Virtual instruments v01#01 july aug 2005 by george

JULY/AUGUST 2005 - VOL. 1 NO. 1

“

ts high-end audio quality, flexible routing and processing capabilities, advanced performance features, sophisticated editing tools, powerful search functions, and streamlined user interface make it once again the indisputable gold standard in software samplers.”

Electronic Musician, May 2005

We couldn’t have said it better. For more GigaStudio information, visit www.tascamgiga.com ©2005 TASCAM, a division of TEAC America. All Rights Reserved. All specifications are subject to change without notice. GigaStudio is a trademark of TASCAM/TEAC.

www.tascamgiga.com

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From the

Editor elcome to the premiere issue of Virtual Instruments Magazine. It’s extremely exciting to finally see this project get off the ground, and we hope you get a lot out of it. Our goal is to put out the finest music industry magazine ever. So what is Virtual Instruments all about? If you’re reading this, you probably understand that softwarebased music creation is the hot new frontier. It’s become a huge deal all around the world, with something like 750 virtual instruments on the market. And counting. There’s now a passionate group of musicians scattered all over the planet—myself included—who live and breathe sample libraries and virtual instruments. We have a whole new way of creating music, a whole new musical medium, and a whole new set of creative possibilities. And along with that, a whole new set of musical skills and technical challenges. In short, it’s high time the world of softsynths and samplers had its own magazine to spearhead the revolution. Being way into this, and having been the editor of a music industry magazine (Recording) for 10-1/2 years—and written for others such as Mix for several more—I felt like the one to do that. So I’ve assembled a team of the best, most knowledgeable writers in our industry, along with a very capable staff to put out the magazine you’re holding in your hands. Our mission is simple: to help you get the absolute most out of your softwarebased musical instrument set-ups, whether you’re first thinking about taking the plunge or you’re an experienced pro running huge orchestral libraries on multiple computers. What you see here is just one month’s worth of that; it’ll take a few issues to expose our full repertoire. For example, you’ll certainly see reviews of effects plug-ins and V.I.-related hardware such as sound cards and controllers. Naturally, the article mix will always depend on what’s new and exciting.

t this point I’d like to thank all the advertisers who took the leap of faith to support us without even having seen the first issue. Please rush out and buy their wares as a reward!

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I’d also like to thank the fine writers who contributed to this issue. It’s an absolute pleasure working with Jim Aikin, whose articles are even greater now than they were when I first read them in Keyboard magazine back in the mid-’80s. Chris Meyer is one of the brightest, most insightful, and colorful writers and people around. My friend Dave Moulton is the audio gurus’ guru. Dave Govett is The Guy when it comes to GigaStudio and many other things, the always articulate Bruce Richardson writes from a wealth of real-world experience, and Ashif “King Idiot” Hakik is a walking clever idea factory with body piercings. General expert Lee Sherman and dashing Frederick Russ (who runs www.VI-Control.net, an excellent discussion forum) are doing a great job for us, and we’re very pleased to have them. And three cheers to our lovely and talented core staff, who have worked their tails off to make this magazine happen: art director Lachlan Westfall, web designer Denise Young, and ad/marketing manager Carl Marinoff. Thanks everyone.

lease email to let us know what you think, what you’d like to see, technical questions, tips, “I wish they’d make…” feature and product ideas…you get the idea. NB@VirtualInstrumentsMag.com. With your help we’ll have a lively Letters section next issue. And for heaven’s sake, make an honest man or woman out of yourself and subscribe! Not only is it much less expensive than the newsstand, you’ll be helping us print more pages and more frequent issues. To encourage that, we have the Mungo Giveaway promotion you’ll find at the center of this magazine. Finally, don’t forget the website: www.VirtualInstrumentsMag.com. Among other things, you won’t want to miss the audio and graphic files that go with some of the articles. Happy reading.—NB

6 July/August 2005

The World of Softsynths and Samplers by Nick Batzdorf

EDITOR/PUBLISHER: ART DIRECTOR:

Nick Batzdorf

A big overview of this exciting new musical medium. Getting started—what’s available, the various product categories, what to expect from different computers, and generally how to integrate software-based musical instruments into your universe.

Lachlan Westfall/Quiet Earth Design

AD/MARKETING MANAGER: WEB DESIGNER:

Carl Marinoff

Denise Young/DMY Studios

Jim Aikin, David Govett, Ashif “King Idiot” Hakik, Chris Meyer, Dave Moulton, Bruce Richardson, Frederick Russ, Lee Sherman.

CONTRIBUTORS:

PUBLISHING CONSULTANT:

Ross Garnick

ADVERTISING CONTACT: Carl Marinoff 818/590-0018. CM@VirtualInstrumentsMag.com

The Age of Reason by Jim Aikin

A Very Deep Clinic on Propellerhead’s popular stand-alone music station. Create stereo and filtered delays, become a monster Maelstrom programmer, ReWire tricks, the pleasures of polyrhythms, and lots more.

SUBSCRIPTIONS/ADDRESS CHANGES: 818/905-5434, 1-800/ViMagzn, subscribe@VirtualInstrumentsMag.com. The best method is to subscribe via our website: www.VirtualInstrumentsMag.com. LETTERS TO THE EDITOR:

NB@VirtualInstrumentsMag.com

query NB@VirtualInstrumentsMag.com or call 818/905-5434

WRITING FOR VIRTUAL INSTRUMENTS MAGAZINE:

Virtual Instruments Magazine is published bi-monthly by Virtual Instruments, Inc., 3849 Ventura Canyon, Sherman Oaks, CA 91423. 818/905-5434, 1-800/ViMagzn. NB@VirtualInstrumentsMag.com.

44 Rigors of the Road Rig by Bruce Richardson

A from-the-trenches report on putting together a VI rig for live use—everything from what power supply to use in the computer to racking it up.

DISTRIBUTOR: Rider Circulation Services, 3700 Eagle Rock Blvd., Los Angeles, CA 90065. 323/344-1200. Bipad: 05792, UPC: 0 744 70 05792 5 07

Standard disclaimer: Virtual Instruments Magazine and its staff can’t be held legally responsible for the magazine’s contents or guarantee the return of articles and graphics submitted. Reasonable care is taken to ensure accuracy. All trademarks belong to their owners. Everything in here is subject to international copyright protection, and you may not copy or imitate anything without permission. © 2005 Virtual Instruments, Inc.

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38 Interview: Film Composer Klaus Badelt’s Shrinking Studio The composer of “Constantine” and over 25 other major Hollywood films (“Pirates of the Carribean,” “Catwoman,” “The Recruit”…) discusses his evolution from a wall-length rack full of gear to a single Apple G5 running Logic Pro.

contents VI

PREMIERE ISSUE

with King 24 Sampling Idiot. Part 1: Modern

V1.N1

Live and 56 Combining Virtual Instruments

features for old libraries

by Dave Moulton

How to create the right space so that instruments recorded in completely different environments will mesh seamlessly with your VIs and sample libraries.

by Ashif “King Idiot” Hakik

Older sample libraries and keyboard instruments can be programmed so they work alongside your new libraries. One of the world’s most sought-after sample library programmers details how to create legato programs, wet and dry samples, and release triggers.

Trends Synful’s 64 Reconstructive Phrase Modeling by Frederick Russ

3 Finesse: 50 GigaStudio Taking the GigaPulse

This new additive synthesis virtual instrument analyzes the incoming MIDI data and uses a database of phrases to decide what note fragments to splice together. Could this technology dethrone sampling?

by David Govett

The man who wrote the manual goes beyond it to examine the power of the GigaPulse convolution processor in Giga 3. Another Very Deep Clinic.

22, 28, 37

Routing with the MOTU PCI-424 card, altering VSL string runs in Melodyne, and using the mouse wheel in Logic Pro 7.

reviews VI 10

Apple GarageBand and Jam Pack 4: Orchestra

Ableton Operator add-on FM synth for Live

Zero-G Beats Working in Cuba

by Nick Batzdorf

by Lee Sherman

by Chris Meyer

An overview of this important entrylevel program, along with the latest add-on content package. Yes, there really is a $99 orchestral library—with sounds that truly belie the price.

Up until now, nobody has figured out how to put a friendly face on FM programming. This synth is a lot easier than the DX-7.

This is more than a loop library—this virtual instrument documents a musical genre. 34

20 18

Random Tips

East West Quantum Leap Colossus virtual instrument by Nick Batzdorf

A self-contained, high quality desert island library that covers all the bases and then some.

Cycling ’74 Cycles by Chris Meyer

Some unique loops to freshen the pallet. These rank high on the don’t-leave-onyour-shelf list.

Spectrasonics Stylus RMX and SAGE Expanders 1-5 by Nick Batzdorf

Exceptional grooves in a brilliantlyexecuted virtual instrument bring working with loops to a whole new level.

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f e a t u r e VI

World of Softsynths and Samplers The

music technology revolution.

The first one came in with the information era. In 1983, when synthesizers started to come under microprocessor

By Nick Batzdorf

control, the industry adopted MIDI as the common control language. Yamaha introduced the famous DX-7 synthesizer, and it brought digital synthesis into the mainstream. MIDI sequencers changed the way music was composed, and their ability to fix every mistake and play with perfect timing changed the way we hear music. Drum machines got some musicians thinking in patterns, which for better and worse laid the groundwork for looporiented music. The synth craze was on. Samplers—which are technically not synthesizers, since they use digital recordings as the basis for their sounds—started becoming affordable a couple of years later. The music software and soundware industries got going in full force. These were the high-flying keyboard du jour days, with each model leapfrogging the previous one. That era lasted a few years. A lot of incredible digital instruments came out, many of which are still in use today. These instruments have microprocessors, memory, storage, operating systems, software,

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they waddle and quack…all the elements of a personal computer. And there’s no ducking the fact that they are computers. But the hardware is proprietary, which means that the development was done by companies with the funds it takes to manufacture hardware. Meanwhile, standard computers were constantly getting more and more powerful. Digital audio workstations (DAWs) began taking over as a recording medium, first with the help of add-on hardware acceleration (Digidesign Pro Tools), later with or without it. By the mid-’90s, a cottage industry had materialized to take advantage of the DAW migration, writing DSP plug-ins—equalizers, compressors, delays, and so on. Then at some point before the turn of the century, standard personal computers became powerful enough to run synthesizers and samplers in software. Five years later, they have so

f e a t u r e VI much power that software-based musical instruments—Virtual Instruments—are often more capable than dedicated hardware ones. It still takes a lot of skill to develop an instrument, but it doesn’t take the huge investment required to build hardware, nor does it take the manpower and time to code an operating system from scratch. We’re seeing a lot of innovation as a result, only now it’s not only coming from large companies—some of which didn’t start out that way—but also from the proverbial 1-person shop in some small Eastern European mountain village you’ve never heard of. The pioneering spirit of the early ’80s is back. This is the second great wave in digital music technology. It’s more gentle than the first one, but it’s having a profound effect on the way music is created at all levels—and on the music itself. The virtual instrument phenomenon has resulted in some very interesting instruments, and more are coming out all the time. But on a practical level, probably the biggest difference between working with hardware synthesizers 20 years ago and software synthesizers now is that we have so many of them at our immediate disposal. Sampling, on the other hand, has undergone a major revolution.

Along came Giga The basic concept of using samples to imitate an instrument—as opposed to doing weird stuff—is pretty simple: record an instrument playing a note, and then assign that to the corresponding note on a keyboard (or other MIDI controller). For added detail, record the instrument playing at different levels, and assign them to that same note to be triggered at appropriate velocities (i.e. the harder you play, the louder the sample you trigger). There were excellent samples available in the early ’80s, but they were limited by the amount of memory that was available, which translates to recording time. Samples had to be loaded into RAM for playback, so it wasn’t possible to sample, say, every piano note at 16

Taking the plunge is surprisingly easy: all you need is a reasonably late model computer, a controller keyboard, and a sound card for high-quality monitoring.

sampler called GigaSampler. It completely blew the lid off the pot. (GigaSampler is now called GigaStudio and NemeSys is owned by TASCAM.) Instead of loading samples into RAM, Giga streams them off hard drives. It still needs to load a little bit into a head-start RAM buffer, but the recording time is essentially unlimited. All of a sudden it became possible to sample as many variations of an instrument as necessary to capture its expressive range. So instead of having a choice between, say, short and

It still takes a lot of skill to develop a software instrument, but it doesn't take the huge investment required to build hardware. The pioneering spirit of the early ’80s is back. velocity levels with their full decay to nothing (which can take over a minute). To save recording time, the compromises were to record a sample every few notes and pitch-shift it, restrict the number of velocity layers, and find a clear section of the wave form to loop repeatedly. While the amount of available RAM grew from kilobytes to Megabytes, this was a limitation of both hardware and early software samplers. Then out of nowhere a company called NemeSys came along with a Windows software

long string ensemble programs, we now have programs with literally dozens of individual violin, viola, etc. playing techniques—known as articulations. Each one is sampled at several different velocity layers, and the programming details go on from there. Giga injected new life into the field of sampling, and it attracted a lot of excellent developers. Sampled sounds have become so much better that it’s almost like a whole new technology. Giga started a worldwide craze. “MIDI programming”—that is, piecing together phrases

using different articulations, rather than just playing parts in real time (using one-size-fits-all programs)—has become an important skill of its own, and not just for orchestral libraries. A whole new musical medium was born. The process of composition has changed forever. Sample libraries used to be measured in Megabytes and come on a CD. Now they’re measured in Gigabytes and come on multiple DVDs. While there are other excellent orchestral sample libraries, the pinnacle of the whole sampling field in many ways is the Vienna Symphony Orchestra, which is currently up to 230GB. These people have sampled every instrument of the orchestra in impeccable detail, including the transitions between every two notes up to an octave apart; what they’ve done is just incredible. Today there are several other software samplers that feature disk-streaming, including the EXS24 sampler built into Apple Logic Pro, Steinberg Halion, Native Instruments Kontakt, and MOTU MachFive. You’ll also find a lot of libraries—including some reviewed in this issue—sold as “virtual instruments,” meaning that they come attached to an OEM “player” version of one of these samplers. The advantage is that you don’t need the sampler to run the library; the disadvantage is that the ability to tweak is generally limited. We refer to large disk-streaming libraries as modern sample libraries.

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f e a t u r e VI The entry level: hardware The list of hardware you need to get started isn’t very long: a suitable computer with a basic sound card aka audio interface if you want high quality monitoring, a MIDI keyboard (or other controller), and a basic MIDI interface if one isn’t built into the keyboard or sound card. As to computers, any PC or Mac released in the past five years—maybe even earlier—is capable of doing something useful with V.I.s, but the more power and memory (up to the maximum your system can address) you have the better. Depending on the instruments and plug-ins you’re running, a large-scale arrangement can bring any machine to its knees; on the other hand, it’s remarkable how much you can get out of a single machine. Software instruments always spec minimum system requirements on their boxes, but in general you’ll constantly be running into the minimum machine’s limits. It’s a good policy to contact the developer of the software you plan on using to find out how much you can expect to eke out of a given computer. You should be able to get a feel for this by reading some of the reviews in this issue, for example the East West Quantum Leap Colossus review on page 18. For playing V.I.s, one of the important specs is latency: the delay between playing a note on your keyboard and hearing it come out your speakers. Up to a point, performance and the size of the RAM buffer every sound card requires are on opposite sides of a scale. The more powerful your computer is, the lower a buffer you’ll be able to set without hearing clicks and pops as you start pushing the number of voices (simultaneous notes). Sound cards come in all shapes and sizes, starting at well under $100 and continuing upwards of $12,000 for a Digidesign Pro Tools HD system. While it’s not ideal, you can even monitor V.I.s through a computer’s built-in audio card (hopefully through decent head-

At 230GB, the Vienna Symphonic Library represents the pinnacle of sampling. This screen capture from Tascam GigaStudio 3 shows how extensive it is.

Some recent keyboard controllers connect to a USB port on your computer. Others have a MIDI Out port, in which case you’ll need a basic MIDI interface to give your computer a MIDI In. These boxes aren’t expensive.

Taking the plunge: software Most virtual instruments are plug-ins, which by definition means that they’re add-ons to a host application. These hosts are usually digital audio sequencers, which are programs that

“MIDI programming”—that is, piecing together phrases using different articulations—has become an important skill of its own, and not just for orchestral libraries. A whole new musical medium was born. phones or a home stereo system for better sound); Mac G5s have optical digital outputs built in as well as their analog ones, and they’re fine if you have a way of converting them to analog (again, such as with the digital inputs on a home stereo). Any MIDI keyboard will work as a controller, whether it’s a workstation with built-in sounds of its own or a dedicated controller keyboard. Bear in mind that keyboards with weighted keys that feel like a piano are good for keyboard parts, but they can be hard to control if you’re playing something like woodwinds, and vice versa.

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record, play, edit, mix, etc. live audio and MIDI side-by-side. The terms “digital audio sequencer” and “digital audio workstation” or more commonly “DAW” are interchangeable. The most widely used digital audio sequencers are Apple Logic, MOTU Digital Performer, Digidesign Pro Tools, Steinberg Cubase, and Cakewalk Sonar; you’ll see all of them mentioned a lot in this magazine. However, there are many other kinds of programs that can host V.I.s., including 2-track audio editors; all-in-one environments such as Ableton Live, Propellerhead Reason, Arturio

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Storm, FL Studio, Sony ACID; stand-alone hosts that do nothing else such as Steinberg V-Stack; notation programs with links to V.I.s such as Sibelius…and many more we’re leaving out. Most people are best off with a digital audio sequencer, but for example Ableton Live is a great all-in-one program if you’re into loop composition and performance. Propellerhead Reason (see clinic in this issue) is a great all-inone groove-based composition environment with some really interesting synths and effects, and it doesn’t require all that hefty a computer. Most sound cards and many other programs these days come with very generous “lite” version software bundles, and that’s a great way to get your feet wet before investing heavily in software. So is Apple GarageBand, which comes with every Mac and is very inexpensive as part of iLife (see the following article). Garritan Personal Orchestra, all the TASCAM audio interfaces, Digidesign’s MBox…these all include serious software bundles. A lot of musicians use a variety of programs to take advantage of what each has to offer. ReWire, a protocol developed by Propellerhead, allows two programs running on the same computer to be in complete lock-step. You set up a ReWire master and slave, and MIDI, audio, sync, and transport control get handled transparently. If you work in Logic and want to treat Reason as an external sound generator rather than a complete environment, for example, you can just ReWire it into Logic’s mixer. Without wanting to be grossly obsequious to our advertisers, the best way to see what’s out there is to go to a major retailer’s website like

f e a t u r e VI www.Audiomidi.com or www.Sweetwater.com and poke around their software sections. Then go to developers’ websites and investigate their demos.

Formats There’s a variety of formats to be aware of, among them formats for software to communicate with audio hardware, plug-in formats, and sampler formats. While they don’t make for a very perky article and we won’t list them here, it’s important to understand the distinction. It’s the plug-in format that we’re listing at the top of each review; that tells you which hosts are compatible with the plug-in being reviewed. One machine Those are the three factors that determine how much you can squeeze out of a computer: processing power, hard disk access, and memory addressing. Computer processing power is the most important resource for synthesizers and DSP plug-ins—something we haven’t really discussed in this article, but nonetheless an essential part of software-based music creation. With certain notable exceptions, however, you probably won’t constantly rub up against the processing limits of a higher-end computer, such as a 2.8GHz Pentium 4 or a Mac G5, if you’re just running synths. We want to be careful not to rule out the possibility, however.

Software samplers don’t use up a lot of processing horsepower just to stream voices, but sometimes they have DSP built in that does require CPU muscle. The latest trend is for samplers to have convolution processors built in, and that really uses processing power. (See the sidebars to the Giga clinic in this issue for more about convolution processing.) The speed at which a computer’s hard disk subsystem is able to deliver audio to the sampler determines how many voices—that is, simultaneous notes—you can play. Today’s machines are delivering in excess of 256 stereo voices off standard drives. If you’re playing sustained keyboard arpeggios, you can quickly run out of voices; single bass drum hits don’t use up any polyphony worth talking about. What is a concern for samplers is RAM, even though they’re streaming the bulk of the recordings off a hard drive. The reason is the head-start buffer we discussed earlier—it still takes memory to load programs; the more samples in a program, the larger the buffer. If you’re running modern sample libraries (again, modern means they’re large and and they stream off disks), you need to try different articulations for notes or groups of notes. You don’t want to have to stop to unload and load programs in the throes of sequencing; that would be like putting a violin in its case and picking up a different one every time you wanted to switch bow directions.

Fig 3: A basic multi-computer set-up. The mixer is optional, and you can share the keyboard and monitor with a KVM (keyboard/video/mouse) switch.

So even if you don’t play all the articulations you have loaded—and you won’t—the name of the game is to have as many programs cued up and ready to play as possible. That means the sampler must be able to access as much RAM as possible. Both GigaStudio and Kontakt can load approximately 1 - 1.2GB of samples in a Windows XP machine with 2GB of RAM installed; installing more RAM doesn’t change this. That includes whatever RAM a host program uses if you’re running these samplers inside one (many people run Gigas on separate machines with nothing else going). It’s not possible to say how many programs that translates to in the real world, because it depends on the program, but 32 - 40 is a reasonable minimum. That figure could easily double if all you’re loading is single percussion hits. A G5 Mac with 4.5 to 5GB of RAM installed, running OS X Panther or Tiger, is equivalent to about 2-1/2 Windows XP machines; a G4 with 1.5 or 2GB installed in Panther or Tiger is roughly equivalent to a single XP machine.

Multiple machines and beyond In order to load big templates, a lot of musicians who use larger libraries—orchestral or otherwise—run multiple computers. They do that in part for the additional processing power and voice count, but mainly for the memory access. If you have an older machine available, you can certainly put it to use just by adding a sound card and a MIDI interface. Don’t get scared, though. You can run a heck of a lot on a single machine. If you look at the diagram showing a basic 2-computer set-up, you’ll see that there are lots of cables to run. The latest technology that’s emerging is to run audio and/or MIDI over an ethernet network cable. That reduces the number of cables and is generally convenient, plus it means you don’t need a sound card and MIDI interface on every machine. Some of the networking products to investigate include MIDI Over LAN (www.Musiclab.com); Apple’s Network MIDI (in OS X Tiger); FX Teleport (www.FXMax.com); Yamaha’s mLan; and Steinberg’s VST System Link. Logic Audio Pro version 7 has a feature called Logic Nodes that offloads Logic-format DSP plug-ins onto another Mac on the network. Version 7.1 also has a tantalizingly undocumented plug-in called AUNetSend that sends audio over a network. Bzz The latest computer industry buzzword is “64-bit.” What’s important to you and me is 64-bit memory addressing, which means virtually unlimited memory access for loading sample starts. Maybe that will bring us closer to the elusive goal of being able to run everything we want on a single machine. And everything we’ve talked about is what this magazine is all about.

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Apple GarageBand and Jam Pack 4: Symphony Orchestra An overview of this important entry-level program, along with the latest add-on content package. Yes, there really is a $99 orchestral library—with sounds that truly belie the price.

t’s hard to imagine an easier entry into the world of softsynths and samplers than Apple GarageBand. Even without any of the add-on Jam Packs we’re starting to review in this issue, this deceptively simple loop-oriented digital audio sequencer comes packed with a hulking selection of everything you need to get started: software instruments, loops, and effects. So we’ll start with a basic run-down of GarageBand for those who haven’t tried it.

Review by Nick Batzdorf

Price: GarageBand, $79; Jam Pack 4, $99

Company: Apple Computer, 1 Infinite Loop, Cupertino, CA 95014. 408/996-1010. www.Apple.com

The first one’s free Apple figures about 50% of all American households have someone who plays an instrument, and they’d like every one of these people to be making music with Macs. So GarageBand 2 comes free on every new Mac as part of their pot-sweetening iLife ’05 collection, along with iDVD, iTunes, iMovie HD, and iPhoto. Or you just buy the entire iLife package (GarageBand included) for $79. Part of Apple’s imperial design is for you to

start having way too much fun with GarageBand, prompting you to upgrade to Logic Express or preferably Logic Pro, their more advanced digital audio sequencers. It’s no coincidence that you can use GarageBand’s Apple Loops and software instruments in these programs, and also in Soundtrack Pro—which in my opinion makes it a no-brainer for every Logic user to get GarageBand just for its content. (Actually, you can use the included audio versions of Apple Loops in any audio program—they’re just AIFF files.)

Toys R’nt us Because non-musicians can stack up loops to use GarageBand as the musical equivalent of painting by the numbers, it might seem like a toy. It’s not. While the program does lack some basic features, most notably the ability to change meters and tempos during the song, it’s a surprisingly capable tool. GarageBand has two types of tracks: Real Instrument tracks (meaning audio, for placing audio loops or recording live input) and Software Instrument tracks, which contain MIDI loops and recordings to trigger sample playback instruments. Rather than inserting the Software Instruments on a track—which is how you use them in Logic—they define the track itself in GarageBand. (GarageBand can also host AU format plugins and virtual instruments, and you can stream audio into it using the ReWire protocol—see “The World of Softsynths and Samplers” elsewhere in this issue.) The interface couldn’t be more friendly. If you drag a MIDI or audio loop from GarageBand’s loop browser, the program automatically creates an appropriate track to play the loop and matches it to the tempo. There’s also a preference to “convert” MIDI loops to audio as soon as you drag them onto the playing field. (There’s actually no conversion—the audio version is substituted for the MIDI version with effects.)

Platform: Mac G3 600MHz+ (G4 or G5 strongly recommended for software instruments); OS 10.3.4+; Jam Pack instruments work with Logic, Logic Express, Soundtrack/Pro, and GarageBand; loops can be used with any program.

License: 1 user, 1 machine, royaltyfree use in audio and video projects. Fig. 1: GarageBand’s loop browser makes it easy to find your way through its extensive library of riffs.

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Fig. 2: Clicking on the scissors brings up an edit window showing the track selection above. In this case an audio track is selected.

What’s really handy is that you audition the instruments with your MIDI keyboard right in the Open dialog. This would be a brilliant feature for Logic’s integral EXS24 sampler, which incidentally can open GarageBand Software Instrument programs for more extensive editing than the player provides. GarageBand has both piano roll and notation-based editing for MIDI regions. You can quantize notes, edit five common controllers, and there are sliders for transposing and adjusting the velocities of selected notes. For audio, there are rather innocuous-looking sliders that, if you think about it, are pretty amazing. One slider transposes the whole region over an octave range in either direction, and it doesn’t sound bad at all. The other two are Enhance Tuning and Enhance Timing. This absurdly inexpensive program can actually autocorrect intonation and quantize audio to a specified grid. And it sounds pretty good, as long as you don’t push it beyond reason. There’s also a guitar tuner, along with various standard effects like eq and reverb, and a guitar amp simulator. I have just two complaints about GarageBand. First, it isn’t able to route its audio to specific outputs on MOTU audio hardware— unlike every other program in OS X. Instead, the sound comes out the first two outputs of the first interface connected to the PCI-424 card, which is absolutely no help if those don’t happen to be the outputs you’re using to monitor. (Fixing this could be MOTU’s or Apple’s responsibility, but that makes no difference to the user.)

Second, the GarageBand and Jam Pack installers will only put the content on your startup disk. Real men and women put applications on startup partitions and content on separate drives. Aliases to content moved to another drive will not work. (I actually had to set up a Firewire drive to start up from in order to do this review, since my startup drive had nowhere near enough free space.) In spite of that, it’s almost funny how much GarageBand has to offer for the price. It’s very few features away from being able to cover all the bases you’d expect from a professional composition tool.

Jam Pack basics There are presently four Jam Packs for GarageBand, each of which runs $99. When you consider that each of these libraries is from 3 to 10GB and contains from about 30 to over 100 software instruments, along with (I’m trusting the boxes here) over 2000 loops…well, that’s a silly price. It appears that Apple licensed some if not all of the content from other developers. In general, the Jam Pack software instruments are somewhat “lighter” than you’d find in a typical modern sample library. They tend to include fewer articulations, or a given program might have a couple of velocity layers instead of three or four. But that’s not a dis, because the flip side is that this makes these libraries very manageable. I had no problem getting useful results out of GarageBand on a G4/500-upgraded PowerBook that’s approaching five years of age—which is about 100 in computer years. (However, the bulk of this review was done on a dual 2.5GHz G5 and a dual 1GHz G4; see the box with this article for system requirements.) Another advantage to these libraries’ relatively compact nature is that all the instruments are designed to be played in real time. You’ll only be doing minimal articulation-switching in the Orchestral Jam Pack, and pretty much none in the others. And the instruments load quickly, since they’re not huge. Apple struck the right balance between complexity and practicality, in my opinion. The Jam Packs still have considerable subtlety, for example some of the programs have release samples, and overall they’re at least as deep as the factory programs on any keyboard workstation from before the modern sampling era. (Again, please see “The World of Softsynths and Samplers” earlier in this issue for an explanation of how sampling has undergone a major revolution over the past five years.) Jam Pack 4 – Symphony Orchestra

Software instruments How comprehensive can a $100 orchestral library be? You’d be surprised. We’ll get to the

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r e v i e w VI loops later, but Jam Pack 4’s software instruments cover every instrument of the orchestra, and the sounds are totally usable—considerably better overall than you’d get out of a keyboard module. Of course the variety of articulations is limited compared to the next tier of orchestral libraries, but there’s plenty here for a skilled programmer to produce convincing orchestrations. Most of the software instrument programs in this Pack use the pitch wheel to control expression (actually volume). You can also use MIDI cc7, but most people like to reserve that for setting an overall “trim” level and then use cc11, expression, for “riding” the level. It’s more common to use a slider for expression, since the pitch wheel springs back to its center when you let it go, but it actually works very well. This library uses the mod wheel to switch between as many as five different articulations. For example, violin Section 1—a very nice sampled string section—has medium sustained notes, staccato, sus tremolo, 1/2 step trills, whole step trills, and pizzicato all up at once. It’s pretty hard move the mod wheel to the right position when there are five articulations up, but three works great—as in the French horn section, which switches between long notes, short notes, and short crescendos. But it’s not necessary to spend hours practicing mod wheel switches—these articulations are available separately, labeled Xtra. If you own Logic Pro, you can build your own program switches in the EXS24 editor. It’s also worth mentioning that Apple chose not to limit the ranges of the lowest and highest samples in these software instruments. You can play string bass way up the keyboard or glock all the way down, for example. As a gauge of Symphony Orchestra’s depth, there’s a total of about 38 different string articulations, divided among seven programs. Subjectively, the strings are way better than they deserve to be in a $99 library. In addition to harp, and violins 1 and 2 ( a smaller section), viola, cello, and bass sections, there’s a String Ensemble section. You can use the ensemble to play the parts before splitting the MIDI notes to individual section tracks, although it’s not very easy to control where notes get pasted in GarageBand.

There are also eleven wonderful church organs, a nice celeste, a decent if somewhat brittle Steinway piano, a harpsichord recorded too closely that has excellent release samples, a keyboardful of good orchestral percussion, crotales, tubular bells, three different timpani articulations (again, available under the mod wheel and separately), glock, marimiba, and xylophone. As with the strings, there’s a brass ensemble as well as individual sections. Both solo and section French Horns are included, the tuba is solo, and the trombones and trumpets are sections. These are not cheesy-synthy trumpets—I was surprised at how good the 2-layer trumpet section sounds. The long horn and trombone programs both suffer somewhat from swelling articulations that get “sucked” back into the track when you play a new note, but you can control that somewhat with the expression (pitch) wheel. Jam Pack 4’s woodwinds include solo bassoon, English horn, oboe, and piccolo, and both solo and section flute and clarinet. All but the bassoon, which I’d subjectively rate only

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Orchestral loops The first thing that struck me about the loops included with this Jam Pack is that they’re not the clichés I was expecting. These are all short riffs of varying length (1 – 16 bars) that you can build upon, with descriptive names like “Adversary,” “Escape,” “Cinematic Transition.” With all the audio loops in this collection, you get the entire mix and then “stems” of individual parts (broken into high strings, medium strings, etc.); the MIDI loops naturally have each individual part separated. Full versions of the demo songs that the loops were extracted from are included on the DVD so you can hear the entire arrangement. Most of these riffs loop seamlessly, a few don’t—and probably aren’t intended to. The Symphony Orchestra is not a “construction kit” like the other Jam Packs; its loops, which almost certainly come from different com-

This deceptively simple loop-oriented digital audio sequencer comes with a hulking selection of everything you need to get started: software instruments, loops, and effects. average, sound very nice indeed—as long as you’re playing the right sorts of lines for the articulations included. The woodwinds are where I really noticed the limited number of articulations in this library. For example, you can play gorgeous legato phrases with the English horn, but the staccato articulation is too bright to blend with the legato one in the middle of a phrase. The same applies to the solo flute and clarinet, plus they also have ”sucking” legato articulations. This is less of a problem with the section clarinet, but it’s still present. To be clear, there’s definitely a place for swelling articulations; the problem is when you use them for successive notes that aren’t long

Fig. 3: Notation editing for MIDI tracks is a new feature in GarageBand 2. It does a pretty good job of interpreting the rhythms that were played.

and sustained. And as long as you “write to the library,” these woodwind programs sound really nice.

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posers, aren’t all designed to work together even if they’re put in the same key. On the other hand there are always going to be some interesting accidents whenever you layer two pieces of music. This is an old music editor’s trick. In general, I found the loop content interesting and worth listening to for ideas. Some people will undoubtedly score videos with them. But this part of the library is all so subjective that it’s best left to your ears to decide what you like and don’t like about this collection.

Conclusion The Symphony Orchestra Jam Pack is one of the best deals around, even if it is missing a couple of important woodwind and brass articulations. Anyone looking to get his/her feet wet with orchestral writing would find it an ideal starting point. Just the string programs, which don’t sound like cheesy synth strings, are worth the price. Musicians who already have other libraries could find this Pack useful as part of a laptopfriendly portable writing rig; just add something like a 2-octave M-Audio Oxygen8 MIDI keyboard and you’re good to go. Most importantly, though, GarageBand has the potential to bring lots of musicians into our fold, and that can only be a good thing for us all. Next month: Jam Packs 1, 2, and 3.

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The Age of Reason Secret Techniques of Power Users Exposed! by Jim Aikin

he so-called “virtual workstation” has carved a fat niche for

itself in the music world. Being able to craft a song from start to finish in one piece of software without ever having to wrestle with compatibility conflict is an attractive proposition. Propellerhead Reason wasn’t the first virtual workstation program, nor is it the most feature-laden. Nevertheless, Reason has become the big gorilla in this particular circus. It sounds great, of course, but the real explanation for its success lies in its user interface, which emulates rackmount hardware with an almost obsessive attention to detail. Not only does this make Reason easy to learn and use, it gives the experience of working with the program a kind of tactile, physical dimension that musicians find gratifying. 14

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Because Reason’s interface is so clear, there are almost no hidden features to trip up the novice. On the other hand, the program is capable of far more than you may suspect. In this clinic we’ll reveal a few of the techniques beloved by Reason power users. This is not an introduction to Reason, nor is it a product review. I’m going to assume that you have the program and have read the manual. Much of Reason’s power stems from its “rear-panel” connections. You can do far more with these than connect a sound module to the mixer. For instance…

Stereo delays The DDL-1 delay line is not a stereo delay. Its left and right inputs and outputs are purely for convenience in processing stereo signals. If you want a ping-pong delay (one that bounces the signal from left to right and back), here’s how to set it up. Take the output of your chosen synth or Redrum channel and patch it into a Spider Audio splitter channel. Create two DDL-1s and give each of them an input from the Spider. Set the delay time and wet/dry mix of the two DDL-1s to taste (the delay times should not be identical). Patch the DDL-1 outputs into your Mixer. (This may have been done for you when you created them.) Now you can use either the Mixer channels’ panpots or the panpots on the DDL-1 panels to pan one delay hard left and the other hard right. With this patch, the feedback setting on the long delay should probably be set lower than the feedback on the short delay, so that they’ll fade out at about the same time. For an interesting sonic wrinkle, set both of the DDL-1s’ wet/dry knobs fully wet and patch the Spider Audio splitter to yet another Mixer input. This will give you a dry signal that hasn’t passed through the DDL-1s. Now patch a

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Scream 4 distortion effect between the left outputs of both DDL-1s and the mixer. Because the Scream is a true stereo processor, it can grind up one DDL’s output in its left channel and the other in its right channel, and pass the results on to the mixer with hard panning. This concept is illustrated in the online file Screamin In Rhythm.rns. This file also uses a couple of Matrix pattern sequencers to modulate the Scream, as described below in the section “You scream, I scream.”

Filtered delay The sound of a filtered delay, in which the successive echoes get darker as the highs are filtered out, is a classic. Reason’s DDL-1 lacks a filter circuit, but you can create one for yourself. Here’s how to do it. Note: This patch can cause feedback! Before setting it up, I recommend that you turn your monitor system way down. The ingredients are as follows: a sound source (such as a Redrum module), a DDL-1, a PEQ-2 for filtering purposes, two Spider Audio Merger/Splitters, and an extra Mixer 14:2. The concept is that the Mixer is going to be used as an attenuator for the feedback loop, so turn the Feedback knob on the DDL-1 all the way down and leave its dry/wet knob all the way wet. For clarity, start by renaming one of the Spiders “Delay Source” and the other “Delay Output.” Rename the Mixer “Delay Loop.” To set up the patch, make the following connections: Output of sound source module to Delay Source splitter. One output of the Delay Source splitter to your main mixer. (That’s the dry signal.) Another output of the Delay Source splitter to the Delay Source merger. Delay Source merger output to the PEQ input. PEQ output to DDL input. DDL output to Delay Output splitter. One output from Delay Output splitter to your main mixer. (That’s the wet signal path.) Another output of the Delay Output splitter to a channel input on the Delay Loop Mixer.

Before completing the loop, hit the Tab key and drastically lower the channel fader on the Delay Loop Mixer. (If you’ve been following instructions, it will be labeled Delay Out.)

one active oscillator, and while the filters are both switched on, the oscillator isn’t being routed through either of them. Choose something other than the sine, square, sawtooth, or triangle wave, turn the motion knob all the way The Delay Loop Mixer’s master output back to the left, and then move the to the Delay Source merger input. index slider slowly left and right while holding a note on your MIDI That’s the patch. To use it, start your sound keyboard. You’ll hear a variety of source, choose a suitable EQ curve on the PEQ, waveforms. When the motion knob and then raise the level of the channel fader on isn’t all the way to the left, the the Delay Loop Mixer gradually until you oscillator will sweep through these achieve the desired level of delay feedback. waves. With this patch you can use the PEQ to boost Next, while leaving the motion selected bands — you aren’t limited to high knob all the way to the left, park rolloff. Note, however, that boosting a band the index slider at any spot and with the PEQ has a side effect: it increases the turn the shift knob up slowly. This feedback level. The gain knob on the PEQ can will change the timbre of the wave drive the patch into runaway feedback, so be by introducing higher harmonics. careful. Since there are more than 80 You can download this patch from the VI waves, this experimentation will website. It’s called Filtered Delay.rns. take a little time. Once you have an idea what a single Malström Dive into the Malström oscillator will do, you’ll be ready to To become a demon Malström programmer, try things like turning on the waveyou’ll need to master the oscillator waveforms shaper and modulating the shift and the signal flow. The latter is shown plainly and index from the mod wheel. on the front panel; you just have to think a bit The easiest way to figure out Malström is to start with its default On glancing at Malström’s two about which buttons are lighted and where the Init patch and check out the waveforms (see text). Mod generators (equivalent to arrows point. (Hint: when the buttons on the inputs to its filter section. These can be used LFOs in other synths), you may think they can’t right side of the oscillators are not lighted, that even if the Malström is not playing notes. By control too many aspects of the sound—there doesn’t mean the oscillators are switched off. It triggering the filter envelope from a sequencer are only three output knobs for Mod A and just means their outputs aren’t being processed track, however (while leaving both oscillators four for Mod B. But there are some quasi-hidby the filters.) switched off), you can add rhythmic envelope den modulation possibilities. First, like other The waveforms are not visible, so the best or Mod generator sweeps to whatever sound is the velocity and mod wheel inputs on the left way to learn them is to experiment. Proceed as being processed by the Malström. The patch side, the Mod generators have A/both/B follows: Mal as Filter.rns shows what can happen when switches. Using this switch on Mod A, for Start with the Init Patch, which is loaded the Malström is pressed into service as an instance, you could direct it to modulate only when you create a Malström. This patch has external filter for a Subtractor. the pitch, index, or shift of Osc B. But what if you want Mod A to modulate Filter B at the same time it’s You scream, I scream modulating Osc A? Impossible, right? As good as Reason’s sounds are, the Scream There’s not even a filter knob in Mod 4 Sound Destruction Unit increases its sonic A. So hit the Tab key and take a look palette by a factor of a hundred. Running a hoat the back panel. There’s an output hum kick/snare rhythm pattern through for the signal coming from Mod A, Scream will turn it into an industrial monster and just to the left of it is an input for covered with radioactive slime. filter control. This input has its own One approach is to route several drums A/both/B switch. So patch Mod A to through different Scream units, but with a the filter input by dragging a patch slower computer, inserting several of them may cord with the mouse, switch the input add too much CPU overhead. On the other to the B destination, adjust the hand, you may not want exactly the same flaamount of filter modulation to taste, vor of distortion on the kick as on the snare. and you’re good to go. The solution, as usual with Reason, is to hit the The one limitation of this techTab key and work with the jacks on the back. nique is that your patch cord routing Create a Matrix step sequencer, make its patwon’t be saved when you save your tern the same length as your drum loop, and new Malström patch. It will be saved patch its note and curve outputs into the P1, with the Reason song, however. And P2, and/or Damage jacks on the back of the version 3.0 can save complex patchScream. By editing the Matrix steps, you can es that include rear-panel routings in change the sound of the Scream from one its new Combinator plug-in. If you drum hit to another. don’t have 3.0, you might want to The file Delay Scream.rns illustrates two of create a little text file containing my favorite techniques. Running a solo synth notes on your Reason patches. tone through a delay line before it reaches the The Filtered Delay patch looks like this, but don’t try to trace Another cool resource, before we Scream distortion causes it to intermodulate the patch cords by eye, just follow the step-by-step directions move on: the Malström has audio with itself. In addition, cranking up the resoin this article.

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nance of a tone before sending it to a distortion effect will cause the pitch to warble and gargle in an unstable way.

ReWire Solutions Using Reason as a ReWire slave device is ludicrously easy, though the details may differ depending on which host application you’re using. If you’re using Cubase SX, for instance, you launch Cubase first and then launch Reason as usual. With Sonar, however, Reason is launched from Sonar’s Insert menu. Next, find the dialog box in the host app where you switch on as many Reason audio channels as you need. If you’re planning to use only Reason’s effects for its sounds, route its synths into a Mixer module in the normal way, then patch this to the Hardware Interface. This way you’ll need only one stereo pair of inputs in your host app, which will receive the output of the Reason Mixer. To process various Reason synths through different plug-ins in your host, simply activate more channels in the host and patch the synths directly to the Hardware Interface, skipping the Mixer. I normally use Cubase to sequence the tracks that play individual Reason modules. Reason becomes strictly a sound module (more or less as if it were a VSTi). Knob and fader moves can be recorded from Reason’s front-panel controls

c l i n i c panel controls. Note that Reason’s MIDI Remote Mapping feature, which is supposed to allow you to assign panel controls to the CC number and channel of your choice, works only with external MIDI arriving from hardware, not with MIDI data arriving from a ReWire host sequencer. I recommend saving all the materials for a given song in the same folder. When saving the Reason .rns file for the song, put it in the same folder with the sequencer files. This way you can

Because Reason’s interface is so clear, there are almost no hidden features to trip up the novice. On the other hand, the program is capable of far more than you may suspect. back up the whole project (you do back up after every work session, don’t you?) more easily. When using Reason strictly as a sound module, routing MIDI correctly requires an extra step. The trick is to create a track in Reason’s sequencer that isn’t assigned to any of the Reason synths. I usually call this the “duff track.” When a new Reason synth is created, Reason automatically creates an internal sequence track for it and assigns the MIDI input to that track. This is not useful in ReWire mode with Cubase, because the MIDI data from the host will reach the module directly, while the input from external MIDI hardware will be routed to the track and from the track to the module. So after creating a new module, click on the left end of the duff track to move the MIDI icon back to it. If you don’t do this, you’ll hear doubled notes during recording.

The pleasures of polyrhythms Most Reason users record exclusively in 4/4 time. Nothing wrong with that. By default, Reason’s pattern sequencers (Matrix and Redrum) give you 16 16th-notes — that is, one-measure patterns. Extending Matrix to The Scream 4 Destruction unit increases Reason's sonic pallet by a 32 steps or Redrum to 32 or 64 steps gives you 2-measure or 4factor of a hundred. measure patterns, which again into the host sequencer; all you have to do is will be fine for many types of dance music. But select the Reason device as the MIDI input for Reason lets you get much more sophisticated if the track, put the sequencer in record mode, you dare. switch to Reason, and move the controls as If you need a hi-hat pattern that runs in needed. (Not all sequencers support this featriplets, for instance, create a separate Redrum ture, however.) for the hats, make its patterns 24 steps long, If you look in the MIDI Implementation and set the clock knob to “1/16T.” Mission Charts document (installed with Reason) you’ll accomplished. But it’s possible to get fancier. find the controller assignments for the various Since Redrum has a separate gate in and gate 16

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out for each channel, you can trigger the sounds from a single kit using different timing resolutions. This type of patching would be the tool of choice for adding a 32nd-note hi-hat flourish at the end of every other bar, for instance. Just set the second Redrum to 32ndnote resolution, make its patterns 64 steps long, enter a couple of notes at the end of the pattern using any channel, and route that channel’s gate out to the hi-hat channel’s gate in on the main Redrum. This technique can also be used to combine duple and triple rhythms in a single Redrum

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channel. The file Duple Triple Hat.rns shows how to set it up.

Learn the shortcuts One of the best ways to become a Reason power user is to get used to using the handy keyboard shortcuts: When creating a new module, hold down the shift key if you don’t want Reason to create some sort of default patch cord routing. To shift a Matrix or Redrum pattern one step at a time to the left or right, use Ctrl-J and CtrlK (Mac Command-J and Command-K). To transpose the pitches in a Matrix pattern up or down, use Ctrl/Command-U and Ctrl/Command-D. To program Redrum dynamics quickly, leave the switch set at Medium. To create a Hard step, hold shift while clicking the button. For a Soft step, hold Alt (Mac Option) while clicking the step. Rock on... In this article I’ve only touched on a few of the many subtle and complex things you can do with Reason. If you want to go deeper, I can recommend Kurt Kurasaki’s excellent book Power Tools for Reason 2.5 (published by Backbeat Books). This is not, strictly speaking, a shameless plug. While it’s true that I edited the book, I earn no royalties from it. And because I did edit it, I took the time to try out all of the dozens of slick Reason patches that Kurt included. They rock. Read the manual, buy the book, get creative, and you’ll rock too.

Jim Aikin is the author of Power Tools for Synthesizer Programming and Chords and Harmony, both from Backbeat Books. When not writing about music technology, he plays electric cello. For more about Jim’s varied activities, you can visit him online at www.musicwords.net.

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East West Quantum Leap Colossus virtual instrument A self-contained, high quality desert island library that covers all the bases Review by Nick Batzdorf

Price: $995

Company: East West (www.soundsonline.com)

Platforms: Native Instruments Kontakt player - Windows XP: VST, RTAS, DXi, ASIO, DirectSound; Mac OS X 10.2.6+: VST, Audio Units, RTAS, Core Audio, Core MIDI (stand-alone)

License: Challenge/response limited to two installs. Free to use as part of a musical composition; may not be resold as another library.

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he concept sure makes a lot of sense: a huge modern sample library that covers all the bases with one product (modern = it’s big and it streams off hard drives). And since Quantum Leap has been releasing highquality modern sample libraries for a few years, they’re in as good a position as anyone to be the ones to do that. Actually, measured strictly by disk space, only half of Colossus’ colossal 32GB of content comes from Quantum Leap libraries; producer Nick Phoenix sampled a whole lot of new material for this product. But much of the content comes from his well-established and popular libraries: East West Quantum Leap Symphony Orchestra, Stormdrum, Hardcore Bass, Voices of the Apocalypse, Ra, and others. Colossus really is a self-contained desert island library, with enough range to handle almost any kind of professional project you could think of. It has everything: drums/bass/guitar/keys, some rhythm loops,

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percussion, all the orchestral instruments, pitched and non-pitched ethnic instruments, synth sounds and drones, choir…and and and.

Workstation du jour Colossus comes in a dedicated Native Instruments (NI) Kompakt Player, an OEM engine derived from their full Kontakt sampler, so you don’t need anything to run it other than a host—in fact you can even run it standalone for live use. However, you can open Colossus instruments in Kontakt or Kontakt 2, which permits more extensive editing. While the Colossus instrument doesn’t pass Logic Audio 7.1’s AU validation test as of this writing, it actually works fine in Logic. Most of this review was done in Logic on a dual 2.5GHz G5 Mac running OS X 10.4.1, but I regularly run NI players in Digital Performer and Pro Tools too, and also on the Windows machines in my studio. As modern sample libraries go, Colossus isn’t especially demanding of the computer it runs on. That’s because its performance programming is relatively uncomplicated, for example there are no keyswitch programs that eat up oodles of memory, and very few programs with recorded reverb that use a lot of polyphony. You should have no problem running large-scale compositions on a single machine—PC or Mac—provided it has enough RAM installed. As a very rough guideline, a machine with 1.5 2GB should be able to load four instances of the 8-part multitimbral Colossus instrument, which of course is 32 instruments; the test G5, which is overstocked with 5GB of RAM, was able to fill over ten instances inside Logic before the system became unstable—80+ instruments, which is really impressive. Colossus is being presented as a modern-day equivalent to the keyboard workstations we relied upon for years, and part of that concept is that its instruments are almost all playable in real time. So rather than loading, say, eight different violin articulations—an entire instance of the player just for one instrument—and splitting notes to program your phrases, you just play a one-size-fits-all “strings” program in real time. In many cases that approach exchanges some subtlety for ease of use, but the choice makes sense given the instrument’s intended application. Having said that, the programming in Colossus takes good advantage of the features (CONTINUED ON PAGE 63)

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Cycles Some unique loops to freshen the pallet

Review by Chris Meyer

cycles vol. 01: sustained encounters

cycles vol. 02: unnatural rhythm

cycles vol. 03: incidental gesture

$99 each

Cycling ‘74 (www.cycling74.com) 379A Clementina Street, San Francisco, CA 94103. 415/974-1818

platform: DVD-ROM of 24-bit .wav samples + audio CD, many REX versions included

license: Free to use as long as combined with other sounds or not resold as another library.

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ibraries of industrial or unusual electronic sounds are a bit of a guilty pleasure for many of us: they’re interesting to listen to, and we think they’re going to push us to be more edgy in our own compositions…but at the end of the day many of them sit on our shelves, all but unused. I’m happy to say the cycles collections are a pleasure we do not have to feel guilty about—they are perhaps the most useable libraries I’ve encountered in this genre so far. Some forward thinking has been put into these libraries. Each has a DVD with 24-bit .wav files, in 44.1 and 48 kHz versions, as well as a normal audio CD that includes unfaded, untrimmed copies of the samples. All samples can be looped, even if they are not naturally “loops” themselves (in other words, the linear ones have clean fades at the head and tail). All the libraries come with nicely-formatted PDF and Excel indexes of the samples, which include (where applicable) tempo, duration in bars, and tonal center. Okay—but what about the sounds? sustained encounters features evolving drones, sound environments, and “pads” that range from lush, dark, and mysterious to industrial in nature (the latter not necessarily being grating—more like being in a spacecraft’s engine room or communication center). There are over 100 “events” or loops that range from seven seconds to a minute, plus 18 longer pieces that range from 40 seconds to over two minutes in length—including several provided as both stereo and 5.1 surround mixes.

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Frankly, given their experimental nature and relative lack of rhythm or clear tonal center, this is the library that would be hardest to apply to a wide variety of music. I mainly hear these as bridges or extended intros for darker pieces. unnatural rhythm contains roughly 300 rhythmic loops, ranging from machines (real and imagined) to interestingly-looped acoustic phenomena (such as pennies rolling around a jar) to raw bleeps and hisses from Buchla and Serge synthesizers. Typical tempo range is wide, centering around 70 to 120 bpm; most of the loops are from one to eight bars in length, although 23 are presented as “xtended” versions ranging from 16 to 96 (!) bars. REX versions of many loops are also provided. REX is Propellerhead ReCycle’s format; ReCycled loops have been sliced into individual beats so they can be played at any tempo. This happens automatically in any REX-compatible host program. I’m happy if I find a handful of loops in new libraries that are promising for my compositions; I easily found a couple of dozen here. There aren’t a lot of variations presented, but the longer loops have nice evolution, meaning you could cut them into smaller phrases. Here again, the tonality is more often lush than jarring. Not many composers would use this library for their entire rhythmic beds, but it

r e v i e w VI does provide nice intros, breakdowns, and unusual layers. incidental gesture is a thoroughly pleasant surprise. This is the most melodic of these libraries, containing a number of blown, plucked, and struck acoustic instruments (as well as some natural sounds and purely elec-

tronic confections) playing meandering lines, then heavily processed to create otherworldly sonic narratives. Most of the performances are one to two minutes long with listed tempos and tonal centers; the longer performances have also been edited into phrases. Of course, the problem

Beats Working in Cuba sound library + virtual instrument

Review by Chris Meyer

Beats Working in Cuba

($399.95)

Zero-G (www.zero-g.co.uk; distributed in US by 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. Challenge/response installer limited to two unique installs.

license: Free to use as part of a musical composition; may not be resold as another library.

his is not just a loop library, it documents a musical genre. Producers Barry Sage and Gonzalo Lasheras Garcia traveled to Havana, Cuba where they recorded local professionals playing examples of “the twelve prime Afro/Hispanic rhythms of the Cuban music culture, including: Cha Cha Cha, Danzon, Son Montuno, Bolero, Pilon, Son Traditional, Guajira Son, Guaracha, Mambo, Songo, Timba, Conga Habanera (Carnival), and Conga Moderna,” often providing modern (with drum kit) and traditional (sans kit) versions of each. In addition to over six gigabytes of 24-bit loops, you get an informative HTML file covering details of the players, the instruments, the styles, the recording process, and the samples themselves. There’s even a DVD documentary included! Performances of each style are broken into roughly 20 to 60 2- or 4-bar segments (including an intro and ending, as well as verses, choruses, or just pieces of a linear performance). Numerous mixes of each performance are provided, including a full mix plus each instrument

with this (and any of the cycles libraries) is that their utter uniqueness will make them easier to spot when others use them. But they are so far off the beaten path, such occurrences will probably be rare. But come to think of it, forget you read this—I’m keeping these libraries to myself.

soloed. These are then presented dry, live, and often in a surround version (provided as two stereo pairs). The samples are played back through the familiar Native Instrument Intakt virtual instrument [see sidebar]; where the individual hits can be isolated, they are sliced using Intakt’s Beat Machine, complete with MIDI files to export of the pattern for each sliced performance. Sometimes these individual hits are mapped to an Intakt keyboard as well. You’ve heard the saying “presentation is everything”—and with this many segments and variations available, presentation becomes quite a challenge. The answer Zero-G decided upon was to present each 2- or 4-bar segment as a folder of discrete Intakt patches. Each single segment then has up to two dozen individual patches. For example, Conga Moderna provides separate patches for congas dry, congas live, congas LsRs (surround pair), bells dry, bells live, bells LsRs, kick, snare, hi-hat, cowbell, toms, drum overhead mics, drum front ambient mics, drum LsRs, and a full stereo mix. There’s usually one patch per segment that provides several useful mixes and breakdowns of that segment, but there are essentially no patches that make more than one segment available at the same time. This means if you want to recreate a, say, 30-segment performance in your piece, you need to load 30 patches, one at a time (a few patches present three segments back-to-back; oh, for more of those…). If you want to combine two mixes, double that. Since the segments were cut out of a linear performance, not all of them loop naturally by themselves, making it a bit harder to “cheat” by repeating a smaller number of segments. One more detail: a chart is provided, showing how to offset each loop in time to achieve the most authentic feel. In other words, you shouldn’t just start each loop on the downbeat—and the offset changes per segment.

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r e v i e w VI Those familiar with piecing together individual articulations in an orchestral library may be comfortable building a performance this way; a more groove-oriented person like myself may not. Personally, I would have preferred that a single patch contain a single mixdown of as many different segments as possible. Then provide me alternate patches—with the same segment-to-key mapping—that have alternate instrument breakdowns and mixes. That way, once I had built a MIDI file that played back

the segments in the desired order, I could then just load different patches to hear different breakdowns and mixes. If you are looking for a definitive library of Cuban rhythms, Beats Working in Cuba will blow you away. If you are looking for a way to quickly add an Afro/Hispanic flair to a piece, this same library may stop you dead in your tracks. Zero-G is looking into updating Beats Working in Cuba to provide more performance-oriented patches, so all of us can be happy.

Scroll Wheel Support in Logic Pro 7

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Most people aren’t aware of the mouse/trackpad scroll wheel functions in Logic Pro 7. They’re very convenient. Try them— you’ll say, “I didn’t know that!”

modifier

scroll wheel function

none or shift/command

scroll screen vertically

shift or command

scroll screen horizontally

option

zoom vertically

shift/option or command/option

zoom horizontally

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Intakt Overview

any higher-end sound libraries now come integrated with a virtual instrument sample player. Part of the reason is to provide a form of copy protection for the samples residing behind the curtain, but they can also present cool sound warpage opportunities to extend the original sounds. Some of the most popular VIs for this task are Intakt and Kompakt from Native Instruments (www.nativeinstruments.com). Intakt is optimized for rhythm loop libraries; both Percussive Adventures and Beats Working in Cuba (reviewed in this issue) use it. The limitation is that the included version can only play the library it came bundled with; you can upgrade to the full version of Intakt—which can load any compatible library—for $129 ($100 off), and to Kontakt—a fuller-featured multitimbral sampler—for $249 ($200 off). Intakt works as a stand-alone program or a plug-in; in the latter case, it immediately picks up the tempo from the host application. It contains three different algorithms to stretch the loop to match the requested tempo: Sampler, which changes pitch to match; Beat Machine, where the loop is sliced into individual hits that are then triggered at the correct times; and Time Machine, which time-stretches and compresses the loop while keeping constant pitch without using discrete slices. Intakt also features nice subtractive synth engine to feed the samples through. It includes a pitch envelope, AHDSR envelope, envelope follower, two LFOs with multiple waveforms (which can be tempo-synchronized), two filters featuring multiple pole and pass configurations, a master filter that can also function as a 3-band EQ, and an effects section that includes a synchronizable delay plus lo-fi and distortion sections. Intakt is easy to use, providing all controls in a single window. This window can be a bit large, especially when trying to view a sequencer or other program behind it, but you can “collapse” sections to consume less real estate. When using one instance of it as a plugin for the Mac version of Ableton’s Live, I found it took up about 7-8% of the CPU load on both a 1.5 GHz PowerBook and dual 1.8 GHz G5; adding a second instance took up about 30% (!). You can trigger multiple loops in one instance of the plug-in, but you are at the mercy of the library developer to arrange useful sound banks with multiple parts in the same patch. Otherwise, you may be better resampling its output into your compositional application of choice.

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online

www.virtualinstrumentsmag.com

Sampling With

King Idiot

Part 1: Modern features for old libraries

Whether they’re in keyboard instruments or in hardware samplers, older sample libraries lack many of the advanced features you find today. Here’s how to update your favorite sounds so they can co-exist with your modern software instruments. Please go to the downloads section of www.Virtualinstrumentsmag.com and grab the example files.

e’ve seen revolutionary advances in the

realm of sample playback over the past

few years. The features in today’s samplers have opened up a whole new breed of options. There are expressive features like real-time timestretching control, multi-layered crossfades within a single patch, lots of control over integrated effects, as well as general options that were available but weren’t always easy to set up—until now.

By Ashif “King Idiot” Hakik

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The old sample libraries we were happily using before end up buried at the bottom of a moving box. Well, I’m here to tell you non-sample tweakers and rookie patch programmers to dust off those old CDs, Zip drives, and—yes—floppies. Get ready to give your favorite forgotten libraries new life with some TLC in your DAW. In this issue we’re going to focus things you can do to the raw samples themselves outside

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the actual sampler, manipulating them in an audio editor. We’ll detail some options for exporting the samples and then re-importing the reborn samples. For this article I’m going to describe how to accomplish the edits in Native Instruments Kontakt and Steinberg Wavelab on a PC, but these techniques work with any sampler and audio editor. You will have to learn the basics of your sampler on your own, though, since this is a magazine rather than an instruction manual. We’ve uploaded some samples and instruments generously given to us by the fine folks at Project SAM. These samples are rough test samples of a trumpet section they used as proof of concept of their sample method, and not part of their release products, but they’ll be perfect for us here. Download them at www.VirtualInstrumentsMag.com.

Sanctioned cloning The first thing you need to do is make a copy of the instrument you want to work with—an actual physical copy of the samples.

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old sounds work with your modern libraries, and you don’t want to stop and run just this one sound through a reverb. First, open the new set of samples in your waveform editor, and apply your preferred reverb settings to each sample. As the focus of this article is sample manipulation, I won’t go into detail about what reverb you should use, or what reverb settings are best—although I’d advise against using a spring reverb setting if you’re trying to emulate a real concert hall! Wavelab has a batch processing feature that will process all the samples with one set of reverb settings. Batch processors are timesavers, but if you don’t have one, don’t fret— there’s hope in the form of scripting applications like QuicKeys (Fig. 1). Once the processing is finished, all you need to do is reload the samples into Kontakt (or your sampler), and the samples will retain all the tuning info volume information and playback settings from the original. You can rename the instrument to whatever you want now. It’s also easy to layer the original and processed instruments into Kontakt and trigger them on the same MIDI channel. (See Fig. 2) You can use MIDI to adjust the balance between the original and new instruments in real time. Many of the older sample libraries that focused on live instruments were recorded in fairly dry rooms. You may prefer the wet samples to the original. Fig. 1: QuicKeys for Windows. This macro utility from CE Software, also available for Mac, can save a lot of time whenever you’re doing anything repetitive—which happens a lot when you’re doing the same thing to dozens of samples.

As a rule, it’s a bad idea to burn your bridges by working on the original samples. In Kontakt the easiest way to do this is to load an instrument, rename the instrument (for this example we’ll add the word “edit1” to the end of the name so it becomes “SAM Trpts TEST_VI1_edit1”), and then save it, selecting “patch + samples”. This will make a complete duplicate of the patch and the samples with a new file name and a new sample folder. Then remove the instrument from your sample player (in Kontakt, just reset the Multi). From here on you’re going to be able to do quite a few things to the samples: add reverb, put the samples through a ring modulator, time stretch/compress, run the samples through a vocoder, even all of the above at the same time. Or other things. You can also perform waveform edits across all the samples to alter the way they play back in your sampler. For instance, you can change the attack characteristics of the samples by changing just the speed of the start of the sample. Or you could add another sample on top of it. It’s up to you whether you want to create new or strange sounds, or stick to more traditional effects. We’re going to stick to some more practical things in this article—one of the

easiest being to add reverb, so let’s start there.

The wetter the better Let’s make a new set of samples that’s a completely “wet” version of the trumpets. We’ll put them in a big hall with your favorite reverb effect. You could just run the whole program through a reverb unit, but there are advantages to applying the reverb directly to the file. Among them: you don’t have to use processing power later; and you can spend the time matching your samples to others in your collection now so you can just load the program later. After all, the goal is to make your favorite

The headless sample Have you ever played a string line and found that some note transitions just don’t happen fast enough, no matter how much you overlap notes in your MIDI sequencer? You know, that dreaded note-to-note “sucking” effect that literally sucks you back to reality and reminds you that your musical cue is being played by a machine with no sense of the music? What’s causing this effect—and what is missing in the samples—is the natural tendency for the string players to move the bow slightly faster and with slightly more pressure on notes after the beginning of a phrase. Along the same lines, a trumpet player doesn’t always re-attack every note with his breath—sometimes he closes a valve that causes it to change the pitch. But with sam-

Fig. 2: Wet and dry versions of the same program in Kontakt. Both are on MIDI channel 1 so they trigger together; you adjust the relative balance with the volume controls.

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Fig. 3: Getting rid of the first 200ms of audio in Wavelab to create a sample with no attack.

ples, a new note starts playing every time you press a key on your keyboard. I’m going to outline how to make “legato” variations of the samples and add them back to existing instruments. There are many different ways to create legato samples, but we’re going to keep it simple. This article is about working on reimporting samples rather than audio editing; I’ll save that for a future article just about legato. Start by making a clone instrument and opening the cloned instrument’s samples in your audio editor. Select about 200 milliseconds of audio, starting from the very beginning of the sample. Now delete that portion of the sample, and save it. Do this to every sample in the cloned instrument’s sample folder. (See Fig. 3) Wavelab has a feature called Auto Split that can speed up this process. It can batch-process all the samples to have their “heads” lopped off, saving the time of opening and editing each file by hand. But again, QuicKeys can automate this process if you’re not using Wavelab. What we’ve done is remove the attack characteristic of the sample. This is an old trick that some older hardware samplers did internally. It’s simple, but it can be very effective for some instruments. We’re through with the waveform editing now, so open the headless instrument in Kontakt or your sampler and save the cloned instrument under a different name.

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Now it’s time to adjust the programming to make this new instrument work with the existing one. While the actions that need to be taken are actually simple, the descriptions can be a bit tricky to read, so just take your time

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and focus on doing each step correctly. Don’t forget to look at the pretty green pictures! We’re going to force the instrument to play the headless samples when the mod wheel is above a value of 64 (center position), and to play the normal samples at any value below 64. In other words, we’re making the mod wheel act as an on/off switch for legato. You can use any other MIDI Continuous Controller (cc) if you want, in fact sustain pedal (cc64) is the most commonly used controller for this type of programming feature. The following is Kontakt-centric, but the steps are similar in all samplers. First, click on the Edit button of the original instrument, then click on Group Start Options in the source module. Change the setting from Always to Start On Controller. Set the CC# to 1 (mod wheel). Set the value range to Between 0 and 64. (See Fig. 4) Now click on the headless instrument’s Edit button, and again click on Group Start Options in its source module. Change the setting from Always to Start On Controller. Again, set the CC# to 1 (mod wheel). But this time set the values to Between 65 and 127. Thats it. Setting both programs to the same MIDI channel will now force the legato samples to be played with the mod wheel all the way up and the standard samples with with it all the way down. You’ll need to adjust the attack time of the new legato group to taste. Usually 30 - 50ms is a good place to start, with varying results depending on the instrument. While you’re there, try assigning the mod wheel to the release time of the instrument, so the legato programs have a longer release. Tweaking the note endings can really help create believable melody lines. (See Fig. 5)

Fig. 4: Setting the program to trigger only when the mod wheel (cc#1) is between 0 and 64.

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The trick is to start phrases or groups of notes to be played in one bow or breath with the standard instrument, and then switch to the headless instrument for subsequent notes in the phrase or group.

Release triggers Now let’s try something a bit more complex, using concepts from both of the above techniques. This time we’re going to create release triggers for instruments—even ones that don’t have any decay or reverb in the samples to start with. Release triggers are samples that play back when you release the note, that is when a MIDI Note-off message is received by the sampler. These can be instrument resonances, sounds that occur naturally when you release a key (as with harpsichords), or they can be natural ambience decays. We’re going to create ambience decays to make it sound like the instrument was recorded in a big space. Start by making a renamed copy of the instrument you created earlier with wet samples. Load its samples in your waveform editor. We’re going to edit the sample by deleting all the audio except for the decay/reverb tail. This isn’t a process that can be done by the numbers, so it will take some time listening and editing to find the right spot. One trick is that it can be easier to hear where the decay starts if you listen to the samples backwards. So you could try reversing the sample, cropping the

Fig. 5: Adjust the legato instrument’s attack time. While you’re at it, try assigning the mod wheel to control the release time too.

decay portion (which of course is rising when reversed), and then reversing it back the right way, leaving just the decay. Parts of this process can be scripted in QuicKeys to speed things up. Once you’re done editing the samples, do the now-familiar routine. Save the cloned

instrument under a different name, and then load up the original and cloned instruments with the original in the first slot and the new one in the second. As shown in Fig. 6, click on the release trail instrument’s Edit button, and click on the RLS.

Changing the scales of VSL string runs in Melodyne

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elemony’s Melodyne allows you to treat monophonic (i.e. single-line, not chords) audio as if it were MIDI data. It can change the pitch, timing, and formants independently of each other. It can quantize audio. It can correct pitch, of course. It’s a remarkable creative tool. One of the many things you can use Melodyne for is taking sampled scale runs—such as the violin section runs in the Vienna Symphonic Library’s Performance Set—and changing the scales they play. While VSL comes with a big selection of standard string runs, it’s not possible for it to include every permutation you can imagine. This graphic shows a C major scale run that’s been changed to a diminished whole tone scale (1/2 step, 1 step, 1/2 step, 1 step…). Melodyne has a very hard time separating the notes in this type of material, since it’s very imprecise by design. Not only are there 14 violins playing a very fast run, they’re playing it legato. But all that fudginess works in our favor, because the new scale doesn’t have to be very precise either. All you have to do is separate the notes somewhere near where they switch, This 14-violin diminished whole tone scale run used to be a C major scale before it and nobody can hear the difference; it all slurs right by very met Melodyne. quickly. You can also time-stretch or –compress the run very easily if it isn’t quite the right tempo. Melodyne also makes it easy to adjust Several other libraries also have string runs, of course. Even the the levels of individual notes. And so on. The point is that you can Orkestra soundbank that comes with Propellerhead Reason has create your own library of runs to augment the ones that come them. with your sample library.

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Trigger button in the Kontakt Source module. Now adjust the attack envelope in the ADSR module so the release sample doesnâ&#x20AC;&#x2122;t come in too abruptly; 80ms or so is usually okay. Set the release envelope high enough to let the sample play outâ&#x20AC;&#x201D;two seconds should be good for an orchestral hall trail. Adjustments should be made to the attack envelope of the release trigger instrument as well as the release time of the original instrument. You want to try and make it so that the release triggers arenâ&#x20AC;&#x2122;t too abrupt and you get a fairly smooth transition between the two. Itâ&#x20AC;&#x2122;s all a matter of taste. And thatâ&#x20AC;&#x2122;s it. You now have a combination of instruments that will play back a release trigger sample when you release a key on your keyboard.

Become an Idiot I suggest spending a little time familiarizing yourself with your software sampler, even if youâ&#x20AC;&#x2122;re using Kontakt and are able to follow the steps in this article directly. Weâ&#x20AC;&#x2122;ve skipped a lot of details for the sake of reaching the goal. If

Fig. 6: Adjusting the release trigger.

What if your â&#x20AC;&#x153;old sample libraryâ&#x20AC;? is a keyboard? The tricks in this article assume that you only have to import the programs from your old libraries into your software sampler. But you can also sample the ROM sounds from a keyboard or module you ownâ&#x20AC;&#x201D; taking care not to violate any license agreements. The way to do this is to program MIDI notes in your sequencer to trigger the keyboard at various velocity levels, and then record the result. Depending on how critical you are about the sound, you may not need to sample every half step. But a minor third is about the maximum youâ&#x20AC;&#x2122;d normally want to go. After programming this â&#x20AC;&#x153;scale,â&#x20AC;? set all the velocities to the first velocity level you want to sample and record the instrument. Then select all the notes and set the velocity to the next level you want. And so on. If this seems tedious, thereâ&#x20AC;&#x2122;s a great program called AutoSampler (www.Redmatica.com) that will do all this and build an instrument automatically for the EXS24 sampler built into Logic. Youâ&#x20AC;&#x2122;re not a Logic user, you say? Import the EXS program into your sam4,3AD?6)MAG PDF !pler.

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jHEFk LEJgNGNN

you spend time understanding why things work the way they do, you can come up with variations of the techniques that may be more suited to your needs. Those of you who are versed in sample editing techniques and are wondering why my way of doing things is different from yours, just wait for future articles in this series. Weâ&#x20AC;&#x2122;ll have a good time getting down and dirty. Ashif â&#x20AC;&#x153;King Idiotâ&#x20AC;? Hakik is an award-nominated composer whose credits include several major video games. He does sound effects design, heâ&#x20AC;&#x2122;s orchestrating the upcoming QueensrĂżche record, and heâ&#x20AC;&#x2122;s edited/programmed several sample libraries you probably use.

Producer wanted. This band plays your songs exactly the way you want them.

Steinberg is a registered trademark of Steinberg Media Technologies GmbH. All specifications are subject to change at any time without prior notice. ÂŠ 2005 Steinberg Media Technologies GmbH.

Virtual Bassist and Groove Agent 2 give you a perfectly integrated rhythm section, powering your songs with infectious grooves in any of a huge range of styles. The Groove Agent 2 virtual drummer offers both the hottest and most popular beats from the last 50 years of music history. Virtual Bassist puts a huge range of electric bass intros, fills and variations at your fingertips, and because theyâ&#x20AC;&#x2122;re not based on dronelike MIDI parts but real phrases played by real musicians, they sound all the more lively and dynamic. And thanks to the innovative new GrooveMatch feature, your Virtual Bassist parts are dynamically adapted to existing drum grooves. The brand new HALion Player is your gateway to the ever-growing number of professional HALion sample libraries, and even comes complete with a full, studio grade HALion library on DVD to get you started. Your band is waiting. Tell them what to play.

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Chicken Systems Translator Pro. The ultimate program for dealing with professional sampler formats. Take any sampler CDROM/disk in one format, and Translator can convert its samples, presets, or banks into another sampler’s format. Translator reads all the popular formats and performs a number of intelligent tricks that no other conversion routine can do. 44 source formats and destination formats are currently supported in the Windows version! (Mac version also available—winner’s choice) www.Chickensys.com

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Spectrasonics Stylus RMX and SAGE Xpanders Exceptional grooves in a brilliantly-executed virtual instrument

Review by Nick Batzdorf

Stylus RMX, $299; SAGE Xpanders, $99 each.

Spectrasonics (www.Spectrasonics.net), distributed by Ilio Entertainments, P.O. Box 6211, Malibu, CA 90265. 818/707-7222, 800/747-4546.

Platforms: Mac 10.2.6+ AU, RTAS, VST; Windows 2000, XP VST.

pectrasonics’ first Stylus release became widely popular because of its loops and sounds. Its successor Stylus RMX, which subsumes “classic” Stylus, is an extremely sophisticated virtual instrument. Stylus RMX advances the way you can work with rhythm loops to a whole new level. Just saying that RMX is a RAM-based, multitimbral sample and loop player with extensive processing, a great effects rack, an intelligent randomizer, and oodles of great, really wellorganized content (plus optional expansion content and the ability to import your own) doesn’t tell the whole story. But it is the high concept. The lower concept requires some explanation.

License: Free to use as long as not made into another library. Need not be combined with other sounds.

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Slices While the content in RMX is derived from groove performances, called Suites, its loops aren’t traditional audio regions. Instead, the performances are cut up into time slices (usual-

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ly spanning an 8th or 16th note) similar to the way Propellerhead’s ReCycle program works. Spectrasonics is very good at this, and they call their proprietary slicing method Groove Control™. Taken to the extreme, each individual slice could be independently filtered, pitch-shifted, reversed, panned, etc.—and run through its own chain of RMX effects units. It can also be sent to its own output if the host sequencer supports that, but with all the integrated processing you may find that unnecessary. Everything in Stylus RMX can be triggered by MIDI, which opens up all the host sequencer’s editing functions to the loops. That means you can shuffle the order of slices, use the sequencer’s quantizing functions, and so on. Pretty much every RMX parameter can be automated in the sequencer or controlled in real time using the simple “watch and wiggle” MIDI Learn function. In the default MIDI mode, called Slice Menu, each successive slice in the pattern is assigned

r e v i e w VI to the next ascending chromatic note. So playing a chromatic scale in rhythm, starting on the first note in the pattern (usually C1), will play the pattern. You audition everything inside RMX itself without using MIDI; while this is reportedly not how the software actually works internally, the effect is the same as if the grooves were being triggered by Standard MIDI files (SMFs) that reside inside the virtual instrument, and they’re synced to the host sequencer’s tempo. In supporting hosts—including Logic 7, which I used most of the time for this review—you do drag actual MIDI files directly from the RMX instrument onto sequencer tracks to build rhythm parts; in other programs like Pro Tools you drag the MIDI files onto the desktop and import them. Spectrasonics calls RMX’ innards SAGE: Spectrasonics Advanced Groove Engine. A utility to convert older Spectrasonics or Ilio (their distributor) Groove Control libraries to SAGE is included. You can also use the utility to convert ReCycle’s REX files, but the results aren’t guaranteed; Groove Control libraries are processed really well, so the loops can play at pretty much any tempo and still sound as good as they do at their original tempo. The MIDI files simply trigger at the host sequencer’s current tempo, and this method has several advantages over timestretching/compressing unsliced loops. Several sub-loops, called Elements, are included with most of the Suites. These might be A and B variations of the main pattern, fills, submixes, a beat without snare or bass, lighter versions of the loop, or maybe just one or two of the parts in it. Even though the Elements are either 2-bar or 4-bar patterns, there are many ways of constructing longer phrases and cus-

Stylus' effects rack, here with a Tape Slammer, a Wa-Wah, and a BPM Delay set up.

Real time There are several ways of performing RMX in real time. The default RMX MIDI mode is Slice Menu, but it has an additional MIDI mode called Groove Menu in which the Elements in a Suite are triggered by MIDI keys; so rather than triggering individual slices, you’re triggering loops. Groove Menu mode makes it easy to string

Just saying that RMX is a RAM-based, multi-timbral sample and loop player with extensive processing, a great effects rack, an intelligent randomizer, and oodles of great, really well-organized content (plus optional expansion content and the ability to import your own) doesn’t tell the whole story. tomizing the patterns. One is simply to layer Elements from different Suites, another is to rearrange the slices in a sequencer, and there are others. There’s another view in the browser called Groove Elements, which has loops grouped by instrument type. This makes it easy to find, say, a 16th note tambourine or shaker part with accents on 2 and 4, or a bongo pattern. Groove Elements contains a vast amount of excellent material, and you can use it on its own or mix and match it with anything else in RMX.

together patterns in real time just by triggering them at the appropriate point (quantized to the next note or bar, etc., if you want). Elements play as long as you hold their assigned keys down; you can layer them by playing multiple notes, play only a portion of them (maybe just the hits on the downbeats), stop and start them, and so on. This allows for real-time pattern performance. Another way of playing Stylus RMX is to use Kit mode, in which the same samples used to create the Suites are mapped to the keyboard

following the General MIDI mappings. This turns RMX into a sound module. It doesn’t have the detail you find in modern drum sample libraries—lots of velocity layers, separate left and right hand samples, and what have you— but the sounds are great. And in this context (mostly processed sounds) you don’t miss that. RMX has an extremely clever interface from top to bottom, but I do have one wish: a closer relationship between Kit mode and the patternbased modes. I often found myself wanting to play my own grooves using the sounds in a Suite I like, but switching to Kit mode gives you a blank slate rather than sounds from the current suite. (Opening another RMX instance on a separate track avoids having the Suite disappear, but the point is that Kit mode isn’t related.) While you can play the slices in a Suite from the keyboard without loading them into Kit mode, the time slices often contain more than one sound, e.g. bass drum and hi-hat. Perhaps preset Kits with the same names as their corresponding Suites would be a solution—which you could assemble yourself, of course, but that could be daunting. RMX is 8-part multitimbral, with each of the eight parts assigned to a separate MIDI channel. A part can contain a standard Element or a Sound Menu instrument—which we’ll explain next—so you can both layer grooves and play along with them on different MIDI channels. If you need more than eight parts, you simply open another RMX on a different track.

Sound Menus While Kit mode uses the same samples as the Suites, there’s a huge additional library in RMX called Sound Menus with its own content.

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r e v i e w VI Spectrasonics says there are over 10,000 sounds here, and we’ll have to take their word for that. There are 17 Sound Menu Suites that range from percussion instruments (Suites of claps, hi-hats, snares, toms, etc.) to Cinematic “sound design” hits to Retro Hits (big band falls, etc.), Guitar Bits (processed guitar chunkas)…a huge amount of excellent material. Each of these 17 Suites contains several Elements, each of which

ment in the traditional way, RMX lets you select the slices you want to work on—meaning the slices you want to filter, effect, randomize, etc.—by creating Edit Groups. You can create Edit Groups for downbeats, backbeats, every specified note value (for example every quarter note, or the second 16th note on each beat, etc.), just one Element, one MIDI note, every slice…suffice it to say that there’s a way to catch just what you want 99% of the time.

The Chaos designer is an intelligent randomizer with several self-explanatory parameters. You can apply Chaos selectively to anything from an individual time slice to backbeats to the entire groove.

brings up a keyboard full of sounds, up to five octaves—i.e. up to 60 sounds. Some of these sounds are straight, but most of them are processed and electronic/synthy. The sound content here is great and it covers a wide range. What’s more, you could run every Sound Menu sound through its own independent effects and processing chain.

Editing It’s very easy to get around RMX. In addition to its browser, which is accessible from all screens, the instrument has four screens: Edit, Chaos (the intelligent randomizer we’ll cover shortly), FX, and a Mixer. In the Edit screen you have control over amplitude, filtering, overall panning, and the pitch. Each of the first three can be controlled with an LFO that has a choice of eight standard waveforms. There are also 4-stage envelopes for amplitude, filtering, and pitch. RMX has two different filters: a more rudimentary Master filter, and a very nice-sounding multimode Power filter that uses more processor. There’s also a pitch control, along with a sample start slider and a Reverse button. Finally, there are handy buttons for doubling and halving the tempo, which you can do repeatedly in either direction. While the editing and in fact the whole instrument are very easy to understand and use, RMX comes with a concise series of QuickTime video tutorials by producer Eric Persing to get you up and running. But while Eric is a very charming and engaging personality, I do admit to wishing for a printed manual as well, both for speed of learning and for reference. [Hold the presses: Version 1.3 due out this summer will have an integrated reference manual that can be printed.] Rather than separating notes by the instru36

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What’s really useful is that edit groups can be soloed and muted. All the tweaks in the Edit, Chaos, and (for insert effects) the FX windows apply only to the current Edit Group. To do something global, you simply select the Main edit group. Would that all user interfaces were this easy to deal with.

Chaos RMX’ randomizer has six self-explanatory sliders that you raise to, as Persing puts it in the tutorials, “increase the probability” of the parameter changing: pattern, repeat, reverse, timing, pitch, and dynamics. This lets you create endless variations of all the Elements; when you find one you like, press the Capture button and drag it to a sequencer track (again, if your sequencer supports this).

tweaks or radical transformations. My favorites are the crunchy Flame Distortion and nasty Spring Verb, but they’re all very usable. While no substitute for a great convolution reverb processor, even the PRO-verb sounds fine. RMX has four shared auxes and a master, plus you can insert effects on each edit group. There are three slots in the “rack” in each of these paths. Version 1.2 added both single and 3-rack effects presets. Not just the effects, but the whole instrument is very efficient. When it’s just sitting there idle, it takes up no CPU; the same applies to inserted effects processors that aren’t passing audio. Playing one Element takes up about 4% on the review dual 2.5GHz Mac G5. A PRO-verb instance takes up an additional 8% or so, and turning on the Power Filter eats up about 3%. The mixer is straightforward: each of the eight parts in Stylus RMX has its own channel strip. This is where you balance levels, pan parts, send them to the four FX auxes, and route them to separate outputs for external processing.

Libraries The base library with Stylus RMX, which incidentally stands for “remix,” is about 7GB. It includes the “classic” Stylus sounds and the new RMX library. With all the Stylus libraries, you get the overwhelming sense that its beats are right on the money. When you’re looking for a readily identifiable contemporary sound, Stylus has it— regardless of the particular style. There’s considerable variety in all the RMX libraries, but the roughly 200 grooves in the classic Stylus library mostly have a hip-hoppy feel. This collection contains mainly retroprocessed real drum sounds, and almost all the beats have slightly swung 16ths or 8ths. They have a great feel. The new RMX library uses a lot of electronic and heavily processed percussion sounds. You’ll

Taken to the extreme, each individual time slice could be independently filtered, pitch-shifted, reversed, panned, etc.—and run through its own chain of RMX effects units. You can easily control the amount of disarray Chaos creates. Since it only affects the currently selected edit group, you can, say, vary the pitch of just the backbeats, randomize the hihat pattern, or vary the timing of a bongo part. Chaos can be weird, subtle, or it can simply avoid stagnant loops. You can also use Chaos live on patterns being triggered in MIDI Groove mode.

FX and mixer RMX’ effects work very well with rhythm parts. Please see the graphic showing the available list on page 35; there’s enough there for subtle

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find a wide range of interesting grooves, from funny to powerful to danceable to dramatic and in between. Many of the grooves have an electronic/ethnic percussion ensemble feel. There are five optional SAGE expanders (“Xpanders”) for Stylus RMX. Each is about 1GB in size, so there’s less content than in the core libraries, but it’s still a heck of a lot. For example, the Retro Funk Xpander has 54 Suites, 507 Elements, 305 Fills, 44 extras (bongos, conga, shaker, tambourines), and 85 Multi patches…a feature we didn’t mention; you can save the multitimbral state of the instrument as a Multi.

r e v i e w VI Each Xpander includes some “bonus content” from Spectrasonics’ Bizarre Guitar, Distorted Reality, Vocal Planet libraries. The Bizarre Guitar material here is mainly scratchy wah or otherwise interestingly processed short licks—really funky ones—although there are a couple of more ambient Elements. Most of the Vocal Planet loops are mouth percussion (processed vocal noises), but there are also some really spirited claps, stomps, and exuberant noises from a gospel choir. The Distorted Reality grooves are stylistically similar to the base RMX library—electronic percussion—but the rhythms tend to be simpler. You can layer them for complexity, of course, and save the result as a Multi. It’s worth pointing out that due to the way some of the drum kits in the Xpanders are recorded with ambience, a few of them can glitch if you randomize the patterns in the RMX Chaos generator. But most of them work fine. The Metamorphosis SAGE Xpander is in the same vein as the RMX core library, and pretty

much everything you can say about one applies to the other. These grooves are extremely creative, they lead you to ideas, and they’re unique. Backbeat is a collection of traditional drum beats, but they’re really good ones. It has a wide selection of grooves on many different types of kits, from standard studio drum sets to unmuffled jazz kits. This library’s content isn’t ground-breaking, but it’s really well done; anyone who’s looking for totally appropriate, wellplayed and recorded drum loops should be very happy with it. You could describe Liquid Grooves as the acoustic equivalent of Metamorphosis. Its grooves feature percussion, but it’s usually combined with a light drum set, in some cases played with what sound like nylon brushes. Liquid Grooves is just outstanding. Burning Grooves features drummer Abe Laboriel, Jr. These beats use standard drum set, but the effects and sounds are all different and all very interesting. Laboriel’s patterns are all

qute original, and due to dynamics, even the really busy ones groove like crazy. If you never used it, this Xpander would be worth having just as a source for drum programming ideas. The last Xpander, Retro Funk, takes you right back to the ‘70s. These beats completely funk out in the Harvey Mason/Mike Clark Headhunters East Bay vein. Some of the patterns groove so well it drives you crazy! Funky funky.

Conclusion It’s one thing for the content to be exceptional, to say nothing of being meticulously recorded, edited, and processed. But the Stylus RMX virtual instrument really revolutionizes the way you can work with loops. And on top of that it’s quick and easy to use. As a reformed drummer, I personally never really liked working with rhythm loops before. This instrument has changed my mind. Stylus RMX is one of the hottest virtual instruments around right now, and deservedly so.

MOTU PCI-424 Card and CueMix Console Routing with Third-party Software

OTU’s popular PCI-424 card features an onboard digital monitor mixer that can be as large as 96 x 96, depending on the audio interfaces connected to it (MOTU makes several). This mixer, which has virtually no latency, is controlled by the CueMix Console software shown here. There’s some confusion about how the CueMix mixer integrates with third-party audio software. If you don’t understand the routing, it’s easy to find yourself monitoring a signal with normal computer latency, or to hear two signals that chorus/comb-filter together. The confusion is that there are really two potential paths to your speakers: through the CueMix Console monitor mixer on the PCI-424 card, and/or through your DAW software. Here’s how it works. The physical input signal goes to the Mute and Input Trim controls at the top of the mixer first; if it’s muted there, you will hear nothing. After the input Mute and Trim, the signal gets split into two. One path goes on to whatever software is running on the computer (whether MOTU’s or someone else’s); the other path goes to the latency-free CueMix monitor mixer. All the onscreen faders and knobs on the light grey background control the CueMix monitor mixer. Your DAW software is blissfully ignorant of the CueMix mixer, so if you have software monitoring turned on, you’ll hear the input signal twice: once directly through the CueMix mixer, and again coming back out the computer (with added latency). The combination of the two signals is what

random

tip

2 To Computer

Input Signal

3 From Computer

causes the chorused sound we don’t want. You must mute the signal here or in your DAW’s channel strip—pick one. There are two sets of meters on the CueMix mixer, marked Mix and Out. The Mix meters show the output of the CueMix Console mixer; the Out meters show what’s

in the Mix meters summed with whatever’s coming from the computer—which includes local virtual instrument outputs, recorded tracks, system sounds, and so on. There’s no level control here for the computer audio, and it would be confusing if there were; you adjust it in your DAW.

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Klaus

Badelt

From wall-sized rack to a single PowerMac G5: one of Hollywood’s hottest young composers traces the evolution of his workflow. laus Badelt is an extremely

busy composer who’s written

music for over 25 major Hollywood films and counting. His credits include “Pirates of the Caribbean” (2003), Werner Herzog’s “Invincible” (2001), “The Recruit” (2003), “The

Walk into Badelt’s old studio at the place formerly known as Media Ventures (he’s since moved), and one of the first things you notice is—without exaggeration—a wall-length, headheight rack full of hardware units. There are more samplers, effects, and outboard equipment than you’ve ever seen. And not a single one of those boxes has been powered on for a couple of years. The only unit with lights on is a solitary MOTU 2408 Mk III audio interface at the very

Time Machine” (2002), “K-19: The Widowmaker” (2002), “Ned Kelly” (2003), “Basic” (2003), The In-laws” (2003), “Catwoman” (2004), and most recently “Constantine” (2005). At the moment he’s working on the score for “Curious George”…as well as other films in the mill. Badelt is equally adept writing for orchestra and working with softsynths and samplers, and he has an inside reputation for being one of the best MIDI sequencer programmers in the world. His score to “The Recruit,” for instance, makes extensive and intricate use of loops, and the “Constantine” score is heavily electronic. Both are extremely effective.

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end of the rack. It’s connected to a PowerMac G5 and five (actually 5.1) speakers. Without prompting, Badelt traces the evolution of his working set-up. Badelt: The very first requirement for film scoring is that you need to be flexible. The film changes, opinions change. Your mind changes. You’re always rewriting or polishing until the end. At the beginning it was tons of E-mu samplers—16 E-mus, 18 or 20 Rolands, going into

i n t e r v i e w VI a handful of 02Rs (Yamaha digital mixers), and we could recall that set-up with the O2R. We thought that was really cool. The only thing is that you have to define your pallet at the beginning of the project. You can’t go back and change sounds once they’ve been recorded as audio, so we had to think in advance what would be good for the film And then we’d load all these sounds. I had to have so many samplers because I had to have all the sounds ready all the time. It would take too long to go load another sound for another cue, and too many errors would creep in. If you had a synthesizer, you had to store the sounds... It was a complete nightmare. Whenever you hit Play when the director was there...well, I would record it to audio to play for the director. Too many parameters, too many tracks— the pallet would be 300, 400 tracks. So it was good but not great. The next generation started when GigaStudio came out. So there you could get rid of all the samplers, you had much better sounds, it would all be networked…suddenly you were in the computer environment, so you would not have to look for that E-mu sound from that film we did five years ago, we would have the library, we would have storage capability…so that was pretty cool. At that stage I got rid of all the 02Rs and bought Pro Tools systems to do live mixing. But Pro Tools wasn’t really built to do live mixing then, at least not for our purposes.

VI: What were the problems—not enough inputs? Badelt: At the beginning we didn’t have enough inputs, so I used a Creamware Pulsar to act like a recallable patchbay, ADAT lightpipe to

then recording that—you’re either mixing live or recording. In other words, what they call Aux Inputs—live inputs—can not be easily converted to a track where you can actually record it. So you’re always working with tracks that are in Record Enable, and then there’s no

The very first requirement for film scoring is that you need to be flexible. The film changes, opinions change. Your mind changes. You’re always rewriting or polishing until the end. ADAT lightpipe, from 90-something channels into 48. So I would save the patch for each cue. It was a great system at the time, because I could finally take my tracks I was writing and record them dry. I would record my live feed as audio, go to a mix studio, and be able to not only mix my stuff but change what I mixed before. To be able to go back one generation, I would record each group of tracks of audio with all the effects wet. You know, you were locked into what you had. But you could deliver all the tracks to the mix studio, including all the plug-ins and everything. Still, there were some practical issues. Pro Tools doesn’t provide a way of mixing live and

automation in Record Enable. These little things then turn into it not being ideal for our application. You can get around this by bussing, but I may have 60 cues in a film, and for each cue I would have to set this up. And then you still had only a limited number of inputs, and each individual sound had its input. So we often had to go back to the writing room and record if we wanted to make a hi-hat louder. And still you have to load your pallet. I was using Orion (Synapse Software); you could run all the internal instruments and VST instruments as well—it was my VST instrument host. We couldn’t use Steinberg V-Stack because it wasn’t possible to slave delays and other timebased effects to MIDI Time Code. So the next generation. I just wanted to be more flexible again. I want to put a delay on just one string track—anything with mixing and writing. One day Emagic came by, we were working on “Pirates of the Carribbean,” and they said, “What do you think of our EXS sampler?” And I was laughing at them! I had 14 GigaStudios, each of them had four MIDI ports. They said, “You can load 64 samplers.” That’s not even one Giga! But after they were gone, we thought about it and analyzed our tracks a little bit, and found out that the average cue has 40 tracks maximum. So the actual used instruments at one time are not that many. I think “Catwoman” was the first project that was

Fig 1: Badelt’s main Logic screen for “Constantine.”

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i n t e r v i e w VI done completely in Logic, including mixing the orchestra and delivering the final master. VI: What sorts of things were you requesting from Emagic [as Logic’s developer was known prior to being purchased by Apple] before they showed you EXS? Badelt: The funny thing was when they came, I remember we said, “Come have a look what we’re doing here,” and they came to the mixing room and saw us working there with a few Logic Controls [MIDI hardware control surfaces] and mixing the orchestra. You could see in their faces: whoa! “You guys are really doing this, right?!” And so they looked at each other and said, “Oh shit!” because we were pushing the envelope so much that it would reach the limits. But it worked. VI: What was missing?

Badelt: For instance loading up complete track set-ups—channel strips and data. You couldn’t really copy and paste between two Logic Songs. These things were put in since then. They’re still working on being able to copy parts of Environments. They’ve now put in the Apple Loops capability. For our purposes it’s better to have it all integrated in one. Now loops are just another audio track. “Constantine” was to a certain degree based on Apple Loop tracks. We recorded an electric cello player, and we made lots of loops that could go in other cues. VI: How are you able to get by with so many fewer sounds loaded than you had available in Giga? Badelt: The Giga revolution was great. Sounds were unlimited, and the company was very collaborative—developer Jim Van Buskirk

put in a lot of work. We had maybe 100 GigaStudios at our place altogether, and I was using maybe 14. But still we had the issue that I couldn’t recall cues; you can’t load 14 samplers, it would take too long. So after we realized we only used 40 tracks in a cue, we gave EXS a try. You never learn if you don’t use it on a project. So I wrote a small film—there was not that much orchestra, not a big action adventure, and I tried to do it with everything coming from a dual G5. I kept a few Gigas running in the back just in case, but it turned out that it worked amazingly well. It’s very efficient and integrated with Logic. For example, when you use the same sounds in five cues, EXS doesn’t load the sounds five times, it loads them once. They in memory. We converted our whole library into EXS— with all our hundreds of thousands of samples. It’s not done yet! But most of it is. It’s just amazing how your work will change this way. Okay, I have an idea, next track I need…let’s try this guitar. You open a track, plug in the guitar—which can be very fast now. No, let’s try a different instrument, a synth sound. Let’s add some ambience. Then the next time you use electric guitar with some ambience simulation. And you could do that between each cue, with full recall. And the mixing is in the writing now. So the fader you have when you write is the channel strip of the mix. VI: Do you have a separate person mixing now? Badelt: Yes. VI: You give him or her the Logic file.

Fig 2: Klaus Badelt’s Logic Environment for “Pirates of the Caribbean.” Each of the icons in this sound pallet points to an independent external MIDI port and channel, in this case on one of about 18 fully-loaded GigaStudio machines. Command/clicking on an icon instantly assigns it to the currently selected track in the Arrange window (not shown here), making it easy to manage a vast template without pre-assigning every single sound to a track. Badelt now uses a single G5 instead, which doesn’t allow you to work this way. Having everything on one machine has the advantages Badelt outlines in this interview, and he is able to pull it off for a variety of unique reasons. For example, he only uses a private orchestral library that loads very quickly; he has programmers on hand converting everything to Logic’s EXS24 format; and all his cues get sent to an engineer for mixing, so he doesn’t need to run a heavy load of processing plug-ins. But before you try this at home, kids, note that you can only keep a fraction of the number of sounds and instruments cued up and ready to play. While you can run a lot on a top-end machine today, many if not most professinoal composers working with large sample libraries and processor-intensive virtual instruments prefer to split the load across at least a couple of computers.

Badelt: That’s what we do, right. We have the same Logic set-up in the mixing room. He opens the sounds. And that’s another big key: you don’t start the mix from scratch, you open what I did. And of course I’ve spent some time with the mix already at that point. He can do what an engineer does well—eq and so on—but it’s based on what I did. It used to take a long time just to get it the point where I had it. VI: You normally use about 40 tracks, but you must have way more than that in your template loaded up and ready to play. Badelt: Oh, hundreds. The palette in Logic (in the Environment) is by definition external instruments. [See Fig. 2] But now I’m using internal instruments and I only load the ones I need. The Arrange page before was hundreds of tracks, and only some of them were used. The cool thing in Logic is that you can configure it any way you want. No two Logic users set it up the same way—if I go to someone who uses it, they constantly surprise me with what they’re doing with it. (CONTINUED ON PAGE 60)

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Ableton Operator add-on FM Sythesizer for Live

By Lee Sherman

Price: $149

Company: www.ableton.com. Distributed in U.S. by M-Audio, 5795 Martin Rd., Irwindale, CA 91706. 626/633-9050.

Platforms: Runs only inside Ableton Live, on Mac or Windows.

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perator is an add-on FM synthesizer virtual instrument for Live, Ableton’s unique and popular Mac and Windows real-time music production program. It works only within Live, but you can stream Live into other sequencing programs via ReWire (please see “The World of Softsynths and Samplers” elsewhere in this issue for an explanation of ReWire). For those unfamiliar with Live, it is a specialized MIDI and digital audio sequencer that’s set up for triggering and processing patterns and loops in real time. It can also record. Operator uses FM—frequency modulation— synthesis, which generates complicated waveforms by using oscillating sine waves from one sine wave generator to modulate the signal coming out of another sine wave generator. A “circuit” with two or more sine wave building blocks is called an operator, hence the name. The famous Yamaha DX-7 from 1983 used FM synthesis, and it was notoriously difficult to

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program. Since then we’ve seen several attempts to make FM easier to program, notably Native Instruments’ FM7, but no one has really rethought the paradigm while still delivering the goods—until now. Ableton’s implementation is so easy even a child could grasp it. Picture Operator as a DX-7 with onscreen knobs that invite you to dive in (as opposed to membrane switches and a couple of sliders that accessed menu upon menu). FM synthesis is represented graphically with a series of colored building blocks that clearly indicate which oscillator is modulating the other. Operator takes full advantage of Live’s hands-on real-time control, low latency, and envelope-based automation features. At its simplest level, Operator hides the complexity of FM programming behind a pair of macro controls, Time and Tone, that can be immediately called into play to modify a sound’s characteristics in real time. Its straightforward layout provides controls for the most

immediate FM parameters up-front, while allowing single click access to its graphical envelope editors. This synthesizer can produce the full range of FM sounds, from percussive to bell-tones, punchy bass, harmonically rich pads, and realistic electric pianos. It sounds absolutely fantastic, and if you just buy it for quick access to some killer presets within Live, you’ll love it. But that would be missing the point.

Hearing carriers Operator’s voice architecture consists of four oscillators, labeled A – D, plus a resonant multimode filter (multimode means it can function

We’ve seen several attempts to make FM easier to program, but no one has really rethought the paradigm while still delivering the goods—until now. as a lowpass, bandpass, highpass, or notch filter). There’s a wide variety of waveforms to choose from, including variations of sine waveforms that allow it to mimic the “whooshing” aliasing artifacts of hardware FM synthesizers. (The DX-7 had six operators, but it had no alternative waveforms or filtering.) Instead of trying to minimize aliasing as most virtual analog synths do, Ableton has gone to great lengths to harness it so that it can be used to musical advantage. Perhaps this is not surprising, given that the software’s conceptualist was the experimental musician Robert Henke, also known as Monolake. Operator even lets you control the level of aliasing. By assigning this control to a MIDI controller, you can go from mellow to nasty at the turn of a knob. Operator is primarily an FM synth, but it does include the subtractive algorithms and waveforms (saw, square, triangle, noise) necessary to provide a reasonable simulation of analog sounds, adding tremendously to the software’s versatility.

The envelope, please There’s more secret sauce. Each of the four oscillators has its own envelope, as do the LFO, filter, and pitch. Interestingly, envelopes can not only hard-sync to the song’s tempo, they can be made to loop completely out of sync, repeat individually at their own rate, or repeat at the rate specified, allowing for some incredibly complex textures. It’s great to see Ableton applying some of its innovative thinking to instrument design.

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f e a t u r e VI

Rigors of the Road Rig Despite the bum rap, softwarebased musical instruments can be reliable on stage—as long as you put together a reliable rig. Part 1: the computer itself. By Bruce Richardson

othing has had such an enormous impact on music production in recent years as software synthesis and sampling. These tools are stun-

ning in their scope and sound. Yet this software playground rarely leaves the studio, and dedicated hardware keyboards continue to reign supreme in live settings. In 2005, does this really make any sense? Hardly. Software offers superior flexibility, configuration, and creative potential on the stage. While the variables introduced in a computing environment appear staggering compared to the relative simplicity of a single-purpose synth or sampler, they are not insurmountable—and more musicians are discovering the pleasures of a computer-based performance rig every day. Digital stagecraft has certainly arrived. But how does one get there? How do we set up a rig we can really depend upon? Surprisingly, it’s not so hard, if we consider the challenges of the stage when designing our systems. 44

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Solidarity The first step in building a dependable rig is outfitting the computer itself. Few mainstream, off-the-shelf computers are ideal for this. They have far too many convenience features for home and office that lie at cross-purposes to the lean, mean, mission-critical systems one needs for stage work. With Windows machines, that leaves us with two primary choices: the do-it-yourself route, or purchasing a turnkey machine put together by specialists for the purpose. (Please see the sidebars to this article for other alternatives.) It is of infinite value to know exactly what is in a machine, where these parts can be purchased, and how to replace them in a pinch. I personally prefer to put machines together myself for that reason. But even if you choose a turnkey machine, knowing what you need and want is crucial to satisfaction, from the power cord all the way to the audio outputs. I avoid laptops. Their portability is attractive, but in many ways that convenience is only skin deep. Sample-based systems want fast, huge disk capacities. There’s no room in a laptop for that, so external drives tend to be a must. Laptops survive drops poorly. They’re inconvenient to place onstage, since monitor and keyboard input are fixed to one another. And by the time you’ve strung the peripherals you’ll need to make a laptop a stageworthy companion, is it really more convenient? Each cable becomes a potential failure point, not to mention a trip hazard. It’s a personal decision, but for the purposes of this article I will focus on what I consider to be a rig that is idealized for roadability, and that’s a rackmount system. Components Since Macs are sold pre-configured (you can’t put them together by selecting a case, motherboard, power supply, fans, etc.), most but not all of this section applies to Windows machines only. Please see the sidebar for more about making Macs roadworthy. Starting with the computer case, I recommend a 4-space rackmount. One can get smaller units, even single-space, but the tradeoffs start to become similar to those associated with laptops. A 4-space case allows a full complement of mainstream hardware in standard sizes. Unless they’re very deep, smaller cases may limit choices in one way or another, which usually equals more expensive parts with less capability. Why rackmount in general? Why not a nice tower case? Rackmounts certainly do cost more, but I have found the cost offset by the convenience and sturdiness. On the road, a rackmount can screw into a caster rack and transport with ease. For local gigging I don’t rack my CPU unit, because I use it in the studio as well. Instead, I have a flight case for it. In the studio, it lives in a Raxxess Isoraxx on a rack-shelf unit. When I pack for the gig, I just undo the cabling, slide out the computer, and put it into its flight case. This is convenient for me. I don’t have to

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f e a t u r e VI unbolt the computer in and out of a rack, and in practice I have not found a compelling reason that it makes a difference on stage. When I set up, I place the computer on the stage, and the rack either atop it or in close proximity. It only becomes worth it for me to rack the computer when someone else is doing the lifting and toting. In fact, I have been known to forego the flight case on a gig with easy ins and outs. The rackmount case in and of itself provides easy carrying and good protection. The biggest dangers in going sans-case are reduced shock protection in the event of a bump or drop, and the possibility of damaging rear connectors. My last vehicle was not quite large enough to accommodate the CPU’s flight case along with everything else, but I have since solved that problem (and must offer the

hole. This allows you to speed down the fans in cool and quiet environments, while providing as much cooling as possible when the heat’s on. Cases with big, grille-covered fan holes cool better than the “poked hole” variety. Anything that impedes the flow of air reduces cooling efficiency. I don’t recommend cooling exotica involving liquids for a road machine, for all the obvious reasons. Heatpipe solutions will work in some environments, but forced air works most dependably. I favor CPU cooling solutions based on large fans and large heatsinks, since these are usually more dependable than smaller, higher RPM fans (as well as quieter). Use wire ties to secure all internal cabling to anchor points, taking care not to kink or bind cabling. But in doing so, leave slack in critical

The most redundancy one can afford is always the right answer. observation that the Honda Element seems to have been invented just for musicians). So for extra insurance I now use the case most of the time. Moving on to the computer’s stuffing, I believe it is worth spending a little more money to buy locally if at all possible. Choosing the guts is all about risk reduction— not to say that the risk of problems is high in a well-tested system. In practice, I have found it to be low. Nevertheless, less risk is better. The parts likely to be stocked in your local computer superstore will be stocked in other communities. Some of the more esoteric parts available online are not. Should you encounter failure on the road, or at the eleventh hour in a local club, being able to dispatch a loyal friend to fetch a part is advantageous. Even if it’s same-brand-different-part, you increase the odds of quick recovery. The drivers may be similar, even the same. System impact will likely be similar. Power supplies should be beefy and quiet. Don’t skimp, because extra hard drives, audio hardware, etc. all put some load on the system. These days, 400 watts is a reasonable minimum. Put a quiet, variable RPM fan in every

locations. In particular, you’ll want to be able to get the drive cage out and accessible without taking loose a lot of wiring. Secure the bulk of the cabling’s weight so that in cartage, chances of cables shaking loose are minimized. Always facilitate the maximum airflow through the case. Stages can get very hot, as can outdoor venues, so make certain that the CPU, drives, and other hot-running components get the maximum airflow possible. As with anything else, balance is the key. Tie down just enough to make things secure and open, but not enough to prevent quick maintenance.

Audio hardware My live rack currently contains a power/light unit, a multieffects hardware unit, a pitch-toMIDI converter, a 12-channel mixer, and an Echo Layla 20 audio interface. The choice of the Layla 20 lies partly in its good working relationship with my choices of software for live use (Tascam GigaStudio and Native Instruments Reactor), and partly in the fact that I already own three of them. Because I have upgraded the studio systems to newer models, I am able to keep a Layla 20 in both

4. A face but no head: Yet another alternative is Muse Research’s Receptor Open Labs’ Neko incorprates a keyboard, and a computer with proprietary software. (www.museresearch.com) is specially-designed Windows computer that replaces the monitor and keyboard with an LED screen and front-panel knobs and buttons.

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Macs on stage: The most common way to road-proof a Mac is simply to put it in a flight case, wedged into the foam lining and held in place by friction. As long as the front and rear lids are removed while it’s on, cooling is not a problem. If you prefer to rack-mount a Mac, Marathon Computer (www.Marathoncomputer.com) has attachments for blue and white G3s, G4s, and G5s.

the live rack and in the studio. This has been ideal, and it leaves me with two extra PCI-cards and an extra breakout box in case one goes bad. Here again, we’re leveraging expense against failure and that equation is personal. The most redundancy one can afford is always the right answer. When choosing your audio interface, no matter what the brand, ask yourself a series of questions. Foremost, does it work well with your choice of software? How many discrete outputs do you desire from the software you’re using? Do you want to route audio into the system for effects purposes? Do you want it to also be your MIDI interface? Do you desire any additional digital I/O or clocking functions? I use both of my main performance applications to process microphone input, and I sometimes route multiple output pairs, hence my personal choice serves me very well. Internet discussion groups are a good source for information about putting together equipment from different manufacturers, since there’s a good chance somebody is using the same combination. Keep your video interface simple. No music software requires particularly cutting edge graphics performance, and high-performance video cards generate lots of heat. Some need fans, and these small, high-RPM fans are very prone to failure. The cards of a generation or three back are more than sufficient. Built-in motherboard video is invariably good enough, unless you run into a problem with some specific manufacturer vs. your software choices. Again, the internet is chock-full of discussion on which

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f e a t u r e VI hardware works well with what. Do your homework, and you will be happier. I use older NVidia-based cards in my machines, and they have worked well for me.

Monitors My current monitor is a generic 15” LCD. No need to spend big bucks here, so look for bargains. CRT monitors are cheaper, but their bulk and weight makes them inconvenient to move and awkward to place on stage. Anything can happen on the bandstand. The bigger they are, the harder they fall. And the dorkier they look. I have a flight case for this monitor, which is handily enough the same exact case I use to transport the CPU unit. Put directional arrows on the case so that it will be packed properly in case it’s not you doing the packing. Protect the face of the monitor with a piece of foam if you are storing other items in the case that can move around. LCD screens are not fragile, but they are vulnerable to pressure and sharpobject injury. I use an amp stand to hold it up in live use, but one can also get quite creative with custom

Pre-configured computers: Several companies put together Windows computers tested and configured for musical applications, often with specific software—e.g. GigaStudio or Kontakt—and sometimes even specific sample libraries in mind. The BYC VisionDAW (www.visiondaw.com) is an example of one we’ve had excellent luck with.

The hood A great monitor hood can be fashioned out of corrugated plastic signboard, black being the most opaque, but any color will reduce direct glare. Cut a sheet of signboard to the total of your monitor’s top, left, and right side dimensions. Make the cut such that the corrugations in the signboard are running perpendicular to the screen—we’ll use them as bending guides for good sharp corners. Leave about

I avoid laptops. Their portability is attractive, but in many ways that convenience is only skin deep. VESA mount solutions. There are many nice mounts on the market, and if your appetite runs to the exotic, any machine shop worth its salt could manufacture whatever mount you desire. I am currently investigating a mic stand mount. Consider the possibility of visibility-reducing glare before you actually start gigging with your new rig. A low-key club date is likely no problem. But if you work with more elaborate stage lighting, you may need a monitor hood. I have been reduced to shrouding my head with a towel in the Texas heat, like a large-format photographer, on an outdoor gig where the sun was setting at my back. Anticipate your situation as much as possible. I looked very dorky that day.

12” of signboard in the dimension parallel to the interior ribs; we’ll trim it later. Now apply good quality Velcro to the factory edge (the one perpendicular to the ribs). Apply the opposite-sex velcro to those edges of the monitor, and starting on the left side, attach the signboard, letting the excess stick straight up. Bend the signboard carefully on the corner and stick down the top dimension, then again to attach the right side. When finished, you’ll have a boxy hood. Now determine the depth you’d like your hood to be, remove it, and use a straightedge and a box-knife to trim your hood to the desired overhang. Angle the sides, if desired, since lighting tends to mostly shine downwards. You want the least amount of overhang that will do the job. It’s a cheap solution, so

Muse Research’s Receptor has a proprietary operating system, but it hosts standard VST instruments and plug-ins. You can connect a keyboard and monitor by ethernet when you’re back in the studio.

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you can always build a couple. They’ll fold nice and flat in your monitor case.

Headlessness All this said, there are advocates of essentially black-box solutions, sans monitors and even input devices like keyboards and mice. One can set up software to autoload, and create a machine that simply turns on and off. To me, the essence of digital stagecraft is using every possible tool the medium provides, and the visual component is the primary interface of modern software. To deprive myself of this would be unthinkable. The simplest example is the best for this: I use a screen-manipulated Theremin emulation in one of my Reaktor setups. It’s great fun, and without the visual component, it would be difficult to set up for live use. I want the ability to manipulate the environment in ways I may not have pre-conceived. And the thought of flying blind frightens me. Keyboard and mouse On stage, the trackball is a far more useful input device than a mouse, since it can be Velcroed to a static point. I also Velcro the computer’s keyboard to my 88-key controller. This has worked well for me. I have seen people adapt drawers, etc., to racks or stands. The important thing is to find a location that works for you, and to make it secure for performance. Conclusions Once you have a good dependable computer system built and have adapted it to the stage, what’s left is the thoroughly fun part: picking the software tools you’ll use to perform, and the keyboards and other controllers you’ll use to make them go. The great thing about a computer on stage is that there are really no fixed limits on what you can hook up as an input device, or how many input devices you can string up to create your own personal jungle gym of musical adventure. Bruce Richardson is a Dallas-based composer, producer, and performer. He has scored hundreds of productions in television, film, and theater; produced world-class artists; and has appeared as a sideman on stages and in arenas all over the world.

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TASCAM GigaStudio:

Taking the GigaPulse Routing instruments through GigaPulse, the built-in convolution processor in GigaStudio 3* isn’t as convoluted as it seems. The man who wrote most of the GigaStudio 3 manual explains. By Dave Govett

Convolution processing is not just limited to rooms. GigaPulse comes with impulses of a variety of microphones, DSP gear, instrument body resonance, and more. You can also create your own impulses and do all kinds of crazy things (see “Getting giggy with Giga”).

You could say I’m the first user of Giga technology, since I persuaded the guys at NemeSys (the original developer before TASCAM acquired the company) to let me beta test it. They ended up hiring me, and I did a few years of tech support, making training videos, documentation, doing clinics, and I laid out the first couple of dozen Giga-format sample libraries. Now I do contract work on Giga for TASCAM—especially documentation.

GigaStudio was the first sampler to have an impulse processor built in, but the idea is catching on in other products, and convolution technology is the cutting edge of sampling. A good example of it is in the upcoming GigaViolin. This library will offer one set of samples—dry, body-less strings—and a dozen or so violin bodies to choose from.

hat a timely concept for a music industry magazine! I’m very glad to be here dishing out info on GigaStudio, a subject dear to my heart.

For this article I’m going to go over the fantastic GigaPulse convolution processor included with GigaStudio 3.0, sharing some recommendations for getting the best sound and most efficiency out of it. Please be sure to check out the audio examples on the Virtual Instruments website (www.VitualInstrumentsmag.com). GigaPulse interface

You can then enable any or all of the microphone models in GigaPulse and even try different pickup patterns and filters. After that, all this can be placed in a room. But for this article we’re going to concentrate on the basics of using GigaPulse for the room and hall environments. Advance placement There are two main ways to assign instrument positions in a GigaPulse Hall Environment: Point Source, and Stereo Spread - Multiple Position.

Point Source:

Stereo Spread-Multiple Position:

Setting up point source

In a nutshell, convolution technology allows environments—or anything else that affects sound—to be sampled, and its attributes applied to other sounds or instruments. You can place a dry recorded instrument into any impulsed room or hall, and it will sound virtually the same as if it had actually been recorded in that space. (*GigaPulse will also be available as a stand-alone VST plug-in in the near future.)

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The first thing to do is select an instrument and load it into Giga. For this example I’m going to use the new French-style oboe from VSL (Vienna Symphonic Library), included with Giga. These samples were recorded fairly dry to allow for good control over how much ambience you want, but they still have a little bit of air. That allows the sound to breathe a bit and gives it some authentic spacial cues. We’ll start by loading a mono version of the oboe into Giga on MIDI channel 1, Port 1, to demonstrate the point source method of using

GigaPulse. If you listen to the audio examples on the site, I think you’ll be amazed at how good a mono instrument can sound. Please check out Oboe-Mono-Dry.mp3 for reference.

The Medium Hall program

The MIDI Channel output of the oboe is assigned to DSP channels 1 & 2 in Giga. MIDI Channel output assigned to DSP mixer channels 1&2

Go to the DSP Mixer Channel pair 1 & 2 and expand it by clicking on the little triangle at the top right corner of the channel. Use the channel insert to open up the GigaPulse. While te uses more processing power than sharing the processor, I prefer to use inserts for GigaPulse rooms instead of Auxiliary Sends. This puts 100% of the signal in the room the way it would happen in real life. Opening a GigaPulse

The hall was sampled with five mics and 18 speakers. You can simply choose any of the 18 positions on the stage with the mouse.

That brings up the GigaPulse Interface, which should look like this: The default GigaPulse Interface

osition-2 (Left-Close)

Position-8 (Right-Close)

Position-11 (Left-Far)

Position-17 (Right-Far)

Be sure to check out the audio examples of various positions on the stage using the Mono Oboe.

For this example, I’m using the “Medium Hall.” I particularly like this hall because it’s fairly ambient without getting too reverberant.

Oboe-Mono-GP-Pos-2.mp3

left up close

Oboe-Mono-GP-Pos-8.mp3

right up close

Oboe-Mono-GP-Pos-11.mp3

left farther Back

Oboe-Mono-GP-Pos-17.mp3

right farther back

Selecting the Medium Hall

The two arcs of positions on the stage give a nice left-to-right spread, as well as some depth positioning—which is where impulse technology really shines compared to traditional reverb. It’s the front-to-back image that really creates the illusion of space, more than the left-to-right position. Perspective controller If the front or back positions are too close or far for an instrument, you can use the Perspective slider to loosen or tighten the sound (up to the extremes of the recorded positions). This is another reason to use GigaPulse as an insert instead of a shared resource on an aux send.

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The Perspective controller

Channel assignments Now the question comes up about how to assign the various mics to the rear or center channels of the system. That is where the “Mic to Mixer Routing” matrix comes in.

Moving the slider to the left tightens and brings the sound closer; moving it right loosens and pushes the sound farther away.

First, set the drop-down to “7-Channel” mode for surround sound. From top to bottom we have the mic positions; from left to right we have the channel assignments, which are the seven adjacent channels to the starting channel.

You can start with a close position and make it wetter, or start with a far position and make it drier using the Perspective slider. Always try both options to see which one works best. Enabling more mic positions for surround So far we’ve only dealt with the default 2-mic setup, however these halls have been “impulsed” in 7-channel surround. To enable any of these channels, simply click on the mic position with the mouse.

5 Mic positions enabled

For example, if we insert a GigaPulse on input channels 1 & 2, then the channels in this matrix will be channels 1 through 7; insert the GigaPulse on inputs channels 15 & 16, and these channels will be channels 15-21. Simply click on the lights to the right of the mix positions to assign any mic position to any of these seven adjacent channels. Then you can go to each of those adjacent channels and assign them to any available physical outputs, or even to a group channel.

All 7 Mic positions enabled

CPU usage This is done at the bottom of each channel. Keep in mind that each extra channel uses additional CPU processing. However, the Giga engineers are well known for making their programs efficient; GigaPulse is fairly thrifty with the processor considering what it is accomplishing. On my 2.8Ghz Pentium 4 machine, I find that each mic position in the medium hall adds an extra 3% to the CPU usage. If I were to dedicate this machine to nothing but GigaPulse, I could get four instances of five channels each—and that’s in full 5-speaker surround, all at the same time on a single machine. (80% of the CPU is being used at that point.)

What is convolution and why is it the latest craze?

hile there’s still a place for the conventional reverb units we’ve been using for the past 25 years, convolution processing has brought a new level of realism to reverb and other kinds of modeling. Convolution actually allows you to sample anything that has an effect on sound—a room or hall, a guitar amp, mic, musical instrument body…anything. Until a few years ago, real-time convolution processing was only available in very high-end hardware units from Sony and Yamaha. Then a few years ago a Dutch company called Audio Ease introduced a software unit called Altiverb, so named because it uses the AltiVec coprocessor on the chips in all G4 and G5 Macs. Altiverb is still an extremely viable contender, but now there are several others as well—in fact Logic Audio Pro even comes with one (Space Designer). Following Gigapulse’s lead, convolution processing is now finding its way into virtual instruments (Native Instruments Kontakt 2, MOTU’s forthcoming Symphonic Instrument, and others will certainly follow). This is an exciting development for sampling, not just for reverb but because of things like instrument body modeling (see text).

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Assigning physical outputs

Now we have Mixer channels 1-8 assigned to hardware outputs 1-8. This gives us seven discrete physical outputs that can then be routed to the speakers. If we use a group channel, then that group channel can be assigned to a physical output as well. So the signal path starts at the MIDI channel, works its way through the mixer, and goes on to the sound card physical outputs, or to group channels first and then the physical outputs. The MIDI channels are assigned to DSP Mixer Channels in the MIDI Channel & Port Screen. Mixer channel is assigned to physical output Or the Mixer channel is assigned to group channel. Then the Group is assigned to the physical outputs.

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Oboe-Stereo-GP-Close-Center.mp3 shows what this sounds like. To adjust the left right position of the oboe, simply pan it using the MIDI or volume panning. These examples show what this sounds like: Oboe-Stereo-GP-Close-Left.mp3

Stereo spread-multiple position

Oboe-Stereo-GP-Close-Right.mp3

To really hear a stereo sample sparkle and shine, try the multiple select, stereo spread method. This lets you keep the air and the positional cues from the stereo samples as you place them in the GigaPulse environment.

The same placement idea applies to the rear stage positions. To position the sound farther back, simply assign the left mic to stage position 11 and the right mic to stage position 17.

You can still use the point source method on stereo samples if you wish, but I find that the stereo samples really benefit with this method. This is especially true of ensemble instruments like a strings, which have a wider stage position than a point-source instrument like a solo woodwind.

These examples show what it sounds like farther back with different panning:

First, load up the stereo version of the oboe. Check out the example: Oboe-Stereo-Dry.mp3 for reference.

Oboe-Stereo-GP-FarCenter.mp3

Instead of just clicking on the stage positions, this time we’ll use the Control key to lock the pairs of positions to pairs of microphones. Choosing mic positions that are farther apart widens the stereo spread, and conversely closer mics narrow it. To do this, hold down the Control key and click on the left-front mic position. Then continue holding down Control and click on stage position 2. Do the same thing for the right mic position and assign it to stage position 6. The colors of the positions change to show which mic is assigned to which stage position.

Oboe-Stereo-GP-Far-Left.mp3 Oboe-Stereo-GP-Far-Right.mp3 You can widen or narrow the stereo spread by using different stage positions. It’s also possible to be creative and combine left close with far right or any combination. However, a symmetrical setup is right for most situations.

Now when we play the oboe it has a more live stereo sound in the center.

Narrow Spread:

Close & Far Spread: (CONTINUED ON PAGE 62)

How convolution works:

f you play a full-bandwidth, very sharp sound through a speaker in a room (or through anything else) and sample the reverb decay, you get an impulse response of the room. In other words, you get a recording of the room’s audio signature— its response to being excited at every frequency. Impulses are theoretical, infinitely short noise spikes that contain all the audible frequencies. In reality, sounds that are similar to impulses (such as starter pistols) are hard for speakers to reproduce, so for acoustic spaces it’s better to to use sine wave sweeps that are time-compressed back into a single spike after recording; Audio Ease’s Altiverb convolution processor actually comes with a sweep-generating/recording/processing utility for making your own samples this way. There are some differences in the convolution playback engines on the market, but

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the quality of the impulse response samples themselves is by far the greatest factor in the quality of the sound. Impulse responses are simply audio files, of course. Here’s a good way to picture how convolution processing works; special thanks to Arjen of Audio Ease for his help explaining this rather difficult concept. Sampling—digital recording—works by taking 44,100 “readings” of the source per second (or 48,000, etc., depending on the sampling rate). Each one of those readings, or samples, represents a voltage; the continuous waveform you hear on playback is produced by “drawing a line” to connect each one of those voltage numbers on an imaginary bar graph. What convolution does is take each one of the samples in the source recording and scale it—that is, multiply it—relative to the value of each successive sample in the impulse response file. So each sample in

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effect gets its own reverb. If the source sample is short, it’s repeated to make up for it not being as long as the impulse response, or vice versa (it makes no difference because 1x2 is the same as 2x1). Because each sample has to be processed 44,100 times every second (depending on the sample rate), it’s easy to understand why convolution processing requires a lot of CPU. And the longer the impulse response is, the more CPU it takes, since there are more samples to process. Mathematically, convolution models exactly what would happen if you played an instrument in the room in which you recorded the impulse response; in theory there’s no difference in the result. Reality is a little more complicated, but convolution reverbs are stunningly realistic—and getting more so as the technology advances and more processing power becomes available.

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Combining Live and Virtual Instruments Creating an ensemble of instruments recorded in different spaces presents some problems. Here’s how to get around them.

By Dave Moulton

ample libraries have come a long way since the days when they really wanted to be hidden in a mix. You either had to mix them in low or disguise them with some live instruments in the foreground. Yet while that’s no longer true, live instruments can breathe a lot of life and expression into a piece, in fact even a small percussion part can add a lot. And there are still times when you want to layer real instruments on top of sampled ones to create an ensemble, for example if you’re creating a pop brass ensemble. (There are some intriguing big band brass libraries on the horizon, but as of now nothing available is totally satisfactory.) That sets up what this article is really about: combining live tracks with synth/sampled tracks. Why would we do such a cheesy thing?

Sometimes we find we really like the room ambience we hear in the studio and try to capture it by hanging up a stereo pair of room mics, only to be really surprised at how trashy it sounds when we check it out on monitors in the control room. (To save money, doh!) Can it possibly work? (Surprisingly well!) How do we approach this? (Very carefully.)

About live sound sources If we’re going to generate the equivalent of real live sounds, and or integrate synth sounds with live ones, we need to know a little about those live sounds. Here are some verities: • when we record a live instrument with a 56

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microphone, the instrument and the microphone are both in a room, and some distance from each other; • when the microphone is within the “near field” of the instrument (i.e. within a distance equal to the instrument’s largest vibrating or resonating dimension) we don’t “capture” the entire sound of the instrument, and what we get may not be representative of what we think the instrument should really sound like; • when the microphone is outside the “near field” of the instrument it may very well pick up reverberance from the room, particularly if it is a small room; • therefore when we capture the “real” sound of an instrument, we often also capture the sound of the room in which it is recorded, which is often something other than what we want; • and finally, we need to keep in mind that microphones perceive sounds in profoundly different ways than our ears do, and that a sound whose quality has been altered by the process of going through a microphone AND a loudspeaker AND another, different room to the listener is without doubt going to sound a lot different than the original sound did. The acoustics and psychoacoustics of the situation require it. When we are struggling with any kind of production that involves combining live sounds and sampling or synthesis (or even if we’re just struggling with sampling), we need to keep these verities in mind.

Recording live sources When we record live sources, particularly in overdubbing/layering situations, we need to devote considerable care to mic choice and placement, for the reasons cited above. Again, we are balancing direct versus room sound while obsessing about the particular tone quality we’re getting, which may not be (a) what it sounds like to our ears in the studio, or (b) exactly what we had in mind or wanted. Tiny changes in physical setup result in big, often illogical changes in sound character. Sometimes we find we really like the room ambience we hear in the studio and try to capture it by hanging up a stereo pair of room mics, only to be really surprised at how trashy it sounds when we check it out on monitors in the control room. My experience is this: when we’re doing multitrack production, we need to close-mic sounds and suppress room tone as much as possible when we’re tracking and overdubbing, except in very special cases. When we want room tone, ambience, or reverberance, we generally do best to add it in mixdown or postproduction. When we’re doing live acoustical recording (classical music, some kinds of acoustic music, live club recordings, etc.), we try to capture the live ambience as part of the recording (via one of a number of techniques). Here we’re less concerned with details of timbre of an individual instrument than we are about the balance between instruments and the balance between

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f e a t u r e VI the instruments and the room tone and/or reverberance.

Orchestration and studio orchestration: the fine art of misdirection It’s worth thinking for a moment about why we do this stuff. We’re making up music, right? If we are doing it the traditional way, we write dots on a page that musicians play. We usually make

Why we need to record dry tracks At this point it’s clear why we desire dry tracks to do this. If the tracks have room tone, that tone is going to become a significant part of the resulting sound, and there is no reasonable way out of it. That’s okay if you really like and can live with that particular room tone. Meanwhile, for my kind of tweaky music production work, sampled sounds with room

Figure 1: spreader. Caption: A basic spreader program creates “walls” with two discrete delays, panned hard L and R. It’s barely noticeable when you get it right—but you sure notice it if you mute it.

a decision as to who plays what. Sometimes we say to ourselves something like, “I really want this part to be like screaming lead guitar, except with a harmonic doubling a third below.” Welcome to the art of orchestration. Orchestration is the act of assigning different sounds to different instruments, as well as making sure those sounds are playable by those instruments. It’s a wonderful craft, and if you’re good at it, it can be a lot of fun. Something similar happens in the recording studio when we are producing recordings there. We combine, double, mix, and otherwise combine instruments to get different sounds and perspectives (not to mention all the signal processing stuff we do when suitably inspired). I call this “studio orchestration.” We are deciding how it is going to sound, not by calling up various instruments, but by the way we mic instruments, mix instruments together, and/or process instruments. The reason I belabor this is because in multitrack production, we usually do live/synth doublings like this for one of several pretty basic reasons: (a) we want to make one instrument sound like a group of instruments (chorusing/doubling, the Moron Block ‘n Tackle Choir, etc.), (b) we want to add a sense of spaciousness (spreaders, explained later; spread-triple-tracks; room ambience; reverb; etc.), or (c) we want to add some ear candy effect to a given musical voice (really tight live unison doublings in mono, or a really interesting extra coloration on an instrumental part). These things are all at the heart of studio orchestration. Now these doublings and triplings are all tried and true effects that mostly have become wonderfully effective clichés. Often, however, we can’t use them because we don’t have the resources available (usually either enough players or enough time), so we resort to synths or samplers to fill in the doublings we want.

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tone are really a shaky premise—unless I don’t plan on doubling them with anything else, because now we have Sampled Rooms whether we want them or not. Uh-oh! Sample libraries often give you a choice of wet or dry mic positions; my personal preference is to opt for the driest and add back moisture in mixing.

er—changs the sample patch slightly for each line so they have small but audible differences. If the tracks are harmonies rather than unison, make the live track the top note in the chord, with the sampled harmony parts below. Again, think about relative levels and EQ, although they won’t be quite as important. After you have them tracked and you are happy with the performance timing, you might wish to slip the synth doublings later in time by 3 or 4 milliseconds relative to the live track, which will further direct the listener’s attention to the musical quality of the live voice. Finally, tight gain riding and panning on individual notes can be a beautiful thing. On important notes, bring the levels up for emphasis, and/or pan them left and right to open up the sound, just for that note. This is easy to do in a DAW. The same is true with synth tracks. They may be more cheeseball than sampled tracks, but you can make a virtue of that by making the patch sufficiently weird or bizarro to enhance the live track, if you can be subtle about it.

Dealing with stereo Stereo can really be your friend here. You can pan parts to left, right, and/or center for more spaciousness. You can add ambience to the lead track only, or to all the tracks (which will have the added benefit of misdirecting the listener some more). You can add stereo reverb to just the live track, leaving the synth tracks dry (and less noticeable), while also increasing

You might wish to slip the synth doublings later in time by 3 or 4 milliseconds relative to the live track, which will further direct the listener’s attention to the musical quality of the live voice. There’s another way to look at this, though. For a great deal of film and broadcast audio production, the quality of the supplied room tone is just fine, and having it aboard from the beginning saves at least one step and probably a number of them. Anyway, when we are tracking and/or collecting our own samples, this means a dead studio and close miking.

Mixing live and sampled or synth tracks If you’re layering a live track on top of samples in order to create a larger sound, for example building a pop horn section with only one real trumpet, then obviously the live instrument has to be the “leader.” We want its liveness to cover the simplification of the sampled horns and misdirect the listener’s attention from their essential cheeseball character. So make it louder and make it brighter (boost EQ in the 2-4kHz range). If the sampled tracks are in unison with the live, make them softer (say, 2-3dB) and also gently roll off their top end. Make sure, also, that the sampled tracks aren’t phase-locked clones of one anoth-

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spaciousness and envelopment. And there’s always the panning stuff I mentioned above.

Spreaders Please look at Figure 1, the screen dump from Pro Tools that accompanies this article. A spreader is a cool little trick for “adding a room” to a sound when reverb or ambience is too much. You simply generate the first lateral early reflections for left and right. In your DAW, set up a stereo aux input or whatever the equivalent is called in your software. Insert a stereo medium slap delay with different delays, say 29 and 33ms—actual values depend on the song, the tempo, and the mood, and are selected by trial, error, and ear—panned hard left and right. Make the earlier of the delays approximately 3dB softer. Send all the signals you want to “spread” to said aux input via one of the internal buses. Adjust the send level to taste. When you get it right, you barely notice it until you mute it, at which point the dry track becomes surprisingly flat, one-dimensional, and comparatively lifeless. If you want a more complex spreader, you can add more delays (four or six is good, eight

is usually a little much). Work out a volley of lateral early delays from a room the size of a small club. Remember, the earlier of two nearly identical delay times from opposing channels should be made 3-5dB softer. This is to offset the precedence effect, which tends to “pull” the spreader ambience toward the side with the earlier time. Do this by ear, sitting on the median plane (i.e. centered perfectly between your monitors). Interestingly, I find that these simple primitive spreaders give me about 80% of all the short-term room ambience I want. The place that they don’t work is with impulsive sounds, such as snare drums, when the volley of delay times is usually revealed as a “smear” of snare drum hits. They do work great with vocals, electric bass, lead guitar, and horn tracks.

Ambience Ambience is short-term reverb, the kind of sonic decay that happens in small rooms and/or comparatively dead rooms. It can be a really attractive kind of sonic perfume that gives a palpable sense of “realness” to the sound without imposing a “wash” of decaying reverberance over (or under) everything. I find 150-300ms decay time—it might be called reverb time on your box or plug-in—is about as long as you want to go, and predelay, if any, should be quite short (no more than 30ms). The point about both spreaders and ambience is that when you send a group of live and synth/sampled sounds together to one of these effects, they acquire a similar character and tend to unify into a comparatively believable ensemble or section. That cheesy synthetic quality begins to really go away, and meanwhile you haven’t done anything that limits your options much for the rest of the mix.

often apply both ambience patches and large hall patches to various sounds in my mixes.

If you aren’t sufficiently confused by now, try surround! Surround makes all of this a bit more tricky. Because you now have five channels (well, four, anyway if you tend to skip the center channel), it is possible to spread stuff out more and make it clearer. Unfortunately, that also reveals the warts a bit more. For your live/synth sections, you probably want to devote some spreading or ambience to them in stereo, whether you assign that stereo to front or back. Don’t try to be really sophisticated (not yet, anyway!) with full surround ambiences and spreaders (which are actually a lot of fun, if immensely time consuming) until you are really sure of what you are doing. Also, for sustained pads it is immensely tempting to put one part in each channel and then wash the whole thing with reverb. If you get it right, this can be wonderfully enveloping. However, you run the risk of revealing your cheeseball synth tracks all too clearly (as in “Hmmm. That violin in Left Front is pretty nice, but it’s a shame it’s all muddied up by the cheeseball synth tracks everywhere.”) The payoff1 Mostly, what we’re trying to do in these cases is develop a sense of ensemble size, rich-

ness, and solidity. We want the assembled live, synth, and sampled parts to feel like part of a real band, in a real space, playing real music. It’s an illusion, of course. Some whacko things to help that illusion along include some of the following tricks: Build up a section feel by double- or tripletracking each part in unison on a separate track, using patches that mimic individual instruments rather than sections. There comes a point, while you are burning through an immense number of tracks, when it really begins to “feel” like a section! If you have enough sends and speakers, send your section tracks to an array of individual loudspeakers set on chairs in the studio, arrayed as if they are an ensemble of players or singers. Then record them in stereo, using whichever of the three main stereo mic techniques floats your boat the most! It can feel remarkably live and nobody will ever figure out how you did it (except me, of course!). Mainly, concentrate on developing the illusion! It’s a musical thing more than a technical thing. Use your ears. And keep trying until it feels really good. Thanks for listening. Dave Moulton is virtually an instrumental cheeseball these days. You can complain to him about anything at his website www.moultonlabs.com.

Dealing with reverb It’s a little tougher dealing with reverb. Reverb times begin at about 750ms (fairly dry) and go out to about 2.5 seconds (wet). Such reverb is essential to support most atmospheric string or sustained keyboard pads. Naturally, it covers a multitude of sins— including your sampled string sounds if you’re layering live strings on top of them. The problem is that it can cover a lot of other stuff as well if you aren’t careful. Today’s convolution reverbs work well for this, but standard reverb plug-ins may not sound all that good in this application. Personally, I’m a reverb junkie. I love the stuff and what it does. However, I’ve learned through a lot of painful experience to restrain my more crass impulses. The trick for most pop work is to use reverb sparingly—either for effect, sudden relaxation, or as a lush sort of background cushion for more dry upfront sounds in the lead in slower, less heavy material. For the kinds of things we’re trying to accomplish in this article, reverb isn’t necessarily the best fix. Happily, when you need it you can also apply the reverb to stuff that already has spreader treatments or ambience. They actually help the reverb, as a general rule. I VIRTUAL

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i n t e r v i e w VI BADELT (CONTINUED FROM PAGE 40)

So I defined all the sounds I used in all the GigaStudios on one graphical Environment page. [Again, please see Figure 2.] They were wonderfully sorted in layers like stings, brass, guitars, orchestral percussion, such and such. So if you had an idea, you didn’t know which sound would be perfect, you just look at it graphically. So you command/click [which assigns the instrument to the currently selected track]—this one? Maybe. This one? No. And you really had your whole pallet at your fingertips. VI: But you can’t do that now the way you’re working with EXS. Badelt: They are about to change that too. Right now my pallet is not that graphic, but still it’s quite a well organized hierarchy of sounds. It gives me a hierarchical view of what I can do, which is quite inspiring too. What they’re doing, the idea is that a channel strip setting is represented like a Favorite object, so basically by double-clicking on it you select it and it loads a track. And then you can create a pallet again. Logic loads so fast. I can load the biggest sounds in just a few seconds. It’s very optimized how they do it. We have sounds with so many layers, we reach the limits of the EXS constantly.

Now they have to make it a really good sampler. It’s a sample playback engine. I’m sure it can not keep up with the flexibility of Kontakt, MachFive, and Halion. But I prefer it integrated.

ing engineer who does film work all the time and the right hall. The real thing. But I have lots of it—CD-ROMs of drums and loops. But mostly we create new sounds in a project.

VI: Are you using using other virtual instruments than the ones in Logic?

VI: Do you spend a lot of time programming sounds for each project?

Badelt: The Logic ones are so good. Just look at the ES2—you can spend weeks with the ES2. Now with the different modeling synths, I haven’t gotten to the ground floor. It’s amazing.

Badelt: Well, I have special people for that. We have at least one sound programmer in a project. It’s a full-time job. VI: You still have Pro Tools in your rig, even

"What do you think of our EXS sampler?” I was laughing at them! I had 14 GigaStudios, each of them had four MIDI ports. They said, “You can load 64 samplers.” That’s not even one Giga! But I use the Spectrasonics stuff a lot. It’s great. Albino. I love these synths—the weirder they are, the better. To find a tiny little sound in one cue, now you can. To get us all these high and supernatural sounds, the bread and butter sounds are actually great. I use synths a lot in the template. And I use Kontakt a lot too, actually. I use Impakt quite a bit too, which is Kontakt, just a different GUI. VI: Do you use any of the commercially available orchestral libraries, or just your own custom sounds? Badelt: Basically not the commercial libraries. Maybe to augment a sound here and there, but you won’t even hear it. We just had the opportunity to record in a really good hall, most of the people I now use when I go to London, the same engineer. It’s about the recording, about the right record-

though you’re not using it for mixing anymore. Badelt: I use Pro Tools as a video player, playing back effects and dialog and picture. That’s the one thing I don’t do inside Logic yet. I like video at such a high quality for projection that it’s just taxing the computer too much. I’m waiting for the quadruple 7GHz! We’re writing music for films now just like the Avid guys cut: you need one more editor, you add one more Avid, connected by a fibre channel drive in another room. It’s scaleable, just another room opens up. That’s what we’re doing now. If you have two weeks or three weeks to write “Constantine,” you need three or four “outlet” composers—arrangers, whatever you call them. Open another room, put another G5 in there, clone our library, and you’re ready to go. Before it would mean build another studio. Not much more to it anymore. It doesn’t help you writing a good tune! You still have to do that, unfortunately. Maybe one day! I’ll ask them to put a preference or two: good cue, bad cue…

Fig 3: Badelt’s Logic Audio Environment for “Constantine.” This is all running on a single G5.

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enabled and then enable the surround to capture sections to disk when you are finished composing. (If you use Giga that way rather than having it on “slave” machines.)

(CONTINUED FROM PAGE 54)

Economy while writing Happy convolution For reference reverb while composing, try using just two instances of GigaPulse and sharing them: one for the up-close spread, the second for the far back spread.

Be sure to check out the other rooms and reverb units. GigaPulse ships with several rooms & halls, as well as some very expensive reverb unit impulses and a guitar amp, among other things. These programs have different mic or parameter selection options to explore. But they’re much simpler than the halls we’ve covered here. David Govett has been using Giga since before it was released. As a composer, he worked with game composing company Fatman for eight years. Dave has programmed many of the top sample libraries.

Close.

Far.

This also applies to the other mic positions, which can be assigned in the same way to stage positions. Here, the surround mics are assigned to the same positions as the close ones: The best way to do this is to use two group channels. Put one instance in the insert of each group, one for close-up, the other for farther back. Then you can assign any mixer channel output— i.e. any instrument—to either of the two groups to position it on the stage. Instruments can be panned left to right by simply using the MIDI or mixer channel pan control. To save resources, you can compose with two of the mic positions

TRENDS (CONTINUED FROM PAGE 64)

Inventions Lindemann has been working on Synful Orchestra for roughly eight years and has had his additive synthesizer running for the last five or six. Along the way he’s been awarded several patents. While additive synthesis is nothing new, the Synful approach to dealing with its parameters is unique. The complicated part is searching and finding the fragments, and then splicing them together transparently. A single note may have 100 – 150 harmonics and a typical piece of music 10 – 20,000, and that requires a lot of computation. Lindemann created and subsequently patented an efficient way of doing additive synthesis. The subject of another patent deals with how to smooth out the timbre; simple crossfading isn’t nearly transparent enough. Lindemann developed a vocabulary to label each phrase and catalogue it. These phrases then get crunched by the computer, which starts by extracting the harmonics. It takes about 25 minutes of calculations to produce a completed instrument, which consists of a database that the synthesizer plays in real time. 62

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Getting giggy with Giga

igaPulse comes with a lot of meticulously-sampled impulse response files. You can make and import your own as well; Giga 3.1, due out this summer, makes it really easy to import files, and there are instructions on TASCAM’s website for earlier versions (look in the Giga Support area). If you have a plug-in and/or outboard effects or processing chain you like for a program, for example, you could sample the result and “burn” an impulse response file. You don’t need to use sine wave sweeps for this—a full-scale digital click (which you can create with your DAW’s pencil tool) should work fine. You can also have a raucously good time creating your own effects. The obvious way is to find interesting things and sample them, whether it’s the inside of a piano, a cardboard tube, a toilet, or someone’s huge mouth. Just keep it clean. Here’s a tip from King Idiot: how about sampling some of your entire mix and running a signal through that? The result is an interesting ambient effect.

De...lay Actually, the initial Synful Orchestra interface has a Magic Button (“delay for expression”), which is a 1-second lookahead delay that allows it to calculate note transitions more accurately by looking at the incoming MIDI data. It would be more elegant if the sequencer could tell Synful what notes are coming up in the track, but there hasn’t been any demand for such a feature before now. In the meantime the solution is to sequence the instrument with the Magic Button turned off, slide the track forward one second, and then turn the button on to listen and tweak notes before mixing. Plans Lindemann plans to round out the existing orchestra with such instruments as string section, staccato strings, harmonics, and various tremolos and different modes of playing. Upcoming products may include jazz instruments, trumpet, and various saxophones, along with various iconic-type data fragments and sounds—perhaps a John Coltrane sound for example. As of this writing, Synful is available for Windows only, but a Macintosh version is due for release in July.

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The future? Whether the Synful technology is a replacement for sampling is partly a matter of opinion. It certainly won’t be a total replacement; Lindemann feels that sampling works well for instruments like piano and percussion, which aren’t changed by the player after the note starts. Synful’s strength is in modeling expression and note transitions. Despite its limitations, “some people” feel sampling has an inherent life. Since Synful is tied to the original recordings, the sound is always going to be subjective. At the moment, some of the instruments in its orchestra are arguably better than others. What’s certain is that Synful is a very interesting technology that is likely to wield some influence. It makes writing “to the library”— writing to accentuate a given library’s strengths and avoid its weaknesses—totally unnecessary. And it’s much less demanding of computers. At the very least, Synful is an exciting new tool to add to our arsenal. You can download a functional demo and listen to mp3 examples from www.Synful.com. Frederick Russ is an accomplished orchestral composer. He runs the discussion forum at www.vi-control.net/forum and invites you to take part.

r e v i e w VI COLOSSUS (CONTINUED FROM PAGE 18)

in the NI sampler engine. For example there are plenty of velocity layers, and some of the programs auto-alternate samples every time you hit a key to avoid exact repetitions. Also, many programs have extra performance elements like pitch glide samples leading into bass and guitar notes when you play extra hard. Other than velocity, you play Colossus with the sustain pedal, modwheel crossfades when labeled, and then use MIDI CC11 (expression) for riding the volume. Beyond that, the Kompakt player has a variety of parameters and effects you can use to shape the sounds—traditional synthesis parameters (filters, modulation, envelopes), tuning, etc. These features are available independently for each of the eight instruments an instance of the player can load. Colossus leaves most of that up to you by default, but it’s very easy to use.

Sounds As you can see from the screen dump with this review, there are 18 banks in Colossus, each with a long list of instruments. There’s also a General MIDI bank that uses mostly the same instruments. The Colossus library is very well organized and therefore takes virtually no time to learn. On the other hand, there’s so much in it that every time you browse through the programs you discover something else. If you get bored—which you won’t—it’s very easy to layer sounds on the same MIDI channel, and that opens up a lot of possibilities. And now we got to some subjective comments about the sounds, starting with the acoustic and electric drums. There’s a good selection of nice kits, starting with a really funky ’60s kit to brush and stick jazz kits with a characteristically tiny (probably 18”) open bass drum, on up to big rocking heavy impact kits, techno kits, 808-style hip-hop kits, and so on. All are cleanly recorded and tastefully processed; these drums stand up to pretty much any sampled drums on the market. The Kompakt player instrument doesn’t let you replace, say, the cymbal from one kit with one in another—you have to load the whole kit on another MIDI channel and play the cymbal separately—but it does have a really useful Group Edit feature. That means you can select, say, the toms (which are subjectively too low in some of the kits) and raise their levels. Colossus’ programmers did a good job of creating logical Groups for you to tweak. There are some really nice electric basses from the Hardcore Bass library, and the acoustic bass is outstanding. Though they’re already amped, I found that running some of the electric basses through a bass amp plug-in makes them even better (I used the new one included with Logic 7.1). Here again, the Group Edit feature lets you adjust the velocity response to stop some of the slap and bend samples from triggering too easily.

There’s a nice selection of acoustic guitars to choose from, some of which take advantage of the virtual instrument’s round robin feature. Colossus also has banjo, mandolin, and uke, along with a big selection of electric guitars. Some of the electrics are recorded with amp tone including spring reverb, others are ready to run through a guitar amp plug-in. Ethnic instruments, both pitched and nonpitched, are a big strength of this library. There are too many highlights to list, but whether it’s sitar (an amazing program) or kalimba, these programs make it really easy to get the right flavor. Great stuff. Colossus includes a variety of “keyboard” instruments, from accordian to reed organ to the mallet instruments from the East West

from the Quantum Leap Brass library. It’s not totally lousy, but it’s not at the same level as most of the other content here. So far nobody has done this very satisfactorily, although there are some libraries on the horizon. In addition to men’s choir shouting (“Oh!” etc.), Colossus has separate men’s and women’s choir programs, with syllables like ah, ee, and oh, mostly with modwheel crossfades into progressively brighter sounds or from ah to mm. That makes the choir sound alive, which is necessary since these programs are looped. You can also get some great choir sounds by layering programs. And finally we get to the orchestral sounds, which come from the Gold and Platinum versions of the East West Quantum Leap

Colossus is being presented as a modern-day equivalent to the keyboard workstations we relied upon for years, and part of that concept is that its instruments are almost all playable in real time. Quantum Leap Symphony Orchestra library (EWQLSO). I promise not to moan about the orchestral vibes not having pedal-up samples— which is particularly unfortunate given how good they sound otherwise. While the new age ensemble programs aren’t my personal cup of tea, they’re very competent synth/acoustic compound sounds. What is my cup of tea is the name of the first one: Andreus Kugelschreiber. The Stormdrone instruments are just outstanding. And I mean outstanding. These are 4way evolving drones that you crossfade with the mod wheel, grouped into high morphs, low morphs, and low-to-high morphs. You could easily score a scene or commercial with these programs. Hidden deep under a hierarchical menu labeled “disk utility” are six hard-pounding Stormdrum percussion ensemble loops. Colossus includes a newly recorded Fazioli piano and a pretty decent Steinway B sampled piano. The Fazioli isn’t sampled in enough detail, and perhaps it lacks some of the clangorous power of a real Fazioli, but it blends in with the EWQLSO orchestral samples in this library very nicely. There are also some excellent electric pianos in this category, including an ’80s one that out-’80s the ’80s, and a wonderfully detailed Rhodes suitcase 88 (complete with tine-damping release samples). Colossus also has a full bank of vintage organs that should make keyboard players very happy. The synth bass, lead, and pad groups are somewhat traditional analog sounds, but they sound really good—none of that thin, digital sound you get with too many sampled analog programs. They’re also programmed nicely, with modwheel crossfades between overlapping sounds and just the right velocity response. One indication of how far sampling has come in recent years is the older pop brass

Symphony Orchestra library, only they don’t have the recorded hall reverb trails (“release trails”). Some of the included programs, like the solo cello, are just gorgeous; others are decent if not spectacular, such as the solo clarinet; still others, like the piccolo, I’d put in the Very Good (B+) category. All the instruments of the orchestra are represented, with the exception of percussion such as drums and cymbals. The orchestral programs in this collection are good choices, since they cover a lot of range. You can certainly find a more flowing legato violin patch, for example, but you couldn’t use it for fast passages; the Colossus violin program covers both situations reasonably well. If you just want to add a symphonic sound to songs when you play live, you should be happy with the high-quality orchestral instruments in this library. Serious orchestral composers will want to look at dedicated orchestra libraries, however. They have way more articulations, and they separate the ensemble string and brass programs into individual instruments—although, to be honest, many people wouldn’t notice the difference if you mix in a little of the solo strings with the string ensemble programs.

The end of sample libraries There’s no such thing as the last sample library you need to buy, since there are libraries that cover every category in great detail, and there will always be different varieties of everything. But this library is complete enough to stand alone. Given its vast range of high-quality instruments, Colossus is equally ideal as the foundation of a professional sample library and as a fill-in-the-gaps collection. While its $1000 list price is a considerable investment for most musicians, in this case I consider it a bargain.

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Synful Orchestra This new additive synthesis virtual instrument analyzes the incoming MIDI data and uses a database of phrases to decide what note fragments to splice together. Could this technology dethrone sampling? By Frederick Russ

isitors to the NAMM Convention this winter who made the trip down to a distant booth in Hall E—where new companies are stationed—may have stumbled across a stunning new emulative synthesizer. This instrument, called Synful Orchestra, was demonstrated ably by company president Eric Lindemann. It turns out that Lindemann is an industry veteran, having designed electronic music machines for the past 35 years. His resume includes having designed the original Waveframe sampler in 1986, designing a computer music workstation for Pierre Boulez at IRCAM in Paris, contributing DSP code for the Linn Drum Machine, he had a stint at Euphonics—as well as having played keyboards for the Osmonds and the 5th Dimension, and being a trained composer. The remarkable thing about Synful is that depending on the instrument, it sounds pretty close to a modern orchestral sample library with dozens of sampled articulations. What’s different is that it only requires one track in a sequencer— the instrument seems to be “changing articulations” (which is not what it’s doing, as you’ll see) and modeling note transitions automatically. While Synful’s sounds are inextricably tied to the sampled recordings it starts out with, it is neither a sampler nor a physical modeling synthesizer. Instead, it’s an additive synthesizer— with some extremely complicated twists.

Overview Lindemann explains it in his own words: “For each instrument there is a database containing phrases of real playing—recordings 64

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of, for example, a violin playing three or four seconds of some fast-flowing passages from a symphony. The database has numerous examples, representing a whole variety of ways that the violin is played. Let’s think of these as recordings for now, although they’re not really stored that way. “The program analyzes incoming MIDI data real time. Synful then extracts as much information about the phrase that is being played as possible. It looks at the separation between notes, the overlapping of notes to determine legato playing, velocity, the position of pedals, pitch wheel, and so forth. “Suppose for example that it determines, ‘okay, what’s being played here is a legato transition between B and D, the first one is a quarter of a second long and the second note is a second long.’ Ideally it would find the perfect fragment exactly like that in the database, but it’s probably only going to find something similar. If it finds a fragment going from Bb to D, it will have to change Bb to B, preserving the nice transition. That segment needs to be spliced onto the previous segment. “What’s happening is a lot of searching for little fragments, splicing fragments together, and morphing fragments in pitch and time to be synthesized. Usually Synful Orchestra will be doing three or four splices per note, depending on the length of the note, the note transition, etc. “Those splices have to sound like one note, so all of this requires a flexible representation of the sound. Rather than being stored as samples, the sound is stored in the additive synthesis model, as time-varying harmonics. The computer is also modulating harmonics and noise elements, which are stored separately, such as the attack noises of a bow, reed noises, as well as the sustained brass or sustained bow noise.” Storing the sounds this way means that the entire Synful Orchestra is only 32MB; a comparable orchestral sample library could be 1000 times the size.

(CONTINUED ON PAGE 62)

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