Virtual Instruments V01#03 Dec-Jan 2006

OUTSTANDING
PRODUCTS REVIEWED:
EAST WEST QUANTUM LEAP
SYMPHONIC CHOIRS PROMINY
LPC ELECTRIC DISTORTION AND
CLEAN GUITARS LIBRARY NOTRE
DAME DE BUDAPEST PIPE ORGAN
LIBRARY QUANTUM LEAP
PERCUSSIVE ADVENTURES 2
DECEMBER/JANUARY 2005/6 - VOL. 1 NO. 3
www.VirtualInstrumentsMag.com
USA $4.95
CANADA $6.50
THE WORLD OF SOFTSYNTHS AND SAMPLERS
THE WORLD OF SOFTSYNTHS AND SAMPLERS
Cakewalks Project5 all-in-one workstation
A Very Deep Clinic
The Making of
GigaStudio:
behind the
revolution
Exclusive, free algorithmic songwriting software download from www.VirtualInstrumentsMag.com.
WIN a DrumCore plus 3 DrummerPacks in our latest MUNGO GIVEAWAY!
WIN a DrumCore plus 3 DrummerPacks in our latest MUNGO GIVEAWAY!
Native Instruments
Kontakt 2 sampler
a Very Deep Clinic and a review
The New
Sampling

2 V I R T U A L I NS T R U ME NT S
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Editor/publisher: Nick Batzdorf
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T
here are two phenomena that led to the virtual
instruments revolution that caused you to be reading
this magazine. One, of course, is that personal com-
puters have enough power to run V.I.s.
The other is The New Samplinglarge, disk-streaming
sample libraries, or modern sample libraries as we refer to
them. That revolution started after Nemesis GigaSampler
(now TASCAM GigaStudio) came along a few years ago, and
were happy to be able to present an interview with the peo-
ple who invented Giga: Jim Van Buskirk and Joe Bibbo.
Today, modern sample libraries either come in their own
V.I. players or theyre programmed for specific sampler
platformsas you can see from the reviews in these pages.
(Some people differentiate between the two by calling the
player versions V.I.s, but around here we dont bother.)
Either way, its fair to say that as wonderful as sampling
technology is, many of the more advanced libraries have
been ahead of their performance interfaces for some time.
Modern sample libraries usually have lots of articulations to
choose from, and the available means for choosing the
right ones in real time have been limited; making sample
libraries sing is usually as much a matter of programming
as it is playing.
Thats about to change. Among other tricks you can
read about in this months Trends column (page 64),
Vienna Symphonic Librarys new Vienna Instruments player
loads as many articulations as you need for an instrument
in one instance of the plug-in. It puts everything on one
sequencer track and one MIDI channel; that lets you select
articulations on the fly.
But the really interesting trick is that it automatically
switches to the right articulations a lot of the time. It does
that by sensing whether youre playing legato, repeating
notes, playing a pattern or trill, orand this is the clever
oneit knows how fast youre playing, and from that it
knows whether to play a shorter or longer articulation.
Now, that doesnt mean the whole problem is solved.
Youll still have to do some programming, if for no other
reason than that it may take more than ten fingers (and
one brain) to deal with all the switching you need to do in
spite of VSLs performance detection. But this is a pretty
exciting step in the right direction.
And the best part is that its already possible to incorpo-
rate some of these ideas using existing tools such as Giga
and Native Instruments Kontakt 2. If history is an indica-
tion, we can also expect to see leapfrogging advances in
other sample players. East West has already announced
that their upcoming player will include speed-sensing and
other innovations, for example.
2006 looks like a very exciting year for our industry. All
of us at Virtual Instruments wish you a happy, healthy, and
prosperous one. VI
Standard disclaimer: Virtual Instruments Magazine
and its staff cant be held legally responsible for
the magazines 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 sub-
ject to international copyright protection, and
you may not copy or imitate anything without
permission.
2005 Virtual Instruments, Inc.
First DAW
by Nick Batzdorf
A beginners guide to putting together a DAW (digital audio
workstation) for softsynths and samplers. Part 2: audio and
MIDI hardware
Letters
Launch
Introductions, updates, news
6
12
10
40
December/January 2005/06
V1.N3
Buick Sessions:
Sound design in music
production
by Peter Buick
Download a free algorithmic lyric and music program from
34
Very Deep Clinic:
Power in Project5
by Jim Aikin
Discover some less-obvious features of Cakewalks all-in-one
virtual workstation
16
Thinking About Reverb
by Dave Moulton
An audio gurus gurus reflection on reflection
Moderately Deep
Clinic:
Disk-streaming
Settings in Native
Instruments
Kontaktseries
Samplers and Players
by David Das
Adjusting those mysterious advanced DFD settings to get
the best performance out of your machines
24
V I R T U A L I NS T R U ME NT S 5
VI
VI
contents
reviews
51, 56
Notre Dame de
Budapest Pipe
Organ
by Bruce Richardson
One of the worlds most deva-
statingly impressive sample libraries
East West
Quantum Leap
Symphonic
Choirs
by Nick Batzdorf
Great singers, great engineer, three
separate mic positions in a great hall,
and a utility program for making it
sing the words you want.
Native
Instruments
Kontakt 2
by Doyle Donehoo
A major revision of this popular
sampler adds convolution reverb,
support for 64 MIDI channels, a new
interface, and a whole laundry list of
new features.
Percussive
Adventures 2
by Chris Meyer
The second volume of the popular
original loop and hit library is eclec-
tic and fresh.
Prominy Les Paul
Custom Electric
Distortion and
Clean Guitar
library
by Craig Sharmat
The worlds largest single-instrument
sample library isnt all about size.
December/january 2005/06
V1.N3
Interview:
GigaInventors Jim Van
Buskirk and Joe Bibbo
The men responsible for GigaStudio and the new sampling
revolution take a look backward, forward, and straight ahead
44
Random Tips:
Groove quantizing music that isnt a groove;
Memory access in Mac G5s
Trends: VSLs Vienna
Instruments Player
VSLs new player brings their huge sample library under
real-time controlon one sequencer track and MIDI channel
64
20
28
48
50
52
random
tip
Lay of the LAN
I primarily subscribed to your mag for the
Lay of the Lan article by Monte McGuire in
your 9-10/05 issue, and I have no problem with
that, but what I do have some difficulty with is
the actual peripheral implications. As a tiny
example, I cant imagine how you can use a
number of different computers without their
monitors, but equally I cant imagine someone
who has 5 or 6 GigaStudio dedicated machines
plus their DAW having a room full of monitors
either. The answer to that particular one (Can
one monitor serve several machines?) seems to
be really difficult to find out.
Thanks for a brilliant magazine and I look
forward to a great deal of pleasure reading it
and benefiting from it over the months and
years to come!
John Sullivan
Cheshire, England
Thanks very much, John.
Youre right to assume that you can share
monitors between multiple computers, along
with keyboards, mice, and often peripherals
such as printers. There are two ways to go
about that: over a network, and using a KVM
(keyboard, video, mouse) switch. Or a combi-
nation of the two.
We touched upon the network method in
the sidebar on page 54 of our last issue.
There are many Mac and Windows freeware
or shareware programs that use the VNC pro-
tocol, which displays the remote computer in
a window on your master machine; you just
click in the window and access it as if it were
the active window on the local computer.
Apple Remote Access is a commercial pro-
gram for Macs that does the same thing, and
Windows has remote access capability built in
too.
The problem is that VNC is fine for things
like loading programs, but its too sluggish for
anything more extensive than that; you
wouldnt want to operate a DAW over VNC.
So as enticing as the idea of just using ether-
net is, most people who use multiple comput-
ers wouldnt be happy without a KVM switch.
Ive seen 2-port PS2 and VGA KVM switch-
es (with keyboard hot keys) as low as $29 and
USB 2-port models in the $60 range, but
there are a lot of 2- or 4-port models in the
$120 200 range. There are also some high-
end ones from companies such as
www.Gefen.com, who specialize in unusual
solutions for our industry.
In addition to quality, there are features
that raise the price of KVMs. One is dual-
monitor switching, or digital DVI monitor
connections rather than analog VGA. Many
KVM switches regardless of price can be oper-
ated by hot keys from your keyboard, for
example you might hit control, shift, option
in rapid succession followed by a number to
specify which machine to switch to; others
work by wired or wireless remote controls.
There are also some compatibility issues.
Solid as our KVM switch (IOGear Miniview III
USB) is, it isnt happy with the Apple Mighty
Mouse. Gefens switches reportedly have no
such issues. We plan to take a look at a few
models side-by-side in the near future. In the
meantime you could check out www.kvm-
switchesonline.com. They carry a wide variety
of models.
Sometimes it can be challenging to figure
out these set-ups, especially when youre
using multiple monitors. In our studio, for
example, we have two monitors and four
machines: a G5 Mac (the main machine) and
three Windows computers.
What we do is connect one
monitor to the G5 all the
time using a DVI connec-
tion, which is noticeably
sharper. The second monitor
is on a KVM switch that
goes between the G5 (for
dual-monitor operation) and
the three Windows
machines.
This works perfectly,
except for one thing: you
can always see the main G5
screen, but you cant oper-
ate it while the keyboard
and mouse are switched
over to one of the Windows
VI
l e t t e r s
Letters
write to:
nb@virtualinstrumentsmag.com
Gefen makes higher-end KVM switchers in all
shapes and sizes. This model switches two DVI
monitors between four computers.
The IOgear 4-port MiniView SE
USB shares one VGA (analog)
monitor, keyboard, and mouse as
well as USB peripherals between
four computers. It sells in the
vicinity of $150.
Gig Firewire 800 7200 RPM Raid Drive. Id love
to learn what the settings in Native Instruments
Kontact, Kompact, etc. do as it relates to Direct
From Disk. I frequently get pops and clicks when
loading and playing back samples. I use Logic
Pro 7.1 as a host. It performs better as a stand
alone instrument than as a plug-in. But there is
no documentation on optimizing those settings.
I realize that a G5 with more RAM will help but
until Apple makes a more powerful laptop this is
the best you can get. A article addressing laptop
optimization would be great.
2. While I use East West Quantum Leap
Symphony Orchestra Gold and soon Pro XP, an
article teaching how to effectively work with an
orchestral library and the many articulations
would be real helpful. Im sure that much of it
would apply to any Orchestral library. I also
expect that anyone spending the money on a
library like that is probably using (or will spring
for) Kontact 2 rather than using the supplied
Kompakt interface.
Anyway, keep up the good work.
Michael D. LaPoff
Fort Wayne, IN
Thanks Michael. Were first covering the
second topic, working with orchestral
libraries, in depth
next issue. That skill
is an important part
of this new musical
medium, and youre
right that we need
more of it.
Hopefully youll
find the clinic in
these pages about
optimizing Kontakt-
family DFD settings
helpful with the first
problem. You may
need larger buffers
to give the system
more breathing
room.
Another thought
is that you might try
using 7200 RPM
FireWire drives
rather than either
the 5400 RPM ones
or the RAID set-up. It might seem counterin-
tuitive, but RAID isnt always better on Macs
for streamingalthough that is supposed to
change with some newer software.
Also, do you have a FireWire interface in
the same bus as your hard drives? You might
try picking up an inexpensive PC slot FireWire
card so you can separate the drives and inter-
face on different busses. Your PowerBook
should have enough processing power to run
those libraries without clicks and pops, and
you should be able to load three instances of
the EWQLSC Gold player in 1.5GB of RAM
pretty comfortably.
Im not sure its true that every EWQLSO
user just runs out and buys Kontakt 2. The
player works very well.
More what we want
A great magazine. How bout a per maga-
zine focus section for we laptop users who are
relatively new to pro audioin particular Id like
to see a survey of audio interfaces that connect
to my mac. Right now Im just taking the audio
output of my PowerBook straight to my volume
pedal then to my amp. Sounds pretty good,
though Im sure there must be a better way.
Don
Somewhere where theres email
The First DAW article on hardware in this
issue should be a good starting point, but we
didnt mention that a lot of audio interfaces
are designed with guitarists in mind. Line6 is
the most obvious example (see picture of
their TonePort UX1) but other companies
have them too.NB VI
VI
l e t t e r s
computers. So were about to experiment
with a donationware open-source program
called Synergy
(http://synergy2.sourceforge.net/index.html),
which sends just the keyboard and mouse
over ethernet to (in our case) the Windows
machines.
The keyboard and mouse require far less
bandwidth than video, and weve seen
Synergy working really well; you just move
the mouse cursor past a prescribed edge of
one of the monitors, and the cursor appears
on the next computers monitor. What
remains to be seen is how reliable a solution
this is over time.
What we want
I love the magazine. Here are a couple of
subjects Id like to learn more about.
1. Optimization of a laptop based VI system.
Specifically my problem involves using East West
libraries (Colossus, RA, EWQLSO, Symphonic
Choirs). Im using a PowerBook G4 1.67 with
1.5 gig RAM and 2 external 2-1/2 inch Firewire
400, 5400 RPM drives. Also have a G-RAID 500
A blatant example of a guitar-oriented audio inter-
face: the Line6 TonePort UX1.
1 0 V I R T U A L I NS T R U ME NT S
Launch
Introductions, updates, news
VI
l a u n c h
Garritan Personal
Orchestra Sibelius
Edition
This newest version of GPO integrates
seamlessly with the Sibelius notation
program, automatically reading and
interpreting performance indications
from the score (articulations, playing
techniques, slurs, dynamics). The
library comes with a dedicated, 32-part
multitimbral Native Instruments Kontakt
sample player, with reverb and several
parameters to adjust. Export your music
to an audio file or convert it to an MP3.
$219.
http://www.sibelius.com/products/gp
o_sibelius/ 888-4-SIBELIUS or
www.Garritan.com 360/376-5766
The FireWire version of MOTUs 828mkII ($795) is still very much around, but now theyve
released an otherwise identical USB 2.0 version. Features include a total of 20 inputs and 22
outputs, including eight 24-bit 96kHz analog I/Os, 8 x 8 ADAT optical and S/PDIF digital I/O,
CueMix DSP near-zero latency monitoring, stand-alone operation, a MIDI I/O, all the standard
Mac and Windows drivers, and MOTUs AudioDesk Mac audio software.
www.MOTU.com 617/576-2760
BIAS Peak Pro 5 and Pro XT 5
The latest versions of BIAS 2-track editor features VST and AU Instrument support, new pitch
shifting algorithms, 100% Redbook Replication-ready CD burning, a new playlist design, auto-
matic plug-in latency compensationand about an 8-inch list of additional new features. Peak
Pro 5 lists for $599, and Pro XT 5 (which includes the Master Perfection Suite of six new plug-
ins) is $1199.
MOTU 828mkII High-Speed USB 2.0 Audio Interface
E-MU Modern Symphonic Orchestra
This new 10GB orchestral library has been programmed for E-MUs Emulator X and Proteus
X Desktop Instruments. It cycles through multiple samples with real-time control over bowing
techniques and articulations. $329.99
www.emu.com 888/372-1372
Vienna Symphonic Library Symphonic Cube
VSL has expanded their orchestral library into a 550GB 24-bit/44.1kHz collection.
The new version is now sold in ten instruments: Solo Strings, Chamber Strings,
Orchestral Strings I and II, Harps, Woodwinds I and II, Brass I and II, and Percussion.
Each of the ten instruments comes with its own Vienna Instrument
VST/AU/stand-alone player, featuring Performance Detection that interprets inter-
vals, repeated notes, patterns, and speed to switch to the right articulation auto-
matically. Please see Trends, p. 66 for a more in-depth look at the Vienna
Instrument.
www.VSL.co.at or www.Ilio.com 800/747-4546
V I R T U A L I NS T R U ME NT S 1 1
VI
l a u n c h
Digidesign Pro
Tools 7 Software
In addition to substantially
greater native processing efficien-
cy on Mac OS X, the latest version
of the Pro Tools for Mac and
Windows XP now has Instrument
tracks that make it much simpler
to use virtual instruments. It also
features real-time MIDI processing
(quantizing, changing note dura-
tions and velocities, transposition,
and timing), new groove quanti-
zation features, sample-based
MIDI tracks, mirrored MIDI editing
(edits apply to copies as well as
the original), and more.
The Pro Tools software works
with Digidesign HD, 002/R and
M-Box 2 systems, and theres now
a $300 M-Powered version for
most M-Audio interfaces.
www.Digidesign.com 650/731-
6300
IK Multimedia Sonic
Refills for Reason 3
Ik Multimedia and Sonic Reality have
introduced a 20-volume collection of
newly-enhanced sounds for
Propellerhead Reason. Each of these
refills is $59 or the whole set is $499.
The list includes Synths, Rhythm
Section, Pianos and Organs,
Symphonic, Retro Keys, Acoustic Folk, and more.
www.Sonicrefills.com 954/846-9101
VirtuosoWorks Nutcracker and
Messiah Scores
Users of the Notion music composition and performance soft-
ware can load these performance-read orchestrations for Notoins
interpretive playback engine, triggering samples of the London
Symphony Orchestra recorded at Abbey Road Studios. You can
conduct the score in real time.
www.Notionmusic.com
USB Classical BoomBox and EthnicBoomBox
for Apple GarageBand
ClassicalBoomBox is a virtual instrument plug-in with 4GB of orchestral instru-
ments and Apple Loops, and EthnicBoomBox has over 5000 Apple Loops and 100
instruments of sounds from Africa, Asia, the Balkans, Indonesia, Spain, and the
Mediterranean. You can also use the loops from these $99 collections in Logic Pro
7, Final Cut, and other Apple Loops-compatible software.
www.Ilio.com 818/707-7222
Chris Hein Horns Vol. 1
East West has announced this $400, 3.5GB sampled brass
instrument library. CHH has alto and tenor saxes, trumpet, trom-
bone, and trumpet section. Up to 44 playing techniques are
included, all of which can be accessed by keyswitches in a single
program. Lots of MIDI control is available, including Growl, Air,
Keys, Spit, Dirty Notes, Breath Release-Trigger, and more.
www.Soundsonline.com 310/271-6969
VI
f e a t u r e
by Nick Batzdorf
One exception: the latest trend is for nota-
tion software such as Notion, Geniesoft
Overture 4, Sibelius, and Make Musics Finale
to interface directly with orchestral sample
libraries. Notion has its own library, Geniesoft
works or is en route to working with several
orchestral libraries, and the other two even
have integrated versions of Garritan Personal
Orchestra.
These notation programs dont even
require a MIDI keyboard for inputyou can
mouse in the notes, and theyll trigger the
samples with the expression you enterbut
theyre all capable of recording MIDI in real
time or accepting note entry from a MIDI
keyboard. Most people will want to use a
MIDI keyboard.
Now, sometimes two or all three of the
above-mentioned functionsaudio, MIDI,
keyboardare combined in one piece of
equipment. A good example is the Alesis
Photon X-25 controller on the cover of our
last issue, which is a small keyboard with inte-
grated audio and MIDI interfaces (or the
other way around, depending on your per-
spective). M-Audio and Edirol/Roland offer
products in the same category. Likewise, a lot
of the audio interfaces being sold today
include a MIDI I/O or two. (I/O = input/out-
put.)
MIDI controller keyboards deserve their
own article and well leave them for a couple
of issues down the line. Also, no hardware
discussion would be complete if we didnt
mention that unless youre content listening
to computer speakers, youll eventually want
some studio monitors (and an external amp
or integrated amps to power them) that are
designed for our kinds of activities.
Note that studio monitor isnt a very spe-
cific term. Some home stereo speakers are
also sold as studio monitors, while others
arent really built to withstand the abuse that
synthesizers and samplers can dish out. The
main difference for serious music production
is that studio monitors areor should be
designed to reveal whats going on, warts and
all, while some home stereo speakers may
part 2: audio and
MIDI hardware
A beginners guide to putting together
a DAW (digital audio workstation) for
softsynths and samplers
First DAW
This month were going to discuss the hard-
ware necessary to turn the computers we
looked at in Part 1 into V.I. Machines.
That radical transformation sounds pretty
major, but as well explain, it basically means
getting MIDI in and audio out.
Those functions require up to three pieces of
equipment: an audio interface, which is highly
recommended for high quality monitoring but
not 100% mandatory; a MIDI interface if your
controller keyboard doesnt plug directly into
the computers USB port; and a keyboard (or
other instrument controller, such as a wind or
guitar controller) to trigger the virtual instru-
ments running on the computer.
intentionally enhance whats there by impos-
ing a pleasant-sounding bass and treble
boost.
Built-in sound
The first option is not to add any audio hard-
ware and just use your computers built-in
audio. While even the most inexpensive audio
interfaces will sound considerably better, you
might be able to get started without one.
Some Windows software such as
Cakewalks Sonar Home Studio and the
Native Instruments synths/samplers will work
with the Windows DirectSound drivers, which
in turn work with most motherboards around
today. Virtually all Mac software uses the OS
X CoreAudio driver, which can access the
built-in audio hardware, so any Mac capable
of running OS X can work stand-alone.
But again, built-in audio is not really an
ideal solution. The exception is that Mac G5s
have S/PDIF format digital outputs that pass
the audio over optical cable to another device
with a digital-to-analog converter, bypassing
the computers built-in audio. That device is
most likely to be a consumer stereo receiver
that accepts optical S/PDIF.
Otherwise the G5 optical digital output is
actually a higher-end solution than many
sound cards, since you need some kind of
digital-to-analog converter to hear the sound.
D/A converters with optical S/PDIF inputs are
built into some digital mixers, and also into
devices like the Presonus Central Station
monitor controller, which includes a lot
of other useful functions. But one
digresses; lets talk about MIDI.
MIDI
For the uninitiated, MIDI (Multiple
Instrument Digital Interface) is the universal
control language for electronic instruments
thats been the standard for the past 22 years.
Unless youre using the direct notation soft-
ware-to-sampler method described earlier to
input music, youll need a controller keyboard
or other instrument that sends MIDI. That
instrument might be a synthesizer with a
MIDI output, or it could be a keyboard that
makes no sound on its own but sends MIDI
to trigger your V.I.s
If your controller has a USB connection to
plug straight into a computer, that means it
has a built-in MIDI interface; the label doesnt
say MIDI, but thats whats going down the
USB cable. Most instruments have MIDI out-
puts, though, in which case youll either need
a stand-alone MIDI interface or one built into
your audio interface.
Chances are
good that if
youre using a lot
of hardware
instruments, you
already have a
MIDI interface
with several ins
and outs. But for
V.I.s you probably
only need a very
simple model,
possibly one with
just a single in
and out. MIDI
interfaces start at
about $40 and go
up to $ several
hundred for an 8
x 8 model with
sync features for
working with tape
machines and
stand-alone
recorders.
MOTU, Edirol, M-Audio, EgoSys, and
Yamaha make most of the stand-alone MIDI
interfaces available today.
Driving the hardware
Before deciding on the audio interface fea-
tures you need, its necessary to figure out
how the unit will connect to your computer
and whether it has software drivers for the
programs youre going to use. With one seri-
ously notable exceptionMacintosh comput-
ers and their PCI card slots (oi veh!)thats
not complicated.
First the software. Under Mac OS X, every
Mac-compatible audio interface works with
the OS integrated CoreAudio drivers. End of
The RME HDSP9652 PCI card includes three 8-
channel ADAT optical digital I/Os and two MIDI
I/Os, onboard digital mixing, and more. The expan-
sion board adds two more ADAT optical I/Os.
The M-Audio Project Mix and TASCAM FW1884 combine 18-input FireWire interfaces with motorized control
surfaces that work with most if not all DAW software. The faders can also be assigned to control soft synth
and sampler parameters.
VI
f e a t u r e
slot connector, a.k.a. the PCMCIA slot, which
is similar internally to PCI.
Historically, internal computer slots have
changed every few years, forcing another
round of obsolescence. Thats certainly not a
recommendation to avoid PCI audio inter-
faces, just a way of bringing up that there are
now 66mHz PCI slots; 33mHz slots; 3.3-volt
and 5-volt slots; 4x and 8x PCI slots used by
graphics cards, called AGP slots; PCI-e slots;
and now faster PCI-Express slots.
Most motherboards on Windows machines
have standard PCI slots that work with pretty
much any PCI card. Its important to check
before you purchase anything, but compati-
bility isnt a huge issue for Windows
machines.
Macs are another story. First, no PCI cards
work in the latest generation of dual-core
G5s; as of this writing, you must use an exter-
nal FireWire or USB interface with these
machines. PCI-Express is a nice format, but its
not backwards-compatible with PCI.
It gets murkier. Some PCI cards that
worked in Mac G4s dont work in G5s, which
have both regular PCI slots and PCI-e slots.
Would you move over, please? Id like to
hide under the bed with you and those poor
audio hardware developers who have to keep
up with these changes.
But seriously. Lots of musicians all over the
world work on Macs with PCI audio interfaces
every day; you just have to make sure that
what youre using is compatible with your
Mac. And once again, Windows users dont
really have to be as concerned.
FireWire and USB
All FireWire and USB devices in the known
universe are external, meaning that they live
outside the computers case.
FireWire comes in two varieties: the more
common FireWire 400, the original protocol;
and FireWire 800, which has twice the band-
width. Weve all been conditioned to think
higher numbers are better, but in reality the
extra bandwidth doesnt make an audio inter-
face twice as fast. However, it could mean
you can hook up two of the same interfaces
(for more I/O channels) without clogging the
bus, or maybe you could hook up a hard
drive to the same bus without bogging it
down.
So given a choice between a FireWire 800
and 400 model, sure youd take the 800. But
400 is absolutely fine; the only FireWire 800
interface on the market were aware of is the
RME Fireface 800, and it reportedly works fine
in either a FireWire 400 or 800 port. Also note
that Macs have one FireWire bus thats shared
by the 800 and 400 ports, so using a 400
device in any port makes everyone run at 400
speed.
USB 1.0 is a much slower protocol than
FireWire, but its fine for a couple of ins and
outswhich may be all you need if youre
just monitoring V.I.sor occasionally, such as
in the Digidesign MBox 2, four I/Os. One
thing to be aware of is the difference
between actual discrete channels into the
computer and inputs into some form of mixer
on the audio interface. If you were to see a
USB 1.0 interface with 16 inputsand we
dont know of onethats what would be
going on.
As an aside, you can find mixers with
FireWire connections into the computer
onboard. Mackies Onyx series and Alesis
Multimix series are in this category, and they
would be a good solution if you have a lot of
hardware instruments or record live in addi-
tion to using V.I.s. Mackies digital mixers, as
well as TASCAMs and Yamahas, can also con-
nect directly by FireWire.
USB 2.0 is roughly the same speed as
FireWire 400. There are only a few USB 2.0
interfaces on the market. Edirol has one, and
MOTU has a USB 2.0 version of their 828mkII
interface, and the USB 2.0 Traveler was just
announced as we went to press. On paper,
MOTUs interfaces appear to be comparable
story; any audio interface that works on Mac
will work out of the box.
On Windows, virtually all software supports
ASIO drivers (ASIO is a format developed by
Steinberg that became a standard). Almost all
interfaces support ASIO, so there isnt too
much to worry about.
The exception is TASCAMs GigaStudio
sampler, which bypasses Windows and there-
fore uses its own GSIF format drivers. Youll
need a GSIF-compatible sound card to use
GigaStudio stand-alone, i.e. if youre dedicat-
ing a computer to running GigaStudio on its
own, with your DAW on another machine.
If youre streaming Giga into a DAW run-
ning on the same computers onscreen mixer
using the ReWire protocol, however, then the
host DAWs ASIO drivers take over and you
dont need a GSIF-compatible sound card.
PCI cards
All audio interfaces connect to the comput-
er by a PCI card, or by a FireWire (400 or
800) or USB (1 or 2) cable. There are also a
few interfaces that connect to the laptop PC
VI
f e a t u r e
(CONTINUED ON PAGE 58)
MOTUs 828mkII is now available in two versions: FireWire and USB 2.0. These boxes feature
onboard mixing and a wealth of I/O.
Metric Halos Mobile I/O FireWire interfaces and MOTUs FireWire or USB 2.0 Traveler can be powered by the
bus on a laptop computerno AC is required on location.
VI
m o d e r a t e l y d e e p c l i n i c
However, RAM-based sampling limits you to
the amount of memory your computer can
access, and modern sample libraries are much
too large and complicated to fit in such a small
space. So Kontakt also has a disk-streaming
mode called DFD (Direct From Disk). When
DFD is active, only the first portion (the attack)
of each sample is loaded into RAM.
In this way a large instrument that takes up
1GB on disk might only take up 100MB of
RAM. Load times are much faster, since the
computer is only loading a small portion of
the full sample set into RAM. The biggest
benefit, though, is that the computer has
more RAM available and can therefore load
many more instruments simultaneously.
On that subject, the best thing you can do
if youre running any samplerfrom RAM or
streaming from diskis add as much RAM to
your computers as possible, up to the limit of
what theyre capable of accessing. Under
most circumstances, Windows XP machines
can use up to 2GB. Mac G4s can hold 1.5GB
or 2GB depending on the model; theres a
Random Tip elsewhere in this issue about the
right amount to install in a G5. Its impossible
to run most modern sample libraries with less
than about 1GB of RAM nowadays.
DFD for all
In the DFD tab of the Options dialog box,
youll find a few options to tweak the DFDs
setup. Bear in mind that while DFD is ready
to be used at any time, in Kontakt 2 its only
actually in use when an instrument that has
DFD mode chosen in the Source module is
loaded. You can mix and match instruments
(and groups within instruments) that do and
by David Das
Disk-streaming Settings in Native
Instruments Kontaktseries
Samplers and Players
Adjusting those mysterious advanced DFD settings
to get the best performance out of your machines
Chances are good that if youre reading VI, you
own at least one Native Instruments Kontakt-
series instrument. Even if you dont own Kontakt
itself, so many popular sample libraries come in
integrated Kontakt-family players that its hard to
avoid in todays V.I. world.
If youre only using Kontakt (either the original
version, player versions of the original
Kontakt/Kompakt family, or Kontakt 2) for smaller
instruments that use very short samples, you can
run in Sampler mode. That means everything
gets loaded into RAM, and Sampler mode allows
you to use certain looping options and do things
like playing samples backwards.
dont use DFD freely. In the Kontakt 1 series,
DFD is on or off globally.
Figs. 1 and 2 show the basic and Expert
DFD setup modes in Kontakt 2; Figs. 3 and 4
show the regular and Expert DFD screens in
the East West Quantum Leap RA player, which
is based on Kontakt 1. The settings are the
same in the entire Kontakt 1 family, including
Kompakt 1.
In the Kontakt 2 basic screen there is only
one control: the amount of memory you
want to dedicate to DFD. You can request
fewer voices, which demands less memory
and more processing power (due to the
smaller buffer), or more voices and memory
as youll see if you try dragging the slider left
and right.
Despite the poor performance label in
Kontakt 2 (which really means heavier CPU
load) there are no right or wrong settings for
DFD. Different combinations of computers,
hard drives, and sample libraries may work
better with different DFD values.
Unless youre using a library with instruc-
tions that suggest otherwise, it should come
as no surprise that the place to start is with
DFD at its default settings. In Kontakt 2 thats
with the slider a little under a third of the way
to the right; in the Kontakt 1 family thats the
Normal (medium polyphony/medium memo-
ry) setting.
Dont be afraid of setting Kontakt for too
many voices; the worst that can happen is
that you will begin hearing audio artifacts
(pops and clicks), which is a sign that your
computer system is trying to work harder
than it can and you need to back off.
Experts only need apply
At the bottom of the DFD page in Kontakt
2 is an ominous-looking button labeled Expert
Mode with the caption, Use this ONLY if you
know exactly what you are doing! (This
warning isnt in the Kontakt 1 family.)
In Kontakt 2, the expert settings reveal a
couple of new settingschannel buffer size,
and the number of reserved channel buffers
and from that information it calculates the
amount of DFD memory required. Kontakt 1
has three adjustments: Preload Buffer Size,
Voice Buffer Size, and Reserved Voices.
The first thing Kontakt does is load the first
chunk of each sample into RAM (the preload
buffer). Then it fetches the remainder of each
sample from the hard drive. Actually, Kontakt
does this in chunks. For example, the preload
buffer may load the first 100 milliseconds of
each sample into RAM; then while its playing
that, it goes to fetch milliseconds 101200;
while its playing that, it goes to fetch mil-
liseconds 201300; and so on.
But many factors determine the streaming
performance. The sample rate and format dic-
tate exactly how large the samples are, and
so does the number of output destinations.
While the majority of people work with mono
or stereo samples, Kontakt 2 is also capable of
handling larger multichannel surround for-
mats.
Figs. 1 & 2: The basic DFD mode and frightening Expert DFD mode screens in Kontakt 2.
Figs. 3 & 4: The basic and Expert mode screens in this Kompakt Player (from Quantum Leap RA) are the same in all Kontakt 1-series instruments.
VI
The Channel Buffer Size setting allows you
to adjust how many channels you want to
reserve and how much memory per channel
should be reserved. For example, if you
reserve 1024 channels, you could theoretical-
ly play back 512 stereo voices, or 256 4-chan-
nel voices, or 64 16-channel voices, and so
on. This parameter is known as the Voice
Buffer Size parameter in Kontakt 1.
(Kontakt 1 and family can work in surround
if you load the front and rear programs in dif-
ferent slots, trigger them together by assign-
ing them to the same MIDI channel, but
route them to different Kontakt outputs.)
Note that the Channel/Voice Buffer setting
is a maximum. As youll see in Figs. 5 and 6,
the actual amount of memory the computer
grabs goes up as you play and the buffer fills
up to its maximum. If youre happily playing
lots of programs youve loaded into the avail-
able RAM but then suddenly start hearing
clicks and pops, this is one possible culprit (in
which case you just need to unload some-
thing to free up enough memory, and usually
restart Kontakt).
Simple strategy
Now that you understand the concepts,
adjusting the DFD settings is straightforward.
Lets take a look at an extreme example.
In terms of DFD performance, probably the
most demanding Kontakt-format library is the
East West Quantum Leap Symphony
Orchestra, which is recorded with hall reverb
that rings when you release the notes and
therefore uses a lot of polyphony. The
Platinum version of this library is recorded
from three stereo mic positions that you mix
and match, so you could conceivably have six
24-bit voices for every note.
For that reason, the instructions recom-
mend a setting of a 192KB preload buffer,
384KB voice buffer, and the maximum 256
reserved stereo voices. However, many users
have reduced those settingsespecially the
first twoand have been able to free more
memory for loading programs.
Its that simple. You just weigh the number
of voices youd like to play, the amount of
memory that requires, and the performance
of your system.
And now that youve made it through this
article, you know exactly what youre doing
and are officially qualified to click on the
Expert Mode button. VI
David Das is a composer and producer based
in Los Angeles. He can be visited on the web at
daviddas.com. This article began as an excerpt
from Davids excellent new book Kontakt 2
Power!, published by Thomson Course
Technology (www.courseptr.com); here its been
adapted and also expanded to include the other
Kontakt-series instruments.
Small channel buffers require more seek
operations from the hard drive. Big channel
buffers stress the hard drive less. But hard
drives always have certain block sizes that
they are best at delivering, and block sizes
that are smaller or larger than a certain hard
drives optimum block size may mean degrad-
ed performance. Theres no way to know how
a hard drive will perform, short of running
some very specialized tests on it.
Figs. 5 & 6: An instance of Kontakt 1 is loaded
inside Steinbergs V-Stack VST host, and the two
Windows Task Manager screen dumps show the
memory use before and after playing for a minute.
The increase of over 35MB illustrates how the
Kontakt voice buffer is a maximum, not a fixed set-
ting. If you have a machine loaded close to the
limit of its RAM and suddenly start hearing clicks
and pops when a sequence is playing, this is proba-
bly the reason.
VI
Review by
Bruce Richardson
VI
r e v i e w
Notre Dame de Budapest Pipe
Organ sample library, $459;
compact version, $129.
Publisher: Shirokuma Ltd., 1192
Budapest, Baross u. 4/1. Hungary.
www.ndb.hu.
US Distribution: East West
Sound Warehouse
(www.Soundsonline.com), MTLC
(www.mtlc.net)
European Distribution:
Time+Space (www.timespace.com),
Best Service (http://www.bestser-
vice.de)
Japanese Distribution: Crypton
(www.crypton.co.jp)
Format: GigaStudio 3 (preferably)
or 2.
License: One machine, one user.
Notre Dame de
Budapest Pipe
Organ Samples
W
hat if I said you could have a really
huge organ for only a few hundred
dollars? Would you go for it?
Minds out of the gutter, please
In this case Im talking about the Notre
Dame de Budapest Pipe Organ Samples for
GigaStudio 2.5 and 3.0, and it is a monster
organ (actually a couple of organs) indeed.
A whopping 23 gigabytes of sample data is
mapped into one giant orgasm for organ
enthusiasts and orchestrators alike.
One complaint heard about the pipe organ
sample libraries up until now is that they lack
the ability to mimic common organ tech-
niques, like swellbox, tremolo, crescendo, etc.
Not so with NDB, which includes all the
important combinations of stops, along with
a set of tools sufficient to cover all the cover
organ literature from pre-baroque to post-
modern. Also included are impulse files for
TASCAMS GigaPulse convolution engine in
GigaStudio 3, along with recorded release
tails that provide the two cathedrals own
acoustics as flavor for your work.
Giga
I was able to load and install the samples
on my GigaStudio machine easily. After
refreshing the QuickSound Database, I found
loads of options for creating organ sounds.
These include the basic GIG files, which are
collections of individual programs that share
the same samples, and the new to Giga 3
GSI file, which loads more elaborate set-ups
of controllers, effects, etc. To get the pro-
grammed organ-specific techniques, there are
GSI files that are configurable to your specific
controller set-up.
In fact, most of
how you will ulti-
mately use this col-
lection hinges upon
what your musical
goal requires.
Composers looking
for some pipe organ
flavor may choose to
use the straight GIG
files. Organists
might set up an
elaborate series of
controllers and ped-
als to emulate the
performance condi-
tions of an actual
organ console.
One nice aspect is
a comprehensive set
of extra-musical
noises the organs
make. Using all the
available tools, one
could emulate an
organ performance
down to the most
minute details. Using the impulses provided,
one could even emulate the organists
approach to the bench.
However one chooses to approach the col-
lection, it has more than sufficient amenities
to get the job done, in high style.
Wow
The folks at NDB have created nothing
short of a masterpiece. In recording these two
organs, a very consistent and lush soundstage
was created by NDB Architect and Lead
Engineer Csaba Huszty. These tones are con-
sistently full, vibrant, and reverberant, but not
the least bit washed out.
The best I can describe the feeling is that
sitting down and playing this instrument for
One of the worlds most devastatingly
impressive sample libraries
the first time brought a huge grin to my face.
It sounded exactly the way I hoped it would,
that singular point on the scale of transparen-
cy versus saturation that a really amazing pipe
organ possesses. There is plenty of cathedral
resonance to give a massiveness to the sound,
but at the same time the precise chiff of the
attacks and release triggers gives NDB a
grounded, solid image that never floats off
into a wash.
Mr. Huszty made the interesting choice of a
pair of Neumann U87s for his microphone
array, which may account for a great deal of
the balance of clarity and lushness. Perhaps
inspired would be a better description,
since this somewhat unorthodox choice of
microphones delivered an end result that
other sample sets have not matched. If I had
to describe the sound in one word, it would
be stunning.
In fact, people almost universally have one
thing to say when I demonstrate this sample
set for them. They say, Wow. This collection
has the huge, full, rich wow factor of the origi-
nal Peter Ewers collection for GigaSamplers
first incarnation, but the NDB organs manage
to retain a pinpoint soundstage accuracy. You
hear individual pipes coming from distinctive
points in the soundstage, combining to give
the fullness you want without washing out...a
very tricky balance, well achieved by Mr.
Huszty and his team.
List price for the full collection is $459
according to the manufacturers website. A
compact version is available for $129, which
ships on one DVD versus the four of the full
version, and which does not implement the
impulse designs, or the advanced swellbox,
tremolo, etc., techniques. Still, for someone
who is looking for some excellent organ sam-
ples to enhance his palette, this could be just
the ticket.
Greatness thrust upon us
The Notre Dame de Budapest Pipe Organ
Samples is without a doubt the finest set of
pipe organ samples I have heard or played to
date. You can lose yourself for hours in the
sheer grandeur of the playing experience.
And as if thats not enough, you get an
information-crammed booklet loaded with
everything from general facts about pipe
organs, to detailed photos of the instruments,
to charts of what stops and combinations are
appropriate for what period of literature.
Another nice plus is that a portion of the pro-
ceeds of each sale goes towards the upkeep
and maintenance of these wonderful source
instruments.
This is not only a great set of pipe organ
samples, but a great example of what a truly
excellent sample library can be. VI
You hear individual pipes coming from distinctive
points in the soundstage, combining to give the
fullness you want without washing out...a very
tricky balance, well achieved. You can lose
yourself for hours in the sheer grandeur of
the playing experience.
by Jim Aikin
VI
v e r y d e e p c l i n i c
T
hanks to the huge increases in comput-
er CPU speed and the cleverness of a lot
of hard working software engineers, it
really is possible to start with a moment of
musical inspiration and take it all the way to a
finished, mixed composition without ever
leaving the comfort of your desktop. The line
between self-contained virtual workstations
and DAW (digital audio workstation)
sequencers has blurred; every program offers
its own set of featuressynthesizers, effects,
automation, and more.
Cakewalks Project5 is emerging as one of
the leaders in this field. It has a unique com-
bination of powerful modules and a friendly
workspace for sequencing. In this tutorial
well take a look at some of Project5s hidden
features, and show you some tricks for get-
ting more music from it.
Needless to say, the suggestions below are
only the tip of a large iceberg. You can learn
more about the programs capabilities at
www.cakewalk.com/Products/Project5/default.
asp.
Hidden dimensions
Project5s Dimension synth is a very capa-
ble instrument, and worth getting to know. A
full Dimension tutorial would take many
pages, but here are a few ideas to get you
started.
The Matrix: Regarded. Dimensions MIDI
Matrix (see Figure 2) is where you add real-
time modulation to Dimension patches. Open
the Matrix by clicking the little MIDI jack-
shaped button in the top row of the
Power in
Project5
Discover some less-obvious
features of Cakewalks all-in-one
virtual workstation
Figure 1: The big picture in
Project5 includes MIDI and audio
tracks (upper area), plug-in synths
like the drum sampler Velocity
(center), effects such as the reverb
(lower left), and an audio editor
(lower right).
VI
Dimension interface. The Help file explains
how to use the Matrix, but it doesnt explain
the peculiarities of some of the modulation
routings.
For instance, the Sample Offset destina-
tions can be used to offset the beginning of
sample playback so that when you play a key,
you never hear the beginning of the sample,
only the later part. This trick has been used in
sample-based drum machines for many years
so that at low velocities, part of the attack
transient of the drum will be skipped, result-
ing in a less punchy sound. It can also be
used with string orchestra samples, in which
low velocities will preserve a gradual attack
while high velocities start sample playback at
a later point for a more aggressive sound.
The string orchestra effect (higher velocities
causing a later sample start time) are easy in
Dimensionbut you might think at first that
Dimension cant go the other direction for
velocity-controller percussion, because nega-
tive depth settings dont do anything with
this type of modulation. But Jesse Jost of
Cakewalk had this solution up his sleeve:
First, choose velocity as the source for a
Matrix routing and Sample Offset as the des-
tination. Set the amount to about -5000.
Next, choose CC 127 as a source and Sample
Offset as a destination, and set the depth to
5000. In Dimension, CC 127 always has a
value of 127, so in this routing its setting the
sample start point to a later value, allowing
velocity to modulate it backward.
Try this effect with the Electric Kicks &
Snares patch. Youll find that it sounds good
with the snares, but not so good with the
kicks. Kicks tend to click because the sample
start point wont always be at a zero-crossing,
depending on the velocity.
The solution is to switch on the amplitude
envelope and right-click in the envelope area
to create an attack segment.
Move the breakpoint close to
the left edge of the window and
then zoom in (using the mouse
wheel) to move it even closer.
Watch the Time value in the
data display above the enve-
lope. An attack time as short as
0.5ms will get rid of the click
without making the drum sound
mushy.
Envelope control. Each seg-
ment of each Dimension enve-
lope can respond to MIDI veloci-
ty, MIDI key number, or both.
This feature opens up a lot of
possibilities for designing instru-
ments that are more playable
from the keyboard. For exam-
ple, you can give high-velocity
notes a longer envelope decay
time, or give the low notes on
the keyboard a longer release
time. Both of these effects are
characteristic of the piano and
other acoustic instruments.
The manuals description of
how to do this is a bit terse. Maybe youre
smarter than I am, but when I first read that
pressing V on the keyboard would show red
velocity-amount bars beneath the envelope,
and then when I pressed V and didnt see any
red bars, I assumed the feature wasnt imple-
mented. In truth the feature works fine, but in
most of the factory patches the velocity
amount is zero. When its zero, the red bars
are invisible.
The procedure for adding velocity response
is simple: click anywhere in the envelope win-
dow, press V, and then click above or below
an envelope segment and drag up or down.
A dim red bar appears (see Figure 3). The
same technique works when you press K on
the keyboard, except that the bars are blue-
grey and control keyboard tracking amount.
Other invisible commands worth remem-
bering: select an envelope breakpoint and
type S to make it the sustain level. Choose
another point before the S point and type L
to create a looping envelope.
Multisample editing. Dimension plays multi-
samples (sets of WAV files in which various
files are mapped to individual keys or key
zones across the keyboard), but unlike a sam-
pler, it has no user interface with which you
can change the mapping of samples or load
your own samples. Or does it? If you think
you can only use Dimension to play the mul-
tisamples provided with Project5, think again.
Heres how to edit multisample layouts and
create your own:
Launch your favorite text editor. Notepad
will work fine; I use a very nice freeware pro-
gram called NoteTab Light. Its easier if you
dont use a full-featured word processor for
this type of work. If you must use a word
processor, be sure to save your work as text-
only files with no formatting.
Navigate to the directory where your
Dimension waveform data is stored. Open the
Multisamples directory and choose a subdi-
rectory. In some of the subdirectories youll
see files with the extension .sfz. Open one of
these files in the text editor. (Before you try
editing the factory .sfz files, I suggest making
a backup copy in some other folder on your
hard drive.) You might want to start with the
file Gibson Les Paul Full.sfz, in the 05 - Guitars
Figure 2: The MIDI Matrix gives the
Dimenion synthesizer 16 real-time
modulation routings.
Figure 3: The dim red bar in Dimensions envelope window shows that velocity modulation is active for that
envelope segment.
VI
directory. This is the multisample used in the
Les Paul (bright) patch.
The first few lines (see Figure 4) begin with
double slash-marks (//). These lines are com-
ments. Theyre included purely for human
reading, and have no effect on the actual
functioning of the file. The rest of the lines
are computer code.
The lines beginning with the <region> des-
ignation are key zones. If you compare the
first line with the directory structure in
Windows Explorer, youll see that this line
directs Dimension to use a sample called E1
LOWH.wav in the Gibson Les Paul/LOWH
subdirectory. The rest of the line (key=e2
lokey=c2 tune=4) gives Dimension some
essential information about how to play the
WAV file. Its root key (the key at which it will
play its original pitch) is e2 (MIDI key 40).
The bottom key is c2 (MIDI key 36, the low-
est key on a standard five-octave MIDI key-
board), and the fine-tuning value is 4.
Lets suppose that for musical reasons you
need to play a low B with this patch. All you
need to do is change the middle value so that
it reads lokey=b1. (Tip: its usually important
in computer code not to put any extra spaces
around the equals sign, and to follow the
upper/lowercase scheme of the letters.) Save
the .sfz fileagain, being careful to save as a
text-only file if youre using a word processor.
Then reload the Les Paul (bright) patch in
Dimension. Youll find that the lowest sample
now goes down an extra key, to B1 (MIDI key
35).
To examine Dimensions velocity cross-
switching, open the file Grand Piano 3V
m3rd.sfz. Here youll find three separate
groups of regions defined. The first group has
the parameter hivel=54, the second has
lovel=55 hivel=107, and the third has
lovel=108. If you know a bit about MIDI key
velocity (it ranges from 0 to 127) youll have
no trouble interpreting this data. This file also
demonstrates that MIDI note numbers can be
used to indicate key zones in place of
letter/octave designations.
Its the Spectre, Phil...
Spectral Transformer is certainly one of the
most complex effects processors in any virtual
workstation program. The easy way to work
with it is just to select it in the Add FX menu
and start messing with the controls. When
you get done being amazed, however, youll
probably notice that Spectral Transformers
output is delayed by a noticeable amount. In
any type of rhythmic music, delay (also
known as latency) can be a deadly problem.
You may also notice that Spectral Transformer
eats up CPU bandwidth as if it were gua-
camole dip.
You can reduce (but not eliminate) Spectral
Transformers latency and ease the CPU hit
somewhat by changing the analysis settings.
This can only be done when the Power but-
ton (in the lower right corner of the plug-in
window) is switched off. The best way to
understand the choices in the analysis settings
box (see Figure 5) is to launch Project5s Help
file and read the Spectral Transformer section.
Read it about three times. Eventually it may
start to make sense.
Changing the analysis settings doesnt just
affect the latency and the CPU load; it also
changes the sound of the effect. If you find a
sound you want to keep, but cant seem to
get it to play along with your song because
its audibly late or doesnt leave any CPU
power for other instruments, the solution is
the Bounce to Track command. Right-click on
the track being processed by Spectral
Transformer and choose this command, and a
new audio track will be created with the spec-
trally transformed sound. After muting the
original track, save your Spectral Transformer
preset and unload the plug-in.
That takes care of the CPU load problem.
But the bounced track will still be lateper-
haps as much as 400ms, which in todays
high-pressure music is about as bad as being
in the next county.
The solution is simple. In the track view,
switch off the snap-to grid. Zoom in on the
left end of the bounced audio clip, drag the
Figure 4: Editing one of Dimensions .sfz multisample setup files in a text editor.
Figure 5: In Spectral Transformers analysis settings box (foreground), you can choose your own compromise
between tight frequency response, tight rhythmic response, and latency. Separate effects are loaded into the
four slots in the upper area.
Review by
Nick Batzdorf
VI
r e v i e w
E
ast West Quantum Leap Symphonic
Choirs (EWQLSC) was produced by the
same people responsible for the EWQL-
SO(rchestra): Nick Phoenix and Doug Rogers.
It was also recorded extremely well by the
same engineer: Keith Johnson.
This team applied the same approach they
used for EWQLSO, recording the choir in a
nice hall simultaneously from three mic posi-
tions and triggering the natural reverb trail
when release the note. And as with EWQLSO,
the gigantic sound of this library just knocks
you over when you first sit down and play a
few chords.
What really makes the library work well is
the WordBuilder utility program. The develop-
ers figured out all the sounds we use in lan-
guage, and recorded them. You enter the
words you want in WordBuilder, syllable by
syllable, and then each successive note you
play triggers the next syllable. Its not quite
that simple, but thats the principle.
EWQLSC includes soprano, alto, tenor, bass
(SATB), and boys choirs, as well as soprano,
alto, and boy soloists. It comes in its own
Native Instruments Kompakt player, which
has in all the standard plug-in formats and
works on Mac and Windows.
Performance
While well worth it, this library is quite
demanding of computer resources. First, the
release trails ring when you release notes, and
that uses some polyphony even though choir
is normally only a 4-part instrument (SATB).
On top of that, many of the programs have
modwheel crossfades between layers, and
that uses a lot of voices.
Then you need as much memory as your
computer can access. Thats especially true of
the WordBuilder Multi programs, which are
spread across four MIDI channels in the
Kompakt instrument. With three velocity lay-
ers of every vowel and consonant known to
man, these programs have a huge number of
samples.
There are also lower-memory versions,
but for example a single Full mic position
East West Quantum Leap
Symphonic Choirs
A massive sampled choir library with a
utility that makes it sing words
East West Quantum Leap
Symphonic Choirs, $995.
Distributor: East West
(www.soundsonline.com)
Platform: Mac OS 10.2.6+ or
Windows XP; standalone (through
Core Audio, DirectSound, or ASIO)
or as VST, DXi, ASIO, Audio Units,
or RTAS plug-ins.
License: Challenge/response
installer limited to two unique
installs.
modwheel-crossfade Alto choir takes up about
420MB (at the default disk-streaming set-
tingsplease see the clinic on DFD settings in
this issue for more about that). On top of that,
WordBuilder takes up about almost 200MB,
but thats outside the DAWs memory space.
Thats on a dual G5 Mac with 5GB of RAM
installed. I was able to load one stereo mic
position of an entire choir (SATB) plus
WordBuilder in this machine quite comfort-
ably, but the choir (plus the DAW itself) took
up about 2.15GB out of Logics approximate-
ly 3GB of available memory, which while
impressive doesnt leave you all that much
room for other instruments. (See the Random
Tip on G5 memory elsewhere in this issue for
a discussion of how this works.)
Youll probably want to do some bouncing
to disk to free up memory, and if youre
working with a single Windows machine
youll have to bounce in order to get a full
choir. In fact, if youre really taking advantage
of the three phase-locked mic positions the
way theyre designedmixing and matching
them to get the right balance between defini-
tion/intimacy and size/ambience, or throwing
the Surround mic positions in the rear chan-
nelsthen youll definitely need more than
one machine for a full choir.
Programming
EWQLSC has two types of programs: the
above-mentioned WordBuilder Multis and
standard single-channel programs. For each
choir, there are standard programs with indi-
vidual consonants (e.g. mmmm), about eight
different vowel sounds (ooh, ah, eh, etc.),
and some very dramatic special effects pro-
grams. Each of these is available from three
mic positions, of course.
Different people work different ways, but
one practical way of dealing with EWQLSC is
not to use the big WordBuilder Multi pro-
grams to play melodies while youre compos-
ing, but instead just use standard aah (or
whatever) programs. Then when youre ready
to get serious with the words, you can
replace the choirs one at a time with the
WordBuilder Multi, and bounce the result to
disk to free up memory. (For the uninitiated,
that means recording it to disk as an audio
file, which takes virtually no memory or
horsepower and only two voices.)
A Full Chorus Church choir is only 32MB,
for example, while a regular alto program
would generally be under 50MB. So in the
worst case youre loading 200MB for SATB
and those programs still sound really nice.
Many of the EWQLSC programs use the
modwheel to crossfade between layers. These
are very easy to control, and the programs
brighter and more intense as you raise the
wheel. Just a little bit of this adds a lot of
expression.
Whats really impressive is the under-the-
hood programming in conjunction with
WordBuilder. This is a very complicated
library, but the programing makes it very easy
to deal with.
WordBuilder
You only have to listen to a computer
speak text to know that syllables arent pro-
nounced the same way every time theyre
encountered. Its the same with singing, and
WordBuilders function is not just to trigger
the syllables you enter, but also to let you
tweak each one.
You can either type words in English, pho-
netically, or by using WordBuilders own pho-
netic code called Votox. The latter two also
have a menu of syllables to select with the
mouse. While initially the least familiar of the
three formats, Votox is actually the most
direct path to the sound youre hearing in
your head.
Each syllable is represented by a bar on its
own track, with a timeline above going from
note-on to note-off. You can adjust the posi-
tion and length of each bar, and there are
shortcuts to enter values or learn them via
MIDI. Theres also a breakpoint level graph
under each syllable, which lets you go in and
fine-tune it (crossfading, etc.). If youre using
melismas (ha-a-a-a-a-a-a le lul yah), you can
make the a-a-a-s all legato with one com-
mand.
With most modern sample libraries it takes
time to learn all the different articulations;
with EWQLSC it takes time to get good at
WordBuilder. The program itself is very easy
to useits the musical judgment that takes
practice.
However, the latest version comes with a
drop-down menu with preset Latin phrases.
In all honesty, even with a lot of tweaking,
the language you enter wont always be
100% understandable. What these preset
phrases do is make the sampled choir more
than just oohs and aahs. Sometimes nobody
really knows whats being sung, but the
words just add a vaguely Latin vibe. East West
has a library of user phrases on their website
for the same general purpose.
In Mac OS X, WordBuilder receives MIDI
from your sequencer via an internal MIDI bus
it creates, processes the MIDI, and sends it
back out on another internal MIDI bus
(although you could just as easily run it on a
separate machine via regular MIDI). Its not
terribly complicated to set up, but I did find
myself having to reload WordBuilder and
check connections almost every time I loaded
a new a new Multi in the EWQLSC player.
With the caveat that it can be a little fussy,
WordBuilder is a terribly clever program that
makes a seemingly complicated function very
easy.
Opinions
The SATB choirs in this library are just gor-
geous, the boys choir is okay, and the solo
voices are especially useful in unison with
other instruments. Choir is such an incredible
sound that you could sit on the keyboard and
it would sound nice, but this library goes well
beyond that.
If you listen to work people have done with
EWQLSC, youll find that it can sound fantas-
tic, and it can also sound just plain wrong
not that everyone would always agree which
is which. Thats because the human voice is
so complicated. Its certainly not necessary to
bury EWQLSC, but in my opinion it generally
sounds best when accompanied by other
instruments.
So in the hands of a good musician,
EWQLSC can sound just like a real choir; used
poorly, it will show its underwear. Either way,
EWQLSC does everything right: good singers,
top-notch recording, and very good program-
ming.
What none of this says is how spectacular it
is to play. VI
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VI
f e a t u r e
W
elcome to sound design in music
production! This series is not meant
as a de facto rule book, guide to
good taste, or even something you wont
think is Chinese water torture. We all have dif-
ferent musical tastes and ideas.
And that is great.
But over the years Ive seen so many peo-
ple struggle at their computers to come up
with somethinganything in fact. Or if they
do, its a struggle to finish the idea without
ruining what was good about it to start with.
We all go through this production night-
mare. But when you have deadlines to meet,
you need solutions, not sympathy.
Thus well be offering you some custom
tools for Windows XP and Mac OS X via the
Virtual Instruments Magazine website. These
are not intended to be commercial solutions,
but that doesnt mean you wont have a hit
using them. It also opens the arena for YOU
to interact and contribute elements you have
made from them into the production pool.
We have two main goals: inspiration and
production. So lets get on with it!
Inspiration 101
Logically we should start with rhythm first,
but instead were covering pitch.
There are only so many techniques for
devising a tune or hook. If youre very lucky,
you can hear notes in your head. For the rest
of us, we usually pick some notes out of a
chord, and for the clever ones, then extend
this out in to a scale. You might also have a
chord sequence, and then pick the melody
from the notes of the chords (but its effec-
tively still scale-related).
Most of us will subconsciously (or blatantly)
piece together several tunes we have heard
from somewhere else. It could be a collage of
anything, from childhood nursery rhymes to a
classical music phrase we heard last month.
Given these techniques, you can see the
high risk for unoriginality and why self-com-
posed album tracks can seem all the same. So
how can we be inspired with something total-
ly unique from our own personal style?
The voice of God
Algorithmic music composers are one
answer. They tend to use random events (or
sometimes a picture) as the seed, and then
put them through some sort of scale correc-
tion, or a good taste filter. But most of the
time they sound like plinky-plonk alien cat
fartsscale corrected but still smelling of
cheese!
Words
Words are a great source of inspiration. But
normally they arent related to music. Well,
not till now. And the great thing about words
is we take things with a meaning and make
them say something else, musically.
You can take them from anywhereyour
own poetry, a novel, the Bible, newspapers,
TV, or radio. Strangely, cooking programs are
great for this! Foreign languages are equally
valid.
The words wont be copyrighted the way
we are going to use them. And the original
speaker is highly unlikely to have thought
they were speaking a hit pop riff anyway.
Algorithms
The way you translate spoken word into
pitch, is up to you. The point is what ever
technique you devise, you apply it in the
same way to every source word. Thats what
will give us the musicality, rather than ran-
domness.
You could make a table of words in a
spreadsheet (like Microsoft Excel) and assign
each a pitch. You could just take the vowels
from words and use those as the basis for
BUICK SESSIONS:
Sound design in music
production
by Peter Buick
Download a free algorithmic lyric and music program from
more
online
www.virtualinstrumentsmag.com
pitch. You could do a similar thing with consonants.
But this process might have all the excitement of doing a Times/Sun
crossword puzzle for you. Plus you have to devise an algorithm (formu-
la) that sounds good.
Enter the Virtual Instruments Magazine Mind Meld program, written
by yours truly. But as hes not here, Ill accept the award for him.
You can download it free from the VM web site. Or if you have a
slow Internet connection, or want the extra sound packs, if you donate
you can get a CD in the post.
This isnt a commercial product, and it is not offered as a sequencing
playback engine. It was designed simply to inspire you with note
names that you can input in to your real sequencer (or brain). But it
does (crudely) play back samples, so you can get a good feel for what
it is creating and mold it to your own production values. Sometimes
you may need to play it through twice to buffer it.
There are more detailed PDF instructions with the program, but lets
extract some salient points.
The algorithm Ive chosen in this version is simply to relate the char-
acter length of each word to a pitch in a scale. To make it more inter-
esting, the number of vowels in each word determine the octave of the
note. I wanted to route something to velocity as well, but I ran out of
time.
Whats interesting about this is that the natural flow of words will
often create useful musical phrases. Although fred, feed, and fioe
will produce the same pitch, because one has more vowels in it, in
Vowel Mode it will produce an octave shift.
You should find that when you hear it back, you will start to edit the
words with more characters or change the number of vowels, or the
base word, to create a riff you prefer. The Mash function will re-arrange
the sentence for you; the Mess function will choose from your entire
library of phrases to widen your vocabulary even further.
It is like learning a new language. But it doesnt require a musical
degree. And very few sentences will be musical rubbish. I hope that
youll find that thinking in letters will give you immense musical free-
dom. All of that musical training goes out of the window. You could do
an extra layer of translation between letters, music, and music theory,
but that would be defeating the purpose. The point is to create things
you wouldnt normally.
Certainly once you have a seed and youre in your main sequencer,
you will need to start the musical process. But that is what makes it
your song and not mine, or a random number generators. This version
is limited to equal length notes, but you can set each run to a different
number of pulses. Note durations and velocities can be a job for your
sequencer.
I hope youll find Mind Meld a lot of fun, and very inspiring. Any
positive feedback would be great! And you can share your own ele-
ment creations via the VM forum. A complete song could just be a few
paragraphs. One for each part. Yes, why not drums and percussion!
Its not common to find pleasing pitch sequences in nature. So I
hope youll give some thought on this technique of taking the spoken
word and making it, well, sing without language.
Production 202
Once we have a seed for an idea, this is the ideal cue for us to think
about production techniques.
The main page demonstrating the conceptual process.
In the word library you can type a new sentence in the scroll box and then
Mess or Mash it to the number of elements set.
Changing the output notes effectively controls the scale and melody. Preview the riff from the Audition page. The Vowel to Octave feature can be
enabled from here.
VI
f e a t u r e
Next well transfer this phrase to our main sequencer (Steinberg
Cubase in my case). Because we set an element length of 12, in 4/4 we
have a natural gap at the end of a 2-bar sequence in 1/8 notes. We can
of course slip the phrase anywhere within this frame, or add or repeat
notes to fill it out. Or use a different time signature or note length. Or
any combination, or a version of each on a separate track (as layers, or
for different parts).
To my taste the phrase currently works best as a bass line. But well
extrapolate a melody from it using some of our production 202 tech-
niques. First we make a copy and route that to a melody sound. Piano
should keep our options (and imagination) open.
Concentrating on the bass line, we change the two repeat notes into
a single longer note. Also the bass line probably shouldnt stop. It
doesnt need the breathing room of the melody, so we extend the
phrase to the bar end using some of the same notes. Its transposed
two octaves into the bass range of the piano.
Next, looking at the melody we decide to change one of the double
notes to a new note that hasnt been used yet. We tried octaves and
borrowing existing notes, but it didnt gel. We also add a 2-note
answer at the end of the phrase. Ultimately this will use a different
texture, but this will do for now.
There are many challenges in production, including not distracting
from the message. Reinforcing the message. Getting the very best from
the talent. Creating a unique sound (both musically and sonically) that
is fresh but not too radical. Generally making the pieces fit together in
an interesting way, that develops but that doesnt stroll off in a tan-
gent. And something that can stand beside the competition.
Comparative loudness, fashion trends, market expectations, emotional
planes, and so on.
The more options we know about, the more chances we have of
getting there. Enter the production 202 guide map.
Choices
One of the most important options is knowing when to say no. But
we need to take that down to (below) a beat level musically and to a
frequency range sonically. When coupled with the paradigm less is
more, what do we get?
Lets start to apply this to our Mind Meld word phrase.
Using a Mess translation, our phrase came out as:
then I two hill or then I two hill rock fries rock a Rye lonely scrubbelduff
lonelyck fries rock a Rye lonely scrubbelduff lonely
With the standard algorithm this produced the musical phrase:
D, A, C, D, D, E, D, A, C, F, Rest, F
VI
f e a t u r e
The Mind Meld Mess translation creates a phrase from all the words in our
word library.
We only selected 12 elements in our phrase, but well use the gap for some
needed breathing space in the melody riff.
Our Mind Meld riff in Cubase, in Key and Score Editor views.
Our edited bass riff with extension.
tions, applying half tempo, and then deleting a pattern of
the remaining notes. We transposed it into a key. Editing
the pitch of the phrase is always an option, of course.
We kept each stage as a copy, just in case that could
lead somewhere else later.
Wrapping up
If you listen to the test MP3
(MIXDOWN_MindMeld_Test.mp3) on the web site, you
can hear some of the options we came up with.
The original word phrase is also available as mindmeld
phrase SPOKEN.mp3 in different characterizations,
which is entertaining itself. A keen samplist could also cre-
ate a MindMeld sample set of each word and contribute
it to the pool. The waves can then of course be made into
a sampler program.
Note that this was all generated from our Mind Meld word phrase;
we never used a keyboard. The rhythms and pitches are all derived by
manipulating or editing layers of the original phrase.
Musically, its highly questionable. But the point is we started this
project with nothing in mind at all. We threw a few words together
and within half an hour, and we have something we can apply our pro-
duction techniques to. Original music will always be about experimen-
tation and having to apply your repertoire of production techniques
and your good taste filter to mold it into art.
This is just the tip of the iceberg. But at least you might now consid-
er that there is an iceberg out there somewhere! VI
So have we followed any rules here, or just tinkered? Every note
should be considered in production terms, but the double note was a
huge trigger point. Do we leave it as two of the same, or do we
change or remove one of them? Or do we assign it a different texture
(i.e. through velocity)?
For the bass line, we half removed it by merging the two notes and
extending the length. This effectively changes the rhythm, which for a
bass line seemed appropriate at the time.
For the lead line, we decided to make the second note stand out. We
cheated slightly and changed the pitch, rather than the texture or
rhythm. But it is another valid option.
Now we are going to use the production techniques more closely.
The Dissection technique
We take the rhythm of the word phrase and turn it in
to a hi-hat part by assigning all notes to a fixed note.
Make the line monophonic using polyphony restriction,
and transpose to your desired percussion sound.
Now we have a mimic rhythm, but its boring and
robotic. We can use our splitting concept to make certain
notes have a different velocity. Its your choice what pat-
tern you use; the pattern could stem from the words, or
it could be inspired by something else.
You may want to make certain notes unique by assign-
ing them to a different sound or texture (either layered or
exclusive). Or add a single beat effect to them. Dont
underestimate the power of those sentences!
We then copied our hi-hat part and made it into a
plod sub bass line by changing the notes to fixed dura-
A test arrangement in Cubase, starting to assign the variations in our phrase
production to textures
.when you have deadlines
to meet, you need solutions,
not sympathy.
Our edited lead line with passing note and ending answer.
.plinky plonk alien cat farts,
scale corrected, but still
smelling of cheese!
VI
f e a t u r e
I
ve spent a good part of the last 20 years
pondering the phenomenon we call rever-
berance, as well as developing my skills at
using it in music production for a variety of
applications. Much of my research has had to
do with loudspeaker performance and the
interaction of loudspeakers with rooms, which
has led me down some odd paths and to
some interesting insights.
Without boring you with all my adventures
and misadventures, Id like to make some
fairly basic observations about reverberance.
To begin with, the wash of sound we hear as
reverb is the leftover residue of a sound in a
space that comes well after the sound itself
has occurred. We often find that to be
enveloped in that reverberant wash is a pro-
foundly satisfying and pleasurable experience.
Before that reverb trail began, however,
some other things went on acoustically in the
space, as well as in our auditory perceptual
mechanism. The main acoustic things that
happened were a volley of early reflections of
the sound source. These arrive at the listener
from a variety of different directions, at differ-
ent times.
Early reflections have two characteristics:
they all are phase-locked with the original
sound (the direct sound), allowing them to
fuse with it; and they all occur after the
direct sound and before the onset of audible
reverberance (or within approximately 50 mil-
liseconds of the direct sounds arrival at the
listener). The number of reflections, the times
at which they arrive, and the angles from
which they arrive are determined by the size
and topology of the room, and the relative
positions of the sound source and listener in
the room.
What is profoundly important about this
volley of early reflections is that it provides a
rich array of data about the timbre of the
sound source (as it radiates in all directions),
its position relative to the listener, the bound-
aries of the enclosing space, and a fair
amount of information about what those
boundaries are made of. Our auditory system
actually uses this array of information to cre-
ate an integrated sensory construct of the
sound, identifying it and locating it in space.
At the same time, the system creates an effec-
tive sensory construct of the environment.
This all happens pre-consciously and quite
seamlessly. Mostly, we are unaware of the
integration occurring. We dont notice how
clearly and easily we hear even in sonically
chaotic small spaces. Also, we are not aware
that our use of early reflections to create this
sensory construct is asymmetrical. We make a
great deal of use of lateral (horizontal) reflec-
tions, but are somewhat confused/distracted
by vertical reflections (particularly from low
ceilings). Why this should be so is way
beyond the scope of this article.
One final, very curious thing about rever-
berance, at least in music production, is that
we dont seem to mind at all the existence of
multiple simultaneous reverberances. Not
only dont we mind, we actually seem to
enjoy the cognitive dissonance inherent in
having, all at the same time, a long decay on
the lead vocal, a different decay on the snare
track, some short room ambience on the
horns, and a slap echo on the hand-claps. I
have no idea why this is so, but it clearly is.
Better mixes
We can create much more powerful and
effective multitrack mixes if we keep these
An audio gurus gurus
reflection on reflection
Thinking
About
Reverb
By Dave Moulton
VI
f e a t u r e
things in mind. There are several features to
think about and address.
The direct sound is that dry mono track
(usually its mono, anyway). We usually pan it
somewhere between full left and full right,
depending on musical issues and a few
recording traditions. In most sample libraries
(and most live overdubs in project studios) its
recorded dry, with little or no room sound.
The first reflections are two lateral reflec-
tions, from left and right. Normally, we create
these with a spreader (a pair of very early
delays panned left and right, fed by an aux
bus). These define the basic dimensions of a
virtual room that the dry sound is in. They are
never shorter than about 12 milliseconds, and
seldom longer than 40ms. They are almost
never exactly the same, but usually have simi-
lar values. These first reflections are not used
on all tracks.
The more complex volley of early reflec-
tions that follow can be thought of as short
room ambience and can be generated by any
normal reverb device, although some are much
better than others. I usually dedicate a reverb
engine to this short ambience. The point of
this is to create a convincing room-ness with-
in which dry sounds can exist convincingly.
This ambience is rich in reflections from
across the stereo panorama that decay during
no more than a second, and usually less, such
as 750 ms. Often I will splice this short
ambience to a longer reverberant wash for
instruments that seem to need longer rever-
berance.
Finally, we have traditional reverberation.
Three concerns exist with such reverb, driven
by the nature of the music in production.
They are reverb time, (whatever that really
means), predelay (the amount of time it takes
for the reverb to build up in level after the
direct sound), and the spectral quality of the
decaying reverb trail (it is generally assumed
that reverb should become darker as it
decays, as happens in most better concert
halls). It is worth noting that not all music
requires these reverberant trails.
Typical reverb times run between 1 and
2.5 seconds, predelays run from about 30ms
up to 100ms, and there will be a spectrum
control capability that changes the spectrum
of the generalized feedback in the algorithm,
so that it gets darker or brighter as it decays
(both applications have their uses). Happy
tweaking!
So when we are slaving away in postpro-
duction, we have four sort of micro-time
zones (relative to each musical event) to be
concerned about (see Fig. 1): the direct
sound, two early lateral delays (maybe), ambi-
ence (probably), and reeeverbbbbbbb
(maybe, but it depends on the track and the
music).
Our success in this enterprise will depend
on how effectively we can manage time in
these various micro-zones.
Convolution reverbs vs.
traditional reverb engines
Traditional digital reverbs use mathematical
algorithms to simulate the sorts of behaviors
that we encounter in a variety of rooms, halls,
and spaces. Unfortunately for us, literal mim-
icry of large hall behavior is computationally
expensive to a point where we simply dont
do it, so traditional reverb engines indulge in
a number of tricks and sleights-of-ear to
reduce the computation needed for accept-
able reverb.
Meanwhile, the developers of traditional
reverbs quickly found that we users wanted
flexibility. The algorithms must be easily and
effectively manipulated. In that traditional
reverb engine, the algorithms have therefore
come to include complex arrays of delay times,
feedback loops with filtering, output taps in
great stereophonic profusion and usually some
randomization of time to (a) prevent persistent
feedback loops, ringing and tonal centers
from developing, and (b) to add a livens and
variability to the reverberant quality. The values
of all these are usually under the control of the
user, to some extent at least.
The best traditional reverb engines have
done this well and are, to my mind, a pleas-
ure to use. However, they are not the real
thing, just low-cost simulations, and however
nice they may sound, that can be audibly
obvious.
The number of reflections, the times at which they
arrive, and the angles from which they arrive are
determined by the size and topology of the room,
and the relative positions of the sound source and
listener in the room.
Figure 1. The four micro-time zones we need to be concerned with when filling in reverberance, particularly with synthesizers and samplers. The times, of course, are
approximate and need to be tweaked by ear for each musical context.
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f e a t u r e
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The inventors of GigaStudio take a look
backward, forward, and straight ahead
T
heres a very good
chance you wouldnt be
reading this magazine if
GigaSampler (now TASCAM
GigaStudio) hadnt come along
a few years ago and revolution-
ized sampling. Gigas big
advance was streaming sam-
ples off hard drives instead of
playing them from RAM, which
meant they could be as long as
necessary to capture the full
expressive range of the instru-
ment being sampled.
That greatly improved the quality
of the sounds, which led to the
new musical medium this maga-
zine is all about. Today there are
other streaming samplers, but
Giga is still going strong and
breaking new ground.
Giga was the product of a tiny
company called Nemesis before
it became part of TASCAM. Two
of the people who invented it,
Jim Van Buskirk and Joe Bibbo,
still work on it.
GigaInventors:
Jim Van Buskirk
and Joe Bibbo
Pic caption: Left to right: Giga developers Jim Van Buskirk, Joe Bibbo,
and Robert Bischof. Out of the pic but on the team: Scott Mitchell.
You were working on virtual instru-
ments back in the day before the day.
Jim Van Buskirk: This stuff started out
with a realization that you could conceivably
do signal processing on a general-purpose
CPU like an X86. This was when the 386 was
in its reign and the 486 was the hot new
thing. It was during a time that Joe and I
were working at Texas Instruments Signal
Processing Group doing software synthesizers
that ran on DSP. (Later we moved to
Rockwell.)
But we realized that the host processor was
becoming a viable computing resource as
well, and the memory restrictions went away
to a large extent when we went to the host-
based system. The trade-off was computation
for memory.
Why is that? One would think youd
have more power using a dedicated,
optimized processor.
You could get more MIPS of computing
power, but the systemat least for sample-
based instrumentsseemed to be memory-
starved, especially on those embedded-type
DSP designs that we were working on.
One of the things that was prevalent at the
time was that multimedia chipsets were start-
ing to become popular for gaming, and we
started putting together solutions that run in
software that would emulate the behavior of
what chipsets were doing on the motherboard.
At the time, the General MIDI stuff was
starting to become popular, and there were
some 2-operator FM chips on the mother-
board. General MIDI was starting to replace
those for better music authoring in games.
So we started looking into how we could
make a General MIDI solution sound better
than the chipsets, largely by virtue of the larger
memory. We knew we had kind of a weak com-
puting engine, but we had all this memory.
I think that was one of the things that led
us to some of the caching-type innovations
that we have done. We came up with caching
algorithms to make us sound better than the
actual chipset.
The timeframe was the early 90s. Actually
Id just come out of graduate school, and we
went to work for Texas Instruments, which
was the DSP Powerhousethey were the
leader in digital signal processorsand the
DSP technology, the chipsets were just com-
ing into their own. When we started, the
Digital Processing group was small. Now it
dominates Texas Instruments, a multi-billion
dollar company. DSPs were just kind of find-
ing their place in
the world of tech-
nology.
Joe Bibbo: Its
kind of strange
that we kind of
abandoned the
dedicated DSP
processors and
went to X86. If
you put them
side-by-side, the
X86 is not a good
processor com-
pared to a dedi-
cated, special-
purpose piece of
silicon.
But it had this
advantage that it
was on every
desktop. So
accessibility beats
out performance,
and the cost of
your hardware is
free, because everybodys got one, right? You
dont have to sell them a piece of hardware,
because theyve got it in their system.
The whole memory issue becomes a real
cost-prohibitive thing, because when you put
memory onto a system and it only serves one
purpose, say a reverb, then it gets very
expensive. But if youre using the host mem-
ory, then its used for reverb, its used for your
filters, its used for every functionit has a
multipurpose value, and therefore it becomes
much cheaper.
So that whole cost model made program-
ming on the X86 or on the desktop a very
good solution, even though DSP technology
was really coming into its own at the time.
How did that lead to GigaSampler?
Was it an outgrowth of this?
Jim: Technology-wise it was, because we
started looking at caching really early on. This
was at a time when we had proven software
wavetables; we had shipped them on several
million motherboards by then. And it got
under the Microsoft radar.
We had kind of a jolt right around 1995,
when Microsoft came and announced, Oh,
wed better put this into the operating sys-
tem. What that meant to us was, well,
theyre going to put it into Windows, theyre
going to give it away for free and, well, there
goes our measly software royalty that we
derived our livelihood from.
We were very scared that we would be out
of business overnight. So we looked into
going into the high end, going into the pro-
fessional market, because the only we could
make money was with a value-add.
It had to be better than Microsofts, and to
the computer guys, the HPs of the
worldquality is a hard sell to them. But in
the professional market its completely the
opposite: quality is the sell, and people will
pay as much money as they need to if its
that markedly better.
So thats basically the environment under
which Nemesis was able to spin off from
Rockwell, where we were working at the
time. We were to develop into the profession-
al market, to continue the value-add strategy
in the computer market, which we were not
going to participate in at all.
Of course, it was a passion of ours as well,
that this is really what its all about. Going
after the quality in the professional market
was a dream.
Giga changed the way people write
music, at least for media. How much of
that had you forseen? Im sure it took
everyone by surprise how far the
libraries pushed the limits, how big and
incredibly detailed they got.
Well, at first it was if it was software, it
wasnt as goodthat was sort of the percep-
tion. It had to be a big old piece of gear that
was sitting in a rack.
We had no hardware support eitherwe
didnt have a single manufacturer out there
willing to write drivers compatible with our
product. So we actually had to write some low-
level hooks to use a single 2-channel S/PDIF
cardthe SoundBlaster Gold at the time!and
hope that would bootstrap us into enough
market share that we could attract a profes-
Fig. 1: iMIDI (for Intelligent MIDI) rules are a new feature in Giga 3. Versions 3.1 or
later include the DEF (Dynamic Expression Filter) mentioned in the interview.
I remember the day Jim came up to me and said,
You know, it might be kind of cool if we could
feed the engine from samples off the disk.
sional sound card maker to support us. Because
we were a little start-up company, we didnt
have enough money to make it ourselves.
As far as how we thought the composers
would catch on to that, we really didnt know
until we actually built some really cool libraries.
Perhaps the GigaPiano was the first one.
Thats all it came with originally.
Yeah. And then peoplemostly independ-
entsstarted to realize what you could do
now that the whole memory barrier has been
blown wide open. And of course then what
follows, once the sounds get that good, is
that the composers are now interested.
It was a little kernel of a thing that started
to snowball.
What was it you envisioned when
you came up with the idea?
Joe: I remember the day Jim came up to
me and said, You know, it might be kind of
cool if we could feed the engine from sam-
ples off the disk.
And I really kind of thought he was crazy,
because at the time we were trying to get the
engine to reasonable polyphony. I distinctly
remember a conversation about how great it
would be to combine the control of MIDI
with the quality of audio.
At the time, you had your samples in RAM,
they were limited, you had to loop themwe
did a few caching things that gave our sam-
ples a little bit of depth, but nothing to the
extent that the Endless Wave patent [from
Rockwell] achieved.
So you had these samples of endless length
that were controlled by MIDI. That was the
motivation. At the time I didnt think it was
achievable. And probably at the time it
wasnt! But it took some time to develop, and
by the time the processors and the hard disks
came around, we were getting decent
polyphony, and thats when the GigaPiano
came out.
And the machines got faster so you
were able to run at a lower latency.
Latency was a big issue, and thats why we
ended up architecting it underneath
Windows: to achieve that real-time feel and
get a decent amount of polyphony and
stream stuff off the disk.
Is there an advantage to using dual-
core processors for Giga?
There is an advantage in the sense that any
time you have two processes going, they can
take advantage of a dual processor. So if you
run Giga and a bunch of VST plug-ins, theyre
going to be on multiple threads, or processes.
Theyre going to take advantage of the two
processors.
If you run a lot of disk-streaming, the bot-
tleneck is the actual hard drive, not the soft-
ware, so you dont get too much advantage
there with a dual processor. But if youre run-
ning a lot of plug-ins with your sampler or
youre running your sequencer on the same
system, yes, you do get an advantage.
Jim: For the long term for our model-based
instruments using convolution, its a good
thing. The dual-core stuff is looking very
favorable when you have a combination of
convolution and streaming.
Of course, the first true modeled instru-
ment will be the Giga Violin. Thats coming
out soon. Its not just sampling, its sampling
and modeling.
The other big question is memory
access. I know you worked very hard to
improve it in Giga 3. As a matter of
fact, I get a 20% improvement on one
of my machines over Giga 2 with the
same amount of memory. But its still
an issuewere still using multiple
computers not so much for the voices,
but to have lots of programs loaded.
What weve seen with the technology is
that disks are not getting any faster, theyre
getting larger; CPUs are getting faster. So the
reason people use up memory even in a disk-
caching system is to buffer up the data. Now,
you can make the buffer smaller and you can
get less polyphony, or you can try and allo-
cate more memory.
But the problem with 32-bit Windows and
with 32-bit processors is theres a theoretical
limit to how much memory you can get, and
thats 2 gigs. You can never get to that theo-
retical limit, so its somewhere less than 2
gigs. Thats why you dont see motherboards
populated with much more memory.
Now, the thing thats right around the cor-
ner is 64-bit operating systems and proces-
sors. That throws away the theoretical 2 gig
limit, and youre up to terrabytes; now your
sampler can use as much memory as you can
stick on your motherboard.
I think its our strategy to take advantage of
the next generation of processors and operat-
ing systems to address those memory limita-
tions.
Why is the theoretical limit in
Windows 2GB and 4GB per program on
Mac G5s?
Well, in a 32-bit operating system you get
4 gigs total. Windows breaks up the memory:
the kernel gets 2 gig and an application gets
the upper 2 gigs. When they architected this
10, 15, 20 years ago, they figured 2 gigs was
going to be more than enough for anybody.
Were in a transition point. But we still have
probably the smallest buffer size and the best
efficiency coming out of that small buffer, so
even though we have this temporary limita-
tion, we do pretty well with it. Even if you
can allocate bigger memories with some of
the other samplers, youre still not loading
any more samples, because youre allocating
larger buffers.
Is your buffer static for each sample,
or does it change depending on whats
loaded?
Its static. I mean, its dynamic in the sense
that if the sample is extra-small, it will shrink
or grow to the size of the sample. But once
you get to streaming samples, it gets to a
point where it doesnt matter how big the
sample is, its going to be a static size.
How is it that youre able to load
Giga to one cowbell short of the point
where it says I cant load any more,
and it runs absolutely reliably? Is that
because youre bypassing Windows?
Yeah. Once you start getting into the upper
memory, then everybody in the user mode
Fig. 2: The Sonic Implants Symphonic Collection uses the DEF and also the Portmento Reshaping Tool, which
is easy to edit in the iMIDI Rules Manager.
VI
i n t e r v i e w
starts to slow down. Windows is not a very
good real-time operating system, and youre at
the mercy of the OS. But because were sitting
at the kernel, underneath the OS, our engine
continues to run at a very high priority.
GigaPulse, the integrated convolu-
tion processor, has some new features.
Sonic Implants uses it for their Legato
mode, for example.
Joe: DEF (Dynamic Expression Filter). Thats
really cool. If youre playing a large cross-fade
instrument with multiple velocity layers, to
get really good dynamics you can change the
controller value. So its not a just a dead thing
after youve played the note. But youve basi-
cally lost your velocity, because you have to
position your controller at note-on and youre
basically going to get that response.
One of the cool things about DEF is that you
get both. You get the after-note-on dynamics
and timbral variation, and you get velocity. The
reason is that its an 8th-order phase-corrected
morphing filter that can do things, such as very
accurately, say, take a double forte sample and
change it into a pianissimo.
What typically happens when you design
your DEF instruments is youll cover more of
the ground with less of the samples. Youll
pull the higher velocity split points lowerthe
double fortes, fortes, mezzosgiving them
more of the upper velocity range and letting
the filter give you the continuous expression.
By doing that you can get more of the veloci-
ty response, you can bias the velocity
response by changing the DEF as you play.
And then you can vary it after youve hit the
note and youre holding it. Its tied to a con-
troller, which can be anything including chan-
nel aftertouch.
Some people are still kind of used to a
straight cross-fade paradigm. If youre using
DEF it does take more computation per voice.
But if youre doing it instead of a cross-fade,
for example one stereo note with four vel-
ocity layers would be eight individual voices
per note; if youre doing it with DEF it would
only be two voices.
8th-order means what, a right angle?
[Each order in a filter increases the
cut-off slope logarithmicallya 1st-
order filter is 6dB per octave, a 2nd-
order is 12dB/8ve, 3rd-order is
24dB/8ve]
It can be a right angle, but its designed to
convert samples from one dynamic layer to
another and sound very realistic without los-
ing the air on top. In the past, if you ever
tried to do that with a multimode filter or
something, especially a lowpass, youre going
to lose the air on top on your pianissimo
maybe that little bit of felt on the piano ham-
mer, for instance. Its a beautiful sound, and
youll lose that.
But it is truly a morphing response. There
are coefficients that you set up at the lowest
dynamics, at the midpoint dynamics, and at
the high dynamics. At every point in
between, those coefficients are dynamically
morphing. So its not like a cross-fade of three
different filters, its a real morphing
responsethe actual filter response morphs
from one point to the other. And that is what
gives it the realism.
What about the Portmento
Reshaping Filter?
Its a similar sort of morphing filter para-
digm. Its a little trickier to get it dialed in,
and it is very instrument-dependent. The idea
is to chase the resonant peak, so you can
actually slide the notes. If you have it loaded
with the right coefficients, you can align it so
that as it slides through, you have some
counter-measure to those resonant peaks, so
you can eliminate that sort of munchkin
effect. If youre willing to really spend the
time with it, you can get some really good
portamento with it.
The tricky thing is when you gliss a note in
a room, youre transposing the reverb, but
you instantly notice that you dont get that
nice Lyle Mays effect, where hes bending the
notes and you get this really big sound. But if
you use a room in the convolution processor
thats pretty similar, and also set it up preserv-
ing the spatial imagery, you can position the
left and right channels independently when
you hold down the Control keythen you
can get really close to that sound, even if
youve sampled it in a room.
So if you put the left ones really close to
one of the close-proximity left channels and
the right really close to one of the right-prox-
imity channels, it preserves the spatial cues
that are in the original sample. Its not theo-
retically perfect, but it sounds good.
Any closing thoughts?
Jim: A couple of things about our team.
Joe and I have been working together since
the early 90s. In the 94 time frame, both
Scott Mitchell and Robert Bischof joined the
team. This is the same team thats been
together through the whole vision, the
inventing of the artto innovate and not
imitateits not changed. TASCAM is really
doing a good job of letting us continue to
innovate, investing in things you wont find
anywhere else.
Its always easy when theres a market out
there for a product that does a certain thing.
They say, Oh, theyre selling a billion a
month, weve gotta have one of those. But
when you have a crazy new idea that nobody
has on any sales forecast and you try to get
people to invest in it, its a much harder
thing. But thats the way weve operated all
along, and TASCAMs being real cool about it.
The same team, the same working environ-
ment, the same amount of artistry and cre-
ativity thats done it in the pasthopefully
well be able to continue to achieve. The envi-
ronments really good for it. VI
N
ative Instruments Kontakt sampler
started out as a memory-based player.
Its big strength was how it allowed
you to time-stretch, slice and dice, and
process loops and samples, plus it had some
unique processing features.
Then they added Direct From Disk (DFD)
disk-streaming technology so it could handle
modern sample libraries, and developers of
these libraries began licensing OEM Kontakt-
based players for their sample libraries. This
allowed them to copy-protect their material,
and the no-sampler-required feature made
their products more attractive to users.
Early versions of Kontakt and its whole fam-
ily of players had some disk-streaming prob-
lems, but by somewhere around version 1.5 it
was running smoothly on both Windows and
Mac OS X. Naturally, it was then time for a
major upgrade of the productwhich brings
us to Kontakt 2.
K2 has a number of new features, but
some of the most glaring ones are: a re-
worked interface, including an onscreen
mixer with a channel strip, complete with
inserts and aux sends for each output; many
new effects modules you can drag and drop
onto the mixer; an Undo command when
editing; up to 64 programs on separate MIDI
channels per instance; a built-in convolution
processor; up to 16-channel surround sound;
Review by
Doyle W. Donehoo
Native Instruments Kontakt 2
sampler, $449; update from
original Kontakt, $149.
Native Instruments USA, 5631A
Hollywood Blvd., Los Angeles, CA
90028. 323/467-5260.
www.NativeInstruments.de
Formats: Windows XP, Mac OS
10.2.6 or higher. VST, Audio Units,
RTAS, DXi, Core Audio, ASIO,
DirectSound.
License: Online challenge/
response installation.
Two machines, one user.
support for 24-bit/192kHz samples; the ability
to import many sample formats; a new pro-
grammable script processor for creating all
kinds of responses to MIDI input; and an
excellent bundled 2-DVD sample library, one
of which is filled entirely with instruments
from the Vienna Symphonic Library.
System requirements
The minimum system recommendation for
Kontakt 2 is a 1.4+ GHz computer with 512
MB RAM running Windows XP, or Mac G4
800 MHz with 512 MB RAM. In reality, you
should have a much more powerful comput-
er. (While this review was done on a PC, I
have witnessed the use of K2 in Macs, and
everything is essentially the same.) The newly
revamped audio engine takes advantage of
dual processors, and the AltiVec co-processor
in Mac G4s and G5s.
K2 can make use of just about any sound
card. On Mac OS X it uses the built-in Core
Audio driver; on Windows its compatible with
ASIO, DirectSound, and MME. A lot of high-
end cards provide all these drivers and more,
but theres also a good number of lower-end
PC cards that support low-latency ASIO and
MME drivers with decent audio specifications.
Native Instruments (NI) promises about 5ms
of latency on a decently fast computer with
ASIO drivers, which is probably the best
Native Instruments
Kontakt 2
VI
r e v i e w
format to use if you have a choice.
With a very fast computer and a top end
audio card, lower latencies are possible. Up to
a point, the processing power requirements
and buffer sizes are on opposite sides of a
scale for audio output; the larger the buffer,
the higher the latency, and more powerful
computers can get by with smaller buffers. 5 -
10ms of latency is nearly unnoticeable and
more than good enough for most audio tasks.
Installation and reg
Installing K2 is trivial, and anybody who
has had any experience installing software
should not have a problem. In most cases it
will just be a matter of popping in the install
CD and following instructions when the
installer loads. At worst you may have to go
to the CD and launch the installer manually.
I had the program installed, launched, and
playing its first instrument literally within min-
utes. After setting the audio and MIDI in the
setup dialog, I was ready to roll. K2 saw all of
my drivers and MIDI ports (including my
Musiclab MIDIOverLan MIDI ports, which
work over ethernet instead of a hardware
MIDI interface), so configuration was no great
chore.
The next task was registration, and NI has
provided a separate registration application to
smooth the way. By following the instruc-
tions, I was able to get full authorization via
email within minutes.
As with every piece of software, its a good
idea to check for updates immediately after
registering. In this case they will improve the
performance of the program.
For those leery of buying something before
knowing if it is going to work, you can down-
load the full-featured demo version of the
program, and after K2 is installed you will
have a generous 30 days to try it out.
Interface first
impressions
K2 operates as a stand-alone program, or
as a VSTi 2.0, Dxi, DirectConnect, Audio Unit,
RTAS, or MAS plug-in. With support for 64
MIDI channels on separate ports, you no
longer need to run it inside a host if youre
dedicating a machine to it and want to access
more than 16 MIDI channels (i.e. if you want
to use more than 16 independent instrument
parts).
While the new version will be perfectly
familiar if youre used to the original one, the
interface has been improved. Probably the
most notable addition is an output mixer,
complete with excellent metering, auxiliary
effects sends, and an output fader.
The left third of the screen is an improved
browser, a tab-controlled window dealing
mostly with files and effects. The right two-
thirds is the virtual instrument rack, which is
really four sets of 16 channels worth of sam-
ple programs.
Each of the four racks allows multiple
instruments to share MIDI ports and channels
for layered sounds. You can save entire con-
figurations of programs and effects, and load
them all at once.
While fly-by or contextual help would be
very useful, the 213-page manual can quickly
clear up any question that may arise, and the
interface is fairly intuitive. The only issues I
found were related to K2s windows: in stand-
alone mode, you can position the K2 window
on the desktop to your liking, but K2 does
not remember where that position is from
session to session. Also, while theres a choice
of three sizes to display the entire application,
you cant scale the size of the K2 window.
Nonetheless, K2 is a marvel of efficient use
of space in a limited window area. NI has
managed to create a virtual rack interface that
is easy to understand and brings all the major
features within a few mouse clicks.
K2 uses drag and drop to good effectin
fact you can drag and drop effectsgiving
you a variety of ways to interact with the pro-
gram and load instruments. For example, you
can drag and drop an instrument into K2
from the Windows File Explorer, or drag an
instrument into the virtual rack from the K2
file browser on the left side of the screen. Or
you can just use the Save/Load button on the
top of the screen.
Getting an instrument loaded and testing it
is as easy as dragging an instrument into the
rack, bringing up the virtual keyboard by hit-
ting the Keyboard button, and mouse-clicking
on a key to see what kind of sound it makes.
In the original version you had to click Edit to
expand the instrument, then open up the the
Keymapping area to get at the virtual key-
board.
Instruments in the virtual rack can be dis-
played in three states: in a space-saving nar-
row band, in a larger size resembling a single
real-world rack space, or expanded into the
edit mode alluded to above, which displays
the various settings and effects. Many of the
instrument modules can be collapsed or
expanded for the best use of interface real
estate.
Library
Native Instruments has added a lot of value
to K2 by including a generous, extensive
sound library thats complete enough to cre-
ate professional music projects right out of
the box. Some users will see the K2 library
collection as the primary incentive to buy the
instrument.
The library comes on two DVDs and totals
just over 15 gigabytes of data. There is a
comprehensive selection of orchestral instru-
ments from the excellent Vienna Symphonic
Library (VSL), and this library alone takes up
one of the two DVD Kontakt Library disks.
Among other companies, East West, Zero-G,
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Percussive Adventures 2
Sound library and virtual instrument
O
ften, rhythm libraries are clearly based
around a drum kit, drum machines,
ethnic instruments, or industrial
sounds. You may hear a bit of cross-breeding,
but often its an obvious flavor, such as hey,
lets add a tabla to this house groove; we can
be really cool and feed it through a filter as
well! (Not that I havent done this myself.)
However, in Percussive Adventures 2, Kurt
Wortman and Tony Humeckea.k.a. Beta
Rhythm Farmare completely at home mix-
ing kits, metals, ethnic instruments, and stu-
dio processing in the same groove. The eclec-
tic result sounds completely unforced and
often fresh as a result.
This 3 gigabyte library is presented as 70
pieces, many of which are rhythmic-based,
but it also includes a few sound environ-
Review By Chris Meyer
ments, free-time improvisations, and collec-
tions of layers to add on top (such as the very
useful Movers and Shakers). Each rhythmic
piece typically contains a linear performance
averaging 12 to 16 bars, including an intro,
middle, and ending (which you can slice
apart later if you desire). The same patch also
contains another dozen or two breakdowns,
variations, and layers that can be used with
other grooves.
Often, you also get a selection of hits, pads,
and endings thrown into the mix. Most of
these variations are four bars in length (nice),
but even the 2-bar loops are so detailed that
you dont become bored quickly. Tempos
range from 65 to 181 bpm, centering around
90 to 130. Some pieces contain melodic parts
or layers that can be played melodically.
Percussive Adventures 2,
$399.95.
Distributor: East West
(www.soundsonline.com)
Platform: Mac OS 10.2.6 or
Windows XP; standalone (through
Core Audio, DirectSound, or ASIO)
or as VST, DXi, ASIO, RTAS, or
Audio Units plug-in.
License: May be used in owners
musical compositions, but not as
part of sample or music libraries.
Challenge/response installer limited
to two unique installs.
The update to the popular original PA
is eclectic and fresh
random
tip
This library uses the common Native
Instruments Intakt front end (please see
www.VirtualInstrumentsMag.com for more
about Intakt); most loops here rely on their
granular Time Machine, although a few are
also supplied in sliced Beat Machine format.
The odd loop variation takes advantage of
Intakts processing capabilities, such as creat-
ing gritty lo-fi versions.
Some of you may be familiar with the origi-
nal Percussive Adventures library, which con-
sisted of mainly cinematic-flavored pieces
crafted by Kurt and Tony as well as Kim
Edmundson and Jorge Patrono (executive
O
r to put it another way, G5s can hold 8 or (in the latest dual-
dual processor model) 16GB of RAM; can you really access 8GB
or 16GB of samples? The answer is no, but its more complicated
than that.
In OS X 10.3 and 10.4 (Panther and Tiger), each program can
access a theoretical maximum of 4GB of RAM. By program we
mean the host DAW; all the softsynths, samplers, and plug-ins its
hosting run inside that 4GB memory space.
From that 4GB you must subtract about 500MB of overhead for
Apple system libraries and frameworks that are shared by all open
programs. And in the real world (at least in our real world) you need
to subtract another 500MB or more for stability.
With breathing room, that translates to between 2.75 and 3GB of
actual memory access in the real world. 2.75GB is pretty conservative,
and the actual performance varies from session to session. Note that
you could still load a considerable number of synthesizers and plug-
ins on top of thiswere only talking about memory access.
Now, OS X itself takes about 500GB, but that memory is outside
the 4GB of application memory. So if you have at least 4.5GB
installed, you can max out your system with one program: 4GB for
the program and .5GB for the OS outside the program.
However, it pays to install more than 4.5GB. For one, each pro-
gram you open can access its own 4GB of RAM. So if you have 5GB
installed, for example, you can run something like Ableton Live or
Propellerhead Reason outside your DAW program and stream its out-
put into your DAWs mixer using the ReWire protocol. Memory over
4.5GB can also be used for standard activities like opening a browser
or checking email.
Furthermore, OS X caches disk reads, and it will grab available RAM
for that. This should greatly improve the performance; or looking at
the glass half empty, when it doesnt have RAM available, it starts
using virtual memorymeaning that it starts pretending your hard
disk is RAM.
OS X always uses some virtual memory, but its not good for real-
time applications for it to start caching disk reads to the diskor
worse yet, swap some of your DAWs memory space (either program
code or sample data) to the hard disk. That can and probably will
lead to instability.
So does that mean you should put in 8GB or even 16GB in a new
dual-dual G5? Not necessarily, because theres some point at which
the benefits of RAM caching dwindle. 4.5GB of RAM is pretty good, 5
or 6GB is very good, but 8GB is not a waste of moneythe OS will
use it all.
The reason for the 4GB-per-program limit has to do with the 32-bit
operating system being able to access that much memory only. But
the G5 is a 64-bit processor, and OS X is supposed to be a 64-bit
operating system.
Thats not a lie. Command-line programs can have 64-bit memory
access, which is for all intents and purposes limited only by what its
practical to install in a computer. But anything that uses the Macs
graphic interfacei.e. every program we use for musicis still a 32-
bit program. That may change in 2006.
In the meantime, 2.75GB of RAM isnt enough to load an entire
modern orchestral library, but its still pretty darn good.
Special thanks to Andrea Gotti of Redmatica Software for explaining all
this so lucidly.
How much memory does it make sense to
install in a Mac G5 so you can load the
maximum number of sample programs?
Apple Activity Monitor reading of a session loaded into Logic Audio on a
dual 2.5GHz G5 with 5GB of RAM installed. The session has 11 fully-
loaded instances of the 8-slot) East West Colossus player loaded (88 instru-
ments), along with 26 EXS24 samplers loaded with the biggest programs
we could find, which are Vienna Symphonic Library Performance Legatos.
This has pretty much reached the limit of how many sampler programs
that can be loaded, but theres plenty of processing power left over for run-
ning convolution reverbs, synthesizers, and plug-ins.
produced by Christopher Page). Interestingly,
although this follow-up contains only two of
the composers from the original version, it
features a much wider range of styles and
sounds, touching on rock, jazz, Latin, Arabic,
post-something hip-hop, film scores, and
many undefined points in between and
beyondsometimes all in the same piece!
What is constant is an underlying funkiness
often missing from ethnic-flavored grooves.
Most of the loops are intricate with a strong
flavor, which means theyre not going to melt
into the background of your next ballad, but
on the other hand they can give your compo-
sitions a distinctive spice. The recording quali-
ty is very sharp, with an appropriate amount
of ambience mixed in.
For those on a budget, East West provides
Percussive Adventure 2 LE for $129.95 as part
of their ProSamples line. This 1 gigabyte
library provides all of the full performance
versions of the pieces, but not the break-
downs and variations. An LE owner can
upgrade to the full version for $299.95but
if you can swing it, Id buy the full version
from the get-go. VI
VI
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VI
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T
o put the significance of Prominys new
Les Paul Custom Electric Distortion and
Clean Guitar library in perspective, lets
start with some history.
Guitar samples have been around almost as
long as sampling itself, but early attempts to
sample electric guitar were primitive at best.
You could hope for some guitar flavor, but full
tracks with these guitar samples always
showed flaws. Being a part-time session gui-
tarist, I felt safe in the knowledge that I only
had to compete with other sometimes inebri-
ated 6-string slingers.
As samplers started to be able to hold
more memory, companies become more
ambitious in their approach. Libraries by Hans
Zimmer, Sonic Implants, and Quantum Leap
let you construct full guitar tracks, and some-
times they sounded quite realistic. Still, even
the average listener could make out that
something was not quite right after just a
short listen. As they said about saxophone
libraries, I often heard fellow composers men-
tion no one is going to get sampled guitar
right.
State of the art
Then Vienna Symphonic Library introduced
their Overdrive distorted guitar library, and
Prominy
Les Paul Custom
Review by
Craig Sharmat
Prominy Les Paul Custom
Electric Distortion and Clean
Guitar Library, $599; clean or
distorted versions alone, $349
each.
Prominy, Inc. 949/502-7944.
www.Prominy.com
1 foliage, Irvine, CA 92603
Formats: GigaStudio 3.0 (prefer-
ably) or 2.0; Kontakt 2.
License: no restriction to the num-
ber of machines; must give credit
on commercially-released audio or
audio-visual products.
Electric Distortion and Clean Guitar library
what had seemed impossible was no longer
so. Among other tricks, VSL uses their famous
Performance Tools to insert samples of the
transition between two notes when appropri-
ate. While Overdrive is limited to one specific
guitar and amp sound, a skilled MIDI pro-
grammer can create very realistic simulations.
Drinking has since taken on new meaning,
and now two other formidable electric guitar
libraries have hit the market in the past few
months: Bela D Medias Liquid Distortion
(something Im familiar with occasionally),
which features a Paul Reed Smith guitar, and
now the subject of this review: Prominys LPC
(Les Paul Custom).
The point of this article is not to make a
comparison between these two recent
libraries, but to review Prominys offering with
the absence of further drinking references.
Prominy LPC
LPCs clean and distorted versions are avail-
able individually, or you can purchase both
together for less than the individual prices. All
the articulations are identical; both versions
were recorded at the same time.
It would be a mistake to write the clean
library off as a lightweight afterthought, how-
ever. These samples can serve as an excellent
rhythm bed, or theyre great for clean solos in
blues, rock, or jazz (the octaves work especially
nicely for Wes Montgomery-style playing).
Perhaps more importantly, these articula-
tions were recorded while the player was lis-
tening through a Marshall amp, so they have
a real rock attitude. That means they can be
re-amped any way you like (e.g. through a
plug-in or a real amp) and end up with a real
feel.
I installed Electric Distortion & Clean
Guitar into a PC with a 2.8MHz Pentium 4
and 2GB of RAM, running TASCAM
GigaStudio 3.1. As we went to press, Prominy
announced a Native Instruments Kontakt 2
version as well (see sidebar). One of the
things I noticed right away was the immense
size of the library: 60 gigs of guitar samples,
probably making it the largest sample library
of any simple instrument on the market.
This of course has pros and cons. On one
hand, you will need to make some serious
storage room for LPC, possibly adding
another hard drive for it. Depending on how
serious you get with your MIDI programming,
it could take up most or even all of a single
computers memory as well.
On the other hand, this quantity of
detailed articulations makes LPC a truly versa-
tile virtual electric guitar library.
Power
Having some electric guitar knowledge will
serve you well with this library, although it
has plenty of simplified patches for those with
a limited understanding of the guitar.
The first patch I opened is a so-called
players instrument, and its impressive. This
patch contains long notes, staccatos, trills,
power chords and a number of other tech-
niques. These articulations are accessed
through keyswitches, which as most of you
know are temporary on the fly program
changes triggered from an unused portion of
the keyboard (usually the very bottom
octave). This single patch would be fine for
many of my programming needs alone, but it
just scratches the surface of the library.
There is a multitude of keyswitch patches,
and these can be altered by mod wheel
velocity andextremely importantlychan-
nel pressure (meaning aftertouch, i.e. you
press down on keys after the note has sound-
ed to send control changes). The channel
pressure allows you to add vibrato at any
time you are holding a note or chord. This
function is used throughout the library, so it is
something to be aware of.
Points of light
This library is too vast to detail all its fea-
tures, but here are some highlights.
The slide patches are spectacular. These are
finger slides up and down the fretboard to
targeted notesreal-time slides from note to
note, power chord to power chord, up or
down. There is no octave limitation as in
other librariesthe full string range is includ-
ed; you simply play it and it works, and you
can add vibrato at any time.
The muted notes sound great, and the
auto round robin feature (which substitutes a
different sample for every attack) is simple
and effective. This means you can play mutes
without a machine gun effect.
The trills sound fine, though I wish they
were looped so you could play them indefi-
nitely. LPC includes pitch bends of varying
depth and speed. There are also fret slides of
all intervals and speeds. In fact, it seems
unlikely any articulation you can play on a 6-
string electric guitar has been missed, or
could not be some how worked out with all
the tools available.
A small complaint is that although every-
thing is nicely organized, some of the patch
lists get long. For example, every string with
every bend at any speed is broken down to
an individual patch, and you will have to
scroll through these.
Working with LPC
While LPC is a huge library, it can be very
efficient. Many patches have a lot of features
programmed into them. For example, a single
patch even without keyswitching can contain
multiple velocities, vibrato that you introduce
using aftertouch, and then you can select a
different version of the same patch with the
mod wheelperhaps shorter or stopped
notes.
On the other hand, guitar can be a com-
plex instrument to program, and to getting a
completely realistic performance over a long
period of time may demand that you use
many articulations. This used to mean setting
up a number MIDI tracks to accommodate
those patches. Now with GigaStudio 3s
stacked instruments (see sidebar), you can
access a large number of articulations on one
MIDI channel.
Another interesting feature of the library is
the keyboard-guitar set-up. You can have six
keyswitches corresponding to the six strings.
That lets you play notes on the keyboard as
realistically as you would on the guitar. Some
pretty good knowledge of how a guitar is
played would be a prerequisite for getting the
best results from this feature. But even if you
dont play guitar, when you start working
LPC in Giga
LPC takes advantage of some of GigaStudio 3.1s advanced features in ways that no other
library has yet done.
The seamless slide patches are one example. These patches automatically insert actual sam-
ples of slides from every note to each other one on the same string, all the way up the fret-
board. No external tool is requiredyou just load the program and play.
Many multi-dimensional patches are scattered throughout the library, and Prominy has
recently incorporated a new feature in GS 3.1 that allows you to stack patches together. What
this means from a user standpoint is that you can be playing a major chord, then keyswitch
to some other chord type without switching MIDI channels. Or you might be switching
between various single-note programs, or a combination of chords and single notes.
This can significantly reduce the number of tracks you need to set up in your sequencer to
play the library.
VI
r e v i e w
with the range limitation of a guitar string,
youll catch on pretty quickly.
Sound
So how does it sound? In two words: really
good. The Les Paul was recorded into a
Marshall 1959SLP. If programmed well, it will
be indistinguishable from a real guitar player.
Were not talking pre-made patterns here
this is all do it yourself, I hear this in my
head, and LPC can do it. The full version
(meaning both clean and distorted libraries)
LPC in Kontakt 2
vs. Giga
Prominy announced a Kontakt 2 version of
the library as we went to press; you just
download the K2 programs from their
website in order to use it. Using the library
is almost the same in both platformsfor
example theyve managed to reproduce
the LPC Realtime Legato Slide in K2but
developer Akihito Okawa explained the
differences to us.
- K2 doesnt have Giga 3s Stack
Instrument feature, which means you
need to edit the Kontakt-format .nki file if
you want to customize it as explained in
the other sidebar with this article (e.g. to
get lots of different playing techniques on
one MIDI channel).
- On the other hand, some of the .nki files
include more controllers and switches
than the Giga version.
- K2 has a purge feature that removes
samples that arent being used from mem-
ory to save RAM and allow you to load
more programs (see K2 review elsewhere
in this issue). That can be useful if youve
loaded a lot of guitar playing techniques
into memory but arent using them.
Conclusion
Prominys LPC is the most complete electric
guitar library ever made. It sounds great and
breaks new technical boundaries with the use
of Giga Studio 3.
If you use a lot of guitar on your tracks and
cant find a real guitar player who showers
occasionally, or if you just want to do it your-
self, I cant recommend this library highly
enough. It uses up a lot of hard drive space,
but thats the price of having no compromis-
es. LPC truly sounds like you have a real gui-
tar player living in your studio, only with
none of the downsides. VI
As a guitarist, Craig Sharmats (www.score-
dog.tv) credits include Ronnie Laws (for whom
he served as musical director), the Pointer
Sisters, and Randy Crawford. His list of scoring
credits spans dozens of live action and animated
TV shows, films, trailers, commercials, and most
recently several Playstation games. Craig also
arranged and programmed on Rick Brauns lat-
est album Yours Truly.
can pretty much can cover any style of popu-
lar guitar that a standard Les Paul in tradition-
al tuning can dish out.
One thing about the library that I found to
be a limitation is that it doesnt have any
dropped tuning power chords. So much of
todays rock music uses them to get a really
heavy rhythm guitar foundation that this is
my biggest gripe about the library. However,
the power chords in LPC sound great, and
after talking to Prominy I get the sense that
they will continue to support this product.
VI
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This aticle was adapted from a
review that appears on
www.SonicControl.com. It is used
with their kind permission.
For instance, I recently produced and mas-
tered a classical piano recording on which we
never could get a reasonable and convincing
reverberance to add to our deliberately fairly
dry recording. The artist could always tell it
was false and needed for us to get rid of it.
Finally, we went with the existing acoustic
room ambience that had been captured in
the recording, and that was that.
These types of problems have led to an
alternative solution, the so-called convolution
reverb. Convolution reverbs use a different
premise. Any audio or acoustic device or
room has a behavior over time. When we
inject an impulse (an instantaneous spike of
broadband energy) into a device or room,
that device/room transforms that spike over
time.
This transformation is called the devices or
rooms impulse response. In the case of a
room, the impulse response is equivalent to
the reverberance of the room when excited
by an instantaneous spike of energy. Because
its hard to reproduce spikes, in reality its bet-
ter to use a sine wave sweep covering all the
frequencies you want to sample, then time-
compress the recording after the fact to simu-
late the spike. This is done in software that
comes with some convolution processors.
In the digital realm, this impulse response
can be thought of as essentially a specific and
constrained data-set of temporal behaviors. If
we take an audio signal and transform it with
that impulse response, that set of behaviors,
the result will be an audio signal with all of
the temporal qualities of that device or room.
This transformation is called convolution.
There was an explanation of how convolution
works in the first issue of this magazine.
The implications of this are really attractive.
You go into your favorite concert hall, fire off
an impulse and record the result. From that
recording, you derive the impulse response.
Then you convolve your audio recording with
that impulse response and, voila, it is as if
your track was played in that hall, quite liter-
ally!
Does it work? Absolutely! Are there limita-
tions? You bet. Is it better than conventional
reverbs? Well, naturally, it depends.
In an undeveloped convolution reverb pro-
gram, all youve got is the convolution. So
you have basically one listening and source
position in the hall and one hall behavior. If
that is perfect for your music, youre fine
However, it has been our general experi-
ence that we want (and need) to tweak vari-
ous values to get the reverberance we want.
Keep in mind those four micro-time zones I
mentioned earlier. They are all pretty tender
and touchy to get just right. One size hall
definitely doesnt fit all.
Recently, however, I have had a chance to
play with a couple of current convolution
plug-ins (Altiverb V5 and Waves IR-1) and
they have both pretty much taken care of the
THINKING REVERB
(CONTINUED FROM PAGE 38)
VI
f e a t u r e
tweakability concerns I have. The actual con-
volution is embedded in a larger algorithm,
and multiple impulse responses are taken so
that you can sort of morph between them,
within reason. (See the Altiverb 5 review in
our previous issue.)
Which should you choose? This isnt a
product review, and Im not going to hold
forth with an opinion, except to say that I feel
confident that I could work comfortably and
effectively with my Lexicon 960 or Altiverb V5
or with Waves IR-1 for all normal production
work.
Working with reverbs
Your ambience times are probably going to
be between 150 and 800ms, depending on
the music. You might use some pre-delay, but
it will probably be short. If you are like me,
you usually will choose a fair amount of low
frequency emphasis, which seems to make
the overall sound fatter. I usually listen for
body around the sound, a sense of spacious-
ness and envelopment that supports the
sound without being obvious. There should
be no obvious trail to the ambience, and no
sonic energy stepping on subsequent notes.
Working with my Lexicon, I just pick any
Room preset and edit to taste from there.
With a convolution reverb I would select a
reasonably sized so that it can be shortened if
necessary. My sense is that convolution works
better when it is being shortened than when
it is being extended in length (which requires
some sort of looping).
Long tails
If you plan for it, you can splice your
ambience to a longer reverb when desired (you
simply send the signal to both engines). Or
not! Before starting a mix, I tend to set up both
a short and a long reverb engine that will work
independently or together as needed.
When working with sample libraries, you
already have some ambience in the record-
ing, and you dont normally want to dupli-
cate that in a reverb processor.
The primary thing to watch out for with
the longer reverb is to use a sizeable predelay
(80 - 100ms?) to get the reverb out of the
way of the ambience you have already or
thats already in your samples; de-emphasize
any early reflections provided for in your pro-
gram; and then start tweaking for taste with
spectrum, reverb time, and the various other
options you have available.
Spreaders
On page 58 of the first issue of VI, I dis-
cussed spreaders, which are discrete medium
delays (maybe 29 and 33ms) hard-panned
left and right, with the earlier one 3-5dB soft-
er. This is to offset whats known as the prece-
dence effect, which pulls the image over to
the earlier delay. These delays simulate the
sides of the room youre in. You can also use
more than two delays for a richer effect.
I find spreaders to be one of the great
audio bargains, and quite easy to work with.
However, restraint is called for. Dont use
them on percussive material (well, maybe on
the snare drum, but be careful!) and set the
level low enough so that you only hear them
by their absence when you mute them. I find
the spreader works great on the bass track,
vocals, and leads.
Payoff
Where this gets to be a whole lot of fun is
when you automate the send levels and sig-
nal processing on the sends (compression,
typically) and maybe on the reverb return (eq
or compression). Theres no reason that
reverb has to be static.
Convolution reverbs are intended to be
used in-line rather than on a send, i.e. the
whole signal goes through them and you
adjust the wet/dry balance or perhaps the
level of the tail. But the following ideas can
easily be adapted to them as well.
Lets say you have a rapid-fire melody that
Groove quantizing
music that isnt a
groove
Q
uantizingthat is, auto-correcting the timing of performanceshas been an essential
feature of MIDI sequencers since they first came along in the early 1980s. But stan-
dard quantizing, percentage quantizing, humanizing (randomizing), and traditional groove
quantizing operations are all intended for music with a beat. What if youre sequencing a
melodic line and want to make it feel right?
Fig. 1: The original part before quantizing.
It was played sloppily and its timing needs
fixing.
Fig. 2: Here it is after being quantized to
16th notes. You cant tell by looking, but it
doesnt sound right at all; the perfect preci-
sion is too stiff. We could randomize or
percentage quantize the part (so the
notes would get moved less than 100% of
the way to the nearest 16th note), but it
would still sound sort of quantized. Those
operations dont usually sound as natural
as one would expect.
Fig. 3: The answer is to tap in a series of
16th notes while singing the music, and
use them to create a groove quantizing
grid. It sounds fantastic. (We left the 16th
notes in just to demonstrate how the music
gets aligned to them.)
Note that every 16th note in the groove is ahead of the beat, just like in the original part. Thats
because the instrument is a sampled violin, and it speaks slowly. In this example we used that actual
violin sound to create the grid, but you wouldnt want to do that if you were creating a grid that
would work on instruments with sharper attacks.
Now if you need to double the part with another instrument, you can randomize the notes you used
for the first groove to create a groove variation, or just randomize the groove quantized part. That
will separate the two instruments attacks so the combination doesnt sound synthy, but the feel will
be decidedly un-quantized.
random
tip
VI
f e a t u r e
needs ambience but no reverb trail, until a
sustained note that is the high point of the
whole cue. Keep the long reverb send level
down until just after the note is hit, and then
ramp it up until the end of the note to excite
the long reverb, pulling it back down after
that. Ill often do stuff like that on guitar
leads, automating by hand to highlight the
sustained notes while keeping the runs either
dry or ambient.
Heavy limiting on the send bus can also be
very useful. Used with a fairly low threshold
and a really fast release, you can end up with
little reverb while the performer is playing, but
much more when he or she stops. Try a thresh-
old of 35 dBFS, an attack time of .5ms, and a
release time of 2-5ms on the send bus, plus
maybe some makeup gain to compensate for
the compressor lowering the level. This really
will help keep the reverb out of the way, except
when you want it between phrases.
If this isnt enough reverb action to float
your boat, and you are dedicating a reverb to
a single track, you can compress the reverb
return as a ducker, using the track itself as the
key (sidechain) input. This will lower the
reverb level while the track is playing and
raise it when its not.
Remember, a compressor is essentially an
automatic volume-reducer that kicks in when
the signal rises above the set threshold. The
higher the input level, the more gain reduc-
tion.
A compressors sidechain lets you use a
separate signal from the one going through
the compressors main path to trigger com-
pression. So the louder the input signal in the
sidechain gets, the more gain reduction gets
applied to the reverb signal being com-
pressed; the quieter the input, the louder the
reverb.
Again youll want a fairly low threshold,
but quite long attack and release times (100
500ms), so that the reverb wash doesnt get
chunky on you with its pumping up and
down in level. Now you can have lots of
reverb between phrases and reasonably dry
sound during phrases, kind of like a noise gate
in reverse. The news guys do this all the time
to mute mics on people who arent currently
talking.
The object of all this is to keep a grip on
the amount of energy coming back at you at
any moment from each of those four time
zones. Lets take a final look at this.
The direct sound is well known and com-
paratively straightforward to deal with, if not
necessarily easy (we do it all the time). It is
the track or the mix or both.
Spreaders, for those critical first two reflec-
tions, project the dry sound a little forward and
give it a real sense of body and presence. Use it
to illuminate your lead instruments and appro-
priate fills, and also to build a little early front-
to-back depth. Make it so you dont really
notice it until you mute it and wonder, What
in the world just went wrong with my mix?
Early ambience is a wonderful thickener or
fattener for sounds that dont quite work oth-
erwise. It enriches them without spraying
reverberant wash over all the succeeding
events. It is particularly appropriate for all our
synths and sample libraries.
Note that many sample libraries include
early reflection versions of samples. If those
work for you, fine. They are the sample pro-
ducers attempt to address this need for you.
Keep in mind, though, that you can do it for
yourself, and with a little practice you can
bring the quality of that ambience into a very
close approximation to what you really want.
Finally, we get to the longer reverbs that
started all thisthose rich, luxy, sensuous
decays that add such beauty and envelop-
ment to so many sounds. These are an add-
on, a lingering musical perfume if you will.
Use them to great effect by being both spar-
ing and judicious. Dont be afraid to really
hose the mix when the music calls for it, but
do keep in mind that a little can go an awful
long way.
Happy trails. VI
Dave Moulton is a reverb verb junky junky
junky junky who repeats himself a lot lot. You
can complain to him about anything at his web-
site www.moultonlabs.com.
to the FireWire versions. Its too early to have
an opinion about whether youre better off
with USB 2.0 or FireWire, given a choice.
It used to be that USB interfaces were the
inexpensive ones, and FireWire interfaces
were considerably more expensive because
they took more development time, but nowa-
days thats not necessarily true. For example,
the Apogee Mini-DAC is an excellent-sound-
ing USB box that has two outputs only (i.e.
no inputs), and its $1000. Likewise, FireWire
interfaces used to start in the middle price
range, but there are now units in the $200s
(the Presonus inspire 1394) and even below
(Behringer).
PCI vs. FireWire or USB
Latency is the delay between when you
play a note and when it comes out the speak-
ers. Anything under maybe 20 milliseconds
(depending on the players sensibilities) is
unnoticeable to most people. As long as the
audio interface you choose has decent driv-
ersand at this advanced stage, all the main-
stream ones doyou should be able to
achieve excellent latency on any machine that
meets the specs listed on the software box.
The conventional wisdom was always that
since PCI is a faster connection to the com-
puters processor(s), it resulted in lower laten-
cies than you could get with FireWire or USB
2 interfaces. While FireWire does use a small
amount of the computers horsepower to run,
the load is not enough to force you to raise
the RAM buffer and increase the latency; in
practice, todays FireWire and even USB inter-
faces perform just fine.
(Within a narrow range, the processing
power required for streaming audio is on the
opposite side of a scale from the RAM buffer
setting, which increases the latency as it goes
up. In other words, you used to need a top-
FIRST DAW
Digidesigns MBox 2 USB interface has two mic ins,
two analog outs, and two digital I/Os. It comes
with the Pro Tools LE software but can also be used
with standard drivers.
Audio Interface
Computer
Input
Signal(s)
To Speakers
IN OUT
To computer
From computer
Fig. 1: Audio interfaces with a direct monitoring feature split the inputs into two paths. One goes into the
computer for recording and processing, the other goes directly to the output, where its combined with the
V.I. outputs and already-recorded tracks coming from the computer. This bypasses the few milliseconds of
delay when the signal has to travel into and back out of the computer. Signals being direct-monitored must
be muted in the DAW mixer so you dont hear them twice.
VI
f e a t u r e
of-the line computer to set a low buffer and
get low latency. Todays computers are faster,
so thats not really a factor.)
Another adage is that the inside of a com-
puter is a terrible environment for analog
audio, since its so noisy with all those stray
electrons bouncing around. For that reason,
most of the better audio interfaces use break-
out boxes. MOTUs PCI-424 card lets you
attach up to four of their audio interface
boxes, for example, connected by cables sev-
eral feet long.
However, there are cards with excellent
analog noise specs that dont use breakout
boxes. Lynx and RME are two companies that
make cards in that category.
With FireWire or USB interfaces noise is a
non-issue, of course, since theyre all external.
In short, there are no religious reasons to
choose PCI vs. FireWire or USB. The choice
comes down to other factors.
Ins and outs
Audio interfaces come in all shapes, sizes,
and prices. At maybe $10,000 depending on
the configuration, the top of the line is proba-
bly Digidesign Pro Tools HD, which has add-
on DSP cards for running plug-ins and V.I.s
and a choice of high-end interfaces with lots
of I/O. The other end of the spectrum is
around $100 or even less for a very simple 2
x 2 USB interface. M-Audio and Edirol have
models in that category.
In between is a whole world of interfaces.
Weve mentioned several companies in this
article, but some of the others with interfaces
worth checking out include Echo,
Creamware, EWI, Terratec, and Marian. With
apologies, there are surely others that weve
missed.
Strictly speaking, all you need for V.I. use is
a stereo pair of outputs for monitoring, or
possibly more if you work in surround. Youll
overdubbing with mics (as opposed to plug-
ging in instruments directly).
Probably the most important question is
how many of what kinds of I/O you need. We
discussed outputs, but if youre doing MIDI-
plus-live overdubs production, youll need
mic inputs, hopefully with phantom power so
you can use condenser mics. (That is, unless
youre using higher-end external mic pre-
amps, which only need line-level inputs.)
Likewise, if youre recording electric instru-
ments directly, youll need instrument-level
inputs. Line-level signals like hardware synthe-
sizers (dont tell anybody, but we still use
some of those too) requireyes, line-level
inputs.
Different rigs require different I/O configu-
rations. If you have a more advanced set-up
with slave computers running stand-alone
V.I.s, for example, you probably want digital
inputs on your master machine (the one with
your DAW) and digital outputs on your slave
machines.
MOTUs 2408MK III has three 8-channel
optical digital inputs, making it ideal for that;
since it uses the above-mentioned PCI-424
card, you can connect up to four of them if
you need more inputs. Frontier Designs
Wavecenter PCI card is quite inexpensive, and
it has digital I/O and 2 x 2 MIDI onlyno
analogmaking it a good choice for slave
machines. RME also has models with optical
I/O in various sizes.
also need a volume control for your monitors,
which not all interfaces provide.
But chances are that most readers will be
recording some live instruments as well,
which obviously requires inputs. There are
very few audio interfaces with outputs only,
so thats not a problem. Youll also need
somewhere to plug in headphones if youre
MIDI interfaces range in size
from 1 x 1 (Yamaha UX16) to
3 x 3 (Edirol 3ex) to 8 x 8
(M-Audio Midisport 8x8).
Analog mixers with FireWire outputs bridge the gap between live and virtual instruments. Shown here are
the Alesis MultiMix 8 (they make a 12 and 16 as well), and the Mackie FireWire option card for their
Onyx series of mixers.
VI
f e a t u r e
Specs
With some important caveats, digital trans-
fers dont have any sound of their own. But
when it comes to analog recording and play-
back, it should go without saying that not all
audio interfaces sound equal. Most of them
sound quite good these days, but some are
still clearly better than others.
This can be due to any number of reasons,
and the only way to judge an audio interface
is to listen to it in comparison to others.
Unfortunately a magazine article like this cant
really be much help here, but please watch
for upcoming reviews.
CDs use 16-bit audio at a 44.1kHz sam-
pling rate. Virtually all audio interfaces on the
market today offer 24-bit resolution as a pro-
duction format, and many sample libraries are
recorded at 24-bit resolution. What does that
mean?
Without venturing off too deeply into digi-
tal theory, there are two numbers to be aware
of when considering audio interfaces: the
number of bits, which determines the resolu-
tion of an audio file (more bits = more deci-
mal places available to describe the precise
level at any given point); and the sampling
rate, which ostensibly determines the highest
frequency that can be recorded. However, the
goal of extended sampling rates is to make
the audible frequencies sound better, rather
than to reproduce tones above the cutoff of
human hearing.
All audio interfaces can operate 44.1kHz
and 48kHz (which is the video standard).
Some can operate at 96kHz or even 192kHz.
There are no sample libraries we know of that
have been released at those high sample
rates, although several were initially recorded
that way. It takes much more computer pro-
cessing to move twice or four times as much
data around, and the performance goes
down linearly; nobody wants to load a quar-
ter of the sample programs into memory and
live with 25% of the polyphony.
There are inexpensive 192kHz interfaces,
and there are high-end ones that operate at
44.1 and 48kHz. Bottom line, these are
important numbers to be aware of, but not to
obsess over.
Direct monitoring
Please check out Fig 1. Running audio
through a computer and back out takes a
small fraction of a second, but that delay
known as latencycan be distracting when
whole clip slightly to the left (earlier in the
song) so that it lines up with the other track
rhythmically, and youre done. If the clip
starts at the very beginning of the song, youll
have to drag the left end of the clip to the
right before dragging the entire clip back to
the left, because Project5 very sensibly wont
let a clip start before bar 1, beat 1.
If CPU usage is not a problem, you can
achieve the same result by dragging the
source material (the input for the Spectral
Transformer) a bit earlier in the track.
MIDI overdubbing
Lets say youve created a great kick/snare
pattern using one of Project5s drumboxes, and
now you want to add a hi-hat beat to the same
MIDI pattern. There are two ways to record new
MIDI notes into an existing pattern.
First, you can record directly into the
youre monitoring live signals. Thats not to
say it automatically will be problem, but it
can be. Live signals include mic inputs and
everything else, including the output of slave
computers.
Following MOTUs lead, more and more
developers are offering no-latency monitoring
built into their audio interfaces. This is usually
called direct monitoring.
What happens is that the input signal is
split into two paths. One path goes into the
computer, the other goes directly to the inter-
faces output, where its combined with the
signal coming out of the computer. This
bypasses the computer so you can listen to it
with no delay.
You can still process and record the signal
going into the computer, but youd normally
mute its output inside your DAWs onscreen
mixer to avoid monitoring it twice. Then
when its recorded, or when youre ready to
mix, you unmute it in the DAW mixer and
turn off direct monitoring.
No-latency direct monitoring can be a very
useful feature.
Other considerations
Many people like tactile control, and all but
one of TASCAMs audio interfaces feature
integrated control surfaces with faders, some
of them motorized. M-Audio and Digidesign
also have interfaces with motorized faders; M-
PROJECT 5
Would you move over, please? Id like to hide
under the bed with you and those poor audio
hardware developers who have to keep up with
these changes.
Audios model lets you program the faders to
control V.I.s, while Digidesigns 002 is dedi-
cated and optimized for its bundled Pro Tools
LE software.
Maybe you want a portable audio inter-
face, which might look a portable bus-pow-
ered model that doesnt require plugging in.
Metric Halos interfaces are known for their
high quality sound, and MOTUs Traveler is
bus-powered. E-mus 1616M Laptop
DigitalAudio System is another option.
The other important consideration if youre
just getting started is that many interfaces
come bundled with lite software versions
which are usually quite serious. For openers,
all Macs include GarageBand, which is a great
entry into the world of softsynths and sam-
plers.
But the bundles are all different. For exam-
ple, E-mus interfaces come with a huge soft-
ware bundle, including Cakewalk Sonar LE,
Ableton Live Lite, amp simulators from IK
Multimedia, SFX Machine LT, Minnetonkas
Diskwelderoh, and E-mus Proteus X LE
Desktop Sound Module.
Mackies Spike comes with their Traktion 2
software, an interesting new DAW.
Digidesigns interfaces come with Pro Tools or
Pro Tools LE, along with some plug-ins. You
can run a version of Pro Tools LE, called Pro
Tools M-Powered, on M-Audios interfaces.
MOTUs interfaces come with their
Audiodesk Mac software, and a lot of inter-
faces include Steinberg Cubasis. This is not a
comprehensive list of what comes with all
these interfaces, just an indication that theyre
worth more than their hardware.
Gasp.
So many interfaces, so little time.
Sometimes it can be harder to find an inter-
face without features you dont need than
with the ones you do need!
Next issue well look at software. VI
Pattern editor window. While doing this, how-
ever, youll be able to monitor only the cur-
rent pattern, not the whole arrangement
(which might include bass, keys, and what-
not) that go with it in the song. This may
work well in some situations, but other times
you may want to monitor the whole arrange-
ment while recording new bits.
You can do this by recording into the track
rather than into the pattern editor. But when
you overdub new MIDI into a track, youll
always create a new pattern rather than
recording additional notes into the existing
pattern. Project5 is happy to play back several
overlapping MIDI patterns on the same track
at the same timebut copying two overlap-
ping clips at the same time in order to extend
the pattern for 16 or 32 bars tends to get
messy. What if you want to record into the
track (while listening to the arrangement) but
you also want to end up with a single MIDI
pattern containing both the older material
and the new material?
VI
f e a t u r e
There are three ways to do this. First, you
can easily copy the MIDI data from the newly
recorded pattern and paste it into the earlier
pattern, after which the newly recorded pat-
tern can be deleted from the track. There are
limitations in this approach, however. First,
you can only copy one controller data type at
a time, and second, youll have to manually
drag the copied data into the correct rhyth-
mic alignment after pasting it into the new
pattern.
Heres a better approach: simply use the
mouse to drag over the two overlapping pat-
terns in the track display to select them (it
doesnt even matter if one of them is entirely
hidden by the other) and then use the
Combine Selected Clips command in the Edit
menu. This command creates an entirely new
clip containing the data from both of the
other clips.
Theres an even better way, though: right-
click on the Track Controls box for the MIDI
instrument and choose Create Layer from the
pop-up menu. This adds a new sub-track
immediately below the first, assigned to the
same instrument. Now you can add an over-
VI
Figure 6: Just about any parameter in any device
can be automated in Project5. These pop-up sub-
menus show a few of the available choices for the
Dimension synth.
dub and still see both the old pattern and the
new one, because they wont overlap in a sin-
gle track lane.
Handling automation
Project5 lets you assign almost any param-
eter as a destination for automation, so the
musical possibilities are truly vast. Need to
change the filter cutoff in a Dimension patch
when the drums enter? Just click the triangle
to the right of the instrument or effect name
in the Track Inspector, select the
desired destination for one of the
horizontal sliders (see Figure 6), and
record the change.
Eight sliders are provided for
synths, and four for effects, but these
are just control strips. When theyre
reassigned to other parameters, previ-
ously recorded automation data wont be
lost. To view and edit track automation data,
just click on the selector box in the Track
Controls box (see Figure 7) and choose a data
type from the In Use submenu.
Just to make things more interesting,
Project5 lets you record automation data
either into the track or into individual pat-
terns. The advantage to using patterns is that
you can easily copy a four-bar automation
pattern throughout a track.
After choosing the desired parameter in the
dropdown menu in the top center of the pat-
tern editor, you can draw the desired curve in
the editor.
Its probably a good idea to avoid automat-
ing the same parameter in both patterns and
the track. This will force Project5 to work too
hard, and youll probably hear a garbled
sound.
Unfortunately, Project5 doesnt support
copying and pasting from one controller type
to another. If youve recorded a controller
sweep that you like into a track, and you
want to apply it to several parameters at
once, youre stuck. But if the controller data is
in a pattern, re-using controller sweeps is
easy.
Copy the pattern (and delete any notes
that are in it), right-click on the controller
data, and choose Reassign from the pop-up
menu. To layer the controller pattern with
others, go to the track and use the Create
Layer command. Now you can park your new
controller-data-only pattern in a separate
layer, leaving the patterns with notes undis-
turbed.
A mad nPulse
The nPulse drum module is not complex,
but it has a couple of features that make it
more capable than you may realize. First, any
or all of the 12 sound sources can be set to
respond to the same MIDI key. This makes it
easy to create complex layered sounds.
Second, nPulse has five audio outputs. This
lets you apply different effects to various per-
cussion sounds.
Heres how to do it. After selecting output
2, 3, 4, or 5 on the panel for one of the sub-
modules, go to the Track Controls box (see
Figure 8) and select that output as the active
one. Next, switch off Bypass for the output in
the Track Inspector column (see Figure 9).
Now you can add effects in the usual way.
They will only be applied to the currently
active output as selected in the track proper-
ties box.
Progress5
The rapid increase in music power of soft-
ware like Project5 is truly inspiring. If you take
the time to learn your way around the pro-
gram, youll reap rich musical rewards. In this
article weve touched on only a few of
Project5s myriad features. The Groove
Matrix, Device Chains, PSYN, Cyclone, and
other cool tools await discovery, so rock on!
VI
Jim Aikin writes about music technology for
various magazines and websites. Hes the series
editor of the Power Tools music technology
books from Backbeat Books, and the author of
Power Tools for Synthesizer Programming. You
can visit him online at www.musicwords.net.
Figure 7: To display a particular automation lane, choose it in the menu that appears below the Track
Controls box.
Figure 8: After sending an nPulse percussion source to a separate
output, select the output as shown here.
Figure 9: After selecting output 2 in Figure 6, use
this submenu to un-bypass the output so it can be
heard.
Sonic Implants, and Northstar have con-
tributed to the more varied material on the
other DVD. Some of Native Instruments own
sounds are in there as well.
The VSL instruments come in 28 different
groups separated into directories, with each
instrument type having from one to four sub-
groups and each sub group containing one to
nine instruments. Some careful calculations
on my fingers told me that there are 435 VSL
instruments in the library. Every instrument in
a traditional orchestra is represented, so you
can create complete orchestrations.
These VSL instruments include special lega-
to and repetition instruments that make use
of the Kontakt Script Processor (KSP), a
sophisticated MIDI processing feature that
well explain later. Everyone knows what lega-
to instruments are, and repetition instruments
alternate between several samples of the
same note to avoid the infamous machine
gun effect.
Other instruments in the library are in the
general categories of acoustic and electric
pianos, harpsichords, acoustic and electronic
drums, percussion, guitars, basses, synthesiz-
ers, surround banks, and Kontakt Multis that
load several instruments in different slots.
There are also banks of KSP-enabled instru-
ments, which may be a real indicator of
things to come in sample libraries.
There are a staggering 627 instruments in
the library (in addition to the VSL instruments),
some purely sample playback-based, others
based on loops or on the power of the K2
sound engine to mangle sounds in interesting
ways. I found the Post Musical Instruments
Steinway D piano and the Sonic Reality 5.1
Surround Cathedral Organ particularly nice.
The plethora of rock and synthetic instru-
ments in the library should keep the most die-
hard modern music maniac happy and busy
for hours auditioning all of the fine sounding
instruments in this collection. Without a
doubt, there is something for everyone in the
K2 Library.
Effects, filters,
modulators
After browsing sounds, youll probably
want to investigate the large assortment of
effects, filters, and modulators, easily accessi-
ble from drop-down lists or from the left-side
file browser. K2 has master audio outputs and
auxes that can hold four insert slots each, and
the instruments each have three rack spaces
that hold eight insert/send slots each.
All this flexibility means you can really
make some twisted sounds. Adding an effect
is as easy as using the drop-down list in one
of the slots, or dragging and dropping an
effect from the left side browser. Most of the
included effects, filters, and modulators are
KONTAKT REVIEW
fine for standard tasks. If you want to use
higher-end third-party plug-ins, youll need to
use the K2 as a plug-in and insert them in
your host DAWs channel strips; the stand-
alone K2 doesnt host third-party plug-ins.
The most advanced plug-in in K2 is
Convolution, the convolution reverb, which
sounds very good. It has a fairly comprehen-
sive set of impulses accessible from a drop-
down list and can accept outside impulses.
I was able to drag and drop impulses from
the Windows file explorer right into
Convolution. Some of my impulses sounded
far better in K2 than in other third-party con-
volution reverbs.
However, I suggest that Convolution be
shared on the master Output insert slots
whenever possible due to its CPU intensive
nature. Lower-end computers may run out of
breath and produce noise when attempting
to use Convolution. Make sure you have the
latest K2 updates installed, because they really
improve Convolution performance.
Another effect of interest is the Stereo
Modeler. One application would be for peo-
ple who use stereo orchestral instruments,
because even the best libraries suffer from
stereo drift from note to note. Stereo Modeler
allows you to take, say, a violin section instru-
ment, narrow its stereo field down (to say a
Spread of -40.8 for those trying this at home)
to reduce or eliminate stereo drift, pan it to
the left for proper orchestral seating, then
VI
r e v i e w
TRENDS
place the violin section a few rows back from
the conductor position in a spacious room
using a good impulse in Convolution. Nice.
Surround sound has not been overlooked,
and K2 provides a number of features and
plug-ins supporting surround, including audio
routing for 16-channel surround sound.
Kontakt can also play 256 voices per instru-
ment (16 instruments maximum) on 32 audio
channels, depending on the power of the
host computer.
Importing and purging
K2 has the ability to import instruments
from 25 library formats. These include AKAI S-
1000 to S-6000, GigaStudio, Roland S-5x,
EXS24, Mac Aiff, Reaktor Map, SoundFont2,
Reason NN-XT, Acid, and Wav.
Taking a cue from Steinberg HALion, K2
has a Sample Purge function that reduces the
memory requirements and increases perform-
ance. After you play it your arrangement to
show it which samples are being used, it
purges any unused samples from memory.
Mapping and looping
Its a testament to K2 that without consult-
ing a manual, I was able to create an instru-
ment in a matter of minutes using trial and
error monkey logic alone. Spreading out sam-
ples on the virtual keyboard and creating
velocity layers is as simple as mouse clicking
and dragging samples.
While looping is intrinsically more compli-
cated than mapping, I found the Loop Editor
to be fairly straightforward and easy to use.
Finding good loop points can be easy or a
chore depending on the wave, so its nice to
have a simple responsive tool to work with.
Both the Mapping and Loop Editors can be
opened as separate detached windows when
using K2 in stand-alone mode.
The Group Editor is a convenience for cre-
ating instruments. It allows groups of samples
to be processed a particular way under cer-
tain circumstances. For example, a group of
low-velocity-triggered samples could have eq
applied differently from samples triggered at
higher velocities. This is another tool that
invites creativity in instrument and sound
design.
One longstanding complaint about the
Kontakt family was that release samples are
triggered when you lift your fingers from the
keyboard but not when you release the sus-
tain pedal. This has been fixed in Kontakt 2,
which is a big improvement for libraries like
Art Vista Virtual Grand and East West
Quantum Leap Symphonic Choirs.
The Performance Tool has three modes that
are now automatic in Vienna Instruments:
Legato (hold one note while playing the next
one, and a sample of the transition between
the two is inserted); Alternation (successive
notes automatically switch between two sets
of samples of the same note, such as up- and
down-bows); and Repetition, which steps
through a series of samples of the same note
to avoid the infamous every note sounds the
same effect.
Repetition mode is dare one say fussy to
program, and the other modes require that
you load separate versions of the programs to
use them. Vienna Instruments now does all
that automatically, making sure you dont
play the same sample twice in a row, sensing
when youre playing repeated notes and pat-
terns, and when you need a legato transition.
This works automatically even when you play
Playback engines
K2 has no less than four instrument sample
playback engines: Sampler Mode, Time
Machine I and II, Tone Machine, and Beat
Machine. Sampler Mode is the basic playback
engine, Time Machine is for time stretching
and pitch-shifting in real time, Tone Machine is
for imbuing samples with new tonal variations,
and Beat Machine specializes in wave slicing.
Most people will be content with Sample
Mode for playback, but everyone into sculp-
turing new sounds and instruments will be
most interested in the possibilities presented
by the other playback engines.
KSP
The Kontakt Script Processor (KSP) is a
MIDI processor that can affect all kinds of K2
parameters. You can roll up your sleeves and
program your own scriptswhich is not
something youll learn how to do in an after-
noonor the program comes with a healthy
selection of scripts you can insert into one of
the five script slots available for each pro-
gram. It isnt terribly difficult to customize
scripts.
There are script presets to randomize the
pitch, quantize the input, constrain incoming
notes to a specified scale, create MIDI echo,
trigger chords from a single note, play
ascending or descending harp glisses or
arpeggios, adjust the incoming velocityand
on and on. There are many more.
Its also possible to program scripts that do
things like alternating samples, or selecting
different articulations based on incoming per-
formance criteriain other words scripts can
make performances more realistic. As an
example of how far KSP can go, a fellow
named Oliver Frappier has written a script to
trigger overtone samples when you release
piano keys
(http://www.audiolivepro.com/download/ove
rtones_release_piano_script5.1.txt).
Some of the included example KSP instru-
ments are quite impressive.
Conclusions
We havent touched upon K2s granular
synthesis, 19 different filter models, keyboard
shortcuts, multiple Undo/Redo (Kontakt 1 had
no Undo function), and some other functions.
However, by now you should have a good
take on the primary improvements.
Kontakt 2s great strengths are its now-solid
disk-streaming engine, excellent bundled
library, and wealth of sound-manipulating fea-
tures. It can also open all the sample libraries
that come in OEM Kontakt-family players, mak-
ing its extra editing features and convolution
processing engine available to them.
K2 is a formidable upgrade to a sampler
that already performed very well. VI
Doyle W. Donehoo (www.sierra-trails.com) is
a game music composer and was also a soft-
ware engineer for many years. He is a serious
practitioner of sample-based instrument orches-
tration and sound engineering, a sample instru-
ment developer and feature writer. He has lec-
tured at the Game Developer Conference (GDC)
on the subject of virtual orchestration, and is a
beta tester for numerous music-related products.
trills up to a 4
th
(which of course arent really
trills).
But the pice de resistance is the Speed
control, which senses how fast youre playing
and switches to the appropriate articulation.
No more switching between fast and slow
programs. What a great idea.
The bar
Its possible with a little ingenuity to inte-
grate some of VSLs ideas into other samplers,
and well probably see people doing that in
the near future. East West has already
announced that their upcoming player will
have a Speed control when it comes out mid-
year, and it would be surprising if we didnt
see that happening with other samplers too.
Meanwhile, VSL has introduced an innova-
tive new product that answers the most glar-
ing problems with sampling technology. 2006
promises to be an exciting year for us. VI
Some users will see the K2 library
collection as the primary incentive to
buy the instrument.
VI
r e v i e w
E
ver since The New Sampling revolution,
sampling technology has been some-
what ahead of its performance inter-
face. In spite of some great features like
keyswitching (on-the-fly program changes
triggered by notes on an unused area of the
keyboard), taking advantage of all the differ-
ent articulations in a modern sample library
requires quite a bit of programming; its not
really feasible just to perform all the parts.
The dedi-
cated player
that each
section in
Vienna
Symphonic
Librarys
Symphonic
Cube
update
comes in
(see Trends,
page 10)
changes
that. Vienna
Instruments,
as the player
is called,
makes it
possible to
do a lot more playing in real time.
To understand the significance, you have to
understand that the standard way of working
is to set up dozens of tracks in a sequencer,
each pointing at a separate articulation
loaded into a software sampler, and move
notes or fragments of phrases onto the track
with the best articulation. Its easy to have
well over a dozen tracks just for, say, violin.
Vienna Instruments loads all the articula-
tions you need for an instrument in a single
instance of the player. It works on one
sequencer track and one MIDI channel, and it
does a lot of the articulation-switching auto-
matically.
Up and down and
sideways
Vienna Instruments structure is arranged in
two dimensions: Vertical and Horizontal.
You switch between programs on the Vertical
axis with your choice of controller. To use the
example in VSLs videos, you might switch
between staccato, short detach, and long
detach violin programs with the modwheel
up, in the center, and all the way down. But
of course there many other available con-
trollers.
(Note that each of these programs is a
complete muti-velocity, mapped program
the kind youd normally assign to a sequencer
track and MIDI channel.)
You can then set up multiple Vertical units
and move between them on the Horizontal
axis, using keyswitches. The second Vertical
unit in line could be a set of articulations with
variations of another performance technique,
but it doesnt have to be. In VSLs example,
they have three of the same legato programs
set up as the second Vertical unit, so you can
keyswitch horizontally between staccato and
legato, short detach and legato, and long
detach and legato. (The legato program is
only loaded into RAM once.)
In another vertical unit the Vienna
Instruments player can assign fast scale runs
to corresponding keyswitches, so hitting C
brings up all the C modal runs (C Ionian/
Major, D Dorian, E Phrygian, etc.), C# brings
up the C# runs, and so on. Then you use
another keyswitch to toggle between ascend-
ing and descending runs.
These entire Matrixes can then be save,
and multiple Matrixes can be called up with
yet another set of keyswitches.
Real-time control
Previous versions of VSL use a Performance
Tool to intercept incoming MIDI. On the
Windows GigaSampler VSL version the
Performance Tool is a separate program, and
on Mac its is integrated into Logic Audio
Pros EXS24 sampler.
VSLs Vienna
Instruments Player
VSLs new player brings their huge sample library
under real-time controlon one sequencer track and
MIDI channel
VIt r e n d s

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