available for $49.99 (about 30). This is about the
price of a single, one-hour piano lesson with a decent tutor. For my own musical needs, I know which will produce faster
results.
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Some musical background does help in using AutoTheory but, providing you have the music in your head, whatever your
level of technical (in)competence on the piano, AutoTheory can help you realise those ideas in your MIDI sequencer. While
there are alternatives available, AutoTheory is a very useful little tool.
.
Alternatives
There are alternatives to AutoTheory, and a number of the major DAW/sequencers include some sort of auto chord
feature. For example, Cubase includes the Chorder MIDI plug-in and, as explained in the November 2013 Cubase
workshop, you can also use the Chord Track to transform your random keyboard doodling into something more coherent.
Both of these are, however, more complex to configure than AutoTheory.
Perhaps a closer comparison is with the various MIDI performance apps that are available for iOS (and which can send

explained
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MIDI data to your desktop virtual instruments) such as Chordion, ChordPolyPad or SoundPrism Pro. All of these allow you
to trigger full chords using a single finger and each also offers the ability to add melody or bass lines that are harmonically
correct. The apps themselves are pocket-money priced but you do, of course, also need access to a suitable iOS device
such as an iPad or iPhone.
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In this article:
Dyn Be The Glory
Power Games
Testing
Opinion
Alternatives
Dynacord
PM502 935
pros
A lot of functionality in a
small space.
Good technical
performance.
Compact and light.
USB-stick music playback
and USB stereo recording.
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Dynacord PM502
Powered Mixer
Buy PDF
Reviews : Mixer
Dynacord apply their typically thoughtful design to a

compact desk, which could be just the ticket for small gigs
and installations.
Paul White
sually, whenever a Dynacord product turns up for review I have to ask for help to lift it out of the box, but not in this
case. The PM502 is a small and lightweight powered mixer designed to meet both installation and mobile musician
requirements but, typically of Dynacord, it has a number of novel features, not least a hybrid analogue/digital
architecture and the ability to feed 100V line systems for installation use.
The main housing is moulded from plastic, and the front panel is angled back slightly for optimum visibility when used on a
desktop. Overall it measures 395 x 244 x 241 mm and weighs just 4.9kg. All the connections, other than the headphone jack
Most users will be paying for (irritatingly a consumer 3.5mm type) and a memory-stick USB socket, are on the rear panel, which sits above the power
some features they dont
bulge holding the electronics. Optional extras include a footswitch to bypass the internal effects and a rackmounting kit.
cons
need.
Needs a case for protection
in transit.
No pan controls.
summary
This is a very flexible and
compact mixer but it may
appeal more to the
installation market than to
gigging musicians.
information
935 including VAT.
Bosch Security Systems
+44 (0)800 1695739
www.dynacord.com
As far as the mixer is concerned, the front end is analogue with conventional and familiar knobs and functions. However, the
master section is digital, and hosts a flexible effects section, before feeding a fan-cooled, dual-channel Class-D power
amplifier capable of delivering 450W (continuous sine wave) per channel into a 4 load. The amp can be fed from any of the
three outputs (of which more later), providing the option to use the internal amps to feed passive stage monitors and one of
the other outputs to feed a main amped speaker system, or vice versa.
Dyn Be The Glory

There are five mic/line channels with combi XLR/jack inputs, each kitted out with a compressor/limiter and talk-over function,
the latter ducking the line inputs to give the mic channels priority. Four depths of ducking may be selected via the DSP menu.
Note that, unlike some mixers, the compressor settings (again accessed via the DSP menu) affect all the mic/line channels
when active. Phantom power is also switchable globally. The maximum microphone gain with all controls maxed out is 65dB,
and a mic amp EIN of -129dB (A-weighted) is specified.
Three further line channels, each fed from a pair of quarter-inch jacks, are available, and there are also two sets of RCA
phono connectors labelled CD1 and CD2, plus a USB port on the front panel that allows the direct playback of stereo MP3 or
WAV audio files from a USB stick. These three inputs share the existing stereo line-in channels. Overload warning LEDs are
present on all channels, while the line inputs also have dual-sensitivity gain switches. A further USB port on the rear of the unit
allows for recording of the stereo output to a computer.
Conventional rotary controls are provided on all channels for input gain, the three-band EQ, effect send, aux send and
volume, but everything else is controlled from the master sections Parameter turn and press control knob and its LCD
window. Above this are four small panels providing further status information when illuminated. Unusually there are no pan
controls all the mic/line channels are centred, and the stereo inputs panned hard left and right.
The EQ section comprises high- and low-shelving filters at

12kHz and 60Hz, respectively, and a fixed mid bell filter
operating at 2.4kHz. The shelving filters have a 15dB range and
the mid a 12dB range. In the master section the graphic EQ
frequencies are set at 63Hz, 160Hz, 400Hz, 1kHz, 2.5kHz, 6kHz
and 12kHz, each with a 10dB range. The graphic EQs can be
set to a mode called Sliders Link in the menu, whereby the onscreen faders follow the position of the physical faders.
Additional low-cut filters can be inserted into the mic channels,
also via the DSP menu, with a choice of 80Hz, 18dB/octave or
150Hz, 12dB/octave responses, and there are further global lowcut filters in the amplifier path that can be set to 50 or 70 Hz
(both with 18dB/octave slopes). Each channel has one aux send
and one dedicated effect send routed to the internal effects.
Including the aux, there are three line outs in total, any of
which can be internally routed to the on-board Class-D
amplifier.
Between the mixer channels and the master section is the

effects strip, which hosts the effect level control and effect-to-aux
send, as well as an effect on/off button. Above this are three buttons for global compression on/bypass, talkover, and mute,
which silences all five mic/line channels at once.
Power Games
Both conventional speaker outputs (4 to 8 ) are on Speakon connectors, while the 100V line Direct Drive speaker outputs
are on a connector block. When not in Direct Drive mode, this connector block is in parallel with the Speakon outs. The two
internal power amplifiers can also be split and used for different audio signals, as routed from the master outputs, making it
possible, for example, to run a mono PA and to power a passive stage monitor at the same time.
In the master section, the on-board effects provide 32 editable

presets and include the necessary delay and reverb options that
are a mainstay of live performance, as well as a few chorus
effects. The number of editable parameters is small but
adequately flexible to meet most requirements. However, theres
no tap-tempo delay button, which is somewhat remiss given how
useful such things are. LPN speaker processing, standing for
Low-Pass-Notch filters, is also accessed via the DSP section,
and is claimed to correct the frequency and phase response of
the connected loudspeakers. According to Dynacord, the LPN
filter optimises the transient response of the connected speaker
to produce a more powerful sound from small speakers, but there
are few clues as to how this is achieved. However, it does seem
to help to maintain clarity.
There are three line-level outputs, on TRS jacks using ground-compensated balancing. Each has its own seven-band
graphic EQ (adjusted by a set of familiar graphic sliders in the master section) and an adjustable peak limiter.
Master A, which can be set to mono or stereo, is fed post the master A volume control and may be used to feed the internal
amplifiers, external power amplifiers or powered speakers. Master output B, switchable pre or post the master volume control,
has a delay option, adjustable up to 150ms, for use in distributed speaker systems, but it may also be used to feed stage
monitors or a subwoofer. Similarly the aux output can be sourced pre or post fader and may be used to feed monitors. By
default this is fed from the aux bus, but the routing to all the master outs and the input source for the internal amplifiers can be
changed via the DSP menu.
Comprehensive amplifier protection is built in, and a yellow indicator warns that the amplifiers are running at their limit. The
limiter warning LED also lights if the amplifier reducers its output due to overheating or a shorted speaker. The technical spec
is impressive, with a 15Hz-22kHz response (-3dB) from any mixer input to any mixer output. The headroom is generous in all
sections and the noise and distortion figures are comparable with those of other well-designed mixers.
Testing
Operationally the mixer is very straightforward, and even the digital section can be explored in a few moments without
recourse to the manual. Having three outputs (if you include the aux), each with individual graphic EQ, could be useful in
multi-speaker setups, as could the delay facility. Menu options include the channel filter, compressors and ducking, plus the
ability to set a low-cut filter in the output. Other pages access basic setup functions including the option to set an anti-tamper
password.
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On the whole the mixer behaves extremely well, delivers plenty of clean power, and the effects are well suited to live
performance without being over-complicated though having a three-band EQ with a fixed mid might be considered a little
restrictive. On an ergonomic note its worth pointing out that the mixer needs to be transported in a solid case as theres
nothing to protect the front-panel controls. The front panel itself is also a little glossy, which might be an issue under some
lighting conditions.
As with many DSP multi-function sections, the extra features are welcome but the means to access them can be a little
slow. For example, the transport controls for the MP3 playback are on the screen so any precision stopping and starting relies
on you being on the right page and being quick with the cursor controls. That said, the MP3 playback worked with no fuss. I
fed it a FAT32 formatted memory stick with a few MP3 songs on it and it read them right away and displayed both the titles
and the elapsed playing time. Similarly, the USB recording worked right off, appearing as a generic USB interface in Logic Pro,
allowing recordings to be made at 32, 44.1 or 48 kHz sample rates, at 16-bit.
Opinion
Depending on how you look at it, the Dynacord PM502 might seem to be an expensive way to buy a handful of mixer
channels and a power stage or it could be the perfect solution for your needs. On the one hand you could buy a pair of
powered speakers and a small mixer with effects for around the same price, but in an installation situation all those extra
features really come into their own, especially the additional master outputs, the ducking and the 100V line-drive capability,
which is often used in churches or other locations where multiple speakers need to be strung out over a long distance. DJs will
also find the ducking useful.
Theres no denying that Dynacords engineers really think their designs through and then deliver a solidly engineered
solution, but for mobile music use, youd definitely need a case for this mixer to protect the knobs. Properly protected,
however, the PM502 should give many years of reliable service, and if you already have some good passive PA speakers,
then the format makes a lot of sense providing there are enough input channels to meet your needs.
.
Alternatives
Powered mixers are available from a number of manufacturers including Studiomaster, but none that I know of include a
100 Volt line output option, as the PM502 does.
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In this article:
The Nest Generation
Egg Box
Alternatives
Munro Sonic
Egg 150 $2999
pros
Very nicely engineered.
Compact yet not at all smallsounding.
Excellent sound quality;
detailed and three
dimensional without being
harsh.
cons
The cost is a bit on the high
side for home studios.
summary
The new Eggs sound very
similar to the first generation
other than for some subtle
improvements to the
electronics in the amp pack.
Be assured that the shape
of the speakers is not just a
gimmick.
information
$2999 per pair.
Fingerprint Audio +1 512
847 5696
www.fingerprintaudio.com
www.sonic
distribution.com
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Munro Sonic Egg 150

Active Monitors
Reviews : Monitors
Formerly sold under the SE brand, these Andy Munro

designed Egg speakers have been given a major update. So
whats new?
Buy PDF
Paul White
f this monitor system looks somewhat familiar, thats because its first incarnation was launched over two years ago. The
concept for the Egg speakers was hatched by acoustic designer Andy Munro, James IshmaevYoung of Sonic Distribution
and SE Electronics owner Siwei Zou. Andy Munro is well known for his commercial studio designs, and also his previous
involvement with Dynaudio loudspeakers.
This new model adheres to the original active speakers with external amp pack paradigm, but behind the scenes there
have been a few significant changes not least that manufacturing has now been moved from China to the UK, under the
Munro Sonic name, with both cosmetic and circuitry upgrades to the power amplifier section to further reduce noise and
improve performance.
The Nest Generation

As with the original Eggs, the cabinet is a ported, eggshaped, 14litre moulding. The shape provides both mechanical rigidity
and a significant reduction in diffraction, as there are no sharp cabinet edges. The curved surfaces also break up internal
reflections and suppress standing waves, so despite the technical challenges in manufacturing this type of cabinet, there are
many potential benefits. The reflex port is tuned to 51Hz, and the overall frequency response of the speakers is 45Hz to
20kHz at the 3dB points. As before, there are recessed LEDs that are visible only from the sweet spot, making it easy to
physically align the speakers, which come with their own adjustable bases.
Handling the highs is a oneinch silkdome tweeter with a
neodymium magnet, while the lows and mids are delivered via a
165mm polypropylenecone woofer. As far as I can tell these drivers
are the same as those fitted to the original Eggs the most noticeable
changes are in the amp pack.
Putting the amplifiers outside the speakers is a departure from the
activespeaker norm for speakers of this size its a configuration
most often seen in largescale studio monitor systems but there are
good reasons for doing it. As well as avoiding potential problems due to
mechanical vibration, this approach means the user can put the
controls close at hand. However, a separate monitor controller may still
be necessary as there are no dim or mono switches on the amp.
All the active crossover circuitry (2.1kHz) and the four 50W amplifiers
powering the two biamplified channels are located in this stylish case,
which then connects to the speakers via the included threemetre
heavyduty Speakon speaker cables. Though the amplifiers may seem
modestly powered when compared with other active monitors, they
pack plenty of punch, with their 35V power rails providing adequate
headroom and allowing a maximum SPL, per pair of speakers and at
one metre, of 118dB.
Egg Box
All the circuitry is traditional analogue, an approach Andy Munro has taken to avoid problems being introduced by digital
conversion systems and digital volume controls. Its also my own experience that many DSP-controlled monitors are
inherently noisier than their analogue counterparts. The dual-power-supply design is also linear rather than switchingmode,
and protection against overload and short circuits is built in. Extra thermal protection is provided for the tweeter in the form of
an automatic cutout that resets automatically when safe conditions are restored.
As before, the control panel is pretty uncluttered, but the seriously chunky metal fascia is noticeably thicker and the knobs
have been redesigned. Input switching for swapping between two sources (aux and main) has been retained, where the inputs
can be on either balanced XLR or unbalanced RCA phonos but theres no jack option, something that could have been
redressed by using combi input sockets.
Separate left- and right-channel LF and HF EQ trim pots are included on the rear panel for matching the speakers to the
room (up to 10dB of cut at 63Hz and, for the highs, +1 to 5 dB at 10kHz), plus theres a large midband control on the front
panel, originally intended to emulate the midrange hump of typical hifi and NS10type speakers. This can be switched flat
or set to introduce a preset amount of cut or boost at around 2kHz. In this new incarnation, the mid-EQ characteristics have
been tweaked slightly to have a wider response, but the purpose is the same and the amount of cut or boost is quite subtle.
Both main and aux inputs have large level controls, and theres a rearpanel slide switch to activate or kill the speaker sweet
spot alignment LEDs. A power switch with blue LED rings occupies the centre of the panel.
Sonically the Eggs come over as easy to listen to, with smooth highs and mids complimented by a solid but not hyped bass
end. The first impression may be that the sound lacks a little definition when heard alongside other active monitors, but then
you soon realise thats not the case the highs are all there, but they just dont sound aggressive. You can hear the same
thing with other topoftheirgame monitors from the likes of ATC, PMC, Neumann, AVI and so on theres an initial
impression that something is missing, until you realise that whats actually missing is the unwanted highfrequency distortion
and phase smearing that makes some budget monitors sound unduly forward. The Eggs also sound like much bigger
monitors than their appearance suggests, with confident lows that dont suffer from flabby overtuning.
Within their price range the Eggs tick all the right boxes, with great stereo imaging, a good sense of depth perspective, an
opensounding clarity and a solid, tight bass end. They make good mixes sound great but reveal every flaw in imperfect ones,
just as an effective monitor should, and though the cost has increased slightly compared with the original Eggs, the amplifier
pack improvements and the shift to UK manufacturing more than justify it. Maybe the Eggs cant guarantee an albumin the
charts, but they certainly wont get in your way!
.
Alternatives
In terms of quality these monitors are up against the best in class, which includes models from ATC, AVI, Genelec,
Neumann and PMC.
AVI Neutron
Five
2.1 Monitor System
This
interesting
monitor
system uses
with the subwoofer.
Tannoy Reveal
601A
Monitoring Tips
Monitors vs Hi-Fi?
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Studio Nearfield
Reference Monitors
Building to a
price
inevitably
entails
ones...
Quested V3110
Three-way Active
Monitors
Sometimes,
a dose of
oldfashioned
good engineering
Adam A7X
Monitors
Their A7
nearfield
monitors
received
manufacturer Adam
for improvement.
PMC TB2S AII

Active Nearfield
Monitors
PMC broke
new ground
a decade
ago with
PMCs new activated
Avantone Active
MixCube
Secondary Reference
Monitors
Avantone
have added
on-board
amplification
take on the classic
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Native Instruments Action Strikes

Kontakt Instrument
Dave Stewart

Like its predecessor Action Strings, the Action Strikes percussion library is a collaboration between
software giants Native Instruments and the production company Sonuscore, a division of computer games music specialists
Dynamedion. Todays games market demands cinematic production values and a big, Hollywoodstyle sound, so with that in
mind, Action Strikes readymade phrases are designed to bring instant orchestral drama to music scores.
Following a welltrodden path, the library combines Japanese taikos (performed by the Dusseldorfbased Wadokyo
ensemble) with standard orchestral percussion, whipping up the ingredients into readytogo rhythm patterns which obligingly
sync to your host tempo. The instruments were recorded from two perspectives: the close-miking contains a fair amount of
hall ambience of a very agreeable kind, while the room position sounds more distant.
As well as taikos of all sizes, the library contains African and
Indian drums, ethnic and hand percussion, rock bass drum, hi
hat and cymbals, concert toms, metals, gongs, shakers and even
a waterphone, which contributes some eerie squeaks, slithers
and groans. No timpani or tuned percussion instruments are
included, but thats excusable in a library designed primarily for
rhythm generation.
Action Strikes percussion ensembles pound out exciting,
repeated twobar rhythms capable of many variations: the
rhythm patterns comprise simple eighthnote figures, more
complex 16thnote patterns, triplets and odd time signatures.
Swing feel can be added via an adjustable onscreen control.
Featuring various combinations of drums, shakers and metals
and sporting names like Heroic Drums, Monster Attack and
War Ensemble, the patterns are thunderous, propulsive,
dynamic and (as it says on the tin) optimised for action scenes.
Digging deeper, each ensemble has a Low, Mid and High
section which can operate independently, enabling you to mix
and match patterns the high section houses lighter, quieter material. Each loaded pattern has five rhythmic variations and
a pair of single hits so you can program your own fills. Further onthefly variations can be triggered by a set of five
keyswitches, four of which vary the number and position of accents in a bar, while the fifth triggers a short ending lick.
The librarys Instrument and Hits patches open up more options. The first features individual single instruments playing a
full set of rhythm patterns, single hits, flams and an excellent set of single and multiple grace notes, while the latter contains
12 useful singlehit multiinstrument combinations with individual controls for each sample.
No timestretching is used in Action Strikes: rather than recording real performances, the loops were constructed from
multidynamic single hits, which means you can drastically increase or decrease the tempo without affecting the samples
pitch or sound quality. This accounts for the librarys relatively small (3GB) footprint; it also explains why the rhythm patterns
loop perfectly and feel absolutely great, making this brilliantly executed library a highly effective compositional tool. Dave
Stewart
Audio Example
$299
www.nativeinstruments.com
.
Fri 19 Dec 2014
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In this article:
Making An Impression
Browser & Navigation
In DAWs
Light Guide & Smart
Play
The Komplete
Solution?
Conclusion
Alternatives
Komplete & Maschine
New In Komplete 10
NI
Komplete Kontrol S49 $599
pros
Turns Komplete into a
unified sound source and
instrument.
Full integration with
Maschine (with v2.2
update).
Light Guide is both beautiful
and incredibly useful.
Scales and Chords modes
are endless fun and
inspiration, especially paired
with...
...the arpeggiator!
DAW control with hardware
track/instance switching in
Live, Cubase, Nuendo and
Logic.
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NI Komplete Kontrol S49

Controller Keyboard
Reviews : MIDI Controller
NI turn their hybrid expertise to keyboard design with an

innovative controller for Komplete and Maschine.
Buy PDF
Simon Sherbourne
I have more or less defined the shape of modern hybrid music production and performance with their Maschine and
Traktor Kontrol ranges. The Komplete Kontrol SSeries is the obvious next step, bringing a hardware frontend to the
mighty Komplete instruments bundle, and a high-quality keyboard option for Maschine users or NIcentric users of
other DAWs.
Youve probably already figured this is more than a MIDI control keyboard, so what is it? First, its not Kore reincarnate, at
least not entirely. Komplete Kontrol focuses on one aspect of the Kore vision: providing a unified user experience across all
NIs many and various software instruments. All the presets from Komplete are accessible from a single browser, with plugin
parameters premapped across the eight rotary encoders. The hardware also offers some special performance sauce from
the arpeggiator and autoscale/chord features, and intelligent integration with many DAWs.
Making An Impression
Out of the box the Kontrol S is a pleasingly narrow, low-profile slab of black. Feel and build quality are similar to Maschine
solid and sleek, with the same mixture of glossy plastic and brushed-metal surfaces. The keyboards panel couldnt be more
different than most of its contemporaries. Rather than packing every square inch with knobs, sliders, button and pads, all the
SSeries keyboards have just eight main rotaries, two modest clusters of buttons, and a selector knob.
While nicely crafted and classy in appearance, the KontrolS is

rather unassuming... until you switch it on. With power (AC not
USB, Im afraid) a wave of colours washes across the surface as
the Light Guide key lights come to life causing grins and oohs
from geeks and nongeeks alike in our house. The touch
cons
sensitive knobs offer just enough resistance to feel expensive,
No simple patch sorting,
and each is accompanied by a beautifully sharp monochrome
favouriting, or saving.
display. These displays show mapped parameters, with values
Komplete Kontrol software
indicated by a horizontal slider graphic. In place of pitch and mod
will not host individually
wheels youll find touch strips with LED position indicators. This
purchased NI plugins, or
might not suit traditionalists, but I liked being able to jump directly
third-party instruments.
to a value, and the option to set pitchbend springiness. The
Expensive.
Limited DAW control in non OEM Fatar keys befit the price tag, with a solid fast semi
Mackie Control hosts.
weighted action. I tested the midsized S49 version.
summary
The SSeries keyboards can function as standalone MIDI
The ultimate keyboard for
controllers, with the Controller Editor software enabling you to define the knobs and touch strips, and create colourcoded
Maschine or Komplete
users.
keyboard splits. Nice. The rest of the surfaces buttons and all the advanced functionality are only available when paired
with Komplete or Maschine (see the Komplete & Maschine box for more on Maschine integration). Outside of Maschine the
information
software component is Komplete Kontrol, which runs as a standalone app or a VST/AU/AAX plugin. Komplete Kontrol hosts
Komplete Kontrol S25
all your Komplete instruments, presents a unified preset browser, and also drives performance features like the arpeggiator.
$499, S49 $599, S61 $699.
Each instance of Komplete Kontrol hosts a single instrument plugin, which by default is shown in a simplified panel view,
info@native
usually focusing on any macros that link to the physical knobs. There is also an extended view where youll find all the plug
intruments.de
www.native
instruments.com
ins controls, and an Edit view which gives you access to things
like Kontakt or Reaktors file systems.
A Komplete Kontrol license is sent to you if you have Komplete
9 or 10 activated in your NI account. Thats right: the keyboards
are purely for Komplete; individually purchased NI plugins are
not eligible, and Komplete Kontrol does not host thirdparty
plugins.
Browser & Navigation

Core to the Komplete Kontrol experience is the browser. In the
software this is the same as the Instrument tab in Maschines
browser, allowing you to search for presets across the entire
Komplete bundle, or within specific products or categories. You
can further filter by Type and Mode then browse the list of
The Komplete Kontrol browser overlay screen.
patches that fit all your criteria. On the hardware, the righthand
button cluster is dedicated to mousefree browsing. The Browse
button opens a graphical overlay on the computer screen, providing instant access to the browser in the current instance of
Komplete Kontrol, regardless of whether the plugin window is open. You then use the four cursor buttons to drill down
through the browsers categories, and the large, clickable knob to scroll and choose from lists. At any point you can enter the
main results list and choose a preset. Once youve selected a sound, pressing Browse again brings you back to the patch list,
but you can also step through patches at any time with the Next and Previous Preset buttons probably the most used
buttons on my S49 during testing!
While effective, the browser navigation system took some practice. In particular, its not intuitive to move back and forward
between the cursors and scroller. I found myself wishing the scroller was a joystick as well. The scroller also takes a bit of
cranking to move around fast some acceleration or even inertia would help zip through longer lists. One other request
would be a button (or Shifted button) to open the current Komplete Kontrol plugin window. This is v1, though, remember, and
the interface will likely get refined further. Overall the hybrid browser and premapped controls work well at making the
Keyboard+Komplete combo feel like a unified workstation.
In DAWs
To run Komplete instruments in your DAW you insert the Komplete Kontrol plugin and open them from there. In Ableton Live,
Komplete Kontrol is inserted as an Instrument Rack device supplied by NI. In all DAWs you get full control over the plugin
from the hardware; in a few you also get certain Advanced Integration features: transport control, track selection, track auto
focus, and instance switching. When available, the dedicated transport buttons on the keyboard will operate your DAW. Track
selection lets you use the cursor keys in the Navigate section to move between tracks. With the autofocus feature, the
keyboard will automatically take control of any instance of Komplete Kontrol on the selected track. If there is no KK plugin on
the track, the keyboard switches to standard MIDI control mode ready to play other instruments.
Live arguably gets the best integration, supporting the S
Series directly via a remote script. In Logic, Cubase and Nuendo
you get all the advanced features via Mackie Control support.
This works great, but will likely be problematic if you have
another MCUbased controller attached. Most other DAWs will
get transport control only, again via the MCU protocol. This
rather leaves Pro Tools out of the party, as it only supports HUI
and Hypercontrol.

explained
While there are obvious advantages to having all of Komplete

in a single plugin, I cant help being wary. I got burned when NI
walked away from Kore: I have v1 and 2 Kore controllers lying
dormant in a drawer, and a large number of unfinished Live
Komplete 10s new Kontour synth.
projects that I can no longer open because I hosted all my plug
ins in Kore (I know, I should work faster!). In Pro Tools, plugin
hosts can also be a pain for automation, as the names of any automated parameters in the hosted plugin become generic
numbers. So theres a certain amount of trust involved in committing to a single plugin that may not be supported in the
future. Its a trend in the industry though, with Waves, Slate, SoundToys and many more developing shell plugins for all their
products.
Light Guide & Smart Play

Light from the SSeries Light Guide LEDs spills beautifully down the white keys and it does look glorious. But beyond the
cool factor, this feature really works as an idea and has been well implemented across the Komplete instruments. Kontakt
instruments use light zones to indicate splits and performance modifiers. Drum instruments use colours to indicate types of
sounds (kicks are red, hats are blue, etc.) and are consistent across products. Even having unassigned keys staying dark is
super useful. Keys light up brighter when they are played, and to indicate triggering from MIDI or the arpeggiator.
The lights are exploited nicely by the Scale and Arp modes that NI are collectively calling Smart Play. Both modes can be
enabled and edited directly from the keyboard. The Arp does everything youd like an Arp to do, including variable swing,
dynamics and gate length. At the basic level, the Scale button puts you into a performance mode that limits notes to a scale.
The neat thing is that the Light Guide shows you which keys are valid. As well as helping us musically challenged music
producers get it right, this is a brilliant educational feature. It also lets you cheat hilariously, because if you do play notes that
arent in the scale they are just rounded to the nearest one that is. And if thats still too difficult, theres an Easy Key Mode,
which maps your chosen scale to the white keys!
Almost hidden in the Scale settings is a wondrous treasure:
chord modes. Harmonic Chord mode will play chords based on
any root note you play in your chosen scale. An encoder controls
the chord type, so you can perform lovely chord progressions
using single keys and judicious knob twiddling. Again, Light
Guide provides a fascinatingly educational accompaniment,
showing you which keys you would have played to get the chord
if you werent so incompetent. Finally, there is Chord Set mode,
which spills a series of predefined chord variations into each
octave, again based on the chosen scale. A knob provides eight
major and minor variations. All thats left for you to do is load a
nice Rhodes patch and mash random keys.
The Polyplex drum machine, also new in Komplete 10.
One thing that should be noted about all the Smart Play functions is that they are playback operations, generated by the
Komplete Kontrol plugin (or Maschine). This means that the MIDI output only comprises the keys you play; you cant record
the actual arpeggiated notes or use these functions with thirdparty instruments. Any realtime adjustments you make to the
arp or scales/chords are recorded as automation in your host. So its a matter of perspective really: theres an advantage to
being able to change the modes after the event (or apply them to existing sequences), while some have flagged this as a
limitation.
The Komplete Solution?
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The burning question I had about the Komplete Kontrol package was about its scope, and where it might go. Right now, its
brief is quite tightly focused: it turns Komplete into a hardware hybrid, provides an integrated keyboard option for Maschine,
and gives you a very nice keyboard and a handful of MIDI controls as a bonus. Of course we always want to know whats
next, and like other exKore users I was interested to know if Komplete Kontrol would inherit more Kore features, like generic
plugin hosting, multiple plugins, internal effects, live features and so on. Looking at it again now, though, I see that
Maschine is in fact the true successor to Kore. I suspect NI will be cautious not to cram too much into Komplete Kontrol, which
was ultimately what made Kore unsustainable. If thats right, Komplete Kontrol will likely remain more akin to a selfcontained
synth workstation keyboard than a generic platform, and on reflection thats probably a good thing.

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Having said that, there are some smaller-scale improvements

that would make the system more useful. My primary issue is the
limited ability to save into the library. The Komplete Kontrol
browser is mostly a oneway street. In Maschine Im used to
rightclicking on any Sound instance and Saving As into the
library theres no mirror function in Komplete Kontrol. NI
explained a way to save updated patches by revealing the
instrument plugins own filing system, then saving out into a
directory. This directory can then be declared in the Preferences
along with any thirdparty sample content. This must then be
Komplete 10s Rounds is an instrument in its own right.
rescanned each time you save something. Once its done your
patches appear in the User tab sorted by instrument, which is good, but should be much easier to achieve. Theres also no
ability to map the knobs yourself. Any patch you save will have the same knob assignments as the patch you started from.
Ironically, its easy to map the knobs to thirdparty synths when youre in standard MIDI mode. Id definitely like to see some
improvement here, and if I could have one thing it would be a favourites system (which is essentially what I use Save As for
in Maschine). Faced with the huge expanse of Kompletes library, I want to be able to star patches that I like and get back to
them quickly. Finally, I imagine NI will face pressure from users to support thirdparty plugins in Komplete Kontrol and, at the
very least, NI plugins purchased outside of the Komplete bundle.
Conclusion
During this review, more than anything I wanted to really get where NI are coming from with the Komplete Kontrol SSeries. It
hit me when I realised that all Id done after three nights was sit there in a daze drifting through the ocean that is Komplete,
reexperiencing years of favourite Reaktor instruments, grinning at the new Rounds synth, just playing the keyboard. First and
foremost, the SSeries makes Komplete an instrument. While the keyboard is sold separately, in most respects its not
competitive as a pure MIDI controller, although the ability to create colourcoded key zones is unique and would be great for
live use. However, the keyboard is not entirely dependent on Komplete as it also achieves its full potential when coupled with
Maschine.
Light Guide is much more than a gimmick and the integrated browser and DAW track/instance navigation really do provide a
handson hardware experience. The encoders and the displays are lovely, although the lack of programmability meant I found
myself paging or reaching for the mouse more than Id like. Yes, the keyboard is expensive (though I remember the first few
versions of Komplete costing more than the SSeries keyboard and Komplete 10 put together). There are many cheaper
keyboards with more controls, with a more generic approach. But, as Apple have shown us, an elegant solution in a controlled
ecosystem can be worth a big premium to many, even while being too restrictive for others. For my money, once Maschine 2.2
is out, this setup is going to be very hard to beat.
.
Alternatives
Competition for the SSeries depends on your usage. As a dedicated front end for Komplete, the SSeries of course has
unique features. You could, however, use other MIDI keyboards with the Komplete Kontrol software, you just wouldnt get
the dedicated hardware tieins and key lights, for example. Likewise, you can use any keyboard with Maschine, but the S
Series has the dedicated mapping. So it all depends how in bed you are with NI; if you use a wide range of other
instruments and dont work in Maschine, there are an awful lot of control keyboards out there. The higher end, with
intelligent integration and automapping schemes, includes the Novation SL, Korg Triton Taktile, Nektar Panorama, M
Audio Axiom AIR, Akai MAX and many more besides.
A strong direct competitor is the Arturia Keylab, which is also a nice semiweighted keyboard, forms a hybrid team with
the Analog Laboratory plugin, and has more controls and pads for less money. If youre a composer/keyboard player
mostly interested in the SSeries with Komplete as an overall instrument, the package could compete with traditional
synth/sampler workstations like the Roland FA08 and Korg PS3X. Finally, if youre trying to decide if you need Maschine or
Komplete Kontrol S, its easy. Maschine is what you need if you want a full production/arrangement environment.
Komplete Kontrol is pure sound source and performance control.
Komplete & Maschine

A key question I had coming to Komplete Kontrol was how it works with Maschine. In thirdparty DAWs Komplete Kontrol
runs as a plugin, bringing the Komplete browser frontend with it. Early tests in Maschine were a deadend (the
Komplete Kontrol plugin could not be inserted at all), then I discovered that full integration comes with Maschine 2.2,
which will hopefully be out by the time you read this. With the update the SSeries becomes a powerful enhancement to
the Maschine environment, beyond the usual benefits of having a keyboard hooked into Maschine. The library and
browser will be unified, so youll be able to access everything from the builtin Maschine browser, but also use the
Navigate buttons and HUD that come with Komplete Kontrol to browse both the Komplete and Maschine libraries at once.
Maschines pad/bank colours will be reflected on the keyboard and show what notes are playing from Maschine. The
Smart Play functionality (the Arp/Chords/Scales performance features) will be shared across the environment and
accessible independently from Maschine and the keyboard. Several commentators have suggested that it makes no
sense that you can buy the SSeries keyboard without Komplete, but it does when you see that the keyboard brings all of
its benefits to Maschine.
New In Komplete 10
Komplete Kontrol is compatible with both Komplete 10 and Komplete 9 (with a library update). I was lucky enough to test it
with 10 Ultimate, meaning many enjoyable hours lost in the new instruments and libraries. Many of Kompletes
components have been reviewed in these pages before, but whats new? The headliners are three new Reaktorbased
instruments.
Kontour is a particularly playable and tweakable Phase Modbased synth. Two oscillators with shapers can be
combined and crossmodulated, and blended with a comb filter and regular filter. The result is kind of a grungy organic
synth with a character somewhere between FM and Physical Modelling. The name comes from the synths front-panel
performance view, which has four modulators, each with builtin motion sequencers. Of course these are mapped to the
knobs on the KK keyboard.
Polyplex is a simple-to-use, but deceptively powerful drum machine, again with immediacy and handson playability a
priority. It has eight pads, each of which can stack up to four samples, and each with its own effects. Seven variations can
be recalled from MIDI notes, and the whole instrument is playable from two colourcoded octaves on the keyboard.
Randomisation is available at the kit, pad, and parameter level so you can generate new kits and sounds quickly. Theres
a great builtin palette of sounds for pop, club and urban styles, with samples instantly swappable from sliders within the
pads.
Rounds is the star of the show for me; you really have to experience it to get what it does, but Ill try to sum it up! The
main panel looks a bit like eight Trivial Pursuit wheels. Each of the four cheeses within each wheel can be assigned to
one of Rounds 16 synth voices. Various performance modes then decide how these engines are triggered, stacked and
sequenced, for example. Triggering can be static, or cycled through the cheeses and/or wheels based on time, successive
notes, or internal sequencer. This architecture can be used in many different ways, from massive trance arpeggiators, to
slow shifting soundscapes, to fullblown multitrack loops. There are two basic synth engine types: a traditional two
oscillator subtractive analogue type, and a twooperator FM digital. You get eight of each to play with. Again, Rounds is
highly integrated with the KK keyboard. The C0 octave provides override selection of wheels and cheeses. The notes can
be used like keyswitches for selecting different sounds, but you can hold multiple keys to stack sound elements or change
sequences. Rounds is pure genius and I predict youll be hearing it on everything for the next year.
On the sampler side, The Definitive Piano Collection is new, plus the Session Horns and Drumlab Kontakt instruments
are now included. On top of these Ultimate gains Rise & Hit, Action Strikes, Kinetic Metal, and Cuba. Two new effects
plugins, the Supercharger tube compressor and Driver distortion filter round out Komplete 10, with Ultimate also getting
Molekular.
Komplete is simply a nobrainer, the only question you have to ask yourself is whether to get the regular or Ultimate
edition. The primary difference is in how many Kontakt libraries you get (and its a metric shed load with Ultimate), so if
youre primarily into the synths then the standard bundle should be enough. Having said that, Id miss the Razor and
Skanner XT Reaktor synths and some of the extra effects plugins that come with Ultimate.
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Information
Fri 19 Dec 2014
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In this article:
Overview
Connectivity &
Controls
Bench Test
In Use
Verdict
Alternatives
Ocean Audio
Signature Three $2895
pros
Classy-sounding inductor
EQ.
Impressive technical
performance.
Decent build quality.
Easy to use.
cons
None really, although some
users might wish to see a
shelving option.
summary
Malcolm Tofts general
approach to EQ design
hasnt changed a huge
amount over the years
and thats no bad thing, as
his products usually sound
good. This version is
unusual in offering more
inductor EQ bands than
others. It sounds great, is
well built and is fairly priced.
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Ocean Audio Signature Three

Dual-channel Equaliser
Reviews : Processor
The Signature Three offers something no other EQ does

four inductor-based bands per channel.
Buy PDF
Hugh Robjohns
cean Audio is the latest venture of Malcom Toft, who is perhaps best known for his involvement in the Trident ARange and Series 80 consoles of the 70s and 80s, and their subsequent reinvention with MTA, Trident-MTA, and
both Toft Audio Designs and Trident Audio Developments within the PMI group. Ocean Audios flagship product is
another large analogue console, the Ark, available in various frame sizes and in two forms, with and without 500-series
module slots. However, I suspect the bulk of the companys business will relate to its other product lines, which include
various 500-series preamps, EQs and routing modules, and three Signature Series rackmount products. This Signature
Series currently comprises a channel strip, a stereo FET-based compressor, and a stereo inductor-based four-band equaliser
the last being the subject of this review.
Overview
In the world of electronics, frequency-selective circuitry generally relies upon two kinds of reactive components: capacitors,
which pass high frequencies but inhibit low frequencies, and inductors, which do the exact opposite. Although inductors are a
fundamental element (mainly in the LF and LMF sections) in most early equalisers, including the classic Pultec EQs and Neve
channel strips, they tend to be physically large and expensive things. Modern devices tend to use electronic trickery to avoid
using physical inductors, reducing both size and cost, so its interesting and unusual that Malcolm Toft has built the Signature
Three equaliser with inductors employed throughout all four of its EQ bands. The handbook explains that inductors lend a
certain distinctive sound character, and the peaking responses obtained allow precise frequency control.
Another unusual aspect of the Signature Three is that the rest

of the circuitry comprises seven all-discrete five-transistor gain
stages in each channel, constructed entirely with conventional
components there are neither op amps nor SMDs here! Again,
the handbook lists the benefits as being a simpler, cleaner signal
path, and the ability to run at higher power-rail voltages for
information
increased headroom. That theory is perfectly valid and theres
$2895
certainly no disadvantage in this approach but there are, in
Ocean Audio +44 1626
fact, countless examples of high-end, op amp-based products
650 017
contact@oceanaudio.co.uk that deliver exceptional technical and sonic performance. And while some classic transistor-based equipment does indeed run
on elevated power rails (eg. early Calrec and modern GML preamps run on 24V rails), a peep inside the Signature Three
www.oceanaudio.co.uk
suggests it actually generates 18V rails about the same as most op amp-based products. Arguably, the only step towards
modernity is the use of an OEM switched-mode power module, which means the unit works on all mains voltages between 80
and 260 V AC. I was pleased to note that the mains safety earth is taken straight to a chassis bolt. Whatever the arguments
for and against the old-school design, the Signature Three certainly delivers the required results.
Connectivity & Controls

The rear-panel electronically balanced input and output connections comprise paralleled quarter-inch TRS jacks and XLRs,
and theres a central IEC mains inlet and an illuminated on-off rocker switch. A large red LED on the front panel confirms the
units powered status. Two more red LEDs, one per channel and labelled Peak, illuminate when the output signal exceeds
15dBu a good 18dB below the clipping level. Below these peak LEDs, each channel has its own independent push-button,
which illuminates green when the corresponding EQ is in-circuit.
The remaining controls are logically arranged, with the four
bands continuous cut/boost gain controls positioned directly
above their corresponding frequency selection rotary switches.
The gain controls span a nominal 15dB range, and all have
centre detents at the unity gain position. All four EQ bands are
fixed peaking types there are no shelf options here and all
four are provided with six selectable centre frequencies. The
handbook provides no specifications for the filter bandwidths, but
rough and ready calculations suggest that all four bands are the
same, with a bandwidth of just over one octave (Q=1.2) at high
The rear panel includes some thoughtful touches such as
duplicating the legend upside down for anyone who is leaning
boost/cut settings, and progressively wider bandwidths at lower
over the top of the unit when plumbing things in!
gain settings. The centre-frequency selections offered on each
band overlap nicely, and are spaced to allow meaningful musical
tweaking. For the record, Ive listed the values. LF: 50, 80, 100, 150, 300 and 600 Hz; LMF: 125, 250, 500, 1000, 2000, and
4000 Hz; HMF: 2.5, 3, 5, 6, 7, and 8 kHz; and HF: 8, 9, 10, 12, 14, and 16 kHz.
Bench Test
After running a set of Audio Precision bench tests on the Signature Three, I was impressed with the technical performance.
The frequency response at the -3dB points extends between 10Hz and 50kHz, and theres a subtle air lift of about +1dB
above 10kHz, which helps to make the unit sound very open and airy when switched into circuit, even with all the band EQs at
their unity positions. The THD+N figure was slightly different for the two channels, achieving 0.0025 percent in the right
channel and 0.004 percent in the left (at 0dBu) but both are better than the claimed 0.005 percent. With increasing input
levels the THD+N figure increased, naturally, reaching 0.01 percent on both channels at +12dBu, and 0.06 percent at +24dBu.
This last is slightly higher than the claimed 0.05 percent at +27dBu, but nothing to worry about.
My Audio Precision system runs out of puff at +26dBu, but there was no sign of clipping at that level, and Im inclined to
believe the specified +28dBu limit, which is far more than anyone would require in practice. The signal-to-noise ratio was,
again, slightly different for the two channels, but measured around 94dB (A-weighted) and 88dB flat comfortably better than
the claimed specifications, and implying a potential dynamic range in excess of 120dB when driven to its fullest extent. With
more conventional peak levels, the dynamic range still exceeds 115dB, which is excellent. The crosstalk between channels at
both 1kHz and 10kHz measured around 109dB normally the figure is higher at 1kHz, but these are still good results, and
the only test which disappointed slightly was the common-mode rejection ratio, which delivered 53dB at 1kHz and 36dB at
10kHz, although I doubt that will cause anyone any problems.
In Use
The Signature Three is perfectly logical and straightforward to use, thanks to a clear and spacious control layout, although it
perhaps suits dual-channel use more readily than true stereo applications, simply because of the inherent difficulties of
accurately matching the individual band gain settings on the two channels, as well as the practical aspect of switching the two
channels in and out simultaneously to assess the effectiveness of the current settings. That said, when using it across a
stereo bus during the review tests I achieved good results.
The gain range and frequency selectivity of each band is more
than sufficient to allow quite precise surgical and corrective
equalisation, when required. But when used with more modest
gain settings the broader bandwidths and generously overlapping
bands allow subtle and very musical creative tonal shaping. I
didnt mind the absence of shelf-equalisation options at the top
and bottom of the frequency range, and in fact the peaking
response characteristic proved more useful when trying to shape
and control the low end. At the high end, the relatively wide
peaking response bandwidth still allows gentle air lifts to be introduced without any difficulty.

explained
Competitions!
Verdict
There is something very attractive in the sound character of inductor-based equalisers thats hard to describe they just
sound more natural and right to my ears, especially at the bass end of things. Its probably tied up in the mysteries of phase
shifts and waveform distortion, but whatever it is, the Signature Three delivers it well. Overall, this is an unusual inductorbased dual-channel equaliser with all-discrete transistor circuitry that delivers a classy and attractive sound, with well-sorted
EQ options. It is very well made here in the UK and achieves excellent technical performance. Whats not to like? .
Alternatives
I cant think of any other EQ that offers four bands, each featuring inductors, but there are plenty of other inductor EQs
offered by the likes of Neve, Rupert Neve Designs, Cartec and A-Designs, as well as the companies with which Malcolm
Toft has been involved in the past, Trident and Toft Audio Designs.
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Panda Audio MidiBeam

Wireless MIDI Interface
Reviews : MIDI Interface
Paul White

Panda Audios MidiBeam wireless MIDI interface does pretty much what it says on the tin. Theres
a small MIDI transmitter powered by two AAA batteries, with a battery life of over 65 hours. The receiver gets its power from a
mini USB socket, and can deliver MIDI either via USB to a computer or through a conventional five-pin DIN. The USB cable
isnt supplied, unfortunately, and for stand-alone use without a computer youll also need a USB power adaptor.
Operating on the 4.2GHz Wi-Fi band, the system needs no licence to operate and has a range of up to 250 metres in open
space, though the indoor range will invariably be less. Pairing buttons on each unit allow them to recognise each other, and
one of 76 possible channels is selected automatically to avoid interference, meaning that multiple systems can coexist. Most
importantly, the wireless system only adds around a third of a millisecond to the existing MIDI response time.
This obviously has applications in live sound: those with battery-powered instruments who wish to move around the stage
are obvious potential adopters, but there are also studio applications, such as when your MIDI controller is in one room and
the MIDI system it controls is in another. For example, an electronic drum kit in the studio could send MIDI to trigger samples
in the control room.
The MIDI plug on the transmitter is at the end of a short captive cable, and Velcro tape may be used to fix the transmitter to
the instrument. A single power switch is the only control on the transmitter, and the LED flashes to provide a warning when the
battery has around 10 percent capacity left.
The receiver is recognised by computers as a class-compliant
MIDI interface, so it needs no special drivers. As an added
bonus, in addition to its MIDI output connector, theres a MIDI In
port that can be used in non-wireless mode, to allow the receiver
to double as a basic USB wired-MIDI interface. Although the
procedure is not described in the accompanying quick-start
manual, the radio power can be set to one of four increments,
and the transmitter can also be set to power off automatically if
its not used for a certain time, with options of 10, 30 or 60
minutes.
The manual directs us to the companys web site, the support
section of which includes both a fully detailed manual and a
series of brief MIDI SysEx files that need to be sent from a
computer to the devices (which must be switched on and paired)
in order to change the settings. The system worked without a
hitch once Id fitted the batteries and paired the transmitter with
the receiver.
The Panda Audio MidiBeam receiver (left) and transmitter

(right).
Indoors, the range is about the same as or greater than youd expect for your wireless router, and the design includes a
safeguard against hanging notes if the transmitter moves out of range. By default the power is at maximum and the transmitter
doesnt shut off until you switch it off, so if youre happy with that theres no need to do anything else.
Wireless MIDI has been around for a while, but this system offers especially low delay times, which in normal use will be
negligible in comparison with the delays inherent in MIDI systems and keyboard-scanning circuitry. If you need wireless MIDI,
then this is a very practical solution. Paul White
126.87 including tax and worldwide shipping
www.pandamidi.com
.
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In this article:
Easychair
Man Bites Zombie
Lukas Creswell-Rost
Lunacybot
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Chris Korff
Easychair
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Clouds Thick, Whereabouts Unknown

I forget who it was that said, There are only two possible answers to the question Do you like Jonathan Richman?: Yes, and
Jonathan who? Members of Londons small but determined country scene seem to fall into the former category. Years ago,
the Rockingbirds hymned his praises on their self-titled debut album, and now Easychair who played for many years as a
bluegrass duo in London pubs have their own Jonathan-themed song.
This, of course, is a reliable indicator of good taste, so its no
surprise that Clouds Thick, Whereabouts Unknown is worth
listening to. Apart from sporadic eruptions of banjo, however,
Andy Brewer and Anton Pigott have mostly left bluegrass behind,
in favour of an idiosyncratic, pastoral and identifiably English
songwriting style which nicely balances the melancholy against
the humorous. I cant fault the playing or the production, and
theres a happy absence of trans-Atlantic accentry. It sounds as
though Easychair have made the album they wanted to make,
rather than aiming for the elusive commercial appeal, and its all
the better for that. Sam Inglis
Man Bites Zombie
Man Bites Zombie

Popular culture is a fickle beast. One minute everyones making
high-budget games and TV shows about organised crime, then
its pirates, and right now it seems to be zombies: witness the
fantastic Playstation epic The Last Of Us, TVs The Walking
Dead, Plants Vs Zombies on the iPad... and riding the crest of the
undead wave are South Wales-based electro-rock concept band, Man Bites Zombie. Perhaps surprisingly, its actually rather
good.
The zombie apocalypse aesthetic translates well to the concept-album format. Not only are there plenty of opportunities for
gruesome lyrics about the best way to neutralise the afflicted, but Man Bites Zombie have also recorded some nice faux-news
snippets and audio diaries to use as song introductions.
The music is produced well, in that glitchy, slightly industrial

breakbeat/metal kind of way favoured by action-movie music
supervisors and Top Gear producers. The performances are
good too the drums especially, which often use electronic
sounds but which were played in live on a V-Drum kit, which
helps maintain the real band feel. Chris Korff
www.manbiteszombie.net
Lukas CreswellRost
Go Dream
Im not entirely sure how to describe Go Dream, other than that
its very good. The albums creator, Lukas Creswell-Rost,
apparently recorded all of it at his own studio in Berlin (the one
exception being the James Yates drum recordings, which sound
excellent), and its quite the feat, especially in terms of
songwriting and arrangement.
Theres definitely an 80s vibe, courtesy of some very long
reverbs and distinctly of-that-time synth patches, but this is no
nostalgia fest some adventurous production techniques, such as heavily distorted drums and contemporary electronica
elements see to that. Though there are definitely some pop sensibilities on show, I wouldnt call it catchy its a little too
sophisticated for that, but then I prefer music that presents something of a challenge. Chris Korff
www.lukascreswellrost.com
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Ploytec PL2
MIDI Powered Duophonic Synthesizer
Reviews : Sound Module
Can this tiny box deliver a big sound?

Paul White
At every trade show we come across at least one product that seems to be half barking mad and half genius. This qualifies.
First shown in 2013 but now shipping, German company Ploytecs Pi Lambda Squared (PL2) is powered entirely via its MIDI
input and based around some type of lo-fi sound chip. Its output is a simple mono RCA phono connector and it is only around
twice the size of a 9V battery. It also has a mini-USB port that can be used to administer software updates or to provide power
where MIDI bus power isnt available. Although all the sounds are based on just four waveforms derived from square waves,
and there are only two tone oscillators, the synth ships with 32 factory presets and room for 32 more user patches. One side
of the single-sheet instruction manual (which is about the size of a cracker joke) is taken up with MIDI information that allows
real-time control over a surprisingly large number of parameters. Further user information is available on the companys web
site (www.ploytec.com).
The digital filter has three modes (LPF, BPF, HPF), while the output stage includes a digitally controlled analogue low-pass
filter plus some intentional distortion to smooth out some of the nastier artifacts (of which there are many!) and to add more of
an analogue character though the sonic imperfections of this synth contribute to its gratifyingly raw vibe. In additional to the
expected filter, portamento and envelope parameters are pulse-width modulation and the ability to sync the LFO to MIDI
Clock.
Any basic synth tends to sound somewhat dry and stark until
you add effects such as delay or reverb, and this one is no
exception, though it is capable of both analogue-esque bass
sounds and some spectacularly gnarly leads. And the ability to
apply pulse-width modulation makes it sound fatter and more
complex than you might expect. The instructions are sparse, but
a little experimentation revealed that it responded to MIDI
channel 1 and gave a one-octave shift in response to a full pitchbend wheel-up movement.
The designers are actually proud of the amount of aliasing and
quantisation noise, but its up to you whether you consider this to
be exciting and industrial or cheap and nasty! Certainly some of
the notes fizz into graceless extinction at lower velocity levels,
while MIDI-controlled dynamic parameter changes tend to sound
ragged and glitchy, but were told that is all part of the charm.
Changes made to patches 33 to 64 are stored automatically as
soon as you switch to a new patch, while patches up to 32 are
factory presets some of which are stacked mono sounds,
others are duophonic. Realistically you could get a sweeter-sounding synth plug-in for the same cost as this synth, with even
some of the free ones sounding significantly more sophisticated, but if you like the notion of synth sounds informed by the lowdefinition sound chips of yesteryear, then this might just be your idea of fun. Paul White
77 plus shipping
www.usb-audio.com
.
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In this article:
XILS Lab XILS4
Digital Brain Multiplex
Vocoder
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Plug-in Folder
Reviews : VST Instruments
Gordon Reid
Buy PDF
XILS Lab XILS4

Formats: Mac & PC VST, AAX & RTAS; Mac AU
It can be difficult to improve upon a proven product but, when the chaps at XILS Lab decided to enhance their XILS3 v2 soft
synth (which was based in large part upon the EMS VCS3), they hit on a rather spiffing wheeze. Rather than tweak their
model of the VCS3 still further, they decided to adopt the concepts of 1969s legendary EMS VCS4. Only ever built as a
prototype, this combined two VCS3s positioned sidebyside, plus a keyboard, mixer and additional signal processing. (A
second VCS4 prototype resurfaced some years ago, but this was merely a modified VCS3 and was ditched by EMS in favour
of the Synthi A and KS.)
You can license XILS4 using iLok or a donglebased eLicenser, or
(for the appropriate cost) upgrade an existing XILS3 or XILS3 v2
licence, whereupon the protection will allow you to run all three synths
simultaneously if desired. I upgraded XILS3, which proved to be quick
and painless although, as before, XILS4 doesnt run as a standalone
application, only as an RTAS, AU, AAX or VST2.4 plugin.
Users of the original XILS3 will find numerous changes in XILS4
although, to be fair, many of these were introduced in XILS3 v2. Indeed,
if you treat XILS4 as a pair of XILS3 v2 synthesizers that can be used in
series or parallel, with a single sequencer but additional options in the
keyboard and sequencer matrices, you wont go far wrong.
As a user of the original XILS3, I particularly like the redesigned and
more colourful GUI, and the new virtual patch pins also deserve
special mention. Whereas the original XILS3 provided eight types of pin
offering preset amounts of positive and negative connection between
modules, the pins in XILS3 v2 and XILS4 act as potentiometers,
allowing you to dial in (where appropriate) the wanted polarity and level
of the connection, making it possible to finetune your sounds in ways
that werent previously feasible. Im also grateful for small but important
The pins in the XILS4 matrix offer continuously
variable resistance values.
upgrades such as the larger, popup patch matrices that allow ageing,
hyperopic keyboard players to see whats connected to what, and by
how much. On a more trivial level, but nonetheless welcome, I like the addition of a phaser in the effects panel, because this
was an effect widely used with EMS synths.
No two VCS3s sound or respond identically, so Im not overly concerned that neither XILS3 nor XILS4 precisely emulate my
VCS3, especially since theyre polyphonic and offer all manner of modern programming and performance options. Im also
largely unconcerned by the odd bug or two, which will no doubt be ironed out in time. Ultimately, XILS4 is a remarkably deep
and powerful soft synth, capable of prodigious feats of synthesis and external signal processing. If youre prepared to put
some time and effort into it, youll find it to be a fascinating and flexible instrument, and I doubt that youll ever plumb all of its
possibilities or fail to discover new sounds using it. Gordon Reid
179 (approximately $223).
www.xilslab.com
Digital Brain Multiplex Vocoder

Formats: hosts Mac & PC VST; Mac AU
Multiplex Vocoder is not a plugin, but a standalone application
that can host other plugins. Its job is to turn pretty much
anything you throw at it into a vocoder carrier signal which can
then trigger up to five vocoders at a time. As its name suggests, it
does this using a simple but effective matrix system which can
route multiple inputs to different vocoders and mix the outputs. It
does all this using whatever VST or AU instruments are installed
in your system, as well as giving you the option of adding plugin
effects.
For anyone who has never played with a vocoder, it requires
two sound sources to work. The first is the sound to be
processed, which in this case is any VST or Audio Unit
instrument on your computer. This needs to be triggered by your
MIDI keyboard, and the MIDI input will usually dictate the notes
that are played as well. The second sound source is called the modulator: this is most often a vocal, though any signal can be
used. This is then analysed to create a filter signature which is then applied to the first sound. It is only when you speak into
the mic that sound will be triggered, and the filtering will make the carrier sound follow the inflections of your voice. In Multiplex
Vocoder, modulator signals can be derived from one of two microphone inputs or an audio file player, which allows you to play
any file from your computer into Multiplex Vocoder and use it as the carrier signal.
So why do we need Multiplex Vocoder? Well, this applications

great strength is its flexibility, thanks to its ability to address
multiple instruments at the same time. Filling the five slots with
five instances of Native Instruments Massive, I was able to build
up a rich layered sound, using slightly different presets to make a
more complex tone. You can of course use completely different
instruments to get even more variety, and if you want to stretch
the tonal palette still further, you can add up to two effects plug
ins per instrument. Each instrument slot has a threeband
parametric equaliser section as well as high and lowpass
filters. An output mixer allows you to vary the mix between the
various instrument slots. Each slot can also be muted although, sadly, there is no solo function, which would help when you
are trying to work out which sound is which. There is a master channel also with its own equaliser and plugin slots, which
come in handy for a bus compressor to keep your synthesized excesses under control!
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The input section allows you to set levels for the two microphone and one playback channels. The microphone inputs have
a highpass filter option which can help with maintaining intelligibility. There is also an option that allows you to morph from
the dry signal to the wet. The top right of the screen gives you more matrix options, allowing different instrument slots to
respond to the different inputs. You can of course record your outputs, and in fact there are several ways to do so. As its not a
plugin itself, to make Multiplex Vocoders output appear within your DAW you will have to route the signal in and out using a
thirdparty router such as Soundflower (Mac) or Jack Audio (Windows). This is relatively easy and shouldnt put you off.
I found Multiplex Vocoder simple to use, and had it running quite quickly and happily using the builtin microphone on my
laptop. You do have to remember that anything the microphone picks up will trigger a sound from any note the keyboard is
holding down, so an external microphone with a tight polar pattern will help. It is such a cheap bit of software Im sure it will
make its way into many peoples tool kits. Even if you dont use vocoders that often, I can guarantee youll have fun with this
one! Jon Burton
$25
www.digitalbraininstruments.com
.
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Fri 19 Dec 2014
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In this article:
Match Making
BIAS Cut
Custom Eyes
After Effects?
BIAS Curious?
Alternatives
Matching Up
Positive Grid
BIAS $99/$199
pros
Very slick user interface.
Amp-design features are
excellent.
Amp Match technology has
considerable potential.
Sounds very good.
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Positive Grid BIAS

Guitar Amp Simulator Plug-in For Mac OS & Windows
Reviews : VST plug-ins
With component-level adjustment and amp matching

features, BIAS can target almost any guitar amp sound
real or imagined!
Buy PDF
John Walden
ove em or hate em, virtual guitar rigs are now a part of the music-technology furniture. For many home, personal or
project studios where cranking up an amp is not an option, they are a necessary convenience. However, they also offer
a huge range of tonal variations that is difficult to match unless you happen to have a warehouse full of classic guitar
hardware. And to many ears (my own included), the quality of the results can be more than good enough to be used in a
commercial music-production context.
In the Mac and PC realm there are a number of well-established products: Line 6s POD Farm, IK Multimedias AmpliTube,
Waves GTR, Peaveys ReValver and Native Instruments Guitar Rig Pro would be among the obvious candidates. However,
No stompbox or rack effects. we now have a new contender in the shape of BIAS from Positive Grid. I say new but BIAS is interesting on two counts, the
first of which is that it actually started life as an iPad app. Unlike software from the likes of Line 6 or IK Multimedia that has
summary
been ported from desktop to iOS, BIAS is coming in the other direction.
cons
Though its not quite a

complete guitar setup in
software, the novel ampdesign and amp-matching
technology in BIAS will
make it very attractive to any
serious guitar tone-head.
information
BIAS Desktop $99; BIAS
Professional $199.
support@positivegrid.com
www.positivegrid.com
Test Spec
BIAS Professional
1.0.8.2160.
Apple iMac with 3.5GHz
quad-core i7 CPU and
32GB RAM, running Mac
OS 10.9.5, with Focusrite
Scarlett 8i6 interface.
Tested with Steinberg
Cubase 7.5.20.
Second, BIAS isnt quite from the same mould as other virtual
guitar rigs. While it does amp, cab and microphone modelling so
that you can craft a guitar tone, it also offers an amp design
function. You can tweak components within the preamp or the
power amp, and because its all virtual, it comes without the risk
of instant electrocution and sudden death.
Match Making
BIAS is actually available in two flavours. The more affordable
BIAS Desktop gives you the amp modelling and amp design
features that are the basis of the iOS version. There are 36
default amp models supplied to get you started, and cabinet and
microphone modelling are also included. BIAS Professional adds
a further element: Amp Match Technology. In principle, this
attempts a similar function to the very sophisticated hardwarebased Kemper Profiling Amp. In BIAS Professional, the
dedicated Amp Match module allows you to sample both your actual sound (the source) and the sound you would like to
recreate in a model (the target) and then applies a processing algorithm to transform the source into the target.
Cramming all the equipment included within BIAS into your studio is somewhat easier than with the equivalent hardware: a
fairly modest 460MB download, a speedy installation and a simple online activation process will have you up and running
quickly. With support for VST (32- or 64-bit), 64-bit AAX, AU (32- and 64-bit) or RTAS plug-in formats, just about everyone
ought to be catered for.
BIAS Cut
Whichever version of BIAS you choose, the core features are the amp/cab/microphone modelling and the ability to design
your own amp. You can access all of these elements from the main plug-in window, which is split into four horizontal areas.
The topmost strip provides access to the preset system and settings menu. Also in this strip is an option called Tone Cloud
which lets you download BIAS creations from other users and upload your own models. This works very well, and Tones are
also compatible with the iOS version of BIAS.
The next strip contains a visual representation of the signal chain through the various modelling elements, starting with the
front panel of the amp and running through to the Amp Match component. The signal chain is fixed, but you can drop either of
the two EQ modules and the Amp Match module in and out as required simply by dragging them up or down.
Click on any specific component within the signal chain and it becomes the focus of the next strip. Its here that you get to
tweak the amp model at the component level, whether thats the visual appearance of the Custom Panel, the valve types used
in the preamp and power amp, or the transformer. Finally, the strip along the base of the window provides controls to set the
input and output levels, configure the hum reduction, noise gate and room control options and configure and use the quick
snap panel. The latter allows you to select up to eight presets for easy recall.
Custom Eyes
Each of the modelling components provides you with editable parameters. So, for example, if you select the Custom Panel,
not only can you tweak the virtual knobs to adjust the tone as you would on a real amp, but if you click on the small edit icon
located top-right of the panel, you can then customise various elements of the amps appearance. In this module, this feature
is mostly cosmetic, but it does allow you to get the visual vibe right as a useful reminder of what kind of tone a particular
preset is intended to create.
The rest is far from cosmetic, though. For example, select the
preamp and a whole host of components and settings can be
adjusted. While you would probably need a PhD in electronics to
make any informed choices when configuring the equivalent
options in the design of a real (hardware) amp, in BIAS you are
free to simply experiment and see where it leads you. Want a
more overdriven sound? Then wind up the input tube gain or add
an extra tube stage. Its easy to do, and nothing explodes.
While the preamp and power amp contain perhaps the largest
number of tweakable options, there are still choices available
within the tone stack and the transformer. Indeed, if you really
are serious about crafting your ideal guitar tone, there are
endless hours of fun to be had. And if you just want to start
gently, each component is also supplied with a set of presets so
you can, for example, mix and match from about a dozen
Preamp presets, eight Tone Stack presets, and so on.
Build your own amp? BIAS allows you to wield your virtual
screwdriver and soldering iron in a safe environment.
Beyond the amp itself, the editing options are perhaps more
conventional. There is a series of cabinet models spanning the
usual suspects from 1x12 up to 8x10, with plenty of stops in
between. Each colours the sound in its own distinctive fashion.
You also get a choice of two different microphone models and
the option to position these virtual mics in front of the cab. The
mic position can influence the tone and volume of the output
and, with the room simulation option engaged, you can get some
very nice real room ambience into your sound.
As with a real amp and most virtual guitar rigs, the more gain
you use, the noisier things can become. Having used BIAS
alongside both ReValver and Guitar Rig, I dont think there is
All of the key amp stages are editable within BIAS.
much to choose between them on this front, but I certainly found
both the Hum Removal and Noise Gate options useful to have once I was into (and beyond) metal territory.
After Effects?
So far, so good. The virtual amp design features are very well executed, and the Amp Matching feature of the Professional
version is both interesting and, with a little care and effort on behalf of the user, very capable. However, if you are in the
market for a single does it all virtual guitar rig plug-in, BIAS is currently missing one key component: a collection of virtual
stompbox or rack effects. This is exactly as in the iOS version of BIAS but, in that case, Positive Grid have a companion app
called JamUp Pro that provides stompbox effects; you can import BIAS amp models into JamUp to create a complete guitar
rig signal chain. Im not sure quite how Positive Grid might address this issue with the Mac and PC versions of BIAS, though
they obviously have the signal-processing expertise, as JamUps stompbox effects are very good.
BIAS Curious?
Having used BIAS under iOS for quite a while, I wasnt surprised to find that the desktop version is a very slick piece of
software. To my ears at least, it is perfectly at home beside the more established virtual guitar software plug-ins mentioned
earlier. Sonically, therefore, BIAS is most certainly a contender. For complete tone nuts, the ability to dig into your (virtual)
amps inner workings is going to prove a lot of fun. The fact that the interface makes this so easy to do, and yet still gives you
plenty of options, is very impressive.
Equally, in the Professional version, the Amp Match technology is straightforward to use. That said, it is not a magic bullet to
emulating the sounds of your guitar heroes; it requires quite a bit of care and attention (and possibly a bit of luck) to match a
sound from a favourite track. Matching a real amp is perhaps a more realistic proposition but that does imply that you might
have some amp sounds that are better than the presets built into the software and worth capturing in the first place! Its clever
technology, and very interesting to explore, but perhaps a quite specialist function.
If you just want a versatile amp simulator that supplies a broad palette of pre-configured guitar rigs, its lack of effects means
perhaps BIAS isnt the most obvious choice. However, for true tone-heads, who want to craft how their virtual amp sounds and
responds, with the exception of some component editing available in ReValver, BIAS is in a class of its own. In principle at
least, the Amp Matching may well appeal to the same crowd. The interface is also a pleasure to use.
The feature set in BIAS overlaps with, but doesnt mimic, whats available in the obvious competition. It is well worth
auditioning already but, if Positive Grid can add that missing effects element, BIAS will offer both direct competition for the
established plug-ins as well as the amp-design element that sets it apart from the herd.
.
Alternatives
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Among rival all-in-one virtual guitar rig plug-ins, Native Instruments Guitar Rig Pro, Line 6s POD Farm Platinum, IK
Multimedias AmpliTube 3, Waves GTR3, Scuffhams S-Gear and Peaveys ReValver are probably the most obvious
competitors. ReValver 4 is also an interesting comparison because it includes some amp-design and sound matching
technology. If amp matching is your main interest, the main option is the Kemper Profiling Amp, but this is a hardware unit
that is much more expensive than any of the plug-ins.
Matching Up
If you stretch to the Professional version of BIAS, you get Positive Grids Amp Match technology. The Amp Matching
process is, in principle, pretty straightforward to use, although there are a number of steps involved, and the best way to
approach it may depend a little on your specific DAW. In essence, however, BIAS captures an audio sample of your
current sound (the source), an audio sample of the target sound, and then applies some processing to match the two.
The resulting profile can then be applied to the source so that it sounds like the target.
I had a go at matching both a real amp and a tone from a pre-recorded guitar track. Even with a bit of experimentation,
the potential of the technology is pretty obvious but, equally, you soon realise that getting good results does require a bit of

explained
Competitions!
S8 monitors
Win Munro Sonic Egg
150 (Global Except
Americas)
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Christmas bundle
(Americas Only)
care. First, in order to make things as simple as possible for the matching algorithm, the closer the sound of the BIAS
model that you start with is to the target, the more likely you are to end up with a good result. Second, you need to make
sure that your guitar input signal is as free as possible from hum and noise.
Third, you want your target sound free of effects just
the sound of the guitar through an amp. This last issue
might be more significant when trying to match guitar tones
from a commercial guitar track; the trick will be finding a bit
of the track where the guitar is both fairly isolated in the
mix, and not soaked in delay, chorus or reverb. You cant
get an ideal snippet from all your favourite rock classics,
but you can get close with some. I gave this a go with a
couple of examples, neither of which were perfect
candidates, but perhaps typical of what users might like to
try. The results were not bad, and I suspect that if I had
spent a little more time getting the source tone closer to
the desired target so the matching process didnt have to
do quite so much corrective work, the results could easily
have been improved.
BIAS Professional includes the Amp Match component,

allowing you to profile amp sounds and attempt to match
them within BIAS.
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In this article:
Old School
Internal Affairs
Listening In
Alternatives
Prodipe
Pro 8-3W $699
pros
Wide bandwidth.
Able to play loud.
Well-judged tonal balance.
cons
Question marks over
manufacturing consistency.
Cost saving leads to
manufacturing and
performance compromises.
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Prodipe Pro 8-3W

Three-way Active Monitors
Reviews : Monitors
Three-way speakers tend to reside at the higher-priced end

of the market can Prodipe succeed with a budgetfriendly option?
Buy PDF
Phil Ward
any of the active monitors Ive written about within these pages have come with both decidedly eye-watering and
wallet-emptying prices. The subject of this review however could hardly be more different. In fact, despite being
designed to do pretty much the same job as those speakers, the Prodipe Pro 8-3W would set you back as little as 10
percent of the price of some of them. So, does it cut the proverbial mustard and get close to the performance of its far more
expensive predecessors?
There is of course an almost unlimited variety of active nearfield monitors that can be had for the same sort of outlay as the
Pro 8-3W, and the review pages of Sound On Sound are populated by a regular supply of them. The arrival of another
The 8-3W is a brave attempt competitively priced two-way, active nearfield monitor can be guaranteed almost as surely as the rising of the moon, an
England batting collapse and disappointment at the World Cup. So, whats different about the Pro 8-3W? Well, its a three-way
at three-way for entry-level
two-way money. The result
design for a start and, it has to be said, it looks an awful lot of speaker for the money.
summary
is compromised and has

some rough edges, but its
fundamentally a capable
monitor.
information
$699 per pair.
www.prodipe.com
Theres an argument to be made that the prevalence of two-way speakers those with just a bass/mid driver and a tweeter
generally represents a triumph of pragmatism over electro-acoustic engineering. You see, any bass/mid driver big enough
to provide acceptable low-frequency bandwidth and power struggles to reach far enough up the audio bandwidth to meet a
typical tweeter. And similarly, tweeter diaphragms small and light enough to provide high-frequency response that extends
past 20kHz, and without serious dispersion issues, are typically operating close to their mechanical limits down where they
meet the bass/mid driver. Its like opposing politicians reaching for a reluctant handshake its often not entirely convincing.
Employing a mid-range driver seems like itd be a no-brainer then. But, and its a big but, things arent that simple. Leaving
aside the cost of a third driver, and the rather obvious point that, with a compact speaker, theres often no space on the front
for it, adding more filter stages in the electronics, adding a third amplifier channel, managing driver integration issues and
incorporating a separate sealed internal enclosure in the cabinet, all conspire to drive designs towards the two-way solution.
Three-way is often just too expensive and complicated; theres too much to go pear-shaped, especially if youre doing it on a
tight budget.
All of which brings me back to the Pro 8-3W. Being a three-way
design, the Pro 8-3W is necessarily larger than the typical
compact nearfield no surprise there, its designers had to find
somewhere to put the mid-range driver. It is in fact quite a big
speaker, and if you find yourself tempted to try a pair, first make
sure they will fit your studio space. In cost-effective speakerdesign terms, a larger cabinet actually falls into double-edged
sword territory: while generous internal volume means extended
low-frequency bandwidth is easier to achieve, it also brings larger
cabinet panels that can vibrate along with the music and
significantly colour the sound. Internal bracing that might
ameliorate the resonance of large wooden panels is often simply
unaffordable if youre targeting the budget end of the market.
Old School
The Pro 8-3W front panel carries a 200mm bass driver, a 100mm
mid-range driver and a 25mm tweeter arranged with the midrange driver and tweeter vertically arrayed alongside the bass
driver. The side edges of the front panel are softened, which will
help a little with high-frequency diffraction. The Pro 8-3W is
supplied in mirror-image pairs, and Prodipe recommend that they
are used with their vertical mid-range/tweeter arrays outermost.
None of the drivers obviously incorporate any esoteric
technology or materials; at the same time, however, they all look
to be competently designed units. There is something old-school
about them though the tweeter has a coated textile dome
buried down a short, horn-shaped throat, and both the bass and
mid-range drivers have pressed steel chassis. The bass driver
The rear panel is a sparse affair, featuring only the power and
diaphragm is a very traditional-looking coated paper cone and
audio inputs, and an input gain trim.
dust-cap, and the mid-range diaphragm a similarly traditional
thermoplastic cone. Of course, in describing a speaker as old-school, Im not necessarily being critical. For example, at a
Munich high-end hi-fi exhibition a few years ago, among all the massively sophisticated and high-tech speakers featuring
ceramic and even diamond-diaphragm drivers, the best sound to my ears came from a pair of ATC speakers incorporating
exclusively old-school diaphragm materials. Some lessons taught in the old school were undoubtedly good ones.
Around the back of the Pro 8-3W is the usual amplifier heatsink and connection panel. Perhaps not unexpectedly at the
price, the Pro 8-3W offers just the bare minimum of rear-panel facilities mains socket, mains switch, power indicator,
balanced and unbalanced analogue inputs and a gain control. Theres no digital inputs or DSP malarkey, the lack of which part
of me celebrates. The gain control offers 6dB of adjustment but lacks detents, so theres no way to be sure that both
speakers of a pair are at the same gain unless you wind the control fully up or down. The Pro 8-3W amplifiers are specified at
90W, 40W and 20W for the bass, mid-range and tweeter sections respectively. (If youve ever wondered about amplifier power
ratings for different sections of active speakers, the differences simply reflect the relative inherent electro-acoustic efficiencies
of the drivers.) Back around the front of the Pro 8-3W, a couple of detented knobs (with legends and detents that dont quite
line up) provide some fine HF and mid-range level adjustment.
Internal Affairs
A quick look inside one of the Pro 8-3Ws, achieved by removing one of the bass drivers, revealed what appears to me to be
some evidence of cost savings. First, all the drivers are secured by wood screws rather than bolts and T-nuts. Wood screws
are not inherently bad, in particular, to secure the relatively small and light mid-range driver and tweeter, however the bass
driver is big and heavy and, when secured with just four screws that cant be tightened particularly well thanks to the risk of
the thread stripping, it can never be really intimately attached to the cabinet. Second, the Pro 8-3W cabinet is constructed
from 15mm MDF rather than the more usual 18mm board, and despite the cabinets relatively large panel dimensions, there
are no internal bracing ribs. A sharp knuckle wrap on the cabinet reveals it to be pretty resonant. Third, the cabinets contain
just the bare minimum of internal polyester wadding for damping. And lastly, while theres a flared plastic flange on the outer
end of the reflex port that will help delay the onset of port distortion and compression, the inner end of the cardboard tube is
just cut straight. The tube is also very long and of relatively small diameter. This is good in one respect as it means the port is
tuned to a relatively low frequency, so is less likely to mess audibly with the speakers time domain performance, but its
unfortunate in another respect in that longer, smaller port tubes become non-linear and turbulent earlier than shorter, bigger
ones.
One of the endlessly fascinating aspects of speaker design (no, really!) is the challenge of balancing all the conflicting
engineering constraints and design requirements and hitting the cost target, and I think Prodipe should be applauded for even
attempting a three-way active nearfield at such a competitive price. Theres one, presumably cost-saving, measure on the Pro
8-3W however that I think falls into the cost-saving too far category. Its also a measure that Id imagine saves so little that it
would probably be almost impossible to quantify. I mentioned earlier that one of the extras needed in a three-way speaker is
a separate internal enclosure for the mid-range driver, and thats what the Pro 8-3W has. The only problem is that no attempt
has been made to seal it from the pressure changes inside the main enclosure. Theres a hole where the mid-range
connection cables enter. It just needed a spot of hot melt glue after the cables went through (and yes, I was tempted).
Now, I perhaps know what youre thinking. Youre thinking that it doesnt matter about sealing the mid-range enclosure
because the main enclosure has a big hole (the reflex port), so therell be no significant internal pressure changes to worry
about. Youd be wrong Im afraid. In fact, peak internal pressure changes (at port resonant frequency) can be higher in a
ported speaker than in an equivalent closed-box speaker. So in the Pro 8-3W, the mid-range diaphragm will be driven
backwards and forwards at frequencies around the port resonance. Now this wont result in any significant low-frequency
output from the mid-range driver, however it will mean that some of the drivers available linear movement will be needlessly
used-up, and it will mean increased levels of inter-modulation distortion as the ports resonant frequency is imprinted on the
output of the mid-range driver. It will also mean the Pro 8-3Ws intended low-frequency performance will be compromised. And
it could be fixed with just a drop of hot-melt glue.
Monitoring Tips
Monitors vs Hi-Fi?
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On discovering the inter-enclosure leak we asked Prodipe to comment. Their response was that the leak was a
manufacturing fault, neither intended nor typical, and they supplied a second pair of monitors. The second pair duly arrived,
and while the cable access hole between the mid-range and bass enclosures had some hot-melt glue applied, it wasnt perfect
and there was still a small leak although probably small enough to be of little significance. Sealing the mid-range enclosure
is clearly something Prodipe need to tighten-up on.
Listening In
So in its design and manufacture the Pro 8-3W is, unsurprisingly perhaps, fundamentally influenced by the cost savings
necessary to produce a three-way speaker at an entry-level price. The proof is in the listening, however, so how did the
second pair of Pro 8-3Ws perform? If youve read the previous paragraphs you perhaps will be expecting me to recount a less
than happy tale here however, another of the endlessly fascinating aspects of speaker design (yup, thats right, there are
two!) is that sometimes speakers defy the odds and pull off a more than the sum of their parts trick. So, despite some rough
edges, in particular what I suspect is the aural signature of those unbraced enclosure panels playing along with the music in
the 200 to 400 Hz region, the Pro 8-3W does some things quite well. In particular I think it genuinely demonstrates one of the
advantages of three-way over two-way; the bass driver can be optimised for reproducing low frequencies rather than be
compromised by having to do the mid-range as well. As a result the Pro 8-3W sounds like a big speaker with a wide
bandwidth. Theres an authority to the way it reproduces low frequencies, and an ability to effortlessly play loud, that you rarely
get with compact, entry-level two-way speakers. Having said that, I was never quite happy with the time domain performance
of the low end. I tried with the ports both blocked (with socks, now youve asked) and open, but the bass always seemed to
have a slightly slurred quality that made any qualitative assessment of low-frequency mix elements more difficult than
necessary. Further up the range, the Pro 8-3W lacks the kind of extreme clarity of the best monitors that enables you to feel
that you can hear every little detail right down to the noise-floor of a mix, but generally it equips itself competently, with a welljudged overall tonal balance, nicely focussed stereo images and drivers that just get on with the job without drawing attention
to themselves.
I was left feeling slightly frustrated by the Pro 8-3W. Theres potentially a genuinely good active monitor struggling to make
itself heard, but to my mind, as things stand, it is compromised a little too much by the cost savings made to reach the price
point. Part of me was sorely tempted to reach for a glue gun, some MDF off-cuts for cabinet bracing and some extra enclosure
wadding in order to pimp the review pair of Pro 8-3Ws and knock off a few of those rough edges. But of course thats not
what writing a Sound On Sound equipment review is about. What it is about, however, is giving recommendations, or
otherwise. As things stand, Id have to tell you to proceed with caution. If you need a nearfield monitor that plays loud and
offers wider LF bandwidth than most more compact monitors at the price, and you can live with the rough edges, then the Pro
8-3W could well have your name on it. I, however, would be keen to hear a Pro 8-3W MkII... What about it Prodipe?
.
Alternatives
The 8-3W is priced slap bang in the middle of the immensely crowded entry-level active nearfield market, so models from
Adam, Dynaudio, Eve Audio, Event, Focal, Genelec, KRK, Mackie, Yamaha and many others are all alternatives on price.
There are, however, very few three-way competitors at a roughly similar price, one of which, however, is the Event 2030.
AVI Neutron
Five
2.1 Monitor System
This
interesting
monitor
system uses
with the subwoofer.
Tannoy Reveal
601A
Studio Nearfield
Reference Monitors
Building to a
price
inevitably
entails
ones...
Quested V3110
Three-way Active
Monitors
Sometimes,
a dose of
oldfashioned
good engineering
Adam A7X
Monitors
Their A7
nearfield
monitors
received
manufacturer Adam
for improvement.
PMC TB2S AII

Active Nearfield
Monitors
PMC broke
new ground
a decade
ago with
PMCs new activated
Avantone Active
MixCube
Secondary Reference
Monitors
Avantone
have added
on-board
amplification
take on the classic
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Samplecraze Low End

eBook Review
Reviews : Books / Publications / CDs / DVDs / Website
This extensive and in-depth eBook explains so much more

than how to process your kicks and bass sounds!

Martin Walker
Choosing, moulding and manipulating kick and bass sounds can make or break dance, pop, urban and hip-hop music, and
this comprehensive 310-page Low End eBook from producer, remixer, sound designer and SOS contributor Eddie Bazil aims
to dispel the myths.
Focusing primarily on the frequency spectrum between 30 and 300 Hz, the intro neatly covers the basics of fundamentals,
overtones, masking, and the frequency ranges occupied by various bass and drum instruments, before moving on to examine
how to clean up redundant and overlapping frequencies to reduce low-end clutter. We then move on to compressing the bass
sound with a side-chain driven from the kick drum, exploring different EDM styles from enhancement to more extreme
pumping effects, and various urban-music techniques to merge both kick and bass, using tonal drum sounds. To give you an
idea of the depth of this book, this chapter alone runs to 46 pages. Like all the others, it includes loads of screenshots and
associated audio examples, using plug-ins from 16 different developers, so you can hear exactly what the text is explaining
and instantly get your head around the various improvements.
Next we turn to Filtering & EQ and, once again, we leap straight into practical ways to improve separation of our low
sounds, using cuts and boosts at particular frequencies, plus compression and limiting to shape the sound envelopes. This
time, the demos follow six audio stems that, together, form a complex dance groove, and which need manipulating, and in
some cases mauling quite severely, to achieve a homogenised whole.
Working In Parallel covers drum layering, using phase cancellation and EQ to
avoid masking effects, parallel (New York) compression, sample-nudging of layers to
thicken and tighten sounds, and the use of mic preamp and amp/cab simulators for
further blending. Dynamic Management, meanwhile, introduces us to the merits of
splitting sounds into attack and body elements, scooping/hollowing techniques that
create space for low-end movement and percussive reverb, and using saturation to
add definition to your various layers.
The chapter on signal paths focuses on maintaining clarity, even when layering or
changing the sound colour or when adding motion, while Middle & Side explores
various ways to add width/depth, side-to-side and front-to-back movement, and the
use of Mid/Side psychoacoustic effects, which sound amazing as your beats swell and
contract in different directions!
Concluding the book are specialist chapters on how to make the best use of classic
808 and 909 drum sounds in the mix, more advanced mix-bus techniques on the
combined low-end mix, adding groove bounce, and ways to massage pad sounds to
prevent them destroying your now beautiful low end.
So this is not just a book about bass instruments, then! Theres a lot to take in, and
occasionally you might find you get a little lost switching back and forth between the
before/after audio demos while following the host of examples but the rewards are
worthwhile, and perseverance really does pay off here. Being picky, some might argue that theres a little too much reliance on
FabFilters products for some of the treatments, but I can see why Eddies chosen them: their graph-based GUIs display
exactly how the signal has changed, and the excellent audio results speak for themselves.
Overall then, its quite obvious that a huge amount of effort has gone into writing this book, and working through the various
treatment and remix demonstrations also gives you a huge insight into how producers repurpose the same sounds to make
them fit different mixes and genres. While I thought I already knew a fair bit about many of these techniques, I gained a huge
amount of new knowledge to help me combine my own low-end components to maximum effect, and most of all how to inject
life and movement into my grooves. I shall be returning to this Low End eBook again and again and at just $25 its an
absolute steal. Martin Walker
$25
www.samplecraze.com
.
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Small Signal Audio Design 2nd Edition

Book Review
Reviews : Books / Publications / CDs / DVDs / Website
Douglas Selfs excellent technical reference book for solidstate audio-gear designers has been updated.

Hugh Robjohns
Since reviewing the original version of this superb book back in June 2011, Ive found myself referring to it regularly. For those
not aware of author Douglas Selfs credentials, he studied engineering at Cambridge and psycho-acoustics in Sussex, worked
at the sharp end of professional mixing console and highend hifi design for many years, and has several audio-technology
patents to his name. He now works as an audio design consultant, and has authored several excellent audio electronics
books, including the Audio Power Amplifier Design Handbook, and Self On Audio. This new edition of Small Signal Audio
Design (SSAD) expands considerably on its predecessor, with roughly 200 more pages and five more chapters! Ill
concentrate on those new additions to gain a picture of the full breadth of content you can access the original review free
on our web site (http://sosm.ag/douglas-self-ssad-1st-ed).
The opening chapter on electronic basics is unchanged, but the second chapter on electronic components (and their
foibles) has gained 10 pages, and Chapter 3 on discrete transistor circuitry has been expanded three-fold. The first all-new
chapter (5) concerns the use of low-voltage op amps running on 5V and 3.3V single-rail supplies (for battery- and USBpowered devices, for example). This includes a typically thorough evaluation of the performance of suitable op amps, and
some helpful design tips and strategies, particularly regarding grounding practices.
The most developed section of the book, by a considerable margin, is the
section on preamp design for record-player pickups, which has easily
doubled in size, with the original and new material subdivided into five
chapters instead of two. Chapter headings now cover moving magnet
levels and RIAA equalisation, archival and non-standard EQ (for 78s, wax
cylinders and so forth), discrete circuitry stages, noise and distortion, and
moving-coil head amplifiers. As Self says in his preface, the fact that it
takes four whole chapters to cover the process of extracting a reasonable
signal from a record groove indicates to me that there is something amiss
with the whole concept. To which I must add that thats without even
considering the implausible mechanical issues!
Another chapter which has been expanded and split into two concerns
volume controls, where the original section on active volume controls has
been expanded by 10 pages, and the balance-control portion allocated a
chapter of its own. The Tone Control and EQ chapter has also been
expanded to include some very clever new low-impedance designs which
minimise noise. The chapter on mixers gains more on routing systems,
balanced summing amps, and level-indication techniques too, and the
audio-interfacing chapters now include more on using instrumentation
amplifier techniques to advantage, as well as some clever output topologies
including ground-cancelling and zero-distortion transformer techniques. Headphone amplifiers gain an entirely new chapter, as
well, covering hybrid solutions and class-A designs with ultra-low distortion, while the final chapter on interfacing with the
digital domain now examines the use of microcontrollers for housekeeping and other key functions.
Douglas Self provides solid, well-explained technical information throughout the book, all gained from years of experience
and a thorough understanding of the entire topic, and all verified by measurement and practical applications in the field. His
book exudes skilful engineering on every page, and I found it a very refreshing, enjoyable and inspirational read. Self writes in
his preface that he dares to hope that he has moved analogue audio design a bit further forward. He certainly has for me, and
if you have the slightest interest in audio circuit design this book has to be considered an essential reference. Very highly
recommended. Hugh Robjohns
$83.95.
www.douglas-self.com
www.focalpress.com
.
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Sonokinetic The Carnival

Kontakt Instrument
Nick Magnus

Roll up, roll up... A couple of years ago Sonokinetic released Carousel, a rather splendid Kontakt
hosted fairground organ. (The Carnival is programmed for Kontakt 4.2.4 and above, and is not compatible with the free
Kontakt player.) The Carnival is very much a sister instrument to Carousel; whereas Carousel is the larger, more bombastic
type of organ, The Carnival presents two smaller-scale barrel organs, simply named Big and Small, each accessible from the
onepage GUI. The Big is an electrically driven instrument, whilst the Small is a hand-operated type. Selecting either organ is
simply a matter of clicking on the words Big or Small on the interface. The Small organ features single ranks of pipes,
offering four registrations, each with its own key range; Bas runs from Bb0 to A2, Bourdon and Viool occupy Bb2 to A4,
Piccolo goes from Bb4 to G6. A Double button adds a lower octave to the Bourdon, Viool and Piccolo.
The Big organ has a fuller, more demonstrative tone, utilising at least two ranks of pipes (the documentation doesnt
elaborate on this). It also has four registrations, plus percussion. Although three of the registrations are named similarly to the
Small organ, these occupy different key ranges. The Bas ranks are tuned in octaves, running from C1 to C#2. Extending from
D2 to F#3 is the Accomp, similar in tone to the Bas but slightly brighter, also tuned to octaves. G3 to F5 is occupied by both
the Bourdon and Viool. Being the melody section, these are considerably brighter and louder than the lower registrations, with
the ranks tuned in unison. As on the Small organ, the Bourdon and Viool can be doubled in this case its an octave above,
not below. The Tremolo button also applies to the Bourdon and Viool, triggering a repeated note effect for that extra touch of
period authenticity far easier than repeatplaying notes by hand at a consistent rate! The Percussion registration offers four
rather strange, lofi sounds from G#0 to A0; these could be described as triangle, bass drum, cymbal and snare, but theyre
open to interpretation! All registrations of both organs can be activated/deactivated either by clicking on the GUI or via
keyswitches.
Both organs offer control over bellows noise and reverb amount; they also
share clever scripting features intended to make the creation of authentic
performances easier. Those characterful runs often heard in fairground music
can be triggered automatically using the pitch wheel. Five types of run are
provided: pentatonic major, pentatonic minor, straight major and minor, and the
classic Mr Kite chromatic run. Playing a note and moving the wheel upward
triggers an upward run; moving the wheel downwards you guessed it
triggers a downward run. Automated major or minor trills can also be invoked
by moving the mod wheel. Trills and runs are both temposynced. The Carnival
does tend to get a bit confused when attempting to execute runs and trills whilst
playing an entire performance in one instance, so the best workaround is to
program each registrations parts in separate instances on different MIDI
channels. This approach also makes adjusting the registrations relative volume
levels easy. If youre looking for a way to score those Victorian freak-show
scenes, The Carnival is your personal organ grinder. The monkey costs extra.
Nick Magnus
Audio Example
84.53
www.sonokinetic.net
.
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In this article:
Overview
Design & Construction
The Channel Strip
The Indie 500
Masters & Monitoring
In Use
Alternatives
Multiplication Stations
SSL
XLDesk From $19,999
pros
Huge 44/52 input count for
summing.
Versatile VHD preamps.
Freely accessible 16-slot
500series rack.
Clever 500series module
internal routing options.
Builtin Gseries stereo bus
compressor (with sidechain
high-pass filter).
Four stereo mix buses.
42 channel and mix-bus
balanced inserts.
Direct outputs on all
channels and mix buses.
Pragmatic aux/cue and
talkback facilities.
Versatile monitor section.
Intriguing possibilities for
future upgrades/expansion.
cons
Potential for lowlevel fan
noise.
Lack of labelling on
borrowed 500 slots.
No direct metering of
channel PFL or AFL signals.
Additional expense involved
in installation
cabling/patchbay and filling
500series slots.
summary
The XLDesk sits neatly
between the diminutive and
basic XDesk and the far
more complex Matrix 2 and
AWS consoles. With 24
input channels, eight mic
preamps, 44 mixdown
inputs and four stereo mix
buses, this is a much larger
desk than its footprint
suggests. Although it omits
DAW control integration and
fader automation, it does
incorporate a 16slot 500
series rack, and provides
very comprehensive I/O
connectivity.
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SSL XLDesk
24channel Analogue Mixer
Reviews : Mixer
The XL-Desk is the first Solid State Logic mixer that can
host hardware processors in the ubiquitous 500-series
format.
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Hugh Robjohns
SL describe their new XLDesk as the smart dumb analogue console and that sums it up pretty well, because
although its a traditional oldschool console in many ways, its also the result of some very smart design. There are
two common trends in analogue console design at the moment: the integration of increasingly comprehensive DAW
control and automation facilities, and the inclusion of API 500series slots, to allow the users choice of preamps and
processors to be incorporated into the console. For now, at least (I detected hints of future possibilities), the XLDesk ignores
the world of DAW control and automation. Instead, the design teams focus has been on creating SSLs first console to cater
for the 500 series. Sixteen module slots are placed inline with the console channels, and there are two more which, by
default, house the most recent version of the companys Gseries stereo bus compressor (which now boasts a highpass
filter in the sidechain).
The XLDesk may be purchased with all 500series slots empty, or with eight or all 16 slots preloaded with SSLs Eseries
EQ modules. SSLs VHD mic preamps (of which more later) are built in to the first eight of the 16 mono input channels, but
any 500series mic preamps can be plumbed and patched in. If you really wanted to, you could fill all 16 slots with preamps,
patching the last eight into the four stereo line inputs, giving you a possible total of 24 mic preamps for tracking, with all
preamps hosted in and powered by the console.
Overview
In essence, the XLDesk is a generously expanded version of SSLs XDesk (http://sosm.ag/ssl-xdesk), but with scope for yet
more expansion. In a way, the XL-Desk is aimed at a similar market to that catered for by APIs The Box (http://sosm.ag/apithebox): the sort of smaller professional studio thats becoming increasingly common these days. While that device has plenty
to commend it, including the signature API sound, the XL-Desk offers its own sonic signature and far greater versatility. You
might think of it as a big summing box that includes a versatile big console front end and monitoring section, with a wealth of
analogue patching opportunities.
Many will probably reject this console simply because it lacks
DAW control and automation, but this wont trouble SSL, as they
have those areas well catered for at a range of prices, in the form
of the Nucleus, Sigma, Matrix 2, AWS and Duality consoles.
Instead, the XL-Desk is intended for those who want relatively
modest tracking functionality and expansive analogue summing
(using their DAWs level automation) but who require more
versatility and channels than offered by the XDesk. The circuitry
uses exactly the same SuperAnalogue technology thats
employed in all current SSL analogue consoles and much of their
outboard, and so it shares the same ultraclean family sonic
signature. And for anyone whos really desperate to automate this desk, its perfectly feasible to hook up SSLs Sigma remote
controlled summing box (http://sosm.ag/ssl-sigma).
Design & Construction

A freestanding console, the XLDesk is intended to sit on a (sturdy) table top. It measures 1015mm (40 inches) wide by
812mm (32 inches) deep, with a maximum height of 251mm (10 inches), and its quite a heavy beast 41kg, but twice that
when boxed up. Its definitely a twoperson lift!
Two internal 200W universal power supplies (one for the console, one for the 500series rack) are fed from a single IEC
inlet, accepting 100V240V AC and consuming under 300W. The 500series racks supply is generously overspecified, to
guarantee compatibility with even the most currenthungry thirdparty modules. Three quiet fans at the back of the console
draw air in through slots under the front armrest, and expel it from the top of the rear panel. One fan is for the power
supplies, while the other two cool the console electronics and 500series slots. I suppose the fans might become audible in a
really quiet room, but theyre certainly much quieter than a typical Mac Pros fan, and definitely quieter than the air
conditioning in SSLs demo room! (I only became aware of lowlevel fan noise when I crouched down behind the desk to
examine the connectivity.) The fans do their job well: the power supply heatsink was barely warm to the touch after several
hours use, and the control surface remained reassuringly cool throughout.
$19,999 (unloaded),
$27,000 (with eight SSL E
series EQ modules), or
$32,999 (16 EQ modules).
Solid State Logic +44
(0)1865 842300
Virtually all audio connectivity is via 25pin Dsubs (27 of them in total), all wired to the ubiquitous AES59 (Tascam)
sales@solidstatelogic.com
standard. The only exceptions are a pair of XLRs, which duplicate the main monitor outputs, and a third XLR to accept an
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external studio Listen Mic input (for which phantom power is selectable via a DIP switch on the rear panel). The reason for so
many Dsubs is that the mono channels all have dual inputs, everything has a direct output, and there are no fewer than 40
balanced insert points! In addition, each 500series slot has its own independent I/O connections.
On the rear are a USB port for console diagnostics and two unused Dsubs mysteriously labelled AJ and CJ. These
cryptic letters are the initials of the consoles designers, Andy Jackson and Chris Jenkins, and the handbook only refers to
them as being a secret, although the block diagram shows external bus access for future channel expansion options, which
would occupy at least one of these sockets. Clearly, SSL have planned for some level of future development.
Twenty-four input channels are spread across 20 channel strips (16 mono and four stereo). Each input channel is equipped
with two mono aux and one stereo cue sends, just as on the XDesk, but theres no dedicated channel EQ or other
processing; thats what the 500series rack slots are for. However, in typical SSL fashion, the XLDesk is much bigger than it
looks: the dualinput topology allows a nominal 44 inputs for mixdown, and there are ways to bring more inputs than that into
play (see below). Input channels can be routed to the main stereo mix bus (A) as usual, but there are also three more stereo
subbuses (B, C and D) which can be used to create discrete stems for separate section processing before being mixed back
into the main A bus, or recorded back into the DAW.
The 16 mono input channels are provided with a 20dB input level trim control, and an adjacent multi
colour LED indicates the signal level. The levels at which the colour changes varies, depending on whether
the consoles meters are configured for maximum levels of +24dBu (standard) or +18dBu (European
broadcast), but theyre always green below 24dBu, yellow above the nominal operating level, and red
when within -2dB of the selected peak level. Cleverly, the light from these LEDs reflects off the
corresponding trim knobs, improving their visibility considerably.
Pushbuttons select each channels input source (main input or DAW return), as well as inverting polarity
and enabling two separate balanced insertpoints. The first insert loop is intended to access the 500series
rack slot directly above the corresponding channel strip, while the second allows other subsequent
outboard processing. Both these inserts, as well as the corresponding 500series inputs and outputs, are
provided on separate Dsubs on the consoles rear panel, and considerable patching flexibility is afforded if
theyre wired to an external patchbay (the manual details a suitable halfnormalled layout).
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The Channel Strip

The main input for the first eight mono channels comes, as mentioned, from builtin VHD mic preamps (as
employed in SSLs Duality, XRack modules, and the Alpha VHD rack preamp), while channels 916 have
balanced line inputs. VHD is an acronym for Variable Harmonic Drive, which generates predominantly
second- or third-harmonic (valve-like or transistor-like) distortion, to add some sonic character. The amount
of distortion is determined largely by the gain setting, hence the need for a channel trim control post
preamp to tame the peak levels! Usefully, the VHD processing can also be switched out, to provide a clean
9000consolestyle SuperAnalogue preamp, if preferred. Naturally, the VHD preamps can accommodate
line inputs, too, making them useful for adding character to line sources such as synths.
Between +15 and +75 dB of gain is available from the preamps, and other buttons in the preamp section
select a 20dB pad, 48V phantom power, and a HiZ option. The last is not quite what you might think,
though. It raises the input impedance from 1.2k to 10k, which is great for ribbon mics and linelevel
sources but not appropriate for electric guitars and other instruments, which typically require an input
impedance of between 250k and 1M.

explained
Competitions!
The standard
mono channel
strip: the first
eight of these
include the
VHD preamp
at the top, and
all have a 500
series slot
positioned
above.
Some potential customers might question the inclusion of eight builtin preamps, arguing either that theyd rather use
external or 500series preamps, or that theyd prefer a preamp on each and every channel. However, some degree of
pragmatic compromise is needed to deliver a certain level of flexibility at a target cost, and I think SSL have got it about right.
Importantly, the inclusion of the preamps means that the fully loaded version, with EQs on every channel, is ready to use for
both tracking and mixing right out of the box.
The eight builtin VHD preamps are sufficient to make it completely hasslefree to track modest setups, including a drum
kit, in a typical project studio. And for anyone wanting a different configuration, the impressively versatile connectivity makes it
very easy to hook up external mic preamps, when needed, and to bypass the internal preamps completely.
The remaining channel facilities are much the same as those of the XDesk, with the two mono aux
sends and a stereo cue, as already mentioned. The latter can be switched pre/post-fader on individual
channels, while the mono aux sends default to postfader, with global prefader switching via buttons in
the aux master section. Usefully, the stereo cue signal can be derived either from the channels main
signal or from the second (DAW) input so if the channel path is currently using the main input, the Alt
Cue option selects the DAW return (and vice versa). Not only does this increase the versatility of the
cue send system, it also provides additional inputs for mixdown, because the cue bus output can be
routed back into the main A mix bus via a button in the master section.
As with all the companys analogue consoles, the channel faders carry a DC voltage rather than
audio, to control SSLs MDAC circuitry (see box), but as the XLDesk has no automation, these faders
are not motorised. This helps to keep costs down a motorised fader costs five times as much! The
use of MDACs is driven largely by the economies of scale and ease of production that comes with their
universal use across all SSL analogue consoles, but perhaps it also opens up interesting possibilities for
future automation options in the XLDesk. After all, the Sigma uses MDACs too...
Each channels panned output can be routed to any (or all) of the four stereo mix buses, and large
illuminated solo and cut lozenge buttons are provided above the 100mm faders scribble-strip space.
The solo button can be configured in the master section to operate either as a destructive soloinplace
(muting the other channels), or in a mixsafe stereo AFL (afterfader listen) mode. An elegant solo
safe facility is provided to isolate specific channels from being muted when a solo is activated, and
theres also a soloclear button. These are nice, bigconsole features!
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series EQ: the
console can be
bought with or
without these
installed in the
500series
channel slots.
The four additional stereo input channels are not as flexible as the mono channels, as they dont have
alternative inputs and, more importantly, they dont have inserts or 500series slot options. (Although
you can use 500series modules as the input source if you wish, there are no dedicated 500series
facilities in the channels themselves.) However, the 20dB input trim control is retained, and separate
left and right polarityinversion buttons are provided. Theres also a mono button, a pan/balance
control, and the same ABCD stereo bus routing, solo, cut and fader facilities, as well as the two auxes and the stereo cue
send.
Every mono and stereo input channel, as well as the two stereo effects returns and the four stereo mix buses, has a
balanced direct output, with independent pre/post-fader switching for the mono and stereo channel direct outputs. Theres
even a direct output from the Listen Mic compressor! The idea of all these direct outputs is so that all channel signals and mix
stems can be recorded directly into the DAW when tracking, or when building and archiving mixes.
When it comes to mixdown summing, the obvious inputs comprise 16 main input channels, plus 16 alternative inputs via the
cue bus, plus the four stereo input channels and the two stereo returns. That lot adds up to a total of 44 mixdown inputs.
However, the ability to sum the insert returns in the four mix buses adds another eight inputs, to mix an astonishing 52
sources if you really want to!

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The Indie 500

Looking more closely at the 500series rack reveals that this is actually an entirely separate part of the desk. As described
earlier, it comes complete with its own dedicated power supply, and all module inputs and outputs are accessed on separate
Dsub connectors. The 500series slots are primarily intended to facilitate the introduction of EQ or dynamics modules into
the channel signal paths via the dedicated 500 insert points, and the desk ships with a set of bridging Dsub links which
couple the 500 channel insert sends to the corresponding rack inputs, and the rack outputs back to the corresponding 500
returns. Only the last two 500series slots are permanently hard-wired into the consoles signal path, as these are dedicated
to the preinstalled stereo bus compressor which sits across mix bus A.
Of course, the 500series rack I/O doesnt have to be wired into the consoles signal paths; it can also be wired up to other
external devices, if required ideally via a patchbay. When installing 500series mic preamps in the rack, as discussed
earlier, the module mic inputs are accessed via the relevant input Dsubs. Jumper links are provided (on the backplane,
between the module sockets) to allow stereo operation of compressors in adjacent odd/even slot pairs.
I said that only the last two channels were permanently hardwired into mix bus A, but there are some other hard-wired
signalpath options. The last eight rack slots (916) can be
borrowed in pairs, for dedicated internal insert points in the four
stereo mix buses. So slots 9/10 can be switched into bus A,
11/12 into bus B, and so on the idea being to allow 500series
signal processing to be introduced conveniently into the mix
buses. These borrowed 500series inserts are positioned in the
signal path immediately after the mix bus amp, and are followed
by an external insert point. For bus A, the dedicated GSeries
stereo bus compressor is the last processor in the chain, just
before the bus fader.
For convenient plumbing, almost all of the I/O is presented on

the rear panel using DB25 Dsub connectors. The main
monitor outputs are duplicated on XLRs, and theres also an
XLR talkback input, while the consoles master section
provides both a headphone jack output and an unbalanced
mini-jack input for quick auditioning from smartphones, MP3
players and so on.
While the ability to reallocate these 500series slots to the mix

buses is very useful, it would help a lot if the panel below the relevant modules were marked to indicate which modules were
routed where I found myself having to count along the slots each time to make sure I was tweaking the right controls! There
is panel space to add some homemade labels, of course, but properly engraved panel legends would be more appropriate
on a professional console like this.
Masters & Monitoring

The idea of four stereo buses came from the film industry, where its normal to create stems for dialogue, Foley, effects and
music, but it has become popular in music mixing, for grouping instruments according to type, allowing separate bus
processing to be applied before these are combined into a stereo mix.
In the XLDesk, each of the four mixbus sections is equipped with four illuminated lozenge buttons, the first two activating
the borrowed 500series slot and external insert point already described. A third, labelled Sum, provides a direct path around
the external insert point to allow parallel processing or to introduce additional signals to the mix bus. The fourth button
introduces the builtin Gseries bus compressor module on Mix A, but for the other three mix buses it routes the bus output
back into mix bus A. Mix bus A is equipped with a conventional master fader (driving an MDAC, again), while the other three
buses have rotary controls, along with AFL buttons.
The stereo cue and aux send master section of the desk provides output level knobs (with +10dB gain in hand) and AFL
buttons. The two aux send masters also have buttons to select global pre/post-source selection, while the stereo cue master
has a button to route the bus output back into mix bus A. Two stereo returns are also located in the master section, each with
an input level control, mono button, pan/balance knob, AFL, and routing buttons for the four mix buses and stereo cue.
Although the four stereo mix buses each have their own bargraph meters (scaled from 0dB at the top to 48dB at the
bottom), theres no facility to meter the channel input levels directly, nor the aux/cue/foldback outputs or AFL signals. Of
course, DAWs have channel metering, so its not a dealbreaker by any means, but something that might frustrate
occasionally!
The XLDesks monitoring section provides outputs for main and two mini sets of monitors, with configurable level offset
facilities. The mini 2 output can also be arranged to work in mono, and a separate mono subwoofer feed features bass
management crossovers to split the main/mini signal at 80Hz. The monitor outputs pass through 12dB/octave highpass
filters while the subwoofer output uses an 18dB/octave lowpass filter. The console remembers which speaker outputs are
being used with the sub, and switches the filters and subwoofer output on/off automatically.
Additional monitoring facilities include a mono button and AFL and dim level adjusters, and large illuminated dim and cut
buttons are provided. However, theres no provision for a right-channel polarityreverse or differencesignal monitoring, sadly.
The monitoring source can be selected from the stereo mix A bus, an external rearpanel input, or a frontpanel 3.5mm iJack
socket (for an iPod or similar). A sum button mixes any selected combination of monitor inputs together.
Two large talkback buttons are provided, injecting the internal talkback microphones signal into two foldback outputs. One
of these is normally fed from the stereo cue bus, and the other from aux 1, and both have master level control knobs, and
pushbuttons to replace the cue or aux 1 feeds with the currently selected monitoring section signal. As mentioned, the desk
also features a dedicated Listen Mic input, and another large button routes this signal to the monitoring section to allow the
engineer to hear whats going on in the live room. The Listen Mic signal is heavily squashed by SSLs infamous Listen Mic
compressor, and a direct output is provided for recording.
A frontpanel 6.25mm headphone socket has an associated level control, with push buttons to
select the foldback A and B outputs (sensibly, these outputs are minus the talkback mic signal). With
neither button pressed, the headphone feed defaults to the current monitor selection.
The observant will have spotted a curious metal bar running horizontally above channels 1114, and I had a bit of a Doh!
moment when my SSL host picked up his iPad and propped it against the bar, leaning back on the 500series slots. Of
course, a lot of people now use remote control apps on iPads, like Neyrincks VControl, and this arrangement offers a very
pragmatic way of providing convenient DAW control without having to build the facilities into the console. Smart!
DIP switches on the consoles rear panel configure various options, including setting momentary or latching actions for each
talkback switch, and for the talkback signal to replace or mix with the relevant bus signals. Another DIP configures the meters
peak levels for +24dBu or +18dBu, while several more are for reserved functions.
In Use
Sitting in front of the console, everything falls easily to hand, and when loaded with 16 Eseries EQ
modules it looks and feels like a small but highquality SSL console. I imagine this classic SSL
styling suffers somewhat if the 500series rack is stuffed with an eclectic collection of thirdparty
modules, but that approach has its plus points too.
Though Im not a great fan of processing while tracking, I often highpass-filter sources to keep
some control over unwanted low-frequency muddiness but unfortunately neither the VHD
preamps nor the Eseries EQ modules include highpass filters. Of course, this filtering can be
done easily postrecording in the DAW, and postpreamp filters could be inserted courtesy of third
party 500series modules, but some might find the omission of HPFs frustrating.
The XLDesk is a very simple (dumb) yet versatile (smart) console, but the real flexibility comes
when it is installed with a full patchbay to provide more flexible access to the individual 500series
modules, 42 insert points, direct outputs and so on. It also opens the intriguing possibility of creating
a hybrid console, combining 500-series modules from Neve and API (Im told other brands are
available!) to allow the oft-cited perfect solution of tracking through Neve 1073 or API 512C
preamps, before mixing or summing via the SSL SuperAnalogue infrastructure. Thats an appealing
concept, certainly, although all those additional preamps wouldnt come cheap!
I think the XLDesk is very well designed and targeted towards studios involved in smallscale
tracking and overdubbing, with summingboxstyle mixdowns from automated DAW tracks and
stems. The ability to record individual console mix stems back into the DAW for archiving is also
very appealing. The XLDesk neatly fills the gap between the small and limited XDesk, and the
much more featureladen and expensive Matrix 2 and AWS consoles. Naturally, it wont suit anyone
wanting integrated DAW control or fader automation, but that still
leaves a very sizeable market, and at least theres somewhere to
prop up an iPad running a remotecontrol app!
Despite the relatively

compact format of this
console, it boasts a
number of bigconsole
facilities.
Looking to the future, SSL clearly have development plans for

the XLDesk, even if theyre keeping things tightly under wraps
at the moment. Some of the consoles signal switching appears
to be overseen by an internal microcontroller of some kind
(hence the USB provision for firmware upgrades), the faders
operate MDACs, and there are those mysteriously labelled D
subs on the back panel too. It doesnt seem too extreme to
The desk sections are angled for ease of reach.
imagine that there may be potential already built into this desk
for some level of useful integration with DAWs in a future update even if only mute automation or remote MDAC control
(like the Sigma). Theres obviously also some provision for a possible expander unit to increase the channel counts...
However, when I mused along these lines the SSL gurus just smiled knowingly, so well have to wait and see.
Regardless of what happens in the future, the XLDesk is already a very impressive console, in terms of what it can do,
what it costs, and how it sounds. It should have a great future.
.
Alternatives
There are several consoles that offer 500series slot options in various formats and channel counts. For example, the
Ocean Audio Ark provides two 500series slots per channel, six auxes, eight buses and a stereo master. The Inward
Connections Tree 500 console provides three 500series slots per channel, with four auxes plus a stereo cue, four mix
buses and a stereo master. APIs The Box console provides four mic preamps with two 500series slots, plus 16 summing
channels, with two mono and two stereo aux/cue sends. Purple Audio, AwTac, Tonelux and Petes Place Audio are among
the generous handful of others who offer consoles based on the 500series format.
Multiplication Stations
MDAC is an acronym for Multiplying Digital to Analogue Converter, a unique technology which SSL introduced with the
Duality console. Despite the name, the audio signal path is entirely analogue, but its governed by a digital control signal
thats derived from the fader (or an external data signal). You can think of the MDAC as a kind of voltagecontrolled
amplifier or VCA (although strictly, it is a DCA).
In a conventional DA converter, a digital audio sample describes the required amplitude of the output voltage
corresponding to a specific moment in time, and the DA converter essentially attenuates a peaklevel reference DC
voltage to deliver the required output voltage for each sample thus producing the varying analogue audio output. The
MDAC system works in a similar way the channels analogue audio signal effectively becomes the reference instead of
a fixed voltage. However, the digital control data still instructs the DAC to attenuate the signal by the required amount, so
the audio signal itself remains entirely in the analogue domain.
The benefit of the MDAC approach is that it provides excellent distortion and noise performance, while attenuation
settings can be recalled precisely, and are easily controlled from a DAW or other remote device.
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Let There Be Light
Performance
Alternatives
Studiomaster
Starlight 12ML $392
pros
Provides all you need for
PA, monitoring and lighting
in one box.
Sensibly priced.
Clear vocal sound.
cons
Direction of lighting and
monitor speaker is fixed.
summary
Im pleased that somebody
followed up on this all in
one idea, and happier still
that it comes in such an
affordable form! Maybe the
idea will catch on...
information
$391.80 per speaker.
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Studiomaster Starlight 12ML

PA, Monitor & Lighting System
Reviews : PA system
Could this crafty three-in-one PA and lighting system save

you both time and money?
Buy PDF
Paul White
couple of years ago I was talking to PA manufacturers, trying to promote the idea of a speaker such as this one, and
Paul Belcher, the main designer at Studiomaster, was the only engineer to take up the challenge. My pitch was that,
since many of us now play live in small pubs and clubs where theres barely enough room for the performers, let alone
the gear, wouldnt it make sense to build the PA speakers, some basic foldback monitoring and LED stage lighting into a
single, compact cabinet? That would cut down on the number of stands and amount of cabling required, making for less
clutter and a faster setup time.
Studiomasters first product of this type to be unveiled was the Starlight 15, but this 12-inch version looks like a more
practical proposition for smaller venues, measuring approximately 400 x 340 x 600 mm and weighing 22.7kg. Personally Id
also like to see a model in the range with 10-inch or 2x8-inch speakers, as that seems an ideal size for many smaller pub gigs
but lets see what the Starlight 12 is all about first.
Manufactured in China to keep the price as low as possible, the front-ported cabinet is made from 15mm MDF and has a
very neatly applied fuzzy fabric covering as well as a robust, powder-coated metal front grille and inset side handles. A
standard top-hat pole-mount socket is fitted to the bottom face. Pretty standard so far then. A 12-inch woofer with a 60mm
voice coil sits below a 1.35-inch horn-loaded compression driver, the horn having a one-inch throat and a 60 x 90 degree
dispersion. The drivers are fed 250W of amplifier power via a 4kHz passive crossover. The speakers can operate from 230 or
115 Volt supplies as set by a selector switch.
A further 50W amplifier is present to drive a four-inch monitor speaker, which is
sited in its own internal sealed enclosure and has a limited 200Hz to 8kHz
bandwidth which is similar to many stand-alone mini monitors. Theres no need
for the monitor to go any lower than 200Hz, as the low-end spill from the main
speakers fills in the necessary bass end. This speaker, which also has a metal
protective grille, is fitted into the angled side of the cabinet along with a recessed
strip of six high-intensity, multi-colour LED stage lights fed from an internal lighting
controller. The cabinet side is angled at around 30 degrees, so gives the best
lighting and monitor coverage when the performers are behind, but not too far
behind, the speakers. In a typical small venue this might only be a couple of feet or
so move much further back and you start to lose coverage, unless you angle the
speakers in slightly. Clearly this layout necessitates the speakers being built as
mirror-image pairs so that the monitors and lights point inwards towards the
performers.
Let There Be Light

Integral lenses give the LEDs a 45-degree coverage angle, and the on-board
lighting controller offers 13 programmes comprising seven static colours and six
auto-changing, cross-fading or sound-activated colour changes. The user can adjust
the overall brightness and speed of change for the relevant programmes.
When the speakers are set up on stands, the monitors are at a nominal ear
height. The monitor source can be from the main speaker feed, or you can plug in a
feed from a mixing consoles aux sends if you need a custom monitor mix. All the
controls and connections are on the recessed rear panel. The external monitor
input is on a quarter-inch jack, and the monitor speaker also has its own level
control.
A simple two-channel mixer section serves the input section, with separate level
controls for the inputs and master, an overall three-band EQ and a balanced XLR
link-output connector to feed an additional powered speaker. Theres one mic/line
channel (XLR and jack inputs, respectively), and one line-only channel that can
accept either an XLR or a pair of RCA phono connectors (the latter are summed to
mono for connecting MP3 players and the like). At 30dB mic gain the noise figure is
-87dB, and the circuitry bandwidth extends from 40Hz to 30kHz.
A series of push buttons on the same recessed panel select the lighting
programme (or turn the lighting off), set the light brightness (static modes only) and
set the colour-change speed. Theres also a sensitivity control for those
programmes that employ a sound-to-light mode. Status LEDs indicate the lighting
mode, power on and also monitor peak levels.
Performance
The integral LED lighting column and, to

its left, the four-inch monitor speaker.
Given that these are not expensive speakers, the sound quality is actually pretty
good, though, as is often the case with smaller ported PA speakers, the bass end can get a bit flabby when playing back fullrange material at any volume. Thats where the three-band EQ comes in handy, as the lows can be tamed very easily, and
having EQ also allows for some adjustment to compensate for speaker placement and room acoustics. If you want to run a
techno show then add a sub, but for voice, the speakers cut through well with good
intelligibility, and again the EQ can help fine-tune the result though that 4kHz
crossover frequency does mean that the woofers are approaching the end of their
HF comfort zone. That might be another reason to add a 10-inch version to the
range. More money will buy you a more hi-fi sound, but Ive always been
impressed by Studiomasters ability to squeeze the best performance possible out
of a given budget, and these speakers dont disappoint in that respect. Theyre also
pretty loud, with an SPL rating of 124dB.
If you listen to the monitor speaker in isolation it sounds predictably mid-heavy
and squawky, but thats not how youd hear it in a real gig. The trick is to turn it up
until it restores clarity to what is spilling back from the front, at which point the tonal
balance you hear when performing comes pretty close to what the audience hear
out front. In a small venue where theres no space for conventional monitors, having
what amounts to built-in side fills is a big plus.
Of the lighting options I found the auto colour fading set to a slow speed to be
most useful. It isnt too flickery or flashy, as the sound-to-light versions can be
sometimes, it is less obtrusive than straight colour switching, and less boring than
static colours. Considering there are only six LEDs the lighting is incredibly bright
The Starlight 12MLs built-in input mixer.
and it goes without saying that this is infinitely better than relying on the pubs strip
light. In a perfect world the lights and monitor would be on swivels allowing the user
to aim them as appropriate, but from an engineering standpoint this would have resulted in a far more costly product.
Studiomasters design here is clearly pragmatic, and because of that theyve managed to offer a great deal of functionality for
no more than youd expect to pay for a standard powered speaker. Theres also a slightly cheaper version (Starlight 12L) that
retains the lights and dispenses with the monitoring if you dont need it.
Our real-world test was carried out at a medium-sized pub gig with a four-piece band, using typical backline and a full drum
kit. We decided to use a mirror pair of Starlights for PA, carrying mainly vocals, and also as the sole source of monitoring. The
vocal level was plenty high enough to be heard over the backline and resistance to feedback also turned out to be better than
expected, even with the monitors turned well up. Trusted ears in the audience said the vocal clarity was fine, while the band
members were very happy with the cross-fill monitoring, as everybody could hear the vocals clearly. Not only did this avoid
separate monitoring with all the extra cabling that entails, but it really made setting up very simple and quick. There were
some protests from band members that the LED lighting was a bit like having laser eye surgery, though turning the cabinets
slightly helped avoid having the LEDs shining directly into anyones eyes. When I told the band members what the speakers
cost they were very surprised, as they expected them to cost considerably more. They also thought the idea of having
everything built into one box was a good one, so my madcap concept would seem to be vindicated!
.
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explained
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Alternatives
Currently there is nothing else that combines the roles of main PA, monitors and lighting in one the only alternative is to
buy separate boxes for each.
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TC Electronic PolyTune 2 Mini

Polyphonic Guitar Tuner Pedal
Reviews : Accessory
Paul White

Guitar tuners now come in many guises, from clip-on devices, plug-ins and phone apps to pedals
and rackmount offerings. TC Electronics PolyTune was the first polyphonic guitar tuner pedal however, and provided at-aglance confirmation that each of the six strings were in tune or werent! A half-width mini version (with no battery-power
option) followed, but both designs have since been updated, to provide better visibility and enhanced features.
Available in black or white, and built into a tough metal case, these latest PolyTunes have a much brighter display that
adapts more effectively to the ambient light level, and it can be set to tune either guitar or bass. As before, it accommodates
alternative drop-D and capo tunings, but this time around you can also choose needle or strobe display modes. A mini USB
port allows firmware updates to be loaded from a computer.
As before, a true-bypass switch mutes the output while tuning, but there
have been numerous internal improvements that arent immediately evident
to the user. If you pick one note at a time, the new PolyTune 2 devices
function as standard chromatic tuners. Play multiple strings, however, and
theyll flip to polyphonic mode, with green LEDs lighting up when the guitar is
in tune, and a pair of red LEDs above and below each one indicating sharp
or flat respectively.
A newly added Display button steps through the four options of guitar or
bass and needle or strobe tuning. The needle mode adopts a familiar pointer,
while strobe mode provides a display that drifts to the left if the note is flat,
and to the right if sharp. As you get closer to the correct pitch the strobe
slows, stopping when you are in tune, at which point a couple of arrowheads
also appear. The needle modes 0.5 cents accuracy is good enough for most
purposes, but the strobe mode is accurate to 0.1 cents, which rivals that of
the high-end tuners used in guitar workshops.
To reset the tuner to drop-D tuning, in which the low E string is tuned down
to D, you need to press and hold the footswitch for about three seconds.
Pressing and holding the footswitch again restores normal tuning. To set
capo tunings, the small Tuning button on the side of the tuner case is used to
step through the options, which go as low as tuning the E-string down to B or up as far as the B above, all in semitone steps.
The reference pitch can also be changed from the standard A=440Hz by using the Tuning and Display buttons in
combination. User settings are stored after powering down.
The manual suggests selecting the guitars neck pickup and using the thumb to strum the strings while tuning, but on my
Strat I found it pretty solid on any pickup setting playing with a finger or a pick. I found the display much brighter and easier to
read than in the earlier versions, and although the Mini requires external power, it has the same ubiquitous connector and
voltage requirements as Boss pedals.
While software tuners are fine for some applications, a pedal such as this is ideal for use on a pedalboard or for making
particularly precise guitar adjustments, and the display is now bright enough to see clearly at outdoor gigs. In all, then, these
are excellent and compact tuners. Should you prefer battery power then the full-size PolyTune 2 offers just the same features
as the Mini, but with a larger display area. Paul White
PolyTune 2 Mini $89.99, PolyTune 2 $99.99.
www.tc-electronic.com
.
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In this article:
First Impressions
Can I Kick It?
Accessories
Alternatives
Telefunken
DD4 $1149
pros
Well constructed with an
iconic look.
Good, solid, workmanlike
sound, particularly the M82
on kick.
The microphones are well
suited to the jobs for which
they are intended.
Very nicely and practically
packaged in a case that
actually fits everything
comfortably!
cons
A bit pricey for a set of
microphones when you
could possible pick and
choose a better set by
looking further afield.
summary
A quality set of microphones
that should last a very long
time, and comes with an
excellent supply of useful
accessories.
information
$1149.
Telefunken
Elektroakustik +1 800
882 5919
info@t-funk.com
www.telefunkenelektroakustik.com
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Telefunken DD4
Dynamic Drum Microphone Set
Reviews : Microphone
Renowned microphone brand Telefunken tackle the drummic kit market, with this high-end collection of dynamic
models.
Buy PDF
Jon Burton
he Telefunken drum-mic kit arrived to a small buzz of excitement in the studio. The Telefunken marque has always been
associated with quality sound and construction, so releasing a set of drum microphones didnt seem to sit in with this
high-end, normally vocal-microphone-driven market. Telefunken however, like many other manufacturers, cottoned on
to the fact that there have been huge gaps in the microphone market for a long time, and while other manufacturers started
selling drum microphone kits some time ago, these have always tended to be somewhat downgraded affairs in order to keep
the kit price reasonable. So how have Telefunken fared? Have they gone the same way or headed for the top end of the
market?
Having a kit of microphones like this has many advantages. It gives you instant access to a set of microphones designed to
do a specific job. It also hopefully gives you a clear and coherent sound across the whole drum kit, with microphones
designed to bring out the best in each kit element. Which microphones the manufacturer chooses is usually a reflection of
their speciality. Companies go down different routes supplying complete kits, including overheads, but Telefunken have
instead provided a basic set that covers kick drum, snare and two (DD4) or three toms (DD5). This really is the bare minimum
anyone will ever need. We were sent the DD4 set, with just the pair of tom microphones. Luckily our guest kit had only two
toms, a mounted rack and a floor tom. We did have an extra snare to contend with, but luckily we had no shortage of in-house
microphones.
To supplement the Telefunken we added an extra boundary mic in the bass drum, and a selection of condensers on snare
bottom, hi-hat, overheads and room mic.
First Impressions
On first examination all the microphones have a certain classy look about them. They come in a range of grey to black shades
the kick mic in black, snare in dark grey and toms slightly lighter. The bass-drum mic is actually vaguely reminiscent of a
Neumann U47 in shape. All the microphones share a solidly engineered look; the iconic Telefunken badge does not look out
of place on the weighty metal shells.
The general quality of finish on the set I saw was pretty high,
looking like something that was crafted rather than just churned
out. The big question, however, is how did they sound. Ill take a
look at each microphone individually.
Can I Kick It?

It is traditional to start with the bass drum. The M82 bass drum
microphone that is included in the kit is a substantial affair only
just narrower than the hole in the bass drum we were trying to
place it through. It is cylindrical, 155mm long with a diameter of
63mm. This houses a 32mm dynamic moving-coil capsule. With
a stated frequency range of 25Hz to 18kHz, it is able to stand
sound levels in excess of 146dB. This chunky mic needs a sturdy
stand though, and is not that easy to manoeuvre into position; I
found the mount was more restricting than some as it seemed to offer less than 90 degrees of adjustment. This tends to leave
the mic stand needing to go in front of the drum rather than tucked alongside. Not a problem in the studio, but on tight
stages...
It has two EQ switches: the first of these engages a passive filter that reduces some of the low-mid frequencies around
350Hz, the second switch introduces a high boost which tilts the upper mids and highs, rising from 2kHz and eventually
resulting in a 6dB boost at 10kHz. These are both very useful and clearly marked. I must admit to invariably favouring the lowmid cut, and the M82 sounded great with this switched in. The high boost was more a case of personal preference, but it does
help if you are looking for a more aggressive modern sound. It will suit some users more than others but is a handy feature.
I thought the M82 was a great, solid microphone. It didnt suffer from too soft a low end, which some large-diaphragm
microphones can do, and even with the high-frequency boost I didnt find the microphone at all harsh or aggressive. Saying
that, if I want to go for a clicky kick-drum sound I would be looking at using a boundary microphone inside the drum instead.
Overall though, I thought the M82 was a very adaptable-sounding microphone.
Time to turn our attention to the other two models in the range. The M80 SHB is recommended for snares, and also for
vocals where a compact microphone body is required. As it has a built-in mic clip on the side it would be awkward to hold, but
for stand use it would be ideal. The frequency response, as you can imagine, has been tailored for its two intended roles, and
it has been designed to minimise the proximity effect. It has a stated range for 30Hz to 18kHz, and a supercardioid pick-up
pattern. It features a gently rising top end which helps capture the snares nicely. Able to handle SPLs in excess of 140dB, it
should be able to cope with even the hardest-hitting drummer. Certainly, in our test, it didnt seem at all bothered by even the
punchiest rimshot.
All in all it seemed to do the job well. This is however a really
crowded market and I am not sure what the M80 offered over
some of the more traditional choices of our mics case. In its
favour, though, is a very solid construction and really nice finish.
Tom-toms are handled with a variant of the M80, called the
M81 SH. Similar in appearance to the snare mic, it has the same
textured powder coating but in lighter grey. This provides a more
distinctive look and, once again, the construction is solid and
impressive. The M81 differs from the M80 in that it has been
designed without the rising high end, and so has a flatter overall
response. This does lend the microphone well to toms, giving it a
generally tight sound.
Audio-Technica
AT4047 MP
Each mic ships with its own mounting hardware (both plastic
Having a short body makes it is easier to get this of
and, as pictured, metal) and 5M right-angled XLR cable.
microphone into relatively tight places, but they are still not that
compact compared to other models in the market. It was,
however, relatively easy get a good sound from the mics quickly and easily.
Accessories
Telefunken have bundled in a few important extras with the kit. Not only do the microphones come in a very robust plastic
hard-shell case, they also come with a variety of drum mounts. The M80 and M81 can be mounted on the drum rim using
either the plastic clip-on mount, or the metal thumb-screw mount. Why two types of mounts were needed I am not sure,
although the metal mount was more flexible as it could be used in a greater number of ways.
The mounts hold the microphones OK, but they are quite heavy mics so you need to make sure the weight is distributed
evenly or they may sag. The microphones also come with some nice 5M mic cables that are terminated at one end with a
right-angled XLR connector. This has the major advantages of making the microphones less obtrusive, easier to place
securely, and putting less weight on the rear of the microphone. The overall look with the cables in place is very neat, and the
logo on the connector adds to the classy appearance.
As individual microphones they all perform well, particularly the M82 on bass drum. However, as a set, the added bonus of
the clips, cables and the uniform sound they are able to provide is a draw. If I was investing in a mic set for the first time this
would be a serious contender. It is not a particularly cheap set, but it will be with you for a long time and could easily prove to
be a very wise investment over the years. I had absolutely no problem using these, and the addition of cables and clamps is
very sensible and gives you a professional case of mics that should cope with most jobs easily.
.
Alternatives
Many microphone manufacturers sell sets intended for drums, notably Audix and AKG, but few are of this quality and
come with such a useful set of extras.
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...
MICROPHONES For Sale in
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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:
Concept
Ins & Outs
Tone Controls
Easy Squeezy
On Test
Conclusion
Alternatives
Thermionic Culture
Nightingale 2 $3999
pros
Beautiful and classic
Thermionic Culture valve
sound.
Useful for dual-mic or micplus-DI recording setups.
Dedicated I/O for both
channels and the
compressor.
Thoughtful combination of
EQ controls.
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Thermionic Culture Nightingale 2

Channel Strip
Reviews : Recording Channel
With its parallel inputs, Thermionic Cultures valve-based

channel strip enables all sorts of creative recording
options. And it sounds great, too!
Buy PDF
Matt Houghton
lmost all of Thermionic Cultures valve processors are a little different in their own way, but I cant think of anyone else
who would have come up with a concept like their Nightingale. In fact, on first inspection, it struck me as being a little
odd, yet after spending a while with it a year or so ago, this quirky creation began to make rather wonderful sense. At
that time, I suggested the tiniest of feature enhancements (an EQ bypass switch) to Thermionic Cultures Vic Keary, and they
decided to incorporate that in a new version, the Nightingale 2, which I have in front of me now.
Concept
The basic idea is that you have dual-mic/line/DI channels, each with a three-band valve EQ. Either channel can be fed to the
third channel, which features a valve compressor. Furthermore, the first two channels may be summed, so that both are fed to
the compressor channel. This means that you can use the Nightingale as a conventional tube recording channel
cons
(preamp/EQ/compressor) or specifically to track dual-mic or mic-plus-DI setups (preamp/EQ+preamp/EQ, all running into a
Lacks mid-range EQ cut
compressor). The first two channels each have a polarity-invert function, so the Nightingale 2 would seem, on the face of it, to
facility.
be useful for tracking things like top and bottom snare mics, or dual-mic guitar recordings, for instance, or perhaps a DI-plusThe mono compressor limits miked-bass cab. In each case, you craft the sound while tracking and record to a single channel, in the time-honoured way,
the appeal for use on stereo
avoiding the modern self-inflicted problem of too many options at mixdown.
sources.
summary
Despite the slightly out
there form factor, the
Nightingale 2 is a tracking
channel strip of rare
versatility. Even rarer is that
this versatility is combined
with such a gorgeous, easygoing sonic character.
information
If thats not flexible enough for you already, the compressor channel has its own dedicated line-level input, and every
channel its own line-level output so you could track both mics to separate channels, or choose to use this as three separate
line-level processors while mixing. Maybe three really is a magic number.
Ins & Outs

Each designed around a 12DW7 (ECC832) valve, the Nightingale 2s two preamp channels can accept mic or line signals via
the XLR inputs on the rear, or a DI signal via the dedicated TS socket on the front. For mic/line signals, a knob near the large
GR meter is used to select the appropriate gain setting. The 20-detent gain knobs (indeed all the gain knobs, whether in the
preamps, the EQs or the compressor) are marked 1 to 10. These span the range from full attenuation to +5dB for line
sources, +21dB for the Hi mic setting, and +34dB for the Lo mic setting. Plugging an instrument jack into the DI socket autoselects the DI as the input source. Theres switchable 48V phantom power for each of the preamp channels, via a fourposition toggle switch (off, A, B, A+B), with an LED per channel to indicate the current setting.
$3999.
Independent Audio +1
207 773 2424.
fraser@independentaudio.com
www.independentaudio.com
www.thermionicculture.com
The third channel accepts external line-level sources via a

rear-panel XLR, but also features a source selector, enabling you
to switch from that to the channel A input, channel B input, or
channels A+B summed signal, before routing this to the
compressor.
The power supply is switchable to accept 110V (USA) or 240V
(UK) mains power courtesy of the ubiquitous three-pin IEC inlet.
Tone Controls
The EQs are intended for use as gentle tone-shaping controls.
Ive already described the EQs as three-band, because the active
control comes courtesy of Top, Pres(ence) and Bass knobs, but
technically they could be described as six-band, as switchable
top-cut, low-shelf, and bass-cut filters are also included.
The Top control applies a high-frequency (9.5kHz) bell boost,
with the bandwidth getting narrower the more that boost is applied. Its a continuous control which runs all the way from 0 to
+12 dB, which is quite a hefty boost yet it manages not to sound overly harsh when applied. Pres governs a mid-boost
(2.8kHz) bell EQ, with similar variable-Q characteristics as the Top band, with a maximum gain of +11dB. Bass is another
boost EQ offering 0 to +11 dB of gain, but this ones a little different. Its a varislope design, in which both the slope and
frequency changes with the amount of boost, ranging from flat (off) through a small 90Hz peak at the 5 setting to a heftier
peak at 40Hz when set to maximum.
The Bass Cut and high-pass filter controls are intended to be used in tandem with the Bass boost. The first offers two lowfrequency shelving options, and the idea is that you can use it to cut in the low mid-range, while using the main Bass control to
restore the level of the lower frequencies. Hence, the Bass Cut turnover frequencies being, slightly counter-intuitively, at 200
and 800 Hz: the gentle slopes mean that the first option is 7dB down at 80Hz, while the latter is 6dB down at 200Hz. The highpass filter, of course, ensures that no unwanted nasties are allowed through when boosting the low end. It offers 30 and 80 Hz
options (6dB down at these positions, but with the curve getting progressively steeper lower down the spectrum).
The Top Cut control offers three settings bypass, and two high-shelf attenuation options, one being -6dB at 8kHz and the
other -8dB at the same frequency. This can be used in conjunction with the Top boost knob, in a similar fashion to the bass
controls, but on its own its also a useful tool for gently softening a source, just taking the edge off overly bright sounds.
By any modern standards, these EQ controls are very broad-brush tools, but then theyre not intended to be surgical
instruments rather, theyre for gently massaging the source to achieve a pleasing character, and used in that way, its
amazing quite how forgiving they are. You can achieve some truly beautiful results which sound that little bit larger than life, as
is the case with so much high-quality valve gear. (Ive yet to hear any plug-in do this sort of sweetening job quite so well.)
Easy Squeezy
The compressor, the design of which is based around three separate valves, is as simple to operate as youd expect of an oldfashioned tube design. A Threshold control switches the compressor on or off, and is used to choose one of three threshold
settings. The ratio and attack time have no controls, but the release can be set to slow, medium or fast. Dialling in the desired
amount of compression is then a matter of juggling the input (Comp Gain) and output (O/P Level) controls. The former dictates
how frequently and by how far the signal exceeds the threshold, while the latter is an attenuator. For anyone used to using a
modern compressor with variable threshold and make-up gain, having an attenuator last in line might seem strange, but it
explained
makes good sense when you consider the signal path: in order to get the signal to exceed the threshold, you add gain to the
source, so while the compressor reduces the dynamic range, your signal still ends up being louder than when it went in. The
output level knob allows you to attenuate the signal to pass it on at a sensible level to whatever youve placed next in your
Competitions!
signal chain.
On Test
To kick off my tests, I ran a number of different line-level sources through the preamp/EQ channels, just to see what the
general sound was like. Without even applying any EQ, rock drum loops were lent a lovely vintage-sounding sort of vibe
ever so slightly gritty, but in that pleasing flair sort of way that Ive only ever heard with good valve gear. I soon realised that I
could use the two channels for stereo processing, adding yet another string to the Nightingales bow. The only downside was
that using this in stereo for drums left me also wanting a stereo compressor!
Putting an electric bass sound (a line-level recording) through
the EQ was particularly rewarding: the top band just rolled off
any lingering harshness and twang from the picking of the
strings, while the Bass control, combined with a 30Hz high-pass
filter gave a wonderfully full and warm sound, without ever
feeling overblown or flabby. This only got better when running the
output of that channel into the compressor, which was smooth
yet disciplined in its control, and again offered a distinctly oldschool sound. Lovely. The DI input, which I tested with a lawsuitera Rickenbacker bass copy, yielded results that were just as
pleasing as when using a line source.
Not having a bass amp to hand, I also tried miking up a guitar
cab (Les Paul Studio into a Fender Twin, with an SM57 close
Three devices in one: each preamp/EQ channel has
mic), while at the same time capturing a DI signal (via a DAV
dedicated audio input and output connectors, as does the
DI/Reamp box). I found it reasonably easy to balance the two
compressor channel.
signals. Ideally, given the lack of attenuation options on the main
EQ controls, Id have had a lower option to be available for the high-cut filter. Still, the device added a nice dimension to the DI
signal, making it a little less flat or dull-sounding, and I was able to track to both main channels and the compressor channel to
three separate tracks on my DAW.
Yet for all this, where Thermionics gear has really shone for me in the past is on vocals and on acoustic guitar, and it was
the same with the Nightingale 2. On both sources, the mic pre and EQ combined to make a beautiful recording channel, which
just made tracking so easy. The Mid and Top EQ boosts were particularly useful, adding a nice sense of breath (to the vocals)
and air (to both), while being able to tame the worst excesses of a finger-squeak here and string-scrape there on the guitar.
I hear a lot of people preaching hate about using compression when recording, claiming that its an unnecessary evil given
the low noise-floor of 24-bit recording. Despite the technical case to support that view, I think it a bigger evil to leave too much
of that sort of decision making until mixdown, and I almost always find that applying a little gentle compression is useful, as
long as I leave a little room for manoeuvre when mixing. It makes the job of mixing so much easier, and I find that thinking
about this sort of thing helps me to appraise the sound Im capturing rather better. And its this job of gently compressing
things that the Nightingale 2s compressor is so good at. I say gentle, because thats how it sounds, but it was quite disarming
to watch the GR meter while tracking a vocal: what sounded like an innocent amount of compression turned out to be
somewhere around 9dB! The beauty of the dedicated line input for the compressor is that it can always run the recorded track
back out through it for another pass later on.
Conclusion
Outwardly, this may seem a quirky device, but when you break it down into its constituent parts its a very neat package
indeed. Its an appealing toolbox for general tracking duties, and an excellent one for vocals in particular. With the second
channel, the mix input to the compressor, and the dedicated I/O for all three channels, it extends the functions of a recording
channel in a useful way and, importantly, offers some serious firepower when it comes to mixing.
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Of course, the worth of valve gear isnt just about the functionality: theres also the subtle-yet-beautiful, musically pleasing
sonic effect of running audio signals through thermionic valves. While tube gear doesnt inherently have to sound warm, most
of Thermionics gear does I have a strong hunch that theyre partly to blame for the perception that tubes equal warmth,
simply because their implementations do such a good job of it!
All of this doesnt come cheap, of course but two preamps, two EQs and a compressor, all high-quality valve types, is
actually a lot of gear for the money.
.
Alternatives
Theres not really anything quite like this on the market. There are channel strips, of course Thermionic Cultures own

S8 monitors
Win Munro Sonic Egg
150 (Global Except
Americas)
Win Yamaha White
Christmas bundle
(Americas Only)
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In this article:
Famous Names
Plug & Play
RMX Bandits
Character Actor
Alternatives
Standard Programs
Additional Hardware
Programs
UA
RMX16 $349
pros
Algorithm code bytefor
byte identical to original
hardware.
Precise emulation of
hardwares sonic character.
Handy directtype entering
option for parameter values.
It produces beautiful
sounding and versatile
classic reverbs!
Numerous celebrity presets
included.
cons
Only the nine factory
programs included, not the
nine optional additions.
High DSP load limits the
UAD plugin count
significantly.
High cost although still
far less than a hardware
unit!
summary
UAD have worked directly
with AMSNeve to hit the
mark emulating yet another
piece of classic audio
hardware.
information
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UA RMX16
Reverb Plugin For UAD2 Platform
Buy PDF
Reviews : VST plug-ins
UAs emulation of the AMS digital reverb shows exactly

why the original is still regarded as a classic.
Hugh Robjohns
dvanced Music Systems RMX16 digital reverberation unit is most definitely one of the alltime classic audio
processors. The manufacturers, more usually known as AMS back then (and AMSNeve now), were set up in 1976 by
former aerospace technicians Mark Crabtree and Stuart Nevison, who shared a passion for audio and expertise in
digital electronics. This was at a time when digital audio processing was exciting and expensive cuttingedge technology.
The company really appeared on the industrys radar when they launched their second product, the microprocessor
controlled DMX1580 (15bit) digital delay line, in 1978. This product appeared in several variations over many years, each
with increasingly enhanced capabilities, but of relevance here is the unique addon expander AMS introduced for the 1580 in
September 1981 called the DMX15R.
Digital delay units became extremely popular in the late 70s and early 80s, but digital reverb technology was very new and
very expensive! Most commercial reverb systems at that time still used mechanical plates or springs, and only a few
companies had developed highquality professional digital reverberators. The DMX15R was AMSs first attempt, and it was
an addon 2U rackmount unit which bestowed the 1580 with sophisticated programmable reverberation facilities.
Although immediately popular and immensely capable, the
DMX15R expansion unit wasnt particularly practical, and it was
superseded just six months later, in March 1982, by the RMX16.
In essence, AMS reengineered the 1580/15R combo into a
selfcontained standalone product: the worlds first integrated,
microprocessorcontrolled digital reverberation unit. Its front
panel was the same as that of the 15R, except for the addition of a pair of knobs to set the input and output levels, and the
first edition of its Operating Instructions was still titled RMX16/DMX15R too, since the two units operation and programs were
identical.
Famous Names
The leading lights in the digital reverb arena in 1982 included EMTs 245 and 250, Eventides SP2016, Lexicons 224, Sonys
DRE2000 and Ursa Majors SST282 Space Station many of which are still in use and revered today! However, although
the AMS RMX16 faced pretty stiff competition, it did very well indeed, largely because of its very musical character and superb
usability, not to mention its compact size relative to many of its peers.
We take microprocessor control for granted today, but in the
$349
early 80s it was still quite a new and unfamiliar technology. In
Source Distribution
the RMX16, it was employed to manage the internal reverb
+44(0)20 8962 5080
algorithms, load alternative algorithms, allow realtime display
sales@sourcedistribution.co.uk
and adjustment of each algorithm parameter, and to store and
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recall 99 user parameter presets. The RMX16 could be
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controlled remotely, too, with an optional keypad unit (no

Lexiconstyle faders, sadly), and a revolutionary barcode
scanning wand was available to import new reverb algorithm
data from sheets of printed barcodes. In fact, AMS developed 18
separate reverberation programs in total, but only nine were
stored in permanent ROM within the machine (locations 19).
Three more could be loaded into dedicated batterybackup RAM
(locations 1012) within the rack unit, and another three in the
Remote (locations 3133), if required. (Thirdparty update kits
are now available that install all of the programs in expanded
hardware ROM.)
With the addition of a separate board called the DMX 15R,

AMSs DMX 15-80S delay (centre) could produce digital
reverb. The now-classic RMX16 (bottom) was in effect a
single-box version of this combination, and was hugely
successful. This fine-looking pair occupy rack spaces in
Henson Studios.
Photo: Hannes Bieger
As you would expect, some of the nine factory algorithms

simulated a range of conventional acoustic spaces such as
rooms and concert Halls, while others emulated mechanical plates, and some provided delay/echo and chorus effects.
However, the really popular algorithms that people still reminisce about today were the unnatural ones called Ambience, Non
lin and Reverse. These terms are quite familiar today, but were highly innovative in their day.
Internally, the original RMX16 hardware was crammed full, with 14 large circuit boards mounted vertically in sockets on a
motherboard which covered the entire chassis floor area. Not surprisingly, the unit required forcedair cooling from a rear
panel fan. Technologywise, its monoin/stereoout pseudo16bit audio converters delivered an 18kHz audio bandwidth and
90dB dynamic range again, stateoftheart in its day, and still very usable today but the reverb algorithm number
crunching worked with 28bit words in the puny (by modern standards) 22MHz microprocessor.
Plug & Play

A preloved RMX16 will typically set you back between 1250 and 2500 today, depending on condition and accessories,
such is the continuing demand for this wellengineered classic. However, Universal Audio have now released a plugin
version that makes its core algorithms available on the UAD platform; and although the plugins $349 price tag is still
substantial, its much less than youd pay for the hardware! Apparently, all nine reverb algorithms are byteforbyte identical to
the original 1980s software, having been lovingly ported across by Mark Crabtree and a team of AMSNeve and UAD
engineers. Great care has also been taken in modelling the complex sonic characteristics of the original hardwares analogue
and converter circuitry, too, which played a big part in the overall sound.
At a reported 41.1 percent DSP load for a single stereo instance, the UAD RMX16 plugin is on a par with complex
emulations like the EL7 Fatso and Chandler Gav19T. Indeed, the RMX16 is one of the 10 most powerhungry plugins
currently on the UAD2 platform, requiring more DSP effort than the Ocean Way Studio plugin (38 percent) and way more
than the Lexicon 224 (17 percent), EMT 250 (7.8 percent), and EMT 140 plate (15 percent). The inevitable consequence is
that a UAD2 Quad card maxes out with eight RMX16s (at 44.1kHz), and proportionately half and twice the number on Duo
and Octo cards, respectively.
You might well ask why it takes so much DSP power to emulate a relatively primitive digital device. Although the RMX16
reverb DSP code was sophisticated for the time, its not actually as complex as the code used in the Lexicon 224XL, where a
chorusing process on the reverb tails is intrinsic to the units lush character. However, the reverb algorithm processing was
dependent on the hardware sample rate (which was, if memory serves, 40kHz), and so the UAD implementation may well
involve DSPhungry samplerate conversions to accurately replicate the original algorithms parameters. The RMX16s
converters were actually gainranging 12bit devices, too, and it seems reasonable to presume that the complexity of this
analogue circuitry modelling accounts for much of the plugins surprisingly high DSP demand.
Fortunately, I cant recall any 1980s commercial recording studio ever using more than two RMX16s at a time!
Reverberation is normally employed in an effects loop, receiving auxbussed inputs from many channels and outputting the
processed signal to an effects return channel and this UAD plugin is configured to work in the same way by defaulting to a GLOSSARY: technical terms
fully wet output. However, UA have added a wet/dry balance control so that the plugin can be used as a channel insert; two explained
pushbuttons used for saving and recalling user presets on the hardware unit have been reallocated as Wet Solo and
Wet/Dry Mix controls on the UAD plugin, and preset save/recall duties are taken care of elsewhere in the GUI.
Competitions!
RMX Bandits
In use, the UAD RMX16 is identical to its hardware antecedent. Input and output levels are set using the rotary controls on the
lefthand side, with a small bargraph meter, and all parameter adjustments are made by selecting the required function with
the virtual pushbuttons and then altering the numerical value in one of four ways. Three of these are identical to the
hardware unit: a specific parameter value can be entered using the virtual keypad (followed by the # key to confirm the
change); the Nudge up/down buttons can be used for small changes; or the rotary pot on the lefthand side can be employed.
The new fourth option exclusive to the UAD plugin is to click in any data display window and type in a value from the
computers keyboard, followed by Enter.
The nine core algorithms are recalled by selecting the Program button, followed by a keypad number between 19 and the
# key to confirm. Entering a higher-numbered memory location results in E for error being displayed (the original hardware
would usually crash!) The algorithm name associated with the selected program is also displayed above the nudge buttons.
Not all of the configuration parameters are available for all nine programs, but the reverbs all offer predelay time, decay time,
and high and low EQ. Some of the keys on the keypad serve no function, such as the asterisk and letters CD, but the letters
AB are used to select the left or right channel when programming independent echo delays and feedback.
Its been a while since I used an RMX16, but it was very much the goto reverb unit during my formative years (with the
later, and surprisingly similar, Klark Teknik DN780 as the first reserve). However, the unique character and quality of the
RMX16 reverbs are still much in mind, not least because they are so obvious on virtually every pop album dating from the
mid80s.
The Ambience program was and probably still is the most popular, with its fundamentally neutral but expansivesounding
reverb which blends very naturally with any existing room reverb on the track. It adds a beautiful space to any source and
sounds warm and flattering, but without imposing any significant character of its own. Basically, its a very musical reverb that
just works brilliantly in almost all situations and across a wide range of decaytime settings. As the units first program its no
surprise it became so popular, and often theres simply no need to look any further!
Another very useful feature of the Ambience program is that if the decay
time is set to the maximum 9.9 value, the algorithm has almost infinite sustain,
creating an ethereal choir effect. By dialling down the highfrequency filter, the
sound mellows over time in a very natural way, too. Some RMX16s had a
variation of this program called Freeze (see box), which made it much easier
to switch this mode on and off with or without new sound input.
Program 8, NonLin2, was another popular algorithm for small room effects,
or for adding energy and scale to percussive sounds giving a kind of Phil
Collins quality. In essence, the early reflections dont decay, which has the
effect of extending the duration of short sounds, ending with a kind of gated
quality as the short reverb tail dies quickly away. This was almost an industry
standard effect for 80s drums and percussion! The Reverse1 algorithm
(program 9) was another drumcentric effect, with the reverb character
building and then stopping abruptly. Not surprisingly, the Room, Hall and Plate
programs all do exactly what they say as well, and still sound extremely
believable today, such was the quality and musicality of those original
algorithms.
Character Actor
One of the innovations pioneered in the
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Does the UAD plugin sound identical to the hardware? Its hard to state
RMX16 was the ability to scan in additional
programs using a handheld barcode reader.
categorically, given that two vintage RMX16s would be unlikely to sound
identical to each other! But thats not really the point: what matters is whether
the plugin delivers the same character of effect that made the original hardware so useful and popular, and I have to say an
emphatic yes to that one. The UAD RMX16 programs definitely have exactly the same sound characteristics, and the plugin
delivers the same kind of sound quality, which means you would choose and use it for the same purposes and achieve the
same overall results. Its about character and usability, at the end of the day, and the RMX16 plugin has that in spades. The
user interface is very simple and fast to adjust, just as the hardware original was, largely thanks to the very small number of
parameters on offer. However, if manual tweaking is not your thing, there are as with most UAD plugins more celebrity
presets available than any sane user could possibly use in a lifetime of musicmaking.
I love the UAD RMX16, both for its familiar controllability, and for its superb sound character and versatility. It delivers the
same sonic environments that I remember from the hardware, and although it might be a bit of a DSPhog compared to other
plugin reverbs, I think its definitely worth it. In fact Id suggest that one RMX16 in an effects loop is worth a dozen lesser

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reverbs inserted in individual channels. Highly recommended.
Alternatives
The EMT 250 and Lexicon 224 are contemporary alternatives to the RMX16, and emulations of both are available on the
UAD2 platform, although neither has the same sound character, of course. The Eventide SP2016 and Ursa Major SST
282 Space Station are also available as plugins. There are also unofficial emulations of the Lexicon 224 and 480L from
Native Instruments and Relab.
Standard Programs
The nine standard programs provided in the UAD plugin are exactly the same as those of the RMX16 hardware unit:
Ambience: a very neutral reverb effect somewhere between a very clean, uncoloured plate and a small hall. Can be used
to extend existing reverb very naturally. Setting the decay time to the maximum value (9.9) provides infinite decay, and
new sound can be added to create a sustained choir effect, or a freeze effect with no new input.
Room A1: a typical small live-room sound, with coloration to simulate standing waves.
Hall C1: a classic hall sound with strong early reflections to give a defined size.
Plate A1: generic highdiffusion plate effect.
Hall B3: similar to hall C1 but with less pronounced early reflections.
Chorus 1: a fivevoice effect, with each voice in a different spatial position and with random delays controlled by the pre
delay setting. The effect depth is determined with the Decay control.
Echo: dualchannel echo effect. Predelay sets the echo period, and Decay the amount of regeneration. Each channel is
selected for control independently via the A and B buttons on the keypad. The two channels have different maximum
delay settings.
NonLin 2: this provides a nondecaying reverb pattern for a short period (set by the Decay control), followed by a rapid
decay. The effect helps to increase energy and loudness, particularly for percussive sources, and can also be used as a
small room effect.
Reverse 1: a dualmono effect with diffuse reflections on the left channel, and more discrete reflections on the right. The
reverb effect builds gradually before stopping suddenly.
Additional Hardware Programs

The following additional programs were also available for the hardware RMX16, but had to be loaded into batterybacked
memory via barcode sheets. They are all variations of the nine standard programs, but offered some useful benefits or
alternate characters. Sadly, none of these are currently available for the UA plugin, but they would be trivially simple to
add at some future point if AMSNeve and UAD decide to offer an upgrade option!
Delay: a single mono delay of up to 1.63 seconds (depending on hardware version) available on output 2 only, with
output 1 producing a direct digital source feed. Delay time is set by the Predelay control, and feedback using the Decay
control.
Freeze: a variation of the maxdelay Ambience mode, making it easier to control. The Decay control has three settings:
0.0 is clear effect, 0.1 freeze but accept new audio, 0.2 freeze with no new input. These modes are easy to control via
the nudge buttons.
Hall A1: an early version of the Hall C1/B3 programs, with a slightly coarser decay.
Image P1: similar to NonLin, but with reflections being panned between outputs. Sounds very similar to NonLin when
summed to mono.
Plate B1: a variant of Plate A1.
Reverse 2: a stereooutput variant of Reverse 1.
Room A0: the original version of Room A1, shipped with early units. It has a coarser and brighter top end than Room A1.
Room B1: uses a different algorithm to Room A1/A0 and features a constant diffusion density with time. The Decay
control affects impression of room size.
NonLin 1: a simpler, dualmono version of NonLin 2, with the two mono outputs differing by providing only discrete or
diffuse reflections.
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In this article:
Installation
Navigation
Bend Me Shape Me
Effects Page
Preset Navigation
Beats Page
Conclusion
Alternatives
Say Cheese...
Fairfax Vol 1 ADpak
What Does It Cost?
XLN
Addictive Drums 2 130
pros
Excellent range of superbly
recorded, stylistically
themed ADpaks available.
Twice the number of insert
effects, including an
excellent Transient Shaper
and one of the best
compressors (for drums) I
know of.
Eighteen kitpiece slots, with
kitpiece linking.
Beats page has many ways
to creatively customise MIDI
grooves.
cons
It didnt come out sooner.
summary
AD2 is a worthy upgrade
that addresses the
limitations of the original
version, whilst adding some
valuable new features. The
insert effect chain has been
greatly enhanced, theres
better EQ, greater control
over velocity response, and
numerous ways to indulge in
serious drum sound design.
For those that use MIDI
grooves, the Beats page is
now capable of genuine
creativity. Oh, and it looks
lovely, too.
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XLN Addictive Drums 2

Virtual Drum Instrument
Reviews : Software: Instruments
Its been a long time coming, but the follow up to XLNs

Addictive Drums was worth the wait.
Buy PDF
Nick Magnus
here are plenty of virtual drum instruments to choose from, offering highly detailed, top-of-the-range drum kits lovingly
recorded in expensive studios. Some people use them solely as a demoing or songwriting tool, others see them as a
valid alternative to real drums and put a lot of time and effort into making them an important and featured part of their
music. Whatever your persuasion, its factors such as cost, sound quality, the level of detailed realism, expandability and ease
of use that affect the decision to gravitate towards one virtual drum instrument rather than another.
Since 2006, Addictive Drums (herein referred to as AD) has proven very popular, delivering on all these fronts. Whilst
perhaps not as minutely adjustable or customisable as, say, FXpansions BFD or Toontracks Superior Drummer, its XLN
Audios everything you need and nothing you dont philosophy that has made it so easy to get great results with the minimum
of fuss. Their constantly expanding library embraces many kits from the 1960s to the present day, all eminently tweakable
beyond recognition if desired using well-designed effects and a very attractive and usable interface. Mindful of the constant
demand for newer and better features, Addictive Drums 2 (AD2) addresses some of the limitations of the original version, and
benefits from improvements and additions across the board. You can check out the review of the original version here:
www.soundonsound.com/sos/jul07/articles/addictivedrums.htm.
Installation
Installation of all XLN products now requires you to use their proprietary online installer, first introduced with Addictive Keys.
Once the online installer application is running and youve logged into your account, your existing AD libraries (if any) are
scanned and checked for available updates. These and any AD2 products youve just purchased will show up as update
available or install as appropriate. You can choose to download and install items one at a time, or everything in one go its
up to you and how much you enjoy watching progress bars!
Navigation
AD2s GUI adopts the same approach as Addictive Keys, using picture-based galleries to navigate through the content. Each
kit has two Explorer menus; the Kit menu offers a selection of bread and butter presets generally biased towards more
natural drum sounds, whilst those in the Selections menu tend to be more processed, often demonstrating how radically the
kits can be transformed away from the raw material. Depending on what you have installed, the presets audition buttons will
show either an arrow (the kit is installed and a short MIDI loop plays) or a headphone icon (the kit is not installed and the
same loop plays as an audio demo). The latter gives a taste of what you could have its a useful demo tool, as well as a
shrewd carrot on a stick sales technique!
information
See the What Does It
Cost? box.
info@xlnaudio.com
www.xlnaudio.com
The Kit page gives an overview of all loaded kitpieces. Visually

its less cluttered than before, and rather easier on the eye.
There are now 18 kitpiece slots as opposed to ADs 12, a
welcome expansion providing six cymbal slots and two ride slots,
a vast improvement on the three cymbals and one ride of the
earlier version, which was always a frustration. Additionally, three
new Flexi slots accept any kind of kitpiece, so you can add extra
toms, cymbals or anything else as the situation demands. These
Flexi slots replace the Xtra slot in AD, which was rather limited in
its usefulness. The kick drum and snare can each be linked to
any number of other kitpieces, of any kind. Simply click on either
drums link icon and drag it to the target kitpiece. Making up a
composite dual snare sound is as easy as loading your second
The Studio Prog kit and its associated presets as displayed in
the Explorers Kit view. The arrow buttons play short
snare to a Flexi slot and linking the two together. Flexis also host
examples for auditioning purposes. Quick basic level
a number of additional useful trigger sounds (sine waves, white
adjustments can be made at the top of the page.
noise and uncredited drum machine kick and snare samples)
designed to be used as layering textures, or indeed used in their own right. The larger complement of kitpiece slots has also
caused the mixer to grow in size; there are 10 channels compared to the eight of AD.
Turning to the Edit page, every mixer channel now has two
MultiFX inserts; one pre-EQ and one post-EQ. Joining the
original Compressor/Distortion effects module are three new
effects, beginning with a Noise module for those who dont like
things to be too clean. Eight types of noise including tube hiss,
7ips and 15ips tape hiss, and even Fairfax Studios air
conditioner noise, can be mixed in and shaped with a variable
decay envelope. Tape and Shape are, respectively, a tape
simulator and a transient shaper, the latter being a particularly
effective tool for controlling the snappiness and sustain
characteristics of individual AD2 channels and/or its master
output. These share the same effects module as the Saturation
Limiter (also found in AD) and each effect within the module can
The distinctly neater kit overview page, with its expanded
complement of 18 kitpiece slots. Here, the snare has been
be enabled individually. Both pre and post effects modules can
linked to a second snare occupying the Flexi 3 slot.
explained
be toggled between either Tape/Shape/Saturation or
Compressor/Distortion options, so the order in which processes
occur is up to you. The only difference in structure is found in the Bus and Master channels, where the Noise module is placed
Competitions!
after the post-EQ effect module, as opposed to being first in the chain on all other channels. The EQ module has also been
upgraded to four bands; the two middle bands are peak boost/cut with variable Q, whilst the lowest and highest bands are
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shelving types.
Bend Me Shape Me
New features are found in the Edit pages upper sampler section too. Velocity response of each kitpiece can be adjusted to
access a narrower range of velocity layers, whilst still giving the full scope of velocity to volume response. The adjoining Filter
knob, when set above zero, controls the amount of low-pass filter applied to lower velocities, effectively darkening quieter
hits.
Adjacent to the Pitch control, and available only to the kick and
snare drums, is the Tone Designer (toggled as an alternative to
the Pitch Envelope module). This clever gizmo appears to act as
a frequency-specific envelope-shaper that controls the amplitude
decay curve of a predetermined frequency range. Every kick and
snare kitpiece has its own signature frequencies; these have
been predetermined by XLN, and if you load drums in
succession you can see the frequency graphic is different for
each one. For example, if the decay length of a particular snare
is right, but you feel its metallic ring goes on too long, the Tone
Designer can shorten the envelope of those ringing frequencies
whilst leaving the overall envelope and volume of the snare
untouched. Conversely, if you want more ring you can push Tone
Designers envelope to the ceiling and really bring that character
to the fore. Its often more effective than EQ or a general
envelope adjustment, both of which might compromise the rest
of the drum sound.
The snares Edit page, showing pre-EQ

Compressor/Distortion and post-EQ Tape and Shape effects.
Note the Tone Designer on the snare is reducing the
amplitude of a set of predetermined frequencies over time, in
this case 181ms.
Effects Page
Based on those featured in Addictive Keys, FX1 and FX2 are two identical send effects modules, referred to as Delerbs;
essentially, delay and reverb combined into one effect. A blend slider mixes from delay at one end to reverb at the other, and
anything in between. The delay can be tempo-synced to one of 11 note values, or set in milliseconds, with control over
feedback, swing and ping-pong width. The reverb has ambience, room, hall and plate algorithms, with pre-delay, decay time,
damping, and a swirl parameter basically a chorus for thickening the reverb. Both effects modules are followed by twoband parametric EQ and a choice of pre or post Master fader routings.
Preset Navigation
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AD2 again borrows its Preset Browser from Addictive Keys, with useful tools for filtering presets by category; you can choose
to browse by product, by author (ie. you or XLN), by type, or any combination of these. Within your search terms, an optional
Sound Ideal filter activates a slidable bar that further categorises your search results into Natural at one end of the scale to
Extreme at the other. If you own the full range of AD2 products, the number of Presets is huge, so these filters can speed up
your search for the perfect kit considerably. As with Addictive Keys, your own Presets are continually updated to XLNs Cloud
storage (as long as youre online, you dont need to be logged in). Your Presets are therefore safe even in the event of the
dreaded disk crash; when AD2 is re-installed, it automatically retrieves and reinstates all your Presets. Owners of AD who
wish to port their AD Presets over to AD2 will be pleased to know that AD presets can be dragged from their folder and
dropped straight onto AD2, and they load perfectly.
Beats Page
AD2 comes pre-installed with a selection of MIDI drum grooves drawn from XLNs range of MIDIpaks. Searching for a suitable
beat could be time-consuming, so various search tools help to narrow down the choices. You can search by Library, Category,
Tempo or Time signature; you can also type in a keyword.
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The Grid Search provides a fun and useful way to customise the beats brought
up by a regular search. It works by presenting a three-lane step sequencer, each
lane representing the hi-hat, snare and kick of the currently selected beat. Ticked
orange circles show where each instruments beats fall; click on any step to add
or remove hits, and the search list updates to show any grooves that share this
change. If the hits youve added are unticked, it means there are no grooves in
the library that share this characteristic, in which case clicking Replace for that
lane plays the line youve created. You can then drag and drop this modified MIDI
groove to your DAW.
The Shortlist tab over to the right provides a very handy way of assembling in
one place all the beats youve found and/or created. These methods alone are a
good way to customise the grooves on offer, but the Transform tab takes things
much further. Here, you can alter the dynamic range not only of the entire
groove, but of the individual instruments within. Using the Accent controls, its
possible to increase or reduce dynamics on an eighth- or 16th-note basis, and to
add dynamics to otherwise undynamic hits. You can also invert the dynamics to
completely alter the feel of the groove. Ride, toms, hi-hat and snare can be
reassigned to alternative articulations as well, and timing and velocities can be
randomised. Grooves can be run at half or double time, 75 percent or 133
percent of normal speed, 4/4 can be changed to 6/8 and vice versa; theres an
incredible number of ways to make new grooves from the presets. Two more
features cap off the Beats page: firstly, your own MIDI drum performances can be
recorded straight into AD2 and saved as MIDI files in its browser. Lastly, anything
you play in AD2, whether its a real-time performance, a MIDI drum part playing
in your DAW, a single hit or a groove pattern, can be dragged straight into your
DAW as an audio file of up to 15 seconds in length.
On the Beats page, a MIDI groove is being

transformed in various ways. The overall
dynamic range has been restricted, whilst
the dynamics of beats falling on eighths
have been inverted, changing the emphasis
considerably. The ride cymbal has been
made to play a bell articulation instead of
the tip, and the hi-hat is forced to play the
shaft articulation instead of the tip.
Conclusion
Theres much in AD2 to make the upgrade decision a no-brainer: the additional kitpiece slots, kitpiece linking, the new insert
effects (especially the Transient Shaper) and compatibility with older AD Presets did it for me, and many are sure to find the
massively improved Beats page inspirational as well. XLNs pricing structure is sensible, allowing for newcomers to add kits as
and when they can afford to, rather than paying for a huge core library they might never make use of. And the most important
bit? It sounds bloody marvellous. .
Alternatives
Toontrack Superior Drummer and EZdrummer 2, FXpansion BFD3, Drumasonic, Native Instruments Drummer Series and
Steven Slate Drums all provide detailed, natural-sounding virtual drum kits recorded to a high standard, each with their
own signature sound and features. AD2 is not unique in these respects, so its a case of try before you buy, compare
sounds, features and ease of use, and let your instincts decide for you.
Say Cheese...
A handy snapshot feature allows up to four snapshots of the current state of AD2; subsequent snapshots can be taken,
but these push the earliest one off the list, so its always the most recent four that you can toggle between. This obviates
having to save edits in progress before making further major adjustments you can always go back to a previous
snapshot if those recent changes dont work out.
Fairfax Vol 1 ADpak

To coincide with the release of AD2, XLN introduced a new ADpak, Fairfax Vol 1. This Gretsch Stop Sign kit was recorded
in the legendary Fairfax Studio A (formerly known as Sound City) in Los Angeles, famous as the recording venue of many
seminal albums. Fairfax Vol 1 does indeed sound very American, with a sound XLN describe as beefy, punchy and
muscular. Id go along with that description, and add that the kit is eminently malleable to suit most rock genres. As such it
would make a good choice as a starter kit for newcomers to Addictive Drums. An American Rock MIDIpak is also available
to complement the Fairfax kit.
What Does It Cost?

At the time of writing, existing owners of AD can upgrade to AD2 for 79.95; the cost includes free upgrades of all your
existing content to AD2 format. Installing AD2 doesnt in any way affect your original AD installation you can still run
projects that used AD as before. Bear in mind though that your upgraded kits will need to be downloaded again in the new
format (as Ive mentioned in the Installation section). New owners are offered an attractive deal: 129.95 gets you AD2
plus any ADpak of your choice thats effectively an ADpak for just 50.00 instead of the usual price of 89.95. Artist,
Producer, XXL Studio and Solo bundles are also available with associated cash savings. The original AD ADpaks that
comprised multiple kits are now broken up into individual products, giving a total of 16 ADpaks, eight individual kitpieces
and 27 MIDIpaks.
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ZeroG Electro Cinematic

Multi Format
John Walden

ZeroGs Electro Cinematic library provides approximately 2.5GB of sample material presented in
Acidized WAV, Apple Loops, EXS24, Kontakt and Reason NNXT formats. I explored the library via Kontakt, but all the formats
have access to essentially the same content. As the title suggests, this is a collection of electronicbased sounds and, given
the cinematic bit, there is more than a passing nod to the needs to media composers or sound designers.
In fact, Electro Cinematic is very much a library of two parts. On one hand, there is a substantial selection of drum,
percussion, bass, synth and arpeggio sounds in a loops folder, and these could easily be combined to create a purely
musical composition, albeit with a fairly dark, electronica, vaguely Nine Inch Nails sort of a vibe. On the other hand, there are
four folders containing collections of soundscapes, textures, pads and sound effects. These are much more in the sounddesign territory and, if you needed to add atmospheric soundscapes or ambient noise to your next scifi/industrial apocalypse
score, this would be a good source of material.
That said, because the atmosphere and sonics of both elements blend well, the two
sections of the library can easily be brought together so, if you wanted to add a few
vocals and create some electronica/industrial-inspired songs to stand alone or to sit
within a suitable film or computer game score, Electro Cinematic will have plenty to
inspire.
The sound-design section of the library is pretty much what you might expect and,
while evolving metallic pads and pulsing electronic noise have been done many times
before in sample format, there is plenty of suitably inspiring content here. If you want to
drive your audience off the edge of their seat and behind the sofa via sound, combining a
few of the textures and sound effects supplied here will provide plenty of ways to do it.
However, the highlights for me were within the loops section. There are some very
cool drum/percussion loops amongst this lot and, if you are looking for an instant dose of
rhythmic and dark, then blending combinations of these loops together will certainly do
the trick. I was perhaps less keen on the chord-based loops, but the bass, lead,
sequences and arpeggio loops were all very good indeed.
Electro Cinematic is not going to be for everyone (singer-songwriter types need not
apply) but, whether you are a media composer with a scifi brief or a music producer with
a liking for fairly dark electro/industrial tunes, this is well worth a look. John Walden
Audio Example
$82.99
www.zerog.co.uk
.
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In this article:
Acid Trip
The Big Time
Half Dub
Clubbers Delight
Strong Words
Too Good
Starting Points
Im With The Multi
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David Wrench
Mixing Caribou, FKA Twigs & Jungle
Buy PDF
People + Opinion : Artists / Engineers / Producers / Programmers
Mix engineer David Wrench has the rare ability to bring

discipline to experimental projects, without losing their
leftfield qualities.
Tom Doyle
avid Wrench has been responsible for mixing three of the most acclaimed albums of recent times, namely Caribous
Our Love, FKA Twigss LP1 and Jungles selftitled debut, and currently finds himself much in demand as the goto
guy when it comes to polishing and finishing a project. To reach this point, Wrench took a slightly circuitous route,
beginning with his secondary school days in Holyhead, Wales, when he was taken under the wing of his physicsteacher
turnedrecordproducer Gorwel Owen (Super Furry Animals, Gorkys Zygotic Mynci).
Encouraged by Owen, Wrench ensconced himself in the music room at school, his career path effectively set. They had an
Atari with Notator, he remembers, and a fourtrack and pair of JBL speakers and a rackmount Roland D10. I was just in
there all the time. I was doing music A-level anyway, and no one else seemed to have any interest in using this facility. So
every dinner time, every spare lesson, I was just in there making music.
Acid Trip
It was the late 80s and the dawning of acid house and, like many of his peers, Wrench was wholly inspired by dance music,
and in 1990 he released the first Welshlanguage acid house 12inch, Lledrith Lliw, as Nid Madagascar. In subsequent
years, he put out a sporadic sequence of albums under his own name, ranging in style from the doomfolk of 1997s Blow
Wings Blow to the synthpop of 2005s The Atomic World Of Tomorrow.
Throughout this time, he was also working as an engineer at
Bryn Derwen Studio in Bethesda, North Wales, where he cut his
teeth on a number of very different projects, including albums by
the late Scottish folk singer Jackie Leven and former Teardrop
Explodes singer Julian Cope. If the former taught Wrench about
good musicianship and preparation, then the latter showed the
fledgling engineer that equally impressive results could be
achieved through utter chaos, initially when he was involved in
the sessions for Copes unhinged garagemetal offshoot Brain
Donor.
That was hilarious, but quite intense, Wrench recalls. Hes
so spontaneous and creative. Id never worked with anyone like
him before, so I was thrown initially. I wasnt used to someone
wanting a [Shure] 58 in the control room standing on the control
surface doing their vocals. Id not dealt with the monitors really
loud and having to control feedback while someones recording a
vocal live.
David Wrench in his favoured Strongroom Studio 1.

Photo: Richard Ecclestone
I learned a lot about mixing then. About two or three in the morning at the end of every session wed be putting down these
rough mixes that were really quite deranged. Half the way through it, hed be like, Aw turn the guitar up 6dB, pan it left. Yeah,
thats all right, sounds good. Something else would come in as I unmuted it, remembering it was there. It came to the end of
the week and I asked him where he was getting it mixed. And he goes, Nah, what weve been putting down, theyre the
mixes. And thats cool learning to keep accidents when theyre good and just going with it and not being too uptight about
stuff.
The Big Time

Before long, Bryn Derwen Studio was being hired for many other projects, partly due to the facility itself and partly because of
Wrenchs skills. When producer David Kosten began using the studio for records by Guillemots, Everything Everything and
Bat For Lashes, Wrench was introduced to the joys of majorlabel production budgets and the freedoms they can afford.
Early on I was on a lot of records where it just had to be done in two weeks, he says. So it was good to work on records
where there was time and budget there to explore. Im lucky enough now that thats the case with a lot of records I work on.
Theres enough budget and time to get it right. I always allow time for that, for just exploring.
Still, it was through his indie label connections that Wrench
was first brought to the attention of Canadian psychedelic
electronica artist Dan Snaith, aka Caribou, as someone who
could produce more focused mixes of his laptop productions.
The two first worked together at Musikbox Studios in Kentish
Town, London, on tracks for Caribous second album, 2007s
Andorra, with Wrench helping to make sense of the often
kaleidoscopic and layered arrangements.
Dan had always just mixed himself, and felt that he wanted to
take it up a notch on some of the tracks, says Wrench. Two or
three of the tracks on that record were really complicated.
Massive things with just hundreds of tracks. Theres a song
called Melody Day that was a big mix. We were using bits of EQ
on the studios Neve and bits of outboard compression and EQ,
but then automation within the computer. Thats how I worked for
a long time really, partly because the computers I was working
on werent powerful enough to do what we wanted.
Dan Snaith, aka Caribou, has been a long-standing client of

David Wrenchs.
Photo: Thomas Neukum
He totally smashed the mix on Melody Day, Snaith enthuses. Made it so much more spacious and massivesounding,
and got the vocal sitting nicely in the track. He was probably frustrated, because as I layered things on top of one another
recording tracks, I would never bother to go back and clean things up because theyd be hidden by another layer of sound on
top. I remember asking him if there was anything I should do differently next time. He said that I should edit the sound of the
bus going past the window of my flat out of the vocal track before applying a hall reverb to it! David can mix and engineer
technically right alongside the best of them, but also understands the value of character and eccentricity in music. Its that
combination that makes him so amazing.
Half Dub
The working relationship between David Wrench and Dan Snaith progressed with the mixing of the next Caribou album, Swim,
in 2010, which was done at the formers suggestion back in Wales at Bryn Derwen. It was great to go up to a studio that he
knew inside out, says Snaith, and where he could plug in all sorts of different hardware. I didnt realise how much of the
mixing process was surgical. He would go in and EQ tiny problem areas where different instruments conflicted with each
other.
I generally do lots of little cuts in an EQ, says Wrench. Its mainly reductive. I only add stuff if I feel it really needs it. If I
can do it through reducing EQ, it feels better to me for some reason. If something wants to be brighter, I often think, Well, is it
just that its too muddy? And Ill take a bit of the mud out and itll sound a bit more natural than just brightening some
frequencies. Otherwise its a bit harsh on the ears. And I find I can make stuff sound bigger through just doing lots of EQ cuts.
The psychedelic nature of Caribous music, particularly on Swim, meant the pair found plenty of room for experimentation
while mixing the album. The way we mixed that was quite strange in places, Wrench states. Especially on the track
Odessa. We did a mix, settled on it, stemmed it, but then also did dub mixes. Then wed find the occasional bars of those dub
mixes that were great and wed just chop them into the main mix.
When he was mixing Odessa, says Snaith, he had out this amazing old Russian BBD delay with all the buttons labelled
in Cyrillic letters, and was jamming out the delay on the drum tracks. Then he used little parts of that in the track to add
excitement. It was so amazing watching him getting super into dubbing out the track in a King Tubby style, nodding his head
and both of us jumping around the room with the music cranked up.
Clubbers Delight
For Caribous latest album, 2014s Our Love, a more polished take on multilayered dance music, Wrench and Snaith
borrowed the studio owned by Radiohead producer Nigel Godrich for the mixing sessions, running the tracks through his Bill
Putnam hand-built desk bought from Ocean Way Recording in Los Angeles.
It was interesting, because we did split it out through the
desk, says Wrench, but we then did a blind test with the inthe
box Pro Tools mixes. We could both tell which one was which,
but the difference was so miniscule. Actually what it goes to
show is how incredibly clean that desk is.
To test the lowend balance on Our Love, Wrench and Snaith
took the mixes into empty club spaces in London, namely Plastic
People and Fabric, to check how the tracks were sounding. That
was a real eyeopener, says Snaith. The mix environment gave
us an accurate picture of what was going on, but getting to hear
the tracks in a club was really informative.
We knew it had to sound good in a club, Wrench stresses.
That was absolutely essential with it. So there were issues to
Caribous most recent album was mixed at producer Nigel
work out that only became apparent in a big club system. Things
Godrichs studio, using this console hand built by Bill Putnam.
like the tuning of the bass drum. So wed run off a couple of
It was, says David Wrench, surprisingly clean.
mixes, one with it tuned to 33Hz, one with it tuned to 28Hz, and
work out where it felt good. It varied track to track, just how it sat with the bass, where it felt like it was giving the most punch.
And then also just things like levels of snares and hihats when they were coming in.
Being able to listen in lots of different places and lots of different ways is so crucial to me, Snaith adds. But everywhere
we went, Davids mixes sounded great.
Strong Words
Wrenchs favourite place to mix is Strongroom in Shoreditch, London, because he finds their rooms true and particularly likes
the Genelec 1031s in Strongroom 1. Interestingly, though, because he doesnt really touch its Neve VR60 or array of
outboard, hes basically just using the facility as a room and a pair of monitors.
I went and listened to a load of rooms in London and I just liked the sound in here, he says. I found with a sub in here, I
knew I could take it anywhere. I mean, its a matter of taste, monitoring. But I found with a lot of rooms, I was a bit unsure what
was happening in the really low mids and down into the bass end, and thats so crucial for the work Ive been doing in the last
year. With the Twigs stuff it was absolutely essential to have the sub cause its so subby. I just found that it sounded good
here, and I just liked the vibe.
David Wrench first began mixing what would become FKA Twigss LP1 in Spring 2014. Twigs, or Tahliah Barnett, former
dancer for the likes of Plan B and Kylie Minogue, had been signed to Young Turks Records as an artist/coproducer and
needed someone to pull together the various recordings that shed done with a number of producers, including Paul Epworth,
Emile Haynie (Lana Del Rey, Eminem) and Arca (Kanye West, Bjork).
I loved her stuff, he says. I thought, This music is amazing, its not like anything else Ive ever heard. I was desperate to
be involved with it. It just sounded like the future to me. They played me eight or nine tracks and they all sounded quite
different. Cause theyd all been through different producers, they were done at different studios. They gave me this track One
Time and it was in a bit of a mess and they were just like, Well, see what you can get out of it.
Perhaps best labelled avant garde electronica, topped by Twigss sultry, expressive vocals, LP1 features an unorthodox
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was stuff in all the tracks that unified them, he says. Because
she does all the drum programming and you can tell its her. No
one else programs drums like she does it sounds almost punk
to me and obviously theres her vocals and the way her
backing vocals all fit in.
Obviously Twigs had coproduced everything on it. She
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shell be like, That thing there, Ive got a movement that goes
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says. Mixes that sound unbalanced. I just like records that excite me, and if
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directions in the studio. Twigs will be very specific, but also shell talk in terms
of, Turn the wizard synth up. Youre thinking, What the hell is the wizard
synth? And shell be like, Yknow, the one thats swish swish swish like a
wizards cape? And youre, Oh OK, I know what you mean. Cause theyre all
just labelled synth on there, yknow. Hundreds of tracks. And she knows every
one and how she wants it to be. Shes got a very clear vision and that makes it
easy.
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Around the same time, Wrench was also mixing the eponymous album by West Londons modern funk/soul troupe Jungle.
This project offered different challenges, since the band wanted a slightly murky sound, inspired by leftfield hiphop producers
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We had various versions of songs that we kind of made cuts

and edits of, so they start to sound like sampled loops, explains
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distortion is preferable, yknow. Wed often use the Roland
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Wrench was initially given Jungles third single Busy Earnin to

mix, and feels his first attempt cleaned it up a little too much.
Their rough mixes sounded good, he says. But there was just
an extra 15, 20 percent to get out of it for me. Then I popped it
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youre right, it can go back towards where it was.
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bringing a whole load of extra lowend rumble into a track thats clouding up the mix.
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swimming in modulation or Leslie speaker effects. They really know how to balance their vocals, says Wrench. But Id be
like, Give me a clean vocal as well as your effected vocal. Ninety percent of the time wed use their effected vocals, but then
sometimes mix a bit of the clean one in for clarity if the words were getting a bit lost.
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Like many mixers, these days David Wrench finds himself working almost entirely in the box. Theres a real reason for it
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HD through Adam A7X monitors with a Focal CMS sub, alongside a secondary pair of Yamaha MSP5s. I really love the
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Once hes cleaned up a track in terms of levels, Wrench

works very rapidly through a first draft of a mix. The first
phase is really quite fast and instinctive, he says. Ill run
through a mix drums, bass and so on and just quickly
put EQs on, quickly get a level. I tend to do it all dry at that
point. I wont get into reverbs unless its absolutely intrinsic
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whole thing feeling like I can get a vibe off it, and then I go
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David Wrench makes more use than most mix engineers of
multiband compression. This Waves Linear Phase
Multiband setting was used across the mix bus for FKA
Twigss Two Weeks, preceded by the same companys SSL
EQ and compressor plugins.
Wrench is also a big fan of Valhalla reverbs.
Im With The Multiband

David Wrench tends to work with a small, if select, bunch of plugin processors. Ive got the Waves bundles, he says. I
really like multiband compression and I find it controls things well. Say on a vocal where occasionally it gets a bit peaky,
instead of taking those frequencies out for the whole thing, I just take them out for the points where theyre a problem. You
can lose warmth in a vocal by scooping out the low mid sometimes. In general, if you spend a while setting a multiband
on it, you can get a vocal to sit well.
I use multiband across the whole mix as well, just subtly. Ill have an EQ and I use the SSL compressor, only on a 2:1
with a medium attack but a really fast release and then into a multiband. But I tend to automate the type of compression
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or the drive into the compression between sections of the song when it comes to the chorus, Ill give it a couple of dBs
boost. The multiband will hold it in place, but itll just feel like its driven up. Thats my big secret actually [laughs]. Or
sometimes I will just ride the master fader even after the compressor. All you want to do is lift the chorus by 1dB.
Reverb and delaywise, Wrench tends to favour Valhalla and SoundToys plugins. On the Jungle, Twigs and Caribou
albums, those Valhallas are the reverbs. They sit really well in a mix. Theres really good control on them and they have a
nice stereo field. The Valhalla VintageVerb I use quite a lot. And for things like guitars and shorter reverbs, the
ValhallaRoom. Theyre only $50 each and I really rate them. A big sound on the Caribou record is the Valhalla berMod
across the whole mix to flange certain sections.
For delays, I generally use [SoundToys] Echo Boy. Sometimes Ill put Phase Mistress after the delay so the delay is
phasing slightly through the track. I sometimes use Waves HDelay, but only occasionally. I quite like the EQ of those, but
then they add noise which Ive got to gate out. I wish there was a way of switching the noise off. Ive noticed on the newer
Waves plugins, you can dial in the noise, which is much better. Ive also got some impulse responses. If I work
somewhere cool, Ill take an impulse response off their plate. Ive got a really good [EMT 240] Gold Foil impulse that I use
a lot.
generations of
music
technology (from
National Health
band in the 1970s to hits
with partner Barbara
Gaskin. For his latest
project, he faced the
challenge of bringing his old
multitracks and MIDI
sequences into the
computer age.
Secrets Of The Mix

Engineers:
Humberto Gatica
Inside Track: Michael
Bubl Youre Nobody Till
Somebody Loves You
In a rare
interview,
legendary
engineer and
producer Humberto Gatica
explains how he and singer
Michael Bubl breathed new
life into big-band swing
music with stunning
results.
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In this article:
On The Beach
Coast To Coast
Too Many Tracks
Keeping Track
The DASH For Digital
The Mystery Of The
Missing Compressors
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Inside Track: Tom Lord-Alge

Secrets Of The Mix Engineers
Buy PDF
People + Opinion : Artists / Engineers / Producers / Programmers
For Rivers Cuomo and Weezer, going back to basics meant

a reunion with one of the biggest names in rock mixing:
Tom Lord-Alge.
Paul Tingen
orry guys I didnt realise that I needed you so much / I thought Id get a new audience, I forgot that disco sucks / I
ended up with nobody and I started feeling dumb Take me back, back to the shack / Back to the Strat with the
lightning strap / Kick in the door, more hardcore / Rockin out like its 94.
Thus sings Weezer frontman Rivers Cuomo in Back To The

Shack, the lead single from the bands ninth studio album
Everything Will Be Alright In The End. Rarely has an artist
expressed a mea culpa to his fans in such unequivocal terms.
The quote refers back to Weezers eponymously titled 1994
debut album, also known as The Blue Album. Produced by
former Cars frontman Ric Ocasek, its combination of raw,
grunge-inspired rock and quirky power-pop gained it sales of
over three million and classic album status.
Rather than repeat this formula, however, Cuomos restless
creative spirit pushed him to continually experiment, and the
band took their most dramatic change of direction with 2009s
Raditude and 2010s Hurley. Both were made in collaboration
with producers, artists and songwriters working in the commercial pop, hip-hop and R&B arenas, such as Dr. Luke, Polow da
Don, Jermaine Dupri, Lil Wayne, Desmond Child, Greg Wells and Rick Nowels. Sadly for Cuomo and his bandmates, no new
audience arrived, their original fans were not impressed, and the bands album sales plummeted even faster than those of the
music industry in general.
Cuomos intention on Everything Will Be Alright In The End is
thus crystal-clear: stop the rot by going back to basics and giving
Weezers fans exactly what they want, namely a return to hardhitting melodic rock. As part of this plan, the band once again
hired Ric Ocasek to produce. They also asked Tom Lord-Alge to
mix the new album, reuniting a team that had last worked
together on the third Weezer album, 2001s self-titled effort
known as the Green Album.
On The Beach
Tom Lord-Alge is the younger brother of Chris Lord-Alge, and
between them, they appear to have cornered the market in
mixing heavy, in-your-face, commercial rock. Moreover, both of
them share the same unique mix approach, which sees them
using a Sony 3348 48-track digital tape machine as an
intermediary between Pro Tools and their favoured SSL desks.
Tom Lord-Alges impressive credits list includes Oasis, U2, the
Rolling Stones and Peter Gabriel among many other famous
names.
Studio A at Miamis South Beach Studios.
Tom Lord-Alges first studio experiences date from the mid-80s, when he became assistant to his brother at Unique Studios
in New York. Their working together laid the foundations for their similar approaches, as Tom recalled in a previous interview
in SOS, in April 2000, when he explained that From a technical perspective, Chris taught me everything. Many of the tricks I
still use, I got from him.
In 1995, Tom Lord-Alge was asked to mix an album at South Beach Studios in Miami, and almost literally never left again.
He still resides there today, in Studio A, with pride of place going to a large SSL desk thats not just used as an expensive
coaster or laptop support. To be precise, the desk is a 4064 G+ with Ultimation and E-series EQ. The studio also holds LordAlges favoured Sony 3348HR, which is capable of 24-bit operation at 44.1 and 48 kHz resolutions (the original 3348 was 16bit only). Before diving into the Weezer mix, Lord-Alge explains his general mixing approach, and why the 3348 is still one of
his main mixing tools of choice.
Of course I use Pro Tools, which to me is like a tape machine on steroids. I use plug-ins, automation, its editing capabilities,
and spend a lot of my time working with it. I then go digitally out from Pro Tools to my 3348HR, which functions as my D-A
converter, and then to the SSL. I use the 3348 for several reasons. First of all, I love the way the converters sound. I refer to it
as vintage digital, if there is such a thing! Another reason for still using the 3348 is that I am a strong believer in data backup,
and I make sure that I have multiple copies of everything. I keep the 3348 multitracks of each session, and I print the main
stereo and main instrumental mixes back onto the same multitrack. I also print the mix back into the Pro Tools Session,
running it via a TC Electronic Finalizer, which I call poor mans mastering, but this is just for listening files for the artist and
management.
People have to wake up to the issue that there is no standard

in the way things are recorded or archived. If you come back to a
DAW mix in 10 years, theres no way that you can be sure that
youll still be able to play that mix back the way you mixed it. Its
easier for me to just play back a 3348 multitrack tape than to
deal with the bullshit that goes with opening up an old Pro Tools
Session. A few months ago Avril Lavigne asked me to stem out
music I mixed for her more than 10 years ago. I tried opening the
Pro Tools 5 Session, but although it opened reasonably well, I
quickly decided that it was easier to just load my 3348 multitrack
tape, pull out the recall data and lay the mix out over the console
and do the stems like that.
Coast To Coast
Though Tom Lord-Alge now rarely uses outboard hardware,
he still makes full use of his SSL G desk.

South Beach Studios is part of the Marlin Hotel, which is an art
deco building steeped in rock & roll history (it was owned for a
long time by Chris Blackwell of Island Records fame). The building itself, its near-waterfront location and the Florida climate
make it a very attractive place for bands to come and visit, according to Lord-Alge. What rock band would not want to come
over to Miami Beach for two weeks!? Its not unusual that they stay at my house, which I enjoy. I encourage bands to come
down, because its a lot easier to have them in the room. I can immediately see their reactions while I am working. Having said
that, Weezer stayed in LA, and the entire mix project was done with Rivers and I communicating via email. We never had a
phone conversation, which I saw as, Wow, this guy really trusts what I do. He had a big picture in mind, he knew what he
wanted, and he related this to me in our correspondence. They were after that vintage Weezer sound, and the good news for
me was that the sessions were very well recorded. There wasnt a lot of trickery necessary during mixing. Basically it was a
matter of getting the drums to sound good and the guitars loud and aggressive.
I am a big fan of the band, and because I have worked with them before, I know their sound and what they are looking for.
You can tell by listening to the new record that it was made today, with more bass and high end, but the songs and the
performances hark back to the older records. The main difference was that I used a lot more compression this time, because
these particular songs lent themselves so well to having that kind of spank. I did not feel bashful about being heavy-handed
with compression! The only debate was about the amount of effects Rivers had on his vocal. In general he prefers as little as
possible, and occasionally he wanted me to change the balance and turn a harmony line into a lead vocal, which I thought
was very interesting. There were also some songs on which he had very particular ideas about the way the guitars were
panned, in one case like a string orchestra. Finally, Rivers is a fucking genius, and he constantly updates and changes things.
His songs arent done until they actually are on the record. I knew this would be the case, and had reserved some extra
mixing time to incorporate last-minute changes. It was a fun time.
Too Many Tracks

In this day and age, its unusual for a single mix engineer to do an entire high-profile album, but Tom Lord-Alge takes pride in
doing so often, and was tasked with mixing the entire Weezer record. Back To The Shack was the first song he mixed, which
had certain repercussions for his mix approach.
Back To The Shack was one of the more straight-ahead songs, explains the mixer. If you look
at the Pro Tools screenshots you can see that it was recorded in a modern way, with for example
each guitar part split out over three or more tracks. When you have 10 guitar parts, it can add up to
40 tracks for the guitars alone! The original session was 150-160 tracks, even though the song and
the arrangement are actually quite simple. There were Sessions that came in with 90 channels of
guitars that really only needed to be three or four channels! As work progressed on the album, I
would more and more bounce things down to one audio track per performance, which made it a lot
more manageable. If I showed you a Session for a later mix for the album, youd see that its much
more concise. This made it a lot quicker for me to do what I needed to do.
Robert Plant
'Angel Dance'
Inside Track | Secrets Of
The Mix Engineers
Thirty years after
Led Zeppelin
ended, Robert
Plant has
reached a second career
high. His latest hit album
was tracked and mixed by
Mike Poole, using a mouthwatering selection of vintage
equipment.
Nashville Guitars:
Recording Today's
Country Guitar
Sounds
Interview | Engineers
With country
guitars, what you
hear on the
record is what
was played in the studio. We
asked Nashville's leading
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those tones.
Mike Vernon:
Producing British
Blues
Mike Vernon
produced some
of the greatest
blues records of
all time. A full decade after
retiring, he's back in the
studio with some of the
British blues scene's
brightest lights.
There really is no reason to have that many tracks. Make a commitment and stick to it! But its a
trend amongst modern engineers to record many mics on many different tracks, and keep their
options open. In this case it may have been hard for the engineer to simply keep his head above
water, because Rivers works really quickly and will record so many different bits and pieces. The
engineer may never have gotten the chance to go back in and clean this stuff up and make it a bit
more manageable. But in general modern engineers are under the impression that this is the way to
record. It works, but you end up with super-big sessions that are difficult to manage, without any
apparent benefit from the size. So I keep telling young engineers: if you have something that sounds
great, bounce it down! You can always hide the individual tracks or save them in an archive session,
just in case you do want to go back.
Keeping Track
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Working with the Sony

DASH digital multitrack
means that complex
sessions with
hundreds of tracks
need to be boiled
down to fewer than 48
for mixing, as here
with Weezers Back
To The Shack.
Lord-Alge has a very methodical approach to mixing, and data management is a major part of his
job. He explains: Ive heard horror stories about people losing Sessions or parts of them, and that
shit never happens to me. When a new Session comes in, my assistant, Eddie, will first organise it
to my liking. Hell place it in the order that I like, with percussion and/or any loops at the top, then the
drums, in a particular order, bass, guitar, and so on. He will also relabel things, so that I can
immediately see whats what. I want him to remove what I call the hieroglyphics, which is whatever
guys call the tracks. There are no standards, so they call them these whacky names that nobody but they can figure out. For
me, if a part is the main guitar, I simply want it called main guitar. I call the bass drum foot, so therell be tracks labelled ft. I
am also really anal about removing capital letters, because I dont like these names shouting at me. Its distracting. Eddies job
is to make every session that comes in look in a similar way, so I immediately know what Im looking at, and I dont have to
scroll up and down so much. Basically, when I open a Session up, I want it to look the same as the Session I opened up
yesterday. This makes it easier and faster for me to work with.
Another thing Eddie does is create kick and snare trigger tracks at the top of the Session. In the case of Back To The
Shack theyre called Ft Gate Trg and Snr Gate Trg, and I use these to trigger gates to get rid of any leakage on the bass
drum and snare drum. The tracks can also be used to trigger samples. My assistant will go through the entire Session, beat by
beat, to make sure everything is perfectly aligned and in time. This is very time-consuming, but I want it perfect, and theres no
program that makes it perfect, so it needs to be done by eye and by hand.
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Below these trigger tracks you can see gate dummy tracks,
which are there to make sure the delay compensation, which
comes in when I add plug-ins, does not affect the moment the
gate or sample is triggered. Once my assistant is done, I will if
necessary go through and time-align everything to the snare
drum and make sure there are no phasing issues. The tracks that
I do this with have the prefix fixed. If I want the slight delay that
you naturally get between, say, the snare mic and the overheads
mic, Ill leave it as it is, but in this song I wanted the drums to be
really aggressive and tight, so I time-aligned all the drum mics to
the snare drum mic.
Eddie also makes sure everything is routed to the designated
tracks on the 3348, and the corresponding tracks on the desk,
with one channel for the bass drum, one channel for the snare,
two channels for the toms, two channels for the overheads, one
channel for the bass, and so on. But my assistant does not do
the actual bouncing to the 3348. In fact, apart from when I print
the mix to the 3348 I am just monitoring through it and the desk,
and I spend most of my time mixing in Pro Tools, using plug-ins
and so on. Generally, 80 percent of my work is done with what I
call faders up on the desk, which means that my desk faders dont move, and instead Im working in Pro Tools, where I try to
get as close as I can to a finished mix. Ill be adjusting and riding the levels of all the instruments and the vocals in Pro Tools,
apart from for the drums, because Im using quite a bit of console compression on them, which means that I cant really do the
rides in Pro Tools.
I tend to use plug-ins, rather than volume automation, to
adjust the volume levels of tracks in Pro Tools, which again is a
matter of data management: I want to be able to just glance at
things and see what I have, and not have to think about it,
because Im trying to get a mix rhythm going. Im trying to focus
on the song. I trained myself to have the volume levels on all
tracks in Pro Tools at zero, using a plug-in gain, so theres no
guessing as to where the level is, or if I make a change, where it
was. I get many Pro Tools Sessions in with the bass at -5dB and
the overheads at -12dB and so on, and it becomes very
confusing to look at. Its much easier to run things at unity gain,
and use the plug-ins to set the gain. Like this, the Session
becomes dummy-proof!
Everything Will Be Alright In The End was recorded at LAs
If I am muting a section of a track, Ill tend to mute regions. If I
celebrated Village Recorders.
want to do volume rides on a track, Ill write it into the volume
Photo: Hannes Bieger
automation, and set the track to automation read. This allows me
to immediately see which channels have volume automation,
and if there is no automation, I know theyre all reading zero. Once the mix is close to completion, I slave Pro Tools to the
3348 and Ill start moving faders on the console. At that point I use the console like a mastering console, a finishing device,
adding some top end or other EQ and compression here and there, and as I mentioned, Ill do any necessary drum rides on
the desk as well.
Drums: Drawmer Dynamics, Bomb Factory BF76, Waves SSL E-channel & C4, Avid Digirack EQ III & D-Verb, SSL desk
EQ & compression.
Tom Lord-Alge: This particular track had two complete drum kits, consisting of the main drums, which spread out over many
tracks, and a second drum kit which had the same amount of tracks, although it only played three fills in the entire song. So I
mixed the second drum kit down to just one stereo track, and I indicated that with the checkmark in front of the track Fills. A
checkmark tells me that I compiled several tracks to that track. This was not an unusual event with this album, with several
songs having different drums for the verses and the choruses, or a bridge section, and so on. In general I treat the chorus
drums as God, and I set them up as the main drums on my desk, because you always want the choruses to be big. In the
songs in which the verse drums had a totally different sound, Id mix them in the box and they might then come up on the desk
on just two channels. In this case the fills came up on channels 3-4.
Next in the session are a Fixed hi-hat track and 10 Fixed regular and big room tracks, and the big room tracks are routed
to GatedBigRoomS56. All these big room mics had a lot of sustain, so I grouped them to bus 5-6. The plug-ins on the
GtdBgRmS56 track are the Drawmer gate, which is triggered from the bass drum track near the top of the session, so they
only open when the bass drum played, and they then went through a Bomb Factory 76 compressor, which is pummelling it,
and then theres a Waves E-series SSL EQ. It comes out on my room channels, 7-8. Below the GtdBgRmS56 track are five
fixed mono room mic tracks, which I ran to another sub below, which has the Waves C4 on it to control some of the cymbals,
bringing down some of the harsh frequencies at 2.3kHz. I think this was the only song that had this many room mics. The
regular setup for the album was two stereo room mics and five mono room mics, and this song had three extra stereo room
mics. Theres also an SSL E-series plug-in EQ on the individual mono room tracks, which are rough-in [ie. preliminary] EQs
which I apply regularly.
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Below the mono room mics are the actual kick, snare, tom and overhead
tracks. There are Ft In, Ft Out and Sub Kick bass drum tracks, and on the
Ft In mic I have the Digirack EQ, which I used to suck out the frequency that
gives that sound of a basketball bouncing, which is around 700Hz. I use the
EQ3 because I can get a really tight Q with it. In this case it had an 18dB drop
at 760Hz with a Q of 10. Next is the SSL E-channel EQ, just adding some top
and bottom end, and also adding some gain to balance it with the other two
Ft mics. It seems like I did something similar on the Ft Out mic, though I
used the EQ3 here to take out a lot of the 200Hz range. The SSL E-series
plug-in is again mostly used for gain, because I like to use my Sessions with
all my channels at zero, so plug-ins are either gain enhancers or reducers. All
kick tracks, and also the snare tracks, also have a Drawmer gate plug-in, to
remove some of the leakage. The snare tracks also have the SSL E-series,
just for EQ and gain, and theres a snare reverb track with the D-Verb. On the
SSL desk I was using desk EQ and compression on most of the drums.
Bass: Focusrite D2 & D3.
Weezer have a phenomenal bass player, Scott Shriner, who gave me really
good tones. You can see four channels of the main bass, which were run to a
sub channel called Bass Sub b15, on which I had the Focusrite EQ plug-in,
Trying to recapture the sound that originally
Focusrite compression, and again some Focusrite EQ to add some 300Hz.
brought them success, Weezer turned once
Theres also four bass fill tracks, which come in during the second verse, the
again to former Cars frontman Ric Ocasek as
producer.
bridge and the solo. In the bridge the main bass is doubled with a distorted
bass sound. These four tracks also go to a bass submix track, on which I had
Focusrite compression. It all worked, and the main thing I did was create a stop at the end of the guitar solo, right on the
words take me back.
Guitars: Waves Eddie Kramer Guitar & SSL E-channel, Avid Digirack Expander/Gate, Focusrite D2, SPL Vitalizer, desk EQ.
From top to bottom, theres an Xtra guitar, main guitar 1, main guitar 2, guitar riff, break guitar, guitar solo, high guitar
line and low guitar line. Each of them consists of multiple tracks, and the grand total is about 40 tracks of audio. I took the
two main guitars in the song and panned them left and right, and the Xtra guitar is in the middle, to give the chorus a jump; it
kind of blends in with the bass and creates this totally over-the-top distortion. Generally speaking, Ill spank the hell out of the
latter to give it as much mid-range and bite as possible.
The three Xtra guitar parts go to the submix track below, on which I have the Eddie Kramer Guitar plug-in, which goes
through the Digidesign Expander/Gate, because the Eddie Kramer plug-in is very noisy! But I like what it does to the treble
and mid-range. It adds a nice bite to the guitar. Finally, theres an SSL E-channel EQ, taking out a bit around 200Hz. Again its
a rough-in EQ, though I probably didnt add anything on the console. I had the same signal chain, the Eddie Kramer,
Expander/Gate and SSL E-series EQ, on the two main guitar submix tracks, and the first two plug-ins were also on the two
Riff sub tracks and the solo sub. The high guitar line tracks and Moog go to another submix track, which has the Focusrite
plug-in, for rough-in EQ, again the Eddie Kramer, and then it goes to the SPL MkII Vitalizer, which I use to create some stereo
expansion. I would later have added desk EQ to the guitars.
Vocals: Focusrite D2, Bomb Factory BF76, Avid Digirack Pitch & Expander/Gate, SPL Vitalizer, SoundToys Echo Boy,
Waves CLA Vocals.
The main vocal track has a Focusrite EQ going into the Bomb Factory BF76. I think the BF76 is a great vocal compressor. It
has a great sound. Ive tried loads of other ones, and I find that they all sound very similar, and it really is about which one is
the easiest to control. I like the BF76 because it kind of bites and spits at you. Theres a lead vocal double in the choruses,
which also has the Focusrite and BF76. Below that theres a StrSprdb track, which has the Digirack Pitch, with which I take
the left side one cent down and move the right side one cent up, put a slight delay on either side, and then add the Vitalizer,
again for stereo expansion, and all this creates what I would call a room sound on the vocal. Its very light, and subtle and
wide, and when he sings more loudly, you can hear the ringing in the room. Its one of my tricks.
The Bridge Delay vocal track has a 172ms delay, from the Echo Boy, to give the illusion of Rivers entering a slightly bigger
room in the bridge. The bridge harmonies also have that 172ms delay, and a Focusrite EQ, and the BF76, and below that is a
Bridge Harmony sub, which has my brothers vocal plug-in from the CLA collection, for some more space, and again the
Expander, and the BF76 to compress the whole thing. I had the same setting on the woo-hoos. Rivers doesnt like any effects
on his vocals, he likes them dry. I like that too, but did my best to create some very subtle ambience around his voice.
Stereo mix: SSL desk compression, TC Electronic Finalizer.
At the bottom of the Session is the final mix, updated because I made some minor adjustments to it. Below that are vocal
stem, and harmony, woo-hoo, a cappella, instrumental, vocal up, vocal down and TV track stems. All my sessions are at
44.1/24, which is my favourite setting because of what the 3348 does with it, and because I am a strong believer in data
backup I create multiple copies of everything. My stereo mix and my main instrumental mix get printed to four channels of the
3348, right out of the Quad compressor on my SSL console, and are also printed into the Session that I am printing from. I
print the final mix back into the Session via a TC Electronic Finalizer, which is what I call poor mans mastering, It really is just
to give it a level bump, just a touch of brick-wall limiting that really is more for listening copies for the artists and management
to listen to, so it has at least a real-world volume level when they play it in their car or other audio devices. I have found out
over the years that it needs to sound as close to a final mastered version as possible.
I also run another Pro Tools rig and record the mixes to that at
96/24, straight out of the SSL with nothing in between, apart from
my Black Lion Audio Sparrow Mark 2 A-D converter. I like
mastering to have a hi-res copy as well, even though I am mixing
from 44.1. But when you are coming from an analogue console,
it will be adding additional air, and I found that the 96k version
often comes out really well. Many record companies these days
like to sell hi-res formats online on specialist web sites. I left Ted
Jensen, the mastering engineer, 4-5dB headroom in the files that
I sent him, so he had space to be creative.
Once again, this mix was not about reverbs or other trickery. With Weezer its mostly about compression. Its about getting
Plan B entered
the public eye as
a rapper, but its
as a soul singer
that he has conquered the
charts. He and his
production team revisit the
tortuous story behind The
Defamation Of Strickland
Banks.
Secrets Of The Mix

Engineers: David R
Ferguson
Inside Track: Johnny Cash
| American VI: Aint No
Grave
Sometimes the
simplestsounding music
takes the most
work to get right, and so it
was with Johnny Cashs
posthumous hit album
American VI: Aint No Grave.
Engineer and mixer David R
Ferguson was on hand at
every stage of Rick Rubins
production.
Porcupine Tree
Steven Wilson: Recording
& Marketing Porcupine
Tree
Every new
Porcupine Tree
album sells over
a quarter of a
million copies. And with
founder Steven Wilson in
control of everything from
songwriting to shrinkwrapping, theres no middle
man to take a cut. Read his
valuable advice for SOS
readers wishing to do
likewise...
Phil Thornalley:
Torn
From Rock Producer To
Pop Songwriter
Phil Thornalley
learned his trade
as a rock
engineer and
producer in the 80s. Then
he co-wrote a little-known
song called Torn...
Ray Davies
Five Decades In The
Studio
Legendary
songwriter and
Kinks frontman
Ray Davies got
his first taste of recording in
1964, and hes never looked
back.
The Stargate
Writing &
Production Team
Mikkel Eriksen
From humble
beginnings in
provincial
Norway, the
Stargate team have gone on
to become one of Americas
leading hit factories.
Songwriter and producer
Mikkel Eriksen explains how
their hard work and talent
brought success.
Dave Stewart:
Creating A New
Album From
Archive Material
Time Trial: Bringing
Multitracks and MIDI into
the 21st Century
Dave Stewarts career has
spanned several
generations of
music
technology (from
National Health
band in the 1970s to hits
with partner Barbara
Gaskin. For his latest
project, he faced the
challenge of bringing his old
multitracks and MIDI
sequences into the
computer age.
a hard, in-your-face, over-the-top, kick-your-balls sound.

Everything has to be louder! Make it sound big and keep the
essence of the song. Its all about rock!
Or, as Cuomo has it, Rockin out like its 94.
Rivers Cuomo going Back to the Strat with the lightning

strap...
The DASH For Digital

The DASH (aka Digital Audio Stationary Head) format was introduced by Sony in 1982, with a digital two-track recorder,
using quarter-inch tape and a 24-track recorder, using half-inch tape. Studer also produced machines using the DASH
format. The most famous DASH recorder of all is the 3348, 16-bit, 48-track recorder, introduced by Sony in 1988. The
3348HR (you guessed it, HR stands for High Resolution), which can handle 20- and 24-bit word lengths, and 44.1 and 48
kHz resolutions, saw the light a number of years later. (Studer also introduced a 48-track, 24-bit DASH recorder called the
D827 in 1993, while Tascam jumped on the bandwagon with the DA800/24.)
During the 1990s the 3348 was ubiquitous in professional recording studios around the world, but like all tape recorders,
it was gradually superseded by the DAW. Today Tom Lord-Alge and his brother Chris (see SOS May 2007) are two of the
most prominent remaining users of the 3348. The former recalls, My original 3348HR was the fourth machine ever made,
and I paid 254,000 dollars for it. Holy shit! Its still working perfectly. They are workhorses! To be sure I can keep it running
I bought a second 3348HR a couple of years ago, for five grand. And more recently I purchased a couple of regular 3348s
many of the parts are interchangeable with the 3348HR and one of them was only $500. I also am constantly on the
look-out for tape, and if I see it available, I buy it all up.
The 3348HR came out at a time when everyone was still working in 16-bit, and was a major advance in sound quality. I
still love the way the 3348HR sounds. There is something that the 44.1/24-bit resolution on the Sony does to the top end
that Pro Tools doesnt. It makes it kind of hard and aggressive-sounding, something that I find lacking in modern
equipment. A lot of my clients also like that sound, which I call, as I mentioned elsewhere, vintage digital. Im sure that
there are technically better D-A converters out there, but theres a familiarity about the sound that I really like, because Ive
been working with it for so long. The sound reminds me of the records that were made in the 1990s, that were digitally
recorded and mixed through an analogue console.
The Mystery Of The Missing Compressors

In Tom Lord-Alges previous interview for SOS, in April 2000, he waxed lyrically about the many compressors that he had,
and still has, at his disposal in his room at South Beach studios. They include the Urei 1176, Teletronix LA2A and LA3A,
Neve 2254, 2264 and 33609, Distressor, Inward Connections VacRac, Focusrite Red 3, and Dbx 160 and 163. Fourteen
years later, all these hardware compressors are gathering dust. On the Weezer record, which Lord-Alge describes as
lending itself to a lot of spank the mixer only used plug-ins to do the vast majority of his compression and not only
that, but relied heavily on the humble Bomb Factory BF76 bundled free with Pro Tools. So what happened?
There is no doubt that the quality of plug-ins has become better and better, explains Lord-Alge. Today I am able to get
the same sound that I was getting from my outboard from plug-ins. Once it came to a point that when I was dialling in my
sounds with plug-ins and A/Bed them with my outboard and found that there was very little difference, it was a no-brainer
to use mostly plug-ins. Its much easier, also, with respect to recalling mixes. It also removed the variables of patch cords,
which always were an uncontrollable factor when recalling a mix. You might get your mix 99 percent the same, but one
dodgy patch cord that doesnt put the entire signal through can make a difference. There are so many variables involved in
using analogue equipment that simply arent there in the box. So, for me, dialling in the sounds that I want using plug-ins
was progress. And the BF76 certainly is one of my go-to compressors. I use it instead of four or five other outboard
compressors to get me similar sounds.
Another factor in changing from mixing predominantly on the desk to predominantly in the box is that I gradually
became more and more familiar with Pro Tools and how it operates. Pro Tools also became more and more reliable, and
as I started delving deeper and deeper into it, I forced myself to become comfortable only using plug-ins. The days of me
having all my outboard patched up to my console and using it are gone. Today I use outboard only occasionally, on a
case-by-case basis, if Im looking for something very specific that I cant achieve with a plug-in. I have another set of 192s
[Pro Tools HD interfaces] just for inserts, so I can use my outboard gear as plug-ins, and I can print the outboard effect
back into the session, just to remove the variables of that piece of outboard maybe not working any more in five years
time or if I dont have it any more.
Having said all that, my mixes are still spread out over my console, because of the difference in sound, which is
indescribable. I like to say that mixing in the box sounds like looking at standard-definition television. It looks good, its
clear, but its a bit flat. But add my Sony 3348HR D-A converters and my SSL, and all of a sudden youre looking at highdefinition. All of a sudden theres a depth and a space that wasnt there before. For me its night and day.
Secrets Of The Mix

Engineers:
Humberto Gatica
Inside Track: Michael
Bubl Youre Nobody Till
Somebody Loves You
In a rare
interview,
legendary
engineer and
producer Humberto Gatica
explains how he and singer
Michael Bubl breathed new
life into big-band swing
music with stunning
results.
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Sounding Off
People + Opinion : Sounding Off
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Russ Hepworth-Sawyer
here are some things in life that always follow procedures. A film, for example, is shot, cut, promoted and released. Mix
together amazing ingredients in the right way, bake in the oven for while, then eat cake. The same used to go for
making albums: record, mix, master, press, promote and release. This was the process I had relied upon in my career
as a mastering engineer, but the music industry and listening trends have changed enormously, and continue to do so.
I think the album is a wonderful format. It can be a story, an experience, or simply 40 minutes of pure escapism. Theres an
intended start, middle and end. Its a form in which the mastering engineer (ME) can assist the artist with their statement and
intended story. We can get stuck in ensuring theres the right flow between tracks and that the listener experiences a
cohesive collection of songs. We all have those albums where were already poised, ready to sing along or break out into air
guitar as we know precisely when the next song will begin...
A lot of my clients these days, though, are heading more towards EP work, and albums much less so. I suspect this is so
they can maintain momentum and keep publicity capital by releasing lots of little EPs, via social media, rather than an album,
with its long release cycle and greater expense. But at least with an EP there is a cohesive format that we mastering
engineers can work with for the time being.
However, the current trends of listening to streamed audio and shuffling on mobile apps will only become more prevalent
and, if Im being really pessimistic, this could even see off the EP.
What does this all mean for the lowly mastering engineer? If tracks can
be played in a random order, perhaps all mastering engineers should work
to a single loudness standard, such as R128. While were at it, why not
agree on a master standard of tone, character and so on, so that Royal
Blood can sit next to Vivaldis Four Seasons in a shuffle? As bizarre as this
might sound, the consumer electronics industry might just inflict that
automated process upon us, like they have with automatic volume levelling
between tracks. Just imagine the effect that this could have if used over the
course of a mastered album of carefully segued songs.
What can we do, then? Is this the end of the album? What about the EP,
or even the formal CD single? Should we resign ourselves to a downward
spiral that will see only the song being of importance?
The people behind auto-mastering services such as landr.com clearly
think so. If youve not yet come across it, this is an online service currently
being marketed to music producers that offers algorithmic auto-mastering of
tracks, for much less than the price of an ME. Most of us mastering
engineers have yet to be convinced that this is a good idea, but have our
clients? And will developments in auto-mastering eventually produce a
result thats good enough to satisfy the would-be clients of the future? As
recording artists are forced to change their business models becoming more self-released and self-funded its quite
possible that the art of mastering could be restricted for all but those able to afford it and who see the value of it.
While we might not like the idea of the final musical product reaching a listener in a way that wasnt intended (Pink Floyd
certainly didnt when their material first went on sale in iTunes), we actually do a lot more than juggling track order. We might
create radio edits with a bit more punch, and make sure that the audio meets the technical requirements for a vinyl pressing.
Even when you have no intention of putting out any physical copies, the ME still serves an invaluable function: we are the at
the end of the line when you need a second pair of ears on a mix.
Mastering is an important, personal, communicative art form that ensures beautiful work still reaches our ears despite the
technical challenges of data-reduced audio and the various limitations of playback systems over the past 20 years. And, with
the recent resurgence of vinyl sales, some mastering engineers might still be in work for a little while longer than others.
.
About The Author

Russ is a former SOS contributor and an author for Focal Press. He spends most of his time mastering through his
company MOTTOsound.co.uk, as well as organising some rather fun weekends teaching mastering and mixing.
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The Thrill Of The New

Leader
People + Opinion : Miscellaneous
Paul White
Buy PDF
could swear that we now have at least three Christmases a year Ive only just finished
clearing up after Christmas 2013. But then the same could be said of Januarys Winter NAMM show, because as soon as
we walk in there it seems like we never left.
Ive now spent over six months of my life on Winter NAMM trips, but the excitement never wanes as theres always
something unexpected waiting to be discovered. On the one hand you know that there will be more mics, more monitors and
more same but newer lines on show, but tucked away there are always surprises too. On the live-sound side of show the
wireless mixer seems to be making greater inroads each year with companies such as QSC, Mackie, Line 6 and Behringer all
building compact units for the gigging performer. It also seems that small line-array speakers are on the increase, so it will be
interesting to see if those trends continue this January.
When it comes to pedal effects, it is probably no secret that I was astonished by how well the Electro-Harmonix B9 organ
pedal worked, and that lead me on to discover how great the Neo Ventilator rotary speaker simulators sound. I now have one
of each firmly fixed to my pedalboard, and they are a key part of my live show. Id love to see more pleasant surprises along
the lines of the B9 because if I see one more Tube Screamer clone its going to be me doing the screaming!
When it comes to recording, the escalation in the release timetable of new computers and new operating systems just goes
to show how tied we are to consumer platforms whose designers take far more notice of what social networking users and
gamers need than what we musicians consider important. I have the greatest sympathy for plug-in developers who have to
spend valuable time and resources reworking their plug-ins to cope with new hardware or new operating systems, and
sometimes I wonder how they have time to come up with any new products at all.
Apple have just launched their Yosemite operating system to replace Mavericks, but I havent been brave enough to
upgrade my own studio system, which is still happily running Mountain Lion. Ill see what happens to other users first. I did
upgrade my MacBook Pro to Mavericks, but since then most of its CPU power seems to be going into turning the coloured
beach-ball of death for longer and longer periods of time.
Despite these inevitable obstacles, Im absolutely sure well be seeing a lot of cool and worthy new recording hardware and
software at Winter NAMM 2015, so look out for those online news updates, NAMM videos and forthcoming reviews.
Paul White Editor In Chief .
Fri 19 Dec 2014
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In this article:
Get To The Point
Stay In Control
Through Thick & Thin
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Automation Action
Reaper Tips & Techniques
Buy PDF
Technique : Recording / Mixing
Get to grips with Reapers range of automation facilities.
Barry Watson
rack automation is an increasingly crucial tool in the mix engineers arsenal, and Reaper offers more control over it than
most DAWs. There are alternative ways to display automation envelopes, for instance, and theres a MIDI Learn
function for tweaking plug-in parameters using hardware knobs and faders, to give just a couple of examples. This
tutorial provides a walkthrough guide to the fundamental features and techniques.
Get To The Point

To get started with automation, you first need to enable it on each track you wish to automate: just click on the envelope
button on the bottom left of the track display to reveal the automation configuration. This window displays the list of track
parameters that may be automated, including parameters for each plug-in inserted into the effects chain.
On the automation window, tick Visible and Arm for each parameter you want to control. This allows the parameter to be
automated using an envelope that appears in a lane below the track on the arrange area.
To show or hide automation lanes displayed under the main track, right-click over the automation envelope and tick Hide
envelope. You can show the envelope again later by right-clicking on the tracks envelope button and selecting Show all
active track envelopes.
The automation mode for the track will default to Read/Trim,
meaning that any changes made to the parameters will be read
and played back. Usefully, Reapers Trim mode is independent of
the main volume fader on the track display and mixer, meaning
that the fader can still be used as a global volume control after
the automation has been drawn in. For automating parameters in
real time, Touch is the least destructive of the various modes,
since it writes and remembers changes made to parameters, but
ceases writing as soon as you stop making changes. If youre
really keen, the various automation modes are explained more
fully in the Reaper documentation, at
http://wiki.cockos.com/wiki/index.php/Automation.
Volume and pan are armed for automation, but parameters

for all plug -ins on the track are displayed.
Back on the main screen, were ready to input and edit points
on the automation envelope. To draw new points, shift-click on the envelope line. Shift and drag to move points whilst ignoring
the snap resolution this is ideal for accurate changes that need to occur fractionally before each beat division, for example.
To move a point simply click and drag, while if you click and
drag between envelope points you can make block changes to
the envelope. This might be used for tweaking volume as you
move between a quieter verse and a louder chorus, for example.
To delete points on the envelope, use Alt/Option-click. Using the
same modifier key and clicking on the envelope line between
points allows you to modify the shape of the curve.
A Control/Command-click on the envelope allows you to draw
a freehand shape, something that allows for very quick and
simple automation editing but does tend to generate a large
number of points.
You can make further changes to the shape of curves on the
automation lane. Select multiple points using the shift key and
then right-click over the selection. From the menu, navigate to
Automation envelopes can be shown on top of the waveform.
Set shape for selected points and choose one of the available
Right-click over the track display for an envelope lane to
options (for example, the Slow start/end option gives an Smove it to this Media Lane.
curve). Additionally, you can use this menu to define the default
curve shape so that all subsequent curves drawn follow the chosen shape.
As soon as multiple parameters are being automated per track, screen real estate starts to suffer due to the number of
envelope lanes. Thankfully, Reaper allows you to change display options to show multiple automation envelopes on top of the
waveform: right-click on the tracks envelope button and select Show all visible track envelopes in media lane. Once this
display option is selected, letters V and P on the keyboard can be used to show and hide the envelopes for volume and pan
over the waveform.
Stay In Control
Once the parameters to be automated have been selected, you
might like to explore ways of manipulating them more efficiently,
since it might not be practical to display every automation
envelope on screen permanently. Another neat Reaper feature is
the ability to show plug-in parameters on the track display itself,
offering an overview of settings and the chance to make
additional changes if needed. I use this feature regularly to refer
to EQ settings for each track without having to open the mixer
and relevant plug-in, thereby improving my workflow. You dont
even need to enable automation to do this:
Recording/Mixing Books
Click the envelope button on the chosen track to open the

automation configuration window.
The Acc R track has some EQ controls shown in the track
Select the user interface UI option for each parameter you
display, achieved by selecting UI next to each relevant
wish to view within the track display.
parameter in the automation window. MIDI Learn is set up by
Experiment with resizing the track display with vertical zoom
clicking the Learn button.
and dragging the dividing line. This way you can display more
of the track controls simultaneously show all EQ controls in a single column on the track display, for example.
Recording Techniques Join

in today's discussions:
Pre Amp Connections
Taking this efficiency of control a step further, the MIDI Learn facility can be used to control plug-in parameters in a more
a bit of a odd question
intuitive way. To achieve this, youll need a MIDI keyboard or control surface that has at least one slider or knob or you can
Overdubbing midi in
use a modulation wheel:
Studio One
Capturing drum
Once youve connected your MIDI controller, go to Preferences/MIDI devices. Ensure that the MIDI input is enabled and set
performance with electric
to receive controller commands by ticking the appropriate
kit... recommendations?
box.
SOS Mix Rescue articles
Open the automation configuration window for the relevant

track using the envelope button.
Select MIDI Learn for the parameter you want to tweak using
the MIDI controller and move the slider or knob on the
controller itself. The hardware control will be mapped to that
plug-in parameter.
Through Thick & Thin

There are times youll need to repeat an automation pattern
for example, for repeated filter sweeps or volume changes on
specific sections of a song and this involves copying and
pasting automation points. The simplest way to prepare a
number of points to be copied is to first set the left and right
locators to define a time selection.
After making a time selection, right-click on the envelope and
pick Select all points in time selection from the menu. Following
The Reduce Points tool can be invaluable!
this, right-click once more and choose Copy points. Set the
song position pointer to where you want the end of the copied
point selection to be located and then ensure that the envelope is selected by clicking on it. Then use Control/Command-V to
paste the points to the new location. You can place the pasted automation points more accurately by dragging them while
theyre still selected. Alternatively, use the same menu to delete or disable an automation envelope.
In other situations, you may end up with too many points in an automation envelope. This can happen when using the
freehand draw function or any real-time manipulation with snap turned off (especially when overwriting existing envelopes with
new values). Parameter values may change erratically as a result, possibly causing jerky changes to the sound. A tool for
thinning out data is the answer here, and is also available by right-clicking on an envelope:
Use the left and right locators to specify a time selection on the automation envelope (skip this step to reduce points for the
entire track).
Right-click on the envelope and select Reduce number of points. Use the slider to reduce the points on the envelope to
the required number but be careful not to remove the automation curves completely!
Ive covered only a few of the endless useful facilities here due to the limited space available to me, but delving into contextual
and preferences menus will uncover further options. Try out some of the techniques above to take automation action you
may be surprised at how positive an effect this can have on your workflow! .
DAW Tips from SOS

100s of great articles!
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In this article:
Criteria
Planning
Monitor Wizard
Composition
Get With The
Programme
Stage Planning
Take It To TiMax
Happy Landings
Precedence Effect
Dante
Probe Flight
Arts Council Funding
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Yourself!
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Deep Space
Live Surround-sound Performance
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Technique : Live Sound
For this ambitious project, John Crossley had a full live

band play through a 16-speaker system, to create an
immersive performance inspired by the Rosetta
spacecrafts journey through the solar system.
John Crossley
espite being too young to have experienced Pink Floyds quadraphonic gigs in the 1970s, I have been fascinated with
the possibilities of using surround sound in live performance. To be honest, though, I have always been a bit
disappointed with many of the experimental multi-channel experiences the effect can be impressive, but the choice
of content to show off these systems always seems to me to be a bit too, lets say, challenging!
Why is it that hardly any normal pop and rock gigs are presented in surround sound? Is it because its too technically
challenging? Too expensive? Is the audience too indifferent? Ive made it my mission to explore these questions and to
organise live performances enabling me to try out various surround approaches and see what, if anything, the audience can
get out of it.
With that in mind, I set out to write a composition that was eventually performed, in surround sound, in June this year. The
project was inspired by the journey of Rosetta, the comet chaser satellite, around our solar system, and its 10-year mission to
unlock the secrets of our universe. It was to be composed from a spatial perspective and presented it in an innovative way.
The performance was designed be an immersive experience, both aurally and visually, thanks to the help of some projected
visualisations.
During the course of my research which was the subject of
my masters degree at the University of Derby I explored
several systems, some of which, such as Ambisonics, Wavefield
and of course venerable 5.1, will be familiar to readers. But I was
particularly intrigued by a system I came across that is offered by
a small company based in Cambridge, UK, called Outboard
(http://outboard.co.uk who produce and sell a system called
TiMax. It is essentially a multi-channel matrixing system which
allows the user to route any or all of the inputs to any or all of the
outputs, with control over level and, crucially, timing delay. The
basic TiMax system consists of 16 ins and 16 outs but can be
scaled up to 64x64 I/O.
The two important aspects of the unit are the timing delays, which allow you to take advantage of the precedence effect
(see box), and its programmability, which includes the ability to morph between level and timing settings. What all this means
is that you can feed any number of speaker channels, with the speakers placed pretty much anywhere (they dont even have
to be symmetrical, though its not a bad idea), and you can harness the precedence effect to create positioning and apparent
movement of sounds.
Although the TiMax system has been used in a variety of situations it tends to be mainly used in theatre-type installations.
Robin Whitaker and Dave Haydon from Outboard were very keen to support the project and kindly loaned me a system to
use.
Criteria
I had several criteria before I started. I wanted the concert to consist of original music Ive performed concerts before
playing covers but I wanted there to be more of a creative challenge (although there were several times during the project
when I did wonder what possessed me to make that decision!).
Although I was prepared to use some sequenced tracks, I wanted the majority of the music to be generated live by a band
of musicians. Apart from the TiMax system, I also wanted to try to use equipment thats readily available to all I didnt want
anybody to be able to say: Thats all well and good but we dont have access to nuclear discombooberators, or whatever.
I wanted all the audio (as far as possible) to be distributed
around the theatre digitally over Ethernet. The University had the
worlds first installation of a Focusrite RedNet system, which
uses the Dante protocol (see box) to send multiple channels of
audio around an Ethernet system. We had a variety of units
which we could use, and although not strictly necessary it made
for an elegant and flexible system.
I wanted to record audio and video of the performance, to
generate a live stream of the performance on the night and, most
importantly of all, I wanted the audience to have a truly
enveloping experience, and to demonstrate that all gigs could
sound like this!
Fortunately I was to be able to call upon advice and support
from colleagues at the University particularly from the Sound
and Light programme team. I also enlisted a team of students to
A Yamaha CL5 console was used for the front-of-house mix,

and this was connected directly to the Focusrite RedNet units
help set up, rig and operate the sound, lighting systems and film
recording.
via Ethernet.
Kit Lane, who lectures in Sound & Light and Technical Theatre
acted as Production Manager to organise and manage the
theatrical design and rigging, and he also produced seven
amazing videos, one for each track. The videos were projected
on to a giant screen behind the band, and they were also used
as cutaways in the final edits of the video clips of the
performance, which can be viewed at
www.youtube.com/syncopateTV.
Planning
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The planning stage included organising the technical equipment

and personnel that would be needed, and also getting a focus for
the musical composition and applying for Arts Council Funding
see box.
A late-stage plan showing how all the equipment would be

connected.
Outboard provided the TiMax unit and the University already

had several RedNet interfaces, but getting a sufficient number of speakers was a different story. The theatre has a system
comprising two D&B Q7s plus two D&B QSubs on each side of the proscenium, with a D&B E12D centre fill, so I needed to
find 13 further speakers of sufficient quality and power.
The speakers were kindly loaned to the project by Simon Lewis, a colleague at the University, who runs his own PA hire
company. He was able to provide a set of JBL SRX 712 wedges that were flown around the auditorium, and these were driven
by Yamaha P7000 amplifiers.
Mark Payne of the SFL Group very kindly loaned us a Yamaha CL5 digital mixer for FOH, and Dante cards to enable the M7
to connect to the Dante network.
The Network rig for the show was put together as a dual-star network topology, using the Dante-enabled devices. The
systems input comprised Focusrite RedNet units (three 4s and a 1), racked up with a gigabit switch. This switch was also
connected to the monitor console, a Yamaha M7CL with two Dante MY cards installed. A cable ran to a switch at front of
house, which was connected to the Yamaha CL5, a laptop and the TiMax unit. A really useful feature of Dante is that the
control software can be used over the same network as the audio traffic, allowing control of the RedNet devices remotely
without creating a new network.
In the end, the network supported 32 channels of input, routed to two show consoles and a recording rig. The FOH console
then routed two broadcast feeds to the monitor console, which was then output to web broadcast. A rock & roll mix and
spatial audio was routed to the TiMax unit, as well as the recording rig.
In total the network was supporting well over 100 channels of
audio data carriage, and all without incident!
Monitor Wizard
The main issue was not to generate too much sound on stage
any sound spilling into the audience would dilute the effect of the
multi-speaker system, particularly as the surround effect relies on
sound levels across sets of adjacent speakers. Therefore I
decided we must use in-ear monitors, and that we wouldnt use
an acoustic drum kit. As the band hadnt had any experience in
playing with IEMs it was important to spend time getting used to
playing with them. So I made the decision that all rehearsals
were to be done with IEMs. In fact this turned out to be very
straightforward; I had a multi-channel/multi-input headphone
amp in my setup (a Behringer Pro 8 HA8000), and I took a direct
The programming setup, with a scaled-down version of the
feed or mic from each band member (guitar, bass, vocals,
surround system that was to be used on the night.
drums, trumpet). Everyone brought their own ear buds or
headphones and I was able to set up individual mixes. This enabled me to include the necessary loud click in drummer Bens
mix for synchronisation, whilst the rest of the band had variations to suit.
The only electronic drum kit we had available was ancient and had solid wooden pads Ben smiled politely with gritted
teeth when I asked him to use it! Fortunately Alan Barclay from Absolute Music stepped in and kindly loaned us a Trapps
electronic kit, which had proper mesh heads. I used it to trigger a set of specifically designed acoustic samples in Kontakt
running on a separate Mac Mini with a Focusrite Saffire 6 USB interface; this gave me enough outputs for separate kick and
snare channels, plus a stereo out for all the other drum sounds.
For the performance we had separate on-stage feeds direct from the Dante Ethernet network into the M7 (stage right) and
used individual headphone amps to feed wired headphones for myself, Ben on drums, Ethan on guitar and Kieran on bass,
with wireless IEM beltpack systems for Nigel on trumpet and Kay the lead vocalist. As a result our monitor mixes were superb
I did consider generating binaural versions of the surround sound mix so that the players on stage could share the same
experience as the audience, but realised that not only was that adding further (unnecessary?) technical complexity, but
creating separate monitoring mixes for each player in this way wasnt really feasible.
Composition
The music was all composed over a two-month period using Apple Logic Pro X. Although mainly composed by me, there were
some interesting collaborations; one song was developed with a friend in New Zealand and involved Pro Tools files being sent
across the ether. The title track was specially written for the performance by a young up-and-coming songwriter, Madelaine
Shepherd; and most impressively I managed to persuade two of the European Space Agency scientists to let me record
interviews over Skype and use excerpts in the show.
Knowing that the performance was to be performed in multi-channel surround sound altered the composition process in
some quite interesting ways. String sounds and other synth parts were often stacked so that I would have more sound to fill
the auditorium with. Noises were added and treated with the space in mind, and effects were often set up and manipulated in
a similar way. Thinking about left-right delays became circular! And having the height channels opened many interesting
possibilities for sounds and effects. An example of this was the multi-layered take-off effect in the first song. This was based
DAW Tips from SOS

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on the actual sounds of the Rosetta rocket taking off from the ESAs promotional video, but augmented with about 10 tracks of
rocket sounds and filtered white noise, giving plenty of scope for sound positioning and movement, which climbed up in sync
with the rocket in the video!
The finished compositions were then bounced down as individual tracks and transferred into Pro Tools both for
rehearsals and the show itself. Even though I prefer writing in Logic, Pro Tools for me is a must for recording and mixing.
Using these tracks I was able to organise partial rehearsals where necessary, and produce Soundcloud versions with parts
missing for the band to learn along to. It also meant I could programme and rehearse the surround-sound setup without the
band being needed on every occasion.
Get With The Programme

For the programming of the surround system I needed to create a working scale model of the speaker setup in the theatre. I
had my own idea of where I would like the speakers to be positioned, but I arranged a site visit to establish the practicalities of
locating and, in several cases, flying the speakers.
Derby Theatre is a typical Playhouse-style venue. The stage is about 14m wide by
10m deep with a 10m-wide proscenium opening. The auditorium is about 14m from rear
wall to front of stage, and between 10m and 14m wide. The average height of the
auditorium is roughly 7m.
With these measurements I was able to set up a replica in a rehearsal room at the
University. This was critical to be able to programme position and movements of sounds
that would translate accurately on the night. The setup included two speakers flown
overhead which I hadnt tried before! I had used this method of creating a scaled
down version of the performance space and speaker positions before with TiMax, and
images and movements made this way tended to translate pretty well although some
tweaking is usually necessary when we get in the real space.
Using performances I had captured during early rehearsal and some of the sounds I
generated during the composition process, I was able to mock up a close approximation
of the whole gig sound, which was very helpful in making decisions about balance,
positioning and movement of all the sounds.
Stage Planning
I am a great believer in diagrams, plans and lists Im one of those people who fires up
Excel whenever Im thinking of buying a new effects unit/car/holiday! About halfway
through the project I needed to visualise how it would all connect together in the hall, how we would use RedNet and the
digital desks, and also what equipment we would need and where we would be short of resources. I also like to present things
visually; it helps me and it certainly helps discussions with others. The plan shown is about version five, I think, and its pretty
close to the actual setup used on the night. It was much easier to send this as a reply to the Production Manager when he
asked: What gear are you using and where will it need to be placed?
There were still questions that needed to be answered, however. How many inputs did we need on stage? Could we record
the live show, and if so, how? Which parts of the rig would be handled digitally over Ethernet, and which would be analogue?
As far as inputs were concerned, we planned on 24, using three RedNet 4s (each with eight preamps). For the show we
actually added an extra RedNet 2, which gave me potentially 40 inputs, as I ended up having a few more outputs from my rig
than I anticipated.
The recording turned out to be quite straightforward; a Mac Pro with a RedNet PCIe card running Pro Tools took care of
that. In the plan the recording setup is shown at the back of the hall, but in reality it was on stage with me I felt more
confident that way! Of course thats part of the beauty of using digital audio over Ethernet all the live feeds were available
anywhere in the room.
The RedNet 3 shown in the plan at the FOH position wasnt needed in the
end, as we were fortunate to get the CL5 mixer with built-in Dante interfacing.
The only analogue connections were inputs on stage from instruments, outputs
from the monitor desk to the IEM transmitters, and the outputs from TiMax to
the speakers. The MIDI cable connecting my on-stage Pro Tools rig to the
TiMax unit to trigger cue points was extended using a couple of MIDI-to-XLR
adaptors and a long XLR mic cable and it worked beautifully. I was
considering sending the MIDI wirelessly but chickened out in the end; you can
have too many variables!
Take It To TiMax
TiMax is a multi-channel programmable matrixing unit with control over levels and delay. The unit I was using had 32 channels
and was equipped with Dante, as well as analogue outs for the speaker feeds. The basic idea is to use delays and take
advantage of the precedence effect (see box) to create positioning of audio sources. In actual fact the unit comes equipped
with a built-in hard drive for playback cues, which could be used for backing tracks and sound effects, cues and so on.
However, I was using it in a live mode, treating live inputs from my own playback and sources on stage.
The first part of the process, once your speakers are in position, is to set up
a collection of images. These are effectively snapshots of levels and delays

designed to position a sound source in a particular place. These images can
then be dropped onto any input, triggered in a cue or used as a starting, ending
or passing point in a movement, with the TiMax morphing between the images.
I always start by creating a complete set of images that are going to cover all
my possible needs. I then nominate the TiMax inputs for various jobs
(although these are quite dynamic), so I might have inputs 1+2 be a static widepanned stereo image, inputs 3+4 to be left and right but in the middle of the
auditorium, 5+6 to be wide-panned rear sounds, and so on. Then I will have
several channels that are used for dynamic movements or specific channels or
instruments. These can have their positions altered by using cues to trigger the
morphing actions.
I was also really excited to be using a feature thats appeared in the latest
software release called Panspace. In this, there is a two-dimensional screen
where you first place your prepared images you are able to drop in a JPG of
your actual space! So you can position your images in the venue, and then
create a path around and through your space by clicking hit points with the
mouse. Note that, as well as the 14 speakers positioned roughly in a flat circle,
I had two speakers flown at height above the audience, and I was able to send
sounds up above during their paths.
The TiMax unit at the top of this rack was

used to apply the required delays to all 16
speaker outputs.
I was then able to trigger these movements using MIDI from on stage. For
this to work the TiMax has to be the final piece of kit that connects to the amplifiers/speakers. This means, of course, that you
need to have 100 percent confidence in the system. Having used it a few times I can say Ive never had any problems: it
seems to be very well engineered, it has a redundant Dante port and power supplies, and can operate fully without a
computer connected.
Happy Landings
So did it all work, was it all worth it and what did the audience think? Well, from a sound point of view I was really happy with
the outcome. Certainly its a big wow to have sounds moving around the auditorium, but for me it was the actual sense of
space and envelopment that impressed me the most. Having a choir or string section spread around you with delays and
reverbs correctly positioned is a tremendous experience, although to be honest a lot of that can be quite subtle until you go
back to a stereo track and you ask, Where has all the sound gone?
Thats the other big difference with a standard rig: the whole hall is filled with sound, without blasting the audience from the
front weve all experienced those times when its too loud at the front and not really loud enough at the back! With this sort
of system theres an even sound level throughout the auditorium, and the overall sound is distinctly similar to listening in your
own living room at moderately loud levels loud enough to be exciting but not so loud as to be uncomfortable.
It was quite a lot of work, although much of that was in the preproduction and planning. In turn this meant that the venue setup
was not too different to a normal gig just lots more speakers
and with a few extra bits of kit involved. So could any band do
this easily? Well maybe not easily, but it is certainly doable, and
it would be relatively easy to repeat at different venues with a few
tweaks each night.
What about the audience? I talked to many of them after the
show and we distributed questionnaire sheets and had online
questionnaires available; the feedback I got was really gratifying.
Everyone who responded really enjoyed it, and we had
comments such as This would make me go to a lot more
concerts, and It sounded fantastic I was really in the middle
of the sounds.
So, would I do it again? How does next week fit your
schedule? This time, lets have 24 speakers!
.
In order to maintain the spacial illusion created by the FOH

system, on-stage sound had to be kept to a minimum
which meant using in-ear monitors rather than stage wedges.
Precedence Effect
The precedence effect, also known as the law of the first wave front, describes how, when two sounds from different
locations are heard in quick succession (1-5ms for simple sounds with fast transients, up to 40ms for more complex
sounds), fusion occurs and the brain cant tell them apart. The brain perceives the location as being defined by the sound
that it hears first. Its this phenomenon that is used to localise sounds in reverberant spaces. By delaying sounds spread
across several speakers, it is possible to fool the brain as to the location and/or movement of the sound.
Dante
Dante is a trademark of Audinate (www.audinate.com), and it is a network technology that allows the transmission of very
low-latency multi-channel uncompressed digital audio over standard CAT 5e or CAT 6 Ethernet cable. It has been adopted
by several audio manufacturers, such as Focusrite, Yamaha, Allen & Heath and Soundcraft. It can operate in Unicast
mode (with point-to-point connection) or as a Multicast system, which can send audio streams to several devices
simultaneously.
Simon Durbridge (FOH) talks about the Dante setup for the show: All signal routing for this network was determined by
use of the Dante network control software. All devices were instantly recognised by the network controller, and the network
configuration process was hassle free. All devices were set to run at a uniform implied latency of 5ms, and at a sample
rate of 48kHz. Dante works via a method of instant recognition, where devices connected to the network determine
addressing information automatically, and devices are automatically configured in line with the rest of the network.
After a discussion with Will Hoult at Focusrite, the

RedNet units were configured to output in Multicast mode,
which reduced network traffic. The units were controlled
remotely using a laptop and control software from the frontof-house position. The units were easy to use and flexible,
and the control software allowed me to see what was going
into the stage boxes remotely, which was very useful.
The Yamaha consoles were set up to be configured by
the Dante master unit, which was the control software run
on the laptop. After some tweaking, the network ran
smoothly. The most surprising thing about the adventure of
orchestrating so much data transfer is that, more or less,
everything just worked. Not only that, everything kept
working. The TiMax unit was a very impressive and
powerful piece of kit.
A screenshot of the TiMax control software, showing panning

hit points being used to create an elaborate back-to-front
ping-pong effect.
Probe Flight
By the time you read this, the European Space Agency should hopefully have successfully landed a small probe onto the
surface of the comet Churyumov-Gerasimenko. I first became conscious of the Rosetta satellite early this year when it
was due to be woken from hibernation. The more I read about the 10-year mission the more amazed I was, and I realised
it would make a great inspiration for this project. It had all the ingredients; technology, space and a dash of derring-do! The
fact that I managed to get two of ESAs top scientists involved was an extra bonus. The songs and music loosely tell the
story of the satellite, its mission and what may happen when it finally disappears around the far side of the sun.
Arts Council Funding

There are a few opportunities for funding arts activities if you search hard enough! My first port of call was the Performing
Rights Society Foundation (www.prsformusicfoundation.com), which has a variety of grants available for music-related
activities. I wasnt successful this time as its quite competitive, but its certainly worth checking out your project might
be just what theyre looking for.
I then approached the Arts Council and their Grants For Arts programme (www.artscouncil.org.uk/funding/grants-arts),
and this time I was successful and managed to get a grant that helped cover marketing costs and so on. It also meant that
the performers could all get paid!
I have to say that putting together a grant application is not a walk in the park. There are criteria you have to hit, targets
you have to achieve and reports to write. Be prepared to do a lot of research and several re-writes.
See (& Hear) For Yourself!

On the night, we had students operating three cameras feeding into a Tricaster vision mixer for a live video feed, which
was streamed on the Internet. In order to give the audience out there a taste of what was happening in the theatre, we set
up a binaural dummy head, which was added as an alternative feed which listeners could enjoy on headphones.
You can watch videos of the performance with both stereo and binaural soundtracks, as well as a short documentary
about the show, at www.youtube.com/user/SyncopateTV.
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In this article:
A Magnificent Pair
Hang The Subtlety
Fake Flange
Perky Percussion
Space Singing
Audio Examples
Audio Examples
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Freaky Filter
Studio One Tips & Techniques
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Technique : Studio One Notes
If youre in search of novel effects in Studio One, fire up

the Autofilter plug-in!
Larry the O
tudio Ones bundled plug-ins are generally excellent, and they include a few really interesting effects. I have written
before about Groove Delay and the spiffy things you can do with it. This month, Im going to look at another space
cowboy: Autofilter.
A Magnificent Pair
Envelope-controlled filters have been around for quite a while, but Autofilter has a few really nice tricks up its sleeve. (Do plugins have sleeves?) For a start, the Autofilter plug-in is actually two filters in one, and each filter can be set to one of seven filter
types. Its an interesting selection: Ladder (12, 18 and 24 dB/octave slopes), Analog SVF (12 and 24 dB/octave), Digital SVF
(12dB/octave) and Comb. Ladder is a low-pass filter, presumably more or less modelled on the old Moog filter topology. SVF
stands for state variable filter: an analogue filter topology that can be continuously varied between high-pass at one extreme
to low-pass at the other, with band-pass response in the middle. Obviously, this is a plug-in, so both the Analog SVFs and the
Digital SVF are digital filters, but the two use different algorithms, each of which has its own characteristics.
The two filters in Autofiler can be configured serially, so that
the output of filter 1 feeds the input of filter 2, or in parallel.
Single Cutoff, Resonance, and Drive knobs control both filters,
but the FLT Spread control applies an offset to the filter 2 cutoff
frequency. These filters can sound pretty clean, but I find them
most interesting when the Drive control is turned up. The Drive
knob controls feedback around the filter and adds a bit of
analogueish distortion, though even at 100 percent, the
distortion is fairly restrained.
This slow, syrupy filter sweep sounds sort of like a flanger,

but not exactly!
This is a very flexible filter arrangement, which really comes alive when you add modulation. The Cutoff and Resonance
controls can each be modulated by an envelope follower on the input and an LFO. The Cutoff and Resonance controls have
separate sliders for the modulation sources. The key thing to note is that the sliders do not go from zero at the bottom to fullon at the top; they are at zero at their mid-points, and go positive or negative from there. Forgetting this little fact could leave
you puzzled and frustrated.
The Envelope Length control determines how closely the
envelope of the signal is followed. In practice, I havent yet run
into a situation where this control has made a big sonic
difference, but I can hear some effect. Im sure the effectiveness
of this control varies greatly depending on the source.
The LFO offers the four standard waveforms, but also a 16step sequencer. Controlling a filter with a sequencer is huge fun,
and goes all the way back to the earliest analogue sequencers
controlling modular synthesizers. The LFO rate is pretty wide
ranging. It can be tempo-synced to any value from 1/64th note
Standard jazz drums are completely transformed by this

preset that uses the 16-step sequencer for filter cutoff
modulation. For something just a little less out there,
bringing in a bit of the original will introduce some cymbals,
which are not otherwise heard.
triplet up to four bars, or, when not synced, from 0.1Hz up to 30Hz. Thats wide enough to go from a fairly slow evolution to a
low audio-frequency buzz.
Hang The Subtlety

Now, Im sure there are some very delicate treatments you can get from Autofilter, but Ive been concentrating on the
awesome funky radical filtering effects I can get with it.
The biggest single factor in getting a sound from the Autofilter is the choice of filter types. The sound of a Ladder in series
with an Analog SVF is quite different from the other way around. Putting the Ladder second gives a cleaner sound to my ear,
which could be really nice for adding an effect to a delicate acoustic instrument, while putting the Ladder first feels a bit
funkier, making it useful for processed vocals or synths. Its easy to change them around, so trying different combinations to
find the best one is a good way to work.
The spacing can also create radical differences in the sound all on its own. While spacing can be set over a range of 2
octaves, I often find I get the most intense effects from values under four semitones (a major third). The resolution of the
spacing control is one cent (1/100th of a semitone), and while one cent may not often produce a useful change, small
amounts of 10 cents or more can make a clearly audible difference.
Let me give a few examples of the range of effects available from Autofilter.
Fake Flange
The preset shown in Screen 1 is a slow sweep that gives a little of the feeling of a thick flange, but with a different flavour. Of
course, flanging is swept comb filtering, and Autofilter has a Comb filter type, so you can get even closer to a flanger if you
want. For this demonstration, I wanted a very straightforward organ playing held notes, just to show how interesting Autofilter
can make even a simple source. Audio example A is just the plain organ, which, as you can hear, is indeed pretty plain. Add
Autofilter and you get a sweep as thick and slow as syrup, as in audio example B. This preset uses two band-pass filters set
about half an octave apart; sometimes, setting values in odd relationships to each other, such as a spread of -6.37 semitones,
results in more complex and interesting filtering.
I added some delay to the audio example, just for fun. Reverb
adds even more texture. Autofilter can generate material that is
pretty interesting bare, and gets even richer when you add
further enhancement.
Perky Percussion
To show how completely the sequencer transforms a percussion
track, I took one of the jazz drum loops from the content included
with Studio One Professional, pasted it in four times, and added
Autofilter. Audio example C is the original drums, while audio
example D is the processed version. The preset is shown in
Screen 2.
This very stereo vocal process uses two Autofilter instances

on bus channels. Top: the left-channel preset, with parallel
Where the LFO was free-running for the organ, here it is

filters and tempo-synced modulation. Bottom: the rightchannel preset, with two band-pass filters in series.
synced to the tempo, with a period of two bars. The filters are a
low-pass Ladder cascaded into an Analog SVF that is almost,
but not quite, set to low-pass. These filters are spaced less than an octave apart, and, as with the organ, in series. There is a
moderate amount of Drive being applied to help remove the sound a bit more from its original form. The filter cutoff is
modulated by the sequencer, while the resonance is modulated by the envelope.
Space Singing
Lets take things a step further. I enjoy animating sounds in space so that they move around the stereo field in a less
predictable way than youd get with a standard auto-pan. Ive created this vocal effect with two pre-fader sends on the vocal
channel, feeding bus channels that each have an Autofilter.
The vocal channel fader can be all the way down, since the sends are pre-fader, but in audio example E
(Autofilter_voc_example.wav), I have set the original vocal down at -30dBFS or so, panned to the centre. This is just enough
to add a bit of anchoring in the centre, but not enough to undermine the intensity of the effect.
The sends to the bus channels are not panned, but the bus channels themselves are panned to opposite sides. In screen 3
you see the left channel preset on the top, and the right channel on the bottom. Notice that the left channel Autofilter is temposynced, but the right channel is not. I find that gives the cyclic feeling of an LFO, tempered by the variation of an unrelated
cycle. The two also differ in that the left channel has the filters in parallel, while the right-channel preset has them in series.
Submixing the effects channels (and, if used, the original) using another bus channel makes it easy to add a stereo
compressor, if you need things nice and even. Depending on the mix, it can be interesting to leave some of the peakiness and
have the vocal jump in and out a bit more.
I am also going to find out what Autofilter can do in surround, putting a signal through two Autofilters, one feeding the main
left and right outputs, the other feeding a pair of rear left and right speakers; or running a signal through Autofilter and then
through a phase-decoding circuit like Dolby Pro Logic 2 and putting that in surround. Autofilter is an undervalued plug-in. I
recommend diving into it and just playing around, checking out its dual filters, sequencer and other delights. Its hard to devise
truly unique effects for a mix, but you can find some here.
.
Audio Examples
To hear the audio examples that accompany this article, and download the Autofilter presets used to create them, point
your browser at http://sosm.ag/jan15media.
Audio Examples
Example_A.wav
Example_B.wav
Before and after: a plain organ patch is transformed by the attentions of Autofilter.
Example_C.wav
Example_D.wav
Thanks to the tempo sync facility, Autofilter has plenty to offer rhythmic parts like this jazz drum loop.
Example_E.wav
Here, two instances of Autofilter create a complex, evolving stereo effect on a vocal.
DAW Tips from SOS

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Hits So Tight
And Relax
The Groove, Vol 1
Two Places At Once
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Improve Your Groove

Reason Tips & Techniques
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Technique : Reason Notes
We look at ReGroove and the new quantise features in

Reason 8.
Robin Bigwood
ne of the small but important changes introduced in Reason 8 was to put quantise settings front and centre
(figuratively speaking) in the transport strip. Theres a popup menu offering time divisions from a whole bar down to a
64th note, together with a Shuffle option. The word Quantise below it is not just a label, but a functional button: when
you click it youll quantise any notes or audio slices that are selected in the sequencer.
Its great to have such basic and important functionality so readily to hand. But if theres a downside its that the new system
does nothing to encourage any quantising subtlety, just the rigid, fullon mathematical values. They can tidy up performances
for sure, but the risk is that any sense of naturalness disappears at the same time. So what other options are there? In fact, all
Reasons traditional quantise options still remain, they just take a bit of looking for.
Hits So Tight
For the main event, open the Tool window with a press of the F8 key or, if thats mapped to something else on your computer,
by choosing Show Tool Window from the Window menu. Reason 8 has further reduced whats on offer in this floating palette,
but even if youre using an older version of the application youll find what you need. Under the Sequencer Tools pane, which
looks like a couple of crossed hand tools, theres an allimportant Quantise panel. Reason 8 users will soon notice that the
settings there are interlinked with those in the transport bar; change the value in one location and the other changes too. The
same is true for the Quantise notes during rec option, which when ticked in the Tools window causes Q RECORD to be
selected in the transport bar (and vice versa).
Then, the crucial thing that the Tools window adds is that Amount pop
up menu. It controls the strength of the quantise. 100 percent is an
absolute setting, snapping notes and audio slices precisely into the
corresponding rhythmic grid. But less than 100 percent and the same
events are only pulled toward the same position, to a greater or lesser
extent. In short, a quantise amount of 80 or 90 percent usually gives a
really rhythmic, tidy result, but avoids a soulless, wooden, computerised
quality. You hear it particularly, if subconsciously, on quantised piano
chords. One-hundred percent places all chord notes at precisely the
same moment, in a way that a human player never does. 80 percent fixes
most gross timing errors, but leaves those chord notes a few ticks apart.
The chord still sounds together, but isnt quite, and therefore retains a
natural and believable quality. The same thing works for drums and bass
lines, amongst other things, recorded in realtime.
And Relax
What about humanising an already strictly quantised, arpeggiated or
stepprogrammed part? The Tool window can do that too, with its
Random option in the Quantise panel. Set the Quantise value (the
rhythmic grid) to an appropriate setting for your material, type a numerical
value into the Random field, select what you want to loosen up, and click
Apply (or use the quantise keyboard shortcut: CommandK on Mac, Ctrl
K on Windows.
Groove quantise isnt for all styles of music, but

luckily that isnt all the unusual ReGroove console
does. Here its affecting two of the note lanes driving
ReDrum.
The unit in this Random field is ticks, and Reason uses a timing resolution of
3840 ticks per quarter note (or 240 ticks per 1/16th note, if you prefer). Thats quite
a fine resolution, but youll hear the audible effects of a value of just 10 ticks on a
typical 1/16th-note hihat pattern, for example. Forty or 50 starts to sound really
drunk!
You can heighten the effect further too, because the Tools windows also has a
Random field in its Note Velocity section. Here the amount of randomisation is
expressed as a percentage, with 1 percent almost unnoticeably subtle and 100
percent total chaos. However, for that bit of softening and loosening of intensely
programmed musical phrases, this additional option can really help.
One particular application for these randomisation options is in doubling MIDI
parts, to create a kind of programmed chorus or doubletracking effect. Try this, for
example: with a single arpeggiated synth track (or perhaps a tambourine or
percussion part) visible in the sequencer, rightclick on the track name (rather than
in the actual note lane or on a clip) and choose Duplicate Tracks and Devices;
Reason gives you a carbon copy of your track and its clips. Now, in the mixer, pan
the original a little left and the copy a little right; select some clips in one of the
tracks; in the Tool window dial in a random Quantise amount of, say, 20 ticks; and
click Apply. Result? A believable doubling or doubletracking effect. It also works
great for MIDIplayed guitar parts and percussion.
Theres more subtlety to Reasons

quantise than initially meets the eye, and
its all courtesy of the Tool window.
The Groove, Vol 1

Id hazard a guess that for many Reason users the ReGroove mixer is still a total mystery. Opened with the transport bar
GROOVE button in Reason 8, it looks, at first glance, like an audio mixer. But this thing is actually a control panel for dialling in
timing and groove alterations: conforming recorded MIDI material to different rhythmic feels that are provided as presets, or
which you can generate from your own material.
Although Reasons console approach to groove is surprising, its actually very powerful and quite elegant in its operation
and ease of use. Also, crucially, some of its features are highly useful even if youve no interest in applying any actual grooves
to your music. Thats what well look at now, and leave a more thorough investigation of ReGrooves abilities to next month.
Heres the first crucial fact about the ReGroove mixer: it houses the knob for
controlling the feel of Reasons shuffle or swing option, found on ReDrum, RPG8
arpeggiator and Matrix Pattern Sequencer, as well as in Quantise value popup
menus. Global Shuffle, at the bottom left of the ReGroove Mixer, can be both
increased or backed off to suit your taste. Try programming a quick rhythm in
ReDrum with its Shuffle switched on, and then play with this knob youll soon get
the picture!
Next, lets look at one of those ReGroove channels. Much of it is for working with
preset or usergenerated grooves, but the Slide and Shuffle knobs are active even
when youre not actually working with a groove. Simply put, Slide is a
MIDI/Instrument track-timing adjuster. A turn to the left causes notes to be triggered
earlier than their recorded positions and, to the right, later. Shuffle introduces
rhythmic imbalances either side of straight, according to whether its turned left or
right from centre.
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Both are useful for general programming. To demonstrate this, check out this
course of action, which could be used to enliven a typical stepprogrammed ReDrum
part.
First, with a ReDrum programmed using its on-board step sequencer, right-click a
blank part of it and choose Copy Pattern To Track. That dumps all the note data into
a clip on the ReDrums sequencer track, which will be used to trigger it from now on.
The size and the location of the clip are chosen with the sequencers L and R loop
markers, incidentally. You wont need ReDrums onboard sequencer any more, so
turn off its Enable Pattern Section switch.
Essentially an antiquantise, timing

randomisation puts note and audio
slices less well in time. It can loosen up
velocities too.
Next, with the new note clip selected, open the Tools window and in the Sequencer Tools pane open up the Extract Notes to
Lanes panel. Select Explode, then click Move to separate out all the notes triggering different ReDrum channels onto their
own note lanes in the sequencer.
Youll notice that in each lane theres an empty popup menu that lets you assign
ReGroove Mixer channels to the lane. For a snare drum, say, choose channel A1.
And for a hihat, A2. Enable loop playback and hit play to audition this pattern
repeatedly.
Working in the ReGroove mixer now, try turning up the Slide knob for channel A1.
This is going to push the snare (whose lane was assigned to A1) later. Experiment a
bit to see how different rhythmic feels sound, both late and early. And for channel
A2 (thats linked to the hihat lane), try introducing some shuffle. Everything should
be available, from subtle to superobvious, and youre never bound to using the
blunt weapon that is the Global Shuffle level, or applying a destructive quantise.
Next month, well look at more secrets of the ReGroove...
The Explode function is a quick way of

getting different note pitches onto
separate sequencer note lanes. Once
there the lanes can be allocated a
ReGroove channel, and easily shifted or
shuffled, nondestructively.
Two Places At Once

If you get into using Reasons Quantise Amount popup menu in the Tool window, Reason 8 users need to be aware of
one thing: its setting remains current, and active, even when the window is closed. You could run into problems if youd
previously been quantising something quite loosely, with a value of 50 percent, say, and then closed the Tools window.
Later, when you needed an accurate, precise result again, and applied it from the transport bar, the feature might appear
to be broken. So the answer is to keep that Tools window handy, and check the Amount setting before carrying out any
quantise command.
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In this article:
Prep School
All About That Bass
Kick Start
Snare Crunch
Under The Bridge
Guitars
Vocals
Synths & Effects
Route Manoeuvre
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Alternative Mix
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Max Farrar: we take on a wellcrafted contemporary pop

tune with a massive number of tracks, and create two
different mixes.
Matt Houghton
his month, were treating you to two mixes: my colleague Sam Inglis and I had need of an interesting project on which to
test some gear, and Max Farrars high-trackcount pop track Valley Girls, with its decent recording quality, neat
arrangement and catchy melodic hooks, fitted the bill nicely. Our assessments of Maxs recordings werent miles apart,
but we steered our mixes in different directions you can hear Maxs original, Sams and my mix on the SOS web site
(http://sosm.ag/jan15media). Ive written up my approach, and Sams added some valuable insights in his Alternative Mix
box.
We often preach about getting things right at source, whether that relates to arrangements and instrumentation, capturing
sounds with mics, customising synth presets, or whatever. Max had mostly done a stirling job: the song worked well; hard
work had clearly been invested in the arrangement, and most recordings (the vocals in particular) sounded great. There were
a couple of issues, though. The bass parts long-attack sidechained feel meant it often fought the groove instead of driving
it. While not overly problematic, there were some guitartuning issues in places. I was less concerned about the daunting
number of tracks, as everything seemed to have a place, and mostly it was due to the extensive doubletracking and layering
of backing vocal (BV) parts. Finally, while Maxs mix sounded decent in places, it failed to deliver its payload: the choruses and
outro just didnt make me sit up like they should.
Prep School
Much of the effort in crafting a good mix isnt in the mixing, its more about removing
obstacles to make the mix process hasslefree. Basic project housekeeping is time
well spent, and its best if you can resist the temptation to start mixing at this stage. I
went through, organising, renaming and grouping the sources, placing things in
Folder tracks and so on, to make navigation easy. I also auditioned each source,
listening for noise issues (there were very few) and to see if any editing might be
required.
Most of the work at this stage revolved around the vocals, which were obviously
the star turn. Theyd been sung well and captured with a Neumann U47 tube mic,
but there were still breath noises, lip smacks and a few timing issues, with
consonants from BV layers not being quite as together as Id like. When the timing
of such things on layered vocals isnt bang on, it can drag your ears focus around
the stereo field, which is unpleasant so I went through the BVs, splitting out and
aligning or muting breath sounds and critical consonants to make things tighter,
using the drums as a reference. You can often remove the esses from most layers
completely, leaving just a couple (on centre or oppositionpanned parts). I also
used Cubases built-in Variaudio to iron out a few minor pitch issues, but was careful
not to overdo it; some of the magic thickening in layered vocals is due to the subtle
pitch variations. Id lavish more attention on the lead vocals later, but for now I just
did a little deessing.
Despite the huge number of tracks, the

arrangement has been well thought out
and neatly structured. Roughly the top
third of this screen is taken up by the
vocals alone!
I also set up LCR pan positions for all tracks, and established a rough mix level, based on the loudest section of the song
(the last chorus/outro). I almost always start with the LCR pan system, whereby tracks are set to 100 percent left or right, or
dead centre. I might later narrow some buses in the mix, or move things into the gaps for effect, but the LCR approach sets
up a nice, wide sound stage, leaving plenty of space for critical elements in the centre.
All About That Bass

I started by training my sights on that bothersome bass part,
which I felt would make or break my ability to craft a good mix.
My instinct was to eradicate that long drawnout attack envelope
using compression and distortion, hoping that this would also
make the sound more audible on small speakers and that I could
use a transient shaper to dial in some definition. It didnt work
brilliantly! My next attempt used Variaudio to extract MIDI notes
to play a Cubase Prologue synth patch. I struggled with this too:
detecting the right noteonset times proved tricky, and Id
neglected to set a suitable project tempo before editing. Grrr. I
might have been better off tracking a new part or asking Max to
bounce the part with a different sound but, instead, I opted for
radical equalisation to emphasise the part of the sound I wanted,
and set the compressor and brickwall limiter in VladGs Limiter
No. 6 to hammer things into shape. I then finessed the timing of
the MIDI notes and filtered the Prologue part, blended the two
sounds together, and ran the result through Cubases Chorus to
add a little width. Phew!
The bass sound was really the only major problem with the
recordings. The final sound was achieved by radically EQing
the original part and hitting it hard with an analoguestyle
compressor and limiter, before blending the result with a
triggered synth patch and running the mix through a chorus
Kick Start
plugin.
The drums and percussion occupied a hefty 22 tracks, with several kick and snare samples, forward and reverse cymbals,
hand claps, shakers and some loops. As the sounds worked pretty well, I had only a few aims: first, to make sure the kick and
snare remained strong and audible, combining with the bass to create a firm rhythmic foundation; and then to EQ and balance
the percussion parts so they didnt jar or mask each other. Id EQ the guitars, keys and synths around the rhythm parts, so I
soloed the drums, percussion, bass and vocals and set to work.
The main kick was treated to a hefty 7.5dB boost at 100Hz, another (6dB) at 500Hz, and a deep notch at 200Hz. In lay
terms this emphasised the kicks knock while minimising lowmid mud (the 150250Hz area can be a real battleground!).
Cubases Compressor, with a 20.5ms attack, 208 release, and about 6.5:1 ratio, was then set to yield about 5dB of gain
reduction (I wasnt aiming for 5dB, thats just what sounded right!).
For the kick that plays during the intro, I did a similar thing a lowend boost, a dip at 200Hz (theres something about that
region I dont like on kicks) and a boost higher up, this time at 3.45kHz. The compressor, at 8:1, was almost limiting, but just
tickling the gainreduction meter. An instance of Cubases EnvelopeShaper was used to put more emphasis on the attack.
Snare Crunch

Pre Amp Connections
Overdubbing midi in
Studio One
Capturing drum
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Most snare hits doubled a kick, so I rolled the bottom off the main snare at
about 90Hz. I also rolled off the top end from 6.5kHz, to make room for the
cymbals and whooshing noises that created a nice suckandblow effect.
Sensing that the snare could sound tighter, I removed some of its tail with
Cubases Expander, with a very short attack, a ratio of 3.5:1 and a medium
release. I also experimented with distortion treatments for the snare,
settling on Cubases Bit Crusher, on its 8bit setting with its Mix slider at 32
percent wet. This kept plenty of the real snare sound while also delivering
a nice, crunchy attitude. As the mix progressed, I added a touch of EQ
(2dB boost around 500Hz to 2kHz) and a hint of reverb, courtesy of
Cubases REVerence plugin, which Id set up as a send effect.
With the kick, snare and bass working well together, the basic rhythm
was coming together nicely. The claps, cymbals and shakers didnt sound
bad, but I applied highpass filters as high as I dared; you neednt do this
for every mix, but in a dense pop arrangement it really helps to clear
space. The claps were also treated to their own ambience reverb patch
courtesy of REVerence, partly to push them back, and partly to thicken the
sound. Reverbs are usually used as send effects, but as this was intended
to alter the sounds character I used it as an insert I could later send the
result to any effects I felt were required.
On pop drums, the snare neednt sound too

natural. After experimenting with various distortion
plugins, a bit crusher was used, mixed in at about
30 percent wet to add a little edge and definition.
The kick, snare, shakers, claps and cymbals left little sonic space for the
loops, so I did little more than bracket them with high and lowpass filters
and notch out a few frequencies, so theyd contribute their feel without
clashing with anything. To make it fit with the other sounds, the breakbeat loop was also treated to a healthy HFshelf boost
(9.5dB from 2kHz), the brightening effects excesses being tamed by a 7kHz LPF. Theres no magic in these numbers it just
sounded right.
I routed the drum/percussion parts to a subgroup, where I
used VladGs excellent freebie Limiter No.6 plugin. The dual
aim was to glue the rhythm section together and make it a little
more snappy and aggressive. The main jobs were done by the
plugins compressor, which applied about 7dB of gain reduction
in the loudest section, and the soft clipper, which shaved a
fraction off the peaks in a slightly harder way than you can do
with a limiter it doesnt soften drum transients so much. Its
easy to overdo this sort of thing, and I found myself backing off
the effect a few times as the mix evolved. Later, Id add the
gentlest hint of a smile curve on this bus with Cubases channel
EQ.
Under The Bridge

There was one more kick part during the bridge. Max had
created some lovely synth and vocal parts for this section and I
wanted to strip things back, both to show these off, and to build
the anticipation so that the ensuing chorus/outro section would
demand attention. To this end, I applied some effects on the
backing vocals. A rotary speaker, for example, was automated to
come in on the spoken vocals, and I made extensive use of
level, lowpass filter and frequency automation. What you can
hear going on with the kick in that part is a combination of subtle
All of the drums and percussion were routed to a subgroup,

at which point they were treated to a little dynamic range
reduction to gel things together including some analogue
style clipping, which seemed to leave the impact of the
transients more intact than limiting. The smile EQ was added
later, as the mix evolved.
EQ and an automated Cubase EnvelopeShaper plugin, with the

attack phase increasing from 2dB to a whopping 18.2dB near the
end of the section (making the drum bus processor work hard).
As the bridge ended, I cut everything out except for a reverb tail
and as that tail lands, the chorus comes crashing back in.
Guitars
The two acoustic rhythm guitar parts, each doubletracked, just
needed to drive the rhythm along and add to the sense of width. I
oppositionpanned the doubles to the extremes, left the higher
pitched emphasis part unprocessed, but aggressively high
pass-filtered the lower part, leaving little by way of pitch
information there. On its own it sounded awful, but against the
rest of the track it worked just fine.
The bridge needed to build to emphasise the bigness of the

ensuing chorus. To achieve that, a few effects were used on
the vocals, but a lot of automation was used, here on the
spoken vocal parts but also on the kick drum.
Although some electric guitar parts suffered from slight tuning issues, I didnt bother with pitch processing, as by the time Id
rolled off the bottom end these issues became pretty much inaudible in context. Again, there were two main parts, both
doubletracked, and again I oppositionpanned them, the lower part being highpass filtered at around 120Hz, and the higher
one up at 300Hz. Once Id balanced the two to my satisfaction and added a very short stereo delay patch for one part on a
send, I routed both electric guitars to a dedicated subgroup, on which Id placed an instance of Tokyo Dawn/Variety Of
Sounds Slick EQ. This plugin, another freebie, includes a switchable loudnesscompensation system: when engaged, you
can make EQ boosts or cuts without the perceived level changing. This made it easy to make small tweaks to the overall
electric guitar sound as the mix progressed, without me having to go back to the individual tracks. I ending up adding a bit of
mid boost and some HFshelf attenuation, level automation and I automated the send to one of the two reverbs.
Vocals
Having edited the BVs and bussed the doubles to their own
subgroups, I didnt have a huge amount to do to get them
working reasonably well. Most parts were treated to little more
than a tiny bit of EQ (no more than about 34dB of boost or cut
at any one frequency), compression and reverb before I set to
work on the level automation. The Oooh sounds were treated to
the firmest compression, with about 8dB of gain reduction on that
group bus, but others only needed around 34dB at most. The
trickiest bit was finding the right level for the ad libs, but that was
just a case of being happy with my judgment there was no
great technical challenge there.
That said, while the BVs were working acceptably, I wasnt
entirely satisfied, and after discussing a draft mix with Sam, he
suggested using multiband compression on the BV bus. I tried
and ironed out the few remaining annoyances pretty efficiently.
Ill be using that trick again!
While the lead vocal part in much of the song worked with the
backing vocals, I wanted a different, softer sound for the part
in the intro, and this was achieved with the excellent freebie
plugins Tokyo Dawn/Variety Of Sound Slick EQ and VladGs
Molot compressor, along with automated use of the global
reverb and delay effects. (The Cubase compressor you can
see here was only used to add gain when refining the mix; its
not actually compressing.)
The lead vocals had been tracked as several different parts,

but there was an easy logic to Maxs organisation. A single
tracked part took care of the opening verse, and this one I
treated to a slight midrange emphasis with another instance of Slick EQ, along with some 56dB warmsounding
compression, courtesy of another VladG freebie, Molot. Later, having already automated the level fader on this part, I realised
it was a little low in the mix, and added a gain plugin.
This part was sent to both of my reverb sends. The delays Id set up for the guitar were also used, with the send level being
automated to provide a few spot delay effects, without the delay signal compromising the vocal intelligibility.
For much of the rest of the song, the lead vocal was so closely tied in with the BVs that I ended up routing these parts to the
BV bus. This comprised a doubletracked part, which I panned hard left and right, which left the impression of a thicker
sounding part in the centre I tried panning both to the centre, but this just seemed to sound better. Each of the two parts
dynamic range was controlled by an instance of Cubases Vintage Compressor (an 1176 FET compressor emulation), with
quite a slow attack and medium release but as one take had been sung a little more dynamically than the other, the
maximum amount of gain reduction I applied to each was very different thats something no preset can get right!
Synths & Effects

The synth and various whooshing effects were all in good order, so I referenced Maxs mix to see what hed had in mind, and
decided to stay true to the spirit of what hed done while also indulging my creative urges. The opening blip synth part, for
instance, I ran through a bass patch in Cubases Amp Simulator plugin, to make the tone a little more warm and rounded. I
used pan automation to make the intro move across the stereo stage, and an instance of Cubases Stereo Enhancer to add a
little more overall width, before adding an instance of Cubases EnvelopeShaper and automating the release time (for the intro
only) just for interest. Other than sending a tiny bit of it to the REVelation effect, that was pretty much it, other than to use level
automation to mute the part in one spot where I felt it was superfluous.
Route Manoeuvre
I experimented with some unusual bus routing in this mix.
Working back from the master stereo bus, this was fed by three
separate buses one for the backing vocals and the chorus
lead vocals, with the multi-band compressor patched in, another
one for all the instrument parts, and that subgroup for the earlier
lead vocal too. That enabled me to compress the backing track
independently of the vocals. Why? Well, the track really seemed
to benefit from the bus compression that I had set up, courtesy of
Tokyo Dawn Labs TDR Feedback Compressor II, and from a
touch of stereo-width enhancement, but the vocals seemed to
suffer from this setup. Leaving that all set up but routing the
vocals around the compressor, before everything hit the main
stereo bus, proved an effective solution .
The vocals were sent to a separate bus to the rest of the mix,
I used only three basic send effects on this mix: two reverbs
and a delay (actually, two hardpanned mono delays). The
REVelation send that Ive alluded to already was mostly an
ambience patch with only 12 percent tail, while the REVerence
convolution reverb was set to the LA Studio preset, which I use a
lot. To prevent things getting too splashy, I preceded it with an
instance of Cubase Deesser. That approach allows you to keep
the reverb sound fairly bright, without paying too high a price!
which was treated to dynamics processing and EQ

courtesy of yet more instances of Tokyo Dawn and VladG
plugins to make it work better with the vocals. This and
the vocal bus were then routed to the main stereo bus, which
featured no processing (although a limiter was used to add a
bit of level at the end). This way, the vocals seemed to suffer
less from the bus processing that the style of track seemed to
be crying out for.
It took several iterations of this mix, comparing the results with references and incorporating feedback from Sam and Max,
before I was really happy and I could call it job done. Phew!
.
Remix Reactions
Max Farrar: Awesome mixes! They each have their own flavour. In Matts mix, I really like the rebalance of the harmonies
very crisp and clean and I love that its more audible than my own balancing. I dig the clarity of the mix overall. I
know the bass was kind of a bitch to deal with I realise now it was the wrong bass sound it was too subby, with not
enough meat to it. It has to do with the room modes in my studio; I have a peak then a huge null in my sub, so that bass
happened to sound powerful and mask the fact that I didnt have enough harmonic content in that sound, I think.
I love what Sam did too. The muffled thing was bothering me for sure, and I feel like he really cleared up a lot of the
spectrum! Its somewhat of a departure from the original sound of the track, and I think it goes to show that I dont need to
use so much reverb/delay in my mixes, making them wash out a bit. I suppose it depends on the vibe, but either way, I
loved hearing this. Its helped me get a lot of perspective on mix choices. Thank you both. Much appreciated!
www.maxfarrar.com
Alternative Mix
When first I heard Max Farrars mix of his track Valley

Girls, I was shocked that he felt it was in need of rescuing!
It was a polished production, and obviously very well
recorded. Maxs balance made good sense, there were no
jarring holes or peaks in the frequency spectrum, and there
was lots of nice ear candy to keep the listener interested.
After a few listens, though, I began to understand what
Max meant by its having a muffled quality. There was a
slight softness and sogginess to the mix, which wasnt
helped by a prominent long reverb and a busy
arrangement; and the bass sound, with its slow attack, felt
as though it was dragging the tempo of the track
downwards. I also decided that a low vocal level in Maxs
mix was underselling a fantastic performance and
recording.
Fabfilters ProMB, set up to act as a dynamic EQ to brighten
In terms of rescuing the mix, then, there were three
the lead vocals.
main aspects to consider. The first was the bass sound.
Having failed to reshape its flabby dynamic envelope with compression, I dropped the audio file into Melodyne and used
its analysis to generate a MIDI part. Somewhat to my surprise, this worked perfectly first time. The part itself was good, so
I used this MIDI track to trigger a different bass sound from Fabfilters Twin 2 soft synth. This gave me the attack and low
end thump that was missing from the original, but on its own it sounded rather small, so I blended it back in with a heavily
processed version of the original. I used automation through the song to rebalance the two parts in places, and the
combination made for a big sound with a much more solid and urgent feel.
The second challenge was to demuffle and open up the mix. This was a matter of many small steps rather than one
giant leap, beginning with some pruning of the arrangement: I didnt feel that the blip synth melody at the start needed to
be repeated throughout the song, for example. I then wielded aggressive highpass filtering on sources such as the
acoustic guitar, aiming to clear out the low midrange.
Effectswise, I set up four reverbs and two delays, but rationed their use. Two reverbs were purely for vocals, another
just for the snare; and the only reverb that I used across multiple sources was a very short ambience patch from Acon
Digitals Verberate. This gave the electronic sources a sense of space and air, without losing any of their impact.
Compression can be the enemy of an open mix, and is rarely necessary on electronic instruments in any case, so I used it
very sparingly. Where I felt things could do with thickening up, I turned instead to saturation plugins, such as Fabfilters
Saturn and SoundToys Radiator.
That left the third challenge: restoring the vocals to their rightful place of honour. Apart from applying filtering and other
effects to recreate Maxs breakdown section, I did very little with the backing vocals, beyond bussing them to groups and
thickening them with reverb and more MicroShift. Most of my attention was focused on the lead vocal, which wasnt quite
commanding the mix as Id have liked. I began by chucking out most of the vocal doubles, as the main lead vocal
performance was plenty good enough to stand on its own. It was also very nicely recorded, but I wanted to make it cut
through without it sounding harsh or spitty, and found it hard to do so consistently using conventional EQ and
compression. I eventually hit on the idea of using Fabfilters ProMB as a dynamic equaliser, which made it possible to
brighten the vocal without exaggerating sibilance, and push the midrange without introducing honkiness. Actual level
control I left almost entirely to automation, and having kept the rest of the mix very dry, I felt there was room for a fairly
expansive combination of reverb and delay on the lead vocal. Sam Inglis
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Need For Speed

Sonar Tips & Techniques
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Technique : Sonar Notes
Become a Sonar ninja with these time-saving tips.
Craig Anderton
onar users frequently mention workflow as one of its best attributes, but there are sometimes multiple ways to
accomplish similar tasks. If you learn some of these crucial shortcuts and design philosophies, you can accelerate your
workflow even further.
Take the Multidock, for example. Compared to pre-X-series versions of Sonar, the Multidock (which can collapse to a thin
strip) greatly reduces the need to open and close windows. You can dock anything, from views, like the Console, Piano Roll,
and Step Sequencer; to individual plug-ins and soft synths, each with their own tab. To dock a window, drag it into the
Multidock or click on the button in a windows upper-left corner and select Dock in Multidock.
Choose a docked window by clicking on its tab; the dock expands if it had been collapsed. When expanded, the dock has a
splitter bar at the top so the docked window doesnt have to take up the entire track view. For example, I often set it just high
enough to leave the time ruler visible so its easy to navigate around a project. However, two exceptionally helpful toggle
shortcuts are D, which collapses or expands the dock, and Shift+D, which toggles between the dock taking up the entire track
view or its previously selected height. The D shortcut is ideal for toggling between Console and Track views.
Also useful is the ability to lock a docked window. For
example, in the Piano Roll View, as you choose different MIDI
tracks the PRV will show the currently selected track. However, if
you dock the PRV, right-click on its dock tab, then choose Lock
Contents, then that view will remain in the dock. When you
double-click on another MIDI clip, it will also dock but have its
own tab. (To undock a window, click on its tab and drag out of
the dock.) To step through the tabs, hold down Alt+Ctrl and use
the left/right arrow keys.
Finally, you can undock the entire multidock. If you work with
two screens, you may want to keep track view in one and switch
between the Console and other views in the other.
Lock the contents of multiple views in the dock, and you can
step through them without having to open and close
windows.
Keys To Success
Sonars keyboard shortcuts are extremely flexible in fact, too flexible for those who hit the O key accidentally, find
themselves in envelope offset mode, and start adjusting faders as they normally do. Personally, I like single-key shortcuts, but
if you dont, you can disable them: choose Edit / Preferences / Keyboard Shortcuts. Then type none in the Search field, and
Edit/Select/None appears in the right-hand pane. Click on it, and in the left pane, click on the shortcut you want to disable.
Click on Bind, then Apply, then Close.
The sheer number of potential shortcuts can be daunting, but to find out what a particular key combination does, click on
Locate Key and type the key combination. If it displays as bold type in the left pane, then its already assigned to the function
shown underneath the pane, but if not, its available for assignment. To find a function you want to control, narrow your search
with the Area drop-down menu, then type the functions name into the Search field.
A few reserved shortcuts cant be changed (like Space Bar, and shortcuts involving the Pause key) but anything else can be
assigned, re-assigned, or disabled. (There are reports that you cant assign all shortcuts to None, but if so, I havent reached
that limit.) You can also export a custom shortcut set for later import.
Finally, dont overlook assigning shortcuts to MIDI keys. A mini-keyboard like Korgs Nanokey 2 makes a cool remote
simply transpose it high enough (or low enough) to cover notes you dont use in normal playing.
Zoom With A View

A common DAW function is zooming in and out, and Sonar has many options apart from clicking the + and - buttons next to
the scroll bars like the spinner between these two buttons, which zooms in and out while keeping the view centred on the
Now time. Other useful zooming procedures relate to the time ruler. Hover the cursor over the top half of the Track View time
ruler, and the cursor turns into a magnifying glass; hold down the left mouse button and drag down to zoom in horizontally, or
up to zoom out. This time, the zoom centres around where you clicked, not the Now time. Holding down the right mouse
button provides vertical zooming. To zoom the entire project in or out horizontally to fit within the Track View, double-click with
the magnifying glass in the top half of the time ruler.
Even better, you can undo and redo zooms. Place the cursor
in a space within the track view, then while holding down the
right mouse button, click the left mouse button to undo the zoom.
To redo zoom, hold down the left mouse button and click the
right mouse button.
Speaking of the mouse, in Track View choose Options / Mouse Wheel Zoom Options to set up the various zooming
parameters, like whether you want to zoom in at the Now Time or the cursor when using the mouses scroll wheel. Theres an
option to scroll horizontally and vertically simultaneously, but I prefer to untick it for independent control over horizontal and
vertical zoom. Hold Alt to zoom vertically with the mouse scroll wheel, and Alt+Ctrl to zoom horizontally.
If you choose to zoom horizontally and vertically

simultaneously, then hold Alt while scrolling to zoom in or out.
For any mouse scroll zoom, holding down Shift increases the
scrolling speed.
On My Level
Youre setting up for a mix, and have your basic levels set just
right. So now you start experimenting with the levels, and...
Oops! You cant remember what some of the initial settings were.
Double-clicking on a control returns it to its default value, but
more importantly, you can customise the default value for a
channels volume, pan, gain, aux send, and aux pan controls.
Set the control to the desired default value, right-click on the
control, and for Value, choose Set Snap-To = Current. Now,
whenever you double-click on the control, it returns to the new
default value. Once theres a basic mix level set up, I always set
custom defaults for all the faders.
If you keep hitting a default shortcut accidentally, you can

assign it to select nothing.
However, note that the Inspector, Console, and Track views

can have different defaults. This is incredibly helpful when automating a mix,
because (for example) you can have one guitar level for solos, one for rhythm guitar
parts, and another for when the guitar accompanies a vocal. This is almost like realtime snapshot automation, as selected by a quick double-click. Note that when you
set a fader to its default, the two other faders follow but their default values remain
as you set them.
DAW Tips from SOS

Cubase
Digital Performer
Live
Logic
Pro Tools
Reaper
Reason
Sonar
Hiding To Nothing
In synth folder tracks, the Audio, MIDI, Synths and Hidden boxes act as show/hide
buttons that affect both the console and track views. So if multiple MIDI tracks are
driving drums (say you have a track for each drum), you can de-clutter the console
or track view easily by clicking the MIDI button. This is also simpler than hiding
individual MIDI tracks in console view once the part is done and you want to work
solely on the audio. Furthermore, hiding/showing tracks preserves the height in track
view and strip width in console view, so when you want to show the MIDI again,
everything is as you left it.
For virtual instruments I set up the audio, MIDI, and synth tracks as desired for
editing, then click the number below the Hidden button to make them go away until
needed again. This speeds up workflow compared to resizing and minimising the
individual tracks in the folder. Another track folder feature is the field below the
show/hide buttons: if you double-click in it, it becomes a notepad that holds up to
1024 characters.
You can set separate defaults for

channel faders in the Console View,
Inspector, and Track View. Doubleclicking on any of these chooses the
respective default.
Multi-tasking
Many Sonar operations let you accomplish multiple tasks in one go. For example, the Insert / Multiple Tracks option: with one
operation you can insert as many audio and MIDI tracks as you want simply choose the audio tracks common destination
(typically the master bus for audio) and include a send if desired. For MIDI tracks, choose the common MIDI port (ie.
instrument) and channel.
Selected Track Inputs is another time-saver. I have my main guitar plugged into a specific interface input, and typically
record several tracks with it. By choosing Selected Track Inputs on those tracks, I can Ctrl+click on them and assign them all
to the guitar input. I also use this trick when recording multiple vocal tracks.
Finally, Paste Special allows you to paste audio multiple consecutive times. You can specify the number of repetitions you
want, and also paste clips to a different track, a new track, or paste clips from multiple tracks into one track.
.
Macro Macro Man

AutoHotKey is a free macro creation programme, available to download from www.autohotkey.com. It is ideal for going
beyond Sonars included keyboard shortcuts. Although it takes some coding chops, Sonar user extraordinaire Steve Cook
(aka scook on the Cakewalk forums) contributes scripts from time to time that provide ingenious shortcut solutions.
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Keep Following
Blurred Lines
Automation Relocation
La Mode
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Normal Service
Pro Tools Tips & Techniques
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Technique : Pro Tools Notes
Pro Tools acting funny? Chances are youve just changed

playback mode by accident.
Mike Thornton
ave you ever found that Pro Tools has started behaving strangely, and you have no idea why? In my time providing
24/7 Pro Tools support there were a number of problems of this sort that came up over and over again, and usually,
they could be traced to features having been enabled unintentionally.
The most common of these relate to a block of buttons under the main tool set where you select the Trim tool, Selector,
Grabber and so on. These buttons are very useful, but can easily get clicked on by mistake when you are actually aiming for
the main tools above. Its then that Pro Tools starts to behave oddly for no apparent reason. So, this month, Im going to
explain exactly what each of these buttons does, what symptoms youll experience if you enable them by accident, and which
button to press to get things back to normal!
Keep Following
The two rightmost buttons have names that can seem slightly cryptic: Insertion Follows Playback and Link Timeline and Edit
Selection. In my experience, one of the things many Pro Tools users find confusing is the way playback behaviour can
mysteriously change so that playback starts from an unexpected place on the timeline.
When you hit Stop in Pro Tools, one of two things will happen.
Either the playback cursor will stay where it was when the
transport was stopped, or it will jump back to where it was when
playback started. Which of these happens depends on whether
Insertion Follows Playback is enabled. Most of the users I know
work with Pro Tools with the Insertion Follow Playback disabled,
so that the cursor jumps back to where you started to play from.
If the button is blue then this mode is enabled, and the cursor will
stay at the point where you hit Stop. Now when you hit Play, Pro
Tools will pick up from the point where you stopped.
Many problems with Pro Tools can be traced to users

accidentally hitting one of these buttons below the main Edit
tool selection.
To go into more detail, if you look in the toolbar display in the Edit window you will see a large display of the current position
of the playback cursor and to the right of that, the start, end and length of the edit selection. You will also find the equivalent
information for the timeline selection in the Transport window. If you click in the timeline without dragging, you create an
insertion point, which looks like a blinking vertical line across the selected track. Playback will start from this point, and you will
see a solid line which moves away from the insertion point as playback continues. This is called the playback cursor. Some
people find it helpful to consider the insertion point as an edit selection with no length, so that the edit selection in and out
points are in the same place.
The Insertion Follows Playback button changes the playback behaviour between two states. Unselected, playback begins at
the insertion point, the playback cursor moves away to the right as the audio plays, and when you press Stop, it returns to the
insertion point. With Insertion Follows Playback active, the insertion point follows the playback cursor, so when you stop, the
insertion point position updates to the position of the playback cursor. This makes the Pro Tools transport act in the same way
as a tape machine, with playback continuing from where you last stopped.
Thinking of an edit selection as a stretched-out insertion point makes sense of the otherwise annoying experience of
forgetting you have Insertion Follows Playback enabled, then pressing Play, and losing your carefully set-up edit selection
when playback stops. The insertion point follows the playback cursor position, thus creating a new edit selection (with no
length). Note that this is not Undoable, but choosing Redo from the Edit menu or hitting Command+Option+Z (Mac) or
Ctrl+Alt+Z (Windows) will restore the previous edit selection.
Its particularly easy for the Insertion Follows Playback setting to get changed accidentally if you have Keyboard Command
Focus enabled. You can check this by looking at the top right-hand corner of the tracks section of the Edit window and seeing
whether the a/z button is lit up. Keyboard Command Focus enables a wide range of single-key shortcuts for a whole raft of
features, which enable you to work a lot faster when you know what they do, but can wreak havoc when you dont! You can
find out more about the Command Focus mode in my article from June 2005
(www.soundonsound.com/sos/jun05/articles/protoolsworkshop.htm). With Keyboard Command Focus enabled, the N key
toggles the Insertion Follows Playback button and because the N key is very close to the space bar, it is very easy to hit it
accidentally when starting or stopping the transport. (Without the Command Focus enabled, you can use the shortcut Ctrl+N
on a Mac or Start+N on Windows to toggle Insertion Follows Playback.)
Blurred Lines
Another common problem encountered by Pro Tools users is that
no matter where you place the cursor, Pro Tools always plays
from the same place somewhere else on the timeline. This arises
because there are two kinds of selections in Pro Tools: timeline
and edit. A timeline selection is a selected area of one of the
timescale rulers that runs along the top of the Edit window:
minutes & seconds, bars & beats, samples, and so on. A timeline
selection dictates where playback or recording will start and
where it will stop or loop. This is shown by a pair of arrows in the
Many Pro Tools users perhaps dont realise that its possible
to make different selections in the timeline and in the Edit
timescale ruler, which will be blue or red depending on whether

any tracks are record-armed.
window.
An edit selection, by contrast, is a selection made on one or more tracks within the Edit window. Edit selections control
which audio or MIDI regions will be affected by editing actions: separating a clip, deleting, copying and so on. The in and out
points also appear in the timeline and, when not obscured by the timeline selection arrows, are visible as orange line and a
dot markers.
The most intuitive way of setting Pro Tools up is to link the
timeline and edit selections, so that Pro Tools automatically
creates a timeline selection to match any selection you make in
the Edit window, and vice versa. This means whatever you point
at in the Edit window will play. Many users of other DAWs are
used to making selections exclusively in the timescale ruler, and
while this is perfectly effective, it is unnecessary if timeline and
edit selections are linked. To link edit and timeline selections
either press the button below the Grabber or use the shortcut
Shift+/. If you want to know more about this feature then you can
refer to my article in the May 2010 issue
(www.soundonsound.com/sos/may10/articles/pt_0510.htm).
Automation Relocation
With Link Edit and Timeline Selection disabled, the timeline
selection shows up in blue arrows as usual, but its possible

Third in this months trilogy of eternal Pro Tools issues is the
to have a separate Edit selection which is indicated in the
problem that when you copy or move clips around a Session,
timeline by orange dots and lines.
automation data doesnt come along with them. The solution is
simple: enable the Automation Follows Edit button so its lit up blue. As with the other buttons, most of the time this one is left
in its default on state, which is why it catches us out when hit by mistake. In this case, most of the time, we want the
automation to stay with the clips as we move them around the session. If we have mixed a set of clips and then we move
them, we want that mix data to go with the clips. And remember, this isnt just fader automation, because in Pro Tools every
parameter in every plug-in is automatable: when we move clips around, we usually want all the automation for everything on
that track to move with the clips.
However, there are times when you might not want the automation
to come with the clips. For example, in audio post-production I always
import the OMF or AAF from the video editor into a new Session
created from one of my templates. I use the Import Session Data
function to bring the tracks from the OMF or AAF onto new tracks at
the bottom of my session. Before I hide these and make them
inactive, I copy the clips up onto the relevant tracks in my template:
dialogue, sync sound, music and so on. When I do this, I choose not
If your automation isnt moving when you move clips, be
to bring the automation with the clips, because I find that it is quicker
sure to enable this button.
to start from scratch than to modify the automation the video editor
has created. I always have their automation on my reference tracks if
I need to see what they did, but when copying the clips up onto my work tracks, I disable Automation Follows Edit. (Then do
my best to remember to turn it back on again before I start work on the edit and mix!)
To give another example, in music production, you might want to copy a complete chorus from early in the song to later on,
but to mix it differently. Again, disabling Automation Follows Edit will enable you to copy the chorus without the automation.
La Mode
There are three further buttons to the left of this group, and although these rarely get hit by mistake in my experience, I will
describe them for the sake for completeness. These buttons are, from left to right, Zoom Toggle, Tab to Transient and Mirrored
MIDI Editing.
Mirrored MIDI Editing: When enabled, this lets you edit MIDI clips and have your edits apply to every copy of the same
MIDI clip. This can be particularly useful when editing looped MIDI clips. In this mode, the Mirrored MIDI Editing button
flashes red once as an edit is made, to alert you that your edit is being applied to more than one clip.
Tab to Transient: With this button enabled, you can automatically get the cursor to jump to the next significant transient
using the Tab key. This lets you define selections and play ranges, as well as start and end points for new clips, without
having to zoom in on the waveform. When disabled, the cursor jumps to the next clip boundary or sync mark on that track.
Zoom Toggle: With a selection made when you hit this button, Pro Tools zooms in so that the selection fills the Edit window.
Exactly how it is displayed is determined in the Zoom Toggle settings the Editing tab of the Pro Tools Preferences. You can
read more about Zoom Toggle in my article in the January 2012 issue (www.soundonsound.com/sos/jan12/articles/pt0112.htm).
I hope this article helps you know what to look for when Pro Tools suddenly starts to behave strangely. When things start
going wrong, it is often because you have unintentionally clicked an on-screen button or hit a key, and thankfully, the solution
is often as simple as hitting another button! .
DAW Tips from SOS

Cubase
Digital Performer
Live
Logic
Pro Tools
Reaper
Reason
Sonar
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First Transpositions
Doubling Up
Vocode To Joy
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Perfect Harmony
Logic Tips & Techniques
Technique : Logic Notes
Learn how to create vocal harmony parts and doubling

effects using Flex Pitch and Logics vocoder plugin.
Buy PDF
Geoff Smith
hanks to Flex Pitch, it is relatively easy to create great-sounding vocal harmony parts in Logic Pro X. First of all, we
need a lead vocal to practise with. I have uploaded a suitable track to this months media page
(http://sosm.ag/jan15media), along with some audio examples. Drag the lead vocal onto an audio track in Logic Pro X.
Next, duplicate the lead vocal track and copy the lead vocal audio region onto the new track, so that you have two identical
tracks containing the lead vocal. Now rename the new track BVoc 1.
We are going to go through the process of creating a backing vocal harmony part using the BVoc 1 track. In the menu bar
above the main page, turn on Flex View by clicking the Flex View icon (see the red box in Screen 1). Next, in the BVoc 1
track header, click on the Flex popup menu and choose the Flex Pitch option (outlined blue in Screen 1). Double-click on the
audio region in the BVoc 1 track to bring up the Audio Track Editor. (Make sure its pane is showing the Audio Track Editor
and not the Audio File Editor you can flip between the two by clicking the File and Track buttons at the top of the editing
window.)
Go to the top of the Audio Track Editor and click the Flex View icon so that it is lit. We now have Flex View enabled in both
the main page and the Audio Track Editor. In the Audio Track Editor, make sure you are displaying the Local Inspector, as this
contains the Scale Quantize and Pitch Correction fields that we will use later. If you cant see it, go to the View menu and click
Show Local Inspector.
First Transpositions
Lets begin the process of creating a harmony part by selecting
all the notes and dragging them down by three scale degrees.
Hit Play to hear how the resulting vocal sounds when
harmonised with the original; then try transposing it further.
At this stage it is unlikely that a simple transposition will sound
good without further tweaking, unless you are moving things by a
perfect fourth or fifth. What I tend to look for is a good starting
point, or moments within a particular transposition that I find
interesting. Once you have settled on a starting point, go to the
Audio Track Editors Local Inspector and set the Scale Quantize
field to the key signature of the track. If you are using the vocal
Screen 1: The red boxes outline where to turn Flex Mode on
example included with this article, set Scale Quantize to C sharp
in the main page and the Audio Track Editor. The blue boxes
show the locations of the Flex Mode popup menu from
natural minor. Press Command+A to select all the notes, and
which we choose the Flex algorithm.
then press the Scale Quantize Q button to snap all of the pitches
to the key signature of the track. To further quantise the pitch,
use the Pitch Correction slider: try setting it to 100 percent to hear what it is doing, then adjust it to taste. To alter the harmony
part, keep the Scale Quantize set to the key signature of the track and adjust individual notes by clickdragging them up or
down. Notice that as you move the notes up and down in the Audio Track Editor, Logic snaps the notes to your selected key.
Once you have finalised all the pitches of your harmony vocal line, you may find that the pitch-shifting artifacts make the
vocal sound unnatural. To improve this we can adjust the formants of the vocal part. In the Audio Track Editor, press
Command+A to select all the vocal notes, then go to the bottom right-hand corner of one of the notes and hover over the
node. The mouse handle will display Formant Shift. I found that when creating two lower harmony parts, shifting the formants
between 50 to 100 cents improved the subjective quality of the end result. An alternative place to make your formant
adjustments is within the Track Inspector: go to the Extended Region Parameters box and use the Formant Shift field (see
Screen 2). This is great for making quick overall adjustments to an entire region. Although Logics Flex Pitch sounds pretty
good, I found I still wanted to apply a low-pass filter to the end result to remove high-frequency artifacts. Audio Examples 1
and 2 show the lead vocal with and without two-part faux backing vocals.
Doubling Up
Once you have created your vocal harmony parts, you can then use Flex Pitch to create the illusion that they have been
double-tracked. Duplicate the track and the audio file of one of your harmony parts and once again open the Audio Track
Editor. First, lets recreate the small differences in pitch youd typically get between takes. Go through the vocal part and move
the mouse point to the node above the middle of each note blob. The mouse handle will change to display Fine Pitch. Now,
add small pitch offsets of between 20 to +20 cents to each note.
Next, we are going to recreate the small timing discrepancies that would be present in a normal backing-vocal recording.
Move the mouse pointer to the front of each note, and the mouse handle will change to the resize icon. Now apply a small
amount of adjustment, pushing or pulling the timing of the duplicate vocal part slightly away from the original. If you pan the
original harmony track hard left and the new variation hard right, it gives the illusion of a double-tracked backing vocal and
adds stereo width (see Audio Example 3).
Vocode To Joy
A more unusual approach to vocal doubling is to use a vocoder to generate the double. This can be a great way to add more
weight to a vocal, imparting a synthetic quality that might be desirable in certain dance genres, for example. Call up the lead
vocal track in the Audio Track Editor and set the Flex Mode to Flex Pitch. In the local inspector,
once again set the Scale Quantize field to the key of the track and set the Pitch Correction to 100
percent. This will help clean up the detected pitches of the vocal line before we export this as a
MIDI file. Hit Play in Logic and make sure the lead vocal plays back with the correct melody,
adjusting any of the notes that are incorrect.
From the Edit menu, choose Create MIDI Track from Flex Pitch Data (see Screen 3). This will
create a software instrument track with a MIDI region containing the vocal line. Change the software
instrument to Logics EVOC 20PS Vocoder, doubleclick on the vocoder to bring up its editing
window and from the preset menu choose Clear Voice. From the side-chain input located at the
top right of the plug-in instrument window, choose the audio track with your lead vocal on as the
source. Hit Play to listen to the sound of the vocoder combined with the original lead vocal track. In
my example, I needed to transpose the pitch of the vocal melody up an octave for it to be in the
correct register to combine with the lead vocal track. Open the vocoders editing window and try
tweaking the patch by adjusting the stereo width control to full and increasing the number of bands
the vocoder has to 20 this is done in the Bands field. Audio Example 4 contains just the vocoder
sound; this could be layered in quietly behind a lead vocal in a dance track, to add weight and
width. You can use the vocoder in this way to reinforce the lead vocal, or use the same method to
add a synthetic quality to backing vocals. If you have multiple backing vocal parts, you can get
really nice results by exporting their respective MIDI regions using the Create MIDI Track option
from Flex Pitch Data, and then combining those regions into one region which can be used to drive
the vocoder.
Logic Pro X has very powerful tools for experimenting with the pitch of monophonic audio such as
a lead vocal just be careful how much you show your singer, otherwise they may never record a
harmony for you again! .
Screen 2: From within

the Inspector, the
Formants Shift field
allows you to quickly
adjust the formants of a
region.
DAW Tips from SOS

Cubase
Digital Performer
Live
Logic
Pro Tools
Reaper
Reason
Sonar
Screen 3: In the Audio Track Editor, quantise your vocals

pitch to the key of the song, then export the MIDI note data to
a new track using the Create MIDI Track from Flex Pitch Data
option. This cleaned-up MIDI part is perfect for driving Logics
Vocoder to create a synthesized doubling effect.
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Fri 19 Dec 2014
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In this article:
Separation Anxiety
Setup Overview
Overheads, Kick &
Snare
Toms, Hat, Ride, &
Room
Bass & Guitars
Musicianfriendly Spill
Reduction
Polarity Checks
Regrets, Ive Had A
Few...
Audio Examples
Session Notes:
Brushes Held Like
Hammers
Audio Files
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Session Notes
The Practical Craft Of Recording
Buy PDF
Brushes Held Like Hammers: Our recording engineer looks

at ways to reduce spill without compromising a fullband
tracking session.
Mike Senior
hen recording bands, some SOS readers seem to feel that their task is to eliminate spill at all cost, but this is
frequently to the detriment of the music, in my view, because it usually involves musicianunfriendly tactics such as
increasing the distance between the performers, isolating them from each other acoustically (necessitating
headphone foldback), and/or recording them piecemeal via overdubbing. Neurotic spillphobia is rarely good for your recorded
sonics, either, because it encourages extreme closemiking (typically the enemy of representative timbres) and robs an
ensemble recording of the benefits that judicious acoustic crosstalk can provide, namely fuller tones for each instrument and
an automatic sense of mix glue binding them all together.
Im not dogmatic enough to suggest abandoning acoustic separation as a matter of principle, because it does have an
important role to play in projectstudio productions. I just think its too often used as an excuse for hampering musical
communication or settling for underwhelming mic signals. With this in mind, Ill use this months Session Notes to examine the
mechanics of a fullband date I did recently, where we were able to get a workably lowspill result without disconnecting the
musical ensemble or getting headphones involved.
Separation Anxiety
The group in question, Brushes Held Like Hammers (http://brushesheldlikehammers.bandcamp.com), already had a couple of
selfproduced albums behind them, but were hoping to upgrade their sound to something a bit more commercial. To this end,
they had called on their friend Algis Juknevicius to help them with general production issues, while I handled the engineering
of a threeday recording session at a small studio in the neighbourhood. The aim was to end up with completed backing
tracks for six songs, so we decided to record everyone playing together to make best use of the available time in other
words, wed track live drums, two electric guitars, and electric bass at the same time.
The studio facilities were fairly basic, in that there was scant
acoustic treatment it was more of a rehearsal room than a
studio, to be honest, despite there being plenty of computer
hardware and rack gear stacked about the place. The main
recording room measured about 5 x 8 metres, with a smaller 3 x
5 metre room adjoining it, which we could use as a control room.
A store room was also available as an isolation booth for guitar
cabinets, but that would have required us to use headphone
foldback, something I was loath to do unless separation in the
main room was really a problem.
A worry people have about recording rock bands in one room
is that spill between the mics will make everything sound roomy
and distant, but that really shouldnt be a problem as long as you
dont try to combat the spill by moving the players away from
each other. This may seem counterintuitive surely you should get less spill the further apart you move the players, right?
Strictly speaking, this is true, but moving the players apart often causes more problems than it solves because, in practice, its
more the timedelay of the spill than the sheer amount of it that makes a mix sound washy. Keep the musicians close
together, on the other hand, and all the mics in the rig pick up each instrument at fairly close range, almost like an elaborate
multimiking technique, so the spill combines with each instruments primary mic signals to give a more threedimensional
and lifelike sound, rather than lingering on afterwards like reverb.
So, even though the recording room was reasonably live
sounding (the only acoustic dampening was a handful of home
made panels on the eightfoothigh ceiling), I deliberately tried
to position the instruments as close to each other as I could. The
two guitar cabs were out in front of the kit, about two feet away
from the kick drum (leaving just enough room for me to set up
the drum mics), while the bass cab was centrally placed, as
close to the drums as floor space (for guitarists/stompboxes)
would allow about seven feet away.
Setup Overview
Never having worked with this band before, I played things fairly
safe with the mic setup. My goal was to aim for a complete kit
sound in my stereo overheads, and then supplement that with
stereo room mics and spot mics on the kick, snare, hat, ride, and
both toms (the drummer only used two). However, I wanted to
hedge my bets in case I needed to dry up the sound by relying
on the sound of the close mics more, so planned to dualmic
both the snare and kick, submixing the two mics to a single track
in each case so as to make more efficient use of my 16 available
The oneroom setup for this session. Despite the close

proximity of the instruments, spill was kept low by careful
session layout, sensible use of mic polar patterns, and
placement of a number of improvised acoustic absorbers.
recorder inputs. For the guitars, I was happy to go for a close

miked sound, as much for sonic reasons as from any desire to
reduce spill listeners are now quite accustomed to this kind of
sound on record, unnatural as it is. However, I did dualmike
both cabs to increase the chances of my getting a representative
balance of each ones qualities, because I find its quite difficult
to avoid unnaturally overemphasising some small aspect of a
loudspeakers character when youve got a single mic right up
close.
Miking bass guitar is never an easy job in a budget oneroom
bandrecording situation. Because high frequencies are more
easily absorbed/deflected acoustically than low frequencies, its
very easy for muddysounding spill to build up across all the
mics in a fullband setup. The easiest solution to this is to restrict
each mics lowfrequency bandwidth with highpass filtering
but of course that isnt really an option for any mic on a bass
instrument! A common workaround is therefore to take a DI
signal from the instrument as well as miking its amp, so that you
can highpass filter the microphone signal while still retaining
spillfree lowfrequencies from the DI. This tactic also helps you
work around the poorly controlled structural and acoustic
resonances which usually compromise lowfrequency response
in projectstudio recording spaces, because you can replace the
mic signals low frequencies with the more reliable low end direct
from the instrument. (Its not just the low end of a bass DI thats
useful, though the string character picked up by a DI can help
a bass remain more audible against things like distorted guitars
and drums, especially for end listeners with smaller playback
systems.) I also recorded separate DI signals for both the guitars
as an insurance measure, as it would allow me to re-amp them
later if spill on the guitar mics caused insurmountable problems,
or in case the band changed their minds about guitar tones post
recording.
Here you can see the drum miking in detail. Notice how the
righthand overhead mic is in the shadow of the cymbal as
far as the hihat is concerned.
DAW Tips from SOS
By building an impromptu isolation tunnel over the kick

drum microphones, we were able to shield them from the
worst of the upperspectrum spill from the cymbals and
electric guitars.
Overheads, Kick & Snare

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Whenever Im recording a group of musicians together, I usually start getting sounds for whichever instrument mics are likely
to pick up the most spill: in this case, the drum overheads. These were a pair of Shure KSM141s which Id set to their cardioid
polar pattern (to keep the sound fairly dry) in an either side of the drummers head position. Although I can normally rely on
this configuration providing a reasonably wellbalanced mix of the kit elements, it really didnt work on this occasion, partly
because Mutlu, the drummer, had his hihat set quite high and wasnt shy about hitting it, so the snare and cymbals sounded
weak and distant by comparison. Unfortunately there wasnt too much I felt I could do about this issue at source without
upsetting Mutlus playing style, and there was no option to change hihats for thinner ones. I made a small improvement by
setting up a mattress just to the side of the drum kit to soak up some of the hihat sound bouncing off the closest wall, but
eventually I decided I needed to move the overheads into more of a cymbal spot mics position over the kit, further away from
the hihats strong horizontal plane of dispersion. While I was at it, I tried to position the closer overhead mic so that the hihat
was acoustically shadowed by the cymbal on that side of the kit. The disadvantage of this was that it made the hihat appear
to be located at the other side of the kits stereo image, but I felt this was a justifiable tradeoff in return for a significant
balance improvement, especially since I was pretty sure that the hihat close mic (and indeed hihat spill on the snare mic)
would relocate the instrument more sensibly in the final mix.
The two mics I was using for the kick drum were a Blue Kickball dynamic and an ADK S7 largediaphragm condenser. I
aimed the former mic at the beater contact spot through the resonant heads small cutout, finding a nice chunky attack
character a few inches in front of the drum, while the latter mic was a foot or so further out to capture the lowend weight.
Once we were satisfied with their mixed sound, Algis and I isolated the kickdrum mics from the rest of the kit using a
makeshift isolation tunnel built from a couple of spare freestanding diffusers, a mattress, and some blankets. I regularly take
this precaution when working with bands for the first time, because you never know quite how consistent the drummers
footwork will be, and its easier to use compression to control wayward levels when theres less spill.
Managing separate stands for my two snare close mics (a Shure KSM137 smalldiaphragm cardioid condenser and a
Shure SM57 dynamic) proved rather awkward under the circumstances, so I set one mic up and gaffer-taped the other to that.
Finding enough meat in the sound involved lowering the drums tuning and then placing the mics a good nine inches away
from it, so again we hedged our bets by wrapping an off-cut of acoustic foam around the sides of the little mic cluster to
mitigate the inevitable hihat spill.
Toms, Hat, Ride, & Room

Getting a decent racktom timbre was made easier by the fact that its tone in the overheads was already nice and full, so I
only really needed the close mic in there for a little extra definition. The Superlux Pro 228A I was using for it is fairly cheap and
cheerful, but managed a presentable enough sound very quickly from about five inches away. The spill it was picking up from
the cymbals was nastysounding, though, so another foam off-cut was applied to smooth this off. The same mic sounded
pretty horrible on the floor tom, however, and a Pro 218A from the same drummiking set was even worse: all stick and no
girth. So I decided to swap in an AKG C414BXLS largediaphragm condenser for richer low end, and by selecting a figure
eight polar pattern and placing it horizontally over the tom, I was also able to reject spill from the hihat and cymbals quite
effectively.
Id put Avantone CK1 cardioid condensers on the hihat and ride cymbal, and took a little time to refine their initial positions
to achieve a useful mid-range tone (offsetting the sizzle coming in as spill from other mics) and to balance the needs of both
the stickwork and the sustain components. While spill on the hihat mic was clearly not a huge concern, the spill from other
cymbals on the ride mic was more of an issue, so I decided to try out a little experiment Id been hatching. There are a lot of
offtheshelf mic baffles available these days (the most wellknown being SE Electronics Reflexion Filter range), but Ive
always wondered whether you might be able to bodge together something reasonably effective as a DIY project for a lot less
money. With that in mind, Id found an old plastic bucket, sawed a halfpipe section out of it, and stuck a bit of acoustic foam
on the inside. Mounting this to a spare mic stand allowed me to position the concave foamcovered surface around the mic,
and I was gratified to hear that it did seem to take a useful edge off the spill.

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However, it wasnt without some cost in terms of the sound of

the ride itself, which became a little cloudier into the bargain,
presumably because residual unabsorbed reflections from the
plastic surface were causing phasecancellation effects at the
mic diaphragm. On balance, though, it seemed like a positive
step, so I left the homemade baffle in place. What I didnt notice
at the time, however, was that the baffle had also impacted on
the sound of the nearby floortom mic, overlaying a rather
unpleasant highfrequency combfilteringrelated whistle onto
the cymbal spill it was picking up, a problem that only struck me
when I listened back to the files at home following the session.
Its safe to say that I probably wont be trying that particular stunt
again...
Off-cuts of acoustic foam were used around the snare and
The room mics were the final piece in the drum-kit setup
racktom mics to take the edge off cymbal, and particularly
puzzle. I knew that these would pick up everything in the room to
hihat, spill.
an extent, but I wanted them to catch more ambience from the
drums in this case, so that the kit would sit a little behind the other instruments in the depth perspective. For this reason I
placed the mics behind the kit. The mics I used were a pair of cheap AKG CBL99 boundary mics, and the good thing about
this kind of mic for ambient pickup in small rooms is that you effectively remove one of the walls from the equation the mic
doesnt see any reflections from the surface its mounted on, which weakens combfiltering and roommode-related
colorations on your recording. I initially placed the boundary mics on the rear wall, but that was a non-starter, because theyre
brighter on axis (ie. perpendicular to the surface theyre mounted on) and that was giving me way too much of the cymbals.
Locating the mics on the floor gave a much more representative balance of the kit as a whole, although again with rather too
much hihat level on the closer of the two. This was something that I initially resigned myself to addressing with EQ, but later
in the session I was able to make a substantial improvement here by propping up another small folding mattress behind the
drummer to block the direct sound path between the hihat and the closer room mic.
Bass & Guitars

The Groove Tubes GT57 largediaphragm condenser mic Id selected gave me the warm amp sound I was hoping for,
sensibly complementing the wirier DI signal. The main floortoceiling room resonance was giving an unrepresentative boost
around 70Hz, which made the balance of some of the low notes inconsistent, but I was content to rely on the DI signal for
lowfrequency consistency. Putting the mic at a different height might have helped, but I liked the sound I was getting from a
specific speaker cone in the cabinet and we didnt have any platform robust enough to lift the bass cab onto, so I chose the
lesser of the evils: an overall tone I wanted, but with some processing work required at the low end. By not firing the bass cab
or its close mic directly at the drums, I was able to keep spill levels across the bass and drum mics pretty low.
I wasnt anticipating many problems with spill onto the guitar

close mics, because I was planning to go for a typical up by the
grille cloth rock sound, and the cabinets were both set up with
their backs to the kit, such that they were physically shielding the
mics in front of them. Indeed, once Id worked up a mix of each
of the dualmic setups (a Shure SM57 and a Groove Tubes
GT55 on one amp; another GT55 with a vintage Sennheiser
MD421 on the other) the spill levels were promisingly low even
without any additional baffling, so I immediately discounted the
idea of isolating the cabs into the store room. However, I did rig
up a couple of duvets on the walls beside each guitar cabinet to
round off the tone of the guitar spill being reflected into the drum
overheads, and also placed a duvet across the top of one of the
guitar amps because the offaxis frequency response of its
condenser mic was making the ride cymbal, in particular, sound
undesirably abrasive.
An improvised baffle made from half a plastic bucket lined

with acoustic foam helped reduce hihat spill into the ride
cymbal mic, but at the expense of introducing nasty comb
filtering sideeffects for the floortom mics cymbals spill.
Musicianfriendly Spill Reduction

After all that, then, how good was the separation between the instruments? Well, the best way to answer that question for
yourself is by checking out the audio examples from this session, (see the Audio Examples box for details). However, let me
also put it into practical terms...
We were completely free to drop in replacement sections on any individual performers live take, without any appreciable
conflict between residual spill and afterthefact overdub.
We could have silenced the drums utterly at any point by re-amping the guitar parts, and we would have had no problems
muting the bass and/or guitar parts in pretty much any situation where the drums werent playing completely on their own. (We
recorded multi-track drum and cymbal samples at the end of the session too, which gave additional flexibility in this regard.)
We were at liberty to totally reinvent the bass or guitar sounds
via re-amping, and even if wed decided to change the sound of
the drums quite drastically with mixdown processing, it would
have had negligible effect on the tone of the electric instruments.
In other words, our recording session left us with all the
mixdown and overdubbing flexibility we were ever likely to need.
Crucially, however, this hadnt been achieved by inconveniencing
the musicians, who could still perform as a closeknit unit and
interact naturally during and between takes, without all the
malarkey of headphone monitoring, talkback systems, and click
tracks getting in the way.
.
Although the bass-cab mic position gave a desirable midrange tone, the rooms resonance modes caused problems
with the low consistency, so we recorded a DI signal to work
around this.
Polarity Checks
Wherever you get spill between mics, theres a potential for phaseinteraction between the mic signals, resulting in comb
filtering effects. This can be disastrous for your sonics, but it doesnt have to be. The easiest way to avoid major setbacks
is to get into the habit of checking for the bestsounding polarity setting for each and every mic you set up. On this specific
session, that meant flipping polarity switches on numerous occasions: when mixing the kick, snare, and guitar mic pairs
together; when adding in the kick, snare, hat, ride, tom and room mics to the overheads; and when combining the bass
and electricguitar mics with the drum setup. (The bass DI and mic signals needed checking against each other too.)
Whats vital to remember when doing this, though, is that you mustnt just listen to the sound of the instrument whose mic
youre adding in, because the phaseinteraction between it and other mics may actually affect the sound of any other
instrument too.
Regrets, Ive Had A Few...

While I was content with the way things came out, there were, as usual, a few things I might have done differently given a
second shot. The hihat spill over the other drum mics was my biggest complaint, and with hindsight I think it may have
been worth miking beneath the snare, as the drums acoustic shadow would have worked in my favour. Miking a snare just
from underneath isnt unheard of, by any means Bruce Botnick did this when working with the Doors, for instance. The
main thing is not to focus the mic too closely on the snare wires, and to take care with the phase relationship between the
snare and kick mics, given the increased kick spill. Another tactic I could have tried was putting both the overhead
microphones into the acoustic shadow of one of the cymbals, using a coincident or nearcoincident stereo configuration.
My other mistake was neglecting to properly phasealign
the two snare mics by ear before taping them together in
the heat of the moment I just lined them up by eye, which
proved not to be too smart. The result was that the two mic
signals were offset by around a 10th of a millisecond, and
although this wasnt enough to impact on the drums midrange tone, it did rob me of some useful snap at the high
end. The way to find the best phase match between two
mics is to flip the polarity of one of them and then slowly
move them relative to each other until you find the weakest
mixed sound. Resetting the polarity switch should then give
you the most solid sonic combination. If youre not keen on
spending time doing this during the session, you can
always set up the mics in front of a portable radio
beforehand to find the proper phase match, and then tape
them together (or mark their alignment with masking tape)
so you can rely on it later. Which is what I should have
done!
In order to reduce reflected spill from the guitars being picked

up through the drum overhead mics, we used duvets to cover
the walls beside each guitarist here theyre draped over a
spare mic stand to hold them in place.
Although the guitar mics were placed close enough to the

speakers that they didnt pick up a tremendous amount of
spill, the condenser mic on this amp gave what cymbal spill
remained an unpleasantly harsh tone. The additional duvet
draped across the top of the cab helped remedy this.
Audio Examples
You can find a wealth of audio examples taken from this months recording session on the SOS web site.
sosm.ag/jan15media
Session Notes: Brushes Held Like Hammers Audio Files

The audio files available on this page accompany my article in SOS about recording the indie band Brushes Held Like
Hammers. The first set of files comprise a section of the complete live multi-track recording which you can import into your
own DAW and try mixing for yourself, if you like. The filenames are fairly selfexplanatory, but here are some additional
notes to describe exactly what youre hearing in each case:

RawTracks01_Kick
The kick drum was recorded with two mics in front of the drum, with a makeshift isolation tunnel built around them both to
reject spill from other instruments. The closer mic was a Blue Kickball phantompowered dynamic model, set up a few
inches outside the hole in the drums resonant head and with its lowfrequency switch set for minimum emphasis so that it
would provide a good solid attack. An ADK S7 largediaphragm condenser was placed a little further away with its 18dB
pad switch in, providing lowfrequency extension and tonal warmth. The two mics were mixed to a single recorder track.
RawTracks02_Snare
The snare drum was also recorded with two mics, this time a Shure SM57 dynamic and a Shure KSM137 small
diaphragm condenser gaffertaped together about nine inches from the instrument. An offcut of acoustic foam was
wrapped around the sides of the mics to take the edge off the hihat spill, but as you can hear from 0:40 onwards theres
still plenty of that!
RawTracks03_HiHat
There was so much hihat spill on the other drum mics that I concentrated on trying to capture more mid-range body from
the instrument with its close mic. Theres still more sizzle than I wanted, but that was easy to remove with a lowpass filter
for monitoring and mixing purposes.
RawTracks04_Ride
This file is the ridecymbal mic, although the instrument isnt being played in this particular excerpt. However, the spill on
this mic nonetheless contributes to the overall drumkit sound, so Ive still provided it here so youre able to experiment with
mixing the entire multitrack take for yourself if you wish. (To hear what this mic sounded like with the ride cymbal playing,
check out the BucketBaffle02_RideWith audio file.)
RawTracks05_Overheads
These are the Shure KSM141 overheads, which were set up in a spaced configuration about three feet apart over the kit.
Given the superloud hihat, I set up the closer overhead mic in the acoustic shadow of one of the cymbals, which helped
in terms of the balance, but also made the hihat appear to be coming from the opposite side of the kit. (The overheads
were panned for an audience perspective stereo image, so the hihat should have been on the right.)
RawTracks06_RoomMics
The room mics were a pair of inexpensive AKG CBL99 boundary microphones laid on the floor behind the drummer about
six feet apart. As you can hear at 0:40, the hihat is again over-prominent in the balance (as it was in the room). Later in
the session I tackled this with some further baffling, as you can hear in the MattressBaffle02_RoomWith example file.
RawTracks07_RackTom
This is the mic over the single rack tom, a superbudget Superlux Pro 228 dynamic mic. The first thing to realise is that
the spill is comparatively benign tonally, partly because Id wrapped an off-cut of acoustic foam around the mic to take the
edge off the cymbal spill, especially from the hihat. You can hear the instrument itself playing from 0:28.
RawTracks08_FloorTom
Heres what the floor tom closemic, an AKG C414BXLS largediaphragm condenser, sounded like in the raw. Id
switched it to figureof-eight mode to reject cymbal spill, but what spill remains hasnt been helped by combfiltering
effects caused by sonic reflections from the ride cymbal mics nearby improvised baffle. For a clearer demonstration of
this, check out the BucketBaffle example files.
RawTracks09_BassAmp
The bass amp was miked with a Groove Tubes GT55 largediaphragm condenser mic switched to its cardioid polar
pattern and placed a couple of inches from the grille cloth over the bestsounding speaker. As you can hear from 0:28
onwards, there was an appreciable room resonance between the floor and ceiling at around 70Hz, and this caused the
fundamental frequencies of certain notes to be overplayed in the balance. Besides this, though, I was happy with the fairly
warmsounding amp tone as a complement to the DI signal.
RawTracks10_BassDI
This bass DI signal was taken from the DI output on the players amplifier, rather than from a separate direct box. Although
the low end is less strong than that of the miked signal, the low frequencies are nonetheless much more consistent, so I
ended up using those in the mix, together with some low EQ boost to bring them up in the balance.
RawTracks11_GtrAmp1
This is the combined signal from the first guitar cabs two closeplaced mics: a Groove Tubes GT55 largediaphragm
cardioid condenser and a vintage Sennheiser MD421 dynamic.
RawTracks12_GtrAmp2
The second guitar amp also had two close mics on it: another Groove Tubes GT55 largediaphragm cardioid condenser
and a Shure SM57 dynamic mic.
The remaining files demonstrate the effects of some of the baffling tactics I mention in the article. For all of these examples
I asked the band to play the same song section several times, so that I could move baffling materials around between
takes.
FoamBaffle01_RackTomWithout
In this audio example, you can hear how much cymbal spill was coming down the racktom mic without any baffling in
place.
FoamBaffle02_RackTomWith
With an offcut of acoustic foam wrapped around the racktom mic, the cymbal spill was significantly reduced. Compare
with the FoamBaffle01_RackTomWithout file.
BucketBaffle01_RideWithout
This is what the ridecymbal mic sounded like without any baffling. Notice how the ride cymbal itself is more aggressive,
as is the spill from the other kit components.
BucketBaffle02_RideWith
Heres how the sound was altered by adding in my improvised mic baffle constructed of half a bucket and some acoustic
foam. Overall, the spill feels less prominent, but the ridecymbal sound has also suffered, becoming less open and a little
boxy. On balance, I decided during the session that this seemed an acceptably compromise.
BucketBaffle03_FloorTomWithout
What I didnt notice during the session was that reflections from the back of my DIY bucket baffle had impacted on the
sound of the nearby floor tom mic. Heres what that mic sounded like before the baffle was added. Now compare this with
BucketBaffle04_FloorTomWith
BucketBaffle04_FloorTomWith
The DIY bucket baffle I used for the ridecymbal mic negatively affected the sound of the cymbal spill in the nearby floor
time mic, as you can hear in this audio example. Check out the section from 0:37, for instance, where a nasty 3.5kHz
resonance has developed which wasnt audible before the baffle wa introduced check out the
BucketBaffle03_FloorTomWithout file for a direct comparison.
DuvetBaffles01_OverheadsWith
This is what the drum overhead microphones sounded like during the recording session. Duvets had been placed on the
walls adjacent to each of the electric guitarists to inhibit their sonic reflections from reaching the drum mics.
DuvetBaffles02_OverheadsWithout
If you compare this audio example with DuvetBaffles01_OverheadsWith youll hear how much difference it made when I
hung up the duvets in the recording room, as described in the magazine article.
DuvetBaffles03_GtrGT55With
The duvet baffles werent just preventing guitar spill from getting onto the drum mics, but were also trying to reduce the
drum spill being picked up this guitar mic. To hear how effective this was, compare this audio example (which uses the
baffled recording setup) with the DuvetBaffles03_GtrGT55With demonstration file.
DuvetBaffles04_GtrGT55Without
When the duvet baffles are removed from around the guitars, the degree of cymbal spill immediately rises to a much
higher level, and with a more audible tone too.
MattressBaffle01_RoomWithout
For the first days recording the room mics were picking up too much hihat on the righthand mic (when seen from
audience perspective). Although I was sure Id be able to deal with that at mixdown, I still wasnt that happy about leaving
things that way, so I tweaked the setup for the second day to improve matters.
MattressBaffle02_RoomWith
By using a small foldup mattress to obstruct the sightline between the righthand room mic and the hihat, I was able to
achieve a substantial improvement in the room balance. Compare this example with the MattressBaffle01_RoomWithout
file and listen for how much less hihat you can hear.
For my final set of audio examples, Id like to demonstrate just how much separation we were able to achieve on this
session without splitting up or isolating the performers:
Spill01_RoughBalance
Heres a rough balance of the live multitracks youve already examined in the RawTracks audio examples. Processing is
very minimal: highpass filtering pretty much across the board to avoid a buildup of muddy lowmidrange spill; some
lowpass filtering on tracks which had picked up too much hihat spill, and a bit of gating on the snare close mic from 0:40
for similar reasons; some compression on the snare, drums buss, and master buss; and a halfdozen EQ bands scattered
around elsewhere. No send effects have been used at all any room sound youre hearing is from the recordings
themselves.
Spill02_DrumsSolo
Heres what the mix in the Spill01_RoughBalance file sounds like if you mute the bass and guitar mics. While you can still
clearly tell that this is a live take, few people would guess that the guitar cabs were in fact only a couple of feet from the
kick drum! In practice, this means that we could have completely rerecorded the electric instruments if we hadnt liked the
performances without any problems.
Spill03_BassSolo
Heres the bass part from the Spill01_RoughBalance audio file. Although the cab is only about 67 feet from the guitar
cabs and kick drum, the level of spill is extremely low, partly on account of the spillless DI signal thats mixed with the on
cab mic signal.
Spill04_GtrsSolo
Although I used no DI signal for either of the guitars in the Spill01_RoughBalance audio example, you can hear that their
two dualmic recording rigs still dont pick up any spill from the bass, and only very little from the drumkit mostly cymbal
spill.
Spill05_FinalMix
And heres the same section of the final release, as mixed by Algis Juknevicius. Notice how hes been able to create a
very different bass sound for the opening section, even though all the backing instruments were performed together in the
single room.
Fri 19 Dec 2014
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Technique : Cubase Notes
Set up your own hardwarecontrolled channel strip in

Cubase.
Matt Houghton
oftubes Console 1 system (http://sosm.ag/jun14console1) was an excellent idea. Having handson control over
channel selection and standard channelstrip plugins can make navigating a mix much easier than with a mouse. And
using a standard set of plugins on every channel, a means youre freed from the burden of choice, which can waste
so many of your precious mixing minutes.
Youre forced to use Softubes (admittedly excellent) plugins, though you cant use Cubases bundled plugins or any
other thirdparty ones youve invested in over the years. That got me wondering if I could create my own channel strip in
Cubase, based on whatever plugins I wished to use, and control it using one of my existing MIDI control surfaces. While my
solution may not rival the elegance of Console 1, I did find a way to do precisely what I wanted using nothing but Cubase 7.5s
Generic Remote facility, my chosen plugins and a Behringer BCR2000 MIDI controller. In this article, Ill explain how you can
configure your own channel strip and controller.
Tooling Up
The aim is to create a Cubase project template. It will feature a standard channel strip on each of the mixers audio channels.
Well use Cubases Generic Remote facility to map the BCR2000 so that it can select channels, and have a common set of
hardware controls for tweaking various parameters on the currently selected channel. Well also consider setting up some
other things, such as sendeffects routing.
Why did I choose the BCR2000? Partly because it was to hand, and partly because its
inexpensive. But a few things make it particularly handy. As it boasts so many physical
controls, you can assign plenty of functions to a single button press or knob turn, and the less
you have to think about switching from one bank of controls to another, the freer you are to
get on with the mix.
Complete control of a channel strips functions can use up a lot of controls very quickly
count the knobs and switches on the Waves SSL channel-strip plugin and youll see what I
mean. You can always be more selective in which parameters you want to access with your
controller but, as a rule, the more controls to which you have direct access in one controller
preset, the better. If your controller has only a handful, check the manual to see if theres a
way to scroll to different banks or presets. You may be able to assign the controller to an EQ
in one mode, and press another button to have it control the dynamics section, for example,
allowing you to control a more comprehensive strip than at first youd thought possible.
The BCR2000 offers useful types of control: there are many continuous rotary encoders (32
in total) and assignable buttons (20) and eight of the encoders double up as buttons to add to
that tally. The BCR2000 can receive parameter information back from Cubase and all the
encoders are circled by LEDs so they can indicate the current parameter value on the
selected channel. Rotary encoders are far better suited to our task than the analogue pots
and faders used on cheaper controllers a knob or fader that has a fixed start and end point
can cause problems when switching from one channel to another with different settings, as
the hardware control will be in the wrong place! Novation, Mackie and Icon make or have
made suitable alternative controllers, and theyre not alone. Pretty much anything with a
combination of motorised faders or rotary encoders and buttons will do.
By default, the buttons on the BCR2000 generate MIDI notes (they can be programmed to
transmit other data if you prefer). This means you could try this out with any MIDI piano
keyboard or drum pad, particularly if it also features encoders and/or moving faders.
The other thing youll need, other than a copy of Cubase, is to
decide on the plugin(s) to employ in your standard channel
strip. To keep things simple, Ill use just one (Waves SSL E
Channel) but you can choose as many as you have insert slots
available. Bear in mind, though, that keeping things in a single
plugin (such as a Neve or SSL channel-strip emulation) means
you only have one plugin GUI to open and close, which makes
it easier to see all the key channel parameters at a glance. It also
means you need only a single button to open and close it.
The first step is planning:

choose your channelstrip
plugin(s) and decide which
parameters youd like
controlled by which hardware
controls. I found it helpful to
add labels in an imaging
program, so I could work from
that when setting up the
Generic Remote.
To save your work in the Generic Remote, you need to export

it to an XML file. I lost a few hours work before I realised this!
Happily, though, you can create as many different versions as
you want.
Generic Remote
With your controller (to make explanation easier Ill assume youre using a BCR2000) hooked up to your computer, fire up a
blank Cubase project and go to Devices / Device Setup / Remote Devices and select Generic Remote. If a Generic Remote is
not visible in the list on the left, click the + sign to add one, then select the BCR2000 from the MIDI Input and Output drop
down menus.
Saving a Generic Remote file is not intuitive youd think that hitting Apply would do the job. While that updates the
details for the current project, it seems (on Cubase 7.5.2 running on Windows 7 64bit, at least) not to get recalled when
closing and reopening the project. Instead, all of the settings are stored
in XML files which you must Export to save your settings. Do that now, so
you have a backup of your starting point, and call it genericdefault.xml
or some such. Well use the same approach later to save the changes we
make to a different XML file. You can create as many XML files as you
wish, which means you could create a virtual Neve strip, a virtual API one
or whatever alongside the SSL one. You just need to create a project
template with the right plugins and import the correct XML file in the
Generic Remote.
While setting things up, youll be opening and closing the Generic
Remote Editor frequently, so I suggest creating a keyboard shortcut to
open the Device Setup window. I found that AltD (D for Device) was
easy to remember and wasnt allocated to anything else.
Configuring Your Controller

More Cubase Notes
Now to make the control names in the top half of the grid meaningful. In
the Control Name column type in names for each control you wish to use,
and when done you can delete any surplus rows to avoid confusion. For
example, for my BCR2000, I added two entries for the dualpurpose
encoders 18, one for rotate and another for push. Then I added one
entry each for the other encoders, and one for each button. If you value
your sanity, do this in an ordered, logical way, and make a note of which
button/knob/fader is called what. I found it useful to download a
picture of the BCR2000, open it in an imaging application and
add text labels so I could figure out what I wanted where.
When youre done labelling, you need to allocate a hardware
control to each row in the upper table. This sort of thing was
tedious back in the day, but the Generic Remotes MIDI Learn
facility makes it easy. Make sure the Learn tickbox is ticked, then
click in the Control Name column in the first row, and
twiddle/press the corresponding control. Cubase will fill out most
of the information for you, including the control type, MIDI
channel, note and value range. However, it will not put any data
in the Flags field. This field determines whether you want
Cubase to see signals from any given control and/or feed back
the parameter value so it can be displayed on the BCR2000. You
want both the Receive (on by default) and Transmit flags to be
enabled the first means Cubase will detect data from that
control, the second that it will send channelparameter
information back to the unit, to allow the LED rings to reflect the
current value, for instance.
Youll need to go through and do the same for each and every
hardware control you wish to be able to use, but Id suggest
getting one control up and running and controlling the desired
parameter first, as if you get anything wrong at this stage its a lot
of work to unpick! Its also a good idea to untick Learn when
editing the Flags field I tended to forget it was on when I was
working things out, and inadvertently assigned the wrong control
several times! When youve got one control working, hit Export to
save your Generic Remote as an XML file so you dont lose your
work.
In the upper table all controls need to be set to

Receive in the Flags column, and if you want your
rotary encoders LEDs to reflect the current
channels parameter value, youll need to set it to
Transmit too.
To make your control act only on the current channel, you

need to set the channel category in the lower table to
Selected. For channelselector buttons, youd choose the
relevant channel instead.
Project Template
Selecting the plugin parameter you wish your hardware
To use the Generic Remote as a channel-strip controller, youll

devices control to govern typing in the search field makes
this much quicker than using your mouse to scroll through the
need to place the same plugin(s) in the same insert slot(s) on
options.
every channel. Youre also going to need to create an audio
channel for each track youre going to want your controller to be
able to select. So if you want to use your controller to be able to select 32
channels, youll need to create 32 audio tracks. If you dont do this, youll
find that the option to select those channels is not available in the relevant
Generic Remote menu, and thats the only reason you must create them
now; you can, if you wish, delete surplus channels later, before saving your
project template. When it comes to creating a real project based on this
template, you can either save a channel preset and recall it for new
channels, or just duplicate existing channels.
Click OK to close the Generic Remote window, create an audio channel
and insert your chosen plugin(s). I put the SSL plugin in insert slot 2 so
that Id have slot 1 free for any corrective work a deesser, perhaps, or a
more surgical EQ to nix bothersome resonances without having to move
the SSL plugin to a different slot (and thus break communication with the
BCR2000). I then created another 31 channels (you can create fewer or
more if you wish it really depends on what sort of material youll be
working on) and dragged and dropped (while holding Alt) the SSL plugin to
copy it to the other channels insert slot 2.
Youll probably find that you need to set the
commands from any buttons to Toggle in the

Next, I selected all the channels in the mixer (click on the leftmost one
lower table. If rotary encoders are set to toggle,
and shiftclick the rightmost, engage Qlink at the top of the mixer) and
theyll go from zero to maximum worth bearing
enabled all the effects sends on all channels, but left the actual send values
in mind when troubleshooting!
at zero. This makes it possible to assign each send level to a specific knob
on the BCR2000 for the selected channel, without having to think about enabling or disabling the send. The idea is that you
create up to eight (you may not need or want that many) effects channels as your standard send effects. For example, you
might want a short and long reverb and delay, a widener patch, a parallel
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distortion and so on the choice is yours. Theres nothing to stop you

creating more effects tracks and having different sends for each channel
youll still have direct access to the send level. But having a basic set of
standard send effects can be a useful time saver.
Mapping Plugin Parameters

So, you have Cubase recognising your controller, and you have your
channels, standard processors and sends set up in a project. Now its time
to start assigning plugin parameters to the BCR2000s controls. This is
done in the lower table of the Generic Remote window, the row names of
which match those you created in the top table earlier. While the upper
table tells Cubase what information is coming in, the bottom one tells it
what controls to issue to your project when a knob is twisted or a button
pressed.
Each entry in the Device column needs to be set to VST Mixer. There are
other options in this column that you can explore, which give access to
transport controls, key commands, macros and so on, but as I already
have most such things assigned to keyboard shortcuts, I didnt want to tie
up the controls on the BCR2000 unnecessarily. The three other columns
youll need to tweak are Channel/Category, Value/Action and Flags.
By enabling all sends in your project template, you

wont need to switch them on or off to make use of
your dedicated hardware send controls.
Click in a cell in the Channel/Category column to access the dropdown

menu. For anything associated with the channel strip (plugin parameters, send levels and so on) set this to Selected so that
the control signal will only affect the current channel. In a similar fashion, specify the plugin parameter you wish to control via
the Value/Action dropdown list. This process is tedious if you use a mouse I found typing in the search filter at the top
much speedier. For example, typing Inp whittled the list down to the InputLevel parameter for the Waves SSL plugin on
insert slot 2, and pressing enter assigned that rows control.
For rotary controllers, you dont want anything selected in the Flags field. I found that things were set to Push Button and
Toggle by default so I had to deselect these failure to do so will lead to some strange behaviour of your plugin controls. As
the push buttons are MIDI notes, though, they need to be set to Toggle. That way, each noteon command toggles the status
of a button on your plugin, rather than engaging it momentarily.
As far as the plugin control is concerned, thats pretty much it. You just decide which parameters you want which hardware
controls to govern, select them in the table, and away you go. Do the same to give you control over the channel FX Send
levels. And remember to export your XML file to save your work (boring advice, I know, but repeating it might save you lost
hours!).
Channel Selection
So far, youll only be able to control the plugin when its instantiated on the currently selected channel, but ideally youll want
to be able to select that channel via the BCR2000s buttons. If youve not done so already, repeat the process for assigning
controls to the plugin via MIDI Learn, so that Cubase can see the hardware buttons. This time, in the bottom half of the
Generic Remote window, we dont want to operate on the Selected Channel because weve not yet selected it!
Set the Device value for the control to VST Mixer, the Channel/Category value to the desired audio track (as I mentioned
earlier, you have to have created the tracks already if you wish to be able to select them here), the Value/Action field to
Selected and the Flags to Push Button and Toggle. Press Apply, and your button should now select the specified channel.
Repeat the process for as many channels as you have buttons for. On the Behringer BCR2000, I assigned the two rows of
eight buttons at the top of the unit to the first 16 channels.
Thats not a huge number of channels for a mix is it? Fortunately, the BCR2000 has buttons that allow you to scroll from one
controller preset to another for each preset, you can assign the buttons to a different set of 16 channels. The beauty of this
is that all those controls that you routed to parameters on the currently selected channel neednt change. All you need to do is
make sure that the buttons are operating on a different MIDI channel in each preset (hold Edit, press the button and twiddle
the Ch knob). Once done, you can map the second bank of 16 buttons to select channels 1732, the third to 3348 and so
on. Sixteen channels multiplied by 16 presets (one for each MIDI channel) gives you up to 256 channelselector buttons if you
need them.
If you have a controller with fewer controls that supports this preset/bank switching, you could use one preset for your EQ,
another for compression, another for tape saturation, and so on, or you could assign one preset to control your sendeffect
chains. Youre only really limited by your imagination.
Extras
If you have any controls left over, youll want to assign one to the channel pan control too. You dont really need to control the
Cubase mixer channel gain or fader levels if using the SSL plugin, as that has its own input and output level controls, but you
could do so if your chosen plugins lacked this. If you want to map things like Cubase channel mute and solo (I already had
keyboard shortcuts for this sort of thing, so chose to leave BCR2000 buttons free for other functions) you can find these
commands in the Value/Action column. The Read/Write enable commands might be worth setting up if youre going to use
your controller to record automation. If you really want to go to town, I suggest investigating the options in the Device list.
Quirks & Troubleshooting

When youre done, export your XML file one last time, then save your Cubase Project as a Template. Create a new project
from your Template. If you followed my instructions to the letter, everything should be working perfectly but I bet you have
missed something somewhere! Heres a quick guide to some of the niggles I encountered.
Making plugins appear in a standard place on the screen for all channels is fiddly as, while Cubase remembers the last
place the plugin GUI was placed, it does so separately for each channel. The only route I found to achieve this was to
manually drag the instance for each track to the same place (eg. topleft of the screen) for each and every channel and not
move it again. Instead, assign a BCR2000 button to the task of opening and closing the plugin window on the selected
channel. If your channel strip combines several different plugins, though, this is a pain. And its similarly annoying if you
decide during a mix that youd rather all instances appeared in a different place.
I neglected to restore my BCR2000 to its factory default settings before embarking on this adventure, and I found on
occasion that Id inadvertently set up two different controls to govern the same MIDI parameters on the same channel. That
caused some head scratching at the time! If you encounter this, its pretty easy to track down the offending controllers by
scanning the Address column in the uppermost table of the Generic Remote window for the same values.
I tracked several problems down to having deselected the toggle function when it should have been selected or vice versa. I
also neglected to set the Flags field for the controller to Transmit in the top part, to get the LED visual feedback working
properly. And, try as I might, I couldnt find a way to make the buttons LEDs light when the parameter was active. Not that I
found that to be a huge problem, but do let me know if you figure out how to do that!
Annoyingly, I found that using the full resolution of the rotary encoders meant that once Id moved a parameter away from
its default position, I couldnt return to it using the BCR2000, although I could using the mouse. Thats fine for some
parameters, but for not for all. If its a problem, the trick is to reduce the range of the control very slightly by limiting its input
value to 126 instead of 127 (in the Max Value field in the upper table in the Generic Remote window). This enables you to
return to a neat and tidy zero/unity using your knobs.
One pleasant surprise was that the two subtly differentsounding versions of the Waves SSL plugin (the EChannel and
the GChannel) both worked with the same Generic Remote setup, so I was able to use whichever I preferred on any
channel, and mix and match mono and stereo versions without having to re-map things.
Quick Mix
With the right control surface, the options are pretty much endless. Ive found that working with a standard strip like this can
save a lot of time and dithering (not in the technical sense of the word!) while mixing. And, as I invariably find that I do at least
the first pass of my mixes better when working quickly, its been well worth me spending the time Ive taken to set this up. Now
I just need to resist the temptation to build a hundred different channels and Generic Remotes! .
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The Strings The Thing
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Strung Out
Ableton Live Tips & Techniques
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Technique : Ableton Live Notes
We get plucky with Ableton Lives Tension instrument.
Len Sasso
ike Collision and Corpus, covered in the May 2014 Live column, Tension is a physical-modelling-based instrument
developed in collaboration with Applied Acoustic Systems (AAS). Based on the AAS synth String Studio, Tensions
purpose is to emulate stringed instruments while providing lots of tools to take you beyond the sounds of acoustic
instruments, be they sampled or real. Tension is included in Live Suite and available for $99 as an add-on for other versions. If
you like Tension, String Studio VS2 is worth a look for its added features and extensive preset library. Ill start with some
Tension basics and then explore how to use them to create your own sounds.
The Strings The Thing

Stringed instruments start, of course, with strings under tension; you tune each note by twisting tuning pegs to adjust the
tension on the strings. To play the instrument you pluck, strike or bow a string with an excitator, which starts the string
vibrating. The length and tension of the string determine the rate of vibration, and therefore the pitch, of the sound. For
stringed instruments such as guitar and violin you change the length (and pitch) of the vibrating part of the string by pressing it
against a fingerboard, whereas for others (stringed keyboards for example), you have one or more strings for each note. Other
important factors also affect the sound. The thickness, shape and material of the string influence both timbre and decay. How,
where and with what you start the string vibrating influence timbre as well as volume. Damping (how you stop the string from
vibrating) determines how the sound decays. The body of an acoustic instrument, or pickup design and placement, for an
electric instrument affect timbre.
Tensions interface has two panels: String and Filter/Global.
The rather crowded String panel controls each of the
aforementioned factors. The Filter/Global panel adds a multimode filter with dedicated envelope and LFO. It also houses
controls for how Tension reacts to your MIDI keyboard, along
with global settings for voice management (count, detuning,
transpose, unison and portamento). Ill start with that panel
because it provides a quick way to subtly or radically alter the
sound of factory presets always a good place to start
exploring a new instrument.
1: Tensions Filter/Global panel provides a quick path for

subtly or radically altering Tension presets. These settings
add an auto-wah effect to Tensions default preset.
Insert Tension on an empty MIDI track and take a quick look at the String panel. The default patch uses the Excitator
sections Plectrum (pick) setting and the String section in the middle of the panel controls various string parameters. The other
sections are disabled. Play a note and youll hear a decent emulation of a plucked string that decays slowly as long as the key
is held, and dies out quickly when you release the key. Lowering the String sections Ratio setting increases the release time
try 30 percent to add a little tail to the sound.
Switch to the Filter/Global panel and youll see the Filter,

Envelope and LFO controls lined up on the left with the keyboard
and global settings on the right. Only the filter is turned on, and
that doesnt have any effect because it is a low-pass filter with its
cutoff frequency all the way up. Playing around with the Freq and
Res controls or dragging the round cursor in the display works
just as you would expect from using other Live filters. For
something a bit unusual try the two formant filter types at the
bottom of the drop-down list. The Envelope and LFO are mostly
typical of Live instruments, but notice that you get separate
Vel(ocity) controls for Attack and Sustain and that there are
separate filter cutoff and resonance modulation-amount controls
for the envelope, LFO and key number. Enable these sections
and try the settings from screen 1, which uses a formant filter to
produce a kind of auto-wah effect. You can defeat the effect by
deactivating the Filter section, and thats a good target for
toggling with a footswitch or a MIDI note.
2: The settings on the String panel of Tensions Blues Guitar

preset create a sound much different from the basic pluckedstring sound of the default preset. The yellow dots indicate
the most noticeable changes.
Make It Your Own

One of the advantages physical modelling offers over sample-based instruments is that you can change the characteristics of
the instrument; youre not stuck with the sound of a specific piano or guitar or what have you. Be careful of your ears and
speakers, though, because small changes in physical-modelled parameters will sometimes push levels way over the top.
There are many ways to get to grips with complex control panels like Tensions, but one of the quickest is to compare similar
presets modifying the controls of one and auditioning the difference until all settings match. For example, insert the default
preset in one track and the Blues Guitar preset from the Guitars & Plucked folder in another track. The settings of the Blues
Guitar preset yield a quite different and more authentic guitar sound. Notably, the Damper, Pickup and Body sections are used
and the Excitator and String sections have different settings. Change the default settings one by one to the Blues Guitar
settings and youll find that some are more influential than others (see screen 2).
As another approach, load any of the individual Tension
presets (avoid the racks so you hear just what Tension is doing),
figure out whats causing the most audible features of the sound
and then try some modifications. Here are three examples:
Mallets/Big Kalimba: the Perfect Fifth interval you hear results
from the enigmatically calibrated Global Detune setting of 92
percent in the Unison section of the Filter/Global panel. Try
different Voices and Delay settings Delay is a good target
for MIDI mapping or automation. Use Detune settings of 88,
84 and 80 to dial in intervals of a Perfect Fourth, Major Third
and Minor Third respectively, but beware that there are
3: Enhance Tension presets with Live Audio Effects racks.
Here the Ambient Drops Hold bouncing-hammer preset is
several increments for each numerical value, so hold the
processed by a rack containing Grain Delays, Resonators
Command/Control key for fine tuning.
and reverb.
Bass/Slap Bass: try changing the Mass, Stiffness, Position
and Damping settings in the Excitator section to change the character of the slap.
Pianos & Keys/Mellow Wurly: try lowering the Position in the Pickup section and the Damping amount in the String section.
Also experiment with the Excitator Mass and Stiffness knobs.
Afterthoughts
Some judicious effects processing can do wonders for raw Tension patches. An obvious example is inserting Amp and
Cabinet effects after any guitar sound. Try the Blues Drive Amp preset followed by the Slight Reflections Cabinet for the Blues
Guitar preset in screen 2. Inserting Auto Pan after an electric piano for the classic Rhodes tremolo effect is another obvious
example.
Corpus, which embodies the resonator section of Collision, is particularly well suited for use with Tension. You can use it in
addition to or instead of the Body section of Tension; I usually disable Tensions Body section. For example, load Tensions
Bass/Acoustic Synth Bass preset, disable the Body section, insert Corpus after Tension and try different Corpus presets. Dont
be too concerned about the Corpus preset names for example, most of the drum-category presets work well with any
acoustic-instrument-modelled Tension sound. The presets with Loop in the name work more like special effects, and a little
goes a long way, so try MIDI mapping the mod wheel to Corpus Dry/Wet knob and using the low end of its range. Many
Corpus presets have strong resonances at some frequencies, so you may want to use Corpus Side-chain section to track the
MIDI notes playing Tension.
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Tension can produce special effects that have an acoustic flavour while being unlike any real-world acoustic instrument, and
Lives more exotic Audio Effect racks can often take those to the next level. For example, try following the Tension preset
Effects/Ambient Drops Hold with the Audio Effect rack Space/Chamber Trio Delay. Ambient Drops Hold is a mono preset that
uses the Hammer (bouncing) excitator with extreme Damping to generate a series of repeats. Thats interesting in its own
right, but a bit repetitive. Chamber Trio Delay takes care of that with a couple of Grain Delay and Resonator chains along with
chains for reverb and the dry signal. Automate or MIDI map the racks Dry/Wet macro knob to keep the effect under control.
Slowly playing staccato notes with a low Dry/Wet setting produces an interesting alternative to the bouncing hammer. Another
bouncing-hammer Tension preset, Ambient & Evolving/Moving Stringpad, becomes much more dynamic when followed by the
Audio Effect rack Space/Ambient Space Delay. Here again, modulating the racks Dry/Wet macro knob tames the effect.
.
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Studio SOS
Paul and Hugh head off to The Cave!
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Team SOS travel to deepest, darkest Cornwall, where they

encounter some troublesome low frequencies and tackle a
few common mix quandaries.
Hugh Robjohns
ocated in a small industrial unit just outside St Austell in Cornwall, The Cave studio is owned and run by Beren
Matthews and singer Louella Jade Eke, who took over the building from a previous studio operator. They subsidise the
studio by hiring out the performance area as a rehearsal space, but Berens aim is to get the control room up to the
standard that it could be used for community music projects, as well as for his own band.
When he took over the property, the control room was a simple, plasterboardonstudding construction with a large vocal
booth built into the rear corner. There was no adequate acoustic treatment in the room or in the vocal booth, so when Beren
first called us for advice we suggested he remove the booth to give him more space and then build some mineral wool traps to
hang on the walls and across the corners to try to tame the acoustics. He built some wooden frames to accommodate
mediumdensity Rockwool slabs of 30 and 50 mm thickness, and then covered them with a purple fabric. Four were placed
across the corners, but didnt fit tightly against the walls because they jammed between the ceiling and floor, leaving
significant gaps either side. The rest were hung on the walls or propped up on boxes.
Speakers Of The House

After a long drive, Hugh and I arrived with our bag of tricks, and over the ensuing coffee and oaty biscuits Beren outlined the
issues he saw as being most serious. Primarily, despite all the broadband absorber panels, the monitoring still didnt sound
right, and there was also the issue of positioning Berens Adam A7 monitors and Yamaha NS10s on the small desk to allow
both to produce decent stereo imaging. As Beren had his monitors set up, the Adams were very close together and so didnt
produce such an impressive stereo image as the NS10s. The power amp used to drive the NS10s also produced quite a lot of
physical noise, due to its cooling fans. Hugh suggested that this might be relocated to the void between the control room and
performance area a potential mini-machineroom space!
We experimented with different mounting arrangements for the
NS10s and Adam A7s, to try to improve the stereo image, but
the more we experimented the more we became aware of much
more fundamental room acoustic issues so our first job was to
listen to the room and run a few test signals to see how it was
behaving. Beren was using a subwoofer with his Adam monitors
and had already moved it to where it seemed to be giving the
most even results, but there were still some very obvious bumps
and dips in the rooms response.
The Low Down

We had intended to start with a test tone sequence run from
Hughs test CD, but it turned out that Beren had upgraded his
iMac to include an SSD drive, which had necessitated removing
the CDROM/DVD drive. He also had no Internet connection at
the studio. Undeterred, we created a semitone staircase of tones
Before the visit. The monitors were set up to fire down the
length of the room, as is usually recommended, but their
configuration on the table was suboptimal.
using the EXS24 sampler that comes with Logic Pro X, his DAW of choice. If you dont load any samples it plays a default sine
wave, and with all the notes set to the same velocity, the test quickly revealed some serious dips and peaks in the 120 to 220
Hz range though the deep bass didnt seem too badly behaved. The frequencies of the problematic standing waves
suggested issues related to the height and width of the room, rather than its length.
Our first thought was to try to move the speakers and desk
relative to the front wall, and though this caused some small
changes in the shape of the response, we got no closer to
ironing out the bumps. We also turned the sub off and ran the
Adams full range to avoid having too many issues in play at the
same time, but still we found the same problems. Berens
monitors were already set up to aim down the length of the
rectangular room, which was just under six metres, so we had to
conclude that the problems were due both to room modes and,
to some extent, resonances in the singleskin plasterboard
walls. Given the inevitably limited budget, the best solution for
the walls would be to add another layer of plasterboard fixed with
Green Glue at a later date when the studio has generated some
income. The additional layer would add mass, while the Green
Glue would introduce some muchneeded damping.
Its A Trap!
Supervillain Paul White scrutinises different monitor

arrangements while listening to a sinewave test set up in
Logic Pro X.
However, we needed a more immediate solution, and after some head scratching we decided to try to improve the
performance and effectiveness of the corner traps and existing panel traps. We started by placing the thicker panel traps at
the mirror points, and spaced them a couple of inches from the walls to improve their lowfrequency performance significantly.
We achieved this using some of the timber left over from demolishing the vocal booth, screwing a piece of 50 x 100 mm
studding behind the top of each frame with a smaller block cut from the same material at the bottom. A trip to the local DIY
store produced the necessary screws and plasterboard fixings. The use of such substantial pieces of wood was not only
because they were conveniently to hand we also hoped they would help to brace and dampen the relatively flimsy
plasterboard walls, too, reducing their propensity to resonate. After fixing the mirrorpoint panels we used the same technique
to mount a number of thinner panels further back on the side walls, and on the rear wall above a sofa.
Turning to the corner bass traps, we had our doubts as to how effective they would be as they were simply thin Rockwool
floortoceiling panels wedged across the corners, leaving 100 to 150 mm gaps at each side. Rather than try to cut them
down slightly so that they could be fitted tightly across the corners, which would have been quite a tricky and messy job, we
asked Beren if he had any leftover mineral wool. He only had a few scraps but he did have a lot of insulation-grade fibreglass
that he didnt need, so we suggested he use it to fill as many plastic rubbish sacks as possible with a view to cramming them
behind the existing corner trap panels. It wasnt the prettiest of solutions, but we had enough to fill the front two corners and it
turned out to be surprisingly effective. There is no need to use breathable sacks in this application as low frequencies have
no problem getting through thin plastic bags, and the loft insulation absorbs LF energy very well.
Pre Amp Connections
Overdubbing midi in
Studio One
help please - portable
audio stereo recorders
with XLR
Reinterpreting or
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midi notes
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Piano and
Clarinet/clarinet quartet
The final wall that we needed to treat was the one in front of
the monitors, and Vicoustic had kindly provided a box of eight of
their Cinema Round acoustic panels, each comprising a curved
solid foam block covered in an attractive black velour-style fabric.
We fixed these using spray adhesive in two rows of five and
three (the smaller, lower row being necessary to accommodate a
mains outlet), and they looked great. After treatment the room
certainly sounded a little less live then before, but how about
those bass problems?
On running the steptone tests again we found that the
problems were much diminished, with only a couple of small and
broad dips, and a peak of 5 or 6 dB at 200Hz still in evidence. An
EQ cut of around 5dB at the problem frequency gave us a much
smoother and totally workable result, so after fine-tuning the EQ
plugin settings, we saved it as a preset to be used in Logics
output bus when mixing but with a reminder to bypass it when
bouncing.
DAW Tips from SOS
Its not what it looks like! Hugh Robjohns doing nothing at

all untoward to Beren Matthews subwoofer.
The front wall and mirror point absorbers improved the imaging precision, but if the two sets of speakers were positioned
beside one another the outer set would always sound more spacious than the inner set. To get around this we put the Adam
speakers at the desks edges to keep them well apart, and then used a couple of speaker stands that Beren had lying around
to support the NS10s so that they were directly behind and above the Adams. Switching between the two sets of monitors
now resulted in a consistent stereo image, making it much easier to make reliable balance judgements. Hugh made some final
tweaks to the sub level setting and by this time Beren had managed to borrow a USB CDROM drive so that we could run
Hughs test songs and tone steps to confirm that all was well. It surprised us how much tighter and more even the sound was,
with good stereo imaging from both sets of speakers. We suggested that Beren should get hold of some more insulation to
treat the rear corners you can never have too much bass trapping but for now the room was behaving well enough to
work in.
Mix Fixes
We then turned our attention to some of Berens mixes in progress, which already sounded pretty good, though perhaps a little
lacking at the low end and lower mid-range, possibly because the inaccurate monitoring had lead him to cut the lows more
than necessary. Hed also got into the habit of using a lot of plugins on each source, often in combination, so we tried to
come up with some alternatives that would put less of a load on the CPU of his iMac as well as avoiding the phasiness that
can occur when several EQ plugins are cascaded.
The kick drum in one of Berens mixes lacked any real weight,
so we tried Logics subbass generator combined with EQ to cut
the extreme lows below 40Hz but to add boost in the 90Hz
region, followed by more cut at 150 to 180 Hz to avoid boxiness.
We also demonstrated how an Excitertype plugin could
accentuate the beater click of the kick drum, as well as being
useful for adding life to dull snare drums or overhead tracks
where the cymbals arent cutting through.
Next we tweaked the bass guitar track to give it more punch,
and the trick Beren liked the most was using Logics basic Limiter
plugin with the softknee box ticked to add attitude to both
vocals and drums. It makes a useful alternative to a compressor
when you need something a little more hardhitting. For the
electric guitars, we demonstrated the tonal variety that could be
achieved by reamping the parts via a plugin amp and speaker
emulation, as an alternative to using EQ and compression.
Driving Theory
The final configuration. The NS10s were placed on stands

behind the Adam A7s, which allowed both sets of monitors to
present a good stereo image. Along the wall on the back can
be seen the Vicoustic Cinema Round panels, and hiding
behind the corner traps are sacks of insulation, to help
bolster the bass traps performance.
While we were at it we also showed how a simple overdrive plugin can be used to add a tubelike warmth to sounds such as
bass, drums or even vocals, as long as you keep the amount of drive fairly low so as not to give the game away. Next came
some of our vocal processing tricks. Beren had used various plugins to brighten the vocal sound, but as an alternative we
simply used some shelving EQ lift at around 7kHz combined with 80Hz low cut and the aforementioned limiter. We then went
on to demo some of our double-tracking tricks, specifically using reverb early reflections (with no reverb tail) delayed by 50 to
90 ms to create a diffused slapback sound.
Beren also explained that hed been having some problems when using a compressor for level ducking, as the amount of
gain reduction varies depending on the level of the sidechain input. I showed him an alternative method using Logics gate
instead of a compressor, where setting the gain reduction slider to actually increase the gain by a few dBs when the gate is
closed allows a precise amount of gain reduction to be set. Essentially the signal is always boosted by the amount you set on
the slider, except when a sidechain signal is detected, at which point the gate opens and the gain falls back to unity. The
attack and release controls vary how quickly the ducking comes in and out.

Cubase
Digital Performer
Live
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Pro Tools
Reaper
Reason
Sonar
Easy As ADT
Beren had also bought the Waves Real ADT plugin, which is actually really good, but again we showed him one of our
favourite alternatives that requires a copy of the original track to be delayed and then subjected to pitch correction using
something like Autotune or one of its many imitators in this case Logics own Pitch Correct plugin. When played back
alongside the original part, the pitch-correction plugin provides the necessary pitch variation, while the delay produces the
required doubling effect. I know a lot of engineers use a pitch shifter plugin set to detune by a few cents to do a similar thing,
but the advantage of a pitch-correction plugin is that its always adjusting the amount of pitch shift it applies, so the result has
a much more random and lifelike quality to it. The slower the pitch-correction speed is set, the more subtle the result.
As Beren was using an iMac with no additional monitor screen, we also went over the benefits of using screen sets, which
he was quick to appreciate. If these are created and saved in a Template song, theyll always be available for future projects.
The plan was to leave at a sensible time to avoid the motorway traffic chaos on a Friday afternoon, which we spectacularly
failed to do, but Beren did let me have the wood off-cuts from our endeavours to use in my stove at home, which was much
appreciated!
.
Reader Reaction
Beren Matthews: In a world of faceless Internet forum misinformation, Hugh and Pauls wealth of realworld knowledge,
plus their hands-on approach, was exactly what was needed. A priority for me was ensuring that the mixes for my bands
(GripLike Vice) new EP werent compromised by inaccurate monitoring.
The improvements in monitor position, coupled with the tightening of the room response, mean I can be confident that
what Im hearing is significantly closer to the truth. Additionally, Pauls extracurricular plugin-related titbits provided some
interesting ways of using the oftenoverlooked stock Logic processors. The Gate and Limiter have already proved scarily
useful for slightly unorthodox tasks.
In short, Hugh and Pauls knowledge and expertise is
something that cant be substituted by scouring the depths
of cyberspace. Their help has been invaluable, and for me
to be able to have confidence in my workspace is
priceless.
Baren Matthews enjoying his studios improved acoustics.

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Fri 19 Dec 2014
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In this article:
Clean Bandit
Oliver Heldens
Meghan Trainor
Barbra Streisand
The Vamps
Classic Mix Bill Haley
& His Comets (Were
Gonna) Rock AroundThe
Clock (1954)
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The Mix Review

Commercial Productions Analysed
Mike Senior
Buy PDF
Clean Bandit
Rather Be
A lot of mixes in dance-music genres trade heavily on thick synths, heavy effects and copious distortion to create full-sounding
textures, but at the expense of mix clarity the quality and detail of individual parts are simply obscured behind a layer of
mush. Not so this tune, however, in which mix engineers Jack Patterson & Wez Clarke have managed to keep everything
crystal clear, allowing those lovely string recordings to make a real statement despite their comparatively smooth highfrequency timbre. This is partly because of the restrained upper percussion levels, but I think its also because the mixers
appear to have resisted the temptation to use heavy synth pads to support the choruses, a move which could easily have
submerged those string players under the sonic equivalent of porridge. As such, Id recommend this tune as a mix reference
for any aspiring EDM producer who wants their audience to fully appreciate arrangement intricacies. (That said, I cant help
feeling that the string levels have been slightly overcooked in the second pre-chorus at 2:08-2:23, where they become a bit
distracting from the vocal. Then again, you could argue that the strings make a better hook than the vocal at that point...)
The keyboard part that characterises the opening of this track
(from 0:16) is also unusual in many respects. For a start, it
sounds like its being passed through some kind of vibrato
processor (or perhaps a tape wow/flutter emulator), which is
modulating the pitch so much that the tuning actually begins to
become rather vague. Id guess that its modulating almost a
semitone either way at a rate of about 2-3Hz. Although Clean
Bandit might have done this with a plug-in, I suspect its being
created by some kind of modulation routing within the
synth/sampler patch itself, given that it seems only to kick in after
the onset of each note in other words, the note attacks all
sound in tune, even though the sustain tails dont. Then when the
beat comes in at 0:48, you get what sounds like super-fast note
retriggering giving a cool rhythmic ripping effect, made all the
more effective because it contrasts so strongly with the more
traditional grand piano patch which arrives with the chorus at
1:04. Mike Senior
Oliver Heldens
Gecko (Overdrive)
The hollow sound of this particular bass line is definitely

reminiscent of a square wave to me, and if you look at the first 10
seconds of this track on a spectrum analyser you can see why:
its mostly just the fundamental, third harmonic, and fifth
harmonic which register on the display a characteristic
spectral signature of that wave shape. The sound is almost
certainly being low-pass filtered in some way, though. An
unadulterated square wave is a pretty buzzy-sounding thing,
whereas what we have here is round and mellow, with precious
little energy to speak of above 350Hz. In addition, a high-pass
filter sweep kicks in briefly towards the end of the introduction
(0:12-0:17) to create a transition into the verse vocal entry.
And when we get to 1:18, is the bass deliberately imitating a
heartbeat to support the feel the bass in my heart lyric at 1:24?
Id like to think its intentional, but if so I reckon they might have
missed a trick in keeping it steady throughout the section,
because speeding it up would surely have assisted subliminally
with building excitement into the following chorus at 1:49. One of
the oldest tricks in horror-movie Foley, that... Mike Senior
Meghan Trainor
All About That Bass
I dont think Ive heard such an enjoyable combination of pop

song and social message since Ms Dynamites It Takes More,
but thats certainly not the only thing to admire here. The vocal
lines extended use of faux triplets (ie. a pattern of two dotted
eighths followed by a straight eighth) across the four-four beat
during the pre-chorus is great, for instance, and there are plenty
of ear-catching vocal stylings besides: the stutter edit on bass
at 0:13; the striking drop in pitch for size two at 0:18; and the
pitch scoop at booty back (1:27), which to my ears draws a
rather ironic parallel with the title hook of Robin Thickes fratboy
smash Blurred Lines.
Pre Amp Connections
Overdubbing midi in
Studio One
Capturing drum
The drum programming is worthy of examination as well.

Although a bit of phase cancellation between different sections of
the arrangement confirms that the main part is a loop, its more
engaging than most stock loops on account of the messy little
ratchet-like sounds that lurk around the main hits and somehow
help glue them together with the various clap sounds overlaid on
top. Its as well to notice that there are no cymbals at all in this
arrangement, which is pretty uncommon in chart productions, but
leaves plenty of room for such HF details, as well as for vocal
nuances. The drums are mixed pretty starkly dry in this mix, in contrast with the vocals, which are clearly flattered with an
audible reverb effect. Although the danger with a tactic like this is that the vocals can begin to feel too distant if youre not
careful, the slightly retro reverb timbre and (judging by the stereo Sides signal) touch of effect predelay seem to have avoided
that in this case. Mike Senior
Barbra Streisand
It Had To Be You
Whatever you think of Barbra Streisands music, youve got to

admire her professional longevity, because shes apparently
topped the Billboard album charts at least once every decade
since the 60s her latest number one being her 34th studio
record, the all-duets album Partners. In the absence of an official
single release, Id like to focus on It Had To Be You, because ,
The Vamps, it intrigues me how Streisands voice relates both to
the backing arrangement and to the voice of guest vocalist
Michael Bubl.
The first thing that strikes me is how powerful the fundamental
frequency of Streisands voice often is by comparison with any
more characterful timbral energy further up the spectrum. Check
out 0:23-0:41 (making me glad... up to you to explain), for
instance, and in particular the word blue, where this bottomheavy timbre combines with an unnaturally static pitch-centre
(need I say more?) to generate a creditable impersonation of a
350Hz test tone! Then, when Bubls first phrases zing us with
pop-friendly 4kHz and 7kHz spectral peaks, I cant help feeling
that the two voices would have sat together a whole lot better
had a couple of decibels been dipped out of his vocal channel at both those frequencies. Not that he has no strong
fundamental frequencies, its just that theyre rather more sporadic (eg. the big 175Hz peak on seen at 2:16) and mostly
restricted to those few occasions where hes singing softly.
Sure, the two vocal tones fit together quite snugly when both leads are singing homophonic harmonies for a couple of
phrases from 2:36, but the moment they go their separate ways again at 2:46 they both feel oddly incomplete to me:
Streisands timbre feels rather bloated, Bubls thin and strident. Furthermore, when the horn section starts giving it the full
razzle-dazzle at 2:58, Bubls vocal tone cuts through fairly easily, but Streisands really struggles her high frequencies are
so seriously masked that it sounds like shes singing through a blanket a lot of the time (on with all your faults at 3:27,
particularly). Given how loud she is in the mix, the low-mid-range baggage shes carrying around also makes the rest of the
arrangement sound rather gutless by comparison.
For me, this is a textbook case of somewhere Id have wanted to mult the vocals out to different channels, so that
independent EQs could have been applied to Streisands voice for the solo and harmony sections, as well as to combat the
masking effects of the full-force band texture. Dont just take my word for it, of course have a listen for yourself! With a bit of
EQ automation, you can even experiment with changing Bubls high-mid-range vocal tone by processing the full mix. Mike
Senior
The Vamps
Somebody To You
DAW Tips from SOS

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This song exemplifies nicely one of the simplest pop production

tricks you can use to make your choruses sound bigger than your
verses: widening the stereo spread. So, in this case, each of the
three chorus sections (0:39-1:01, 1:37-1:59 and 2:10-2:54) is
preceded by a section with a narrower stereo picture:
respectively, the first verse and pre-chorus; the second verse and
pre-chorus; and the middle section. Whats a little more
interesting, though, is that the last of these is quite a bit narrower,
and I can see a good justification for this. The producers already
chose to fully showcase the stereo width in the first chorus and,
like so many things at mixdown, theres only so far you can push
stereo width before you start getting problematic side-effects
(such as mono-compatibility problems). Therefore, the only way
to increase the impact of the third chorus in terms of stereo
spread is to increase the contrast between that chorus and the
section which precedes it.
Theres also rather a good example of a straightforward
feedback delay effect being used to increase sustain on the
yeah you! sampled hook. You can hear it most clearly right at
the end of the song, but the delays there throughout, subtly
thickening and extending the massed-vocals texture. Notice also that the delay repeats are tempo-related, which helps them
sink into the mix, and they also retain the stereo positioning of the dry sound. You have to be careful with some plug-ins that
you dont end up with mono delays on stereo input signals, because that can undesirably clutter the centre of your stereo
image. Mike Senior
.
Classic Mix Bill Haley & His Comets (Were Gonna) Rock AroundThe Clock
(1954)
Whatever mic Bill Haley was using here, it sounds like he was all but eating it at times, because you can hear serious
wind-blasts at various points throughout this song. (Im kind of assuming it wasnt a ribbon!) Although you normally
associate wind blasts with plosive sounds such as p and b, this track provides a good demonstration of how many other
consonants can cause similar problems, because you get big low-frequency thuds appearing on three (0:01), four
(0:02), and twelve (0:07) in this case. To put it in perspective, these blasts produce significantly more low end than the
kick drum and bass combined, as you can clearly hear on three at 0:28. Maybe the engineers at Deccas Pythian Temple
studios didnt think anyone would hear it through 50s playback technology. Or maybe they couldnt hear it through their
50s monitor loudspeaker...
Other things to listen for here are the live-to-tape fader ride of the electric guitar into the solo at 0:40, arguably two bars
too early, and the extremely roomy drum sound, most of which were presumably hearing via the instruments spill on other
mics within the large Pythian Temple live room. What
puzzles me is whats producing that super-upfront clicking
sound. I cant imagine its any part of the main kit, because
if it was wed have more dry sound from the rest of the
drums, but theres only one drummer credited on the
session. It cant be anything the lead guitars doing,
because I cant believe hed have been able to keep it going
during his solo. Could it be string slaps from the bass
player? Seems unlikely to me because the bass notes
seem too sustained for this to be possible. Plus this sounds
missing from every live performance Ive seen. Could it be
the great uncredited washboard performance of rock and
roll history? Mike Senior
Fri 19 Dec 2014
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Out With The New
The Unbridgeable Gap
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Notes From The Deadline

TV Music From The Inside
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Its hard to make something new. Its even harder to make

something thats both new and old.

Paul Farrer
n February of this year, Google Maps passes its first decade of being online. And yet during a recent parliamentary debate
about the use of drones, a 76-year-old member of the House Of Lords Digital Skills Committee reported how horrified she
was to discover a few days earlier that on a web page called Google map or something she could find an image of her
own house.
Out With The New

We all know the pace of technological change is dizzying, so much so that even those of us whose careers depend on it find it
hard to keep up. Or, more accurately, we often lack the fight to keep on keeping up. Im now about four versions of Mac OS X
behind where I probably should be, but if you run a computer-based studio system with lots of plug-ins and virtual instruments,
you view each OS and DAW stride forward as a sort of perverse hostage negotiation. On one side, youre tempted by all the
additional features the upgrade offers (even though theyre nearly always shallow and largely aesthetic), but on the other,
youre brought back to earth by the terrifying bump of wondering if its going to permanently disable vast swathes of your sonic
toolkit.
I also wonder if the reason Im feeling so upjaded (an adjective Ive just
invented) by the whole thing is that I worry we may have reached a depressing
peak. For example, I currently own about six big orchestral string libraries
and, as lovely as they all are, I doubt I use more than a handful of the same
string sounds in 99 percent of my music. Who among us can say they have
properly auditioned every single preset in Omnisphere?
Libraries are often so big these days they are supplied on their own hard
drives, which often require powerful computers of their own just to be quick
enough to access all the different sound options. I love technology and what it
does for us, but I could learn to play a cello in the time it would take to properly
go through 500GB of cello samples.
The Unbridgeable Gap

Perhaps the proliferation of retro tech, in the form of new versions of classic
synths, drum machines, and vintage outboard equipment brought back into
service, speaks of an industry that finds the past more comforting than it finds
the future exciting. Or perhaps, in the same way that broadcasters love bringing
old TV formats back to life, it might suggest that we have simply run out of
marketable new ideas. I hope not. One of the hardest things to do is revisit the
past and make it relevant for the present something which will be well-known
to anyone who has ever been asked to rework the theme tune to an old TV
show. Which elements to keep, which to lose and which to enhance?
I told him hed struggle to upgrade to Logic

X using punch cards.
I was asked by a channel recently to revisit a well-known theme tune for a show that ran for nearly 18 years back in the
1950s and 60s. It was a soaring and elegant theme, written for a dour post-war Britain when Princess Margaret was foxy and
people still knew how to cook things with powdered egg. And the brief was almost impossible: to keep the spirit of the existing,
yet make it relate to modern audiences. I felt a bit like a tailor trying to make a pair of studded leather trousers using the floral
pattern material from your grannys chintz curtains.
The first Apple Macintosh computer had a monochrome 175,000-pixel screen. Thirty years later, the current iMac has a 5k
display with 14.7 million pixels, yet costs just four dollars more than the Mac did in 1984. The race forward is inevitable,
wonderful and cheap, and we should embrace it even when it comes at a higher price.
.
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In this article:
Show Me The Money
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Off The Record

Music & Recording Industry News
Music Business : Music Publishing
Despite a recent downturn, the videogame industry still

has much to offer to audio pros.

Dan Daley
ike the music industry, which it tends to emulate, the videogame business has hit a wall. At midyear in 2014, software
sales for the previous generation of consoles declined 57 percent from the same time last year, resulting in a $47 million
overall decrease year over year, according to the most recent report from market researcher NPD Group.
Nonetheless, its still a potentially valuable space for those who produce its audio. That bit of good news comes from a
survey conducted this Autumn by GameSoundCon, which produces an annual conference for videogame music and sound
design. Brian Schmidt, GameSoundCons executive director, wanted a more accurate picture of what the audio domain really
looked like. He noted a survey released earlier in the year by Gamasutra, the UKbased trade site for the industry, which
seemed to indicate that audio professionals in that business earned an average salary of over $95,000 more than any other
personnel category apart from company executives. Was it too good to be true?
Actually, it was, but not by that much. Gamasutras audio survey sample size was quite small, with only 33 respondents who
listed audio professional as their field. Secondly, it was limited to information from fulltime employees, not the freelancers
that make up the majority of the industry. The sample was also limited to Gamasutras own conference attendees list.
GameSoundCon then undertook a more comprehensive study, with input from over 500 audio professionals in the game
industry, drawn from the rolls of the Game Audio Network Guild (GANG). The questions were more comprehensive, too,
focusing not only on compensation but also on work and environment, royalties (or the lack thereof), the use of live musicians
and middleware, and contract terms within the categories of largebudget games, casual games (smaller budget but
professionally produced), and independent games (selffinanced, including Kickstarter projects). The outcome reveals a
business well suited to the ambitious composer who is capable of doing his (or her more on that later) own production,
including sound effects and dialogue.
Show Me The Money

A salaried audio employee in the games industry makes $70,532 per year on average, whereas for freelancers, a single
project in the largebudget category averages $76,822; thats 20 percent less than Gamasutras finding, but still eminently
respectable. The project fee for indie/casual games averages $9830 (only 12 percent reported that they were paid by the
hour; most compensation comes as a flat fee). The survey also found that the mean number of games that freelancers worked
on per year was 4.6. Thats over $45,000 PA.
Its worth noting that salaries and fees are the sole compensation for the vast majority less than two percent of
respondents that worked on bigbudget projects reported perunit royalties. For casual or indie games, this number is a still
slim 17 percent. In some cases, composers were able to retain soundtrack rights, but that was also rare.
The survey also concluded that game music and sound is predominantly a freelance gig (60 percent); that most composers,
particularly for smaller games, also deliver sound effects and/or do integration or other technical work; that even in large
budget games, most (54 percent) of the music is produced virtually as opposed to predominantly by live musicians; and that
women are underrepresented in the industry, at less than five percent. Sound Familiar?
Both the video-game business and the music industry have been declining in sales and revenue, and increasingly rely on
blockbusters for profits. Call Of Duty: Black Ops, which took in over $650 million in the first five days, was that industrys
version of Taylor Swifts 1989. Both industries are also going through their own distribution transitions: as music wrestles with
extracting money from streaming, games are migrating from consoles to handheld devices, with free and freemium models
proliferating. That will continue to challenge revenues in the future, but it still offers substantial opportunities for employment
for audio producers. And overall the industry remains huge: international videogame revenue is estimated to be $81.5 billion
in 2014 (more than double that of the international film industry in 2013), with $22.2 billion of that coming from North America.
One big finding of the survey is the fact that, among all respondents who delivered music, the overwhelming majority of the
music was created by the composer alone. Game audio is primarily a onemanband business, perhaps the ultimate
application of the projectstudio environment, with 91 percent of indie/casual game audio producers working solo. Its also a
onegigatatime landscape, where dozens may bid on the same game and success comes from a combination of talent,
tenacity, perspicacity and, as with everything else to do with music, lots of dumb luck. As the surveys synopsis sums it up,
making an income means working on many smaller projects until your reputation, skills and experience generate demand
from largebudget companies. The good news is: a composer or sound designer can make a decent or even a considerable
income working in games, large or small.
Then theres the issue of how few women are involved in the gaming business. Not just on the audio side, where the survey
found that they represent only about four percent of that category, but in the industry as a whole, where they account for only
11 percent of game designers and three percent of programmers. In addition, as of 2013, females in the industry receive an
average of 27 percent less income than their male counterparts. Its worth noting here because organisations like the
Womens Audio Mission in the US, and Sound Women in the UK, have raised the number and visibility of women in pro audio,
and may be able to do the same in the games universe.
It may not be all fun and games out there, but in an era of niches, the audio side of gaming has a lot of potential for those
with the right combination of skills. .
Fri 19 Dec 2014
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Studio File
RMV Studio, Stockholm
Music Business : Music Publishing
Hannes Bieger
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wedens newest high-end studio is built on very strong foundations, both literally and
metaphorically. The building stands on the solid bedrock that underlies the city of Stockholm, and the people behind
RMV (which is short for Riksmixningsverket) unite several strands of the countrys rich pop history. With RMV, they are
setting out to add yet another chapter to a Swedish success story.
RMV is unusual in several respects, not least its location. The island of Skeppsholmen, only a couple of hundred metres
across, lies right in the centre of the Swedish capital, just a stones throw away from the kings palace. Skeppsholmen is home
to a number of cultural facilities, including a theatre, several museums, an art school and two brick buildings of, in
comparison, much smaller proportions. One of the latter has been the headquarters of ABBAs Benny Andersson for many
years. From this spectacular location, Andersson has run his record company Mono Music and operated his own writing
studio and a dedicated mixing room manned by his long-term engineer Bernard Lhr.
Bomb Factory
When Andersson made plans to build another studio more suitable for recording, he naturally took a look at the second brick
barn next door and after some negotiations with its owner, the Swedish government, RMV was built. Most of the
construction was overseen by Linn Fijal, who graduated from recording and engineering school not long before, and who now
runs RMV both as the studio manager and in-house engineer.
When the team took over the building it was in a very raw condition, with bare brick walls and a dirt floor. Skeppsholmen
once served as a naval base, and this late 19th Century building is believed to have been conceived as an ammunition dump.
The extremely solid architecture supports this theory, as do the number of cannonballs which were found during studio
construction. When planning the studio, great care was taken to maintain the light and open atmosphere of the building, and
with its many doors and windows on ground level, RMV is one of the brightest and friendliest recording venues in the world.
Another unusually large window separates the buildings two largest areas, the control and live rooms, which are
accompanied by a number of smaller booths and a kitchen and office space. The studio was conceived with the idea that most
of the acoustic treatment and technical aspects should be as invisible as possible. All rooms are built on floating floors, and all
of the wooden ceiling panels are mounted at different angles a brilliant idea, yet one which is easy to overlook at first
glance.
Another subtle feature that might not be immediately noticeable is that the colour scheme of the buildings interior is derived
from the bright blue and grey of the vintage 64-channel Neve 8068 console that is its centrepiece. This desk was relocated
from Max Martins Maratone Studios just a few kilometres away, another significant hotspot of the Swedish pop scene.
Original Neve employee Fred Hill was flown in to restore the console to its original beauty a massive task that kept half a
dozen people busy for over a month. However, it proved worth the trouble: Its such a nice desk, its magic! says Linn Fijal.
The maiden session at RMV took place in early 2011, with

Benny Anderssons own 16-piece ensemble, who were followed
shortly after by the Hives. Since then, a lot of jazz and other
mainly acoustic music has been committed to hard drive (and
tape!) at RMV, and the studio has also hosted many string
sessions for both film soundtracks and pop songs, with the string
parts for Loreens Euphoria, which won the Eurovision Song
Contest in 2012, a good example of the latter.
Polar Patterns
Today RMV keeps an equipment arsenal which makes the studio
suitable for a wide range of gigs, with the focus laid a little more
on tracking than on the mixing side of things. The outboard
RMVs Neve desk was relocated only a few kilometres from
collection was hand-picked for quality and versatility, and with
Max Martins studio.
classics such as Pultec and Massenburg EQs and several pairs
of Urei and Universal Audio dynamic processors, RMV covers
the common ground found in nearly all high-class recording facilities around the world. Yet what makes the venue really stand
out, besides the rooms themselves, is the collection of instruments many of which have been part of Benny Anderssons
private collection for decades. There is ABBAs original white-painted stage Minimoog, and also an ultra-rare Yamaha GX1
synthesizer. Only a dozen examples of this predecessor to the legendary CS80 are known to exist outside Japan, and this
particular unit has not only been played on ABBA productions, but was also used by John Paul Jones when Led Zeppelin
recorded their last album at ABBAs Polar Studios. Likewise, some of the microphones also came from ABBAs legendary
recording location, and today the historic Neumann U67 and U47s are accompanied by some of Swedens finest new designs,
such as a pair of Manibus de Geer large-diaphragm mics based around modified AKG CK12 capsules, which impress not only
with their sheer size, but also with their steampunk appearance. RMV also offers a well maintained Steinway D grand piano
from 1959, a Mellotron, and Benny Anderssons own Synclavier,

which he still uses for writing, as well as an assortment of vintage
Ludwig, Slingerland and Zildijan drums and cymbals.
With other facilities such as the TAD Reference One, ATC
SCM300, Genelec 8260A and Yamaha NS10 monitors, and a 24channel CLASP system for tape recording, and most importantly,
with a unique vibe and location, RMV has proved an ideal
recording venue for ensembles of up to 50 musicians.
.
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The old but still powerful New England Digital Synclavier

remains one of Benny Anderssons preferred writing tools.
This original Mellotron is studio manager

and in-house engineer Linn Fijals
favourite thing in the studio.
Fri 1 May 2015
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Q Are mineral-wool acoustic panels safe?

Sound Advice : Recording
Im looking to get some acoustic panels of the rigid fibreglass/Rockwool type for my bedroom
studio. Ive read a few things online about possible dangers in particular about respiratory issues
(though I think carcinogenic impact was proven negative). Now, I know you wouldnt use them if you thought there were a
danger, but theres a lot of guff online about potential issues. As my daughter has slight respiratory issues already, I definitely
dont want to make things worse. Can you offer me any advice on this?
SOS Forum post
SOS Technical Editor Hugh Robjohns replies: Theyre not
inherently carcinogenic, as far as I am aware, but loose fibres
can certainly cause irritation. For that reason, mineral wool
should always be covered with a breathable but tightly woven
fabric that will prevent the release of fibres. If youre making DIY
panels, then spraying the mineral-wool slabs with a diluted PVA
glue helps to keep fibre shedding down, and make sure you
wear a mask while handling the stuff.
Commercial mineral-wool-based panels can smell unpleasant
at first, due to the glues used, so I always unwrap them and
leave in the garage for a week or so, to let the fishy glue smell
dissipate before installation! .
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We often advocate the use of mineral wool for use in DIY

acoustic treatment its safe to use provided you take
sensible precautions.
Fri 19 Dec 2014
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Q How do I tackle the unwanted noise coming from

analogue modelling plug-ins?
Sound Advice : Recording
I like to use a lot of analogue modelling plug-ins when Im mixing, but while I like the sounds Im
getting in general, I always seem to end up with too much noise and its driving me crazy! Whats
the best way to tackle this problem?
Daniel Jones via email
SOS Reviews Editor Matt Houghton replies: The noise could be coming from more than one place, and could be being
amplified by more than one processor, so the first task is to find out where the noise is actually emanating from. Step one is to
turn off the noise generators! These seem to be on by default in too many modelling plug-ins. (Really, who wants them to
sound that authentic? Theyll be making them break down during sessions and charging a virtual repair fee next!) Step two is
to go back and look at the sources youre processing, to see if theres any low-level noise there thats being amplified by the
compression make-up gain thats going on in a lot of these plug-ins. Im not just talking about compressors and limiters, here
anything that features some sort of saturation or distortion will reduce the dynamic range, and when gains applied that will
bring up the noise along with everything else.
Then its time to consider how to tackle the remaining noise. If noise
is prominent while the wanted signal is playing, consider using a
dedicated noise-removal tool like iZotope RX, Waves X-Noise and so
on. These can be highly effective, but theyll sometimes leave unwanted
artifacts. If the noise only bothers you between sections of wanted
sound, then level automation on the individual sources is an obvious
solution if the noise isnt there, it cant be amplified by any plug-in.
Gates do this automatically, of course, but they just dont sound right
to me unless they have a variable floor control, or whatever you prefer
to call it (the bundled one in Cubase, for example, doesnt have this
feature), as the abrupt cutting off of the noise just serves to draw
attention to the fact it was there in the first place. Id rather have noise
right the way through than hear that! But even then, level automation is
a more precise option.
A more natural-sounding technique than a gate is to automate the
frequency of a low-pass filter so that the filter rolls down the spectrum in
sections between the wanted bits of sound. The sound remains, but it is
less noticeable, and the transition between sections is less glaring too.
This is how the old Symetrix noise gates worked, and where
sophisticated noise-removal tools such as iZotope RX arent called for,
or arent working for you for whatever reason, its a useful technique to
try on hissy guitar sounds; Ill bet it will work for you too.
A common cause of unwanted noise is analoguemodelling plug-ins, which are often too authentic! This
UAD Studer model, for example, features a noise
control which, by default, is hidden beneath a panel.
Several Waves plug-ins also have noise switched on
by default, and in some mixes compression and
limiting further down the signal chain can raise this to
annoying levels.
I havent yet found a plug-in that does this automatically for you, but
you can do this in Cockos Reaper using its dynamic automation system
(its called Parameter Modulation; see http://sosm.ag/reaperparametermod for details). This can be set up to make the filter frequency move dynamically in relation to the amplitude of the
source signal so as the vocal phrase finishes, the filter rolls off the more noticeable high-frequency hiss. It takes a bit of
finessing to get it right, but if youre already using Reaper, it could be just the ticket! .
Fri 1 May 2015
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Q Should I buy my own digital mixer for a band tour?

Sound Advice : Mixing

Im the sound engineer with a group called Flyte, who are heading out on a mini headline tour of
150-500 capacity venues next month. I have the chance to acquire a little digital mixer such as a
Behringer X32 rack, but Im not sure if it would be worthwhile. Obviously Im trying to weigh all the factors, like having a
lovingly pre-sculpted FOH and in-ears mix to start every day with, versus the ball-ache of crow-barring the mixer into tiny
booths in the face of reluctant in-house guys. Any advice on this would be gratefully received.
Hugh Fielding via email
SOS contributor & the Prodigys live-sound engineer Jon Burton replies: In an ideal touring situation youd carry the same
equipment to every venue to ensure familiarity and thus speedy operation. If you have a mixer you know well, and which is
already configured with the basic settings you need to start the day, youll save a lot of the time youd otherwise take setting
up a house desk, configuring the monitors and generally wiring it all up. That frees up more time to consider the variables in
each venue the house sound system, the amps and speakers, the unique acoustics. Dont underestimate that, as getting
the house system sounding good, and everything working correctly, can often take up a lot time.
However just turning up with your desk is only a small part of the answer you must also consider the infrastructure that
supports that desk. Having your own gains set in the mixer is only any use if you are using the same sources, for example,
and this means not only the instruments, but also the various microphones and DI boxes that feed their sound into your desk.
You can buy a reasonable 32-channel desk for very little, but
buying all the microphones to go with it may cost you the same
figure again and more.
The next stumbling block is carting all those mics around with
you on tour, and then, of course, you have to have a means
connecting them (placed somewhere on the stage) to your desk,
which might be quite some distance away. There will be an inhouse multicore which will go from the stage to the house desk,
but using this is fraught with difficulties. Will there be enough
channels in each and every venue? If so, do they all work? How
is the cable terminated at the desk end? Will the leads be long
enough to go to your desk? Is it a digital or analogue cable run?
And so on...
With sophisticated digital desks becoming ever more

affordable, it seems tempting for the small touring band to
invest. It may be the right solution, but theres an awful lot
more you need to think about than acquiring the desk!
As you point out, you might also run into space issues. With
smaller venues, the mixer tends to be shoe-horned into a very small area, probably leaving very little space for your desk. Will
you be able (or indeed allowed) to move the house desk? If you do move the house desk, who will mix the support band, and
on what? If you move the multicore to your desk will it have to be patched back for the support act and if so, who will do the
re-patching of the 20-plus channels youre using in the 15 minutes between bands when you are needed on stage to
reposition microphones? Its a lot to ask of anyone, and introduces unnecessary risks.
For all these reasons, if using your own desk, its usually best to also take your own multicore and stage box, so that you
can be completely independent of the house system. This way you can just give the house engineer a left/right mix that he
can run into his desk and into the house system. Great! Except now you need to spend money on a multicore and stage box.
An analogue one will be big and heavy, but relatively inexpensive, whereas a digital one will be small and lightweight, but
expensive. Youll probably need some sub boxes or looms for the drums as well, to save you time and effort.
By now you have easily spent triple the cost of your desk alone, and have gone from a little mixer to a small touring
package. Youve also started taking up a lot more room in the backline van. By bringing in your own desk and cables youve
also reduced the job of the house engineer to lending you a few mic stands and giving you two channels of the house desk for
your mix. My experience of touring at this level (more times than I care to mention) tells me that the house engineer will be
delighted; rather than jump in to help set up an unfamiliar system, hell probably slink off to his warm cubby-hole for a
refreshing cup of tea while he imagines how youll end up hoist with your own petard!
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Having said all this, if you can live with the budget, the risks and the inconveniences described above, a package like this
can be fantastic for this sort of tour but only if you prepare properly. Dont turn up to the first show at three in the afternoon
thinking you can just throw it all together! You need to be there in rehearsals with the band, doing a dry run. You need to label
cables and boxes, check that the leads reach right across larger stages, and that everything works as it should. This is the
role of the production rehearsal, and its best done with the band set up just as they intend to play live. You can then make up
time-saving looms and label them clearly so everything can be interconnected easily and quickly.
If youre creating in-ear monitor mixes, this level of system really comes into its own, as it virtually eliminates the variables
from the stage point of view, and youll be ready with a basic mix straight away. One option you may want to look at is a
monitor-only set up, whereby you carry a desk to use just for the monitors but use the house desk for the FOH mix. This would
still involve carrying a microphone set and stage cables, and youd need a simple microphone split system to go into the
house multicore, but it can work really well, and save you a lot of time. A few times recently, Ive seen this sort of setup but
with the band themselves controlling their own monitors using iPads. Only certain desks will support this, but there are actually
quite a few now, so perhaps thats something to consider.
I know this hasnt really answered the question Im afraid that you really do need to weigh up all of the pros and cons for
yourself but hopefully Ive been able to give you a clearer picture of what you can expect when embarking on this road, and
prompted you into a practical course of action! .
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Q Do surround-sound speakers need to be of the

same type?
Sound Advice : Mixing

Im a sound engineer planning to have a surround-sound setup. The problem is, I already have a
pair of Yamaha MSP5s and I dont really want to spend the money required to have three more of
them for surround use! So, can I have Yamaha HS5s for the centre and rear speakers, and also a different sub (something like
KRK 10S)? Will it really make such a difference?
SOS Forum post
SOS Reviews Editor Matt Houghton replies: Well, depending on what sort of surround material youre mixing on it could be
workable but it will never be ideal, and if youre serious about doing commercial surround work in the longer term youre
going to want speakers designed to be used together. In the meantime, the trick in this sort of setup is to do all of your critical
EQ/balance work in mono or stereo in the first place on your best pair of speakers, and then to pan things out to do your
surround mix safe in the knowledge that your EQ and relative levels already work, and that youll have more space for
separation when mixing in surround. Your speakers might not match perfectly, but youll already have done most of the tonal
work, and youre just trying to get the right idea of positioning. A few years ago when I interviewed Kevin Paul (who, as then
Head Engineer at Mute, had just re-mixed the entire Depeche Mode back catalogue for surround sound), I asked about
budget setups for home-studio owners who wanted to dip their toes in the world of surround-sound. For similar reasons to
those I gave above, he suggested that you could probably get by for a while with a home-cinema surround setup as a
secondary monitoring system, with your critical work being done on your usual higher-quality stereo pair.
SOS Technical Editor Hugh Robjohns adds: Plenty of
professional surround monitoring systems use different, typically
smaller, speakers in the rear channels. However, the critical
aspect is that they are voiced to sound the same as the front
speakers, so that the tonality remains consistent regardless of
where any sound is panned. Most high-end monitor
manufacturers pay a great deal of attention to this aspect,
specifically so that their monitors can be mixed and matched in
the way you describe. However, tonal consistency is likely to be
less well maintained at the budget end of the market, so its
something youll need to assess first hand.
As Matt has said, the work-around is to make all your critical
EQ and balance judgements on the higher-quality front L/R
speakers first, and only after you have the stereo track working
Cheap home-cinema surround systems like this might help
you get a feel for surround-sound mixing, but theyre far from
well to think about re-panning things for surround. You will
the best tool for the job.
probably then notice distracting tonality changes as sounds
move onto the other speakers, but youll have to restrain yourself
from reaching for the EQ controls, as youll otherwise be equalising for the speakers rather than the content! You may well
need to tweak the relative balance of things after panning to compensate for the inherent panning-law effects, but be careful to
tweak only because of panning offsets, not because a speakers response is over or under-emphasising the signal.
Id add a small word of caution about using a domestic home-theatre system. Yes, the five mains speakers will all be
identical and will have the same tonality, which is helpful. However, theyll be compact and most of the bass will be diverted to
the subwoofer via in-built bass-management arrangements. The potential problem is that home-theatre subwoofers are
generally designed with the emphasis on delivering impressive explosions, not tuneful bass. Most seem to have a one-note
bass quality so, once again, make all your bass EQ and balance decisions on your good-quality stereo speakers, not the
home theatre system! .

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Sound On Sound : Est. 1985

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