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DEC./JAN. 2007 - VOL. 2 NO. 6 

THE WORLD OF SOFTSYNTHS AND SAMPLERS 

Very Deep Clinic: 

Inside 

Ableton 

Operator 

Reviews: 

Steinberg Cubase 4 DAW 

Native Instruments FM8 synth 

RME Fireface 400 interface 

Modartt Pianoteq synth 

Big Fish Electro Magnetic Fury and Madahi: African Rhythms 

Akai EWI 4000m 

Electric Wind 

Instrument 

breathes life into 

your synths 

Breakthrough: 7 Gigabytes of samples loaded on one Mac!

From the 

Editor 

Since the last page of this issue is a look forward—what our writers 

see happening in 2007 and beyond—it’s only fitting that we begin 

with a superficial look back at the year that’s almost past. Quite a few 

major products came out, and there have been some pretty important 

developments…and the following is by no means a comprehensive list, 

just a few that come to mind. In no particular order: 

First, the big libraries. East West Quantum Leap Symphonic Orchestra 

got a big update (the XP version), but for me the company’s show-stopper 

is EWQL Symphonic Choirs—an absolutely gorgeous library with its 

own utility program for making the choir sing phrases you construct out 

of syllables. Now, you’re not going to pull off Mozart’s requiem with it, 

but it can do a lot, and even vaguely Latin syllables are a big step 

beyond oohs and aahs (which are also cool, but not the same thing). 

SONiVOX, the company formerly known as Sonic Implants, complemented 

their string section with the rest of the orchestra. Sonic Implants 

Symphonic Collection is a really nice sounding library, and now there 

are three major orchestral libraries (soon to be four when Audio 

Impressions comes out with theirs in January). In addition to the 

GigaStudio version, SISC is now available in Kontakt 2 format with its 

scripting engine (about which more later). 

Editor/publisher: Nick Batzdorf 

Art director: Lachlan Westfall/Quiet Earth Design 

Advertising manager: Len Keeler 

Production manager: Laurie Marans 

Web designer: Denise Young/DMY Studios 

Contributors: Jim Aikin, Jason Scott Alexander, Thomas J. Bergersen, 

Peter Buick, David Das, Peter Dines, Doyle Donehoo, Gary Eskow, Jerry Gerber, 

Paul Gilreath, David Govett, Ashif “King Idiot” Hakik, Mattias Henningson, 

Mark Jenkins, Michael Marans, Monte McGuire, Orren Merton, Chris Meyer, 

Dave Moulton, Zack Price, Frederick Russ, Bruce Richardson, Craig Sharmat, 

Lee Sherman, Dietz Tinhof, Jesse White. 

Advertising contact: Len Keeler 818/590-0018. Len@VirtualInstrumentsMag.com 

Subscriptions/Address changes: 818/905-9101, 1-877/ViMagzn, 

subscribe@VirtualInstrumentsMag.com. The best method is to subscribe via our 

website: www.VirtualInstrumentsMag.com. 

Letters to the editor: NB@VirtualInstrumentsMag.com, or fax: 818/905-5434. 

Writing for Virtual Instruments Magazine: query 

NB@VirtualInstrumentsMag.com or call 818/905-9101. 

The other big orchestra, Vienna Symphonic Library—probably the 

highest-end music software product in the world—came out with one of 

two products that have really advanced the field of sampling: the Vienna 

Instruments player. Of course, the huge update to their sample library 

that goes with it is also no sneezing matter! But this brilliant player 

makes it easy to play, program, and manage their gargantuan library by 

sensing what you play and in many cases switching articulations automatically. 

It’s certain to be influential. 

The other product to change sampling’s face is Native Instruments 

Kontakt 2, specifically its add-on scripting feature. Among many other 

tricks, of a sudden you can add very good legato to any library. There’s 

a lively discussion section about K2 scripting on the VI-Control forum 

(www.VI-control.net), and it’s well worth stopping by to see what’s 

going on—and to see the latest scripts you can download. 

On the hardware front, the big news is of course Intel Macs that can 

switch between Windows and Mac OS on the same machine. How long 

can it be before both are happily running different music software at the 

same time? 

Season’s greetings and Happy New Year from all of us at Virtual 

Instruments magazine.—NB 

Virtual Instruments is published bi-monthly 

for $16.95/year, $26/two years by Virtual 

Instruments, Inc., 3849 Ventura Canyon, 

Sherman Oaks, CA 91423-4710. 818/905- 

9101, 1-877/ViMagzn. 

NB@VirtualInstrumentsMag.com. 

Periodicals Postage Rates are paid at Van 

Nuys, CA, and at additional mailing offices 

under USPS # 023-464. POSTMASTER: 

please send address changes to VIRTUAL 

INSTRUMENTS, 3849 VENTURA 

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Distributor: Rider Circulation Services, 3700 

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UPC: 0 744 70 05792 5 05 

Standard disclaimer: Virtual Instruments 

Magazine and its staff can’t be held legally 

responsible for the magazine’s contents or 

guarantee the return of articles and graphics 

submitted. Reasonable care is taken to ensure 

accuracy. All trademarks belong to their owners. 

Everything in here is subject to international 

copyright protection, and you may not copy or 

imitate anything without permission. 

© 2006 Virtual Instruments, Inc. 

VIRTUAL INSTRUMENTS 5

10 

14 

16 

20 

December/January 2006/7 

V2.N6 

Letters 

Launch 

Introductions, updates, news 

Voltage Control 

by Michael Marans 

Let’s Make Some Noise!?Part B in our series on using your 

synth’s noise generators to create drums, percussion instruments, 

and sound effects. 

VERY DEEP CLINIC: 

Operator on the Line 

by Jim Aikin 

The add-on synth in Ableton Live 6 is a lot more than just FM 

lite 

8 VIRTUAL INSTRUMENTS 

26 

38 

44 

54 

7 Gigs Loaded on 

One Mac 

by Nick Batzdorf 

Accessing 7GB of samples on a single Mac. 

Sequencing Samples 

by Thomas J. Bergersen 

One of the world’s true masters at MIDI programming starts 

a new series. This issue: balance. 

Bent Clicks 

by Paul Gilreath 

One of the most overlooked aspects of a good sequenced 

performance is the tempo. Here are some pointers. 

MIDI Mockup 

Microscope: Nathan 

Furst 

by Frederick Russ 

In this installment of our series on composers and how they 

did their MIDI programming, talented composer Nathan 

Furst discusses “Heart of the Baja Part 2” and “Shark Attack.” 

Download these cues at www.VirtualInstrumentsMag.com 

and follow along.

24 

30 

40 

VI contents 

Native 

Instruments FM8 

by Lee Sherman 

The second coming of the DX-7 gets 

a major update. 

RME Fireface 

400 FireWire 

Audio Interface 


This little half-rack box houses a 24bit/192kHz 

audio interface with 18 

ins and outs. But that’s just the 

beginning, because it can also route 

software tracks all around the room. 

Akai EWI 4000s 


Do you play a wind or brass instrument? 

The Electric Wind Instrument 

is a wonderful controller for playing 

synths. Its latest incarnation of this 

fabulous instrument now includes a 

built-in synth and some new MIDI 

control features—and it comes in at 

half the price. 

48 

52 

56 

Modartt 

Pianoteq 

modeled 

acoustic piano 

by Jason Scott Alexander 

This surprising piano only uses samples 

for things like pedal noise—it’s 

actually modeled from the ground 

up. As a result, every behavior of a 

real piano is accounted for. 

Loop Librarian 

by Chris Meyer 

Loops loops loops and libraries. 

Reviewed: Big Fish Electro Magnetic 

Fury and Mahadhi: African Rhythms. 

Steinberg 

Cubase 4 

by Mark Jenkins 

A look at the update to this massive 

sequencing package. 

61 

December/January 2006/7 

V2.N6 

VI reviews 

Importing DX-7 sounds into Native 

Instruments FM8 

random 

tip 


VI letters 

Letters 

write to: 

nb@virtualinstrumentsmag.com 

Physical modeling 

While I understand completely the notion 

of physical modeling as it applies to traditional 

instruments (i.e. representing the length of 

a trumpet, the materials of which it’s made, 

differential equations of the wave motion, 

etc), I have never understood what it means 

to do physical modeling of an analog synthesizer 

(for example, Yamaha claimed such for 

the An1X). 

What exactly is being physically modeled? 

It seems to me that to do physical modeling 

of an analog synth, you have to model such 

things as the behavior of transistors, resistors, 

capacitors, i.e. the electronic equivalent of the 

length of the trumpet and its materials. 

But I don’t get the sense that that’s what’s 


happening—it seems like they are just “modeling” 

the OUTPUT of oscillators and so forth. 

And if that’s the case, then the Yamaha DX7 

was doing this kind of thing a long long time 

ago. 

David Jameson 

via email 

As your letter points out, it’s not a very precise 

term. And let’s face it: it’s very difficult to 

market a product just by saying it sounds good 

and some rock star—who you’ve just eliminated 

as a potential customer by making him or her 

an endorsee—uses it. It’s far more interesting to 

be able to talk about the technology. “We were 

very careful to do an excellent job recording this 

sample library using good mics” isn’t going to 

turn a lot of heads. 

So yes, we have seen “modeling” used fairly 

liberally as a selling point. That was especially 

true in the mid-’90s, when it was the buzzword. 

Having said that, if the definition of PM is 

using equations and algorithms to simulate the 

behavior of a physical sound source, then what 

Yamaha does did in the VL1 Virtual Acoustic 

synth (which may have been the first commercial 

synth to use PM) absolutely is physical modeling. 

The VL1 uses Digital Waveguide synthesis, 

which actually uses varying delay lines, and 

Korg used it, and at one point Technics used it 

in their WSA-1 synth. 

Applied Acoustic’s Tasman synth and I’m 

guessing Apple’s Sculpture (one of the synths 

that comes with Apple Logic Pro, although 

Apple only touts it as “component modeling”) 

are two current PM softsynths that come to 

mind. Modartt PianoTeq, reviewed in this issue, 

uses PM to create its pianos, but the actual 

technique is a closely held secret. My educated 

guess is that if they haven’t licensed Digital 

Waveguide synthesis, they’re using a different 

variation of the Karplus-Strong plucked string 

model, which uses a noise generator followed by 

a filter with a delay feeding back into it. 

Circling around your original question some 

more, some instruments actually do model the 

various circuit components of the product being 

emulated. Universal Audio does that in their 

plug-ins, for example. Whether a given softsynth 

actually models the oscillator from the synth it’s 

emulating is another question, but to me 

absolute purity isn’t what makes a product 

valid—it’s the bottom line sound. 

Is FM synthesis physical modeling? Who 

knows. What I do know is that a softsynth version 

of the VL1 would be fantastic, because I 

still consider it the pinnacle of synthesis in many 

ways. 

High praise 

I just recently picked up your magazine 

and am very excited about it. I am a police 

officer that has patrolled the [withheld] area 

for many years. I recently was injured and will 

no longer be able to do police work. Music 

composition has helped me through my rehabilitation 

in many ways, and your magazine 

has given me so much information on the 

emusic world. 

I know your mag is not necessarily aimed 

at novices such as me, but please know that it 

is making a difference for me. Even though I 

don’t have any of the highspeed soft synths (I 

use Cakewalk Sonar with its stock instruments), 

with your help I am learning to get 

the most out of what I have, and that is exciting. 

I have a long recovery ahead that will not 

be easy, but emusic is playing a big part in 

keeping me positive and keeping my head on 

straight, and your magazine is a big help. 

Keep up the good work you are doing and 

I can’t wait for the next issue. 

Chris Watson 

via email 

This is one of the nicest letters we’ve received, 

Chris. Thanks so much, and we wish you all the 

best with your rehab. 

The synths in Cakewalk Sonar are absolutely 

nothing to sneeze at, by the way, in fact we 

have a Very Deep Clinic about them in the 

works And we certainly hope novices will enjoy 

the magazine too. Our “First DAW” series hasn’t 

gone away!

VI letters 

Digital puzzle 

[Please see the fab diagram George included 

with his question.] 

What does someone do if they want to use 

the SPDIF outputs on multiple hardware 

synths simultaneously? Most sound cards and 

interfaces have only one SPDIF I/O, so only 

one of these things can be connected at a 

time. 

I’ve done a bit of research. There are a few 

products out there (M-Audio makes one) that 

will let you connect multiple ADAT devices 

and provide a master clock and connection to 

your DAW/soundcard. I also ran across a 

series of product called “digital detanglers” 

by Z-Systems Audio Engineering, but these 

seem to be simply format converters...and 

très expensive. 

I’m looking for something in the “prosumer” 

space that will allow me to hook up 

three or four SPDIF sources (e.g., V-Synth XT, 


Virus TI, etc.) and pipe them into my single 

SPDIF I/O on my Firewire 410. I can’t be the 

only person out there that wants to do this, 

can I? 

Any ideas would be a help. Thanks. 

George Napier 

via email 

First of all, thanks very much for including the 

diagram of your studio. It’s actually more 

involved than necessary to answer your question, 

but it’s always interesting to see peoples’ 

rigs. This is technically a hardware question, of 

course, but since you’re integrating the hardware 

with V.I.s as well… :) 

There are two issues here: digital 

clocking/sync and where you’re going to stick 

the S/PDIF outputs. As to clocking, every device 

in a digital audio system has to be referencing 

the same digital clock, or you’ll get clicks and 

pops. Both the VSynth and the Virus have 

S/PDIF inputs, so the obvious answer would be 

to use the clock from your FireWire interface. 

But the FW410 only has one S/PDIF out, and 

you need two. It so happens that RCA S/PDIF 

digital signal uses 75-Ohm video cable, so you 

can create multiple outputs using a standard 

video distribution amp. Radio Shack makes one 

for $50 that should work fine, but there’s a wide 

range of models on the market. 

Just run S/PDIF cables from the FW410’s out 

to the distribution amp’s in, and then from two 

of its outs to the synths’ S/PDIF inputs. Put the 

synths on digital sync (a.k.a. external sync) and 

that’s solved. Be sure to use 75-Ohm cables, 

though, or the system may not be reliable. 

Your other problem, finding an audio interface 

with multiple S/PDIF inputs, is a little more 

difficult. MOTU used to make a 308 box for 

their PCI-324 and -424 cards, if you can find 

one. Lynx AES 16 is another, and in fact it features 

asynchronous I/O, meaning that it can 

resample the digital inputs that aren’t clocked 

together. 

You don’t want to use USB, but adding an 

inexpensive USB interface that has S/PDIF I/O 

might be the most practical solution. There are 

some very inexpensive ones, actually. 

I also see that you’re also using Native 

Instruments Kore—or maybe not using its controller 

as an interface? It has S/PDIF out but no 

in, so that rules it out as a place to stick your 

hardware synths’ digital outputs. But you could 

make it part of an aggregate device if you wanted 

to use it for more analog inputs. (Aggregate 

devices are seen as one larger audio interface by 

Mac OS X software.) 

If you go into the Aggregate Device editor in 

Audio MIDI Setup, you’ll see a checkbox for 

which device to use as the clock master, and 

then there are options to resample the inputs. 

That’s what you want to do with Kore, since it 

can’t be clocked externally. 

Suggestion box 

An article for a VI issue that would be very 

valuable to me, and to many others as well, 

would be a step by step overview of running 

V.I.s on an external Mac and connecting it to 

a host computer. I know far less about this 

than I would like to; it is time consuming to 

hunt up the information; and there are many 

users, particularly in the Mac community, 

who are using second computers as V.I. hosts. 

If such an article could address things like 

the network feature in Audio MIDI Setup in 

OSX, ways to bring a laptop/Traveller combination 

or similar rig into the main set-up, 

avoiding conflicts with dual installs of the 

same plug/V.I., etc, it would be very much 

appreciated. 

Any chance this might happen soon? 

I’m enjoying the publication, thanks for 

your continued innovation. Been reading 

you since Home & Studio Recording [which 

later became Recording] back in the day. 

Robert R. Martin, PhD 

Victoria B.C., Canada 

Absolutely, great suggestion. You shall have 

it, Sir, and thanks for the compliments. 

This letter is here as an example of what we 

welcome from you readers. Please let us know 

what you want to see: 

NB@VirtualInstrumentsMag.com

VI launch 

Launch 

Introductions, updates, news 

Native Instruments 

KORE 1.1, Kontakt 2.2 

universal binary 

This free update to NI’s KORE 

Universal Sound Platform can now run 

without its controller/audio-MIDI interface 

hooked up to the computer. Other 

new features include a new SingleSound 

format; integration with NI’s new 

Massive synth and the latest updates to 

their other instruments; and a pre-listen 

feature for auditioning sounds before 

they’re loaded. 

The most important new feature in 

v.2.2 of their Kontakt sampler is that it’s 

now a universal binary, which means it 

works on Intel Macs as well as PowerPCs. 

www.native-instruments.com 

Millennia Media 

Orchestral Recording 

Discussion #1 

If you’re interested in learning about 

how to improve the production of your 

sampled orchestral sequences, this discussion 

about large ensemble acoustic 

recording is sure to help. Millennia 

Media makes some of the best analog 

recording processors (mic preamps, etc.) 

in the world. 

www.mil-media.com 


M-Audio NRV10 Analog Mixer/FireWire 

Audio Interface 

This $899 unit combines an 8x2 analog mixer with a 10x10 Pro Tools M-Powered-compatible 

audio interface. Among other features, the unit features a built-in effects section 

with 16 effects that run on its hardware. It also comes with Mac and Windows control software 

that adds dynamics processing plus two VST effects slots to each channel. 

www.m-audio.com 

VSL Saxophones and Elements 

Vienna Instruments libraries, update 

Vienna Symphonic Library’s 24-bit/44.1kHz Saxophones 

Collection ($475) features 35GB+ of soprano, alto, tenor, bari, 

and bass saxes, intended for both legit and jazz use. 

Articulations for jazz and big band are included: tongue slaps, 

growls, screams, extended harmonics, key clicking, and more. 

There are also long and short downward ending phrases triggered 

as release samples. 

Elements ($365, updates $45) is an 18GB expanded version 

of the Horizon Series Glass & Stones library, featuring all kinds of 

mallet and percussion instruments such as glass bells, musical 

glasses, glass harmonica, verrophone, and lithophone. New to 

the library are such mainstays as bass waterphone, and fluttertongued 

blown bottle. 

Both come with the Vienna Instruments player, which was just 

updated to v.1.1, which among other things: loads more quickly 

and has some user interface improvements. This player features 

Performance Detection algorithms that that switch articulations 

automatically by analyzing the intervals, repeated notes, patterns, 

and speed of your playing in real time. 

www.VSL.co.at

McDSP Updates 

McDSP has been 

producing some of 

the best Pro Toolscompatibleplugins 

for a long time. 

Their latest updates 

to the Native and 

HD product lines 

include support for 

Intel Macs, 

improved RTAS efficiency, 

faster interface 

updates and 

loading times, and Pro Tools HD Accel 

card optimizations. The ML4000 ($495) 

is a high-res brick wall look-ahead limiter 

and a multiband dynamics processor. 

Analog Channel LE and ML4000 LE are 

new additions to their $495 Project 

Studio bundle. 

www.mcdsp.com 

Alexander Publishing 

“How Ravel 

Orchestrated: Mother 

Goose Suite” 

This edition includes a book, score, and 

CD, designed for learning orchestration. 

The analyses that follow each movement 

include electronic scoring insights. 

www.professionalorchestration.com 

Digidesign Mbox 2 Mini 

VI launch 

$329 list now buys you a 

USB interface that has both 

phantom powered mic and 

line inputs, and line and headphone 

outputs. The software 

bundle starts with Pro Tools 

LE, and then it includes the 

Xpand! sample playback/synthesis 

workstation, several 

Bomb Factory plug-ins, about 

38 DigiRack DSP plug-ins 

(dynamics, reverb, EQ, pitch shift, delay, and on and on)…and then the fun starts. It also 

includes several lite versions: Ableton Live Lite 4, Propellerhead Reason Adapted, 

FXpansion BFD lite drums, IK Multimedia SampleTank SE, IK Amplitube LE, IK T-Racks EQ, 

Celemony Melodyne Uno, and more. 

www.Digidesign.com 

MPC Lockbox, Big Fish 

DrummerPacks for 

DrumCore 

The SonicEmulations MPC Lockbox loop 

library ($59) is a new expansion pack for 

Submersible Music’s DrumCore loop librarian/player. 

This new DrummerPack features 

hip-hop/DJ-type loops and single hit drum and 

percussion sounds, ready to play by MIDI. 

Big Fish’s “Brain: One Stroke Done” (reviewed 

in our 10-11/06 issue) and “Roots of South 

America 2” libraries ($79 each) are also now 

available as DrumCore DrummerPacks. This pack 

is also available in loops—which are ready to be 

drag/dropped into a DAW—as well as individual 

hits to be triggered by MIDI. 

www.drumcore.com, 

www.bigfishaudio.com, 

www.bandmateloops.com 

Garritan Stradivari Solo Violin 2.0 

The update to this solo violin now comes in a Native 

Instruments Kontakt 2 Player, so you don’t need Kontakt to play it. 

The Garritan Strad features “Sonic Morphing,” which crossfades 

across dynamic layers; lets you control the onset, rate, and speed 

of vibrato; and shape vibrato in real time without the phasing 

problems you get when crossfading normally. $199, upgrades $29 

+ s/h. 

www.garritan.com 


VI feature 

Let’s Make Some Noise!—Part II in our series on 

using your synth’s noise generators to create 

drums, percussion instruments, and sound effects. 


Voltage 

Last issue we concentrated on building a standard 

trap kit. This time around we’ll look at creating 

complementary percussion instruments and immerse 

ourselves in some really bad weather. (We would say, 

“So grab your Macs,” but the pun would be way too 

obvious, and we’re not that desperate….) 

by Michael Marans 

Ooh, Baby, I Got an Itch… 

We’ll start off with two versions of the now 

ubiquitous “record scratch,” only we’ll be 

using virtual vinyl. 

In Synth1 (see Sidebar, Downloading Synth1 

and the Example Patches), call up program 45, 

Simple Click. Play any note, and all you’ll hear 

is a plain old ordinary click?and a rather 

unmusical one at that. 

It’s instructive to analyze the origins of the 

click, so give the Synth1 front panel the onceover 

(Fig. 1). You’ll notice that the Attack, 

Decay, and Sustain parameters of both the filter 

and amplitude envelopes are set to their 

minimum values. Since these values essentially 

shut off the filter (i.e. no harmonic content 

passes through) and the amplifier is “closed,” 

all we hear is the click of the electronics turning 

on for a brief moment when a key is 

pressed. But no “real” sound is actually being 

generated by the synth engine. 

more 

online 

www.virtualinstrumentsmag.com 

Control 

Let’s open the amplifier by setting the 

Decay control to its halfway position. Still 

nothing. Why? 

Look at the filter setting. With LP12 selected 

(a lowpass filter with a 12dB per octave 

slope), no sound is passing through the filter, 

so the settings of the amplifier envelope literally 

have no effect on the sound. Toggle to 

LP24 and play a note; same deal: no sound. 

Now select HP12æa highpass filter with a 

12db per octave slope. Hey! That almost 

sounds like a thick hi-hat hit! 

If you’ve been following these Voltage 

Control articles for a while you’ll immediately 

know why. First, a highpass filter allows all 

sound higher than the filter cutoff frequency 

to pass through and be heard. So with our filter 

closed down to 0Hz, we’re allowing all frequencies 

higher than 0Hz to pass through 

unfiltered. 

Second, Oscillator 2, set to generate noise, 

is the only oscillator sounding (note the fully 

clockwise position of the Oscillator Mix knob). 

As we pointed out last time, hi-hats and cymbals 

are comprised entirely of noise (all frequencies 

playing at equal amplitude) and the 

amplifier envelope is the primary tool used to 

shape that noise into a recognizable sound. In 

this case the amplifier’s Decay setting is creating 

the shape of a medium length hi-hat hit. 

Now back to our regularly scheduled programming: 

record scratches. Starting with our 

boring click, let’s start molding the sound into 

something resembling the mistreatment of 

vinyl. We’ll begin by selecting LP12 for the filter. 

Now turn up the filter envelope’s Attack 

time to about one-third and play a note. 

You’ll hear that we have a rough approxima-

Fig 1: The vinyl “record scratch” patches created in this article began life as a simple, uninspiring electronic 

click. Worse, the click was really nothing more than the sound of Synth1 trying make sound, but the 0 

(minimum/off) settings of the envelope generators make playing a purposefully-generated sound next to 

impossible. Turning the click into record scratches required selecting the proper filter, adjusting the filter and 

amplitude envelopes, setting the filter cutoff point and resonance, modulating the sound with LFOs, and 

adding effectsæin other words, using a variety of voltage controls to turn the plain and the boring into the 

bad and the beautiful. See the main article for details. 

tion of a simple scratch, but the sound cuts 

off rather abruptly. 

Time to enlist the filter’s Decay control. 

Bring it up to about one-third, and you’ll 

notice that you now have the basis of a fairly 

convincing quick scratchæalbeit one that’s 

awfully thick. Adjust the filter Attack and 

Decay parameters to taste, using Attack to 

control the speed of the scratch’s onset and 

Decay to control how long it lasts. You might 

find it helpful to think of the word “Wow” 

while adjusting these parameters. 

Now about that thickness. Please say hello 

to the Bandpass filter (BP12 on Synth1). A 

bandpass filter allows a specified band of frequencies 

both above and below the frequency 

cutoff point to pass through and be heard. 

This type of filter lets us maintain the core 

timbre of the sound while simultaneously 

attenuating unwanted frequenciesæthat is, 

those that are too low and too high. 

So select BP12 and play a few notes. Notice 

how this thinning out of the sound really helps 

home in on the tonality of a scratch. 

The next steps are all “tweaks,” and subject 

to personal preference and musical application. 

First, try increasing the filter cutoff 

and/or the resonance to thin the sound further 

and give it the perception of having a 

higher pitch. (Remember that pure, unfiltered 

noise has no pitch, so accentuating higher or 

lower frequencies will make the sound appear 

to be pitched higher or lower, accordingly.) 

Then add compression (Effect > Comp, on 

Synth1) to make the sound fatter and more 

consistent in volume from start to finish. 

Unison mode and filter SAT (Saturation) can 

thicken the sound considerably, and you can 

add a bit of automated panning using an LFO 

to animate the sound’s stereo image. 

(Turn on LFO1 and note that the sound 

immediately travels across the stereo soundstage, 

in accordance with the fairly rapid 

speed of the LFO. Slowing down the speed 

allows you to move the sound across the 

stereo soundstage with each successive keyon. 

Click on LFO1’s KEY button to retrigger 

the sound from the exact same stereo position 

with each note-on.) 

Our final version of a basic scratch, which 

utilizes all of the above, is stored in Program 

46, Record Scratch 1. Program 47, Record 

Scratch 2, builds on the core sound to create 

a “double scratch” by adding filter velocity, 

and using a stereo delay to split and double 

the sound across the stereo soundstage. 

One of the highlights of the patch: LFO2, 

assigned to Filter (try assigning it to 

Amplitude as well), can be used to add multiple 

scratches by increasing the value of the 

speed control and varying the note-on velocity. 

Truly complex rhythms can be created 

through the use of this control and a bit of 

keyboard dexterity. 

VI 

feature 

Perking up 

The high-pitched strike of a Clave, the classic 

Latin percussion instrument (Program 48, 

Clave F3), can be created using a combination 

of noise and filter self-oscillation. A bit on 

the latter: when the Resonance parameter of 

the filter is set to maximum, the filter cutoff 

frequency is so accentuated that it actually 

produces a tone (generally a pure sine wave). 

The pitch of that tone can then be controlled 

by adjusting the filter’s cutoff frequency. 

(Note that if you want the tone to track 

the keyboard, you need to turn the Filter’s 

track control [TRK] to its maximum position, 

which,on most synthesizers will produce the 

proper voltage to play properly intonated 12tone 

scales.) By combining the tone produced 

by filter resonance with the tone(s) produced 

by the oscillators carefully, you can create 

noise-based timbres with distinctive pitches. 

In creating our Clave sound, we used two 

noise sources. OSC2 is producing pure white 

noise. OSC1, set to a sine wave, has its FM 

(Frequency Modulation) control set to about 

the 50% mark (64 on the MIDI parameter 

value scale). The sidebands (harmonic frequencies)) 

created by feeding the complex 

noise waveform of OSC2 into the pure sine 

wave generated by OSC1æwhich is what the 

FM control is doingæare wild and “random,” 

thereby generating a sound heavily predominated 

by noise. (See Fig. 2.) 

To hear OSC1 on its own, turn both the 

Filter’s RES knob and the OSC MIX control to 

0 (fully counter-clockwise). Note that by 

increasing or decreasing the value of OSC1’s 

FM parameter, you can add as little or as 

much noise to the sound as desired. 

Reselect Program 48 so that you’re back to 

the original Clave patch. If you listen closely, 

you can hear the sound’s noise characteristics, 

though the dominant timbre is the short, 

Let's open the amplifier by setting the Decay 

control to its halfway position. Still no sound. Why? 

pitched strike that’s created by filter resonance. 

Last tweak: toggle between the LP12 and 

LP24 filter settings. You’ll note that LP12 

allows for more low frequency noise content. 

Our preference (strictly personal) is for the 

“cleaner” LP24 sound, played between C3 

and F3. As always, experiment with different 

keyboard ranges and filter settings. 

Give yourselves a hand 

The analog handclap, popularized by 

Simmons way back when (yes, there were 

earlier incarnations, but Simmons is credited 

for creating the handclap that slapped you in 

the face), makes use of several key synthesis 

techniques. Programming a basic handclap is 


VI feature 

Fig. 2: Frequency Modulation (FM) is a type of synthesis (popularized by 

Yamaha’s DX series) in which one wave (a modulator) is fed into another (a carrier) 

to generate a timbre that is harmonically rich. When the modulator and carrier 

are sine waves, the harmonics produced are predictable. In Synth1, the output of 

OSC2 is the modulator, and the carrier is OSC1. Since both of these oscillators are 

using complex waveforms in the Clave patch, the resulting timbre produced when 

using FM comprises hundreds of frequenciesæwhich, due to their number and 

amplitude, are a form of noise. Try raising and lowering the FM amount to hear 

the effect on the “noise” portion of the Clave patch. 

pretty easy; making a knock-you-over-dead 

clap is a multi-step process. 

We’ve programmed two claps. Program 

49, Small Handclap, relies almost exclusively 

on basic filter and envelope settings to create 

the core sound. A highpass filter is used to 

nail the timbre, which if you listen closely has 

many of the spectral characteristics of the hihats 

we constructed last time around. 

To see for yourself the dramatic effect a filter 

has on a sound, toggle through the other 

available modes. Once you leave HP12 (the 

highpass filter) our clap becomes lifeless and 

sounds more like an unwelcome stranger 

knocking at our door. 

Now go back to HP12 and enable Tempo 

Delay. Our settings double the sound and 

spread it across the stereo spectrum. Next, 

enable Chorus/Flange, which thickens the 

sound. For fun you can enable Effect A.D.1, 

which gives you an interesting percussive 

sound, but out of the realm of the clap. 

Now Select Program 50, Big Handclap. 

This is essentially the same program as Small 

Handclap, but when we enabled Effect A.D.1, 

we changed the filter to BP12 (bandpass). 

Doing that allowed us to retain the basic clap 

soundæby cutting out much of the high frequency 

content we heard in the A.D.1 experiment 

in the previous paragraphæwhile at the 

same time thickening the sound considerably. 

Deep into the (Rain) forest 

While a completely different type of instrument 

from the hi-hat?and certainly not metallic 

in any way?the decidedly primitive rain- 


stick has a sound 

that contains a number 

of the same 

spectral components. 

The primary 

difference between 

it and a hi-hat is in 

the shape of the 

envelopes (both filter 

and amplitude). 

So even though the 

rainstick is about as 

organic an instrument 

as one can 

imagine, the bits 

and bytes of our virtual 

analog synthesizer 

can do a very 

convincing job of 

emulating it. 

Let’s start by calling 

up Program 51, 

Medium Hi-Hat. This 

provides our basic 

timbre. (For those of 

you not using 

Synth1, our example 

Medium Hi-Hat 

sound is created 

using noise 

processed through a 

high pass filter with 

a bit of resonance 

added to give it a metallic edge. The filter 

envelope is used to fine tune the timbre, but 

the amplitude envelope provides the majority 

of the hat’s “sharp hit” characteristic.) 

We now need to tweak the envelopes to 

eliminate the hi-hat’s sharp attack and create 

the trademark rainstick sound of “grains” 

(rice, small pebbles, etc.) falling through the 

hollow stick. Start by increasing the amplifier’s 

Attack time to a bit more than halfway-up. 

Play and hold a note. Notice how this simple 

adjustment has a dramatic effect on the 

sound, and has taken us quite far toward the 

rainstick emulation. 

Now we’ll adjust the Amplifier Decay 

timeæincreasing it to about three-fourths of 

its throwæto create the long, smooth sound 

of the grains continuing their journey through 

the stick. 

Okay. We’re pretty close to the basic shape 

of the sound, but it’s not definitively a rainstick. 

To go that extra mile, we’ll need to 

enlist a number of other parameters. 

Call up Program 52, Rainstick, and play a 

note. You’ll notice that the sound is now 

much more refined timbrally (it’s not just 

pouring rain anymoreæwe can hear the individual 

“droplets”). We’ve done this by making 

some minor changes to the filter envelope, 

but more significantly we’ve enlisted the aid 

of the LFOs. 

LFO1 is set to a noise (random) waveform 

with its output routed to the filter, causing 

the filter cutoff point to change rapidly over 

the course of a held note. This rapid randomization 

of the frequency content helps to dis- 

tinguish the “droplets’ ” individuality. LFO2 

uses a high-speed random sine wave assigned 

to amplitude, which further serves to define 

individual droplets. Finally we adjust the FRQ, 

RES, and SAT controls for the desired tonal 

coloration. 

Uh-oh…bad weather ahead 

Mother Nature is pretty good at creating 

impressive noises, and it just so happens that 

two of the most common?lightning strikes and 

thunder?are comprised of ingredients readily 

available on basic analog subtractive synthesizers. 

If you’ve been following this series, you 

know that we always break down sounds into 

basic components, i.e. their core tonality 

(waveform), harmonic content (filter), and 

amplitude (VCA, or Voltage Controlled 

Amplifier), and then apply various voltages 

from modulation sources, such as envelopes 

and LFOs, to manipulate the core ingredients. 

So with those basic principles in mind, let’s don 

our raincoats and call in the thunderclouds. 

A lightning strike usually consists of a sharp 

crack, followed by thunder. The particular 

incarnation we created here (Program 53) has 

the lightning striking right on top of us. In 

other words, the thunder happens simultaneously 

with the initial strike. 

The lightning sound starts with OSC2 set 

to noise, and the filter parameters set to create 

a medium length “crack.” (You should 

experiment with the filter envelope values to 

see how they affect the sound. Hint: Decay 

and Sustain are critical.) Synth1’s filter SAT 

parameter has been used to fatten the crack. 

LP12 is the filter selection, since it allows both 

a great amount of high frequency content to 

pass through for the initial crack, while at the 

same time providing enough filtering for us 

to create the rumbling thunder that follows. 

Now check the settings of the Amplitude 

(CONTINUED ON PAGE 62) 

Downloading and 

installing Synth1 

and the example 

patches 

As with our past articles, we’re using 

the freeware Synth1 for our examples. 

Instructions for downloading and 

installing Synth1 can be found at More 

Online at 

www.virtualinstrumentsmag.com. There 

you’ll also find the Synth1 patch examples 

used in this article, as well as those 

used in our last three installments. You 

can use the synth of your choice, since 

the controls we refer to are common to 

virtually all software synths (at least 

those of the analog subtractive variety).

VI very deep clinic 

by Jim Aikin 


Operator on the Line 

When I first heard that Ableton had released a 4-operator 

FM synth called Operator as an optional add-on for Live, I’ll 

admit I glanced at it and yawned politely. As much as I like 

FM, how could four operators hold their own against instruments 

that have six? 

When persuaded to take a closer look, though, my opinion 

ratcheted upward. The multimode filter and the useful list of 

waveforms open up a lot of sound design possibilities that 

aren’t apparent on the surface. The new version of 

Operator available with Live 6 contains in itself only a few 

small refinements—but the more flexible architecture of 

Live 6 boosts Operator to a whole new level, as you’ll discover 

in this clinic. If you like synthesizers, this little monster 

will truly chop wood and carry water for you. 

The add-on synth in Ableton Live 

6 is a lot more than just FM lite 

The first part of this clinic is for those who 

are new to FM synthesis. It introduces 

Operator’s FM features briefly, and shows 

how to explore the tone colors available with 

FM. After shining a spotlight on a few of 

Operator’s other features, we’ll tackle some 

expert concepts—both FM programming and 

ways to use Operator in conjunction with Live 

6’s new Racks and expanded modulation routings. 

While working on this article, I interviewed 

ace remixer and sound designer Francis Preve 

(www.fap7.com). Francis designed many of 

the factory presets for Operator, and has had 

tracks on Top Ten dance remix CDs. In addition, 

he’s the author/editor of The Remixer’s 

Bible (Backbeat Books). You’ll find his insights 

(uncredited) scattered throughout this article, 

and a few specific tips in the sidebar. Thanks 

Fran! 

FM 101 

FM (frequency modulation) synthesis was 

first made popular by Yamaha in the 1980s. 

In FM, complex sounds are created by having 

one simple waveform modulate another. 

Often the waveforms being produced by the 

oscillators are sine waves, which are about as 

simple as an audio signal can get. The oscillator 

producing the sound we hear is called the 

carrier, and the oscillator being used to

change the tone of the carrier is called the 

modulator. 

A standard FM synth has at least four oscillators, 

and many have six. These can be configured 

in various ways. For instance, with 

four oscillators we might have two 

carrier/modulator pairs, or we might have 

three modulators, all operating on a single 

carrier. The configuration of oscillators is 

called an algorithm. Operator offers a choice 

among 11 algorithms, as shown in Fig. 1. 

Two key factors affect the tone of the carrier: 

the relative frequencies of the carrier 

and modulator, and the amplitude of the 

modulator. Cranking up the level of the 

modulator adds more overtones to the 

sound. While the ratio of the frequencies of 

the carrier and modulator controls which 

overtones will be heard, the formula that 

determines the exact combination of overtones 

produced by a given frequency ratio is 

not simple, and not interesting either. The 

best way to work with FM is to try different 

tunings for the carrier and modulator, and 

listen to the results. 

To give the user control over these factors, 

each oscillator has its own coarse and fine 

tuning knobs and an output amplitude knob. 

In addition, each oscillator has its own amplitude 

envelope, which you’ll see in the center 

panel when you click on one of the oscillators 

in the left column. Unlike an analog-type 

synth, where one amplitude envelope is often 

called on to contour the entire sound, FM 

synths give us separate amplitude control 

over each oscillator. 

The combination of an oscillator and an 

amplitude envelope is called an operator— 

hence the name of the synth. (Memo to 

Ableton: when Antares named their filter 

plug-in Filter, it was not a trend you want to 

emulate. Calling Live’s amazing new sampler 

Sampler... well, maybe all the good names 

have been used up.) 

FM discovery step-by-step 

Let’s walk through the process of creating a 

basic sound in Operator. Launch Live and follow 

along: 

1. Drag Operator from the Browser into an 

empty MIDI track. Don’t drag one of the 

existing presets—we want to start from 

Fig. 1: The algorithms in Operator. The box or boxes on the bottom row of each icon are carriers, and 

those above them are modulators. Algorithms 6 (right end of the top row) and 10 (highlighted) are new in 

Live 6. At first glance 9 and 10 might seem almost identical, but if you click on 9 and look at the “wires,” 

you’ll see that in 9 the modulator affects all three carriers, while in 10 it affects only carrier C. 

scratch, so select Operator itself in the 

Browser. 

2. Click the In button in the MIDI track and 

play your keyboard to verify that Operator is 

operating. You should hear a sine wave. 

3. Looking at the panel, you’ll see that the 

default patch uses algorithm 1, in which all 

four operators are stacked. (It’s visible in the 

lower right area.) But you’re only hearing a 

sine wave, because the levels of operators B, 

C, and D are set to –inf dB. Turn up the Level 

of operator B while playing the keyboard. 

You’ll hear the sound gradually acquire more 

overtones. This is the core of FM synthesis. 

4. After setting the Level of B to a reasonable 

value, such as –15dB, try adjusting the 

Coarse frequencies of A and B. You’ll discover 

that each ratio of A to B produces a distinctive 

combination of overtones. 

5. Return A and B to Coarse settings of 1 

and 1. Then click on the Fine tuning knob of 

B, hold down the computer’s Ctrl key to allow 

VI 

very deep clinic 

smaller adjustments, and move the Fine value 

up to between 3 and 6. You’ll hear the tone 

become animated as the two oscillators are 

detuned from one another. 

6. With operator B still selected, drag the 

envelope sustain level down to zero. Now 

operator B is producing an attack transient. 

Adjust the decay time shorter or longer to 

change the length of the attack transient, and 

try adjusting the Coarse tuning of B to 

change the harmonic character of the transient. 

Leave the envelope set at a medium 

decay length, perhaps about 3 seconds. 

7. Click in the algorithm panel (lower right) 

and select algorithm 7. In this algorithm three 

modulators all affect one carrier. The sound 

won’t have changed yet, because we’ve only 

turned up operator B. So turn up operator C 

to –15dB, turn its envelope sustain down to 

zero, and set the decay time to a much shorter 

value, in the 40-60ms range. Turn its 

Coarse tuning up to about 7. This will give 

each note a very percussive attack. FM synthesis 

is especially good at tuned percussion 

sounds like the one you’ve just created. 

8. Turn up operator D to about –50dB, 

change its Coarse tuning to 8, and give it an 

envelope with a 4-second attack, a 32-second 

decay, and a –15dB sustain. This will produce 

a slow swell containing new overtones as the 

note sustains. 

9. To hear the contribution that each operator 

is making to the composite tone, click 

their colored boxes to mute them one by 

one. Muting operators is extremely useful 

when you’re developing an FM sound. 

10. Add velocity response to some or all of 

the operators to make your new preset more 

playable. 

Three important features 

Operator is sometimes compared to the 

Yamaha TX81z, a popular 1980s-era rack- 

The formula that determines the exact 

combination of overtones produced by a given 

frequency ratio is not simple, and not interesting 

either. 

mount synth that also produced its tones 

using 4-operator FM. But Operator has some 

powerful features that go well beyond what 

was possible in those early days. 

Filter. Operator’s multimode resonant filter 

is a vital sound-shaping tool. In Live 6, the filter 

has been enhanced with a choice between 

12dB and 24dB per octave rolloff slopes. The 

24dB options produce a more definite filtering 

action, while the 12dB options are more 

gentle. 


VI very deep clinic 

Fig. 2: The Operator panel. The filter is selected, 

and its Freq

Now when you load the Rack preset into a 

different song, you’ll only have to make one 

modulation routing—mod wheel to the 

Macro knob. 

Also note that individual envelope segments 

can be modulated using MIDI or a 

Macro knob. This lets you do tricks like 

changing the sound of the note attacks from 

a MIDI slider in subtle and highly controllable 

ways. 

Additive synthesis. Load four 

Operators into an Instrument Rack, choose 

algorithm 11 (four carriers) for each of them, 

and tune the oscillators to Coarse values of 1, 

2, 3, 4, 5, 6, and so on up to 16. Leave the 

waveforms all set to sine waves. Now you 

have a 16-sine-wave additive synthesizer. 

Give some of the operators envelopes that 

decay quickly to zero, and give others slow 

attacks so that the tone swells. Adjust the 

Level of each oscillator to get a pleasing 

blend. Try detuning some of your additive 

overtones slightly. 

FM with a low-frequency carrier. 

When you click the Fixed button on an 

oscillator, it no longer tracks the keyboard. 

This feature has at least three important uses. 

First, if you tune a modulator to a high, fixed 

frequency (above 2kHz or so) and keep it at a 

low level, it will mimic the sound of a fixedfrequency 

formant filter. Second, percussive 

attack transients, such as the sound of a mallet 

hitting a xylophone, are pretty much fixed 

in frequency no matter what note you play, 

so using a couple of fixed-frequency operators 

to produce mallet attacks works well. 

The third use of fixed frequencies is less 

obvious. If you tune a carrier to a low, fixed 

frequency, perhaps a Freq setting of between 

30 and 70Hz and a Multi (multiplier) knob 

setting of 0.01, it will add a chorused quality 

to the tone of the modulator. This sound is a 

Fig. 4: An Instrument Rack with four Operators 

in parallel. Several parameters have been assigned 

to Macro knob 1, which I’ve renamed Mod Wheel. 

Note the small pink dot next to the Level knob for 

operator C; this indicates that it is now under the 

control of a Macro knob. 

Francis Preve’s Quick Tips 

• When using the filter with FM sounds, program the FM section first, before turning 

on the filter. In other words, use the filter to enhance a sound that’s already good, 

not to fix problems. 

• The Spread knob in the pitch section doubles the CPU load of a patch. If your CPU is 

hitting the wall, turn the Spread down to zero. 

• When starting work on a new FM sound, choose algorithm 8 (two carrier/modulator 

pairs). It’s a good basic choice, and you can later change if you need something special. 

When you add the second modulator to a carrier, it’s easy to end up with too 

many overtones. 

• The pitch envelope can be routed to some oscillators but not others, which makes it 

useful for subtle detuning effects and adding accents in envelope loop patches. 

• Unless you know what you’re doing, do not tune a carrier to any Coarse value 

except 0.5, 1, 2, 4, or 8. Those are the octaves of the fundamental. If you tune a carrier 

to 7 and a modulator to 14, for instance, you’ll get a tone that’s difficult to 

retune to concert pitch even with the aid of the Transpose knob. 

• If you’re using Operator with an arpeggiator, check to make sure the R button is 

switched on. If it’s off, the arpeggio will chew up your polyphony while adding CPU 

load. 

• In an FM patch, apply the mod wheel to one or more modulators’ Coarse tuning 

knobs to create PPG-style stepped wave sweeps. 

• Using two Operators in an Instrument Rack will give you 8-operator FM. When 

doing this, choose complementary algorithms to make the most of the composite 

sound. 

• If you need an extra modulator, use the LFO. Set it to a high frequency range, set its 

Rate

VI review 

Native Instruments FM8 

Review by Lee Sherman 

Native Instruments FM8, $339; 

upgrade from FM7, $119. 

www.NativeInstruments.de 

System requirements: Mac OS X 

10.4, G4 1.4 GHz + or Intel Core 

Duo 1.66 GHz +, 512MB RAM; 

Windows XP, 1.4GHz Pentium or 

Athlon, 512MB RAM. 

Formats: Stand-alone, Audio 

Units, VST, DXi, RTAS. 

Copy protection: online authorization 

using included utility to manage 

all Native Instruments auths. 


The second coming of the DX-7 gets a 

major update 

Native Instruments’ FM7 has become a 

staple of modern music production. In 

a way this is the second coming of the 

famous Yamaha DX-7, which dominated the 

world in the early ’80s when it came out. 

While the FM synthesis technique these 

instruments have in common is capable of 

creating some of the cheesiest sounds ever 

recorded, it’s also capable of creating some 

wonderful, unique sounds. As a result it’s 

never really gone out of style. Among other 

things, FM synthesis can produce a metallic 

sheen that’s fantastic for clanging bells, lush 

pads, glassy electric pianos, and tight basses. 

It’s as welcome as ever in today’s music. 

Recreating the sound of the original hardware 

is a lot easier when the original hardware 

was a digital synthesizer. But what’s really 

important is whether the software has 

improved upon the original. 

Indeed, FM7 not only delivered the sound 

of the DX-7 (you can even import all of the 

old patches—see Random Tip: Loading original 

DX sounds into FM8 [which works the 

same way]), it went some way toward 

improving the arcane process of programming 

FM sounds. We’ll explain FM synthesis 

in the next section, but FM7 also provided an 

advanced sound architecture that includes a 

Fig. 1: FM8 now includes the intuitive 

KoreSounds Browser. This allows you to search for 

sounds and filter them according to instrument, 

genre, synthesis type, and musical attributes such 

as timbre and articulation 

free-form FM matrix, additional waveforms for 

the Operators (the original only had sine 

waves), graphical envelopes with nearly 

unlimited stages, host-syncable looping, and 

chorus and delay effects. 

While FM8’s considerably revamped user 

interface no longer resembles the panel of a 

DX-7 (it’s now a soothing gray-on-white color 

scheme), the biggest changes are due to the 

software’s integration with Native 

Instruments’ Kore software 

host/controller/interface. FM8’s own patch 

brower is similar to Kore’s, and its sounds use 

the Kore Single Sound Format. 

If you do open FM8 within Kore, you can 

use complex layers of FM8 and Native 

Instruments programs. KoreSounds can take 

advantage of the new morphing technology 

and multi-effects chains, making for some 

truly massive sounds. 

A tabbed navigator along the right side of 

the software’s main window lets you switch

quickly between the Browser and Attributes 

view, expert and novice programming modes, 

access the new arpeggiator, or bring up the 

effects rack. The Attributes view lets you filter 

your sounds according to instrument, genre, 

synthesis type, and musical attributes like timbre 

and articulation. 

Like all NI instruments, FM8 runs standalone 

or as a plug-in on both Macs and 

Windows XP machines in all the standard formats. 

It is now a universal binary, meaning it 

works on Intel-based Macs. 

Smooth operators 

In FM synthesis, sounds are created by 

modulating one waveform with another, producing 

a more complicated waveform. 

Waveform generators are called Operators; an 

Operator that’s modulating another one is 

called a Modulator, and one that’s being producing 

actual sound is called a Carrier. Any 

Operator can act as either a Carrier or a 

Modulator, and it is the arrangements of 

these waveforms that form the building 

blocks of FM synthesis. 

In FM8, up to eight Operators can be freely 

mixed in any combination, and you can 

choose from a wide variety of waveforms— 

unlike the DX-7, whose Operators produced 

only sine waves and were arranged into 32 

set configurations (called Algorithms). Other 

DX-series instruments had four Operators. 

The last two FM8 Operators are special and 

aren’t found in Yamaha’s implementation of 

FM synthesis. Operator X adds noise and 

waveshaping, while operator Z is an analogstyle 

multimode filter with a filter envelope. 

Also unique to NI’s version of FM are multistage 

envelopes that can be edited graphically. 

The next step would be the ability to 

import your own samples as you can on the 

Yamaha SY77/99; perhaps that will come in 

FM9? 

Instead of only allowing you to view a single 

Operator at a time as you did in FM7, 

FM8’s Expert window now provides an at-a- 

Fig. 2: FM8’s new Expert window provides an ata-glance 

view of Operator parameters. 

glance view of the 

key parameters for 

all Eight Operators 

all at once. 

Alternatively, you 

can choose to drill 

down into the 

more advanced 

parameters for 

individual 

Operators by clicking 

on labeled 

buttons in the 

navigator or the 

FM matrix. The 

FM Matrix provides 

an intuitive 

view of operator 

routing. 

If you’re not interested in dealing with the 

complexities of FM, an Easy/Morph page can 

be used to tweak an overall sound quickly. 

Here you can route LFOs, adjust the amplitude 

envelope, apply effects, and more. 

FM7 was limited to a pair of fairly standard 

effects: chorus and delay; FM8 now has a full 

effects rack that includes 11 high-quality 

effects including talk, wah, phaser and 

flanger, tremolo, reverb, pitch-able delay, 

tube amp, overdrive, and EQs seemingly 

drawn from NI’s Guitar Rig. Why should guitarists 

have all the fun? 

The smooth workflow and visual approach 

to parameter editing remains a huge advantage 

over the old hardware, while the highresolution 

audio engine in FM8 sounds even 

better than its predecessor and far better than 

a real DX7 (due to better D/A convertors, 

oversampling, multi-stage EGs, and the builtin 

effects). 

Motion sickness 

Since it takes some work to master FM programming, 

FM8’s sound morphing features 

are particularly welcome as a serendipitous 

means of coming up with new sounds. The 

sound-morphing feature provides a 4-way 

matrix onto which you drag and drop sounds 

from the browser. 

By clicking and dragging across the matrix, 

you can 

morph your 

current sound 

with the timbres 

of the 

additional 

sounds. (In 

order to 

ensure that 

the original 

sound is still 

playable, only 

the correct 

parameters 

morph.) A 

Randomize 

button generates 

new 

VI review 

Fig. 3: The new arpeggiator looks and acts like 

an old-school analog sequencer, bringing life to 

sounds 

sounds for use in the matrix. The 4-way 

matrix is reminiscent of waveshaping synths 

like the Sequential Prophet VS and the Korg 

VS software but much easier to program. It 

made me wish for a joystick to control it with. 

Fortunately all movements can be automated. 

Additional movement comes from the programmable 

arpeggiator, with its 32-step 

matrix that’s reminiscent of a classic analog 

step-sequencer. It’s the arpeggiator that 

accounts for the many on-board patterns 

you’ll find in the browser. DJs, dance producers, 

and musicians of all kinds will find new 

inspiration from the groove presets. 

One of the great things about FM7 has 

always been the wide availability of additional 

presets. You could purchase one of NIs sound 

packages, choose from hundreds of FM7 presets 

available online, or import patches from 

the DX-7 family (most of which are readily 

available on the Internet). 

FM8 ships with 960 preset sounds for all 

styles and genres, including both original 

FM7 patches, the complete FM7 Sounds Vol. 

1 and Vol. 2 libraries, as well as 200 new 

sounds—enough programs that you may not 

ever need to know what an Operator is. Plus 

there’s the above-mentioned KoreSound format. 

Is eight enough? 

At a time when most new virtual instruments 

seem to play tribute either directly or 

indirectly to their hardware forbears, its gratifying 

to see Native Instruments extend the 

DX-7 legacy. FM8 sounds better, it’s easier to 

program (the interface improvements alone 

are worth the upgrade price), and its sound 

morphing and on-board arpeggiator go far 

for updating digital FM synthesis for today’s 

music. VI 


VI feature 

A breakthrough solution to the problem that’s 

plagued sample library users for the past few 

years, and an explanation for beginners about 

why it’s been a problem. 

The great thing about modern sample libraries is 

that they’re huge; the problem with modern sample 


7 Gigs Loaded 

on One Mac 

libraries is that they’re huge. 

Being huge means they require lots of computer 

memory to run, even if they stream samples off 

hard disks (because they still need a head-start 

memory buffer). We’re on the verge of getting new 

64-bit operating systems from Apple and Microsoft 

that will allow virtually unlimited memory access— 

unlike today’s 32-bit operating systems. 


Meanwhile, for Mac users it turns out 

that an old solution to the problem 

that never worked reliably now does 

work: running stand-alone software samplers 

on the same machine that’s running a DAW. 

This allows each program to have its own 

memory allotment, solving the problem 

before all our software gets adapted to work 

on 64-bit operating systems. 

But first let’s start with some background. 

Those of you who just want to find out how 

to access 7GB of RAM on a G5 or Mac Pro 

might want to skim over the next couple of 

sections. 

Why RAM is your friend 

Having a healthy percentage of the samples 

and programs in these libraries cued up 

and ready to play is the name of the game, 

and it’s always required way more RAM (random 

access memory) than any single computer 

can access. For a good percentage of 

our subscribers—about a third of you, to be 

precise—the solution is to use multiple computers. 

The reason for needing all these samples 

loaded is simple: you want to them ready to 

play even though you won’t actually use all of 

them in a given piece of music; it would be

Fig. 1: 7GB on one machine, and it’s running reliably. This Activity Monitor screen dump shows 6.96GB of 

memory access in a G5 with 8GB loaded. The VSL-Server program is listing samples loaded into the plug-in 

versions of the VSL Vienna Instruments player running inside Logic Pro, which is accessing 1.38GB. (Logic is 

also running other programs, including Spectrasonics Stylus RMX.) 

Four stand-alone copies of the Vienna Instruments player (all named differently, which you don’t see) are 

running outside Logic. In addition, Native Instruments Kore is running several instruments, including 

Toontrack’s EZ-Drummer (a sample-playing instrument). 

While the sequence isn’t playing here, you can see from the idling percentages that the CPU hit isn’t going 

to be outrageous. That’s what’s changed recently to make this possible. 

deadly to have to stop and reload programs 

every time you wanted to, say, switch violin 

bow directions. This is no different from an 

artist sitting down to draw with a box of pastels 

containing dozens of colors; he or she 

certainly won’t use them all in a single picture, 

but who wants to go fishing in closets 

for a lighter shade of green. 

Now, the first logical question is why you 

can’t just install, say, 32 Gigabytes (GB) of 

RAM in one computer—more than you’re 

likely to need—and be done with it. The 

answer is that there’s a finite number of 

memory locations our present 32-bit operating 

systems can address; they only have 

enough bits to map out 4GB of RAM. That 

translates to a real-world maximum of 3GB or 

even slightly less. 

Getting even more real-world, somewhere 

around 1.25GB of RAM is about all you can 

access in a stock Windows XP machine with 

2GB installed. However, on p. 26 of our 6- 

7/05 issue, Mattias Henningson described 

how to access almost 3GB of RAM using “The 

3GB Switch” tweak. Before the 3GB switch 

there was no reason to install more than 2GB 

in a Windows machine; considering that the 

tweak more than doubles a single machine’s 

RAM access—potentially standing in for two 

machines—that’s quite a leap forward. 

Another limitation is that Windows itself 

can’t recognize more than 3GB installed in a 

computer. However, PowerMac G5s under OS 

X 10.3 or higher can recognize up to 8GB of 

installed RAM, depending on the model 

(some G5s only have slots for 4GB). The new 

Intel Mac Pros can hold up to 16GB of RAM. 

But note that there’s a difference between 

recognizing the installed RAM and actually 

VI 

feature 

being able to access it for loading samples. 

While Mac OS X will hold and recognize up 

to 8GB of RAM in a PowerMac G5 or 16GB in 

an Intel-based Mac Pro, its 32-bit operating 

system “only” allows each program to access 

a real-world maximum of about 3GB. 

Part of the reason that only 3GB of the theoretical 

4GB allowed by 32-bit OS X comes to 

work, as we explained in the June/July issue, is 

that system frameworks and libraries (extra 

code used by programs) take some of it. Then 

OS X itself will use about .5GB outside any 

running programs, plus it needs some RAM 

for caching. Without that RAM for caching 

the system will go to virtual memory—meaning 

it will pretend the hard disk is RAM—and 

that’s not good for our real-time applications. 

However… 

That’s 3GB of RAM access for each 

program, though. Unlike Windows, two 

loaded-to-the-gills programs in Mac OS X can 

theoretically access a combined 6GB…except 

that things get unstable when two programs 

are that full. 

In practice—before the solution offered in 

this article—the previous functional loading 

record for a G5 with 8GB of RAM installed 

was between 5 and 5.5GB. That record was 

set running Apple Logic Pro with about 3GB 

plus an additional 2GB in the VSL Vienna 

Instruments player plug-in, which behaves 

like an external program running outside the 

host DAW and has its own memory space. 

(Normally instrument plug-ins’ memory 

comes out of the host DAW’s RAM budget.) 

Now there’s a way to push that concept 

farther. Mac users can take advantage of OS 

X’ ability to allot each program its own memory 

space to run—are you ready for this— 

about 7GB of RAM in a G5 with 8GB 

installed. You can actually load a little more 

than that, but again, things become unreliable. 

This may well work on Intel Macs that have 

up to 16GB of the (very expensive) RAM they 

use installed. We haven’t had a chance to test 

that yet, and not all the software is compati- 

Fig. 2: It’s necessary to make renamed copies 

when you’re running multiple instances of the same 

program. This is handled automatically by the Mac 

OS. 


VI feature 

Fig. 3: The IAC Driver in Audio MIDI Setup. 

Make sure you have enough ports to send MIDI to 

all the stand-alone programs running outside your 

DAW. 

ble as of this writing anyway, but it should be 

possible. 

The only caveat is that you also have to 

keep an eye on the processing power. If you 

run too many programs, you can easily run 

out—in fact that was the factor that prevented 

this trick from working in the past; some 

of the latest programs have quietly become 

more efficient when running outside a DAW. 

By the time you read this, the 64-bit 

Windows Vista will probably be out, and the 

full 64-bit Mac OS X will be imminent. 

Hopefully it won’t take too long for all the 

music software developers to come out with 

versions that feature 64-bit memory access. 

And hopefully the RAM used in the newest 

computers such as the Mac Pro won’t be as 

expensive as it is right now (about $250 per 

GB as of this writing, vs. $100 per GB for the 

PC3200 memory in the G5 in this article— 

which is up from $75 at the time we filled it 

with 8GB). 

Meanwhile, 7GB of RAM access (as shown 

in Fig. 1) is far from shabby. What’s more, 

running samplers outside the DAW can actually 

be advantageous: you don’t have to wait 

for them to load when you fire up a new cue 

in your sequencer. 


Loading 7 

The 7GB trick isn’t complicated: load your 

DAW up as usual, then launch stand-alone 

versions of the sample-playing programs 

you’re going to run outside your DAW. We 

tested the trick with Native Instruments 

Kontakt- and Kontakt 2-family instruments 

(which includes Kontakt 2 Player and also 

embedded libraries such as East West 

Colossus); with the VSL Vienna Instruments 

player; and with Synthogy Ivory Grand. 

All of these instruments have stand-alone 

and plug-in versions, and you can run both of 

them simultaneously. What’s more, you can 

run multiple instances of a stand-alone sample 

player if you make duplicates of the program 

and give each a different name. Just 

highlight the program icon in the Finder and 

hit command/D as shown in Fig. 2; the copy 

will be called “copy of” whatever it is. 

You can also run stand-alone hosts outside 

the DAW. We were able to run Native 

Instruments Kore outside Logic Pro with pretty 

good stability. The one fly in the ointment 

is that not every program is happy when you 

run stand-alone and plug-in versions at the 

same time, or when you run plug-ins in two 

hosts; you’ll have to experiment with what 

gets loaded where. 

This trick seems to work best if you load 

the DAW first, then the stand-alone programs—although 

we’re going to stop short of 

making that a blanket statement, since we 

haven’t tested every combination of software 

on the planet. Anecdotally, the system also 

feels more reliable with a greater number of 

stand-alone program instances, each loaded 

with a moderate amount, than to run few 

instances all loaded to the gills. 

MIDI 

The way you send the stand-alones MIDI is 

to use IAC, the inter-application communication 

feature built into OS X’ Core MIDI. If 

necessary, go into Audio MIDI Setup program 

Fig. 4: Enabling IAC bus 1 in this Kontakt 2 

Player (running SampleLogic AIR).

and click on the IAC driver. Make 

sure it has enough 16-channel ports 

enabled, as shown in Fig. 3. 

The IAC driver is just another 

MIDI interface as far as all parties 

are concerned. Simply assign the 

appropriate IAC port and channel as 

the output of your sequencer track 

and the input of your stand-alone 

instrument, as shown in Figs. 4 and 

4A. Kontakt 2 and its relatives 

receive 64 MIDI channels, so you 

can just keep creating IAC busses 

until you have enough. 

Audio 

There are a few ways to route the 

audio from the stand-alone programs 

into your DAW’s mixer. 

Which one you choose depends on 

the stand-alone program and on 

your audio interface(s). 

RME’s interfaces have a “loopback” 

mode for their onboard mixers 

that let you route software outputs 

back to inputs. Apogee is about to introduce 

the same feature with the Maestro software 

for their Symphony cards. The hardware 

does the routing. 

In our last issue we had a Very Deep Clinic 

on Soundflower (free download from 

http://www.cycling74.com), an inter-application 

audio routing utility. That works very 

well; simply select a Soundflower input as the 

output for the stand-alone and the input to 

your DAW’s mixer. 

You’ll want to create an aggregate device 

Fig. 4a: Just assign each program its own MIDI 

channel as you would with a physical MIDI interface. 

consisting of Soundflower and your audio 

interface(s) in Audio MIDI Setup. Aggregate 

devices link multiple audio drivers so they 

appear to software as one big interface with 

lots of inputs and outputs. Fig. 5 shows a 

slightly over the top Aggregate device using 

MOTU PCI-424 hardware, an RME Fireface 

400, the Mac’s built-in stereo optical digital 

I/O, and 16 Soundflower I/Os. 

You can also just make a physical connection 

between outputs and inputs on your 

audio interface. That can be a sacrifice if you 

don’t have a lot of I/Os, but it’s a perfectly 

workable solution if you have extras. 

Connecting a TOSlink cable between the 

Mac’s digital output and input would give 

you a stereo path into your DAW’s mixer. 

VI feature 

Fig. 5: Creating an Aggregate Device so that 

Soundflower becomes a virtual audio interface. This 

is one way of looping the stand-alone instruments’ 

audio back into your DAW. 

Sometimes it’s necessary to use more than 

one driver. For example, the stand-alone 

Vienna Instruments and Ivory Grand players 

don’t let you select individual outputs—they 

just uses the default pair for the driver it’s 

using. So if you’re running more than one 

instance and you want separate outputs into 

your DAW, you’ll need to use more than one 

driver. Soundflower + the built-in audio would 

be one solution. 

On the other hand, the Native Instruments 

stand-alones can be directed to any or all outputs 

of the selected sound driver. Note that 

all programs can use whatever driver (including 

aggregate ones) they want. 

Alignment of the stars 

One final disclaimer: we’ve tested a few 

combinations of programs and come to the 

conclusion that 7GB is a reasonable estimate 

of the amount of RAM you’ll be able to 

access. Please don’t sic your attorney on us if 

it turns out you can only work with 6.75GB 

on your rig—although you might also be able 

to get better results. 

Bear in mind that there are many combinations 

of hosts and instruments, all of which 

may or may not perform well together. Also 

bear in mind that these are computers; we’re 

dealing with the occult. VI 


VI review 

Review by Nick Batzdorf 

RME Fireface 400, $1199 

www.RME.com, distributed by 

Synthax (www.Synthax.com) 


RME Fireface 400 

FireWire Audio Interface 

This little half-rack box houses a 24-bit/192kHz 

audio interface with 18 ins and outs. But that’s 

just the beginning, because it can also route 

software tracks all around the room. 

There’s not much to say about most 

audio interfaces. They have a certain 

number of ins and outs, hopefully they 

sound good and work reliably, and some of 

them have no-latency monitoring features. 

And some—not the RME Fireface 400 being 

reviewed—come with nice light-version software 

bundles, an important a consideration if 

you’re first getting started. 

The RME Fireface 400, on the other hand, 

is a higher-end FireWire interface than most 

people will start out with, and it has a lot to 

talk about. There’s some pretty serious analog 

and digital engineering packed into this little 

half-rack box; while it doesn’t have a light 

software bundle, it comes with some pretty 

sophisticated Windows and Mac OS X mix- 

ing/routing/metering software to control it. 

It also has a few features and characteristics 

that make it function exceptionally well in 

V.I.-based set-ups. 

The ins and outs 

RME calls the Fireface 400 a 36-channel 

interface, and indeed it has 18 ins and outs: 

eight analog I/Os, 8-channel lightpipe in and 

out, and stereo S/PDIF—plus 2 x 2 MIDI. But 

calling it an 18 x 18 interface doesn’t tell the 

story of its onboard no-latency mixing and 

routing features, because you can also route 

18 software outputs to any or all physical outputs 

at the same time the 18 inputs are getting 

routed to the software. 

What’s more, it has a Loopback mode that

lets you route the software signal assigned to 

an output back to same-numbered input 

(where it can in turn be routed further). This 

allows you to send the output of one software 

program—such as a stand-alone sampler— 

into another one—most likely your main 

DAW—without cables. That’s the first feature 

that could be very useful for V.I. rigs. 

The interface can operate at up to 24bits/192kHz…or 

to be precise it can operate 

at anything between 27kHz and 200kHz due 

to the way it’s engineered. Quad speed sample 

rates (192kHz, etc.) are sent out the 

S/PDIF port—which is also an AES/EBU port— 

and the Fireface supports the S/MUX protocol 

for transferring four channels at 96kHz over 

the ADAT connections. 

All the Fireface 400’s analog I/Os are balanced 

1/4” TRS, with two exceptions: front 

panel mic/line/instrument inputs 1 and 2 are 

on Neutrik combination XLR/TRS jacks; and 

outputs 7 and 8, also on the front panel, can 

Fig. 2: RME’s TotalMix software front end to the 

box’ onboard digital mixer. The top row is the 

physical inputs, the middle row is the software 

tracks’ outputs, and the bottom row is the physical 

outputs. Outputs 3&4 are in Loopback mode, 

which routes their signal to inputs 3&4 so you can 

record or monitor them inside your DAW. Among 

other applications, this allows you to record and 

monitor the output of one software program into 

another—for example if you’re running a standalone 

software sampler outside your DAW. 

Fig. 1: The Fireface Settings control 

panel. Note the AutoSync clock mode, 

which automatically locks to an incoming 

digital signal. Combined with the 

unit’s SteadyClock circuitry, which 

removes the jitter from incoming clock 

so what goes out is very clean. This 

effectively lets you run your whole rig 

from any input without having to worry 

about its quality. 

drive headphones or line-level outputs. 

Front panel inputs 3 and 4 

can accept line or instrument-level 

inputs; all the other analog I/Os are 

line-level jacks on the rear panel. 

400 vs. 800 

The 400 has much in common 

with RME’s Fireface 800, which has 

been out for a while. Apart from 

the 800’s full 1U size, the main differences 

are that the 800 has an 

extra 8-channel lightpipe I/O; it 

isn’t bus-powered like the 400; it 

has more mic inputs (but still eight 

simultaneously active analog ins); it 

can take advantage of the faster 

FireWire 800 bus; and its line inputs 

VI 

review 

have lower noise and distortion specs. 

Those noise specs come with an asterisk, 

however, because one of the first things you 

notice about the 400’s software is that the 

input level readouts are bouncing around at 

about -109dB with nothing connected. 

Translation for the less nerdy: it’s quiet. 

One spec that does make a difference is 

that the 400’s mic inputs have 65dB of gain 

vs. the 800’s scant 50dB. That extra 15dB of 

gain means you can use the 400 comfortably 

with low-output dynamic and especially ribbon 

mics. 

In case you were wondering whether the 

bus provides enough power for mic inputs to 

sound good, that’s not a concern. For one, it 

comes with a switching power supply, so you 

don’t have to use the bus. But FireWire has 

no problem delivering upwards of 25 volts, 

and these mic preamps sound quite transparent 

and solid; I heard no difference running 

the unit bus-powered or plugged in. As a 

matter of fact, the Fireface gets enough juice 

from the bus to get warm to the touch. 

The Fireface 400 is a very nice-sounding 

interface. You really do hear an obvious sound 

quality improvement over garden variety lowerpriced 

units. The difference is especially noticeable 

when you play something like acoustic guitar 

recordings through it—the sound becomes 

detailed and solid rather than harsh. 


VI review 

Fig. 3: In addition to being passed directly to the 

host oomputer, any input can be assigned to any 

or all outputs. That applies to physical inputs and 

“returns” from software tracks. The TotalMix 

Matrix, with inputs on the horizontal axis and outputs 

on the vertical one, shows this (along with the 

level of each bus). 

If you run the 400 at 4x sample rates 

(around 192kHz), the ADAT lightpipe inputs 

disappear, since they max out at 2x (around 

96kHz). It’s hard to run V.I.-intensive sessions 

at anything other than 1x (441.kHz or 48kHz) 

sample rates anyway, so that’s probably not a 

consideration for our purposes; regardless, 

the word clock output can be set to single 

speed when the interface is running at 2x or 

4x speed. 

That means you can continue to use the 

Fireface as a master clock for your whole rig if 

you’re running slave computers and other 

devices at 44.1 or 48kHz. 

Digital 

The Fireface 400 has an important feature 

called SteadyClock that in addition to locking 

up quickly, removes essentially all the jitter 

from incoming digital signals. That means the 

unit should actually sound the same whether 

you clock it with a dedicated diamond-studded 

word clock generator that costs more 

than the unit itself or from the biggest piece 

of digital garbage you can find. 

To see whether SteadyClock really is all 

that, I burned some music onto a CD and 

locked the Fireface 400 to an old and cheap 


CD player’s optical S/PDIF digital output. If 

the CD player wasn’t jittery to start with, optical 

connections—whether S/PDIF or ADAT 

lightpipe—are fine for carrying audio, but 

notoriously jittery as clock sources. That’s 

especially true over longer runs, so I used a 

12’ lightpipe cable. 

Lo and behold, I really couldn’t hear a difference 

between the music coming down 

that lightpipe and the hard disk file with the 

Fireface running under internal sync. That was 

true when monitoring through the Fireface 

and when monitoring through my usual setup, 

which means that the clock leaving the 

Fireface is equally clean either way. RME takes 

this farther with an AutoSync feature that 

locks to an incoming digital signal when you 

turn it on. 

This digital clocking scheme can save a lot 

of switching around in set-ups with multiple 

computers or devices that have digital outs 

but no digital input to clock 

to. It’s really well thought out 

and implemented. 

Software 

You can operate the Fireface 

400 stand-alone without a 

computer. Its last-saved routings 

and functions are active; it 

can convert formats, act as a 

mic preamp, send digital clock, 

and so on. All the settings you 

choose in the Fireface Settings 

utility (for Mac or Windows; 

see Fig. 1) are remembered. 

The other included program, 

the Fireface Mixer (Figs. 

2 and 3), controls the 400’s 

TotalMix onboard digital 

mixer/router. When everything 

is being displayed, the top row 

has faders for all physical 

inputs (which you can name) 

and the middle row is for up 

to 18 software channels headed 

for the 18 physical outputs. 

These two rows can be 

assigned to any or all of the 18 

outputs, which is what the 

bottom row of faders control. 

You can group and pair 

faders, and they’re scaled. So if 

you move one fader in a pair or group, the 

others—which can be at different levels— 

change by the same percentage rather than 

the same number of dB. While this mixer 

doesn’t have dynamic automation built in, it 

can be controlled by MIDI using standard 

Mackie Control protocol and automated 

externally. It also has eight factory and eight 

user preset snapshots, recallable by MIDI program 

changes. 

The unit should sound the same whether you 

clock it with a dedicated diamond-studded word 

clock generator that costs more than the unit itself 

or from the biggest piece of digital garbage you 

can find. 

TotalMix’ interface is slightly different from 

the way most mixers work. Each fader on the 

top two rows can only control/display the 

level going to one output pair at a time, 

selected from a pop-up button. So it’s possible 

for Input 1 to be showing its level going 

to Outputs 3 and 4, Input 2 to be showing its 

level to Outputs 1 and 2, and so on. 

That’s what you want when inputs are simply 

routed to different places; once the routings 

are set up, all you’re concerned with is the

level balance. But this can be confusing when 

channels are routed to multiple destinations, so 

TotalMix has a Submix mode (see Fig. 4) in 

which all the input/software output fader displays 

move to the currently selected output. 

That lets you adjust the total balance going to 

your main monitors, or headphones in a 

recording situation, and so on. (Submix mode 

parallels what MOTU does in CueMix, the software 

that controls its built-in digital mixers.) 

Now, rather than providing 17 aux send 

knobs plus the main fader for each channel, 

RME created a convenient routing matrix (see 

Fig. 3). This overview screen makes it easy to 

assign inputs to multiple outputs and see 

their levels right away. 

The big advantage to having a digital 

mixer built into audio hardware is that it provides 

“no latency” monitoring, avoiding the 

few milliseconds of delay a signal incurs going 

through a computer. (“No latency” means 

the delay is defined by the digital converters, 

and it’s a couple of milliseconds like on a digital 

mixer.) 

That means you’re using the software 

mixer in your DAW and also adding a second 

one built into the audio hardware to control 

the routing around your studio. Switching 

between software and hardware monitoring 

to avoid latency is somewhat of a nuisance on 

Macs—although the TotalMix preset snapshots 

make this much easier than it could be. 

The Windows version of TotalMix supports 

ASIO Direct Monitoring, in which the switching 

is handled by “remote control” from the 

DAW. Given that lots of interfaces have no 

latency monitoring these days, it would be 

great if Apple incorporated a similar standard 

in its systemwide Core Audio driver. 

The Fireface 400 also comes with a DIGIcheck, 

a Windows-only audio analysis program. 

DIGI-check displays levels, information 

about the phase and spectrum, and bit statistics 

and noise. All this is calculated in the 

unit’s hardware—as is all the metering in 

TotalMix, on Mac as well as Windows. 

TotalMix’ meters are very useful, showing 

both average and peak levels, plus they have 

a numeric readout. 

Performance 

PCI slots provide a more direct path to the 

computer, so all things being equal, FireWire 

is always going to have a little more latency. 

Furthermore, on its PCI systems, RME is able 

to use hardware to help stream audio in both 

directions, reducing the CPU load. This is 

explained in the impeccably detailed manual, 

which has a lot of interesting technical information. 

However, the performance of this FireWire 

interface is still very good. At a 128 buffer—at 

which the latency is subjectively not an issue 

when you’re playing V.I.s (at least for me)— 

VI 

review 

the test dual 2.5GHz G5 registered 35% of 

one CPU when playing a stereo audio track. 

By comparison, the MOTU PCI-424 card in 

this machine registered 30% at the same 

buffer setting. That’s not a significant difference. 

Furthermore, FireWire has the advantage of 

being more obsolescence-proof than internal 

computer cards. With PCs you can swap out 

motherboards for one with the PCI slots you 

need, but you can’t do that with Macs. Apple 

Computer has gone through several different 

PCI varieties just in the past three or four 

years—and this isn’t the first card change 

we’ve seen. While each generation has legitimate 

technical improvements over the previous 

one, a lot of musicians express frustration 

over having perfectly good hardware rendered 

“legacy” so quickly. 

It also should go without saying that the 

bus-powered Fireface 400 is intended to work 

well as a portable recording unit. 

In your Fireface 

This is a very well-engineered, good sounding 

interface with a lot of features that make 

it work especially well in virtual instruments 

rigs. It costs more than an entry-level interface, 

but you really do get more. VI 


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VI feature 

Sequencing 

Without doubt the most 

common beginner 

mistake is lack of 

attention to natural 

balance among the 

instruments at play. This 

is a problem even in 

experienced peoples’ 

mockups, and it’s the 

result of an improperly 

balanced orchestral 


Part 1 in our new series on the art of MIDI 

programming: balance 

template. 

by Thomas J. Bergersen 

Samples 

The problem often starts at the sample 

library level (lack of proper/natural 

dynamic relations between the various 

layers, particularly often in the case of crossfading 

patches) and works its way into compositions 

through peoples’ careless handling 

of MIDI CC#11 (expression) or CC#1 (mod 

wheel) data. 

In this article I’ll offer a few pointers that 

can help you out if you’re fighting the 

headaches of improper volume relations 

between your orchestral instrument samples. 

Basics 

All commercially available libraries have 

normalized sample pools (meaning that every 

instrument, from solo bass flute to four horns 

in unison, is set so its loudest point is as loud 

as possible). So the first step is to make sure 

your template is balanced properly. This is a 

rather long and tedious process, since you 

need to rebuild the natural volume relations 

between the patches. 

I’ve split this process into two phases: 

1. Find the proper relative balance between 

each articulation of an instrument in your 

template. Essentially this means you should 

find the proper volume relations between 

your staccs, sustains, runs, tremolos, trills, etc. 

Start with the loudest articulation (top layer 

of aggressive staccs, or sustains for strings) 

and measure the volume of your other articulations 

against this. Flautando patches, sul 

ponticelli, sul tasto etc. will all be relatively 

quiet in the real world, so make sure these 

patches are lower in volume. Just how much 

you need to lower them is hard to say, but 

use your ears— the mids/low mids are generally 

good indications. If you go too loud, the 

mids become too prominent and balance 

between the other articulations is lost. Do this 

for each instrument separately. 

2. Now it’s time to find proper balance 

between the various instrument groups. At 

this stage I like to have a classical recording 

handy, just to benchmark against. The recording 

you’re benchmarking against has to be 

recorded with a simple A/B mic setup (no 

spot mics—simple stereo pair over the conductor) 

and the piece has to be dynamic. It 

needs a triple forte tutti passage, since this is 

how you find the maximum sound level for 

each instrument group. I recommend any 

recording of Gustav Mahler’s 5th from the 

1990s. 

Begin with the first instrument in your 

score, the piccolo flute. Find a part in your 

benchmark recording where it’s playing at full 

blast. Approximate the level in your 

sequencer/orchestral template and make sure 

all the articulations of this instrument retain 

their relative balance. 

Configure any pan data too at this point if 

you want to. Move on to the next. Repeat 

until you’ve done this for all the instruments 

in your template. This could take a few days, 

but it is well worth it. 

If you’re meticulous in your approach, the 

end result will be an almost properly balanced 

orchestral template that is the basic starting 

point. 

Advanced 

Adding to the long list of what I consider to 

be problems with commercial sample libraries, 

we have the normalization issue, which effectively 

kills the natural relation between dynamic 

layers. A good example is the flute, which in 

the real world is an instrument of great dynamic 

variation through its register; you can’t play 

loud in its lower register and you can’t play soft 

in its higher register. 

Due to ruthless normalization, the flute’s 

low range is as loud (in dB) as its high range,

the horns’ lower range is as powerful as its 

mid/high range, and so on. It’s not possible 

to reconstruct the proper natural relations 

unless you know all the instruments very well. 

So if you really want to create a properly 

balanced orchestral template, you’ll have to 

hire a musician for each instrument and 

record them all performing a controlled chromatic 

scale at ppp, mf and fff! Then you analyze 

the sound wave and find the approximate 

dB differentials between notes in each 

part of the instrument’s various registers. 

Finally, you transfer these relations to your 

orchestral library—not exactly a cakewalk, but 

truth is without proper balance, even within 

an instrument itself, you can never go by traditional 

orchestration rules or guidelines. 

There are too many discrepancies between 

real orchestras and today’s orchestral libraries. 

Another technique I use when dealing with 

samples of ensembles in homogenously 

instrumented harmonies (such as a string 

triad comprised of three notes with a 12-violin 

ensemble sustains patch) in order to avoid 

a buildup in volume is to use CC#11 to attenuate 

the output as more and more voices are 

added. 

Fig. 1 is a chart based on my own custom 

samples (your results may vary), and these 

have proven themselves pretty accurate with 

TASCAM GigaStudio’s linear handling of 

CC#11. Again, your results may differ 

depending on your library and the instrument 

in question. 

But in general I would lower the volume 

from 127 to around 100-110 when playing 

triads with a sustain patch. That’ll make sure 

you retain some of the balance in the orchestra. 

Orchestra sizes and MIDI 

There are distinct disadvantages and 

advantages to working with orchestral samples. 

The obvious advantages are the cost, 

perfection (tuning and intonation among 

other things), and flexibility. 

When you’re working with sample libraries 

that have large sections of instruments, such 

as 12 violins or 10 celli, you are stuck with 

these sizes. The act of dividing these groups 

into smaller sections (called divisi, or div. (It.)) 

is impossible, and thus you are stuck with this 

size on single notes. 

Here the advantage is that you can easily 

achieve a huge sound without much effort. A 

huge soaring string line in 1st, 2nd violins, 

and violas is easy to achieve with single-note 

sustains for each section. A triad will sound 

like 36 violins and a 4-part chord will sound 

like 48 violins. 

Or at least this is the general consensus. 

Theoretically speaking this practice is obviously 

wrong, especially when you consider the 

improper balance that is obtained throughout 

the orchestra once you accumulate all these 

instrument groups. 

In reality, however, the result is a bit different. 

While the effect certainly is something to 

be on the lookout for, it is usually not something 

people will perceive as “a bad thing.” If 

overdone, it can lead to the dreaded “organ” 

effect” so much of the frequency range is 

occupied with sounds of similar timbre and 

character that the sense of space between the 

colors is lost. This 

is akin to the effect a painter experiences 

when he or she mixes colors carelessly, resulting 

in a brown mess. This is a typical result 

when layering different samples from various 

libraries. Vibrato is partially to completely lost, 

and it becomes akin to a thick synth pad. 

There are ways to avoid this problem if 

you’re using sample libraries that offer wide 

selections of solo instruments, or sampled 

divisi strings. In the case of solo instrument 

samples, it involves layering multiple tracks of 

various similar articulations to form the 

desired section size. 

The upside is that you have complete control 

over each individual musician’s perform- 

All commercially available libraries 

have normalized sample pools, so the first 

step is to rebuild the natural relationships 

between the patches. 

ance; the downside is that this is a lot of 

work. It becomes more difficult to keep track 

of your work, and the playing techniques, 

intonation, recording techniques, and resonance 

issues all become very apparent very 

quickly. In the case of pre-sampled divisi samples 

you’re obviously going to enjoy a more 

comfortable work procedure. 

Brass instrument sizes in libraries are usually 

within reason: 3-4 trumpets, 3-4 trombones, 

and 4-8 horns. It is common that the library 

also offers solo instruments for each section, 

and obviously a tuba. Some specialized sample 

libraries offer a piccolo trumpet, Wagner 

tuben, euphoniums, Bb and C trumpets, 

Viennese horns, bass trombones, and even 

contrabass trombones. There are certain 

libraries that even offer horns in different section 

sizes from pairs of 2 to 6 in unison. 

With brass I advise you to use solo instruments 

in harmonies of three notes or more. A 

certain synthetic sound becomes increasingly 

audible with larger harmonies consisting of 

two or more instruments in unison per sample. 

The exception is horns: their mellow tone 

lends itself better towards large sections in 

VI feature 

1 voice: CC#11 - value 127 

2 voices: CC#11 - value 114 




Fig. 1: As you add voices, lower the volume 

using MIDI CC#11. This chart was put together for 

TJ’s custom library, but it’s pretty close for all the 

commercially available ones. 

complex harmonies. It’s not uncommon to 

see two horns per note in a triad (comprising 

a 6-horn section), although the traditional use 

of the horn section (consisting of four horns) 

is one horn on each note in a 4-part chord. 

Some people prefer a large horn section, 

others like the space that a more modest 

horn section gives. Experiment with chords 

using both section and solo samples, but 

keep in mind that there are different rules in 

the digital orchestration domain. 

Sampled woodwind instruments are difficult 

to deal with. If you have too many instruments 

of the same timbre playing in unison, 

the personal character of that instrument is 

transformed into a more authoritative but less 

charming by-product. I really recommend 

solo woodwind samples in place of ensemble 

samples, especially in the digital domain 

where balance can be corrected very easily 

using volume control. 

Experiment both with solo and ensemble 

samples of the same instrument and find out 

what you like the most. A solo instrument will 

provide more nuances and a higher level of 

detail to your orchestration, while ensembles 

provide a slightly more sterile yet smooth 

edge—flute ensemble doubling violins for 

example. 

Woodwinds have a tendency to blend better 

in the real world than with samples. This is 

often because the intonation and tuning in 

orchestral samples are too perfect, resulting in 

a battle for the same exact frequencies. 

Consequently the distinct woodwind colors in 

traditional orchestration are easily lost. 

Finally, don’t be afraid to crank up the volume 

of your instrument sections to let them 

be heard, especially when using woodwind 

ensemble samples. Because of their smooth 

nature they tend to disappear in between the 

strings and the brass if the orchestration does 

not offer them their rightful space. 



VI review 

Akai Pro EWI 400s Electric 

Wind Instrument, $1000 list 

(street price about $700) 

www.Akaipro.com 

Akai EWI 4000s Electric 

Wind Instrument 

Do you play a wind or brass instrument? The Electric Wind 

Instrument is a wonderful controller for playing synths. and the 

latest incarnation of this fabulous instrument now includes a 

built-in synth and some new MIDI control features. 

Review by Nick Batzdorf 


It’s hard to believe that the EWI (Electric 

Wind Instrument) has been around for a 

quarter of a century, and its EVI (Electric 

Valve Instrument) brass counterpart even 

longer. What’s more, it’s hard to believe—and 

really great—that Akai has come out with a 

new EWI model about eleven years after the 

introduction of the previous 3020 model. 

The big thing about this new EWI, the 

4000s, is that it’s self-contained—there’s no 

separate rack unit. It has its own built-in modeled 

analog digital synth, effects processing, 

and MIDI I/O. You can use an optional external 

power transformer, but it’s normally powered 

by four AA batteries. It even has a builtin 

1/8” headphone output so you can just 

walk around and play. 

The 4000s maintains the identical high 

quality feel of the previous model, but it adds 

a lot of features and has a street price not 

much more than half the price. While there 

were reportedly 15,000 EWIs and EVIs floating 

around as of 1998, hopefully everything will 

converge with this new model to bring wind 

controllers more to the forefront. They certainly 

deserve it. 

It would also be great if more people 

learned to play the EWI, because it’s really not 

a difficult instrument to pick up. Anyone who 

plays a wind instrument can pretty much play 

it already, brass players aren’t all that far 

Fig. 1: The EWI 4000s is now self-contained with 

no rack unit. It has a built-in modeled analog 

synth, and MIDI out for controlling other instruments. 

While it’s considerably thicker than its 

model 3020 predecessor, cutouts on both sides (like 

the one above where it says EWI 4000s) make it 

feel just the same when you play it. The basic 

design has been around for 25 years, and it’s a 

mature, solid instrument. Anyone who plays a wind 

instrument can just about pick it up and play, brass 

players can adapt pretty easily, and it’s not a difficult 

instrument for beginners either. 

behind, and like everything else in music it 

pays back ten times what you put in. The 

most mundane synth patches can come to 

life when you play them with a wind controller. 

History and overview 

For those of you not familiar with this wonderful 

instrument, we’ll start with some background. 

The original EVIs and then EWIs were built 

by inventor (and virtuoso trumpet player) 

Nyle Steiner starting in the pre-MIDI mid- 

’70s—Steinerphones. Early versions of these 

instruments became popular in sessions 

around the early ’80s, in fact Steiner himself 

used to play EVI in the studios. If you remem-

Fig. 2: Fingerings are the same in every one of 

the instrument’s seven octaves, so once you know 

one you know them all. Then you select which 

octave you’re playing in by putting your thumb 

between two of these rollers—a very natural and 

intuitive interface. The bar at the tip of the thumb 

is the Glide strip, one of the instrument’s continuous 

controllers. Normally the Glide strip turns portmento 

on and off, but the more thumb it feels—the 

harder you squeeze—the higher the value it sends 

out. 

ber the St. Elsewhere TV series, for example, 

Nyle’s EVI playing was a big part of the sound 

of JAC Redford’s terrific scores. Since then 

many big-name jazz musicians have adopted 

the instrument, including Michael Brecker. 

The original EWI/EVI sent out control voltages 

to an analog synth that was part and 

parcel of the system, in fact even the 3020m 

synth, which was part of the rack brain of the 

EWI 3020, is digitally-controlled analog. (Akai 

later came out with a 3030m module that 

used samples.) Originally, MIDI’s 128-step resolution 

seemed to be a problem for synthesizers 

under wind control, but that got sorted 

out and MIDI was added pretty early on so 

you could use it as a controller. 

A lot of professional 

wind players would use 

the EWI with an Oberheim 

Matrix-1000 synthesizer in 

the early days—an experience 

we can now duplicate 

using softsynths 

(arguments about software 

models aside). Later 

(around 1994) the 

Yamaha VL1 Acoustic 

Modeling synth came 

along, and it and its lesser 

relatives became very 

popular as sound sources 

for wind controllers. 

Akai took over manufacturing 

and distributing 

the EWI and EVI about 20 

years ago, and sometime around the same 

time Yamaha came out with the first of their 

more sax-centric WX series of wind controllers; 

other companies have introduced 

trumpet-like controllers more recently. There 

have been several iterations of the EWI, and I 

understand that Nyle Steiner was still involved 

with the design of the 4000s. 

Steiner himself makes a custom instrument 

called the MIDI EVI, but as a widespread 

instrument it seems to be in limbo. However, 

a lot of EVI players are reportedly flocking to 

the 4000s. 

So the EWI has been a mature, solid instrument 

for a long time, and I personally have 

had a longstanding and passionate love affair 

with it. 

Why wind control? 

The obvious answer to that question is 

Because it allows someone who plays a wind 

or brass instrument to use his or her technique 

to play synths and samplers. But there’s 

more to it than that. 

When you trigger a note on a keyboard 

controller, it follows its programmed envelope; 

you use a slider or other physical controller 

like the mod wheel to modify the 

sound after the initial attack—a playing/programming 

technique that’s obviously 100% 

valid. Wind controllers augment keyboards in 

V.I. studios rather than replacing them, for 

many reasons, most notably that the vast 

majority of sample libraries are set up for keyboard 

control… never mind that only keyboards 

let you play independent notes with 

each of your ten fingers. 

However, wind controllers have some great 

advantages. When you play a note with a 

wind controller, it follows your breath (if programmed 

to do so, of course). Innately that’s 

a very physical, human connection, and it 

gives you continuous and total control over 

the sound without even thinking about it— 

just like an acoustic instrument. 

Blow progressively harder and the sound 

shadows your performance, getting louder, 

brighter, and possibly more distorted with 

louder overtones. Tongue the note (“ta”) and 

it has a sharper attack; play connected notes 

VI 

review 

in the same breath without tonguing them 

and they’re automatically connected; play 

successive notes and they all sound different 

because your breath pressure and attack will 

be different. 

Put another way, wind controllers are 

extremely expressive instruments. That’s what 

always overwhelms anyone who tries one for 

the first time. 

Fingers 

Because of its maturity, it’s hardly surprising 

that the basics of the EWI 4000s haven’t 

changed. While a lot is new, its essential feel 

is pretty much identical to its predecessor. 

Rather than the mechanical keys most 

woodwind instruments have, the EWI uses 

capacitance-sensitive buttons like the ones 

you find on some elevators—you just touch 

them and they’re “on.” The instrument 

comes with a neck strap, and you also support 

it by resting its grounding strip on your 

right thumb. 

Some wind players find the lack of keys to 

depress a difficult adjustment, so much so 

that at one point Akai offered add-on faux 

keys. Coming from a background playing 

recorder, I didn’t even notice it, in fact after a 

couple of days I actually felt more comfortable 

on EWI than on recorder. 

The instrument can use the Boehm fingering 

system used by clarinets and saxes, or you can 

use a simplified recorder fingering in which 

each finger is only responsible for one key—it 

never moves to cover other ones. This fingering 

is really easy, for example all fingers on 

gives you a C, and then you step up the major 

scale by lifting successive fingers in order. 

Then you determine the octave you’re 

playing by placing your left thumb between 

any two of a series of eight rollers (see Fig. 2), 

giving the instrument a 7-octave range (plus 

pitch-shifting). Every octave is fingered the 

Fig. 3: The EWI 4000s next to the 3020, the 

previous model. (Normally the 3020 is white—this 

particular one has been painted black.) Not shown: 

the 16” deep 2U rack unit you need to operate the 

3020. Cool-looking instrument, eh? 


VI 

review 

same, so once you know one you know them 

all. Some musicians coming from other wind 

instruments have a hard time avoiding glitches 

when they go over an EWI octave break, 

but again, I personally didn’t even think 

about it when I first learned the instrument. 

In practice, people new to the instrument 

are likely to start with the simplified recorder 

fingering and then gradually incorporate 

alternative fingerings and most likely some 

extra keys to make given passages less awkward. 

Using alternative fingerings for trills is 

routine on wind instruments; the difference is 

that on the EWI they’re in tune. 

What makes the EWI 4000s accessible to 

brass players is a new valve mode designed 

by Matt Traum of Patchman music 

(www.patchmanmusic.com, where you’ll find 

lots of info and wind-oriented soundware) 

that makes the fingerings brass friendly and 

reverses the octave rollers. The EVI uses something 

like a can that you rotate to switch 

octaves, and the reversed roller direction 

makes more sense to someone used to the 

EVI. Very clever. 

Since human beings don’t coordinate their 

fingers and breath with 100.000% precision, 

the instrument has an adjustable key delay 

that produces fewer glitches, i.e. it waits a little 

while before determining which keys 

you’re touching. A setting of maybe 25 milliseconds 

lets you play as fast as you want 

without generating false triggers. 

Lungs 

A lot of kids start off on recorder as their 

first instrument because it’s so quick to learn. 

The EWI is even easier. That’s partly because 

you can play it with an even simpler fingering, 

but mainly because you don’t need to 

develop much breath control to keep the 

instrument in tune and sounding even. 

You make noise by blowing into the 

mouthpiece, which directs the air to a pres- 


sure-sensitive transducer. By definition anything 

you blow into has an embouchure, but 

the EWI doesn’t require a special one—it’s just 

a matter of blowing into it. That’s another 

characteristic that makes it an easy instrument 

to learn. (Embouchure = the way you hold 

your mouth when playing a wind or brass 

instrument.) 

Early EWIs were closed without any air 

venting, so you had to let air out the sides of 

your mouth; like the previous model, the 

4000s’ air system has a small vent, but letting 

air out is still part of the playing technique. 

I’ve read complaints from some wind players 

that this is hard to get used to, but here yet 

again, I personally have never even been conscious 

that I was doing it. 

The breath pressure sensitivity is adjustable 

with great precision over a very wide range. 

You can set it so light that just the air pressure 

in the room makes it play—i.e. it’s on all 

the time, like bagpipes—or at the other 

extreme you have to blow so hard that your 

eyes bulge and your saliva glands hurt. 

Controllers 

In addition to the breath sensor, which in 

the version 2 software can send out MIDI 

Breath (cc2), Volume (cc7), Expression (cc11), 

Aftertouch, and Velocity—any or all of 

those—the EWI has several physical controls 

that affect the 4000s’ internal synth and also 

go out over MIDI. (Unlike previous models, 

the 4000s’ firmware can be updated via MIDI, 

and Akai has already added a considerable 

number of features in version 2.) 

First, there’s pitch bend up and down, 

which you trigger by sliding your right thumb 

up or down from its normal resting point on 

the grounding plate onto Pitch Up and Pitch 

Down plates. Then there’s a Glide strip on the 

far side of the pitch rollers (you can see it in 

Fig. 2). This is normally set to control portmento—press 

it with your left thumb and 

notes…well, glide from one to another. 

The Glide strip senses how much of your 

thumb is making contact with it, so it sends 

higher values when you feel like you’re 

squeezing it harder. In V. 2, both the pitch 

and Glide controllers can be assigned to MIDI 

cc numbers (cc = continuous controller; the 

MIDI spec has room for a lot of them). 

The mouthpiece incorporates a bite sensor, 

which sends a pitch or breath “blip” to trigger 

vibrato. Unfortunately the opportunity to 

separate this control from the breath or pitch 

information was passed by in the new 4000s 

EWI; while pitch blip vibrato is a very good 

feature, this really begs to be a discrete control 

in my opinion. (Windworks Design’s 

BrainBox and upcoming CV-Midi Pro control 

interfaces for the 3020 do allow you to separate 

the blip and the continuous control, 

although you can also combine them the traditional 

way.) 

The four controls we’ve mentioned so far 

are identical to the ones in the 3020. If it 

sounds like using them requires a lot of coordination, 

bear in mind that you’re only using 

them while you’re sustaining notes; they’re 

actually quite easy and intuitive. The only 

possible exception is that it can be hard to 

avoid triggering the Glide strip by mistake. 

For that reason I always set its sensitivity very 

low so it only gets activated in response to a 

very deliberate squeeze. 

There are two additional controls on the 

4000s: Octave and Hold. These on/off buttons 

also function as up/down switches when 

you’re tweaking EWI 4000s settings and 

patch change up/down buttons when you 

move your right thumb onto a metal screw 

on top of the Pitch Up plate. 

But while you’re playing, they’re lighted 

on/off toggles that you engage with your 

right middle and index fingers. The bottom 

one, Octave, causes the instrument to double 

the note you’re playing an octave (or any 

other interval you set) below. A 

red light makes it easy to see 

when either button is engaged. 

Normally, the Hold button sustains 

the first note you play sort of 

like a sostenuto pedal on the 

piano, so you can then other 

notes over it. You can set it to sustain 

until you attack another note 

with your breath (the default) or 

until you disengage the button, 

regardless of what you do with 

your breath. The Hold button can 

also send cc64 (sustain pedal) info 

over MIDI. 

Fig. 4: The UniQuest editor for the 

4000s’ built-in virtual analog synth. 

Notice that breath control takes the 

place of envelopes for the oscillators 

and the multimode filters for the oscillator 

and noise generator. But what 

makes this synth come alive is that it’s 

so responsive to the EWI.

It doesn’t take long to get comfortable 

engaging and disengaging the Octave and 

Hold buttons on the fly. The Hold button 

especially is a lot of fun, opening up all kinds 

of possibilities. You can jam over notes, play 

double stops on string patches, and so on. 

The new body 

To accommodate all the extra circuitry, at 

26-1/4” the EWI 4000s is 4-1/2” longer than 

the 3020. (See Fig. 3) It also has a larger 

perimeter, but due to cutouts in the body the 

keys feel to your fingers like they’re in the 

same locations. Actually, the right pinky keys 

are the same distance apart from one another, 

but it turns out that they’re 1/4” closer to 

the adjacent one on the 4000s, a subtle but 

noticeable difference that I personally find 

more comfortable. 

Though black rather than white, the styling 

is essentially the same as the 3020’s, with a 

high quality plastic body and metal sides. At 

just under 2 lbs. the 4000s is quite a bit heavier, 

but it’s still very comfortable, and it feels 

reassuringly solid. 

Getting around the EWI 4000s’ functions 

is straightforward. You simply hold down 

the Setup button at the top of the instrument, 

use the Hold and Octave buttons to 

scroll through parameters, release the Setup 

button on the parameter you want, and 

then use the Hold and Octave buttons to 

select the value. 

There’s a dedicated button to transpose the 

instrument to any key, which you program 

the same way: hold, and use the Octave and 

Hold buttons to scroll up in half-step increments 

to Eb or down to E. Two other buttons 

let you set the global and individual program 

output levels; and the level of each of the 

three effects: reverb, delay, and chorus—also 

globally and for each program. 

What you’re doing is reflected in of those 

early ‘80s-style 7-segment LED displays (two 

of them). You know, when everything is lit up 

it forms a boxy number 8 or letter B, if the 

middle bar is off it’s a 0 or letter O, etc. 

Initially I was concerned that all the writing 

on the black body is molded, so you can only 

see the button labels when there’s a light 

source from the side, but there are only four 

of them. It would be good if the Glide, Pitch, 

Breath, etc. knobs (accessible under a removable 

plate near the to of the instrument) were 

VI 

review 

painted, but these adjustments are pretty 

much set and forget. 

You can use a wireless MIDI transmitter 

and/or wireless guitar system with the EWI, 

since it’s self-contained and runs on batteries. 

That’s a great feature for performance, and a lot 

of players will gravitate to it just for that alone. 

But the other side of the sword for studio 

use is that you can now end up with as many 

as four cables attached to the EWI: MIDI in 

and out, audio out, and power (although the 

four 9V batteries do last a long time). The single 

wire clip on the 3020 only has to hold a 

single multi-pin cable. On the 4000s, that 

same clip can only hold one cable with three 

others tied to it—not the most elegant solution. 

Furthermore, you need to loop the wires 

to avoid strain if they don’t have right angle 

plugs. 

This instrument really wants a dedicated 

cable to carry all that. If you don’t ground the 

MIDI connections (which makes sticklers mad 

but in practice works fine), you’d only need 

to wire up eight conductors (4 x MIDI, 2 x 

audio, 2 x power). And you’d need to locate 

(CONTINUED ON PAGE 62)

VI feature 


Bent Clicks 

Using the tempo track to enhance performances 

by Paul Gilreath 

It’s curious how someone can spend countless 

hours choosing articulations and tweaking individual 

notes, while ignoring one of the most essential 

elements of a living, exciting musical performance: 

variations in tempo. Sometimes it can take a little time 

to introduce these tempo variations, but it’s very 

easy and well worth it. 

When any musician plays without a click track, slight 

variations in tempo occur naturally throughout the 

course of the piece. If your performance skills are up 

to it, you may be able to simply turn off the click and 

play; those variations will be there by default. After 

that, different sequencers have different ways of lining 

up their beats and barlines to coincide with your 

freely-recorded performance. 

However, playing without the click isn’t 

always possible or necessarily even 

desirable with the sample libraries we 

have today. Especially the larger ones are set 

up to be “programmed” as much as played 

completely in real time. 

So we’re going to analyze why, when, and 

how to program these tempo changes into 

MIDI compositions after the fact. We’ll 

assume the music was recorded to a static 

click track or step-entered with the transport 

stopped. 

Where 

Slowing down (ritardando) or speeding up 

(accelerando) can make a phrase more musical 

or expressive. These deviations from the 

piece’s set tempo can fit in many places within 

a phrase, such as: where the melody is particularly 

touching; where the most tension 

occurs, often right before it resolves; and just 

before an emphatic downbeat (where the 

tempo usually resumes. Ritards often mark 

the ends of phrases, sections, and pieces as 

well. 

Obviously some pieces want more variation 

than others—an orchestral march will have a 

more rigid tempo than perhaps a waltz or a 

more delicate piece; it’s usually just a matter 

of interpretation. 

Tempo tracks 

Most MIDI sequencers have two playback 

modes: using a fixed tempo setting, or following 

a variable tempo track that you’ve 

programmed. The fixed tempo setting may 

be a working mode that you only use while 

recording the parts. It’s easy to evaluate various 

tempi, or perhaps slow down the piece to 

play a difficult line.

The tempo track has various names within 

different sequencing programs—it’s the 

Conductor Track in Digital Performer and Pro 

Tools, the Tempo Track in Cubase, Nuendo 

and Logic, and the Tempo View in Sonar. In 

all four programs, changes can be incorporated 

using a graphic editor (where the tempo 

changes are shown linearly and can be drawn 

directly on the track) or an event editor 

(where the changes are shown numerically in 

a list form just like MIDI note data). 

The following examples use Steinberg 

Nuendo, but the technique is exactly the 

same no matter which software package you 

use. 

How it’s done 

Different musicians work different ways, 

but programming changes is one of the last 

things I do before mixing. Working one section 

at a time, I go through the piece to 

“spot” where the changes should go, dropping 

markers as I go. Then I work phrase by 

phrase. 

My preference is to use the graphical editor 

to make changes. This editor plots tempo on 

the y-axis against time on the x-axis (in either 

seconds and minutes or bars and beats), so a 

piece with no tempo changes would be represented 

as a single straight line. (Fig. 1) 

To implement a change, you simply draw a 

new point on the timeline. If you draw a single 

point, you will get an abrupt change; if 

you want a gradual change (as in a ritard or 

accelerando), you will need to insert several 

points. 

Let’s look at a couple of examples. First, 

let’s apply an abrupt change. Decide on the 

new tempo (in this case 90 BPM), insert a 

point with that value, and you have a completed 

tempo change from 60 BPM to 90 

BPM. (Fig. 2) 

Now let’s look at the more common 

change, a ritard. Usually you’re going along 

at the set tempo, the music slows down, and 

then the originally tempo is resumed. We 

need to insert points on the tempo editor 

that correspond to these three events. 

First, we define where the original tempo 

ends, going into the ritard. In this example, 

there’s a tempo insert point at bar 14 beat 3 

(Fig. 3a) at the original tempo. Next define the 

end point of the ritard where the original 

tempo will resume. This is done by placing a 

second tempo insert point at bar 15, beat 1 

(Fig. 3b), again at the original tempo. 

Finally, we place a tempo insert point 

somewhere between these points to program 

the ritard. This point will represent the last 

two instructions: how much change happens 

and how quickly it occurs. 

Usually the tempo gets slower and slower 

during the ritard until just before the event is 

over and the first (or primo) tempo resumes. 

Consequently, you’ll see the point entered 

near the end of the ritard as shown in Fig. 3c. 

Notice that I’ve changed the curve option to 

ramp instead of jump. This allows the tempo 

to change gradually between the points 

VI feature 

Fig 1. Graphical tempo editor with no tempo change and tempo at 60 BPM 

Fig 2. Graphical tempo editor with change from 60 BPM to 90 BPM 

Fig 3. Graphical tempo editor with end point of first tempo bar 14, beat 3 (15.4b) 


VI feature 


Fig 3b. Fig 3c. 

Fig 3d. 

instead of changing immediately (like it does 

in Fig. 2). Also, when I change the x-axis to 

display time rather than beats, the last point 

of the ritard occurs at 57.446” (Fig 3d). 

The next step is to listen to see if the ritard 

works with the music. In general, the faster 

the tempo, the more successful the singlepoint 

ritard will be; conversely, this type of 

ritard does not work as well in slower pieces. 

In slower pieces, success usually depends 

Fig 4. 

on the accompaniment and in particular the 

rhythm of the accompaniment. If the measure 

contains only slow moving notes, a single-point 

ritard may work fine. However, if 

the measure is filled with a lot of activity, you 

might find that the music really drags toward 

the end. This can make eighths or 16ths 

sound strange and out of context, like a 

music box running out of power. When this 

happens you need to add more points to the 

track to obtain the desired result. 

In Fig. 4a I have added two more points— 

one at beat 4.1 and 

one at beat 4.3. This 

makes it possible to 

create a ritard that 

slows slightly from 

beat 3 to 4.1, slows 

more from beat 4.1 to 

4.3, and then slows 

the most from 4.3.0 to 

4.4.87. When plotted 

against time (Fig. 4b), 

the start of bar 15 

now occurs at 56.724” 

instead of 57.446” 

from above. That means the ritard occurred 

slightly faster than the one in example 3. 

Thus this technique can be used to perfect 

your ritards and it can be used to alter hit 

points when working with media. 

In our next example, let’s insert a final ritard 

for a big finish. To create a dramatic flare, I 

want the second to the last measure (#104), 

which includes sixteen 16th notes in the 

accompaniment, to really slow down starting 

on beat 1. As before I draw a point that represents 

the end of the primo tempo and a point 

that represents the end of the ritard and beginning 

of the first tempo (although this step 

might not be necessary since this is the end of 

the piece) (Fig. 5a). Finally, I insert three points 

as shown in (Fig. 5b). 

The result is a very dramatic ritard, but let’s 

assume that the last few 16th notes sound 

too fast. Up to that point the ritard sounds 

good, so I just need to insert a few more 

points. First I draw a point that represents the 

ending of the present ritard at the point just 

before where the 16ths should slow more. I’ll 

insert this on the existing curve just before the 

second 16th note of beat 4 (Fig. 5c). 

Next, I can either insert another point just 

before my point at beat 4.4.045 or I can just 

move that point to a slower tempo, which is 

the option I choose: lowering the tempo to 

15 BPM (Fig. 5d). This results in the perfect 

ritard. 

For the final example, let’s enter some 

events that represent unintentional tempo 

changes. Very few human beings have perfect 

machine-like time. To simulate that, the goal 

is to change the tempo slightly every so often. 

Subtlety is mandatory for this to work\ 

The easiest way to do this is just to insert a 

tempo change every 6-10 bars. It’s not necessary 

to place these changes at the beginning 

of the bar, but they certainly can go there. 

The values of these points need to be close to 

the original tempo, perhaps within a range of 

+/- 1.5% (total 3%) of the main tempo. And 

the changes need to be in increments of only 

Fig 4b. 

about 1% of the main tempo. 

So if the main tempo is at 90 BPM, then 

the overall range of the tempo changes can 

be about 1.5% faster and 1.5% slower, resulting 

in a range from 88.65 to 91.35 BPM (but 

you don’t need to get out the calculator— 

we’re just being precise for this article). The 

changes shouldn’t be greater than about .09 

BPM. Insert the changes slowly and you can 

go back and forth between values—like 90 to 

89.1 to 90 to 89.5 to 88.65 to 89.5 90.2 to 

90.9, etc. (Fig 6.)

Clicking heels 

Tempo changes don’t have the immediate 

wow factor of great sounds or a well-written 

piece, but they can add an incredible amount 

of realism to your sequences. 

Paul Gilreath has composed music for feature 

film, television, documentarie,s and video 

games. He is the author of the best selling book, 

The Guide to MIDI Orchestration, 3rd edition, a 

700-page book detailing how to achieve maximum 

realism when producing orchestral emula- 

Fig 5a. Fig 5b. 

Fig 5c. Fig 5d. 

tions using sampling 

technology. Gilreath 

lives in Atlanta with 

his wife Channie and 

his two children, 

Quintin and Birdie. 

More information 

about Gilreath and 

the book can be 

found at www.musicworks-atlanta.com. 

VI 

If you just can’t program a ritard to sound natural, this is a simple 

technique that help. It works best when the accompaniment has a 

lot of movement rather than just consisting of sustained note: 

Record a piano performance (MIDI only) of the problem area without 

using a click. Analyze the MIDI events to see how you play the 

notes naturally. Translate that timing into your main sequence. 

You’ll need to mute everything you’ve recorded from the point 

where the ritard starts on, which you can do by automating the piano 

track’s solo button to mute everything else or by cutting all your 

tracks temporarily at that point and sliding them forward out of the 

way (you’ll put them back afterwards, of course). Be sure to play 

along with the recorded tracks before they drop out, and if the music 

resumes its tempo after the ritard, keep going to ensure a natural 

musical transition. 

At this stage you can look at the graphical or event editor to see 

how your performance lined up against the bar lines. Or you can use 

your software’s audio to tempo conversion function (Cubase’s Beat 

Calculator, Digital Performer’s Adjust Beats, Sonar’s Extract Timing/Set 

Measure/Beat At Now, Logic’s Beat Mapping) to help automate the 

process. 

When the Rit Won’t Groove 

VI feature 

Fig 6. 

Typically I do this manually. The way to do this is to write down the 

times in your new performance where each quarter or even eighth note 

falls. Then mute the piano line and go to the tempo track, viewed as 

tempo plotted against beats and measures instead of linear time. 

Enter two tempo points, one for the beginning of the ritard and 

one for the beginning of the measure that follows it if there is one; 

these points will of course be at the original tempo. Now enter a 

point at the first eighth note after beat one. 

Lower the tempo until this eighth begins at about the same time it 

did in the piano performance. Insert a point at beat 2 and pull it 

down like you did for the eighth, then the next eighth note and so 

on until you reach the end of the ritard. 

The result should sound almost exactly like your piano performance. 

You can use this technique for problem phrases or for the 

whole piece. 

Another tool that can be useful is your DAW’s Tap Tempo function, 

which lets you “conduct” the tempo by playing a note on your keyboard, 

which you record and use to build a tempo track. You can 

then go in and tweak the map using the techniques we’ve discussed 

in this article. 


VI review 

Modartt Pianoteq 

modeled acoustic piano 

Review by 

Jason Scott Alexander 

Modartt Pianoteq $249.00 ? 

(about $315 at press time) 

Formats: Windows VST; 

Mac VST, Audio Unit. 

www.modartt.com 

Copy protection: online serial 

number 


Look Ma, no samples 

The idea of physically modeling acoustic 

piano is nothing new. We first saw proof 

of concept in the mid 1990s with 

Yamaha’s Virtual Acoustic technology, together 

with promises of polyphonic hammered 

string models that, unfortunately, were never 

released. Many terrific electro-acoustic models 

have emerged in the ten years that have followed, 

but still no practical solution for a truly 

modeled acoustic piano has been found—up 

to now. 

In what its French creators at Modartt call 

the “fourth generation of the piano” (first: 

Cristofori’s pianoforte in 1698; second: electro-acoustic 

in 1929; third: sampled in 1984), 

Pianoteq is the very first commercial piano 

application where the sound is entirely com- 

Fig. 1: Modartt Pianoteq as it appears when first 

opened. The photos hide “advanced” parameters 

for tweaking the model and changing the sound 

radically. 

puted from discrete models without the use 

of samples whatsoever. (The exception is that 

it uses small samples for acoustic noises such 

as the pedal mechanism.) 

Creating a realistic sounding math-based 

acoustic piano emulation is no small task, 

though, what with all the psycho, acoustical, 

physical, and mechanical variables present. 

Pianoteq tackles each of these variables head 

on.

The grand scheme 

All the complex factors that make piano a 

truly vivid instrument, such as the interaction 

between strings, use of pedals, cabinet resonance, 

position of the hammers, time and 

space within a given room, etc. are taken into 

account. In fact Pianoteq is the first virtual 

piano factory, if you will, in that it can produce 

entirely new “brands” as well as copies 

of historically poignant instruments. It does 

that all in real time. 

The plug-in opens to a clean, compact, 

and rather attractive user interface that’s 

divided into six main sections (Fig. 1). 

Physical modeling is in the top half, where 

you click on photo depictions (Fig. 2) to edit 

Tuning, Voicing, and Instrument Design. 

In the lower half of the display you find virtual 

pedal board controls, along with an 

equalizer with a graphical display and a 

mediocre reverb unit (which you can bypass if 

you prefer to use higher quality plug-in 

effects instead). You can also adjust the velocity 

curve in quite some detail to suit your 

playing style and controller keyboard. 

When conjuring up a new piano sound, the 

Voicing section will likely be your first destination. 

This is where you find the Spectrum 

Profile editor, with sliders similar to organ 

drawbars for adjusting the first eight overtones 

to define your instrument’s timbre, tone 

color, and ultimately character. You can adjust 

the “ringiness” of a piano very subtly; many 

early piano manufacturers used to avoid a 

strong seventh overtone, for example. More 

experimental synthesists can easily turn 

Pianoteq into a harpsi-organ or piano-clavinet 

with just a few mouse strokes. 

Hammer hardness, another extremely 

important parameter, determines how brilliant 

the piano sounds. You set the hardness 

for p, mf, and f dynamic levels, and also the 

amount of hammer noise and soft pedal. 

These parameters together are timbral and 

percussive adjustments rather than changes 

to a layer of samples. The result is very realistic—not 

only do you feel how close the piano 

is to you, but the mechanical behavior of the 

hammer assembly itself. Then there’s the 

Character parameter, which essentially controls 

the irregularity of overtone intensity; 

increasing this may surprise you with a distinctly 

organ-like sound. 

Modartt even points out in their literature 

that this is pretty heady stuff, and understanding 

of basic piano behavior will help you 

adjust Pianoteq. Thankfully they’ve included 

excellent pop-up help floaters for each in the 

myriad of parameters available. 

The Design portion of Pianoteq features 

three soundboard controls that can also affect 

the character of overtones directly: impedance, 

cutoff, and Q. Though impedance can 

drastically alter the length of the piano sound, 

adjusting the piano Size (or harp/string 

length) parameter naturally affects both vibra- 

Creating a realistic sounding math-based 

acoustic piano emulation is no small task, what 

with all the psycho, acoustical, physical, and 

mechanical variables. 

tion time as well as timbre. A short string 

length produces greater inharmonicity, which 

means a very bell-like sound. 

A Global Resonance control governs the 

whole instrument, including strings, soundboard 

and cabinet, while Sympathetic 

Resonance adjusts the weight of the string 

resonance between individual notes. If you 

press and hold a few notes (slowly so they 

don’t make a sound) and then play some 

short notes with the other hand, you can 

actually hear the held notes ringing. Anyone 

who has played a piano will appreciate how 

realistic this. 

Likewise, the Quad Effect slider adjusts the 

level of that distinctive overtone ping an 

VI review 

Fig. 2: Absolutely every the characteristic of a 

real piano is accounted for. 

octave higher when play harder. Both this 

and the Global Resonance are really well 

thought out and implemented features. 

Sitting on the bench 

I have to admit, when I first played 

Pianoteq I wasn’t immediately sold on its 

sound. There seemed to be no getting past 

the fact that its presets sounded boxy and less 

present, and certainly lacking the same stereo 

edge that live sample recordings have. 

While it was certainly impressive that this 

was all coming from a model and not samples, 

I wasn’t going to trade in my gold standard 

GigaStudio collections Granted, I’m a 

pretty harsh critic of piano recordings, piano 

sample libraries, and now virtual pianos; 

piano is the instrument I grew up on. 

As it turns out, comparing the presets sideby-side 

to my favorite Steinway, Bosendorfer, 

and Yamaha libraries was actually the downfall 

in my initial evaluation. You see, Pianoteq 

doesn’t come with “branded” presets, and it 

doesn’t directly attempt to emulate any of 

your favorite piano flavors. 

Instead it comes out of the box fairly neutral 

sounding, or hybrid as it were. Particularly 

in the middle registers, Pianoteq has a tendency 

to sound a little “honky” to me, not 

unlike the way some digital pianos sound 

when played softly. 


VI 

review 

It wasn’t until I began exploring the voicing 

and design provisions in greater detail 

that I began appreciating Pianoteq for its true 

strengths flexibility. After increasing hammer 

hardness for more overtones on the attack, 

adjusting the hammer hardness and pedal 

response at each velocity level to suit my 

Korg Triton Pro X controller, tweaking the 

tone color with the spectrum profile, and 

adding presence through string response and 

some creative tuning adjustments, I was able 

to get very close to my favorite sample 

libraries. 

In fact I was able to get the entire keyboard 

range to sound absolutely stunning, 

and I don’t use that term lightly. The dynamics 

are extremely convincing. 

You can also get creative. For instance, the 

ability to increase the length of the piano 

incrementally means that you can take it from 

A size to D size, or even create a 33-foot 

grand if you like. Yet because with the harmonics 

change as the piano is lengthened, 

the bass notes remain incredibly solid and 

believable. Conversely, I boxed up my perfect 

Steinway D emulation into a respectable 

sounding apartment-sized Heintzman piano, 

detuned Coldplay-esque stage piano, detuned 

honky-tonk, and even a tinkering toy piano. 

But creative sound design is only half of 

Pianoteq’s bag of tricks. Its playability is second 

to none. 


The amazing tuning section gives you all 

the possibilities typically done only by piano 

tuners. For example, you can set diapason 

(415-466 Hz), different kinds of temperaments 

(from equal to well tempered), unison 

tuning (for changing the timbre or color of 

the sound), octave stretching, and direct 

sound duration. Pianoteq also features a pro- 

gressive sustain pedal, allowing partial pedal 

effects such as quarter and half-pedaling; a 

Sostenuto pedal; harmonic pedal; and una 

corda (“one string,” i.e. soft) pedal. 

Pedal play on glissandos sounds incredibly 

lifelike—it’s as though you have your head 

hanging over the harp. Pedaling in general is 

spot-on. And it doesn’t just consist of the artificial 

thunking in a lot of sampled pianos. I 

was especially impressed by the ability of 

Pianoteq to catch a flurry of staccato notes 

with intermittent presses of the sustain pedal 

and treat them accurately. 

Because the plug-in operates with 32-bit 

internal resolution at rates up to 192kHz, 

there’s absolutely no funny digital noise, plus 

the instrument produces a real progressive 

variation of the timbre at 127 MIDI velocities 

The interaction between strings, use of pedals, 

cabinet resonance, position of the hammers, time 

and space within a given room, etc. are all taken 

into account. 

per note. A phase/repetition variation scheme 

ensures that the hammer strikes the strings 

slightly differently with each note, and thanks 

to the wonderful math going on behind the 

sympathetic vibrations algorithms, chords 

large and small truly sound like chords played 

on a real piano as all the individual strings 

interact. 

Left me tickled 

I really like Pianoteq. Its sound is pure, 

vivid, alive with responsiveness, and without 

question the most flexible of any “acoustic” 

piano instrument on the planet today. Thanks 

to its rather modest system requirements, 

Pianoteq will run on any reasonably current 

desktop computer with next to no impact on 

the processor, and the extremely small 8MB 

size and negligible RAM requirements means 

it can run on any modern laptop. (A standalone 

version with studio-grade compression, 

EQ, and reverb would be useful for “straight 

out of the box” stage use—and in fact the 

developers are working on a stand-alone version; 

they also plan a free update this spring.) 

With more likely to come, there are currently 

two free add-on presets available for 

registered Pianoteq users at the Modartt website. 

These include dynamic model constructions 

of an 1812 Pianoforte Schöffstoss 

recorded at Schloss Kremsegg in 

Kremsmünster, Austria and a Johann 

Evangelist Schmidt pianoforte, circa 1790, 

recorded at Handelhaus in Halle, Germany. 

The presets are meticulously modeled with 

authentic tuning (well tempered) along with 

all its charms and artifacts. 

Though it may not capture one hundred 

percent of the image that a stereo sampled 

piano will, it’s the subtle real-time details that 

make Pianoteq sound leaps and bounds more 

realistic for solo work—and win you over 

every time. VI

VI review 

Review by 

Chris Meyer 

Electro Magnetic Fury $99.95 

Big Fish Audio 

(www.bigfishaudio.com) 

platform: 16-bit 44.1 kHz 

AIFF/Apple Loops, WAV, & REX2 

files 

license: May be used in the 

licensee’s own “derivative” live 

performances or recorded compositions, 

but not in a sample or 

music library. 


Loop Librarian 

Loops loops loops loops and libraries 

Big Fish Electro 

Magnetic Fury 

sound library 

Ilike the idea of industrial music, but rarely 

do I find an industrial sample library that 

lives up to the promise of the genre. It 

seems that many try too hard to be abrasive, 

clever, or obscure, while providing isolated 

loops that give you too little variation to build 

an entire, cohesive song around. The promo 

materials for this collection—promising it was 

“downright dangerous” and would “blow 

through your speakers” with “punishing 

drums and beats” and “ear-shattering FX”— 

made me assume we had another entrant in 

the too-abrasive category. 

Therefore, I was pleasantly surprised to 

hear that this library hews much closer to the 

intelligent industrial mindset, combining 

ambient and exotic elements with a pulsing 

groove. 

As is the case with most Big Fish loop 

libraries, Electro Magnetic Fury repeats its 

content in AIFF (Apple Loops), WAV, and 

REX2 flavors. The AIFF version weighs in at 

just over 1.5 gigs of 44.1kHz 16-bit stereo 

files. 

There are 95 construction kits listed by 

tempo and occasionally key. The kits each 

contain a 30+ second demo plus from three 

to 16 component loops, ranging from two to 

12 bars in length (four being the most common). 

Typically, one part is presented in two 

or more variations, and there is usually an 

ambient loop among the components. 

Although these kits are intended primarily 

to be rhythmic backing tracks, there are often 

tonal (dare I say melodic) elements, including 

pitched drums, vocoded rhythms, processed 

traditional instruments, and the occasional 

sequencer. Tempos range from 67 to 200 

bpm, focused around the low 100s. In addition 

to the kits, there is also a folder of 17 

“atmospheres”: lovely 30 second to two 

minute droning, ambient, mostly tonal beds 

with an occasional rhythmic pulse that would 

make great intros or bridges 

As I hinted in the intro, what struck me 

most about this library was that the atmosphere 

wasn’t nearly as harsh and monodimensional 

as I expected. There is indeed an 

abundance of strong grooves and distortion, 

but some pieces are downright ambient, 

while others wander into house or even 

lounge territory. There is also more of a feel-

ing of understated mystery, dread, or power 

than pure chainsaw mayhem. Many kits 

throw in surprising elements such as African 

chants, Middle Eastern percussion, Far Eastern 

strings, military snares and toms, ambient 

hits, and a lot of bandpass-filtered beats. 

The only thing I can hold against this collection 

is a lack of variation, as most components 

are only presented in one version. As a 

Mahadhi: African Rhythms 

$99.95 

Big Fish Audio 

(www.bigfishaudio.com) 

platform: 16-bit 44.1 kHz 

AIFF/Apple Loops, WAV, & REX2 

files 

license: May be used in the 

licensee’s own “derivative” live 

performances or recorded 

compositions, but not in a sample 

library. 

result, quite often you will need to create 

movement by switching elements on and off, 

rather than being able to pull up evolutions 

of the existing parts. Many who create music 

in this style are already used to working this 

way. For those outside of this genre looking 

for additional spices and flavors, I think you’ll 

find a surprising number of useful layers and 

elements to include in your own work. VI 

Let’s set the record straight from the 

start: This is not a pristine set of carefully 

recorded solos and grooves performed 

in isolation to a click track by wellpaid 

session musicians in high-end studios, 

preserved in 24-bit 96 kHz fidelity. The vast 

majority of the loops were recorded in mono, 

close-miked with varying degrees of fidelity in 

live ensemble performance situations, complete 

with some mic bleed and the occasional 

grunt. 

And you know something? It’s great. My 

left brain says it shouldn’t be, while my right 

brain says shut up and dance. 

All loops are available in AIFF, WAV, and 

REX2 formats; I tested the AIFF versions, 

which weigh in at just over a gig. There are 

27 construction kit folders labeled by tempo. 

These include from five to eleven subfolders 

that break out each song by instruments, plus 

a minute-plus 24-bit stereo demo that illustrates 

various combination of instruments. 

(Don’t be fooled, though, as most components 

are 16-bit mono.) 

Each instrument folder contains anywhere 

from one to 47 mostly 2-bar looped phrases 

in 4/4, tallying up to over 1800 loops total. 

With the exception of the occasional kalimba 

folder, you get just the percussive instruments, 

although you can hear through the 

mic bleed that a full ensemble including 

singers was actually performing at the time 

these were recorded. 

As noted, sound quality varies: the conga 

and djembe have particularly nice transients, 

plus the bembe and ogenne have appropriately 

present rattles and buzzes, but the shakers 

and kalimba suffer from exaggerated 

VI review 

Big Fish Mahadhi: African Rhythms 

loop library 

proximity effect by being miked too closely, 

while the talking drum sounds a bit muffled. 

This can make you cringe while listening to 

loops in isolation, but it becomes far less of 

an issue when mixed into a track. 

Someone used to a steady diet of Western 

grooves might listen to real African rhythms 

and hear just a cacophony of competing 

instruments, assuming there’s no way that the 

timing is tight or would line up against a 

steady Western pulse—hey, even I did at first. 

In reality, all those mad percussionists are 

cleverly hitting in-between beats and placing 

emphasis on notes other than The One. 

As a result, these loops work surprisingly 

well as layers mixed in with straight-ahead 

Western drum kit grooves, as they hit the 

spaces in-between, rather than fighting over 

the accents. Although the occasional groove 

needs some timing correction to bring it back 

to a metronomic beat, the vast majority are 

actually in the pocket. 

Another nice thing is that most of the 

loops are closer to backing grooves with variations 

rather than solos, again making them 

work well as layers supporting other rhythms. 

Stripped of their melodic instruments, the 

“songs” come across more as moods or flavors, 

also resulting in them being more flexible 

than anticipated. 

Honestly, my initial impression of this 

library was not good, but now I expect to use 

it quite a bit for texture and spice. While 

recording live like this might have caused the 

fidelity to suffer, the flip side is that the performances 

are exuberant, and authentic emotion 

is all too hard to find when it comes to 

samples. VI 


VI interview 


MIDI MOCKUP MICROSCOPE 

In this installment of our series on composers and how they did their 

MIDI programming, talented composer Nathan Furst discusses “Heart of 

the Baja Part 2” and “Shark Attack.” Download these cues at 

www.VirtualInstrumentsMag.com and follow along. 

At only twenty-eight years old, Nathan Furst has 

already begun to carve out a niche for himself as one 

of Hollywood’s elite young composers. Nathan has 

developed a reputation for blending world instruments 

and electronica soundscapes seamlessly with the 

strength and elegance of a symphony orchestra. He 

has composed themes and underscores for over 20 

film and television projects. This signature style has 

earned Nathan three nominations for his outstanding 

original film and television compositions. 

by Frederick Russ 

more 

online 

www.virtualinstrumentsmag.com 

Tell us about your rig and your sample 

libraries. 

That’s a crazy thing. It’s always in a state of 

flux. I envy the people who run out and buy, 

say, four or eight Vision computers [custom 

machines from www.VisionDAW.com] and 

then don’t touch their systems for another 

four years. I wish I could live like that. I 

can’t—it drives me insane. 

So it’s constantly changing. Currently I 

have just taken four computers out of rotation 

because they have been causing too many 

problems, or they don’t hold enough memory, 

or whatever it is. I just built three computers—myself 

actually. I went to Fry’s, bought a 

motherboard, bought a CPU, bought one of 

those metal rack chassis that look like the 

Vision computers, and just built it. So I have 

these three new machines. 

I’m currently starting a project, and I’m 

going to be using four PC farms, with a fifth 

that kind of floats in there if I need it. I have 

two G5 dual 2.0s. One of them I don’t use 

that much; it basically sits in the background 

as a potential node and a potential [Native 

Instruments] Kontakt 2 secondary.

For the most part, it’s just sitting there as a 

back-up computer. I try to keep it as near to 

my original G5 as possible, so if my first G5 

goes down I take the first one to the shop 

and slide the second one over and keep working. 

I got bit once where I had a Version 1 G5 

and I was in the middle of a deadline, and 

the computer went down and I didn’t know 

what to do. So I literally bought a new computer. 

I kept going, but of course there was 

another hump to get over: installing the software. 

Meantime, my deadline is coming up 

on me. Ever since, I try to keep a secondary 

sequencing machine that is near to my first 

one in case it goes down. 

So right now I am using four PCs, which is 

pared down—I originally had six. And then I 

have a dual 800 Mac that does nothing but 

run Pro Tools. 

Which Pro Tools systems is that you 

are running? 

It’s just a basic Pro Tools Digi 001. While 

I’m writing, Pro Tools runs all my video. I like 

to keep video off my main machine. 

Whatever sequencer I’m writing in, I 

inevitably bring it to its knees, because I’m 

writing these gigantic orchestral things. I 

might have Space Designer [the convolution 

reverb in Apple Logic Pro] and a couple of 

[Spectrasonics] Stylus instances on there— 

plus I try to work at a low latency. 

Logic sends MIDI Time Code to Pro Tools so I 

can go to bar 40 and the picture will follow 

along—as if the video were in the sequencer. At 

the very last stage when all my MIDI tracks are 

playing simultaneously, I just bust the masters 

out to Pro Tools, using it as a tape machine. It 

locks perfectly in sync, and I go into Record 

mode, and that’s how I have my sends that get 

delivered to the stage. It’s instant—it’s so fast, I 

can almost mix in real time. 

Many times on a film they want the percussion 

track separated or they want the choir 

separated. Having the Pro Tools system and 

working it this way allows me to do that. 

What kind of monitoring system do 

you use? 

Probably not the best one! I just do what 

works best for me. I’m a big fan of keeping 

the room as lean as possible, and I like that I 

have successfully driven out any real world 

interview VI 

Fig. 1: Nathan Furst has come up with a very clever way of navigating his huge orchestral template (34 

tracks just for violins alone), as shown in this Logic screenset. See the shrunken window with the colored bars 

on the left? That’s a second Arrange window saved as part of this screenset, and the bars are folders. 

All his string tracks are packed inside the dark grey Strings folder, the Woodwind tracks are packed inside 

the sage green folder, and so on. Clicking on (in this case) the Strings folder opens up the main Arrange window 

to all the string tracks. 

The reason this works is that the main Arrange window is Linked to the same level, meaning that it jumps 

to whatever is being selected in any other window. That’s the little yellow chain link icon at the upper left; to 

prevent the shrunken Arrange window with the folders from scrolling to the current location, it isn’t linked. 

Also note the locked markers corresponding to different events in the show he’s scoring. 

audio (by soundproofing it). 

There’s nothing that enters the real world 

except for what’s coming out of my MOTU 

2408 interface to my monitors, which are currently 

Mackie HR824s. I like them. A lot of 

hardcore engineers call them the “composers’ 

monitors” because they argue that it is not a 

true flat sound and that it makes everything 

sound good. They claim there is a lot of 

sweetening in the monitors—which is probably 

true, but I don’t care because I like it. 

Are you using a mixing controller? 

I have a short-lived Logic Control. It’s a 

great little box, but I have to be honest, I 

don’t go to it that much. I’ve been using 

Logic for 12 to 15 years. 

Let’s talk about “Shark Attack.” From 

0:01 to 0:05, how did you get the brass 

to sound that way? 


interview 

VI 

Fig. 2: This is the part of the same screenset shown in Fig. 1 that’s displayed on a second monitor (as you 

can see behind Furst in the photo with this article). One of the things it shows is the wonder of remote access 

over ethernet. 

At the same time that clicking on the Strings folder brings up the string tracks shown in Fig. 1, on the second 

monitor it brings up an “adaptive mixer” with channel strips for every track that’s being used in the 

Arrange window. 

Note that there’s room in the screenset for two remote computers to be displayed above the mixer. The 

machine at the left is a G4 Mac running Pro Tools LE, which Furst uses as a mixdown machine for recording 

stems (since Pro Tools is usually the delivery format); the one at the right is a Windows XP machine running 

Native Instruments’ Kontakt 2 sampler (you can see the ethernet address 192.168.0.13 at the top of the 

window). 

Clicking in either remote computer window activates it for keyboard and mouse control just as if it were 

running on the local machine. The response over a network is noticeably more sluggish than a wired monitor, 

but it’s perfectly fine for recording stems or just loading sounds into a soft sampler running on a slave 

machine. 

Actually I wrote that in, I think, the summer 

of 2002. At the time I think I had one 

machine that had the Sonic Implants stuff on 

it and SAM Horns. 

And those cool trombones? 

That would have been me custom re-programming 

the SAM stuff, layered with 

Miroslav Vitous. 

How did you do that? 

I grabbed the original files and I basically 

built, like a 20-key keyswitch program. 

Everybody thought I was crazy, but now it’s 

like the norm. At the time, EXS [the sampler 

built into Logic Pro] was really the only instrument 

that allowed you to go that deep. I 

would just go in, grab the original files, and 

remap them. 

Usually I would have a starting point to 

successfully convert, say, an E-mu patch, just 

so I would get the key mapping done correctly. 

And then from there, just start making little 

tweaks with the filters, adding buzz, and 


playing a lot with reverb tones and rooms 

and sort of just discovering how I was going 

to fit these things into the same space. 

That’s the way I started, which is essentially 

the way I work today—basically having one 

“pre-delay” sort of room reverb on a buss, and 

then everybody gets that buss send. It’s prefader. 

So essentially every single instrument has 

its own mix to do, it has its own wet fader and 

its own dry fader. Obviously the trumpets 

would get a lot more wet and almost none of 

the dry signal—that kind of a thing. 

So you’re approaching it like you’re 

standing there—the trumpets are farther 

back, so you’re kind of thinking 

about how the reflections go? 

I was basically thinking What should it 

sound like? I don’t know if this is “right” or 

not; I wasn’t thinking of how it would sound 

if I was standing there, because that is sort of 

irrelevant. If you’ve ever heard a cue or an 

orchestra in a recording studio, if you walk 

back and forth from the stage while they’re 

playing, and then in the 

middle of the cue walk to 

the booth, it sounds very 

different. 

So I don’t really think in 

terms of Sonically how it 

would sound if I was physically 

standing there, I think 

in terms of production. 

How do I produce this cue 

using my favorite composers 

and my favorites 

scores as a standard? 

So you’re listening to 

the orchestra and not 

necessarily trying to 

emulate it? 

Oh absolutely emulating 

it, absolutely. Some of those 

guys have a great sound. I 

was absolutely trying to 

emulate the production 

sound. Not necessarily any 

one in its entirety—not only 

a John Williams sound, or 

only a Don Davis sound, 

but sort of taking the things 

I love from all of them—then take what you 

believe you do, and incorporate all of that. 

You have a lot of clarity in this cue. 

For me anyway, some of that is the writing. 

I try to write—orchestrate—in the sense of 

thinking of it as a spectrum of light. 

Everybody sort of gets a space. And absolutely, 

somebody can cross over to the other person’s 

space, but I try not to do it at the same 

time. That helps create a sense of separation. 

You like panning wide, I’ve noticed. 

Like with the custom library you’re 

working on, everything is panned 

wide—the violins are way to the left. 

Definitely, and maybe a little too much— 

maybe. I’ll say I naturally go a little wide. For 

me it helps me to hear the orchestration a little 

bit better, and also when things are really 

wide, inevitably in almost every score I do 

there is some sort of, shall we say, a specialty 

element. 

Lately I have been hired to do a lot of 

world scores—big orchestra, but there is 

always a sort of a Mideastern or world element. 

That stuff tends to hang in the center a 

little more, and the orchestra is off harder 

panned. I feel I’m able to make sense of what 

is happening a little bit more, as opposed to if 

everything is generally hanging around the 

middle. 

I’m not really re-inventing the wheel, I’m 

emulating my favorite scores and my favorite 

productions. I guess I am re-inventing the 

wheel for myself in the sense of making it 

happen completely in the room with no 

money and no recording schedule, or whatever. 

You’re basically faking it, but absolutely—create 

real estate and create space.

Do you remember what you were 

using back then for ambience? 

Yeah, it’s terrible. Now I can’t stand it, it’s 

way too much ambience. It’s way too washy. I 

was basically using the Waves Gold stuff 

because that was what I had available at the 

time— the TrueVerb plugins for the room and 

using a separate hall verb for the hall. [Audio 

Ease] Altiverb was out, but I didn’t have it 

working then. 

I had made my own TrueVerb preset for 

each section. The strings had one to represent 

them being in the front of the orchestra, 

woodwinds a little behind that, and the horns 

were way in the back, as were the trumpets. 

It was an experiment, shall we say. 

Now you’re using convolution reverb. 

I just use Space Designer [the convolution 

reverb built into Logic]. I like Space Designer 

a lot. 

This is sort of my little rule. To me there are 

way too many options. So if it requires extra 

things that are unnecessary, like hardware 

dongles or if you load one instance and it 

soaks up half your CPU, we’re done. There 

are too many amazing options happening out 

there to have to tolerate things that are little 

annoyances. 

I realize that it is an older piece, but 

it still sounds cool to me. So “Shark 

Attack” is from “Christmas Vacation 

2”—what is that, a movie? 

Yes, it was originally a TV movie, like a special 

event. It was aired on NBC around the 23 

of December. If I were to guess, it’s about 80 

minutes of music—maybe 90. From spot to 

delivery was about four weeks. 

Oh my goodness. So you were averaging... 

…a few minutes a day. I find two minutes a 

day to be very comfortable—I can hang out, I 

can write the music I want to write. I can 

orchestrate it and experiment with the 

orchestration if I am on a two minute a day 

schedule. If I am on something worse, which 

tends to be happening more and more—like 

if I have to come up with four minutes a day 

or more, then I am just writing it the best I 

can. So pretty much when it’s done, it’s done. 

You don’t have time to second guess 

yourself. 

No, there is no time to second guess yourself, 

let alone experiment. On a creative level, 

and also on a practical level, I work a lot with 

themes—I try to create themes because I find 

that is an appropriate tool as well as a useful 

one to be able to recall characters’ themes constantly 

and consistently throughout the film. 

So when I’m writing music, I can create a 

mood with my orchestration, and what is 

happening in certain sections rhythmically— 

sort of sneak around and play with the theme 

Fig. 3: While note-on velocity has been recorded in the sequencer (as evidenced by the orange and red 

notes in Logic’s graphical Matrix editor), Furst turns off the velocity response in all his sample programs, 

instead relying on MIDI continuous control changes to create dynamics. The blue pane shows the fairly radical 

mod wheel riding that shapes each note in this passage. 

This screenset also has a notation editor open; Logic’s windows are interactive, so changes in one are 

immediately reflected in all the others. 

VI 

interview 

and keep incorporating it. The theme helps 

me to be able to write four minutes a day. 

On top of that, some of my favorite scores 

work that way. There’s a sense of consistency. 

You have some interesting ideas 

about creating realistic performances. 

When I am playing something like fast staccato 

stuff, I also like to layer it with half note 

trills and bury them—anything that will help 

me create bad intonation. Because if you 

were to have string players playing those passages, 

there is no way in hell they are going 

to hit the tone. It’s going to be a mess. 

So even though it’s a fast staccato passage, 

I’ll double it with the trill pass and bury it, so 

you don’t hear perfect intonation. If you are 

trying to create a sense of realism, I think it’s 

a common mistake to try and write runs in 

staccato, or to write runs with a short note 

patch of some sort. I prefer to write mine 

with legato, doubled with trills, doubled with 

trem. I find that for me, those are the most 

successful runs. 

I had rebuilt some of the runs in that cue 

from the old Roland libraries. For things like 

the woodwind runs that are happening in 

there, those are probably not performed—by 

me. They are probably just runs I grabbed 

from a library and reprogrammed in EXS. But 

I will play with the runs. 

So if it is a woodwind run that just goes 

up, I will double it myself by, say, playing a 

piccolo with it. Inevitably I’m not fast 

enough, and that produces the sound I want. 

I strive for imperfection. 

I want to talk about breathing 

strings and breathing life into strings. 

You have a small cue here—“Heart of 

the Baja, Part Two.” You’re really emoting. 

That would have been 

a combination of many 

things, including another 

custom library. But for me 

the real trick is that with 

the exception of staccato 

or short notes—even with 

marcato stuff—I never use 

a velocity-sensitive patch. 

To me that is not what it 

would sound like. The first 

thing I tend to do when I 

load a string sample, I disable 

any response to 

velocity. 

Then I rebuild it with 

crossfades only on the 

mod wheel to help me 

get in and out at the very 

tail end of it, almost like 

you can’t really discern 

when it went away. 

Always, everything going 

in and coming out is a 

(CONTINUED ON 

PAGE 62) 


VI review 

Steinberg Cubase 4, $999.99 

Cubase Studio 4, $499.99 

www.Steinberg.net, distributed 

by www.Yamaha.com 

Formats: Mac OSX 10.4 (G4 or 

better) including Intel Macs, and 

Windows XP Home/Professional 

Copy protection: USB dongle 

(comes pre-licensed; upgrade 

requires online update) 

Steinberg Cubase 4 

A look at the update to this massive 

sequencing package 

Review by Mark Jenkins 


Steinberg’s Cubase digital audio 

sequencer is especially popular in its 

native Europe, but it also has a lot of 

users in the US. The program has undergone 

constant development for a lot of years, and 

that shows in the depth of its features and 

refinement. 

The last version of the massive Cubase 

sequencer was SX3, but now the SX prefix 

seems to have been quietly dropped, so the 

new revision is referred to simply as Cubase 4. 

There’s also a Studio version with a strippeddown 

feature set, a much lower price point, 

and an upgrade path to the full version. 

The fact that Cubase (as we’ll simply call 

Cubase 4 from now on) runs on both Mac 

and PC offers some flexibility, and a Steinberg 

USB key—omitted from some of the company’s 

simpler packages—is included. You’ll 

need a DVD drive to install, followed by an 

internet licensing procedure. 

As always with major revisions of complex 

sequencers, the manufacturer lists scores of 

updates (as well as updates to the way it handles 

scores), but the basic look and feel of 

Cubase has not changed very much in this 

Fig.1: The main Project window in Cubase 4 

handles audio, external MIDI, and internal virtual 

instrument tracks with absolute equality. It can be 

made as plain or as multi-colored as you wish. 

revision. The same large Project window (Fig. 

1) handles audio, external MIDI, and internal 

virtual instrument tracks with absolute equality, 

and can be made as plain or as multi-colored 

as you wish. A large Transport window 

can be positioned anywhere, helping to make 

the package ideal for use with dual monitor 

set-ups. 

But beneath the hood Steinberg claims 50 

or more major revisions, and there are certainly 

many new virtual instruments and 

effects included, offering thousands of new 

sounds. Let’s take a look at some of these 

additions and revisions in more detail. 

Mixing and editing 

Cubase was always pretty flexible as 

regards final mixdown, but the addition of a 

new Control Room Mixer window (Fig. 2) will 

help interface the package better to the out-

Fig. 2: The new Control Room Mixer features 

Talkback and Headphone busses, External Inputs, 

and Control Room Monitor busses, all without having 

to leave the software. 

side world. Now there’s a Talkback and a 

Headphone buss, External Inputs and Control 

Room Monitor busses, assuming of course 

you have the external hardware to take 

advantage of all those. 

With up to four Studio Send busses per 

audio channel, the whole cosmetic appearance 

of the mixer (Fig. 3) has been sharpened 

up so it’s easier to read, and insert plug-ins 

can be copied from one channel to another 

simply using drag and 

drop. The individual preset 

name of the plug-in 

effect is also now visible 

(like “BigFlange6” rather 

than just the name of the 

effect). 

Some editing functions 

are also improved. In earlier 

versions of Cubase it 

was difficult to edit controller 

data alongside the 

related notes, so for 

example if you moved 

some notes you may 

have lost contact with 

the vibrato that accompanied 

them. Now notes 

and controllers can be 

edited together, which is 

a huge improvement. 

There’s a new Trim 

Tool to trim start and end 

points for multiple 

events, which is going to 

be useful for complex 

cues. A new Preferences 

setting allows event 

selection to select the relevant Track automatically. 

In fact there’s a whole new type of track 

too: the Instrument Track, combining a MIDI 

input and an audio output for VST instruments 

into a single track and mixer channel. 

One of the most confusing points of the old 

Cubase up to SX3 was that a lot of the mixer 

channels were doubled—for a virtual instrument 

you’d have one for its MIDI activity and 

effects, and one (often far removed from it) 

for its audio output and audio effects. 

This made it hard to pin down exactly 

where an instrumental sound was coming 

from, and made for big mixer displays too 

wide to fit on one or even two monitors. The 

VI 

review 

introduction of this new track type is a huge 

improvement. 

MIDI and VST 

There’s little change to the way the new 

Cubase handles MIDI except in the area of 

scoring. The settings dialog has been simplified 

and the score editor now has clearer 

icons. There are two new score fonts—”Jazz” 

and “Classic”—and song lyrics can now be 

imported from a MIDI file. 

In fact there are several improvements in 

the area of file and preset handling. MP3 

Surround (which is compatible with MP3 

stereo) can now be imported and exported, 

and a new media management system called 

MediaBay makes it much easier to search for 

and modify files and preset sounds within VST 

plug-in instruments via the “SoundFrame” 

Universal Sound Manager. 

Beneath the hood 

Steinberg claims 50 or 

more major revisions, 

and there are certainly 

many new virtual 

instruments and effects 

included. 

In other words, Cubase now accesses and 

searches the preset list of any virtual instruments 

in use, and also adds a preview function 

that if you select a looped sound, plays it 

at the current project tempo. The MediaBay 

browser can help search for audio loops and 

clips, MIDI Files, video files and entire Cubase 

project files on your system too. 

The greatest excitement though probably 

lies in the new Cubase VST (Virtual Studio 

Technology) instruments and effects. In fact 

the standard has now been revised to VST3, 

and some 17 original plug-ins having been 

updated accordingly. VST effects now include 

mono, stereo, and ping-pong delays, nine different 

dynamics controllers including compressor 

and expander plus vintage and multiband 

compressors, nine modulation effects 

including chorus and autopan, an octaver and 

tuner, plus mono-to-stereo and stereo 

enhancer routines. 

Fig. 3: With up to four Studio Send busses per 

audio channel, the whole cosmetic appearance of 

the mixer has been sharpened up so it’s easier to 

read, and insert plug-ins can be copied from one 

channel to another simply using drag and drop. 

The individual preset name of the plug-in effect is 

also now visible, rather than just the effect. 


VI review 

Fig. 4: The Prologue analog-style synth. erful EXS24 sample player built in. 

Fig. 5: Spector is a spectrum filter synth, one of 

These new VST instruments have more 

than 1,000 instrument sounds, drum kits, and 

the new instruments included with Cubase 4. 

While most of these are familiiar, the VST effects to offer, so it’s becoming increasingly 

instruments are completely new. Prologue possible to compose entirely using internal compliant, meaning that it runs on PowerPC 

(Fig. 4) is an analog-style synthesizer, Spector instruments rather than having to spend and Intel Macs. 

(Fig. 5) is a spectrum filter synthesizer, and more money adding plug-in instruments. 

New additions such as the Control Room 

Mystic is an impulse/comb filter synthesizer. Steinberg obviously still has other soft instru- Window and the Media Bay/SoundFrame 

These are all based on individual pages of the ments available—their high-quality Steinberg browser (Fig. 6) combination for searching and 

flexible Steinberg D’Cota softsynth, and offer Grand piano for example—but particularly at accessing the presets of VST instruments and 

various numbers of oscillators and sound syn- the price of a package like Cubase 4, it’s only effects running within Cubase should speed up 

thesis techniques together with hundreds of reasonable to expect to find a wide range of the work process considerably. On the subject 

preset sounds, promising much greater syn- powerful instrumentation included. 

of speeding workflow, for those who like to use 

thesis power than the simpler A1 and other 

key commands as much as possible and avoid 

analog-style synths built into earlier version of Progression 

the use of the mouse, Cubase 4 has a com- 

Cubase. 

Steinberg has made a lot of progress with pletely revised set of key commands; 

Halion One in contrast is a stripped-down Cubase 4. A full printed handbook is now Deactivate All Solo and Unmute All are going 

version of Steinberg’s Halion sample playback supplied, not just a set of PDF files on disk, to be particularly useful, as are being able to 

software. This is a very simple instrument with and between this clearly written 600-page select altenative mixer views and open the 

envelope attack and decay, filter cutoff and monster and the 75 pages of Getting Started Control Room, MediaBay, SoundFrame, and 

resonance settings, however its “ROM” 

info, Cubase 4 is not going to be difficult to Loop Browser windows quickly. 

sounds come from the Yamah Motif key- understand. The cosmetic appearance has 

And if you want to limit the budget a little? 

boards. This helps put Cubase 4 more on a changed, but not all that much (the new VST Cubase Studio for around half the cost offers 

par with Apple Logic 7, which offers the pow- instruments look pretty sleek though, and many of the new improvements. It comes 

rather similar to their Apple Logic equivalents) without the Spector and Mystic instruments, 

and some major annoyances such as the split- Surround facilities, or full MediaBay searching, 

Fig. 6: The Soundframe browser, which is functing of mixer channels between the MIDI and and it has fewer VST effects. But it includes 

tionally much like Native Instruments’ Kore, makes audio facilities of virtual instrument tracks the Halion One sample player and Prologue 

it easy to manage and categorize large libraries of have been eliminated. Most importantly, analog-style synth, and it has similar multi- 

sound by name. 

Cubase 4 is Universal Binary/Intel Mac OSXtracking and scoring abilities. 


Score? 

Cubase works smoothly and reliably without 

any needing any settings in the background, 

and many of the layout aspects that 

were less than perfect—such as the channel 

split between MIDI and audio facilities of virtual 

instrument tracks—have now been 

removed. The new VST instruments in Cubase 

4 are powerful and flexible, and having a 

built-in sample-playing instrument is a massive 

bonus. 

Users on a limited budget can readily 

upgrade from Cubase Studio, while users on 

an unlimited budget can one day upgrade to 

the even more audio/visually-oriented 

Steinberg Nuendo. Cubase 4 is a strong candidate 

for either Mac or PC users as the center 

of any recording studio. VI

andom 

tip 

Importing 

Yamaha 

DX-series 

patches into 

Native 

Instruments 

FM8 

It’s hard to imagine anyone who worked with 

music technology in the 1980s and didn’t own 

a Yamaha DX-series instrument at one time or 

another, most likely a DX-7. That means many 

of us have old projects that use those sounds. 

It also means that billions of people were 

programming sounds for DX instruments, and a 

lot of those sounds are still very good (never 

mind that a lot of them are also still very bad). 

Native Instruments FM7 and FM8 have a great 

feature: the ability to import DX-series sounds 

and play them back at 2006 sound quality. 

Fortunately getting those sounds into FM8 

(or FM7) is a lot easier than choosing which 

ones to import. FM8 can accept System 

Exclusive (SysEx) Data from Yamaha’s DX7, 

DX7II and DX200 synthesizers and convert 

them into its own parameter format. 

There are two approaches; which one you 

choose will depend on whether or not you 

already own a DX series synthesizer. 

A. Connect your DX-series synthesizer to your 

computer via a standard MIDI interface. In 

stand-alone mode (rather than using it as a 

plug-in) FM8 will automatically receive any 

MIDI SysEx data, in other words the sounds just 

go in over MIDI. Single Presets go into the edit 

buffer and you must save them manually; Banks 

are converted into .ksd files and are saved automatically. 

It’s not possible to receive SysEx data 

via MIDI when running FM8 as a plug-in due to 

the current limitations of the plug-in standards. 

B. Load SysEx data from a file. To do this, 

simply click on the Import SysEx button in the 

File menu, navigate to a compatible SysEx file, 

and open it. Note that SysEx files will have a 

.syx extension under Windows. On the Mac, it’s 

usually a good idea to add this extension to the 

file before attempting to import it. VI 

1. First choose “Import SysEx” from the File menu 

2. FM8 converts a DX7 SysEx bank to the .ksd format 

3. The converted DX7 SysEx bank appears in the KoreSounds browser for use in FM8. 


VI 

interview 

NATHAN FURST 

(CONTINUED FROM PAGE 57) 

combination of sample layer crossfading on 

the mod wheel and controller 11 to sort of 

duck in and out. To me that’s the best chance 

you have to making your strings sound real. 

“Though None Go With Me”—that’s a 

more recent piece, isn’t it? 

Yes, that was another TV movie I did probably 

ten or eleven months ago. I can’t say it’s 

the best example of my MIDI production 

work. That score is about 100 minutes of 

music, and I scored that in two weeks. 

I play in the line I am using in the violins, 

and just quickly flesh it out. Because I think in 

terms of intervals, sometimes I have to play 

EWI 


right-angle MIDI plugs, which aren’t all that 

easy to find. 

The synth 

The EWI 4000s contains your basic modeled 

analog synth. As you can see in Fig. 4, 

the screen dump from the included UniQuest 

editor for Mac and PC, it has two identical 

oscillators, each with faders for sawtooth, triangle, 

and square waves. It also has a noise 

generator, which you can use for breath noise 

or just for noise in the traditional way. 

Notice that in place of envelopes, the oscillators 

and filters are controlled by breath. 

There’s also a formant control for the oscillator 

filter with woodwind and string settings. 

The delay/chorus/reverb (in that order, not 

the way it appears) effects section is quite 

basic, but it’s fine for what seems to be its primary 

application: live performance. What you 

VOLTAGE CONTROL 


Envelope; they’re essential in the creation of 

this sound. Though the filter envelope controls 

the timbre of the sound, it is the amplifier 

envelope that truly shapes the sound, 

allowing the crack to happen immediately at 

significant volume, while the thunder gradually 

fades to silence. 

The settings of the Release segments of 

both envelopes allow the sound to be played 

in two different ways. Strike a key and immediately 

let go, and you’ll get a crack followed 

by thunder. Hold the note and you’ll get a 

crack that gradually turns into rain. 

Also note that the Compressor is enabled, 

which helps fatten the sound. Stereo Delay is 

also used; the setting doubles the sound and 

spreads it out over the stereo soundstage. If 

you’re looking for something less dramatic, 

simply turn off the delay. 


two parts at a time. Say, I’ll play a couple of 

parts on the cello and just move the bottom 

note that I played down to the basses. Most 

of the samples at that point are going to be 

in combination of VSL [Vienna Symphonic 

Library], a different custom library, and Sonic 

Implants. 

And I love, love, love Sonic Implants. It’s 

gorgeous and it’s interesting because it’s actually 

very “under-programmed.” Most of the 

patches have no velocity layers, it’s a single 

thing. If there are multiple velocity layers it’s 

only one or not very many—it’s not heavily 

programmed. They got the timbre right when 

they recorded it. To me it’s a gorgeous 

sounding library. 

I also use a lot of SAM. I love everything 

that Maarten Spruijt does. 

don’t see in the screen dump is that all the 

parameter settings are shown when you 

mouse over them, so for example what looks 

like a totally rudimentary delay can be set in 

10 millisecond increments up to 1.27 seconds. 

But the reverb is what it is—you control 

the time, density, and high frequency damping, 

and of course the level, but nothing as 

sophisticated as, say, the type of room. You’ll 

want to use a studio-grade reverb with less 

grain for recording applications; this is a very 

mono reverb for a mono instrument, 

designed for live use. 

While the sound of this synthesizer is good, 

there are some nice factory presets, it’s a lot 

more versatile than you might gather from 

looking at the screen, and it’s great having 

good sounds built right into the instrument…and 

the last thing I want to do is sell it 

short…this is really a pretty traditional synth 

that wouldn’t turn too many heads if it 

weren’t attached to an EWI. 

The Program 54, Rolling Thunder, uses the 

same basic approach as Lightning Strike, but 

the Attack time of the Amplitude Envelope 

has been increased to eliminate the crack and 

the LP24 filter is used to deepen the sound. In 

addition, the two LFOs are enabled, each 

routed to the filter, but programmed with different 

speeds and amounts. This is what creates 

the “rolling” effect as the thunder plays 

through. 

One final note: Filter Tracking, which raises 

the filter cutoff level as you play higher up on 

the keyboard, is enabled. This allows you to 

play deep, low rumbling thunder at the low 

end of the keyboard, and thunder that 

evolves in howling wind at the top. 

Stop the noise! 

Two full-blown articles later, we’ve only just 

begun to explore the possibilities noise offers 

us for sound design. Everything from gunshots 

to flutes to jet planes to A-bombs to 

What about brass? 

Same thing. I always, always, use controllers 

1 [mod wheel] and 11 for virtually 

everything, all the time. Sometimes this produces 

a very noticeable, dramatic effect, and 

sometimes very subtle. I find that the instruments 

sound the most real when they have 

small fluctuations in dynamics—which translates 

into layers as well as the timbre of an 

instrument when going from ƒƒ to mƒ and 

then back again. 

I mostly do this as I go. Very rarely do I go 

back and add more CC info the next day or 

something. I can’t move on to the next bar if 

the legato line doesn’t have the delicate tail 

out that I want to hear. OCD I guess! VI 

And that right there is precisely the point: 

attached to an EWI, it doesn’t take much 

more than a sine wave to sound really good. 

We have at least one article in the works on 

programming instruments for wind control. 

It’s the way to go. 

Also, it’s worth pointing out that the builtin 

synths in previous EWIs were also quite 

conventional. About the only thing the 4000s 

doesn’t have is an external input for controlling 

other synths alongside the built-in one. 

Conclusion 

After 25 years and several generations with 

all the glitches worked out, the EWI is a 

tremendously satisfying instrument to play. 

This latest one has a lot of new things going 

for it, not the least of which is that it’s now 

down to a more accessible street price. 

The EWI is an instrument that opens up a 

whole new world of synthesizer control. I 

can’t recommend it more highly. VI 

drum kits to ethnic percussion to wild and 

wacky sound effects can be generated using 

that annoying stuff we spend so much time 

trying to eliminate. 

So the next time someone asks you about 

the signal-to-noise ratio of your system, puff 

out your chest, crack a big smile, and tell them 

you’ve got the noisiest system on the planetæand 

a host of great sounds to prove it. 

Michael Marans, often accused of being overly 

noisy, wishes to inform VI readers that no animals 

were harmed in the creation of this article. 

However, several windows were shattered and 

his earthquake insurance premiums were raised 

dramatically. VI

SEQUENCING SAMPLES 


Naturally, an intelligent orchestration will 

offer the best results, but it is not “wrong” to 

give particular boosts to certain instruments 

or instrument groups. When doing so it’s 

important to give a slight EQ boost in the 

TRENDS 


Peter Dines: 

1. Linux will make greater inroads into 

computer music with the release of Energy XT 

for Linux. This will spur more developers— 

smaller, more agile ones at least—to try cross 

compiling their products. However, Linux support 

from sound card and MIDI controller 

vendors will continue to be abysmal. 

Peter Buick: 

1. Computers will continue the trend of 

being more powerful for less money. But I see 

64 bit as a (fairly poor) interim to the real deal. 

To gain/retain a unique edge in music production, 

the elite will invest more in DSP card 

power and tactile hardware controllers. The 

mass will remain unimpressed with the deluge 

of “me too” alternatives out there already, but 

still refuse/be reluctant to learn how to make 

more out of what they already have. 

Apple Garage Band and the like will 

become even more powerful (and include 

video) and ultimately allow end user listeners 

to be interactive, instead of passive, as finally 

a multitrack mp3 format (probably including 

video) standard is defined. People will be able 

to do Acid-style re-mixes on their mobile 

phone (okay, by 2008 then). 

By the way, my New Year’s resolutions: I 

will index my sample library (likely to be broken). 

I will listen to more other artists’ music 

(including socio/eco sites like garageband and 

myspace), especially music from other musical 

styles and cultures. I will limit myself to make 

more out of what I already have by having 

one half day programming session per week 

where I will not start a song, no matter how 

inspired I am by my tweak. I won’t eat too 

many mince pies at xmas (likely to be broken). 

I won’t use presets 1-10 any more. 

2. Technology will go full circle and we’ll 

return to “people” interacting —on-line, with 

more virtual studios, rehearsal rooms and 

gigs. Several real-time sharing sites have 

emerged recently, and more are about to. It’s 

time people stopped playing with themselves 

in their bedrooms. Even the porn industry lets 

punters interact with models (i.e. web cam, 

text instructions) in real time. 

IMHO I think the single-wire studio is a fallacy—what 

people want is one single box 

that’s powerful enough to do all. We only tolerate 

multiple boxes because that promise is 

nowhere near true yet—nor is it with 64 bit. 

high frequency range of that instrument/ 

instrument group, and pull down some of the 

mids in order to make it appear slightly closer 

(without saturating the mix) as well. 

Join us next time for some exclusive tips on 

dynamics, including how to sequence a 

crescendo properly. 

David Das: 

I think 2007 will shape up to be the Year of 

Unlimited Processing. 

The strides that have been made in this 

direction are just starting points compared to 

what they will evolve to be. Things to consider: 

• Apple Logic Pro’s nodes (which offload 

processing to remote computers on the netword). 

They only work for Logic plug-ins at 

the moment, but wait until Apple evolves 

them and/or other DAW companies go on 

board with stuff like this. I bet other companies 

will come along and do it better than 

Apple. Soon the only limit to your plug-in 

count will be how many computers you can 

afford—and they’ll all work perfectly integrated. 

(Okay, okay, honestly, this will probably 

not see ultimate fruition in 2007; it’ll be 

longer than that. But it makes for nice copy.) 

• The popularity of powered plug-ins like 

UAD, TC, Waves’ new hardware boxes, and 

(half-finished but always in the race) Metric 

Halo [also SSL Duende]. Prices of these boxes 

are coming down too. Now anyone can add 

boutique plug-in power to a native workstation. 

And they (mostly) sound really good. 

• Apple’s switch to Intel chips, which are a 

more universal platform and more hackable. 

We’re already seeing isolated reports of hackers 

swapping 8-core chips into a Mac. If this 

goes further—like seriously mainstream and 

people start hot-rodding real DAWs that can 

actually take advantage of it—there’s another 

speed limit gone right there. Intel chips are 

readily available to end users. 

I remember back when the first version of 

Altiverb was released and it was famous for 

being the first plug-in (nothing else came 

close) that would bring a then-state-of-the-art 

G4/400 to its knees with one instance... 

All this to say that the bottleneck—CPU 

power—for plug-in fans is being abolished, 

and I bet 2007 will be a banner year for this. 

Jason Scott Alexander: 

1. Definitely! 64-bit is going to be a big 

thing—I’m just not so sure if it will hit full 

stride any time imminent (on the PC that is). 

First we must get MS to roll out Vista in a big 

way. Third party softs won’t be far behind, I 

don’t think. 

2. The whole concept of “chip farms,” or 

motherboards with many (not just two) 

processor sockets. According to the chipmakers, 

they will only be able to cram so many 

“cores” onto a single processor before they 

have to resort to multiple physical processors 

VI 

review 

Thomas J. Bergersen (contact@thomasbergersen.com) 

Bergersen (25) still lives in Norway. 

He is a composer working in the media music 

industry. His credits include hundreds of productions 

in film, TV, trailers, radio, and video 

games, as a composer, orchestrator, and 

arranger. VI 

to quench our thirst for power. We’ll start seeing 

mobos with rows of processor sockets 

similar to the way we currently see PCI-X or 

RAM module sockets. Then you’ll simply add 

processing power either in series or in parallel. 

Another cool concept that’s already in use 

at the enterprise system level (mainframes, 

servers, etc.) is hardware virtualization: taking 

a single physical computer system and, 

through its OS, breaking it up into discretely 

operating and appearing systems. A single 

super duper high-speed processor could operate 

discretely as a very high-end signal 

processor, with drivers to work specifically 

with its own memory and a certain piece of 

hardware I/O, while another “part” of that 

same processor could be told to operate as a 

discrete high-end sampler, again addressing 

its own memory and I/O. 

Just like old studios with lots of hardware 

boxes, this is highly reliable (if one crashes, 

the others don’t) and highly powerful (no 

sharing of resources and much lower system 

overhead for each one). The effects processor 

doesn’t need all the crap that Windows runs 

in the background, and it could run on a very 

lean and mean OS that is different from the 

needs of the sampler. 

A final really cool idea to chew on (and this 

eludes to your everything-down-one-pipe 

concept) is that of companies designing 

breakout DSP that streams extremely high 

bandwidth data directly off the PCI bus to 

outside hardware. Need more? Just add 

another DSP box. Latency and conflicts won’t 

be a problem anymore. 

Frederick Russ: 

Someone will develop a virtual instrument 

player à la Vienna Instruments that it will be 

universal and work with all libraries—Sonic 

Implants, Project SAM, etc. 

Virtual Instruments Magazine will quadruple 

in size and scope, will be available in all 

stores, and go monthly. VI 


VI trends 

Trends 

VI writers stir the tea leaves 

Jim Aikin: 

Thomas J. Bergersen: 

I. have to agree. :) 

1. 64-bit computing—who really cares? 

2. More famous hardware units turned into 

Not musicians, that’s for sure. Manufacturers, software plug-ins (more a wish than a predic- 

you betcha. The more people they can sucker tion), even more bloated software and sample 

into upgrading, the more money they make. libraries, new cross-platform plug-in standard, 

The trend I’m seeing in synthesis—and I new higher resolution MIDI interface standard 

expect this to gather steam in 2007—is a shift (more wishful thinking rather than a predic- 

in emphasis away from the filter and toward tion, though :), tougher copy protection fol- 

the oscillator. For many years, the sound quality 

of the filter has been considered a makelowing 

the Vista OS. 

or-break deal for a synth, but by now the Orren Merton: 

problem of how to make good-sounding digi- 1. Slowing down of exact hardware emulatal 

filters seems to be pretty much solved. tion V.I.s. I think we’ll start to see fewer and 

Meanwhile, advances in computing power fewer models of Moogs, Arps, Prophets, and 

have made it practical to do waveshaping other classics, and more VIs that try to cap- 

within the oscillator itself in complex and ture the zeitgeist of many different analog 

great-sounding ways. Three instruments that I synths, put in a new interface without the 

know of illustrate this: Cakewalk Z3ta+, NI limitations of hardware. 

Massive, and u-he Zebra2. I expect others 

More hardware-based copy protection. 

have appeared that I don’t know about, and Piracy is absolutely ravaging the smaller play- 

Just for fun we sent a very simple note to 

our writers, asking them two questions 

(with leading answers just to get the 

still more will show up before too long. 

I predict that Steinberg’s announcement of 

VST 3 will get a lukewarm response, and that 

ers, and we’re going to continue seeing more 

softsynths using dongles, integrated controllers, 

and other related hardware. 

crystal ball rolling): 

support from other manufacturers for the 

2. Hardware synths will all offer VI front 

1. What predictions do you have for the VI new spec will be spotty, at least until Q2 of ends. We’re already there with the Virus TI 

world in 2007? [NB’s example: 64-bit com- 2008. 

and the Roland SH-201, but as time goes on, 

puting—which you are free to agree with or 2. Wire? Why do you need a wire? The fully all hardware synths will need to offer a VI 

not.] 

wireless studio is only a matter of time. 

front end for integration into the DAW. 

2. How about for the future in general? (Thanks to Dave Smith for suggesting this 

Lee Sherman: 

[NB’s answer: everything will be connected utopia to me.) Walk into the studio, plug in a 1. The few remaining classic instruments 

by a single wire. We’re almost there with pro- power cord, and your hardware can talk to that have yet to be emulated will find their 

grams that send audio and/or MIDI over anybody else’s hardware wirelessly. Audio way into software form. 

Gigabit ethernet, but in a couple of years channels, MIDI, all configured automagically. 2. We’ll see entirely new virtual instruments 

Fibre Channel or something else should make You know what I’d like to see, but don’t that don’t just mimic classic hardware but 

this the norm.] 

expect to see anytime soon: a resurgence of instead go off in exciting directions made 

enthusiasm among the general public for possible by today’s user interface paradigms 

Chris Meyer: 

music made with electrons. It seems to me and processing power. NI’s Massive is a prom- 

1. Sound libaries will continue to grow to that the problem of how to make amazing ising first step in this direction. 

the point that once we've installed them, sounds with computers is totally solved. If I Gary Eskow: 

there won't be room on our internal drives to never installed another plug-in, it would take Obviously, 64 bit computing, although the 

install the next major upgrades to MacOS or me a couple of hundred years to explore the only place it seems necessary to me is with 

Windows next year. 

musical potential of the stuff that’s on my my VSL Vienna Instrument samples... I’m per- 

(Which really brings the need for licensing hard drive today. But the career opportunities fectly happy with the processing I’m getting 

schemes to easily accomodate them being lag very far behind the technology. What we on my dual Opteron pre-64 bit right now. 

installed on external drives that we can move need is... I don’t know, a Keith Emerson for More and more fantastic classic synth mod- 

between our studio towers and live perform- the new millenium, I suppose. Somebody els.ance 

laptops—not—currently the case.) 

who captures and ignites the public imagina- 

2. Um, see parenthetical aside above. tion by jamming with a laptop. 


64 VIRTUAL INSTRUMENTS

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