Waves Plug-Ins Workshop: Mixing by the Bundle

Waves Plug-Ins Workshop: Mixing by the Bundle Barry Wood

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Waves Plug-Ins Workshop: Mixing by the Bundle Barry Wood Publisher and General Manager, Course Technology PTR: Stacy L. Hiquet Associate Director of Marketing: Sarah Panella

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This book is dedicated to Alex Cima, whose classes at Fullerton College provided me with a solid grounding in recording electronics and the physics of sound and also gave me my first experience under pressure behind the mixing board. It’s also dedicated to all the artists whom I’ve worked with over the years: It has been, and will continue to be, a learning experience.

Acknowledgments I’d like to thank Orren Merton for suggesting that I try my hand at writing a book. It’s not something that I would have thought of doing on my own. I need to thank Cathleen Small for helping me appear more literate than I am and Andre Knecht for doing an outstanding job of pointing out the places where I could be clearer or where I was missing something entirely. I also need to thank the people who contributed to the song “Running Aground,” which was recorded as a test bed for this book: n

Mark Smith, who wrote and played the excellent guitar parts—all in just a couple of hours, as usual.

n

Orren Merton, who wrote the lyrics and melody of the song and who sang it as well. Thanks for letting me push your performance in a style somewhat out of your oeuvre.

n

Clare Carnegie, who’s always willing to come over to create and record background vocal parts on the spot.

n

Craig Wheeler, who graciously allowed me to reuse drum tracks that I had originally produced for one of his songs.

And finally, I’d like to thank Ittai Shaked at Waves for taking time to answer questions that only someone at Waves could have answered.

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About the Author Barry Wood has been involved in music and audio in one way or another for as long as he can remember. He grew up playing trombone and currently plays keyboards, Chapman Stick, bass, and percussion. As a composer, he has written music for numerous commercials and industrial videos. He also composed music for the award-winning documentary Earthlings, narrated by Joaquin Phoenix. The band Edison Suit, his collaboration with guitarist Mark Smith, has released two albums and has had songs used in television shows such as MTV’s Real World. In 2010, he received an NEA grant commission to compose a microtonal piece for the Hutchins Consort, a New Violin Family string octet. His recording experience started at the tender age of 11 with a Christmas gift from his grandparents: a portable cassette recorder. In high school, he graduated to a pair of cassette decks and some Y cables. He used these to create multitrack recordings of his Casiotone mini keyboard and Mattel Synsonics drum machine by Ping-Ponging from one tape to the other while adding new parts. A turning point in his audio education came in 1983, when he enrolled in recording and electronic music classes at Fullerton College. There he learned about the technical aspects of recording—the physics of sound, the functions of various types of outboard gear, signal flow, and troubleshooting. Those classes provided a solid basis for his future career in audio. He remains involved at Fullerton College, providing advice on curriculum as a member of their recording arts professional advisory committee. The next major stage of his audio education was as co-owner and head engineer at a commercial 24-track studio, Ashwood Productions. The day-in, day-out process of running recording sessions on analog tape with a non-automated mixing board honed and expanded on the skills he learned at Fullerton College. In 1995, he sold Ashwood and started his current studio, the Other Room (www.otheroom.com), where he dove headfirst into computer-based audio recording with a Mac 6100 and an AudioMedia II card. That system was capable of playing back a whopping four tracks of audio, which brought back memories of his cassette Ping-Pong recording days. Computer recording was a brave new world that introduced a new set of challenges along with the concept of audio plug-ins. After using Waves’ products for several years, he was invited to join their beta-testing team in 2001. Over his career Barry has recorded, mixed, and/or mastered more than 400 albums for a large number of artists in a wide array of styles. On the literary side, he has contributed to articles in Keyboard, Electronic Musician, and Home Recording magazines. He has served as technical editor for seven books about audio by authors such as Mitch Gallagher and Bobby Owsinski. His best-known literary endeavor is his NAMM Oddities website (www.nammoddities.com), where he reviews the innovative, under-appreciated, and sometimes inscrutable products shown at the Winter NAMM show each year. This is his first book.

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Contents Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . xi

Chapter 1 Introduction to Waves Plug-Ins

1

The Waves Bundles ............................................................................................................ 2 Understanding Plug-Ins ...................................................................................................... 6 Mastering the Waves User Interface ................................................................................... 6 Changing Plug-In Values .................................................................................................... 6 Mixing Basics..................................................................................................................... 7 Tracking....................................................................................................................... 7 Editing ......................................................................................................................... 8 Mix Levels and Processing ........................................................................................... 8 Mix Automation .......................................................................................................... 9 Mastering ..................................................................................................................... 9 Brief Discussion of Mixing versus Mastering ..................................................................... 9 The Sample Project .......................................................................................................... 10 Mixes ......................................................................................................................... 10 Individual Tracks ....................................................................................................... 10 Panning ...................................................................................................................... 11 Master Fader.............................................................................................................. 12 Preset Files ................................................................................................................. 13 Downloading.............................................................................................................. 13 What’s Coming Up Next.................................................................................................. 13

Chapter 2 The Musicians 2 Bundle

15

Equalization ..................................................................................................................... 15 Renaissance EQ................................................................................................................ 17 Dynamics ......................................................................................................................... 18 Limiters...................................................................................................................... 18 Compressors............................................................................................................... 19 Gates.......................................................................................................................... 19 Expanders .................................................................................................................. 19

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Contents

vii

Renaissance Compressor ............................................................................................ 20 Renaissance Axx ........................................................................................................ 21 Renaissance Vox ........................................................................................................ 22 Effects .............................................................................................................................. 22 SuperTap.................................................................................................................... 23 Doubler ...................................................................................................................... 24 The Mix ........................................................................................................................... 26 Drums ........................................................................................................................ 27 Loops ......................................................................................................................... 29 Bass............................................................................................................................ 30 Guitars ....................................................................................................................... 31 Keyboards .................................................................................................................. 34 Vocals ........................................................................................................................ 35 What’s Coming Up Next.................................................................................................. 37

Chapter 3 The Silver Bundle

39

Equalization ..................................................................................................................... 39 Q10 Paragraphic Equalizer ........................................................................................ 39 MaxxBass................................................................................................................... 40 Dynamics ......................................................................................................................... 41 C1 Parametric Compander ......................................................................................... 42 L1 Ultramaximizer ..................................................................................................... 43 DeEsser ...................................................................................................................... 44 Effects .............................................................................................................................. 45 MondoMod................................................................................................................ 47 Enigma....................................................................................................................... 48 TrueVerb.................................................................................................................... 49 IR-L Convolution Reverb ........................................................................................... 50 Other Plug-Ins.................................................................................................................. 52 S1 Stereo Imager ........................................................................................................ 52 PAZ PsychoAcoustic Analyzer.................................................................................... 54 The Mix ........................................................................................................................... 56 Drums ........................................................................................................................ 57 Loops ......................................................................................................................... 57 Bass............................................................................................................................ 58 Guitars ....................................................................................................................... 59 Keyboards .................................................................................................................. 62 Vocals ........................................................................................................................ 64 What’s Coming Up Next.................................................................................................. 66

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Waves Plug-Ins W orkshop: Mixing by the Bundle

Chapter 4 The Gold Bundle

67

Equalization ..................................................................................................................... 67 V-EQ3/V-EQ4............................................................................................................ 67 Dynamics ......................................................................................................................... 68 V-Comp ..................................................................................................................... 69 Hybrid Compressor.................................................................................................... 70 C4 Multiband Processor............................................................................................. 71 MV2 Dynamics Processor .......................................................................................... 72 Effects .............................................................................................................................. 73 MetaFlanger ............................................................................................................... 73 Hybrid Delay ............................................................................................................. 74 UltraPitch ................................................................................................................... 76 Renaissance Reverb .................................................................................................... 77 Other Plug-Ins.................................................................................................................. 78 Waves Tune LT.......................................................................................................... 78 Doppler ...................................................................................................................... 81 AudioTrack ................................................................................................................ 81 The Mix ........................................................................................................................... 83 Drums ........................................................................................................................ 84 Loops ......................................................................................................................... 87 Bass............................................................................................................................ 87 Guitars ....................................................................................................................... 88 Keyboards .................................................................................................................. 90 Vocals ........................................................................................................................ 90 What’s Coming Up Next.................................................................................................. 93

Chapter 5 The Platinum Bundle

95

Equalization ..................................................................................................................... 95 PuigTec MEQ-5 ......................................................................................................... 95 PuigTec EQP-1A ........................................................................................................ 96 Linear Phase EQ ........................................................................................................ 97 Renaissance Bass ........................................................................................................ 99 Dynamics ....................................................................................................................... 100 Linear Phase Multiband ........................................................................................... 100 L2 Ultramaximizer ................................................................................................... 102 L3 Ultramaximizer ................................................................................................... 103 MaxxVolume ........................................................................................................... 105 Renaissance DeEsser ................................................................................................ 105 Other Plug-Ins................................................................................................................ 106 Renaissance Channel ................................................................................................ 106

Contents

ix

The Mix ......................................................................................................................... 108 Drums ...................................................................................................................... 109 Loops ....................................................................................................................... 109 Bass.......................................................................................................................... 110 Guitars ..................................................................................................................... 111 Keyboards ................................................................................................................ 112 Vocals ...................................................................................................................... 113 What’s Coming Up Next................................................................................................ 118

Chapter 6 The Diamond Bundle

119

Dynamics ....................................................................................................................... 119 Trans-X.................................................................................................................... 119 L3 Multimaximizer .................................................................................................. 121 Effects ............................................................................................................................ 122 SoundShifter............................................................................................................. 122 Other Plug-Ins................................................................................................................ 123 Center ...................................................................................................................... 123 Morphoder............................................................................................................... 124 Q-Clone ................................................................................................................... 125 X-Noise.................................................................................................................... 127 Z-Noise .................................................................................................................... 130 The Mix ......................................................................................................................... 132 Loops ....................................................................................................................... 133 Guitars ..................................................................................................................... 134 What’s Coming Up Next................................................................................................ 135

Chapter 7 The Horizon Bundle

137

Dynamics ....................................................................................................................... 137 PuigChild 660/670 ................................................................................................... 137 CLA-2A ................................................................................................................... 138 CLA-3A ................................................................................................................... 140 CLA-76 Bluey/Blacky ............................................................................................... 140 L3-16 Multimaximizer ............................................................................................. 142 Vocal Rider .............................................................................................................. 143 Effects ............................................................................................................................ 145 GTR Tuner .............................................................................................................. 145 GTR Stomp.............................................................................................................. 146 GTR Amp ................................................................................................................ 146 GTR Tool Rack ....................................................................................................... 147 Other Plug-Ins................................................................................................................ 149 DeBreath .................................................................................................................. 149

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Waves Plug-Ins W orkshop: Mixing by the Bundle

The Mix ......................................................................................................................... 151 Guitars ..................................................................................................................... 152 Keyboards ................................................................................................................ 156 Vocals ...................................................................................................................... 156 What’s Coming Up Next................................................................................................ 160

Chapter 8 The Mercury Bundle

161

Equalization ................................................................................................................... 161 API 550A/API 550B ................................................................................................. 161 API 560.................................................................................................................... 163 Kramer HLS Channel............................................................................................... 164 LoAir ....................................................................................................................... 166 Dynamics ....................................................................................................................... 167 Kramer PIE Compressor........................................................................................... 167 API 2500.................................................................................................................. 168 C6 Multiband Processor........................................................................................... 170 Effects ............................................................................................................................ 171 IR-1 Convolution Reverb ......................................................................................... 172 Other Plug-Ins................................................................................................................ 173 W43 Noise Suppressor ............................................................................................. 173 WNS Noise Suppressor ............................................................................................ 173 Waves Tune ............................................................................................................. 175 Dorrough Stereo....................................................................................................... 176 The Mix ......................................................................................................................... 177 Drums ...................................................................................................................... 178 Loops ....................................................................................................................... 182 Bass.......................................................................................................................... 182 Guitars ..................................................................................................................... 183 Keyboards ................................................................................................................ 185 Vocals ...................................................................................................................... 188 Wrap Up ........................................................................................................................ 188

Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 189

Introduction Mixing is a unique blend of art and science. It’s difficult to get good results without equal helpings of both technical expertise and good aesthetic instincts. If it was simply a matter of science, there would be a box or a piece of software you could purchase that would magically mix a song for you. But as much as some manufacturers might like you to believe this is possible, it’s just not the case. The thought that blindly turning knobs and flipping switches will enable you to achieve a good mix is akin to the idea that an infinite number of monkeys at an infinite number of typewriters will be able to eventually produce the collected works of Shakespeare. Without an understanding of the tools at your disposal, you’ll spend a lot of time producing inconsistent mixes—and they certainly won’t be Shakespeare! Over the course of this book, I will provide insight into the workings and application of each plugin that Waves offers. By the time you finish the book, you should be on pretty solid technical footing with the Waves version 8.0 suite of plug-ins. But hey, enough of my yakkin’; whaddaya say? Let’s boogie!

Companion Website Downloads You may download the companion website files from www.courseptr.com/downloads. Please note that you will be redirected to our Cengage Learning site.

*If you don’t recognize this quote, put down this book immediately and go watch the film This Is Spinal Tap.

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1

Introduction to Waves Plug-Ins

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early anyone who has used a computer for recording and mixing has heard of the company Waves. From the early days of Digidesign’s 2-track audio editor, Sound Designer II, Waves has been a premier manufacturer of highquality plug-ins. Their offerings now include more than 100 plug-ins organized into a number of different bundles.

Over the course of this book, I will demonstrate the use of various Waves plug-ins by employing them to create a mix of a song. You will be able to download audio files for the song as well as all of the plug-in presets for each version of the mix. You’ll find more details about the demonstration song and downloading the files at the end of this chapter. In Chapter 2, I will mix the song using just the six plug-ins found in the Musicians 2 bundle. In each successive chapter, I will use a progressively more complete Waves bundle to mix the same song, and I will finish up with the Mercury bundle, which includes virtually every plug-in that Waves makes. This book should prove to be an excellent resource for determining whether you would like to purchase a bundle of Waves plug-ins or update your current bundle to a more complete one. Although reading about mixing can certainly convey useful information, there’s no substitute for hands-on and “ears-on” experience. You should familiarize yourself with this book and then go to www.waves.com and take advantage of their seven-day demo. If you request a demo for the Mercury bundle, you will have a week to try out all the plug-ins discussed in this book. Using the preset files, you can see and hear exactly what I did for the mix and get a good feel for how you can use them. To start the demo, you will need an iLok key and an account at www.ilok.com. A Note about Piracy It’s impossible to spend any time in the audio business and not be aware of the extent to which audio software is pirated. What most people don’t realize is that even the audio software companies that are considered large, such as Waves, are really quite small when compared to any number of other industries. That, combined with the fact that the pro audio market is fairly small, means that piracy has a potentially catastrophic effect on the companies that we rely on to produce the tools we use.

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W a v e s Pl u g - I n s Wo r k s h o p : M i x i n g b y t h e B u n dl e

The Waves Bundles Waves has more than 30 different bundles available. This book will focus on seven of them, starting with the humble Musicians 2 bundle and finishing with the nearly all-inclusive Mercury bundle. I selected bundles that follow a consistent progression of price and capabilities. Tables 1.1 through 1.4 show which plug-ins are included in each bundle.

ry M er cu

H

or iz o

n

d D

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m Pl at in u

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4

4

4

4

4

4

Q10 Paragraphic

4

4

4

4

4

4

MaxxBass

4

4

4

4

4

4

V-EQ3

4

4

4

4

4

V-EQ4

4

4

4

4

4

PuigTec MEQ-5

4

4

4

4

PuigTec EQP-1A

4

4

4

4

Linear Phase EQ

4

4

4

4

Linear Phase Lowband EQ

4

4

4

4

Renaissance Bass

4

4

4

4

Renaissance EQ

4

Si lv er

M

us

ici an s

2

Table 1.1 Equalizer Plug-Ins by Bundle

API 550A

4

API 550B

4

API 560

4

Kramer HLS Channel

4

Lo-Air

4

Chapter 1

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In t r od u c t i on t o W a v e s P l u g - I n s

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Table 1.2 Dynamics Plug-Ins by Bundle

Renaissance Compressor

4

4

4

4

4

4

4

Renaissance Axx

4

4

4

4

4

4

4

Renaissance Vox

4

4

4

4

4

C1 Parametric Compander

4

4

4

4

4

4

L1 Ultramaximizer

4

4

4

4

4

4

DeEsser

4

4

4

4

4

4

Hybrid Compressor

4

4

4

4

4

C4 Multiband Processor

4

4

4

4

4

MV2 Dynamics Processor

4

4

4

4

4

Linear Phase Multiband

4

4

4

4

L2 Ultramaximizer

4

4

4

4

L3 Ultramaximizer

4

4

4

4

L3-LL Ultramaximizer

4

4

4

4

MaxxVolume

4

4

4

4

Renaissance DeEsser

4

4

4

4

V-Comp

4

4

4

Trans-X

4

4

4

L3 Multimaximizer

4

4

4

L3-LL Multimaximizer

4

4

4

PuigChild 660/670

4

4

CLA-2A

4

4

CLA-3A

4

4

CLA-76 Bluey/Blacky

4

4

L3-16 Multimaximizer

4

4 (continued )

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Table 1.2 Dynamics Plug-Ins by Bundle (Continued )

4

Vocal Rider

4

Kramer PIE Compressor

4

API 2500

4

C6 Multiband Processor

4

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Table 1.3 Effects Plug-Ins by Bundle

M

4

SuperTap

4

4

4

4

4

4

4

Doubler

4

4

4

4

4

4

4

MondoMod

4

4

4

4

4

4

Enigma

4

4

4

4

4

4

TrueVerb

4

4

4

4

4

4

IR-L Convolution Reverb

4

4

4

4

4

4

MetaFlanger

4

4

4

4

4

Hybrid Delay

4

4

4

4

4

UlraPitch

4

4

4

4

4

Renaissance Reverb

4

4

4

4

4

4

4

4

GTR Tuner

4

4

GTR Amp

4

4

GTR Stomp

4

4

GTR Tool Rack

4

4

SoundShifter

IR-1 Convolution Reverb

4

Chapter 1

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Table 1.4 Other Plug-Ins by Bundle

S1 Stereo Imager

4

4

4

4

4

4

PAZ PsychoAcoustic Analyzer

4

4

4

4

4

4

Waves Tune LT

4

4

4

4

Doppler

4

4

4

4

4

AudioTrack

4

4

4

4

4

4

4

4

4

Center

4

4

4

Morphoder

4

4

4

Q-Clone

4

4

4

X-Noise

4

4

X-Click

4

4

X-Crackle

4

4

X-Hum

4

4

Z-Noise

4

4

Renaissance Channel

DeBreath

4

4

W43 Noise Surpressor

4

WNS Noise Surpressor

4

Waves Tune

4

Durrough Stereo

4

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W a v e s Pl u g - I n s Wo r k s h o p : M i x i n g b y t h e B u n dl e

Understanding Plug-Ins To use plug-ins, you need to run them within your digital audio workstation (DAW) software, such as Pro Tools, Logic, Digital Performer, or Cubase. Plug-ins are essentially bits of software that will process audio in various ways that expand the capabilities of your DAW. You can think of these in the same way that you think of audio processing hardware, in that each device will do something to the signal and then pass that modified signal on to the next processor and eventually to the mix bus. The order in which plug-ins occur on a given track is important because the results of the first processor will be passed on to the next one. Unlike a hardware processor you can only use on one track, you can use a plug-in processor simultaneously on as many tracks as your computer can handle. The only limitations are based on the speed and number of CPU cores that your computer possesses. Most modern systems are capable of running dozens or even hundreds of plugins simultaneously. The sonic differences between plug-ins and their hardware counterparts are often debated loudly, but there is no denying the cost advantages and the perfect recall that plug-ins provide. In the world of hardware, the settings of each device had to be carefully documented to give you any sort of chance of re-creating a mix later. With plug-ins you simply open your project file and resume work.

Mastering the Waves User Interface All Waves plug-ins share some common user-interface features. At the top of each plugin interface, you will find the Waves toolbar shown in Figure 1.1 and described in Table 1.5.

Figure 1.1 The Waves plug-in toolbar.

Changing Plug-In Values There are several ways to change the values of knobs and sliders. You can use the mouse to grab the control and drag to change the value. You can also click to highlight a control and use the mouse scroll wheel (if you have one) to change it. Additionally, you can double-click the value display and then enter a value with your keyboard. There are also a number of hardware control surfaces available that you can program to allow parameter control using actual knobs and buttons. For the most part, I’m comfortable using a mouse to change values, but there are times when I use my Frontier AlphaTrack to record smooth, real-time parameter changes as automation data in my DAW. Any automation parameter that the plug-in presents to the DAW is available for assignment to the AlphaTrack fader or any compatible controller of your choice.

Chapter 1

In t r od u c t i on t o W a v e s P l u g - I n s

7

Table 1.5 Buttons Common to Each Waves Plug-In Command

Description

Undo

The curved arrow button that points to the left is Undo. Clicking on this will undo the last change you made to the plug-in. You can undo the last 32 changes you’ve made to the plug-in using this button.

Redo

The curved arrow button that points to the right is the Redo button. After you’ve undone something, you can click this button again to redo it. When there are no more actions to undo or redo, the appropriate button will dim.

Setup A/B

You can have two complete versions of the plug-in settings. This button allows you to quickly switch between the two sets of settings. This is extremely useful for evaluating different plug-in settings.

Preset Step

The left and right arrow buttons will load the previous or next preset in the current preset list.

A->B

This will copy all the settings of Setup A to Setup B. When you’re unsure about making a change, you can copy your current settings and use those as a starting point for further changes.

Load

This will allow you to select one of the current presets or load individual presets or groups of presets from a file on your computer. If you download the song project, you will use this feature to load the presets that I saved while I created the mixes for this book.

Save

If you would like to save a preset, you can use this button.

?

Clicking on this button will open up the plug-in documentation that the Waves installers placed on your computer.

Mixing Basics A song goes through several production stages on its way to becoming a finished song, ready for public consumption. The following sections provide a brief description of each of these steps. This book will cover the third step, where the focus is on the sonic aspects of the mix—specifically, the use of plug-ins.

Tracking You need to start somewhere. To have something to mix, you have to record all the tracks that may be included in the final mix. These tracks can be recorded using microphones, electronic instruments that are plugged directly into the audio interface, or virtual instruments whose audio is created entirely within your computer. You may also be receiving tracks from someone else to mix. If that person has properly named his tracks, you’ll have a reasonable idea of what you’re working with. More

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W a v e s Pl u g - I n s Wo r k s h o p : M i x i n g b y t h e B u n dl e

often than not, you’ll be presented with a pile of audio files all called “track-n,” and it will be up to you to put the puzzle together. When recording, please make it a practice to name all of your tracks before you start recording. You’ll thank me later.

Editing When you’re finished with tracking, most likely you will need to do some editing. The first thing I do is trim out all the garbage before, after, and between the usable audio on each track. Another staple of editing is comping. This consists of cutting together a single track from multiple takes, where you combine the best parts of each take. There are times when a single track will have a very different feel in different parts of the song. Splitting up these parts onto two or more tracks allows you to set up different processing for each “feel.” For example, splitting a lead vocal into one track for verses and one for choruses is a common practice. Sometimes you have a track—or part of a track—that seems to be ahead or behind the beat. Time-slipping the audio to make it work better in the song can be beneficial. Another important aspect of editing is removing parts of tracks or entire tracks that don’t contribute to the song. You might have spent a great deal of time on a particular track; but if it’s not making the song better, don’t be afraid to cut it down or cut it out completely. The use of DAWs has made it much easier to completely rearrange a song at any point in the recording process. In the analog tape days, it was possible to do this with a razorblade and adhesive tape. This got the job done, but it was neither as easy nor as precise as editing with a computer. The drum tracks on the downloadable sample song are an excellent example of this. The drums were actually tracked for a completely different song and then rearranged to form the bed of this new song. Keep in mind that you can edit the life out of a song. If you line up all the parts so they hit at exactly the same time, it becomes difficult to distinguish the individual sounds. These slight timing differences all contribute to the feel of the song. Remember: If it ain’t broke, don’t fix it.

Mix Levels and Processing This is the stage of mixing on which this book will concentrate. At this point you have all the tracks you want to include in the final mix; this is where you fix sonic problems, improve the sound of tracks, and make everything work together as a cohesive whole. This is the step where everything really falls into place. If your mix doesn’t sound good after you’re finished with this stage, you should consider doing more editing or tracking to correct things that just aren’t working in the mix. Copious use of mix automation and heavy-handed mastering cannot improve your mix as much as making the tracks or the arrangement better.

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Mix Automation This is where you can sink huge amounts of time into a mix. Using a DAW for mixing gives you the ability not only to automate volume changes and panning, but also effect send levels as well as individual parameters within plug-ins. You can spend countless hours making minute changes to your mixes that nobody may ever notice. There’s a reason they call it “tweaking.” When I mix, I like to approach it in a traditional fashion, as if I were using a tape machine and a non-automated mixing board. I won’t use any level, pan, or effect automation until I have a solid mix established. That’s why I’ve created a distinction between this step and the previous one. I’ve found that if I get into automation too early in the process, I end up focusing too much on details and I lose perspective on the overall sound of a mix. In many instances you will find that you can get 90 percent of your mix dialed in before you get into any automation.

Mastering The last stage in the life of your song is mastering. This is where the last polishing and adjustments happen. Ideally, you will hand off your mixes to a mastering engineer who will be able to listen to them objectively and fine-tune them. The better your mixes are, the less the mastering engineer will need to do to them. Mastering can take a mediocre mix and make it sound much better, but it will never reach the quality level that can be achieved by the subtle mastering of a great mix. There are many benefits to using an outside mastering engineer. First of all, the mastering engineer has not been living and breathing your songs for weeks or months the way you have, so he will have a fresh perspective. He will also be listening to your mixes in a well-designed acoustic environment—one with which he is intimately familiar. If you don’t have the budget for professional mastering, it’s a very good idea to give yourself some time between the completion of mixing and the start of mastering to provide yourself with as much perspective as possible. You need to try to listen to the songs with a fresh ear so you can make objective decisions about what needs to be done.

Brief Discussion of Mixing versus Mastering When you’re working at home, it’s easy for the line to blur between mixing and mastering. Why not just throw a couple of plug-ins on the master fader in your mix and call it done? The most compelling reason to keep these two processes segregated is that if the only existing mixes of your songs include significant bus compression and limiting, then there will be little that a future mastering engineer can do to the mixes. You should create 24-bit unmastered mixes that you can archive for the future remastering of your greatest hits. There are times when you may want to use one or more plug-ins on the master fader to impart a particular color to the entire mix. The one thing that is nearly impossible to undo later is heavy compression or limiting, so avoid using those at the mixing stage.

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The Sample Project I’ve produced a song that I use throughout this book as a test bed for using and comparing Waves plug-ins. The song is titled “Running Aground,” and it was written by Mark Smith, Orren Merton, and myself. Here are the performance credits: Mark Smith: acoustic and electric guitars Orren Merton: lead vocals Frank Cotinola: drums Clare Carnegie: background vocals Barry Wood: keyboards, bass, background vocals

Mixes I’ve created 256-Kbps MP3 files of the mixes for each chapter. If you don’t download the individual tracks to try out the presets yourself, you can still hear the end results. There is even a mix for this chapter that consists of just the raw tracks with no processing at all.

Individual Tracks The audio files are provided as 24-bit, 44.1-kHz WAV files. All of the tracks are mono except for Tracks 9–13, which are stereo tracks. Here is a list of the tracks provided: 01-Kick-in.wav

Mic inside the kick drum

02-Kick-out.wav

Mic outside the kick drum

03-Snare-top.wav

Mic above the snare

04-Snare-btm.wav

Mic below the snare

05-Tom-hi.wav

High rack tom

06-Tom-mid.wav

Mid rack tom

07-Tom-low.wav

Low rack tom

08-Hi-hat.wav

Hi-hat

09-Overhead.wav

Stereo overhead mics

10-Room.wav

Stereo room mics

11-Loop1.wav

Bridge drum loop #1

12-Loop2.wav

Bridge drum loop #2

13-Loop3.wav

Bridge drum loop #3

14-Bass.wav

Bass recorded direct

15-Gtr-chords.wav

Distorted guitar

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16-Gtr-picked.wav

Distorted picked guitar

17-Gtr-verse.wav

Slightly distorted picked guitar

18-Gtr-solo.wav

Distorted guitar solo

19-Acoustic

Acoustic guitar

20-Mel-synth

Melody synth

21-Epiano

Electric piano

22-Synth-chorus

Chorus synth

23-Lead-vox-verse

Lead vocals for verses

24-Lead-vox-chorus

Lead vocals for choruses

25-Bvox-mel-F1

Female melody background vocals

26-Bvox-mel-F2

Female melody bvox double

27-Bvox-mel-M1

Male melody background vocals

28-Bvox-mel-M2

Male melody bvox double

29-Bvox-har-F1

Female harmony background vocals

30-Bvox-har-F2

Female harmony bvox double

31-Bvox-har-M1

Male harmony background vocals

32-Bvox-har-M2

Male harmony bvox double

33-Oooh-F1

Female ooohs

34-Oooh-F2

Female ooohs double

35-Oooh-M1

Male ooohs

36-Oooh-M2

Male ooohs double

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All of the audio files start at the same exact time, so you can drop them into your DAW and they will all line up. Because they have been consolidated, there is quite a bit of empty space in the audio files. Feel free to cut up the tracks and rip out the dead space. That will reduce the demands on your computer and hard drive, since that empty audio will not have to be read from disk.

Panning After you have the tracks loaded up in your DAW, you will have to do a little panning to get your mix to sound like the basic mix that I created. Figures 1.2 through 1.4 show the panning as it displays in my DAW of choice, Digital Performer. Different DAWs use different pan laws that control how the signal shifts as it’s panned, so the panning you do is not likely to be exactly the same as mine, but it should be close enough for rock ’n roll. Digital Performer uses pan values that range from 64 (full left) to þ63 (full right). The numbers that you see in Figures 1.2 through 1.4 are on this scale.

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Figure 1.2 The panning for the drum tracks.

Figure 1.3 The panning for the guitar tracks.

Figure 1.4 The panning for the background vocal tracks.

Master Fader Most DAWs use a high-resolution mix bus to sum all of the individual tracks, whether it’s a 32-bit floating-point bus like the one that Digital Performer and Logic use or a 48-bit extended fixed-point bus like Pro Tools HD uses. This higher resolution provides for a great deal of headroom, so you don’t have to worry about signals clipping at the output of each channel or overloading within the mix bus. What you do need to worry about is the final output level of the master fader. When a signal leaves your DAW—whether it’s going to the digital/analog (D/A) converters in your audio interface or being bounced down to a file—it will need to fit into a 16-bit or 24-bit pipe. If your signal is exceeding 0 dB at the output, it will clip, and you will have distortion. Fortunately, there is a simple fix for this: Turn down your master fader. Because the mix bus has a much greater resolution than your output, you can reduce the gain until you’re not clipping, and you will not make any sacrifices in resolution. The last step in setting up the sample mix is to set the master fader to 9 dB. This will attenuate the summed tracks enough to prevent clipping the master outputs and leave you with a little headroom.

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Preset Files I have created collections of Waves plug-in presets for all of the plug-ins that I used in each mix. I’ve created a zip file for each chapter that contains all of the preset files for that chapter’s mix. The names of the preset files are organized by chapter, track, and insert slot. For example, ch2_06-Tom-mid_A_RenEQ2.xps is the setting file for the Chapter 2 mix that is for a Renaissance EQ 2-band inserted on the mid tom track in the first (A) plug-in slot. See Figure 1.5.

Figure 1.5 The directories with the downloadable files.

Downloading The individual tracks, presets, and mixes are available for download on the Course PTR website at www.courseptr.com/downloads. n

The mixes directory contains one MP3 mix for each chapter of the book. The Chapter 1 mix is just the tracks all mixed together with no processing.

n

In the presets directory, you will find a zip file for each chapter.

n

The tracks-mono directory contains all of the mono 24-bit, 44.1-kHz WAV files.

n

The tracks-split-stereo directory contains pairs of mono files that are to be used as stereo files. If your DAW doesn’t support interleaved stereo files, you should download these.

n

The tracks-stereo directory contains interleaved stereo files. Download these files if your DAW supports interleaved files.

The individual audio tracks add up to a little more than a gigabyte, so be prepared for a lengthy download.

What’s Coming Up Next In Chapter 2, it’s time to get our hands dirty and dig into creating the first mix. It will be a challenge because the Musicians II bundle contains only six plug-ins.

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The Musicians 2 Bundle

W

aves has several affordable bundles, so I looked for the one that would be the best choice as a basic bundle for mixing. My selection was the Musicians 2 bundle, which provides many of the staple processors that you need for mixing. Although there are some important processors that are not part of this bundle, you can get a great deal done with the six plug-ins that it provides. The Renaissance series plug-ins that are included in this bundle have long been staple tools that I regularly use when mixing. For the purposes of this book, I’ve created categories that I use to group plug-ins of similar function. The Musicians 2 bundle contains one equalizer plug-in, three dynamics plug-ins, and two effects plug-ins.

Equalization The simplest way to think of equalization is as tone control. Like the bass and treble controls on your stereo, an EQ can make something that’s dull sound brighter and something that’s thin sound fuller. The equalizers used in audio production are typically more complex and more flexible than simple tone controls. With that increased control, you can use an equalizer in a corrective fashion to deal with tracks that might have certain annoying frequencies that you would like to get rid of. Most fledgling engineers fall into the habit of using equalizers only to boost frequencies. For example, the vocals may not be cutting through, so they boost the high end on them. This can be effective, but you can also run into problems with overemphasized sibilants. Another way to deal with that situation is to look for other instruments that are competing in the vocal frequency range and do some subtle equalization cuts to make space for the vocal. You can end up with the vocals sounding nice and present and a mix where all the elements can be heard because they’re not stepping on each other. Three main filter types are commonly found in equalizers: high-/low-pass, shelf, and bell. Figures 2.1 through 2.3 are graphs that illustrate the effect of the filter. The (horizontal) x-axis represents the frequency affected (from low to high), and the (vertical) y-axis shows boost and cut, with the center representing no boost or cut.

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The Parts of a Filter There are three common controls found in filters. n

The frequency setting determines the part of the audio spectrum upon which the filter is acting. This sets the center frequency for bell-curve filters and the corner frequency for shelving and high-pass/low-pass filters.

n

The boost and cut setting controls whether the filter is making the affected audio louder or softer. You will find this control on bell-curve and shelving filters.

n

The Q, or bandwidth, setting lets you control how sharp the EQ curve is. For bell-curve filters, it literally makes the affected area wider or narrower. For some shelving and high-pass/low-pass filters, it causes the slope of the filter to vary from gentle to sharp.

Figure 2.1 High-pass/low-pass filters.

High-pass and low-pass filters are sometimes known by their more intuitive names, low-cut and high-cut. As you can see in Figure 2.1, the shaded area on the left shows you what’s being removed with a high-pass filter, and the shaded area on the right shows you what’s being removed with a low-pass filter. Many microphones and microphone preamplifiers provide a high-pass filter to eliminate unwanted low frequencies when you’re recording. Although these filters are not the most glamorous members of the filter family, their utility is undeniable. If the filter you’re using provides a Q control, it will allow you to change the slope of the filter so you can vary the slope from gentle to steep. A steeper slope will let you be more precise in which frequencies are eliminated. High-pass filters are useful in eliminating unwanted rumble in a track. I don’t use lowpass filters as often, because they cause the track to sound dull almost immediately due to the immediate loss of the high frequencies. You can use a high-pass and a low-pass filter in conjunction for creative effects. The classic thin “telephone” sound can be achieved with a high-pass and low-pass filter used together, much like what you see in Figure 2.1. Shelving filters are similar to high-pass and low-pass filters in that the affected audio is all above or below the filter frequency. The difference is that you specify a boost or cut

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Figure 2.2 High- and low-shelving filters.

amount for a shelving filter. The shaded area on the left in Figure 2.2 shows a low-shelf filter that’s boosting everything below the filter frequency. The shaded area on the right shows a high-shelf filter that’s cutting the frequencies above the filter frequency. Some shelving filters provide a Q that allows you control the slope. Bell-curve filters are the most common type of filter you’ll run into. As the name suggests, the EQ curve is bell shaped. It’s centered at the selected filter frequency and only affects a certain number of frequencies above and below that center frequency. Some bell-curve filters have a fixed Q so you only have control over the frequency and the boost or cut. Bell-curve filters that also give you control over the Q are referred to as parametric EQs. In Figure 2.3, the shaded area on the left depicts a filter that is performing a narrow boost, while the shaded area on the right shows a filter that is performing a wide cut.

Figure 2.3 Bell-curve filters.

Making decisions about what filter type and what settings to use is a big part of the art of mixing. Sometimes I’m still amazed by how much an entire mix can be changed with a subtle EQ applied to a single track.

Renaissance EQ Even though the Musicians 2 bundle includes only one equalizer, the Renaissance EQ, it’s quite a versatile tool capable of almost any tone control needs you have. It comes in 2-band, 4-band, and 6-band versions. When the Renaissance EQ was released, computers had a fraction of the processing power that they now possess. With a modern computer, you probably won’t see any CPU load savings by using the 2- or 4-band version. That being said, I still have the habit of using the simpler versions when I know I’m not going to need six filter bands.

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This EQ strives to emulate the types of curves found in hardware equalizers. At the same Q setting, a bell filter will have a narrower bandwidth when cutting and a somewhat wider one when boosting. The Q setting also affects the shelving, high-pass, and low-pass filters, allowing you to control the slope and even introduce resonant peaks or dips at the corner frequencies. These resonances occur in many hardware EQs and can work really well in many instances. Unlike other EQs, RenEQ has two different types of shelving filters, a normal shelf and a resonant shelf. Each has different slope characteristics, and the Q setting affects each one differently. I suggest you try each one to see which works better on the material at hand. The stereo version of Renaissance EQ has one more trick up its sleeve. If you look at the bottom-right corner of Figure 2.4, you’ll see the Left, Link, and Right buttons. If you turn off the Link button and select each channel individually, the RenEQ can have independent EQ curves for each channel. This isn’t something you need every day, but on occasion it comes in very handy.

Figure 2.4 Renaissance EQ.

Dynamics Compressors, expanders, limiters, and gates all fall into the category of dynamics processors, which act on the volume, or amplitude, of a signal.

Limiters The limiter is the easiest dynamics processor to understand. The vast majority of limiters you will find in plug-in form are brick-wall limiters. The basic idea is that you set a volume threshold, and the limiter will not let the audio get any louder than that threshold. Typically, the threshold is expressed as a negative dB value, such as 6 dB.

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Zero dB represents a full-sample value, or the loudest signal that can be recorded to a file or output to a D/A converter without clipping. Limiters are useful in taming tracks that have sharp peaks that are popping out of a mix. Because digital brick-wall limiters are so good at stopping peaks, they are the last processor used in the mastering chain. They allow the overall level of the track to be boosted right up to full-sample value at the expense of the peaks that used to be there. The availability and abuse of digital limiters are largely responsible for the “volume wars.”

Compressors A compressor is similar to a limiter, but instead of refusing to allow the signal to exceed the threshold, it will simply dial back the volume as it goes over the threshold. The ratio setting controls how quickly the volume is reduced. A 2:1 ratio instructs the compressor to allow the output volume to increase by one decibel for every two decibels of signal increase above the threshold. A 5:1 ratio allows only one decibel of output for every five decibels of signal increase above the threshold. As you can see, the higher the ratio, the more the compressor clamps down on the volume. In the analog world, a compressor with a ratio of 10:1 or greater is considered a limiter. The ratio for a brickwall limiter is expressed as ∞:1, or infinity to 1. Because a compressor’s job is to turn down the volume, you generally need a way to turn up the overall output. Compressors often include a makeup gain control to provide this function. Compressors are useful in a number of situations. Reducing the dynamic range of a track can help the quieter parts from disappearing into the mix. Compressors can also imbue a track with attitude that it was previously lacking. The selection of the right compressor and its settings is another large part of the art of mixing.

Gates At the opposite end of the dynamics spectrum we find the gate. A gate will mute the output when the signal volume falls below the threshold. Gates can useful for automatically eliminating the background noise on a track when the vocal or instrument is not playing. Gates often have attack and release time parameters that control how quickly they open and close. In addition, more sophisticated gates will feature a Hold Time parameter that is typically expressed in milliseconds. It instructs the gate to wait for that period of time before it starts the release action. This can help prevent so-called “gate chatter” that can happen if the signal is hovering right at the threshold level.

Expanders A logic test for audio engineers might contain this question: “A limiter is to a compressor as a gate is to a(n)______.” The answer would be “expander.” Like a compressor, an expander will modify a signal’s level according to a chosen ratio, except that the ratios are inverted, and the affected signals are those below the threshold. For example,

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once a signal has fallen below the threshold, an expansion ratio of 1:2 will cause a signal drop by 2 decibels for each 1-decibel decrease in input level. A gate can be thought of as an expander with a ratio of 1:∞. Sometimes the sound of a gate acting on a track is too extreme and becomes distracting in a mix. In these cases, an expander can help clean up a track without making it sound chopped up.

Renaissance Compressor The Renaissance Compressor, shown in Figure 2.5, was one of the first digital compressors whose sound really impressed me. It has a very simple interface and effective metering, so it’s quite easy to use. The attenuation meter in the center immediately shows how much compression is being applied to the signal.

Figure 2.5 Renaissance Compressor.

In addition to using the traditional ratio setting, you can also affect the behavior of the compressor using the ARC (auto release), the Electro/Opto mode selector, and the Attack and Release settings. The Attack controls how quickly the compressor starts acting once the signal exceeds the threshold. The Release controls how quickly it stops compressing once the signal drops below the threshold. When ARC is enabled, it varies the attack and release times based on the input signal in an effort to mimic the way some hardware compressors work. The Electro and Opto settings also affect the attack and release times. Rather than trying to explain in excruciating detail how they affect the compression, I’m going to suggest that you just try both settings when using this compressor and then choose the one that works best for what you’re compressing.

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Finally, in the upper-right corner of the plug-in there is a button that can be set to Warm or Smooth. If you have the Warm option engaged, the compressor will add low-frequency harmonics that will increase as the gain reduction increases. This is useful because a signal can often sound thinner as it gets compressed more heavily. Using the Smooth setting will not add any harmonics. Under the hood, this plug-in contains both a compressor and a limiter. You can see how hard you’re hitting the limiter by watching the box at the top of the Gain control —the more often it lights up, the more limiting is going on. The limiter occurs after the output Gain setting, so pulling down the gain will cause less (or no) limiting to happen.

Renaissance Axx Compressors don’t get much simpler than the Renaissance Axx, shown in Figure 2.6. There are only three controls: Attack, Threshold, and Gain.

Figure 2.6 Renaissance Axx.

Reducing the Threshold will simultaneously increase the amount of compression and makeup gain. I’ve found that I almost always have to turn down the output Gain to compensate for the substantial volume increase caused by lowering the Threshold. Like the Renaissance Compressor, RenAxx also contains both a compressor and a limiter and has the same limiting indicator above the output meter.

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Renaissance Vox Renaissance Vox, shown in Figure 2.7, is another plug-in with a very spartan interface. You simply have Gate, Comp, and Gain controls. As you would expect, as you pull down the Comp control, you are increasing the amount of compression being applied. As with RenAxx, the compressor in RenVox also has makeup gain that automatically increases as you lower the threshold.

Figure 2.7 Renaissance Vox.

RenVox also includes an output limiter, but there is no indicator to show you how hard it’s working. You’ll just have to use your ears, which is always a good idea in any case. The fact that this plug-in has a gate makes it a particularly useful tool in the Musicians 2 bundle, because it’s the only plug-in in it with that feature. Technically its gate is really an expander, because the attenuation is proportional to how far the signal is below the Gate threshold. With the Gate threshold set all the way to the bottom (at inf), it will never engage. As you move the control up, you will eventually get to the point where the signal level falls below the Gate threshold and the gate starts working. Because the input level is displayed next to the Gate control, it’s easy to set. You do need to be aware that the main gain reduction meter in the middle will show you both the gain reduction created by the compressor as well as the gate.

Effects The plug-ins that fall into the effects category are quite a diverse group. An effect could be a delay, a chorus, a reverb, or any other strange modulation effect or oddball processor.

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Most plug-ins are available in mono-to-mono, mono-to-stereo, and stereo-to-stereo configurations. This refers to the input versus output channels. With EQs and compressors, you simply use a mono plug-in on a mono track and a stereo plug-in on a stereo track. Some effects have the ability to generate stereo output from a mono input. This is where you’ll want to use the mono-to-stereo version of a plug-in. Some DAWs present you with two different mono-to-stereo versions of a plug-in. Normally, the host will create a mono-to-stereo instance by using the stereo version of the plug-in and feeding the mono input to both the left and right inputs within the plug-in. This works fine in most cases, but there are some plug-ins for which Waves provides a special mono-to-stereo version that include additional controls not available in the host-provided mono-to-stereo plug-in. In my DAW, Digital Performer, I see three different versions of Doubler 2 when I bring up the plug-in list on a mono track: n

Doubler 2 (m->s) (mono to stereo)

n

Doubler 2 (s) (stereo)

n

Doubler 2 (s) (mono to stereo)

In Digital Performer, the first one in the list is the special mono-to-stereo plug-in that Waves has provided, and the last one is created by Digital Performer. If you see more than one mono-to-stereo instance on your DAW, you can determine which one is the proper mono-to-stereo plug-in by looking at Doubler. In the Waves version, you will not see the left and right channel-routing controls at the bottom of the plug-in. The Waves version knows that there’s only one input, so it omits those controls. Whenever you have two mono-to-stereo plug-in choices, it’s a good idea to always choose the Waves version.

SuperTap Even though I have literally hundreds of plug-ins on my system from many manufacturers, SuperTap is often my first choice when I need delay (see Figure 2.8). Its simple interface and powerful features provide most of what I need when delay is called for. SuperTap comes in two-tap and six-tap versions; but unless you need to set up complex rhythmic delays, the two-tap version is likely to do everything you need. In case you were wondering, the tap in SuperTap harkens back to the origin of delay effects, when they were produced by feeding a signal to a separate analog tape machine. The time it took the tape to travel from the record head to the playback head determined the delay time. Much like adding another spigot to a pipe, to get an additional echo you would add another playback head. One of my favorite features in SuperTap is the modulation, which adds some nice motion to the delayed sound by varying the delay times. With the Modulation Rate,

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Figure 2.8 SuperTap 6-Taps.

you have control over how fast the modulation is, and the Depth controls how much you’ll change the delay time. Modulation Modulation is something of a catchall term that simply means change. The modulation is often controlled by a low-frequency oscillator, or LFO, which creates constant sweeping changes in an effect parameter. A common example of this is a tremolo effect, which uses an LFO to vary the volume of a signal.

Another feature that’s a personal favorite in SuperTap is the tap filter. The tap filter gives you EQ on each individual tap, which allows you to tailor the sounds of the taps. Putting low-pass or high-shelf filters on the taps can give you a nice retro delay sound, while doing the opposite can create a delay that’s prominent in the mix yet doesn’t take up space in the lower frequencies. The feedback can act as an additional delay line when in Tap Feedback mode; in Normal mode, it controls how much feedback is applied to the existing taps.

Doubler As its name implies, the Doubler plug-in can make your tracks sound as if they’ve been doubled (see Figure 2.9). When you double a track by re-singing or re-playing a part,

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Figure 2.9 Doubler 4.

you’ll get slight variations in timing and pitch. This can thicken the track and make it pop out of the mix. The Doubler plug-in attempts to replicate this effect by delaying and pitch-shifting the original signal. Don’t assume that all that Doubler does is create subtle doubling effects. With its delay, feedback, EQ, octave shifting, and generous modulation controls, it’s capable of generating some pretty strange sounds. Doubler comes in four-voice and two-voice versions, just in case you’re short of CPU power.

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The Mix In each following section, I discuss the various plug-ins that I used to create the mix. I will talk about why I chose each plug-in and what I did with it. If you download the audio tracks and presets, you can bypass each plug-in so you can hear the before-and-after processing effects. I suggest that you do this not only with the track soloed, but also while listening to the entire mix. There will be instances when the processing I did to the track sounds odd on its own but makes sense within the context of the mix. Table 2.1 provides a list of tracks on which I used plug-ins. Slot A is the first insert on the track, and Slot B follows. The order in which the plug-ins occur on a track can

Table 2.1 Musicians 2 Bundle Mix Track

Slot A

01-Kick-in

RenEQ2

06-Tom-mid

RenEQ2

07-Tom-low

RenEQ2

08-Hi-hat

RenEQ4

10-Room

RenEQ2

11-Loop1

RenEQ4

13-Loop3

RenEQ4

14-Bass

RenComp

15-Gtr-chords

RenEQ4

16-Gtr-picked

RenEQ4

18-Gtr-solo

RenVox

19-Acoustic

RenAxx

21-Epiano

Doubler2 (m->s)

22-Synth-chorus

SuperTap2 (m->s)

23-Lead-vox-verse

RenVox

24-Lead-vox-chorus

RenComp

Slot B

RenEQ4

Doubler2 (m->s)

The (m->s) at the end of a plug-in name denotes that a mono-to-stereo instance of the plug-in is to be used.

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affect the end result substantially. If a particular track doesn’t appear in the list, then I haven’t put any plug-ins on it.

Drums In my experience I’ve found that at least half the battle in getting good drum sounds is using a good drummer who’s playing a good kit. In this case, I definitely got both. The other big factor is recording in a nice-sounding room. Beyond that, there are numerous ways to record drums. I’ve recorded them with as few as three microphones and as many 15. Because I recorded these in a great-sounding room, I went ahead and put up some room mics in addition to the overheads and the normal close kit piece mics. The first thing I noticed in the overall dry mix of the drums was that the kick needed a little help in anchoring the song. Listening to the inside and outside kick mics, you can hear that the inside mic is definitely picking up more of the bottom end. To give it more heft, I gave it a fairly narrow 8.3-dB boost at 42 Hz with RenEQ (see Figure 2.10). Combining that with the nice top end that the outside mic was providing makes for a nice, punchy kick sound.

Figure 2.10 The EQ on the kick inside track.

The mid and low toms also needed a little low-end boost to help them be heard over the chunky, distorted guitars, so they each got around 4 dB of boost, as shown in Figure 2.11. I used a higher Q on the mid tom track to boost a narrower range of frequencies so that I didn’t make it too boomy. To dial in the tom EQs, I looped the playback over the pre-chorus and tried different frequency, boost, and Q settings until I was happy with where they were sitting in the mix.

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Figure 2.11 The EQ on the mid tom (left) and low tom (right).

I’ve found that the hi-hat mic often captures much more than is needed for the mix. You have hi-hat in the overhead mic as well as bleed on the snare and tom mics. I almost always end up putting a high-pass filter on the hi-hat track. In this case I’m using that as well as a slight high-shelf boost, as seen in Figure 2.12. What I’m left with is a track that I can use to add definition to the hi-hat as needed.

Figure 2.12 The EQ on the hi-hat track.

The last thing I did to the drums was to give the room-mic track a little high-shelf EQ at 5 kHz. In Figure 2.13, you can see that I pushed the Q to its max value of 1.41 to give me a steep slope and a slight dip below the shelf boost. I found that gave me a little snap to the snare sound that I really liked. The small dip cut down a bit on the upper midrange and cleaned up the sound a bit.

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Figure 2.13 The EQ on the room track.

Loops There are three stereo drum loops that make an appearance at the bridge. Each one of them has quite a bit of low-frequency content, so I decided a culling was in order. After muting the tracks and listening to the bridge with one loop at a time, I decided that Loop 2 was going to be the one that would carry the low end. On Loop 1, I used a low shelf at 92 Hz to reduce the low frequencies there by 7.1 dB. I also gave it a boost at 1.5 kHz to bring out some of the midrange elements (see Figure 2.14).

Figure 2.14 The EQ on the Loop 1 track.

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Loop 3 comes in at the halfway point on the bridge, and it got a similar treatment with a low-shelf cut and a high-end boost, this time at 3.5 kHz (see Figure 2.15).

Figure 2.15 The EQ on the Loop 3 track.

With all the boosts and cuts in place, the loops now worked better as a group, and the low end wasn’t so overpowering that the live drums sounded weak coming in again after the loops.

Bass The comment I made about using a good drummer and a good drum kit also applies to getting good bass tracks. A skilled player with a quality instrument makes all the difference. Unfortunately, that was not the case for the bass track on this song. Because bass is not my primary instrument, my playing technique leaves a lot to be desired. The bass itself is a swap-meet special that may have been an Ibanez once upon a time. I initially tried using RenAxx on the bass, but it didn’t seem to help much. I settled on RenComp (see Figure 2.16) because it gave me more control over its behavior. I set it up with a 2:1 compression ratio and a low threshold so that most of the time it’s acting to even out the dynamics of the performance. I set the attack to about 60 ms to let the attack come through before the compressor kicks in. I also set the release to around 40 ms so it would quickly back off on the gain reduction shortly after the signal fell below the threshold. I used the Smooth character (rather than Warm) because there was already plenty of low end in the track.

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Figure 2.16 RenComp on the bass track.

Guitars The guitar chords track (15-Gtr-chords) is playing big, ringing, distorted chords. To give this track some of its own space, I added a little RenEQ (see Figure 2.17) at 733 Hz and a touch at 7.6 kHz. This emphasized the midrange and gave a little bite on the top that seemed to help it drive the song better.

Figure 2.17 The EQ on the guitar chords track.

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The picked guitar track (16-Gtr-picked) plays the signature guitar part during the choruses and chunky, muted chords in the pre-choruses. Due to the choice of guitar, amp, and miking, this track ended up with way too much low end, particularly on the chunky chords. To deal with this, a high-pass filter once again came to the rescue. I tried to set it so that I wouldn’t lose too much of the low end on the picking parts, but so I would still tame the chunks. It does lose a little bit of power in the picking parts, but because the bass doubles much of the guitar part, you don’t really notice the loss. To give it a little bite, I also added a small boost at 1.5 kHz (see Figure 2.18). Although the difference between the sound of the two sections could be handled with effect automation or by splitting the guitar part onto two tracks, leaving it as one track presents a useful mixing challenge.

Figure 2.18 The EQ on the picked guitar track.

The guitar solo track has some fairly nasty ground-loop noise, which is not uncommon when you’re recording with a lot of distortion. It can be caused by bad grounding in the electrical circuit into which the amplifier is plugged, or it can be caused by the guitar pickups or guitar cable picking up noise from cheap light dimmers or other radio frequency (RF) sources. In any case, we have to deal with it on this track. My solution was to use the gate that’s part of the RenVox plug-in (see Figure 2.19). Just because it’s a plug-in designed for vocals, that doesn’t mean you can’t use it on something else. The gate does a pretty good job of cleaning up the noise in between the guitar phrases in the bridge. When the guitar is playing, you don’t really notice the noise the way you do when the guitar stops playing. I’m not using the compression portion of the plug-in at all. I don’t often need to compress distorted guitars, because the distortion itself does a very good job of smoothing out the dynamics.

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Figure 2.19 RenVox as a gate on the guitar solo track.

With the drums and electric guitars all flailing away, the acoustic guitar needs a little help to be heard. In the last part of the verses, it’s playing a tight percussive part; and in the pre-choruses, it’s playing the figure from the intro. I started with RenAxx (see Figure 2.20) to give it some attitude. The compression brings out the

Figure 2.20 Renaissance Axx on the acoustic guitar track.

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pick sound, which helps the more aggressive strummed parts cut through. Even with this compression, the picking in the pre-choruses is pretty hard to hear, so at this point it is just contributing subliminally. I followed RenAxx with a RenEQ instance (see Figure 2.21) to give the acoustic a fuller sound. Some boosting at 238 Hz and more at 823 Hz gave it a much more round character.

Figure 2.21 Renaissance EQ on the acoustic guitar track.

Keyboards I felt that the electric piano could benefit from being made wider and exhibiting some motion. I used a mono-to-stereo instance of Doubler 2 (see Figure 2.22) to create a

Figure 2.22 Doubler on the electric piano track.

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chorusing effect. I simply panned each of the voices hard left and hard right and detuned them in opposite directions (in other words, sharp and flat). I then added a touch of modulation to the detuning amounts to produce a chorusing effect. The chorus synth was also too dry for my taste. Because the vocals, bass, and guitars were filling up the center of the mix, I used a mono-to-stereo two-tap instance of SuperTap (see Figure 2.23) to spread out and liven up the synth part. The left and right delay times are 192 ms and 264 ms, respectively, and there’s a small amount of modulation to keep it interesting. The synth part was already pretty thick-sounding, so I used bell filters on both of the taps so that they wouldn’t cause the synth part to overpower the other parts of the mix. This is one of those tracks that I call an “icing” track because it adds a little interest and variation to the song rather than being one of the crucial parts of the mix.

Figure 2.23 SuperTap 2-Taps on the chorus synth track.

Vocals The lead vocals on the verse are sung in a quiet voice that is easily lost behind the other elements. The RenVox plug-in (see Figure 2.24) does a nice job of bringing out the subtle parts without making them sound particularly compressed, and the gate does a pretty good job of deemphasizing the breaths in between the vocal phrases. I could have edited out the breaths completely, but I find that the breathing between phrases sounds natural (unless it’s too obtrusive).

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Figure 2.24 RenVox on the verse lead vocal track.

The lead vocals on the chorus are sung in a completely different style from the lead vocals on the verse, so I split them out to their own track in order to process them independently. On this track I started with RenComp (see Figure 2.25) to keep the vocal sitting on top of all the other tracks in the mix. I’m running a 4:1 ratio and getting

Figure 2.25 RenComp on the chorus lead vocal track.

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Figure 2.26 Doubler on the chorus lead vocal track.

between 3 and 6 dB of gain reduction. In this instance I’m using the Warm setting so that the vocal maintains its low end even under heavy compression. I’m following the compressor with mono-to-stereo Doubler 2 (see Figure 2.26) to give just a bit of a doubling effect to it. In this mix I’ve left the background vocals untouched because they were blending pretty well. The panning that I’d set up helped separate them from the lead vocals.

What’s Coming Up Next Although the Musicians 2 bundle provides some solid processors for shaping your mix, there are still some pretty big holes in the tool set, which is to be expected when you have only six plug-ins to work with. The Silver bundle will fill a couple of these holes and will also provide a very useful plug-in that doesn’t do anything to the signal.

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he Silver bundle introduces 11 new plug-ins that provide some important tools missing in the Musicians 2 bundle—namely, reverb and limiting. It also contains several other interesting additions that I’ll discuss in this chapter. The Silver bundle includes all of the plug-ins found in the Musicians 2 bundle except for Renaissance Vox.

Equalization In the Musicians 2 bundle, we had exactly one choice for equalization. The Silver bundle expands on this with two additional plug-ins that fall into the equalizer category— one workhorse EQ and another very specialized low-frequency equalizer.

Q10 Paragraphic Equalizer The Q-series plug-ins were the first products to be released by Waves back in the early 1990s. They come in 1-, 2-, 3-, 4-, 6-, 8-, and 10-band versions (see Figure 3.1). Back in the dark ages of personal computer audio (ca. 1993), the amount of processing power your computer had was extremely limited. The CPU requirements of Q2 versus Q4 were readily apparent, so it was important to choose wisely so as not to bring your computer to its knees. These days you could run hundreds of instances of Q10, and a typical laptop machine would hardly break a sweat. The Q-series plug-ins are very capable equalizers that can cleanly accomplish a wide variety of equalization tasks. When I need to drop in a high-pass filter to eliminate some unwanted rumble in a track, Q2 is often the first plug-in I reach for. The Q control in the filters has a wide range of settings and can create bell filters spanning many octaves all the way down to a very narrow band of frequencies. The Q only applies to the bell filter, which means that the shelving and high-pass/low-pass filters all have fixed slopes. Like the Renaissance EQ, the Q-series plug-ins allow you to create completely different settings for the left and right channels. This is not something you’ll use every day, but when you need it, you’ll appreciate it.

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Figure 3.1 Q10 Paragraphic Equalizer.

MaxxBass Calling MaxxBass (see Figure 3.2) an equalizer is something of a stretch, but since it does primarily affect the frequency content of the signal, I put it into this category. The goal of MaxxBass is to add upper harmonics to the low-frequency content of a signal so that it’s possible to perceive those low frequencies on a playback system that cannot reproduce the fundamental bass frequencies. On a small, tinny speaker, it’s possible for you to “hear” the bass even if the speaker can’t reproduce the bass’s fundamental frequency. Any signal more complex than a sine wave will have harmonic content that’s above the fundamental frequency of the note. Even when you cannot hear the fundamental frequency, if you can hear the harmonics, your brain can fill in the missing information and allow you to perceive the bass. MaxxBass gives you control over the portion of the signal that is passed to the harmonic processor, and also gives you control over both the level and the range of those harmonics. This can be a tricky plug-in to get dialed in, but MaxxBass helps you out

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Figure 3.2 MaxxBass.

here with its monitoring controls. The lower-right of the plug-in contains three buttons: Audio, MaxxBass, and Original Bass. When Audio is selected, you will hear the original signal as well as the generated harmonics. The MaxxBass button will cause only the harmonics to be output. The Original button will let you hear just the signal that’s being fed to the harmonics processor. This plug-in also includes a simple dynamics processor that will compress only the generated harmonics. The Response setting affects both the attack and release of the compressor, with lower values giving you a faster attack and release.

Dynamics In moving from the Musicians 2 bundle to the Silver bundle, we lose the Renaissance Vox plug-in; but in its place we gain three useful dynamics plug-ins. Sidechaining Sidechains are found on many hardware and software compressors. All dynamics processors feature a detection circuit (or code in the case of plug-ins), which determines the amplitude of the signal that it then uses to determine how it’s going to process the audio. Incoming signals are split and fed to both the detector and the processor simultaneously. A sidechain will let you modify the signal that’s being fed to the detection circuit without changing the actual audio that’s being processed. This allows you to do things such as filter out the low frequencies in the sidechain so that the threshold is not affected by the low end. You can also fashion your own de-esser with a compressor sidechain by boosting the sibilant frequencies that you want to remove so that you will get more gain reduction when those frequencies are present. A more thorough explanation of de-essing is provided later in this chapter.

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Some plug-ins and DAWs provide the ability to route an entirely different signal into the detector. One technique that uses this scheme is ducking. The classic use of a ducker causes music to automatically become quieter when a voiceover is occurring. You would do that by putting a compressor on the music and feeding the voiceover to the sidechain input. When there’s talking, the compressor will then reduce the gain on the music.

Digital Performer’s external sidechain routing.

These are the Waves dynamics processors that currently support external sidechain inputs: Renaissance Compressor, Hybrid Compressor, Renaissance Channel, V-Comp, Vocal Rider, C6-SideChain, and API 2500. You can use external sidechains in all sorts of creative ways.

C1 Parametric Compander Like the Q-series equalizers, the C1 plug-ins (see Figure 3.3) also date back to the early days of Waves’ history. Because C1 was created when computers didn’t have much CPU power, Waves offered the C1 in the following configurations: C1 comp-gate, C1 comp-sc, C1 gate, and C1 comp. C1 comp-gate includes all of the processing modules, the compressor, the gate/expander, and the sidechain. C1 comp-sc includes the compressor with the sidechain. C1 gate just has the gate/expander module. C1 comp just contains the compressor. Although these days there’s no reason not to just use C1 comp-gate and forget about the others, I find that if I just need some gating, I will use C1 gate to make it easier for me to remember what I’m doing with the plug-in without having to open it up and look at the settings. When I have a track with particularly troublesome dynamics, the wealth of control that C1 offers will generally provide a way to fix it. As you may have gathered from the look of the plug-in interface, C1 is quite the Swiss Army knife of compressors. Rather than describe in detail each of the plug-in parameters (which would be a book in itself), I encourage you explore the many presets that come with C1. Trying out the presets and playing with their settings will really help you explore some of its more unique features (such as negative ratios and band splitting) and understand the power that they provide.

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Figure 3.3 C1 Parametric Compander.

L1 Ultramaximizer The L1 limiter (see Figure 3.4) is the progenitor of Waves’ substantial limiter offerings. This humble peak limiter helped usher in the era of heavily peak-limited tracks coming out of mastering studios worldwide. Prior to the introduction of the digital brickwall limiter, it was impossible to make a perfect limiter that would absolutely prevent a signal from exceeding the threshold. The combination of the digital delivery medium of the CD and digital brick-wall limiters provided the perfect storm to create really loud audio at the expense of dynamic range. Aside from its part in the “loudness wars,” L1 is still a useful weapon in your audio arsenal. There are often instances when you have a track that contains peaks that just jump too far out of the mix. You can turn down a problem track like that, but then you’re likely to lose the track in the mix. Putting a compressor on the track can help, but if the peaks are loud enough and brief enough, a compressor won’t be fast enough to stop compressing the material following the peak. This is where a limiter can really shine because it can surgically reduce the gain on just the peaks and leave the rest of the track as is.

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Figure 3.4 L1 Ultramaximizer.

The Threshold control on L1 does two things simultaneously: It will lower the threshold where the limiting occurs, and it introduces makeup gain that increases in equal proportion to the lowering of the threshold. This means that as you reduce the threshold, you are also increasing the level of the track. A feature included in L1 is the ability to change both the Threshold and the Output Ceiling simultaneously. You just grab the blue triangles between the two controls, and you’re now changing both controls. This will let you really hear the effect of the limiting because the overall loudness of the track will remain the same while the amount of limiting changes. It’s easy to introduce too much limiting while simply changing the threshold because of the natural tendency to perceive louder as better. L1 comes in two versions, the normal L1 and L1þ. The latter includes Waves’ dithering tool. The only time you should need to use the dithering in L1þ is when you’re using the limiter as the last plug-in on the master fader. Dither is essentially very low-level noise that’s mixed in with the signal to help preserve the lowest bits of your signal. Without dither, you’ll have either truncation or rounding, both of which are less desirable than proper dithering. Personally, I don’t worry about dithering too much when outputting 24-bit files, but it is essential for 16-bit files.

DeEsser The primary use of a de-esser is to reduce unwanted sibilance in a vocal track. Sometimes when you get a vocal sounding nice and present in your mix, the “ess” and “sh” sounds of the words are far too prominent. A de-esser is essentially a compressor with a sidechain that focuses the compression on just the sibilant sounds. Although you can get the same results from the C1 comp-sc plug-in, the DeEsser provides that function in an easy-to-use package.

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The presets provide a good starting point for setting up the DeEsser. You will need to pull down the Threshold setting until you see it acting primarily on the esses. If you’re having trouble getting the results you want, you should try soloing the track and click on the Side Chain button shown in the lower left of Figure 3.5. That will let you hear just the sidechain signal, and that can help you in adjusting the frequency and sidechain filter type until you’re hearing just the “esses” that you’re trying to remove. When you’ve zeroed in on those annoying sibilants, you will see the attenuation meter showing gain reduction on just the sibilants.

Figure 3.5 DeEsser.

If you find that DeEsser is causing compression on parts of the signal that should be left alone, you should try setting the Audio button to Split rather than Wideband. In Split mode the DeEsser will compress just the “ess” portion of the signal. In this mode the Freq setting will set the center frequency of the sidechain bell filter rather than the corner frequency of the high-pass filter that’s provided in Wideband mode. If you switch the Audio mode after setting the sideband frequency, you may need to monitor the sidechain signal again to make sure that you’re properly targeting the sibilants. De-essers are not just for vocals. Any track that has sibilants that are sticking out is a potential target for a de-esser. I’ve been able to use a de-esser to tone down an overbearing hi-hat that’s dominating an overhead track.

Effects You can use effects plug-ins inline either on a specific audio track or on an aux track where several tracks can be routed to it via a send. Equalizers and compressors are typically used as inline effects because their job is to completely change the audio rather

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than to add something to it. When using an effect inline, you will need to control the wet/dry mix within the plug-in to get the desired balance of original signal and effect. Effects such as reverb are often used as send effects, allowing you to add an effect to multiple tracks without having to put the same plug-in inline on every track. When you use an effect on an aux track, you will need to set it so that there is no dry signal in its output (frequently referred as 100% wet), since the track that’s sending is already included in the mix. To make it a little easier for you to replicate each chapter’s mix, I’ll be using more inline effects than I might normally use. This way, it’s just a matter of instantiating the effect on the track and loading the preset, rather than establishing a send bus and an aux track. In this chapter’s mix, I will put a reverb on a send so that I can route multiple guitar tracks to the same reverb. Using the same reverb on multiple tracks can give a mix a more cohesive character because you will have tracks placed within the same virtual space. Phase and Polarity

The concepts of phase and polarity are often misunderstood. A single cycle of a waveform can be described as going through 360 degrees—0 degrees being the start of the cycle and 360 being the end of the cycle. In the case of a symmetric waveform, such as a sine wave, the 180-degree point occurs as the signal passes through the zero point on its way to the negative portion of the wave. If you shift the phase of a signal, you are actually delaying its start.

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The polarity of a signal can have two states, positive and negative. When the polarity of a signal is inverted, what was above the zero line is now below, and what was below is now above. Often people use the phrase “180 degrees out of phase” or “phase invert” synonymously with “polarity reverse.” When you are looking at a steady-state sine wave, the two techniques produce similar results, but you will get very different results with any signal more complex than that. If you duplicate a track and invert the polarity of one of them, they will cancel out each other perfectly when played at the same time. You can get partial phase cancellation with improperly placed mics in a multi-mic recording. There can be enough phase cancellation between the signals to cause problems when listening to them mixed together.

MondoMod MondoMod (see Figure 3.6) is a fun little plug-in that processes a signal with amplitude modulation (AM) and frequency modulation (FM). You might recognize amplitude modulation by its more common name, tremolo, and frequency modulation as vibrato. In addition to processing AM and FM signals, MondoMod can also pan your signal around the stereo field.

Figure 3.6 MondoMod.

In its most mundane application, MondoMod can generate simple tremolo and vibrato, but it can go well beyond those into sounds that can make you downright dizzy. The processing is all driven by an oscillator that can be set to arbitrary frequency or can be synchronized with the tempo of your project. You can also apply a multiplier to easily get various multiples and fractions of the selected speed. You can select one of

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five different waveforms for the oscillator. Both the AM and FM are controlled by this oscillator, but you can offset them from each other by using their phase controls. If one of the phase settings is at 0 and the other is at 180, then the AM and FM will be sweeping in opposite directions. This is another plug-in that deserves some preset test driving to get a feel for the range of sounds it’s capable of producing.

Enigma Waves came up with the perfect name for this plug-in. It’s kind of flanger and kind of a delay, but really it’s neither—yet it can produce sounds reminiscent of both types of effects. This plug-in started life as an internal tool at Waves for developing flangingand phasing-style processors. Fortunately for us, Waves decided to shine it up and release it into the world. Enigma (see Figure 3.7) has two major components—the notch filter processor and a feedback network processor. Each of the components has controls for what is essentially a 2-band equalizer with low-shelf and high-shelf fixed-slope filters. The gain for these filters is controlled with the x Factor, which ranges from 0.10 (full cut) to 2.00 (full gain). At 1.0 there is no boost or cut.

Figure 3.7 Enigma.

The notch filter processor that’s in the top half of the plug-in can be static, or it can be modulated by an oscillator that is identical to the one found in MondoMod. Below the

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modulation oscillator controls, there are Min Freq and Max Freq controls that are represented in the notch filter display by thin vertical gray lines. These two values control the frequency range between which Enigma will sweep. This is another example of the unique control that Enigma provides. The output of the notch processor is sent to the feedback network, which will process the signal and send it back into the input of the notch filter module. The feedback network can be disabled completely by clicking on the Feedback button. To further manipulate the phase of the effect, there are several controls labeled with the Ø symbol that invert the polarity of the associated element. Since effects such as Enigma rely on changing phase relationships between parts of the signal, these polarity changes can be useful in achieving interesting results.

TrueVerb This was Waves’ first foray into the world of reverb. TrueVerb (see Figure 3.8) is what’s referred to as an algorithmic reverb. It consists of code that was written to simulate the sound of reverberation in a room. You are given a plethora of controls over many aspects of the room simulation, which will allow you to tailor the quality of the reverb to fit your needs. It also allows you to create spaces that don’t exist in the real world.

Figure 3.8 TrueVerb.

The two major components of reverb are the early reflections and the reverb tail. The early reflections are the initial short echoes you get when the sound first bounces off the

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nearby walls. This is what gives the listener the first cues as to the size of the room. As sound bounces around the room, you will no longer hear individual echoes as they merge into an aural stew of sound. The amount of time it takes the reverb tail to start is another room-size cue for the listener. The amount of time it takes the tail to fade away also gives the listener a feel for how big and how reflective the walls of the room are. As you increase the Room Size and Distance parameters, you will hear immediate changes to the perceived size of the room and your position within it. The bottom half of the plug-in contains the frequency response controls that let you decide what your virtual room is made of. Highly reflective surfaces, such as glass and rock, will bounce the high frequencies right back at you with little attenuation, while wood- and fabric-covered walls will absorb more of these frequencies. Changes to the Reverb Damping settings here will affect the overall high- and low-frequency reverb time. You can also change the character of the early reflections using the Dimension and Decorrelation controls. With the smaller room sizes, you may find that there is some resonance or notch filtering that is not working with what you’re putting into the reverb. Changing these settings will alter the early reflections and can give you a sound that works better for you. RT60 is a term that you will find in used in the description of the acoustics of a room. The value is expressed in seconds and represents the amount of time that it takes for the volume of the reverberation volume to drop 60 dB in relation to the original sound. A 60-dB drop will basically be where you don’t hear the reverb any longer. This is a handy yardstick to use when comparing different reverbs and when changing the size of a reverb.

IR-L Convolution Reverb Waves IR-L reverb plug-ins (see Figure 3.9) use a technique called convolution to create reverberation. Convolution is very different from the algorithmic approach that TrueVerb uses. Instead of using code written to simulate the sound of a room, a convolution reverb uses measurements that have been made of actual spaces. An impulse or frequency sweep is played in the room and is recorded. Then special programs are used to process those recordings into impulse response files that the convolution reverb uses to reproduce the sound of the space. IR-L comes into two versions, efficient and full. The efficient version uses one convolution for the left channel and one for the right channel. The full version uses two convolutions per channel that are dedicated to processing crosstalk between the channels. The efficient version approximates this crosstalk by giving you a control that feeds some of the left input channel to the right convolution and some of the right input to

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Figure 3.9 IR-L Convolution Reverb.

the left convolution. The full version will require twice the processing power, but it will result in more realistic results. IR-L comes with four impulse-response presets—Hall, Room, Studio, and Plate. If you are a registered owner of IR-L and you are currently covered under the Waves Update Plan (WUP), you can go to www.acoustics.net, where you can download more than 100 impulse-response files that Waves has provided. The interface of IR-L is quite spartan when compared to TrueVerb. The Reverb Time slider will affect the overall reverb time, and the change will be reflected in the RT60 time. The other two controls are Cnv. Start and Cnv. Length, which allow you to trim the start and end of the convolution. Trimming the end will give you a gated reverb-like effect, and trimming the start will eliminate early reflections. If you find that the initial reflections are too bright for your application, a little trimming of the start can tone them down or eliminate them entirely. If you find that you’re pushing your computer’s CPU too hard, you can lighten the load caused by IR-L by clicking on the Full CPU button to change it to Low CPU. This will reduce your computer’s CPU use by around 20 percent, at the expense of the resolution of the convolution process. For certain styles of music, convolution reverbs are ideal; but for denser multitrack mixes, the malleability of an algorithmic reverb can be a better choice. In Chapter 8 you will read about the full version of IR-L, called IR-1, which uses the same impulseresponse files but gives you algorithmic-style control over the sound of the reverb.

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Other Plug-Ins Some plug-ins don’t fit neatly into my categories, so I had to call them something. These run the gamut from spatializing tools to noise reduction and analysis tools. Mid/Side Processing Mid/Side, or M/S, is best known as a stereo recording technique that utilizes a center mic (typically cardioid) and a side mic that must use a figure-8 pickup pattern. The mid mic is pointed directly at the sound source, and the side is positioned so that it picks up the signal to the left and right of the mid mic. Those two tracks can then be processed to create a stereo signal by splitting the figure-8 track out to two channels, with one hard-panned left and the other hard-panned right and with the polarity inverted on one of the channels. Mixing in the side channels will result in a nice stereo sound that has excellent mono compatibility. This is because when summed to mono, the side channels cancel each other out perfectly, and you’re left with the mid channel. Anyone who has recorded a stereo track only to find that in mono the track all but disappears will appreciate the mono compatibility that M/S miking gives you.

You can also use M/S techniques to isolate the mid and side portions of an already recorded stereo signal. The Center plug-in that you will read about in Chapter 6 utilizes this technique internally to accomplish its processing.

S1 Stereo Imager Another set of venerable Waves plug-ins is the S1 Stereo Imaging series (see Figure 3.10). There are three different versions of the plug-in provided: S1 Imager, S1 Shuffler, and S1 MS Matrix. All of the S1 plug-ins require a true stereo input. If you feed them the same signal on the left and right, it will have no effect because S1 manipulates the phase relationships between the two channels to achieve its effect.

Figure 3.10 S1 Stereo Shuffler.

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The primary parameter is the Width control. A Width of 1.00 will leave the stereo field untouched. As you increase the width, you will hear the stereo field expand. Be careful, because it’s easy to be seduced by a nice, wide stereo field, but it comes at the expense of tonal changes and a loss of a solid center. This is an effect that’s best used on individual elements of a mix rather than on an entire mix. The Asymmetry and Rotation parameters will let you shift the overall position of the stereo field. This can be useful for left/right rebalancing when simple panning doesn’t seem to do the trick. The S1 Shuffler adds some controls that affect the bass frequencies. The phase manipulation that’s effective on the higher frequencies doesn’t work very well for spreading out lower frequencies. To accomplish this, the S1 Shuffler splits the low frequencies into a number of bands and alternately pans them left and right. You have control over the how high this shuffling goes with the Freq control and how prominent it is with the Shuffle control. There is also a bass trim that is an overall boost or cut for the bass frequencies. One nice feature in S1 is its ability to handle the M/S decoding for you. If you have a track that’s been recorded using M/S miking, you click on the Input Mode button to change it from L/R to M/S input. Below the Input Mode button are controls that allow you to invert the polarity of each channel. In between the polarity controls is the Channel Swap control. If you decide that you’d like to swap the left and right channels, or if you recorded your M/S track with the side mic on the left channel, you can click this, and the inputs to the plug-in will be swapped. Below the output meters you’ll find a Meter Mode switch that selects either normal left/ right metering or mid/side metering. The M/S metering allows you to see the change in the amount of side signal as you adjust the Width. Finally, the S1 MS Matrix, shown in Figure 3.11, is likely the simplest plug-in you’ll ever see. There are no controls whatsoever, and all it does is decode (or encode) an M/S stereo track. It’s not terribly useful for decoding tracks recorded as an M/S pair since there is no level control for the stereo signal. For that task, you’re better off using one of the other S1 plug-ins that will give you width control. You could, however, sandwich a plug-in, such as Q10 or RenEQ, between a pair of S1 MS Matrix plug-ins to equalize the mid and side

Figure 3.11 S1 MS Matrix.

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portions of a stereo signal separately. Each S1 MS Matrix plug-in will cause a 3-dB drop in level, so you need to be aware that your signal will be 6-dB quieter when processing it in this fashion.

PAZ PsychoAcoustic Analyzer This is the plug-in that I referred to at the end of the previous chapter as a useful tool that doesn’t actually do anything to the signal. PAZ Analyzer (see Figure 3.12) will show you information about your audio.

Figure 3.12 PAZ PsychoAcoustic Analyzer.

When something sounds strange and you can’t pin it down, the PAZ Analyzer can be an invaluable tool. Its frequency analyzer shows you what’s happening across the frequency spectrum, and its position analyzer gives you a good idea of how your mix is spread out across the stereo field. There is also a stereo peak meter with a mono-summed loudness meter between the left and right meters. Both the frequency analyzer and the position analyzer have peak hold indicators, which are represented by the orange lines you see in Figure 3.12. The frequency analyzer display is pretty self-explanatory. Two features that are not necessarily obvious are the zoom and the crosshair cursor. If you click in the frequency display, horizontal and vertical lines will display with a label to show you what the

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frequency and amplitude are at the point. This is great for pinpointing the frequency when you’re setting a de-esser or looking for an annoying resonance in a track. When I’m mixing, I put an instance of PAZ Analyzer on my master fader and leave it open so I have all that information available at a glance. With it on my master fader, I can solo tracks to see what’s going on with them. When I’m mixing I make it a habit to go through and briefly solo each track with the analyzer up, so that I can see whether there is any useless low-frequency information that I need to get rid of with a high-pass filter. Unless you have a substantial subwoofer and a well-designed room, it’s unlikely that you’ll be able to hear anything below 30 to 40 Hz; yet it often exists even on tracks that have no useful information in that range. For the most information, particularly in the lowest frequencies, you should set the LF res to 10 Hz and leave the Weight set to the two horizontal lines that represent no weighting. The position analyzer is a powerful tool for checking the overall phase of your mix. The center 90-degree spread in the display that falls within the blue graph lines represents all of the in-phase audio. If you pan a mono track from the left to the right, it won’t go beyond that 90-degree arc. Everything that occurs beyond that arc is out of phase. That means that the same frequencies must be occurring on the opposite channel at the opposite polarity. This is not necessarily a bad thing—S1, for instance, uses that kind of phase cancellation to spread out a stereo signal. Reverb, acoustic or simulated, will often naturally contain frequencies on opposing channels that are out of phase. In Figure 3.12 you can see the orange peak hold line showing that most of the audio occurs within the normal L/R stereo field, but that there are plenty of peaks that happened well into the out-ofphase areas. You can also see the gray area that indicates the current signal is all nicely within the in-phase area. That’s a pretty normal-looking position meter. What you don’t want to see is sustained activity outside of the L/R field. This would indicate that you’ve got some audio that’s seriously and consistently out of phase and ought to be corrected. That’s when you should start soloing tracks and groups of tracks until you’ve tracked down the offending audio. The PAZ Analyzer plug-in includes all of the modules. PAZ Frequency includes just the frequency analyzer portion, PAZ Position includes just the stereo phase meter, and PAZ Meters only includes the peak and loudness meters. Unless you’re particularly tight on screen space, I suggest that you use the full PAZ Analyzer plug-in.

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The Mix The biggest changes in the mix are largely due to the additional effect plug-ins that we gained access to with the Silver bundle. Overall, you should hear a better stereo field and more depth in the mix. For this mix I decided that I wanted to use the same reverb for several guitar tracks, so I created an auxiliary track to use for the reverb. Table 3.1 shows the send level for each

Table 3.1 Silver Bundle Mix Track

Slot A

Slot B

Slot C

01-Kick-in

RenEQ2

04-Snare-btm

Q1

06-Tom-mid

RenEQ2

07-Tom-low

RenEQ2

08-Hi-hat

RenEQ4

10-Room

RenEQ2

11-Loop1

RenEQ4

13-Loop3

RenEQ4

L1

14-Bass

RenComp

MaxxBass

15-Gtr-chords

RenEQ4

21.5 dB

16-Gtr-picked

RenEQ4

18.0 dB 25.5 dB

17-Gtr-verse 18-Gtr-solo

C1 gate

Q4

19-Acoustic

RenAxx

RenEQ4

21-Epiano

Q4

MondoMod (m->s)

22-Synth-chorus

SuperTap2 (m->s) S1 Shuffler

23-Lead-vox-verse

Q4

24-Lead-vox-chorus Q4 Effects 1

Send 1

Enigma (m->s) 29.0 dB

DeEsser

RenComp

RenComp

TrueVerb (m->s)

IR-L (m->s)

The bolded plug-in names in the table above indicate plug-in or settings changes between the previous chapter and this one. The (m->s) at the end of the plug-in names denotes that a mono-to-stereo instance of the plug-in is to be used.

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track that’s sending to the reverb. The send level is expressed in decibels, where 0 dB means that the full level of the track is being sent to the aux track. When creating the aux track, I set it up with a mono input because the reverb I chose would typically generate a stereo output from a mono input.

Drums When recording a snare with microphones both above and below the drum, you will often find that the two tracks have opposite polarity. This happens because as the stick hits the drum head, it creates a low-pressure zone for the top mic and a high-pressure zone for the bottom mic. If you look at the two snare tracks and zoom in to the sample level, as shown in Figure 3.13, you’ll see that although they have significantly different waveforms, the polarity in the lower frequencies is opposite. This will cause phase cancellation in the low end of the snare and make it sound thinner than it ought to.

Figure 3.13 Close-up of the two snare track waveforms.

Another utility of the Q-series plug-ins is that you can flip the polarity. In a stereo instance, you can change the polarity of the left and right channels independently. I put Q1 on the bottom snare mic and clicked on the þ button to change it to  to invert the polarity. With this simple change, the snare sounded more natural and fuller.

Loops I thought that the EQs I’d used on the loop tracks were still working well to create a nice balance, so I left them in place. However, I did think that Loop3 could be doing more for the mix. That track comes in at the halfway point in the bridge and adds

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rhythmic interest to help build up to the next section. The EQ on Loop3 did a good job of de-emphasizing the kick, but I couldn’t just turn up the track to bring out the rhythmic elements, because the (quite prominent) snare would have overloaded the mix. To deal with this, I added a limiter to Loop3 (see Figure 3.14) and set the threshold to 15 dB, which caused it to act only on the snare. Since the limiter threshold simultaneously provides makeup gain, I needed to reduce the output ceiling to get the proper level in the mix. With the threshold set to 15 dB and the ceiling set to 9.2 dB, the end result was that the rhythmic elements got a 5.8-dB boost, and the limiter kept the snare from overpowering the mix.

Figure 3.14 L1 limiter on the Loop3 track.

It’s important to note that I put the limiter after the equalizer. Had I placed the limiter before the EQ, the limiter would have to be providing quite a bit more limiting, because it would be acting on the bass frequencies as well. Letting the EQ remove the bass frequencies first allowed the limiter to do its job cleanly.

Bass The RenComp I was using on the bass helped out quite a bit in evening out my lessthan-stellar performance, but the track still wasn’t anchoring the mix the way I wanted. I followed RenComp with an instance of MaxxBass (see Figure 3.15), which did a surprisingly good job of giving it the body it needed. Using both the upper harmonic generation and some gentle compression on those harmonics provided the heft that I

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was looking for—that is, after dialing it back until I had the right balance between the original bass and the MaxxBass harmonics. You do need to be careful with MaxxBass; it’s one of those processors where a little bit goes a long way.

Figure 3.15 MaxxBass on the bass track.

My instincts were to put MaxxBass before RenComp to get a more even sound, but I found that I got better results with RenComp first. I’m mentioning this to encourage you to always be open to trying out new tools and techniques. You never know when you’ll find out something new.

Guitars The most substantial change I made to the guitars consisted of setting up a reverb on an auxiliary track. I used a mono-to-stereo instance of IR-L (see Figure 3.16) and loaded the factory Plate reverb preset. Whenever you use an effect on an auxiliary track, you need to make sure that it’s set to 100% wet output; otherwise, you will essentially be turning up the volume of your track as you send more of it to the aux. On the aux track, I used the mono-to-stereo version of IR-L because actual plate reverbs are traditionally mono to stereo, and also because I was planning to send to it from mono tracks. The only time I create reverb aux tracks with stereo inputs is when I’m going to be sending a stereo track to them—most likely in a classical or other liveto-2-track post-production situation. The main tracks to which I was adding reverb were the 15-Gtr-chords and 16-Gtrpicked tracks (see Figure 3.17). When listening to the default Plate on the guitars, I felt as if the reverb tail was a little too long, so I scaled the reverb time using a ratio of 0.80, which resulted in an RT60 time of 1.7 seconds. Although this was a fairly subtle effect, if you mute the aux track while the chorus of the song is playing, you will hear the guitars lose some impact. I’m a big fan of the concept of “less is more” as

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Figure 3.16 IR-L Plate reverb set up on an auxiliary track.

Figure 3.17 Reverb sends on the guitar tracks.

applied to mixing—several subtle changes can add up to a substantial overall change. There are other times, however, when “more is more” is the appropriate route to take. In addition to the two main chorus guitars, I’m using a bit of reverb on 17-Gtr-verse and just a touch on the 19-Acoustic track. I like the fact that the verse is much dryer than the chorus, so I added reverb to them very sparingly.

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The mix in the previous chapter used RenVox on the guitar solo track to gate out the ground-loop noise in between phrases. Because RenVox is not part of the Silver bundle, the gating duties had to be handed over to C1 Gate, which did a fine job of replacing it (see Figure 3.18).

Figure 3.18 C1 Gate being used to remove noise between phrases.

Following the gate I did a little midrange boost with Q4 and then had some fun with Enigma and created a slowly sweeping phaser sound (see Figure 3.19). In the scope of what Enigma is capable of, this is pretty tame, but it does add a great texture to the

Figure 3.19 Enigma used as a phaser on the guitar solo.

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sound of the guitar solo. Even though it’s a mono-to-stereo instance of Enigma, it’s still generating output that’s nearly mono. There are times when big stereo effects can cause a track to lose its focus and impact, and I wanted the guitar solo to still drive hard up the middle of the mix.

Keyboards Although Doubler did a decent job of creating a chorus effect for the electric piano, it still wasn’t quite what I was looking for, and the overall tone wasn’t quite there. I started with Q4 to roll off the low end and notch out 250 Hz so it would sit in the track a little better (see Figure 3.20).

Figure 3.20 Q4 rolling off the low end on the electric piano.

Now that I had MondoMod at my disposal, I decided to remove the Doubler and put more of a vintage electric piano spin on the track (see Figure 3.21). Many of the old Fender Rhodes and Wurlitzer electric-piano models had tremolo effects built into them that would either just modulate the volume of the signal or pan it back and forth between two speakers. MondoMod had no trouble getting that sound. I used a fairly fast 2.42-Hz modulation rate, which is pretty close to the timing of a quarter note at the tempo of the song. Because this song was not recorded to a click, its impossible to perfectly synchronize modulation rates or delay times, but that’s not necessarily a bad thing. Even when I’m working on a tempo-locked tune, I’ll often use free time modulation and delays that are close to the tempo but not right on. Having delays that are slightly faster or slower than the beat can add some interest to the mix.

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Figure 3.21 MondoMod giving the electric piano some vintage vibe.

The AM Depth and the Rotation are the primary ModoMod features that I’m using to achieve my desired effect. The Rotation is set to a range of 45, which causes the signal to pan back and forth across half of the stereo field. I didn’t use a Rotation setting of 90 to go full left and right, because I didn’t want the effect to be distracting. I also threw in a touch of frequency modulation for a little Doppler effect, although it’s so slight that I suspect nobody will ever notice—but then, much of what we obsess over in mixing is never appreciated by the listening public. Such is our lot in life. The Synth Chorus track was sounding pretty good with SuperTap giving it some width, but it still seemed to take up too much space in the track, particularly in the center. To give the guitars and vocals more sonic space in the middle, I followed SuperTap with an S1 Stereo Shuffler plug-in (see Figure 3.22). With the Width set to 1.73—very close to its maximum value of 2.00—S1 was working overtime to push the track out to the

Figure 3.22 S1 Shuffler really spreading out the chorus synth.

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edges. With that much widening going on, mono compatibility is definitely compromised. If you solo the track and sum to mono, you can hear a significant loss of volume. Because this was just an “icing” track, I wasn’t too concerned about it being heard in mono playback. This is definitely not something that I would do to a major element of a mix, but in this case it did a great job of opening up the mix. I also used the BassTrim control to roll off a couple of decibels of low end.

Vocals As I mentioned earlier in the chapter, I put an instance of PAZ Analyzer on my master fader, and I leave the plug-in open so I have all of that information available at a glance. Figure 3.23 shows the frequency spectrum of the lead vocal and the substantial audio happening down below 60 Hz. In a vocal track, that low-frequency audio will do nothing but muddy the mix.

Figure 3.23 PAZ Analyzer showing the soloed lead vocal.

On both lead vocal tracks, I start with an instance of Q4 to cut out that low-end mud. Because the slope on the Q-series plug-ins is fixed, I often stack two high-pass filters at the same frequency to achieve a steeper slope. For both vocal tracks, I used two highpass filters set to 102 Hz. That gave me excellent low-frequency filtering without adversely affecting the low mids of the vocals. On the verse vocal track, I started with Q4 using the pair of high-pass filters, but I also added some gain at 5 kHz to brighten it up (see Figure 3.24). I liked the brighter sound, but the EQ caused the vocal to be a little too sibilant. That’s where a DeEsser comes in handy. You do need to be careful with a DeEsser, because it’s easy to overdo it and cause the vocalist to sound as if he has a lisp. This is another case where you don’t want the processing to draw attention to itself. Because Renaissance Vox is not part of the Silver bundle, I had to replace the instance I had used on this track. RenComp was the natural choice because the compression code in RenVox is based on RenComp. Putting the compressor after the EQ and DeEsser allows it to do a better job because it’s not getting hit with the low-frequency audio or the sibilant peaks.

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Figure 3.24 EQ on the verse lead vocal track.

The chorus lead vocal track also got the Q4 high-pass filtering because it, too, contained the low-end muck that needed cleaning up. I followed that with the RenComp that I used in previous chapter’s mix. For this mix I removed the Doubler and put an instance of TrueVerb in its place to see how some reverb sounded on the track (see Figure 3.25). Because I used it inline rather than as a send, I had to balance the wet and dry signals within the plug-in.

Figure 3.25 TrueVerb on the chorus lead vocal track.

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I was still pretty happy with the background vocal balance, so I’ve left them alone for this mix.

What’s Coming Up Next The tools in the Silver bundle gave us a new view of our mix with the PAZ Analyzer and some nice effects with which to embellish it. The Gold bundle will provide several alternatives to plug-ins we’ve used so far, and we’ll get our first peek at Waves’ analog modeling plug-ins.

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I

n moving from the Silver bundle to the Gold bundle, we nearly double our list of available plug-ins. Most of the new plug-ins expand our existing choices of EQ, dynamics, and effects plug-ins, while the others take us into uncharted territory. In one instance it’s so uncharted you’re unlikely to ever go there in a mix.

Equalization The Gold bundle gives us the V-EQ3 and V-EQ4 EQ plug-ins.

V-EQ3/V-EQ4 Even though you won’t find the name “Neve” anywhere in the V-EQ documentation, the references to 1066, 1073, and 1081 make it obvious that these plug-ins are modeling the venerable Neve equalizers. The reason why you don’t see mention of Neve in the documentation is because it’s still an active trademark, and the use of that name requires a licensing agreement to be reached with its owner. In this case there is already another company making Neve-branded plug-ins, which makes it unlikely that Waves would be able to strike a licensing deal as they have with API and SSL. In any case, it’s all about the sound, not the name. The V-EQ3 (see Figure 4.1) is modeled after the Neve 1073 equalizer. As its name suggests, it’s a 3-band EQ. There is a fourth band, but it’s just a high-pass filter, so I guess it doesn’t really count. Each of the main bands uses concentric controls like the original hardware, where the outer knob sets the frequency and the inner one controls the amount of boost or cut. You can get +/18 dB of filter gain change. The frequency control is not sweepable and can only select specific frequencies, which is right in line with the original Neve hardware. The interface is a little tricky to navigate when you are attempting to click and drag on the outer filter frequency knob, but Waves has thoughtfully allowed you to click on the frequency number markings to set the frequency. If you choose Off for the frequency, then that band will be bypassed. Along the bottom you will find the EQ button, which will bypass the filters. However, even with those bypassed, the plug-in will still impart the modeled harmonic distortion characteristics on the signal. You would have to bypass the plug-in using the host DAW bypass control to completely defeat the processing. The Phase button does the expected

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Figure 4.1 V-EQ3.

and inverts the polarity of the signal. The icing on the cake—or maybe more appropriately, the grease on the driveway—is the Analog button, which simply generates noise that’s added to the signal and does not affect the analog modeling or behavior. Since the Analog button just adds noise, always leave it turned it off. The V-EQ4 (see Figure 4.2) is Waves’ take on the Neve 1081 equalizer. It has four filters with frequency and boost/cut controls, as well as high-pass and low-pass filters. Starting at the left in Figure 4.2, you have the high-pass and low-pass filters, which share the same concentric-style knobs as the others, but in this case, each knob is a frequency selector. The next knob is the low-frequency filter, which can operate as either a low-shelf filter or a bell-curve filter, depending on the state of the button below the knob. The next two are the low mid filter and the high mid filter. These are both bell-curve filters with fairly wide bandwidths, although each can narrow its bandwidth by enabling the Hi-Q buttons. The high-frequency filter can operate as a high-shelf or a bell filter, depending on the button below the knob.

Figure 4.2 V-EQ4.

V-EQ4 also includes the same EQ bypass, polarity invert, and Analog noise buttons as V-EQ3.

Dynamics The Gold bundle broadens our palette of dynamics plug-in choices significantly with our first taste of vintage compression, the introduction of multiband processing, and a unique dynamic shaping tool.

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V-Comp As you may have guessed from the name, the V-Comp plug-in is Waves’ take on a Neve-style compressor; in this case it’s the Neve 2254 bus compressor. The V-Comp plug-in (see Figure 4.3) contains both a limiter and a compressor that can be independently switched into or out of the signal path. The limiter has the traditional Threshold, Attack, and Release controls, but the compressor only has Ratio and Release. The threshold of the compressor is fixed, so you control the amount of compression by changing the level of the signal that is being fed to it. This is right in line with the way the original 2254 works. It was designed as a master bus compressor, and you would control the amount of compression by adjusting the master fader. Waves thoughtfully included an input level control so that you do have some ability to control the amount of compression within the plug-in. I suggest that you turn off the limiter, set the meter to GR to see how much gain reduction is occurring, and work with it a bit to get a good feel for how the compressor sounds and reacts.

Figure 4.3 V-Comp.

A couple of other nice features in V-Comp are the De-Esser and the Analog control. The De-Esser will switch in a fixed sideband EQ that causes the compressor to provide more

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aggressive compression of sibilants. You don’t have any control over the frequency or bandwidth, so it’s a case where it’s either going to work or it won’t. Because the 2254 is a bus compressor, the De-Esser was designed to be transparent enough to take the edge off an overly bright mix; don’t expect that it will fix a highly sibilant vocal. The Analog control doesn’t influence the sound or action of the compressor at all; it just serves to add analog noise to the output, so I generally leave this set to Off.

Hybrid Compressor The Hybrid Compressor, shown in Figure 4.4, was developed after Waves had released their SSL 4000 and API bundles. In developing plug-ins that modeled vintage hardware, Waves engineers learned a thing or two about creating tubes and transformers in software. They took this knowledge and produced a compressor that is, as the name might lead you to believe, a hybrid of vintage sound and modern digital control.

Figure 4.4 Hybrid Compressor.

This compressor features the familiar Threshold, Ratio, Attack, Release, and Output controls that are pretty standard on compressors, but it also adds a few new knobs to twist. The Punch control lets you control how much of the initial transient is allowed to pass through, regardless of the Attack setting. The release time can be set so that it’s tied to the tempo of the song, which can help the compression to work better with the track when used on percussive instruments. The Analog control has four modes that

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are intended to produce qualities associated with analog compressors. Each of the modes has its own distinct effect on the sound, and these are difficult to describe— you’ll just have to audition each one to decide which is appropriate for the audio you’re compressing. Keep in mind that each analog mode also introduces a different amount and type of noise. My favorite feature in H-Comp is the Mix control. It works in the same fashion that the wet/dry control works on an effects plug-in, giving you control over the balance between the original and processed signals. A great compression trick is to mix both the original uncompressed and the compressed signal together in what’s called parallel compression. Using this technique you can go really overboard with the compressor, creating all sorts of compression artifacts that you can then blend in to taste with the Mix control. This gives your track the aggressive over-compressed attitude without completely sacrificing the natural sound.

C4 Multiband Processor Originally, the C4 Multiband Processor, shown in Figure 4.5, was released only for the Pro Tools TDM system. When I asked them, Waves informed me that it would never be available as a native plug-in because computers just didn’t have enough CPU power to run something as demanding as the C4. Things have changed a little since 1999.

Figure 4.5 C4 Multiband Parametric Processor.

The major difference between C4 and the other dynamics processors that I’ve discussed so far is that C4 is really four processors in one. The audio signal is split into four

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different frequency bands, and each band is individually processed. To minimize the sonic coloration that is caused by the phase shift introduced by the filters that split the bands, C4 uses phase-compensated crossover filters. The Range control that is available on each band in C4 is a novel feature that allows you to specify the dynamic range that the processor is allowed to use for each band. A negative range will produce compression, and a positive range will produce expansion. In Figure 4.5, you can see that C4 is allowed to change the low band by nearly 12 dB, while the next band up can only be changed over a 7.4-dB dynamic range. This is a key feature that gives you a great deal of control over exactly how C4 processes your signal. You would think that a plug-in with more than 30 parameters would be difficult to use, but the informative display and some grouped controls make C4 reasonably easy and intuitive. At a glance, the graph shows you vertical gray lines where the band crossover points are, with a purple-shaded area showing the range that each compressor can use. An animated orange line shows you the gain change that’s currently occurring across the spectrum. Because each band has its own makeup gain control, you don’t often need to use an EQ in conjunction with C4—you can just boost or cut bands to make tonal shifts. Waves thoughtfully provided a handy way to change parameters in all four bands at the same time. In the Master section in the lower right, there are master controls for Threshold, Gain, Range, Attack, and Release. If you click on the box next to the master control name and hold down the mouse button, as you move the mouse up and down you will adjust all of the bands simultaneously. This is a great timesaver. After spending some time with C4, you’ll find that multiband processing provides powerful creative and corrective control over signal dynamics and tonal balance.

MV2 Dynamics Processor The MV2 plug-in (see Figure 4.6) does exactly two things: low-level compression and high-level compression. High-level compression is the most common type of compression; it starts acting on the signal after the signal exceeds the threshold. The single slider in MV2 controls both the threshold and the makeup gain, which makes it dead easy to set up. The meter to the left of the slider displays the amount of compression as a red bar. The low-level compression works a bit differently. It does what’s called upward compression, where the signal below the threshold is compressed and then boosted so that the quieter portions of the signal get compressed into a narrower dynamic range. All the audio above the threshold is left unchanged. The meter next to the right of the slider displays the amount of compression as a yellow bar. You do have to be careful with this control when working with noisy tracks. Too much low-level compression will really emphasize the noise.

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Figure 4.6 MV2 Dynamics Processor.

It’s possible to accomplish both of these tasks using instances of a general-purpose plug-in, such as C1, but it’s certainly easier to quickly set up these two types of compression using MV2.

Effects Our selection of effects has been broadened in the Gold bundle with additional entries in the flanging, delay, pitch-shifting, and reverb categories. Although these don’t break radically new ground, they do offer different flavors to spice up our mixes.

MetaFlanger Unlike Enigma, MetaFlanger, shown in Figure 4.7, is a much more straightforward effect that makes it easy to get basic phasing, flanging, and chorus effects. The majority of the controls should be very familiar to you after using Enigma. It has Mix, Feedback,

Figure 4.7 MetaFlanger.

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EQ, modulation Rate and Depth, oscillator Waveform, and Stereo width controls that all do what you’d expect. The Delay control changes the effect from phasing at low values, to flanging at moderate values, to chorusing at higher values. The term flanging originates from the days of analog tape. The effect was accomplished by recording the same signal on two tape machines, and then during playback you would alternately slow down one and then the other by pressing a finger against the flange of the tape reel. As the delay between the two signals approaches zero, you get the extreme “through zero” comb filtering, which is the hallmark of flanging. Normally, the Delay setting in MetaFlanger only applies to the wet (effect) output of MetaFlanger, but when the Tape button is lit, the dry signal is also delayed by the same amount. Because the delay time modulation is only applied to the wet signal, it will cause the effected signal to sweep between occurring earlier and occurring later than the dry signal. Using Tape mode and turning on the Mix Polarity button (denoted by a Ø) is how you create those big, sweeping flanger sounds in which the signal all but disappears at the midpoint of the sweep. This is due to the extreme phase cancellation that occurs as the effected signal approaches the same time as the dry signal.

Hybrid Delay The Hybrid Delay, shown in Figure 4.8, was introduced at the same time as the Hybrid Compressor and shares similar design characteristics in that it emulates various types of vintage delays without modeling a specific device.

Figure 4.8 Hybrid Delay.

As with SuperTap, you can specify the delay time in beats per minute or in milliseconds, or you can lock the delay times to the host DAW’s tempo. H-Delay also has a Tap button that you can click to set the delay time manually. The time in between button taps determines the delay time. It often takes a number of taps to get it to accurately set the delay time you want, but I find that’s because I don’t necessarily click the mouse button in perfect time. Tap tempo can be a great timesaver for projects that are not tempo locked.

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Most of the controls will be familiar to you after using SuperTap and other Waves effects, but H-Delay does have some unique features. Like the Hybrid Compressor, it has an Analog control that serves up variations in the sound of the plug-in. You will just have to play with this setting to see what sounds good for the mix. Many early digital delays used low sample rates, which limited the upper frequencies and also introduced other side effects. The LoFi button will cause the delayed signal to have these types of artifacts. Although these artifacts are undesirable if you want clear, clean echoes, they can create a cool vibe that may work perfectly in a track. Hybrid Delay lets you invert the polarity of the left and right wet signals independently. For short modulating delays, inverting the polarity can create flanging effects much like MetaFlanger. Between the polarity buttons, there is one labeled Ping Pong that, if engaged, will alternate the taps between the left and right channels. I don’t think I need to explain why this type of delay has that name. Additionally, H-Delay provides the ability to control most of its parameters via MIDI. This is a great feature for DAWs that don’t already offer this type of control for all plug-in parameters. When you instantiate the plug-in, it will create a virtual MIDI device called something such as H-Delay (m->s): 23 Lead-vox-verse : Insert D. Route the output of your MIDI controller to this virtual device, and you can then assign MIDI continuous controllers to various parameters. You do this by right-clicking on a parameter (see Figure 4.9), selecting the MIDI Learn option, and then moving the MIDI controller (be it a knob, a slider, or a button) so you can control aspects of the delay in real time. This is most often used in a live performance situation, but because you can record the MIDI data to a track, it becomes possible to “perform” delay setting changes in real time in the studio.

Figure 4.9 Enabling MIDI Learn for the Delay parameter.

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Unlike most similar plug-ins, H-Delay also models the behavior of its hardware counterparts in that when you change the delay time while it’s already processing a signal, the audio in delay buffers gets sped up or slowed down depending on how you’re changing the delay time setting. For some real fun, set up a long delay with lots of feedback and start messing with the delay time.

UltraPitch At first glance, UltraPitch, shown in Figure 4.10, might look a lot like Doubler; but while it can produce similar effects, UltraPitch gets much deeper into pitch-shifting than Doubler does. UltraPitch comes in both three-voice and six-voice versions.

Figure 4.10 UltraPitch.

With UltraPitch you not only have control over the pitch, you also have control over the formant of the signal. The classic tape-speed “chipmunk” effect is an excellent example of pitch-shifting that has no formant correction. The fundamental frequency along with all of its harmonics are simply shifted up or down. Formant correction will modify the harmonics such that you can pitch-shift a signal, and it will sound more like that voice or instrument performing that different register. Speaking of voices and instruments, UltraPitch provides some controls that you can set to help it do its job better. The Mode control is found below the Pitch Detector display

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and allows you to let UltraPitch know something about the behavior of the signal you’re feeding to it. Different instruments have different formant characteristics, and the Link control allows you to choose one from a list. By this time you have probably noticed that Waves does a nice job of creating userinterface conventions and reusing them across many plug-ins. The Mix display looks and works just like the Pan Graph in SuperTap, where it indicates the left/right position and the relative volume of each voice. The Voice Character display shows the relative pitch on the y-axis and the formant shifting on the x-axis. The Pitch Detector display shows you the detected pitch of the input signal in real time. UltraPitch is only capable of creating what are called parallel harmonies, which will rarely generate musically pleasing results as the song moves from one chord to the next. Because of this, I find that I use UltraPitch as an interesting mangling processor much more frequently than I ever use it to create harmonies.

Renaissance Reverb If you’ve spent any time working with TrueVerb, you’ll feel right at home with the Renaissance Reverb, shown in Figure 4.11. In fact, Waves used the early reflection portion of TrueVerb as a starting point for RenVerb, but they made significant improvements to the sound of the reverb tail. Waves also made the interface much friendlier and easier to work with.

Figure 4.11 Renaissance Reverb.

The reverb damping and EQ in TrueVerb were intertwined and somewhat difficult to visualize within the plug-in. With RenVerb each function has its own controls and separate display. Those improved displays and the liberal use of sliders rather than value boxes contribute to making RenVerb easier to set up and understand.

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The majority of the controls act the same way as they do in TrueVerb, so you can refer to Chapter 3 for a refresher. One new and very important control is Diffusion, which determines the composition of the signal that is fed to the reverb tail processor. At a value of 0.0, it’s just the direct signal that’s sent; at 100.0 it’s just the early reflections; and everything in between will be a blend of the two. Higher values tend to make the original signal less distinct in the reverb tail because the early reflections will blur the sound. For realism you can’t beat an impulse response reverb like Waves IR-L, but sometimes realism is not what you need. You can easily shape an algorithmic reverb like RenVerb to make its sound fit the mix. This can often create the perfect space around a track that sounds good but doesn’t necessarily sound real.

Other Plug-Ins Among the new plug-ins in this bundle that don’t fit tidily into my categories are Waves Tune LT, Doppler, and Audio Track. Considering that it would be difficult to find three plug-ins that have so little to do with each other, I think that the “Other PlugIns” section is an appropriate place for them.

Waves Tune LT Like super powers, vocal tuning can be used either for good or for evil. It’s up to you to choose. Waves Tune LT lacks some of the features that the full Waves Tune plug-in has (see Chapter 8), but it’s still quite a capable retuning tool. The Mercury installer does not include the LT version because it already includes the full version; so if you’re trying all of these plug-ins under the seven-day demo, you can either install the individual Tune LT plug-in or just use the full version and pretend the additional features aren’t there. It’s certainly possible to use a tool like Waves Tune to make someone who has terrible pitch sound as if he’s not half bad. That also means it’s impossible to make the person sound more than half good. Even if it means encouraging the singer do more takes than he would like to, you should really try to get the best performance you can out of the person and comp the tracks before resorting to pitch correction. On the other hand, there are times when you’re working with an excellent singer who has delivered a vocal take that’s got a great attitude, but maybe there’s a note or two that’s far enough off key to ruin it. That’s where I find a tool like this is golden. For Tune to do its job, it first must analyze the pitch of the audio it’s going to process. To do this, you must insert the plug-in on a track and play through all the parts of the song where audio is playing on that track. When Tune comes across audio it hasn’t scanned yet, the Waveform Overview panel at the top of the plug-in window will turn orange and will display the text “Scanning,” as shown in Figure 4.12. As it’s scanning,

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Figure 4.12 Waves Tune scanning the audio file.

Tune will be creating a retuning curve based on its default correction parameters for Speed, Note Transition, and Ratio. Speed will determine how quickly Tune will correct the signal, and at low values it will remove vibrato. Note Transition determines how quickly Tune will act when the note changes. Ratio is the overall control for how much correction is applied. Setting Ratio to 50% would simply move notes halfway to the destination pitch rather than all the way. Another important setting is Segmentation controls, which will limit the potential target notes to those in the selected scale. If you’re working with material that’s not too far out of tune, you can leave the segmentation key set to Chromatic, and Tune will find the closest note to correct to. If you have audio with pitch that’s all over the place, it will be beneficial to select the key and scale that the song is using so Tune will only try to correct to notes that exist in the scale. If the track is that far off, you likely will have to do some hand editing to fix it up. When Tune has analyzed the audio, it will display the waveform at the top in the Waveform Overview, and in the Pitch Edit Area below that, it will display the detection graph (shown as an orange line), the correction graph (shown as a green line), and the note segments (shown as orange boxes). If everything sounds good once Tune has analyzed the pitch and applied its default settings, then you’re finished. In practice, I’ve found that it’s rarely that simple. There are two approaches to customizing the pitch correction. The first entails selecting a portion (or all) of the audio in the Pitch Edit Area and manipulating the tuning parameters. As you turn the knobs, you will see the correction graph change so you can see the effects of the different settings. The other approach is to manually adjust pitch segments by working in the Pitch Edit Area and dragging the note segment boxes up and down until you get the curve you want. In Figure 4.13 you can see that I’ve dragged some boxes down to cause a short bit of audio to come down in pitch. To make the tedious job of adjusting and then listening to the corrections a little easier to do, Waves has licensed ReWire from Propellerhead (www.propellerheads.se). ReWire is a technology that allows audio and MIDI data to be communicated between different applications. ReWire also allows one application to control the transport of another, and this is a feature that the Tune plug-in uses. Clicking on the Timeline Ruler (found above the Pitch Edit Area) will cause your DAW to move its playhead to

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Figure 4.13 The Pitch Edit Area showing the detection and correction graphs.

that location. There are also red and green vertical lines (shown in black and white in Figure 4.14) that will set your DAW’s memory start and memory stop locations. With ReWire transport control, you don’t have to leave the Tune plug-in to change those settings. It makes life much easier to be able to set up these loop points when you have to comb through a track to fix multiple pitch issues.

Figure 4.14 Waves Tune LT.

It’s worth noting that whenever Tune does change the pitch of the audio, it can also change the formants (as discussed in the “UltraPitch” section earlier in this chapter) to provide as natural a sound as possible. The Formant button is shown in the upper-left corner of Figure 4.14.

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One other note: If you want to produce the pitch-correction effect overused by multitudes of artists, you’ll have to move up to the full version of Waves Tune to have control over the segmentation tolerance. You can read about that in Chapter 8.

Doppler And now for something completely different. Doppler, shown in Figure 4.15, is about as specialized as a plug-in can get—its only job is to simulate the Doppler Effect. You can set up a “flight path” for a sound, and Doppler will modulate the volume and pitch to make it sound as if it’s flying past you. Although this is pretty cool, I’ve never had the occasion to use it in a mix. Who knows—maybe one of these days a band will ask me to make something sound as if it’s flying by, and Doppler will come to the rescue.

Figure 4.15 Doppler.

AudioTrack AudioTrack, shown in Figure 4.16, is another one of Waves’ early efforts. I put it into the “other” category not because it’s strange, but because it’s both an EQ and a compressor. These types of plug-ins are commonly called channel strip plug-ins because they offer the processing commonly found on every channel of a high-end mixing board. AudioTrack is the first channel strip plug-in that I recall seeing. You can tell it’s old because it still has a processing preset for preparing audio for Internet encoding and delivery by RealPlayer and Shockwave, which are all but forgotten these days. If you took the Q4 equalizer and followed that with a simplified version of the C1 comp-gate, you’d have AudioTrack. It’s pretty common to place the compressor

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Figure 4.16 AudioTrack.

after the EQ, so not being able to change the order isn’t much of a limitation. If you need a little simple EQ and dynamics processing, AudioTrack is a handy tool. I personally like to mix and match my EQs and compressors to have the flexibility to create the tone I want, so I don’t often use channel strip–style plug-ins.

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The Mix For this mix I put most of my effort into working on the drums. I also spent some time on the acoustic guitar and vocals.

Table 4.1 Gold Bundle Mix Track

Slot A

01-Kick-in

RenEQ2

03-Snare-top

V-Comp

04-Snare-btm

Q1

05-Tom-hi

Slot B

Slot C

Slot D

Send 1

C1 comp-gate

06-Tom-mid

RenEQ2

C1 comp-gate

07-Tom-low

RenEQ2

C1 comp-gate

08-Hi-hat

RenEQ4

10-Room

RenEQ2

Drum sub*

H-Comp

11-Loop1

RenEQ4

12-Loop2

V-EQ4

13-Loop3

RenEQ4

L1

14-Bass

RenComp

MaxxBass

15-Gtr-chords

RenEQ4

21.5 dB

16-Gtr-picked

RenEQ4

18.0 dB

17-Gtr-verse

MV2

25.5 dB

18-Gtr-solo

C1 gate

Q4

19-Acoustic

C4

MetaFlanger (m->s)

21-Epiano

Q4

MondoMod (m->s)

Enigma (m->s) 29.0 dB

(continued )

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Table 4.1 Gold Bundle Mix (Continued ) Track

Slot A

Slot B

22-Synth-chorus

SuperTap2 S1 Shuffler (m->s)

Slot C

Slot D

23-Lead-vox-verse Q4

DeEsser

RenComp H-Delay (m->s)

24-Lead-voxchorus

Tune LT

Q4

V-Comp

Effects 1

IR-L (m->s)

Send 1

RenVerb (m->s)

*Drum sub is fed by Tracks 3 through 10. The bolded plug-in names in the table above indicate plug-in or settings changes between the previous chapter and this one. The (m->s) at the end of the plug-in names denotes that a mono-to-stereo instance of the plug-in is to be used.

Drums I decided that it was time to work over the drums, and I was just itching to use the juicy Neve-style V-Comp compressor. My first stop was the snare, and the logical choice was the top mic track (see Figure 4.17). The bottom snare mic is all about the rattle of the snares, and putting a compressor on that track would’ve caused all the grace notes and

Figure 4.17 V-Comp on the snare top track.

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sympathetic snare rattles to be far too prominent. Putting V-Comp on the snare top track fattened it up nicely and brought out the resonance of the drum shell, which in this case made it sound livelier and more in your face. When you close-mike the top of a snare, you often pick up a good deal of ringing. If this had been a snare with that problem, you wouldn’t have wanted to use a compressor on it because the ringing would’ve gotten out of hand. I didn’t want to blunt the attack of the snare, so I turned off the limiter entirely. Because V-Comp doesn’t have a threshold setting, I backed off on the Input level to set the desired amount of compression. I then used the Output level to get the level back up. I turned on the DeEsser, thinking that it might mitigate the hi-hat bleed that was now louder due to the compression, but I didn’t really hear it helping much in that regard. While the snare compression gave me a livelier and thicker snare sound, it also caused the hi-hat bleed to be boosted enough to make the hi-hat overbearing. To compensate for this, I modified the EQ on the hi-hat track (see Figure 4.18). Instead of a high-shelf boost at 8 kHz, I changed that to low-pass filter at 9.9 kHz. The snare track was providing plenty of hi-hat above that frequency now.

Figure 4.18 The modified EQ on the hi-hat.

I still thought that the drums could use a meatier sound overall; it was about time to try some drum bus compression. Now that I had the Hybrid Compressor to work with, I knew I had a tool that could give me the attitude I was looking for without sacrificing the transients. Instead of adding H-Comp to each track (which would have been cumbersome and wouldn’t have sounded very good anyway), I set up an auxiliary track as a drum subgroup. With a subgroup I could use one instance of H-Comp that would

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process the entire kit. This type of bus compression can act like sonic “glue” that creates a more cohesive sound. Subgroups A subgroup is a very useful mixing tool that, like an effects send, allows you to use a single plug-in on multiple tracks. To set up a subgroup, you create an aux track and set its input to an unused stereo bus. You then assign the outputs of all the channels that need to be part of the subgroup to that stereo bus. At this point those tracks are no longer feeding the mix bus directly, but they are all submixed and processed in the subgroup before being passed on to the mix bus.

Live audio engineers routinely create drum, vocal, and instrument subgroups so that they can quickly change the overall mix balance without having to grab multiple handfuls of faders.

The trick to using a compressor in this fashion is getting it to provide compression without causing excess pumping or loss of transients. H-Comp is an excellent choice for this task because its Mix control allows for parallel compression. In Figure 4.19 you can see the settings I’m using for H-Comp. The Ratio is cranked all the way up, and during the choruses I’m getting 12 dB of compression with the peak indicator

Figure 4.19 The Hybrid Compressor on the drum subgroup.

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lighting up. On its own that likely would have been far too much compression for a song like this, but that’s where the Mix control comes in. Setting the mix to 60 gives me 40% uncompressed and 60% compressed signal. That heavy compression mixed in with the unaffected signal makes it much thicker without making it sound flat and lifeless. Another key part of the sound was using the Analog 3 setting, which to my ears was the darkest of the four available settings. Compressing an entire drum kit will often accentuate the cymbals, so the darker setting tamed them a bit while giving the rest of the kit a bigger sound. Initially, I sent all 10 of the drum tracks to the drum subgroup, but I found that the kick was losing too much of the thump. If H-Comp had a sidechain EQ built into it, I may have been able to deal with this by using a high-pass filter on the sidechain; but since it didn’t, I simply chose not to route either of the kick tracks to the drum subgroup. The benefits of subgroup compression also came with some unintended side effects. Because the compression was boosting the low-level portions of the signal, you could now hear the toms ringing in between the kick and snare hits. This caused the drums to become a little masked in the low midrange. To clean this up, I added an instance of C1 comp-gate to each of the tom tracks. Figure 4.20 shows that C1 is providing around 30 dB of attenuation while the tom is not being hit—which is certainly enough to eliminate the ringing. To get the expander to work consistently and not open up when the snare is hit, I set up a sidechain with a bell-curve filter whose center frequency is set to the resonant frequency of the tom. To find that frequency for each drum, I soloed the track and used PAZ to identify it. Because C1 also has a compressor, I went ahead and added a few dB of compression for the peaks. All in all, this really cleaned up the sound of the drums, and I still had nice, punchy toms. Now that I had C1 and then the H-Comp submix processing going, I found that I could back off on the EQ gain on the mid tom.

Loops For this mix I decided to use V-EQ4 to work over the Loop2 track (see Figure 4.21). This is the track that I decided to use as the primary loop, and I wanted to give it a little more grit and substance. I boosted some of the low mids as well as the highs but cut a bit at 1.8 kHz to leave room for what was happening in Loop1. Between the EQ and harmonic distortion of the modeled transformers, Loop1 was carrying the bridge handily.

Bass I was also still pleased with how the bass was standing up to the drums, even with the changes I’d made to them. In fact, cleaning up the ringing in the toms gave the bass a little more space down around 125 Hz.

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Figure 4.20 The C1 comp gate used on the mid tom track.

Figure 4.21 V-EQ4 warming up Loop2.

Guitars Most of the electric guitars were holding their own nicely, but I did decide to try something different on the semi-distorted verse guitar track. Using MV2 (see Figure 4.22), I dialed in some low-level upward compression to bring out the quieter parts of the track. The best way I can describe the effect of this processing is that it sounds as if you’ve moved closer to the guitar amp but you have not increased the apparent volume. You can hear more detail in the picking and a little more texture overall. This complemented the dry, upfront sound of the verses.

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Figure 4.22 MV2 on the verse guitar track.

Although the electric guitars sounded good, I thought that the acoustic guitar was getting kind of lost in the mix. I decided to replace the RenAxx and RenEQ with the C4 plug-in (see Figure 4.23). Not only was I able to replace the two plug-ins with one, I also gained much more control using C4. Using the low-band gain, I pulled down

Figure 4.23 C4 being used on the acoustic guitar track.

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everything below 100 Hz. The low midrange band saw the most compression activity, with a wide range and its attack and release set quite fast so it would get into and out of compression very quickly. The Attack and Release settings for the upper midrange band were much slower and provided more of a leveling type of compression. The high-frequency band had range set fairly narrow, so it was largely left alone with just a couple of dB boost using the Gain control. The overall effect was to make the acoustic guitar cut through more and take up less space in the low end. To lift up the acoustic track a little bit more, I added a mono-to-stereo instance of MetaFlanger (see Figure 4.24). With a slow modulation rate of 0.09 Hz and wet/dry mix of only 14% wet, it created a very subtle effect that largely preserved the natural sound of the acoustic guitar while giving it a little more spaciousness.

Figure 4.24 MetaFlanger on the acoustic guitar track.

Keyboards The keyboards were still working, so I left well enough alone…for now.

Vocals I mentioned in previous chapters how I liked the dry feel of the verses. Now that the drums were filling up more sonic space, I felt that the verse vocals could use a little something to make them pop out of the mix, but I didn’t want them to sound particularly processed. I made a small adjustment to RenComp to provide a little more compression and then followed that with the Hybrid Delay (see Figure 4.25). I used the mono-to-stereo version of it so that I could use its Ping Pong feature. With a short 122 ms delay and the feedback set to zero, the Ping Pong effect creates just two echoes—one on each side. I turned on LoFi and set up a band-pass filter to tuck the echoes back in the mix. Like the MetaFlanger I used on the acoustic guitar, this is an effect that you don’t hear outright, but you definitely notice it’s gone when it’s bypassed. The intonation for the majority of the lead vocal tracks was really good, but on the prechorus the pitch went a little sharp at the beginning of some of the phrases. In Figure 4.26 you can see the three places where I used Tune to bring the vocal into line. When I’m going to be doing some surgical retuning like this, I don’t want Tune to create a correction

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Figure 4.25 Hybrid Delay on the verse lead vocals.

Figure 4.26 The three tuned sections on the chorus lead vocals.

graph that will change the pitch of any of the audio; I just need it to analyze the track. To undo the default tuning that’s applied after Tunes does its initial analysis, I hit the Select All button and then set the Ratio to zero. This will result in a correction graph that is identical to the detection graph. I’m then free to edit pitches by hand. Whenever you are using Waves Tune, you should make it the first plug-in on the track, because processing the vocal prior to pitch analysis can make it more difficult for Tune to identify the pitch. I thought that the chorus lead vocal track could use a little more attitude and bite, so I swapped out RenComp for V-Comp (see Figure 4.27). The settings I used are pretty similar to the ones I used for the V-Comp on the snare track. I’ve got the limiter turned off because it was taking too much of the edge off the vocal. I’m using a pretty low compression ratio of 2:1, but the input is set so that the compressor is acting all the time across the entire dynamic range of the vocal. I opted to use the fastest release time of 400 milliseconds, so the compress got out of the way as quickly as possible. This provided quite a bit of dynamic-range compression, which helped the vocal sit on top of the mix and stay present.

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Figure 4.27 V-Comp giving the chorus lead vocal more attitude.

TrueVerb added a little space around the chorus vocal, but it wasn’t really sitting in the track the way I would have liked. For this mix I replaced TrueVerb with Renaissance Reverb (see Figure 4.28). Instead of using a normal reverb type, such as Hall, Chamber, or Plate, I chose to use the Non-Linear reverb. Because there’s a lot going on in the choruses, I didn’t want a reverb that was going to cloud it up. The Non-Linear

Figure 4.28 The Renaissance Reverb used on the chorus lead vocals.

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algorithm acts a little bit like gated reverb and doesn’t have a tail that lingers. By using a healthy Predelay setting, this reverb is acting a little like a delay but with a more diffuse sound. The vocals get some depth and support without sounding as if they’ve stepped back into a room.

What’s Coming Up Next Moving up to the Platinum bundle, we’ll get another pair of emulations of actual vintage hardware processors, as well as several alternatives for dynamics plug-ins.

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5

The Platinum Bundle

T

he move from the Gold bundle to the Platinum bundle increases the depth and breadth of your choices of equalizer and dynamics plug-ins. Some of the Platinum bundle plug-ins are targeted at mastering, but they can sometimes be useful in mixing. Our long-lost friend from the Musicians 2 bundle, Renaissance Vox, returns to the fold in the Platinum bundle.

Equalization The Platinum bundle hits both ends of the spectrum with a pair of equalizers that are noisy, have coarse controls, and have an uneven frequency response, as well as one that can perform laser-precise equalization with virtually no artifacts.

PuigTec MEQ-5 If you’ve had experience using vintage audio gear, you know that when comparing two units of the same make and model, it’s unlikely that they will sound the same. Aside from circuit revisions and aftermarket mods, the age of the components can have a significant effect on the sound. When a company sets out to model a piece of vintage gear, they need to pick one unit to analyze, so it’s important to pick a good one. Waves worked with renowned producer/engineer Jack Joseph Puig (U2, Black Eyed Peas, Beck, Rolling Stones, Green Day, and others), who provided his favorite Pultec MEQ-5 (see Figure 5.1) and EQP-1A equalizers to Waves for analysis and modeling.

Figure 5.1 PuigTec MEQ-5.

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The PuigTecs are far more than just a pretty interface slapped over the top of Q3. Waves has gone to lengths to model the harmonic distortion, the effect of the transformer, and even the noise floor of the original unit. The harmonic distortion and the transformer both contribute to the unique tone that these EQs produce—they’ve definitely got a character all their own. In Figure 5.2, you can see the PAZ Analyzer display for a 1-kHz sine wave. On the left is just the sine wave, and on the right is the sine wave passing through the MEQ-5 with no boost or cut. You can see the significant harmonics created at 2 kHz and 3 kHz, the somewhat quieter harmonics at 5 kHz and 10 kHz, and the even quieter harmonics at 4 kHz and 7 kHz. When passing a more complex signal through the EQ, you can just imagine what kind of interesting flavor it’s adding to the sound.

Figure 5.2 PuigTec plug-in harmonic distortion.

With the Mains control, you can select AC line noise for 60 Hz (US) or 50 Hz (UK) to have the plug-in generate noise specific to the region of your choice. The 60-Hz noise contains a noticeable 60-Hz hum as well as some broadband noise. The 50-Hz setting seems to just have the broadband noise. I can’t hear the noise influencing the equalizer at all, so I usually opt to simply turn it off. Unlike the equalizers you’ve read about so far, the MEQ-5 filters will not sweep continuously through the spectrum—you can only select from a handful of specific frequencies. This may seem like a serious drawback, but when you think about the fact that thousands of great recordings were made with this limited amount of control, you realize that it’s possible to get great results without pinpoint control. The character that the analog modeling imparts more than makes up for the lack of precise control. The EQ itself is relatively simple. There is a low-mid peak that provides up to 10 dB of boost, a mid dip band that provides up to 10 dB of cut, and a high-mid band that provides 10 dB of boost. I find that this is a great EQ for electric guitars because the frequencies on which it operates cover the guitar range very nicely.

PuigTec EQP-1A The EQP-1A is also a 3-band EQ like the MEQ-5, but this unit provides more controls and a wider range of filter frequencies. When you first see the front panel, it’s not

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Figure 5.3 PuigTec EQP-1A.

obvious which knobs work on which band, so in Figure 5.3 I’ve cut the interface apart to make it clear which controls operate together. The low-frequency shelf filter is interesting in that it has separate boost and attenuation (cut) controls. You might think that they would cancel out if you boost the same amount that you cut, but their curves are not mirror images of each other. Some interesting EQ results are possible by using both of them together to varying degrees. The frequency selector is labeled as CPS, which stands for Cycles Per Second. That’s just a different way of saying hertz (Hz). The high-frequency boost control allows you to set the Q of the filter by dialing in a sharp (high Q) or broad (low Q) bandwidth. The frequency selector here also has an unusual label of KCP, which is a different way of saying kilohertz (kHz). Your options range from 3 kHz up to 16 kHz. The last band is a high-frequency shelf filter that allows you to attenuate starting at 5 kHz, 10 kHz, or 20 kHz. Even though the transformers that were modeled start rolling off the high end at 10 kHz, the Q on this filter is gentle enough that even when set to 20 kHz, it reaches down well into the audio range. Some people think that graphics like those used in the PuigTec plug-ins are just gratuitous eye candy and that the same control can be accomplished using a spartan interface that simply contains numbers and sliders. Although that’s certainly true, I think that the rich interface contributes to the way I interact with the plug-in. For me, the form can certainly influence the function.

Linear Phase EQ The previous two equalizer plug-ins reveled in their un-digitalness, rolling around in the DSP equivalent of mud. At the ritzy end of the of EQ spectrum, the Linear Phase EQ (see Figure 5.4) is as squeaky clean as they come. Most digital filters use an Infinite Impulse Response (IIR) system in the filter code. IIR filters don’t require a great deal of CPU resources to implement, but they do exhibit

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Figure 5.4 Linear Phase broadband EQ.

phase-shift artifacts that are similar to those in analog filters. The greater the boost or cut, the greater the phase shift is. This phase shift is the result of slight delays that are introduced to the portion of the signal that’s being filtered. This phase shift is not necessarily bad and does contribute to the sound of the equalizer. Linear phase filters use Finite Impulse Response (FIR) systems in the filter code. FIR filters require significantly greater CPU resources to do the same job, but they don’t exhibit the phase-shift artifacts that IIR filters do. This allows them to do some serious filtering without adversely affecting the phase relationships between the filtered and unfiltered portions of the signal. Typically, FIR equalizers are used in mastering because they can cleanly change the frequency balance of a mix without introducing additional coloration. However, linear EQs can be useful in mixing when you need to perform some surgical EQ and you don’t want to color the sound. The Linear EQ has two different components, the LinEQ Broadband and the LinEQ Lowband (see Figure 5.5). The Broadband EQ has five bands that range from 258 Hz to 21 kHz and one low-frequency band. The Lowband EQ has three low-frequency bands. If you’ve got a lot of work to do in the low frequencies, then the Lowband will be the right EQ for the job. The Method control is unique to the LinEQ plug-ins. The three settings are Normal, Accurate, and Low Ripple. Unless you’re doing huge boosts or cuts or using very high Q values, your best choice is to use the Normal setting. For extreme EQ you should use either Accurate or Low Ripple. Accurate is the choice to make when you need very

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Figure 5.5 Linear Phase Lowband EQ.

narrow, high boost or cut bell curves or very sharp high-pass or low-pass filters. The Low Ripple cannot produce curves that EQ as sharply, but in the case of high-pass and low-pass filters, the signal outside the filter band will be attenuated by 100 dB or more. With Accurate, the most attenuation you will get outside the band is about 60 dB, which may be just fine for most applications. This may sound very esoteric, but that’s because it is. This type of extreme filtering is rarely needed, so if you do run into it, you’ll have to decide which is the best method for the task at hand. Both of these plug-ins can introduce dither into the output signal. Their default settings dither to 24 bits, so if you’re using a DAW with a 32-bit floating-point or a 48-bit integer mix bus, then you should leave dither set to off.

Renaissance Bass In the Silver bundle (in Chapter 3), you were introduced to MaxxBass. The Renaissance Bass plug-in (see Figure 5.6) is a simplified repackaging of the MaxxBass tool but with some refinements. We all know Waves as a company that produces tools for audio production, but they also license their technology to manufacturers that include Waves’ code in the firmware of various devices that handle audio. As you might imagine, a tool like MaxxBass is useful for dealing with signals that have to travel over band-limited devices, such as cell phones. Tailoring their MaxxBass code for use in these devices has resulted in Waves refining their algorithms, so they decided to bring those refinements back to the audio-production world in the form of Renaissance Bass.

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Figure 5.6 Renaissance Bass.

The Freq slider controls the part of the signal that is passed to the RenBass processor. The In button that’s above the input level meter will control whether the original bass signal will be present in the output. If it’s turned off, then your output will only contain the RenBass processed signal and original signal above the frequency setting. If you’re trying to enhance a low-frequency track, you will want to leave this on. If you’re trying to remove the low end completely, you would leave this off and rely on RenBass to move the low-frequency portion of the signal into a higher range. The intensity will give you more or less processing. The amount will be up to you.

Dynamics Most of the new dynamics plug-ins are tailored to mastering tasks. For mastering plugins, Waves uses algorithms that take a bigger bite out of your available CPU power. Typically in mastering, you’re only using a handful of plug-ins, so it’s not a problem that you can only run a few. As computers have gotten exponentially faster, it has become possible to use these power-hungry plug-ins in mixing. It doesn’t happen often, but it’s good to know they’re available when you need them.

Linear Phase Multiband You might be wondering how linear phase processing applies to a dynamics processor. Multiband dynamics processing splits the signal into separate bands so they each can be processed independently. That band splitting in the Linear Phase Multiband plug-in is accomplished using FIR filters (see Figure 5.7). It makes a great deal of sense to use an FIR filter in this application because you really just want the filter to cleanly separate

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Figure 5.7 Linear Phase Multiband.

the bands. As with other plug-ins that use FIR filters, this precision comes at the cost of higher CPU usage. You can think of LinMB as the more refined sibling of C4 (as seen in Chapter 4). LinMB provides essentially the same controls but has five bands to work with rather than four. LinMB also has a few controls that C4 does not—specifically, Dither, Makeup, and Adaptive. The Dither control is identical to the one found on the Linear Phase Equalizers discussed earlier in this chapter. Makeup can be set to either Manual or Auto. In Auto mode, as you move the threshold setting, there will be an opposite makeup gain added. This attempts to preserve the overall output level. Moving the threshold down will cause more compression to occur so the band gain is increased. I usually use the Manual mode. The Adaptive control is something of an enigmatic beast. The basic concept is that when you have a louder, lower-frequency sound, that sound will mask the sounds in the frequencies above it. For example, a big, booming kick will cause the midrange to sound as if it’s quieter than it really is when that kick hits. The Adaptive processing will modify the threshold settings so that when a loud signal happens in a band, the next band up will have its threshold temporarily raised so that it gets less compression to counter the masking that’s happening. The higher the Adaptive control is set, the more influence the levels will have on the thresholds. It’s kind of a tricky thing to manage, but it can be useful in situations where you’ve got a perceived drop in level when you

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have low-end peaks causing masking. One other note: The Masking control looks like a button because it defaults to the off state. To change it, you have to click and drag up or double-click and enter a value. As with the LinEQ Broadband and LinEQ Lowband, I tend to use LinMB more often in mastering than I do in mixing.

L2 Ultramaximizer As successor to L1, it doesn’t look like the L2 Ultramaximizer (see Figure 5.8) brings anything new to the table. Other than the ARC button, all of the other controls are the same. But under the hood, Waves made some definite improvements in the sound. ARC stands for Auto Release Control, which, when enabled, allows L2 to dynamically alter the release time of the limiter based on the material that’s being limited. With L2, you can do more extreme distortion-free limiting than was possible with L1.

Figure 5.8 L2 Ultramaximizer.

L2 has dither controls very much like those found on L1þ, where you can select the bit depth, dither type, and noise shaping. L1 has bit-depth choices all the way down to 8-bit, but L2 only goes down to 16-bit. For use within a mix, it’s best to leave dither turned off. The only reason to use dither is for final output to 16-bit files. This plug-in would have to be absolutely the last process the audio undergoes before final output to have any benefit. Any additional processing will destroy your carefully dithered signal. Used in mastering, this technological advance moved us even closer to the goal of creating finished products that have no dynamic range. Although I might sound bitter

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about the loudness wars, I have to admit that I did my share of squashing with L2 as soon as it became available. Latency and Delay Compensation The term latency is just a fancy word for delay and is generally used to describe undesirable delay rather than the desirable kind found in delay line devices. In analog circuits processing, latency is virtually nonexistent, but in digital systems even the A/D and D/A converters introduce latency many times greater than that—on the order of several milliseconds. Much greater than that is latency created by the time it takes to perform the DSP processing on a signal. When monitoring live signals through your DAW, this latency can cause all sorts of issues, but fortunately this is largely not an issue when mixing. Most DAWs these days have some sort of automatic delay compensation that determines how much delay the various plug-ins on a track will introduce. They then feed the audio to the track early so that after the signalprocessing delays, the audio will arrive right on time.

Certain plug-ins require a substantial amount of time to do their processing. Plug-ins that use FIR filter systems often generate quite a bit of latency. Compressors that utilize look-ahead detection also require the audio to be delivered early and will have long latency. If you’re using a DAW such as Pro Tools LE, there are limits to the latency compensation. Past a certain point, it will not be able to compensate, and your audio will be late. In these cases, you can either choose a different plug-in or slide the audio track earlier to manually compensate.

L3 Ultramaximizer Even though L2 and L3 are introduced in this book at the same time, in the real world three years elapsed between their introductions. Again, there don’t seem to be many differences between this limiter and its predecessor, but L3 (see Figure 5.9) is very different in how it does its limiting. Internally, it acts like Linear Multiband in that there is a set of FIR filters that split the signal into five different frequency bands. L3 still uses a single peak limiter, but it plays some games with the five bands prior to the limiting that allow it to put an even bigger squeeze on the audio. The Profile choices change the internal settings for the band crossover points, the separation between bands, release time, and something called band priority. In the next chapter, you’ll gain access to these underlying multiband parameters with the L3 Multimaximizer. The L3-LL Ultramaximizer plug-in (see Figure 5.10) is basically the same as the L3 Ultramaximizer with three specific differences. The first one is pretty obvious: The user interface for the plug-in looks totally different. The next relates to the LL in the

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Figure 5.9 L3 Ultramaximizer.

Figure 5.10 L3-LL Ultramaximizer.

name, which stands for Low Latency. The time required for this plug-in to process audio is quite a bit less than the normal L3. If you’re in a situation where high latency is an issue, this could really help. But that reduced latency comes at a price. To get the latency down, they had to forego the FIR filters and use IIR filters to split up the bands. That means there will be a bit of phase shift in the output, which will cause L3-LL to not be as transparent as the normal L3 plug-in.

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MaxxVolume Even though it was covered in the previous chapter, MV2 is the direct descendant of MaxxVolume. In an evolution that parallels the one that MaxxBass and Renaissance Bass followed, the older MaxxVolume features more control than the stripped-down MV2. Like MV2, MaxxVolume’s primary function is to simultaneously provide low-level and highlevel compression. The separate threshold and gain sliders (see Figure 5.11) allow for more fine control over the character of the compression.

Figure 5.11 MaxxVolume.

In addition to those two compressors, MaxxVolume also has a gate and a leveler. The gate functions very much like the gate found in RenVox, where you’ve got a single threshold slider that sets the point where the gating starts. And like RenVox, it’s not really a gate, but a gentler downward expander. The Leveler is a compressor that has very slow attack and release settings so that it will attempt to smooth out the overall levels of the signal. The slider sets the threshold for this leveling. All four of these dynamics processors can be switched in and out using the small lighted button within the slider control itself.

Renaissance DeEsser The Renaissance DeEsser (see Figure 5.12) is further up the evolutionary ladder from the original Waves DeEsser. RenDeEsser works pretty much like the older Waves DeEsser plug-in—you’ve got the basic frequency, threshold, sideband filter type, and wideband/split settings found on de-essers everywhere, but the RenDeEsser adds a

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range setting that goes from 0.0 dB (no attenuation) to 48 dB (a huge amount of attenuation). This range setting will determine how much gain reduction happens once the signal exceeds the threshold. If you set a high threshold and a wide range, you will get infrequent de-essing, but when it happens it will produce a lot of attenuation.

Figure 5.12 Renaissance DeEsser.

When using Split Mode, a DeEsser must divide the signal into multiple bands, so there is the potential for phase shift to rear its ugly head. With RenDeEsser, Waves implemented the same phase-shift compensated filters that they used in C4 (see Chapter 4) to prevent unwanted signal coloration.

Other Plug-Ins The Platinum bundle has one new plug-in that falls into the catchall “Other Plug-Ins” category, and it’s not particularly odd, but it does pack a lot of features into a small package.

Renaissance Channel Like the AudioTrack plug-in that was discussed in the previous chapter, the Renaissance Channel plug-in (see Figure 5.13) contains an equalizer, a compressor, and a gate. RenChannel improves on AudioTrack in both sound quality and signal-routing flexibility. The equalizer in RenChannel is identical to the RenEQ 4-band equalizer, the compressor section allows you to select either the Renaissance Compressor or the Renaissance Vox algorithm, and the gate is essentially the downward expander/gate

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from Renaissance Vox with additional range and release-time controls. RenChannel’s flexible routing scheme allows you to put EQ before or after the compression.

Figure 5.13 Renaissance Channel.

The compressor and gate each have extensive sidechain controls. You have three different sidechain sources to choose from: Int, PreEQ, and Ext. If you have the EQ prior to the compressor, choosing Int will use the already equalized signal as the starting point for the sidechain. If you’ve selected PreEQ or you’ve got EQ following the compressor, then the raw input signal is used for the sidechain. If your DAW supports external sidechains for plug-ins, then Ext will use any signal bussed here as its sidechain. In all cases you can add EQ to the sidechain signal using the pair of dedicated sidechain filters. Waves has really packed a lot of flexibility into this plug-in. Finally, you have output gain, polarity invert, and rotation controls. The Rotation setting is a pan control that’s available on stereo and mono-to-stereo instances of RenChannel.

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The Mix The list of tracks with processing on them just keeps getting longer. This time around it was primarily the guitars and vocals that got attention. I was starting to feel as if the guitars could use a little more edge now that the drums had more attitude. Mixing can get to a point where it’s like the Pillsbury Doughboy—you push your finger in one place, and it squishes out somewhere else. At some point you need to know when to say it’s finished. For this mix, that point will be at the end of Chapter 8.

Table 5.1 Platinum Bundle Mix Track

Slot A

01-Kick-in

RenEQ2

03-Snare-top

V-Comp

04-Snare-btm

Q1

05-Tom-hi

Slot B

Slot C

Slot D

Send 1

C1 comp-gate

06-Tom-mid

RenEQ2

C1 comp-gate

07-Tom-low

RenEQ2

C1 comp-gate

08-Hi-hat

RenEQ4

10-Room

RenEQ2

Drum sub*

H-Comp

11-Loop1

RenEQ4

12-Loop2

V-EQ4

MaxxVolume

13-Loop3

RenEQ4

L1

14-Bass

RenBass

15-Gtr-chords

PuigTec MEQ5

21.5 dB

16-Gtr-picked

PuigTec EQ1PA

18.0 dB

17-Gtr-verse

MV2

25.5 dB

18-Gtr-solo

C1 gate

Q4

19-Acoustic

C4

MetaFlanger (m->s)

21-Epiano

Q4

MondoMod (m->s)

22-Synth-chorus

SuperTap2 (m->s)

S1 Shuffler

23-Lead-vox-verse

Q4

RenDeEsser

RenComp

H-Delay (m->s)

Q4

V-Comp

RenVerb (m->s)

24-Lead-vox-chorus Tune LT 27-Bvox-Mel-M1

Q2

28-Bvox-Mel-M2

Q2

29-Bvox-Har-F1

Q2

30-Bvox-Har-F2

Q2

Enigma (m->s) 29.0 dB

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Bvox sub*

RenDeEsser

33-Oooh-F1

Q2

34-Oooh-F2

Q2

Oooh sub*

RenChannel

Effects 1

IR-L (m->s)

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RenChannel

*Drum sub is fed by Tracks 3 through 10, Bvox sub is fed by Tracks 25 through 32, and Oooh sub is fed by Tracks 33 through 36. The bolded plug-in names in the table above indicate plug-in or settings changes between the previous chapter and this one. The (m->s) at the end of the plug-in names denotes that a mono-to-stereo instance of the plug-in is to be used.

Drums The Hybrid Compressor on the Drum subgroup was still sounding good in the mix even with the additional processing added elsewhere, so I left the drums alone.

Loops For this mix I decided to use MaxxVolume to give the Loop2 track a little more of an edge (see Figure 5.14). Between the low-level upward compression and the high-level downward compression, I was able to maintain the solid low end while bringing out more of the detail and texture of the loop.

Figure 5.14 MaxxVolume on the Loop2 track.

There was no need to use the gate or leveler features in MaxxVolume. The loop doesn’t have anything that requires gating, and the leveler would have tried to smooth out the dynamics too much. The track needs to maintain its punchy sound, and MaxxVolume did a great job in this instance.

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In case you were wondering, using a limiter like I used the Loop3 track back in Chapter 3 would not have worked in this case. The kick portion of the loop was one of its more prominent parts and would have been decimated by a limiter’s fast attack. Speaking of Loop3, I did make one change to that track. You may have noticed that I tend to use a 2-band EQ when I only need one band or a 4-band EQ when I only really need two bands. An example of why I do that occurred while I was comparing the existing L1 plug-in to the L2 and L3. While making the comparisons with Loop3 soloed, I noticed that there was some low-frequency rumble showing up in the PAZ Analyzer between the main kick hits on Loop2. In the interest of cleaning out useless bits of audio, I decided that I needed to add a high-pass filter to the track (see Figure 5.15). Because I’d used RenEQ4 instead of RenEQ2, it was a piece of cake to add that highpass filter. I suppose I could always use Q10 and RenEQ 6-band, but old habits of minimizing CPU hit die hard—even though I’m running an 8-core machine these days.

Figure 5.15 Renaissance EQ 4-band on the Loop3 track.

In this particular application, I couldn’t hear any difference between the sound of L1, L2, and L3 (using the Basic Profile), so I just left L1 in place. L2 and L3 do actually sound better when used in a more normal fashion. The way I’m using a limiter here is an atypical situation.

Bass The bass hadn’t changed since I added MaxxBass (following RenComp) back in Chapter 3. I decided to give RenBass a spin to see how the two plug-ins compared. The improvements that Waves made were indeed substantial. Even with the simplified controls, it was easy to get a great sound. The trickiest part of setting it was finding the right frequency to use. Too low, and it’s muddy; too high, and it gets nasal. Just moving

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it up or down a few Hertz made a big difference. Because the processing is essentially a bell-curve slice of the signal, as you move up the frequency you’re losing the effect in the lower frequencies. In the end, I got rid of both RenComp and MaxxBass, and with a single instance of RenBass I got a result that gave the bass a nice, full sound that minimized the clicky finger noises on the track. See Figure 5.16.

Figure 5.16 Renaissance Bass giving the bass track some heft.

Guitars For this mix I spent some time working on the electric guitar tracks. Now that I had the PuigTec EQs, I wanted to see what they would do. The solo and acoustic guitar tracks were left as is. I’ve found the PuigTec equalizers just sound great on electric guitar tracks. The harmonic distortion and transformer modeling work some magic on guitar tones. For the guitar chords track, I wanted to shape the tone to give it more power but still leave room for the picked guitar track that plays at the same time. With the PuigTec MEQ-5, I added some boost down at 300 Hz to give it a little body and the dip at 700 Hz to make room for the other guitar track (see Figure 5.17). If you recall, when I set up the RenEQ for this track back in Chapter 2, I had an EQ boost at 733 Hz, which goes to show you it’s not a bad idea to reevaluate your earlier mix decisions and try different ideas as the mix progresses. Finally, I added a touch of “fizz” up at 4 kHz to provide a little more edge. Because I had to work to minimize the noise and hum on the guitar solo track, it didn’t make any sense to me to turn around and add noise with an EQ, so I’ve left the Mains set to Off.

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Figure 5.17 PuigTec MEQ-5 on the guitar chords track.

I pressed the PuigTec EQP-1A into service for the distorted picked guitar track (see Figure 5.18). This track suffered from an overabundance of low frequencies that I handled using a high-pass filter in RenEQ. The EQP-1A doesn’t have a high-pass filter, but it does have a low shelf. Since the low shelf has a fairly gentle Q, I selected 20 Hz and used nearly the entire available range of attenuation. To set the amount of attenuation, I kept increasing it until I heard it cutting too much and then backing off a bit.

Figure 5.18 PuigTec EQP-1A on the guitar picked track.

The real fun on the track happens with the high-frequency boost filter. Using the most narrow bandwidth setting, I did quite a large boost at 3 kHz, which gives the guitar significant bite. I have a high-shelf attenuation going at 20 kHz, but given the modeled transformer roll off and the fact that there’s not much above 10 kHz in the guitar track, this cut isn’t doing too much. And as with the PuigTec MEQ-5 I used on the previous guitar track, I’m not adding any Mains noise.

Keyboards The keyboards seemed to still be working in the mix, but never fear—I’m sure I’ll find something to do to them in coming chapters.

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Vocals This time around I made a couple of small tweaks to the lead vocals and finally worked over the background vocals. Now that the drums and guitars were hitting harder, the background vocals were getting lost. In the previous chapter, I set up a subgroup for the drums. For this mix I set up two more subgroups—one subgroup is for the pre-chorus/chorus background vocals (Tracks 25 through 32), and the other subgroup is for the “Ooohs” (Tracks 33 through 36) that come in at the end of the song. As with the drum subgroup, you need to create an auxiliary track and stereo bus to feed it. You then set the output of each set of tracks to the proper subgroup bus. Most of the background vocal processing occurs on the subgroup, but I did a little equalization on a few of the individual background vocal tracks. I’ll discuss these in the coming paragraphs. The verse lead vocal track already had the normal Waves DeEsser on it, and it didn’t sound bad. After giving the Renaissance DeEsser a spin, I could hear a subtle but definite improvement in the sound. When the RenDeEsser was active, it just sounded a little more open and natural. In Figure 5.19, you can see that I’m using a pretty deep range and a fairly high threshold so that, for the most part, it’s only catching the sharper sibilants.

Figure 5.19 Renaissance DeEsser on the lead vocals in the verse.

The chorus lead vocal track seems to have some changes made to it on every mix. That’s usually a sign that it’s not sitting quite right in the track. I thought that the

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tail of the non-linear reverb was a little too bright and was cluttering up the mix. To address this, I pulled down the EQ within the RenVerb plug-in so that there was a 3.7-dB shelf just above 2 kHz (see Figure 5.20). That helped, but I’m not yet convinced that the vocal is where I want it.

Figure 5.20 The modified Renaissance Reverb settings for the chorus lead vocals.

I thought the male background vocal melody (Tracks 27 and 28) could use a boost in the lower midrange to help support the lead vocal. For simple EQ changes like this, I often reach for one of the Q-series plug-ins (see Figure 5.21). A quick 3-dB boost at 507 Hz seemed to do the trick.

Figure 5.21 The Q10 EQ used on the male melody background vocal tracks.

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In the overall mix of the background vocals, it seemed to me that the female harmony parts were sticking out too much. The melody background vocal tracks were simply doubling the lead vocal parts so they blended in nicely. The harmony tracks needed some blending help. Rolling off the low end and pulling down the low mids helped Tracks 29 and 30 blend better (see Figure 5.22).

Figure 5.22 The Q10 EQ used on the female harmony background vocal tracks.

The background vocals now had a little EQ to help them blend as a group. I decided to use Renaissance Channel to do a little shaping to really get them to work for the track (see Figure 5.23). I created a bit of a rollercoaster of an EQ. The PAZ Analyzer showed me that there was some low-end rumble, so a high-pass filter was the first order of business. The lead vocal had strong low-mid content, so I pulled that range down a couple of dB in RenChannel to prevent too much buildup there. I added a 3-dB boost up above 8 kHz to give the background vocals some “air.” To smooth out the dynamics, I choose the RenVox compression algorithm, which was able to get 3 to 6 dB of compression without it sounding overly squashed. This really helped to keep the background vocals powerful through the ends of the phrases, where the singers usually start falling off in volume. As is often the case, there are unintended consequences that become apparent as you start processing a signal. With the compression happening, you could now hear headphone bleed and other noise in between the background vocal phrases. Fortunately, RenChannel also has an expander. Careful setting of the threshold and release time cleaned that right up.

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Figure 5.23 The Renaissance Channel used on the background vocal submix.

I liked the tone that 8-kHz high-shelf boost created, but it also had the unfortunate effect of boosting the sibilants. With eight tracks of background vocals, it’s almost inevitable that you will end up with sibilants that don’t quite all line up at the same time. When you have a large number of vocal tracks, it’s generally a good idea to have the singers back off on the sibilants because of this potential problem. In this case I decided I should use a de-esser to deal with it. The sidebands in RenChannel are fully capable of doing de-essing, but since I liked what the compressor was already doing, I moved the RenChannel plug-in to Slot B and put a Renaissance DeEsser in Slot A to de-ess the vocals prior to the RenChannel processing (see Figure 5.24). In the last 16 bars of the song before the drums stop, there are some “ooohs” that answer the lead/background vocal phrases. The male vocals were overpowering the female vocals, so I gave a 3.2-dB boost at 754 Hz to both of the female vocal tracks. And last but not least, I used an instance of Renaissance Channel on the Oooh vocal submix (see Figure 5.25). I used a similar approach as I did with the background vocal submix. I set up the EQ to get rid of the rumble, pulled down the low mids, and gave them a little high-end bump. I also did some compression to smooth out the dynamics and some expansion to kill the noise between phrases.

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Figure 5.24 The Renaissance DeEsser used on the background vocal submix.

Figure 5.25 The Renaissance Channel used on the Oooh vocal submix.

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What’s Coming Up Next The Platinum bundle expanded your choice of plug-ins in several areas. The majority of the new plug-ins coming in the Diamond bundle are unlike anything you’ve seen so far.

6

The Diamond Bundle

T

he move from the Platinum bundle up to the Diamond bundle doesn’t add any plug-ins that I would consider mixing staples. When working with a bundle that contains in excess of 50 plug-ins, it’s to be expected that many of the new plug-ins are designed to address very specific issues. One specific issue for which the Diamond bundle provides ample tools is noise reduction. If you’ve ever had to deal with a track that was rendered virtually unusable by noise problems, you will immediately see the value of the X-Noise and Z-Noise plug-ins.

Dynamics Our palette of dynamics processors broadens slightly with the addition of a very unique processor and the next evolutionary step up from the L3 Ultramaximizer.

Trans-X The Trans-X plug-in (see Figure 6.1) belongs to a unique breed of dynamics processor that was pioneered by Sound Performance Lab in Germany. The original SPL Transient Designer was (and still is) an analog device that has the ability to independently control the level of the transient and sustained portions of a signal. Waves has come up with their own way to accomplish the task with the Trans-X plug-in. Transient control is useful when you want to emphasize or de-emphasize the attack portion of a signal in a way that sounds very different from what a compressor or a limiter would be capable of. The power of these processors lies in the fact that they can identify transients regardless of their level. A compressor or limiter would only be able to act on transients that were louder than the surrounding audio. Trans-X has four controls: Range, Sensitivity, Duration, and Release. The Range will determine whether you’re boosting or attenuating transients and how big a change you can achieve. You can boost up to 18 dB and cut as much as 24 dB. The Sensitivity and Duration settings work together to determine which transients are processed by Trans-X. With a high Sensitivity setting, Trans-X will process every transient regardless of its amplitude, whereas at lower settings it will ignore the quieter transients. The Duration is expressed in milliseconds. Trans-X will process any transient shorter than the Duration value. Low settings will limit processing to very short

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Figure 6.1 Trans-X Wide.

transients, while progressively higher values will cause longer transients to be processed. The Release control works as it would in a compressor or limiter and determines how quickly it will take Trans-X to return the signal to its normal level after a detected transient’s Duration has elapsed. Trans-X comes in two different configurations—Trans-X Wide and Trans-X Multi— that are related to each other much the way that C1 compares to C4. Trans-X Multi (see Figure 6.2) splits the signal into four frequency bands that each have independent transient shaping controls. Just like C4, Trans-X Multi uses phase-compensated

Figure 6.2 Trans-X Multi.

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crossovers to minimize phase-shift colorization caused by splitting the signal into separate bands. Trans-X also borrows heavily from the intuitive display that Waves uses with the C4 and LinMB plug-ins, which makes it easy to see what’s happening in the processing.

L3 Multimaximizer The L3 Multimaximizer (see Figure 6.3) is Waves’ penultimate weapon in their peaklimiting arsenal. It uses the same processing engine found in the L3 Ultramaximizer that you read about in Chapter 5. The L3 Ultramaximizer uses profiles that act like presets to control the band characteristics, while L3 Multi gives you explicit control over each of these settings. Each band offers controls for Gain, Priority, and Release settings.

Figure 6.3 L3 Multimaximizer.

The Gain works the same way as the band gain in C4 and LinMB and provides an overall band boost or cut. The Release setting also works in a familiar way to set the release time for each band. The Priority is different from what you’ve seen before and acts like an adaptive threshold for a compressor. If one band has a higher priority than another, the band with the lower priority will be squeezed a little bit harder prior to the combined signal being limited. If you find that the audio in one of the bands is sounding particularly squashed compared to the others, you can boost its priority to cause L3 Multi to be more gentle with it. The LL version of the L3 Multimaximizer (see Figure 6.4) follows the same pattern as before. It has the same peak-limiting engine under the hood, but to reduce latency and

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processor hit, it uses phase-compensated IIR crossovers rather than the more transparent FIR crossovers.

Figure 6.4 L3-LL Multimaximizer.

As is the case with the other versions of L3 and to a great extent L2, you won’t find many applications for these plug-ins in a mixing context.

Effects The only new effect that is added when moving from the Platinum bundle to the Diamond bundle is SoundShifter.

SoundShifter The SoundShifter plug-ins (see Figure 6.5) can change the pitch of a signal without changing its speed, they can change the speed without changing the pitch, or they can do any combination of both. Although there are three different versions of the SoundShifter plug-in, you will only see the SoundShifter Pitch plug-in in your list of channel insert plug-ins. The Parametric SoundShifter and Graphic SoundShifter are only available as offline processors, because time compression and expansion simply cannot work as a real-time process in a mix. The basic SoundShifter Pitch plug-in allows you simply to change the pitch, which it does rather well. It has four modes that affect how it approaches its pitch shifting. Using Sync mode will preserve the timing as closely as possible at the expense of

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Figure 6.5 SoundShifter.

tone. Smooth will result in the fewest tonal artifacts, but the timing and transients may suffer. Transient will preserve the quality of the transients at the expense of slight timing changes. And finally, the Punchy mode will attempt to preserve the overall punch of the signal. You’ll have to decide what is the most important aspect of the signal being processed. This is another plug-in that’s not likely to see much action in mixing. I’ve used it to change the key of parts of a song during production when the option doesn’t exist for rerecording them. I’ve also used it when exploring the best key for a song. You can toss it on a preproduction track and see what works best for the singer.

Other Plug-Ins The “Other Plug-Ins” section is well represented in the Diamond bundle, with five new tools that accomplish various tasks.

Center In Chapter 3, I touched on the concept of mid/side miking when I was discussing the S1 plug-in. Not only is it possible to record a stereo signal on two channels as an M/S pair, it’s also possible to take an existing stereo signal and separate the mid and side components. Once the signal has been split, you can process the mid and side portions independently and then recombine them into a normal stereo signal. The Center plug-in (see Figure 6.6) uses a similar process to extract the mid and side elements and allows you to process them in ways not otherwise possible. Center provides basic level controls over the center (mid) and side portions of the signal with the faders on either side of the stereo output meter, but the Low, High, and Punch controls are where the real fun happens. The Low control will shift the low frequencies to either the center or the sides. With the knob straight up at its midpoint, no shifting

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Figure 6.6 Center.

will occur. The Low control sounds as if it’s acting on the audio spectrum up to around 500 Hz, with more shift happening the lower the frequency goes. The High control does the same thing but for the upper frequencies starting at around 8 kHz. The Punch control uses a transient detection algorithm like that found in the TransX plug-in. You can use the control to shift the transients either to the center or to the sides. The horizontal meter at the bottom of the plug-in will show you the level of the detected center signal. The effects of the Punch control are reflected in this meter, so if you’ve shifted them to the sides, this meter will show a lower signal.

Morphoder The Morphoder plug-in (see Figure 6.7) is Waves’ version of the venerable vocoder effect. Dating back to the 1920s, it was developed as a means to encrypt military communications (voice encoder/decoder), but it wasn’t until the 1970s that the effect found its way into music. The basic idea behind the vocoder is that it will take the spectral characteristics of one signal (the modulator) and impose that on another (the carrier). The classic vocoder application is creating robotic voice effects by using a voice as the

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Figure 6.7 Morphoder.

modulator and droning oscillators as the carrier. Kraftwerk fans and Droid phone users will be intimately familiar with this effect. The robot voice is just the tip of the iceberg. Because you can use any signal you like as the carrier or modulator, there are all manner of interesting things you can do with a vocoder. Morphoder will let you select among any of the inputs (left, right, sidechain) to use as the modulator and the carrier. It also has a simple eight-voice synthesizer built in that can be used as the carrier. The synth will play sustained notes, or you can change them via MIDI or track automation. There are several additional controls that adjust the EQ, the filter response, and the output mix of the vocoder. Exploring the presets will give you a good overview of Morphoder’s features. This plug-in is much more likely to be used during production than during the mix, but you never know—sometimes doing something interesting with a vocoder at the mix stage may be just the thing a particular song needs.

Q-Clone Q-Clone (see Figure 6.8) does not have a sound of its own; it’s an equalization doppelganger that assumes the shape of other EQs. It accomplishes this feat by using convolution, which is the same technology that the IR-L Convolution Reverb uses that was discussed in Chapter 3. An impulse signal is passed through an external equalizer and analyzed to produce a filter curve that has the same frequency response as the external equalizer. Q-Clone is not able to capture the non-linearities of the external equalizer, so it will not be able to reproduce any distortion or compression that might be occurring. That’s why it’s so involved to create a plug-in that models a piece of analog hardware— it’s far more than replicating filter curves.

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Figure 6.8 Q-Clone.

If you have a nice external analog or digital equalizer, you can dial in an EQ curve, snapshot it with Q-Clone, and then use Q-Clone instead of the hardware. This frees up the hardware EQ for other tracks. To create the snapshot, you will need to use the QClone Capture plug-in (see Figure 6.9) on a mono auxiliary track. You route the output of the aux track to the hardware EQ so that Q-Capture can send the impulse signal to it. You then route the output of the EQ back into the Q-Capture aux. With Q-Capture transmitting and receiving the impulse through the external equalizer, any instance of Q-Clone can snapshot the curve.

Figure 6.9 Q-Capture.

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X-Noise X-Noise (see Figure 6.10) is actually a suite of plug-ins that each address a different aspect of audio restoration. The suite is made up of X-Noise, X-Click, X-Crackle, and X-Hum. On each of these plug-ins, you’ll find Audio/Difference buttons. Normally you leave them set to Audio so you can hear the processed audio. When they’re set to Difference, you will only hear what is being removed from the signal. This is an extremely useful feature that allows you to make sure you’re not removing any musically useful parts of the audio. As they say, you don’t want to throw the baby out with the bathwater.

Figure 6.10 X-Noise.

When you have a track with constant noise, and the audio you want to keep falls into that same frequency range, you don’t have the option of using filters to remove it, because you’d also be filtering out audio you want to keep. Sometimes you can use a gate to eliminate the noise between phrases and rely on the noise being masked when the keeper audio is playing. When the noise is still audible even when the track is playing, you’ll want to use a broadband noise-reduction tool, such as X-Noise. Broadband noise-reduction tools, such as X-Noise, split the audio into a large number of bands and then apply an expander to each of the bands independently. To do this effectively, X-Noise needs to know how much constant noise there is in each and every band. A noise profile can be generated by playing an isolated sample of the noise while X-Noise is in Learn mode. Once this noise profile has been created, X-Noise will know where to set the expander thresholds. When the signal level of a particular band drops down to the noise threshold, the expander will act and then reduce the noise. When this is happening across many bands simultaneously, you can remove the noise very effectively and transparently without losing the audio you want to keep.

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The Threshold and Reduction sliders work exactly as they would in a single-band expander, but in this case they control the overall behavior of many expanders. The High Shelf filter gives you some control over the noise profile by boosting or cutting the high frequencies of the noise profile. If you’re trying to eliminate some hiss, but you find that you’re losing lower-frequency audio that you want to keep when you have the threshold set low enough to remove the hiss, you should use the High Shelf filter. If you apply a high shelf boost, it will cause X-Noise to start expanding the higher frequencies sooner, which will allow you to raise the overall threshold. The number of bands into which X-Noise splits the audio is determined by the Resolution control. As you increase the Resolution, you will get better noise reduction at the expense of a higher CPU hit. At the higher resolutions, the transients may not sound as sharp; so if you’re hearing this, you should switch to a lower resolution. If you’ve got some isolated noise that you can use to train X-Noise, this plug-in is capable of quite a bit of very transparent noise reduction. When I have a track that has a click or pop in it, I will generally fire up a waveform editor and fix the problem with the Pencil tool by simply redrawing a tiny portion of the waveform. If you have a track that’s plagued by hundreds of these, though, you’d end up with Carpal Tunnel Syndrome by the time you’re finished redrawing them all. X-Click can save you this pain (see Figure 6.11).

Figure 6.11 X-Click.

There are only two controls in X-Click, the Threshold and the Shape. The higher the Threshold, the louder the click must be to be detected and removed. The Shape control tells X-Click what type of clicks to remove. At low values X-Click will only remove very short clicks. Typically, short clicks are the result of some sort of digital clocking

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problem. If you have analog clicks to remove, you should set the Shape to a higher value, because those will last longer than digital clicks. As you play the audio, the last 10 seconds of the waveform will be displayed in the plug-in window. Any time you see a vertical red line, a click has been removed. Crackles refer to small amplitude clicks and pops that you typically hear on an old vinyl album. It’s not really feasible to even attempt to redraw these because there are often thousands of them, and because they are low in amplitude, they are much more difficult to see in a waveform editor. X-Crackle (see Figure 6.12) is another powerful tool with simple controls. The Threshold and Reduction controls determine when X-Crackle starts working. Like X-Click, X-Crackle will show you the waveform of the last 10 seconds of processed audio. Below that is a display that shows green dots when a pop has been removed. The vertical position of the green dots will show you the frequency of the removed pop.

Figure 6.12 X-Crackle.

If you’re trying to clean up some digitized vinyl, you should place X-Click first to remove the large pops and then follow that with X-Crackle to tackle the smaller ones. X-Hum (see Figure 6.13) is essentially a specialized equalizer that is optimized for removing ground-loop noise. A ground loop will create a steady-state tone at either 60 Hz or 50 Hz, depending on the frequency on which the power grid operates in your country. When a ground loop is bad enough, or if it’s subsequently boosted by a guitar amp or another device, it’s likely that harmonics of the ground loop will be created. The center frequencies of the eight bands are tuned to these harmonics. Normally, you will want to keep the Q set very high so as not to affect any more audio than you have to. In extreme cases, you may need to lower the Q to notch out more audio to effectively remove the noise.

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Figure 6.13 X-Hum.

There is also a high-pass filter included in X-Hum so that you can cut out unwanted rumble at the same time.

Z-Noise Z-Noise (see Figure 6.14) is very similar to X-Noise in form and function, but it improves upon X-Noise in several areas. One of the most dramatic differences is that Z-Noise can be effective even when you don’t have an isolated noise sample with which to create a noise profile. The noise profile Extract mode works much like the Learn mode in that you play the audio, and the plug-in analyzes it. It will attempt to recognize the noise in the signal and build a noise profile. It does a

Figure 6.14 Z-Noise.

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reasonable job of doing this, but it cannot build a noise profile nearly as accurately as it can using Learn with isolated noise. When you don’t have the luxury of having an isolated sample of the noise, Z-Noise is an excellent option. If you have noise that changes over time, a static noise profile will not be able to do a good job for the entire file. Z-Noise’s Adaptive mode will continually analyze and generate the noise profile as the audio plays. Where X-Noise has a single high-shelf filter that can modify the noise profile, Z-Noise has five filters that each have four filter types to choose from: low-shelf, high-shelf, bell, and notch. The bell and notch filters are both bell-type filters, but the notch covers a higher Q range than the bell filter. These are used to shape the noise profile, which in turn establishes the individual noise-reduction thresholds over the frequency range. Another feature unique to Z-Noise is its Transients control. If you find that the transients in your signal are getting eliminated because of a high threshold, you can boost the Transients control value. This gives the transients a boost before the noise reduction, and after the noise-reduction processing they are reduced to their original level. This allows the transients to get above the threshold so the noise reduction doesn’t eliminate or reduce them. The Optimize controls work very much like those in SoundShifter, which was discussed earlier in this chapter. Smooth will provide the best-sounding noise reduction, but the transients may suffer slightly. Punch will preserve the transients at the expense of best noise reduction. Normal is a compromise between the other two.

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The Mix To be perfectly honest, the move from the Platinum bundle to the Diamond bundle doesn’t provide many more tools for mixing. The X-Noise and Z-Noise plug-ins are for dealing with problem tracks; Morphoder, SoundShifter, and Q-Clone are specialuse plug-ins; and the L3 Multimaximizer is really just for mastering. That leaves us with TransX and Center as the only real go-to plug-ins for mixing. I did try Center and TransX on nearly every track in the song, but I only found one instance where there was a noticeable improvement in the sound. I could have tried to make them work for the sake of using them, but part of the art of mixing is knowing when to leave well enough alone. These aren’t the kind of tools that you’ll use every day, but they can really save the day when you run into a problem that only they can fix. You’re not going to hear a radical difference between this mix and the one from the previous chapter because I only added or changed two plug-ins over the entire mix. The changes I did make resulted in a subtle improvement in the the mix, so I’ll call it progress.

Table 6.1 Diamond Bundle Mix Track

Slot A

01-Kick-in

RenEQ2

03-Snare-top

V-Comp

04-Snare-btm

Q1

05-Tom-hi

Slot B

Slot C

Slot D

Send 1

C1 comp-gate

06-Tom-mid

RenEQ2

C1 comp-gate

07-Tom-low

RenEQ2

C1 comp-gate

08-Hi-hat

RenEQ4

10-Room

RenEQ2

Drum sub*

H-Comp

11-Loop1

RenEQ4

TransX Wide

12-Loop2

V-EQ4

MaxxVolume

13-Loop3

RenEQ4

L1

14-Bass

RenBass

15-Gtr-chords

PuigTec MEQ5

21.5 dB

16-Gtr-picked

PuigTec EQP1A

18.0 dB

17-Gtr-verse

MV2

25.5 dB

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18-Gtr-solo

Z-Noise

Q4

19-Acoustic

C4

MetaFlanger (m->s)

21-Epiano

Q4

MondoMod (m->s)

22-Synth-chorus

SuperTap2 (m->s)

S1 Shuffler

23-Lead-vox-verse

Q4

RenDeEsser

RenComp

H-Delay (m->s)

24-Lead-vox-chorus

Tune LT

Q4

V-Comp

RenVerb (m->s)

27-Bvox-Mel-M1

Q2

28-Bvox-Mel-M2

Q2

29-Bvox-Har-F1

Q2

30-Bvox-Har-F2

Q2

Bvox sub*

RenDeEsser

33-Oooh-F1

Q2

34-Oooh-F2

Q2

Oooh sub*

RenChannel

Effects 1

IR-L (m->s)

133

Enigma (m->s) 29.0 dB

RenChannel

*Drum sub is fed by Tracks 3 through 10, Bvox sub is fed by Tracks 25 through 32, and Oooh sub is fed by Tracks 33 through 36. The bolded plug-in names in the table indicate plug-in or settings changes between the previous chapter and this one. The (m->s) at the end of the plug-in names denotes that a mono-to-stereo instance of the plug-in is to be used.

Loops The one track that I found that benefited from TransX was the first loop track. This was a support track that had an eighth-note pattern that filled in the spaces between the snare hits. A bit of TransX Wide (see Figure 6.15) allowed those transients to pop out of the mix while leaving the other parts of the loop more in the background.

Figure 6.15 TransX Wide boosting the transients on Loop1.

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Guitars Up to this point, I’d been using expanders of one sort or another to deal with the ground-loop noise on the guitar solo track. These did a good job of removing the noise between phrases where it was most noticeable, and I relied on the noise being masked by other elements while the guitar was playing. If you listened closely, you could still hear the noise—particularly in Bars 15 and 16 of the guitar solo, where there’s a single note that’s held out. As the volume decayed on that guitar, the noise became more exposed. With the Diamond bundle, I finally had some more advanced tools to turn to. Because this was a ground-loop issue, I first tried the 60-Hz preset in X-Hum. It helped a little bit, but at the default Q setting it really didn’t do very much. I tried lowering the Q to affect a wider bandwidth, but I lost too much of the audio I wanted to keep. The problem was that the ground loop started with the guitar or effect pedal and was then overdriven at the amplifier. This took the 60 Hz and created very strong harmonics that permeated the mid frequencies. Having struck out with X-Hum, I moved on to XNoise. The guitar solo track has two seconds of nicely isolated noise at the beginning, which is plenty to train X-Noise. After adjusting the Threshold and Reduction sliders, I was able to remove about 20 dB of the noise and leave the guitar tone largely intact. When the track was soloed, I could still hear a bit of the hum, but it wasn’t bad and would most likely not be heard in the full mix. I could’ve lived with that, but because we also have Z-Noise, I thought it would be a shame not to see what it could do for the track. I went through the same training process with Z-Noise, and I could immediately hear an improvement in the sound. I was able to get 25 to 30 dB of reduction with virtually

Figure 6.16 Z-Noise doing away with the ground loop on the guitar solo track.

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no change in the tone of the guitar. After some careful listening, it seemed as if I was losing a little bit of the pick attack to the noise reduction, so I set the Transients control to 6.3 dB to help it punch through the noise-reduction threshold, and the attack was preserved. Because there wasn’t anything worth keeping down below 40 Hz, I modified the noise profile with a low-shelf boost to cause Z-Noise to apply even more noise reduction in that range. See Figure 6.16.

What’s Coming Up Next The next bundle that I’m covering is the Horizon bundle, which contains the pinnacle of Waves’ peak limiting tools, a bumper crop of savory vintage compressors, and, as is almost expected at this point, a few very unique processors.

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7

The Horizon Bundle

W

aves offers upwards of 30 different bundles of plug-ins. Many of their bundles are collections of specific sets of plug-ins aimed at a particular application or themed sets of plug-ins. For the purposes of this book, I decided to cover their main bundles that each build on the plug-ins from the previous bundle. The Horizon bundle is a slight sideways step because it does not include the X-Noise or Z-Noise plug-ins. However, I do think it makes a good stepping-stone bundle for mixing purposes. In both price and features, it fits nicely between the Diamond and Mercury bundles.

Dynamics A plethora of new compressors come our way in the Horizon bundle. The bulk of them are the result of collaborations with noted engineers Jack Joseph Puig and Chris Lord-Alge, who provided Waves with their favorite hardware for modeling purposes. Although all these compressors do basically the same thing, each one has its own sonic character and reacts differently when presented with the same signal. I suggest you spend some time working with each one to get a feel for what it does with different types of dynamic signals. Fortunately, the interface of each is relatively simple—and in many cases they are very similar—so it doesn’t take much time to learn how to use them. Like the PuigTec plug-ins, all of the vintage modeled compressors have the 50-Hz or 60-Hz noise that you can turn on if you have a burning desire to add noise to your mix.

PuigChild 660/670 The original Fairchild 670 is a 65-pound, 6-rack-space behemoth of a stereo compressor that contains 14 transformers and 20 vacuum tubes. The Fairchild 670’s substantial size and even more substantial price tag make the PuigChild 670 a very attractive alternative (see Figure 7.1). The controls differ from those of other compressors in two ways. The first is that instead of having attack and release time controls, there is a Time Constant switch that controls both attack time and release time. The first four settings use progressively longer attack and release times. Positions 5 and 6 use an automatic release time

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Figure 7.1 PuigChild 670.

algorithm, with 5 being faster than 6. As usual, you should use your ears to see which mode works best for your track. The other departure is the Lat/Ver control. The PuigChild 670 has three stereo modes: Left/Right, Linked, and Lat/Ver. Left/Right mode allows you to set up the left and right channels independently. The Linked mode will disable the lower set of controls and will use the settings on the upper controls for both channels. The Lat/Ver terminology dates back to the time when the Fairchild 670 was being used as a program compressor while cutting vinyl masters. Lat refers to the lateral side-to-side motion of the stylus, and Ver refers to the vertical up-and-down motion. The PuigChild 670 extracts the mid and side portions of the signal using M/S decoding. In Lat/Ver mode, the plug-in will compress the mid signal with the upper channel and the side portion with the lower channel. This allows for applications such as compressing the center portion of the signal much more than the sides. That can give you a more consistent center without sacrificing the overall dynamic feel of the track. With a normal stereo compressor, you are only able to affect the dynamics of the entire signal. The PuigChild also comes in a mono version called the PuigChild 660 (see Figure 7.2) that, for obvious reasons, does not include the Lat/Vert, Left/Right, or Linked controls.

CLA-2A For the CLA plug-ins, Waves worked with acclaimed mixing engineer Chris Lord-Alge, who provided Waves with four of his favorite compressors. If you’ve listened to albums

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Figure 7.2 PuigChild 660.

he has mixed for artists such as Nickelback, Foo Fighters, Alanis Morissette, Snow Patrol, and others, you will realize that he knows a thing or two about compression. First up in the set of Chris Lord-Alge compressor models is the CLA-2A (see Figure 7.3) that’s based on the Teletronix LA-2A Leveling Amplifier. The LA-2A employs an electro-optical attenuation circuit that provides nearly instant gain reduction without undesirable distortion that would normally be present in compressors that use a purely electrical gain reduction circuit. The desirable distortion has been modeled, specifically that from the input and output transformers, as well as the tube in the makeup-gain stage.

Figure 7.3 CLA-2A.

The Peak Reduction control acts as the threshold for the compressor, and the Gain control is the output makeup gain that comes into play after the compression has occurred. The Compress/Limit switch changes the ratio from around 3:1 for compression to around 100:1 for limiting. The limiting here doesn’t act like the digital brickwall limiters that you’ve used so far, such as L1, L2, and L3. Lacking an instantaneous attack, there is no absolute output level that the limiter will not exceed; it simply attenuates it at a very high ratio.

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HiFreq is a sidechain filter control that ranges from 100 down to 0. At 100 it’s not modifying the signal at all, so the detection is working with the full bandwidth signal. This control is supposed to introduce high shelf boost with a corner frequency of 1 kHz. The reason I say “supposed to” is because as you dial it down from 100 to 0, you would expect to hear more compression because you’re boosting the signal into the detector, but you actually get less compression. To my ears it sounds like as you turn it down from 100, you’re actually increasing a low shelf cut. Although the mechanics might be in question, the end results are that the lower you set it, the less the compressor reacts to the low frequencies.

CLA-3A The CLA-3A (see Figure 7.4) is based on Teletronix’s successor to the LA-2A, the LA3A Leveling Amplifier. The biggest difference between the two is that the LA-3A is a solid-state device rather than a tube device. It has the same electro-optical detection circuit, but it has somewhat faster attack and release characteristics when compared to the LA-2A. And, as with the LA-2A, the Comp/Limiter switch provides the only compression ratio control. The effect of HiFreq control settings is much more subtle than with the CLA-2A, and changing them doesn’t affect the overall amount of compression much at all. There is, however, a subtle difference in the release time on lowfrequency content.

Figure 7.4 CLA-3A.

CLA-76 Bluey/Blacky The last of the vintage compressor emulations is the CLA-76, which models two of the revisions of the UREI 1176. The CLA-76 Bluey (see Figure 7.5) is modeled on the 1176

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Figure 7.5 CLA-76 Bluey.

Figure 7.6 CLA-76 Blacky.

Revision B, and the CLA-76 Blacky (see Figure 7.6) is modeled on the 1176LN Revision D. As with the other vintage modeled plug-ins, Waves was provided with the engineer’s favorite units for analysis and comparison purposes during development. The 1176 is another compressor that has a fixed threshold and relies on a combination of the input level and the compression ratio to determine the amount of compression that’s applied. When using any compressor, I always leave the meter set to show me how much gain reduction is happening. With the vintage compressors, this is even more important. The compression ratio is set using the buttons to the left of the VU meter. The available ratios are 20:1, 12:1, 8:1, and 4:1. Long ago, someone figured out that you could get crazy, over-the-top compression by pushing in all four buttons simultaneously. You end up with some low-frequency distortion and some strange pumping that gives you an unnatural yet very cool sound. It’s not something you want to use all the time, but it’s a good trick to keep in your back pocket. I’m guessing it was discovered by some engineer in the UK, because it’s often referred to as the “British Mode.” Waves has thoughtfully added a fifth button that will give you this all-button mode. The 1176 has much faster attack and release times than most other compressors. The attack ranges from 1 millisecond at its slowest to 50 microseconds at its fastest.

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Because a microsecond is 1/1000 of a millisecond, that means that at its fastest, the 1176 has an attack time of 0.005 milliseconds. By comparison, the fastest attack time that the PuigChild does is 0.2 milliseconds, which is 40 times slower. The release time is fairly quick as well, ranging from 1 second down to 50 milliseconds. Both the Attack and Release controls are at their fastest settings when turned fully clockwise to the highest value on the faceplate. The Revision D 1176, which the CLA-76 Blacky is modeled after, is considered by many to be the best sounding of 1176 compressors. The Bluey and Blacky compressors have slightly different attack and release times, and the harmonic distortion is a bit different between the two units. This is another case where you will just want to try them both out and see whether one sounds better than the other for the task at hand.

L3-16 Multimaximizer The L3 Multimaximizer (see Figure 7.7) is the 800-pound gorilla of peak limiters. As a mastering engineer, when I’ve been asked to produce the loudest master I possibly can, I don’t have any choice but to reach for the L3-16. The controls are essentially the same as on the L3 Multimaximizer, so you can refer to Chapter 6 for operational details. Internally, L3-16 splits the audio into 16 different bands and uses a 6-band interface to control the action of those 16 bands. There is no “LL” version of L3-16. I suspect that’s the case because using a phase-compensated IIR crossover filter to split the signal into 16 different bands would yield less than stellar results.

Figure 7.7 L3-16 Multimaximizer.

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Vocal Rider As original and unique as Vocal Rider (see Figure 7.8) is, it could have easily been put into the “Other Plug-Ins” category, but since its job is to deal with vocal dynamics, I decided to list it with the compressors. The short description of Vocal Rider is that it will “listen” to your vocal track and act as if someone is riding a fader to create a consistent output level. Now for the long answer…

Figure 7.8 Vocal Rider.

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To establish the settings for Vocal Rider, you should open up the plug-in and play your song. First set the Vocal Sensitivity knob so that as the vocal plays, the green Ride indicator lights up. If the amber Idle indicator lights during quiet vocal passages, you should turn up the sensitivity. Next you need to set the Target slider at the top to give Vocal Rider an output volume to shoot for. You then fine-tune the behavior by positioning minimum- and maximum-range bounds with the dual-control fader on the left. These will set the limits of how much Vocal Rider is allowed to change the volume of the vocal. Finally, you need to set the Idle Point by dragging the small triangle between the minimum- and maximum-range controls. This will determine where Vocal Rider will return to in between vocal phrases. There is also a Fast/Slow switch that controls the speed of the detection and fader moves. I find that the Slow mode works fine in most instances and provides fader moves that are reasonably transparent. This riding can be written to your DAW as automation data. Writing this automation is as simple as turning on the DAW real-time automation write mode for that track, clicking on the Write button at the bottom of the plug-in window, and playing the track. As you play the sequence, Vocal Rider will write automation data (see Figure 7.9). When that’s finished, you click on the Read button, and Vocal Rider will follow the data that it created. This is useful because it allows you to edit the automation data in your DAW. I will often boost the volume on the vocal at the start of a verse or chorus to focus the listener’s attention on the vocal.

Figure 7.9 Automation data written by Vocal Rider.

When you ride a vocal manually, you will naturally make the vocal louder when the backing track is louder. With a little bit of routing, Vocal Rider can do this as well. To make this work, Vocal Rider needs to have everything but the lead vocal routed to its sidechain input. I do this by creating an auxiliary track to which I route all of the instrument tracks. I then set up a send on that aux that feeds the sidechain input on Vocal Rider. If I leave the aux track fader at unity gain, there will be no change in the mix balance; so once it’s set up, I just leave it. Now that Vocal Rider can “hear” the

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music, you can use the Music Sensitivity knob to adjust how much influence the music has on the overall vocal level. The intensity of the green indicator to the right of the Music Sensitivity knob will tell you how much Vocal Rider is being affected by the music level. When using Vocal Rider, you should insert it as the last plug-in on the track so that it can operate on the final level of the vocal. Vocal Rider kind of does the job of a compressor, but many compressors do contribute to the tone of the vocal, so it’s not uncommon to place a compressor prior to Vocal Rider in the processing chain.

Effects All of the effects plug-ins introduced in the Horizon bundle are part of the GTR bundle. The primary use for these plug-ins is to provide effects and amp simulation for guitar tracks. However, you can use them on anything you like. It’s not uncommon for engineers to use guitar effects in the studio to get interesting tones on non-guitar tracks. When I record guitar tracks, generally a guitarist has his own amp and effects; so I will record the sound of his rig, but I also like to record the direct guitar sound to a separate track so I have the option of using GTR to virtually re-amp the guitar later. Ideally, you should use something like the Waves/PRS guitar interface that will give you a balanced low-impedance output that you can record for the direct track, as well as a 1/4-inch pass-through output to send to pedals and/or amp.

GTR Tuner It’s always good to have a tuner handy, particularly one that’s not going to turn up with a dead battery. I’ve always made it a rule to have all the instruments use the same tuner for a given recording. I have run into too many situations where each player used his or her own tuner, and none of them were quite in tune with each other. If you use Waves’ GTR Tuner (see Figure 7.10), not only will everyone have access to the same tuner, but you’ll also have that same tuner available for subsequent sessions.

Figure 7.10 GTR Tuner.

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GTR Stomp The GTR Stomp virtual pedalboard (see Figure 7.11) comes in 2-, 4-, and 6-pedal configurations and operates like a normal pedalboard, where the input starts on the left and progresses to the right from one pedal to the next. In a mono or stereo instance of GTR Stomp, you simply choose pedals from the list and have fun. Using the mono-tostereo version makes things a little more complicated because you can mix and match mono, stereo, and mono-to-stereo effects. Some of the effects have additional controls when in a non-mono configuration, so you will want to be aware of if and when your effect path goes from mono to stereo.

Figure 7.11 GTR Stomp.

GTR Stomp has a main A/B control up in the Waves toolbar, and each stompbox effect also has its own A/B control. The way that Stomp is designed, you will always have the same set of stompboxes loaded into both the main A and B setups. Switching the toolbar setup from A to B will simply switch all of the individual stompbox setups to their alternate settings. For example, if you’re on Setup A in the toolbar, changing over to B will cause each stompbox that was on A to switch to B, and each stompbox that was on B will switch to A.

GTR Amp GTR Amp (see Figure 7.12) has four configurations to choose from: mono, mono DualCab, mono to stereo, and stereo. The mono configuration is very straightforward because you have a single amp channel and a single cabinet. With the DualCab (GTR Amp 2Cab), you have two separate cabinets that are summed to the mono output. The mono-to-stereo configuration adds panning controls to each cabinet. The stereo instance processes the left and right channels separately and acts as if you have two identical amps that are each feeding their own cabinet.

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Figure 7.12 GTR Amp.

When using any of the dual-cabinet configurations, you can simulate the practice of putting two mics on one amp by selecting the same cabinet for both and then choosing different mics. You can get a variety of guitar tones by inverting the polarity (Phase control) of each cabinet and using the Delay control to change the phase relationship between the two cabinets. If your project is running at a 44.1-kHz or 48-kHz sample rate, you have the option of turning on HD mode. That will cause the plug-in to internally upsample so it can process the signal at a higher sample rate, which will provide higher-fidelity results even after the conversion back to your project sample rate. If you have the spare CPU power, it certainly can’t hurt to run GTR Amp in HD mode.

GTR Tool Rack When you think you’ll need all the GTR plug-ins, you should reach for GTR Tool Rack. It combines Stomp-6, Amp, and Tuner all in one handy plug-in. It also allows you the flexibility of putting effects before or after the amp simply by dragging the amp divider left and right. In Figure 7.13, I have four stompboxes before the amp and two that follow it. GTR Tool Rack also introduces different routing schemes that can be selected separately for the stompboxes that are before the amps and the stompboxes that are after the amps (see Figure 7.14). The default is Cascade, which is how GTR Stomp works with the signal flowing from one box to the next. You can get fancy and run parallel signal chains within GTR Tool Rack by switching the routing from Cascade to Parallel and Split. In these modes, the signal will have two paths so that the odd stompboxes process separately from the even ones. In Split mode, the left input goes to the even boxes, and the right input goes to the odd boxes. In Parallel mode, both the left and

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Figure 7.13 GTR Tool Rack.

right signals are fed to each signal path. This all sounds very complex, but all will be revealed when you click on the Routing View button.

Figure 7.14 GTR Tool Rack in Routing view.

Finally, there are dedicated buttons on the right side that allow you to independently bypass the amps and stompboxes. GTR Stomp, GTR Amp, and GTR Tool Rack all have the same MIDI Learn facility that is found in the Hybrid Delay, which was discussed in Chapter 4. If you have a MIDI-capable foot controller, you can turn these plug-ins into your own virtual pedalboard.

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Other Plug-Ins The Horizon bundle only has one curveball to throw, but it’s a good one.

DeBreath Here’s another example of one of Waves’ unique plug-ins. DeBreath (see Figure 7.15) is a specialized expander that is designed to identify and attenuate the sound of a singer taking a breath. There are times when the combination of the singer, the mic, and the vocal phrasing result in the breaths becoming obnoxiously loud. The use of compression on the vocal track generally makes them even louder. It’s possible to comb through the track and edit out all the breaths, but that will often end up sounding unnatural— the singer has to breathe sometime. Doing volume automation on each of the breaths is even more tedious. It’s this precise situation where DeBreath comes into play.

Figure 7.15 DeBreath.

You target the breaths by adjusting the Breath Threshold and Energy Threshold sliders. For a signal to be considered a breath, it has to be above the Breath Threshold and below the Energy Threshold. In Figure 7.15, you can see the properly identified breaths that are highlighted in the graphs. The Reduction slider controls how much attenuation is applied to detected breaths. The Monitor control works much like the X-Noise Audio/Difference controls, where you can listen either to the processed signal without the breaths or to just the audio that’s being removed. This is useful for fine-tuning the settings.

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When using DeBreath in an exposed spoken-word situation, it becomes very apparent when the breaths are completely removed. This is because there will always be some background noise that you can hear between phrases, and the removal of the breath leaves you with nothing. If the Room Tone button is on, then instead of nothing, some low-level white noise is added when the breath is removed. DeBreath is surprisingly accurate in attenuating breath sounds. I didn’t end up using this plug-in on my vocal track because there wasn’t a problem with the level of the breath noise, but if you drop this plug-in on the verse vocal track and use the default setting, you can see how it accurately identifies and reduces each breath that the singer takes.

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The Mix The introduction of the vintage compressors gave me the impetus to revisit a number of tracks to rebuild the processing around the tone of those compressors. The guitar and lead vocal tracks got all of the attention this time around. The drums, loops, bass, and background vocals didn’t have any changes made to them. I also decided to take the reverb that was set up as an inline effect on the chorus lead vocal and move it to an auxiliary track in a send/return configuration.

Table 7.1 Horizon Bundle Mix Track

Slot A

01-Kick-in

RenEQ2

03-Snare-top

V-Comp

04-Snare-btm

Q1

05-Tom-hi

Slot B

Slot C

Slot D

Send 1

Send 2

C1 comp-gate

06-Tom-mid

RenEQ2

C1 comp-gate

07-Tom-low

RenEQ2

C1 comp-gate

08-Hi-hat

RenEQ4

10-Room

RenEQ2

Drum sub*

H-Comp

11-Loop1

RenEQ4

TransX Wide

12-Loop2

V-EQ4

MaxxVolume

13-Loop3

RenEQ4

L1

14-Bass

RenBass

15-Gtr-chords

PuigTec MEQ5

16-Gtr-picked

PuigChild 660

PuigTec EQP1A

18.0 dB

17-Gtr-verse

MV2

GTR Amp

25.5 dB

18-Gtr-solo

C1 Gate

GTR Stomp 2

19-Acoustic

CLA-3A

PuigTec EQP1A

21.5 dB

Enigma (m->s)

CLA-2A

32.5 dB 29.0 dB 16.4 dB

20-Mel-synth 21-Epiano

Q4

MondoMod (m->s)

22-Synth-chorus

SuperTap2 (m->s)

S1 Shuffler

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Table 7.1 Horizon Bundle Mix (Continued ) Track

Slot A

Slot B

Slot C

Slot D

23-Lead-vox-verse

RChannel

CLA-2A

Vocal Rider

H-Delay (m->s)

24-Lead-vox-chorus

Tune LT

Q4

CLA-76

Vocal Rider

27-Bvox-Mel-M1

Q2

28-Bvox-Mel-M2

Q2

29-Bvox-Har-F1

Q2

30-Bvox-Har-F2

Q2

Bvox sub*

RenDeEsser

33-Oooh-F1

Q2

34-Oooh-F2

Q2

Oooh sub*

RenChannel

Effects 1

IR-L (m->s)

Effects 2

RenVerb (m->s)

Send 1

Send 2

11.4 dB

RenChannel

*Drum sub is fed by Tracks 3 through 10, Bvox sub is fed by Tracks 25 through 32, and Oooh sub is fed by Tracks 33 through 36. The bolded plug-in names in the table above indicate plug-in or settings changes between the previous chapter and this one. The (m->s) at the end of the plug-in names denotes that a mono-to-stereo instance of the plug-in is to be used.

Guitars I changed the processing on every single guitar track. With all these vintage compressors, I was like a kid in a candy store. I liked what the PuigTec EQP-1A was doing for the picked guitar track, but I still felt as if it could use a little more presence in the mix, so I gave some of the vintage compressors a spin. In a situation like this, where you’re not looking for a lot of compression, you can clearly hear the effects of the modeling of the tubes and transformers. After some auditioning, I settled on the PuigChild 660 (see Figure 7.16). The way I’ve got it set up, it’s only averaging about 2 dB of gain reduction, so it’s not making any radical changes to the dynamics. It is, however, adding some very nice analog-like tone to the track. When you bypass the compressor, you don’t hear a change in the volume, but you can certainly hear that it gives the guitar more presence in the mix. In many cases you would normally place the compressor after the equalizer because you want the compressor to provide even dynamics based on the final tone of the track. In this case, I found that it sounded better to put the compressor before the EQ. If you’ll recall, this track is playing chunky staccato chords in the pre-chorus that required some

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Figure 7.16 PuigChild 660 on the picked guitar track.

high-pass filtering to get under control. When I put the compressor after the EQ, those chunky chords were lighter, so they got less compression and were louder than they should be. By putting the compressor before the EQ, the chunky chords got a little more compression and sat in the track better. The verse guitar had a cool, semi-distorted dry sound that I liked, but I felt it could use a little more life. The original track was recorded direct, so the cabinet simulator in GTR Amp was my first stop (see Figure 7.17). Since the guitar track had already been through an amp, I bypassed that portion of the plug-in, which left me with the cabinet and microphone choices. Each one of these produced a strikingly different tone, so it was a matter of auditioning the different combinations until I got something I really liked. I settled on the Neil Citron cabinet with the Electron_2_12_Clean microphone.

Figure 7.17 GTR Amp on the verse guitar.

If you think that 2_12 sounds more like the name for a cabinet than the name of a microphone, you would be correct. For the Citron set, instead of providing different microphone choices, they decided to model specific mic setups on specific cabinets.

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These abstract presets can include models of multiple mics that noted engineer Neil Citron personally set up for each particular cabinet to capture the sound in the way he records guitars. I don’t know exactly what mics are being modeled for the one I chose, but I do know that I like the sound. In the previous chapter, we finally eliminated that annoying ground-loop noise in the guitar solo track. Unfortunately, the X-Noise and Z-Noise plug-ins are not part of the Horizon bundle, so I had to go back to C1 (originally set up in Chapter 3) to simply gate the noise between phrases. It’s not too bad, but you can definitely hear its effect on the guitar tone when the notes are ringing out. With the new GTR toys at my disposal, I decided to replace the Q4 I’d set up back in Chapter 3 with something a little more in character with the track. I dropped in an instance of GTR Stomp 2 in place of Q4 and dialed in some EQ with the Tone stompbox (see Figure 7.18). It does sound different from Q4, but I don’t suspect I’m actually getting anything that I couldn’t have gotten by using a normal equalizer plug-in. I also tried out all the phaser-style stompboxes that GTR Stomp had to offer, but none of them was able to produce an effect cooler than the one I already had going with Enigma.

Figure 7.18 GTR Stomp 2 on the solo guitar.

I was on winning streak with using vintage compressors to add vibe to guitar tracks, so I saw no reason to leave well enough alone. The standout for this track was the CLA2A (see Figure 7.19). It added a nice midrange presence while rounding out the bottom, courtesy of the tube modeling.

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Figure 7.19 CLA-2A on the solo guitar.

The acoustic guitar was something of a problem child, and I wasn’t entirely happy with what C4 was doing for it. I liked what CLA-2A did for the solo guitar track, but I found that the tube modeling in CLA-2A made the acoustic tone too thick. Switching that out with the solid-state CLA-3A gave it the right amount of bite and a tone that wasn’t too tubby. The other guitar tracks had the vintage compressors providing more tone than compression, but in the case of the acoustic guitar, I set up CLA-3A to really dig in (see Figure 7.20). I also set the HiFreq sidechain filter all the way up to 100, where it let a little more of the pick attack through.

Figure 7.20 CLA-3A on the acoustic guitar.

I followed CLA-3A with a PuigTec EQP-1A to do some tonal sculpting (see Figure 7.21). A little boost at 100 Hz and a more substantial one at 3 kHz gave the acoustic a little more bite and body. Had I put the EQ before the compressor, it would have caused even more compression to occur, and I wouldn’t have gotten the edge I was looking for. All in all, I think the new processing chain for the acoustic guitar provided a more natural sound that also contributed more to the mix.

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Figure 7.21 PuigTec EQP-1A on the acoustic guitar.

You might have noticed that the level is pretty hot coming out of the EQP-1A. Running the signal hot like that does introduce a small amount of distortion, but in this case, I was looking for bite, so I didn’t mind a bit of that. I did some testing and found that even if I boosted the signal coming into this plug-in by 40 dB, it only resulted in modest distortion in the output. I suspect that if you did the same thing with a real Pultec, your signal would end up pretty crispy.

Keyboards The only thing I did to the keyboards this time around was to send some of the Mel Synth to the new reverb aux I set up.

Vocals The lead vocal tracks got their processing chains worked over pretty well. The effects remained the same, but the dynamics processing is completely different. The first order of business was to replace Q4 and RenDeEsser with a RenChannel plugin (see Figure 7.22). I duplicated the EQ high-pass/high-shelf filtering pretty easily, but the de-essing took a little more work. I selected the RenComp compression algorithm and then set up an internal sidechain band-pass filter that isolated the sibilant frequency range for the vocal. A little work with the threshold and ratio got the de-essing that I was looking for. If the guitar tracks didn’t make the point that I’m a fan of vintage compressors, maybe the vocal tracks will. CLA-2A made a good replacement for the RenComp that I had been using up until this point (see Figure 7.23). My RenComp settings used the Warm setting so the bottom end wasn’t lost under higher gain-reduction levels. The low-end emphasis that the tube modeling provided filled this role nicely. All in all, I was able to get the same perceived dynamic smoothing with the CLA-2A, but with less actual gain reduction than RenComp was doing.

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Figure 7.22 Renaissance Channel on the verse lead vocal.

Figure 7.23 CLA-2A on the verse lead vocal.

Not only is Vocal Rider one of Waves’ more clever plug-ins, it works amazingly well. The main dynamic change in the music occurs between the verses and the rest of the song, so with the verse and the chorus vocal tracks already split onto separate tracks, there wasn’t

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any need to use the music sensitivity feature in Vocal Rider. It was just a matter of setting the various controls to get the vocal to sit where I wanted. See Figure 7.24.

Figure 7.24 Vocal Rider on the verse lead vocal.

The first two plug-ins on the chorus lead vocal track remained the same. Tune was still doing its job, and there wasn’t any reason to use anything other than Q4 to provide high-pass filtering. The V-Comp plug-in that I switched to in Chapter 4 did a fine job with the vocal, but I knew I could get more attitude out of the vocal with CLA-76 (see Figure 7.25). My first-call compressor for vocals is generally the CLA-2A because I like the warmth and smooth compression that the electro-optical circuit provides. In this case, I wanted the vocal to own the chorus, so something a little more extreme was called for. I normally reserve the use of all-button mode for special effects, but in this case it did some really nice things to the vocal. The all-button mode provides a huge amount of compression, so it’s a good idea to use a fairly fast release time. I needed to use a faster attack with the all-button mode because the peaks that slipped through got quite loud, and the faster attack kept them under control. I tried both the Bluey and the Blacky in this application and thought that the Bluey sounded a little better, so that’s what I went with.

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Figure 7.25 CLA-76 jumping all over the chorus lead vocal.

The Vocal Rider setup for the chorus vocal is very similar to the verse lead vocal track with slight differences in the Target level, Output level, and upper and lower ranges (see Figure 7.26).

Figure 7.26 Vocal Rider on the chorus lead vocal.

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What’s Coming Up Next In the next chapter, we get to play with everything that Waves has got. The Horizon bundle loaded us up with vintage compressors, and the Mercury bundle will provide us with a complementary slate of vintage equalizers. It seems that in each bundle Waves has a surprise or two in the way of unique processors. The Mercury bundle doesn’t disappoint in this regard.

8

The Mercury Bundle

T

he Mercury bundle includes all of the plug-ins that Waves has to offer with the exception of their SSL plug-ins, which are only available as a stand-alone bundle. With all of the excellent analog modeled equalizers and compressors that are already included in the Mercury bundle, I didn’t see a pressing need to cover the SSL plug-ins in this book. Also included in the Mercury bundle are all of the “signature” plug-in bundles from Jack Joseph Puig, Tony Maserati, Chris Lord-Alge, and Eddie Kramer. I’ve also elected not to discuss those bundles because this book is focused on the nuts and bolts of mixing, and those plug-ins use an abstract approach to processing. Behind the scenes, the signature bundle plug-ins are actually made up of several standard Waves plug-ins that have linked controls with preset frequencies and ratios that have been set up by the engineer. Using them is something of a guessing game because you really don’t know what is happening when you a change a control labeled Girth or Presence. These bundles are targeted more at entry-level users, but they are included in Mercury because it has all the Waves plug-ins. I do have to say, though, that I’ve found the Chris Lord-Alge signature plug-ins to be the most intuitive, predictable, and useful of all four of the signature bundles.

Equalization To make up for the lack of new equalizer plug-ins in the last two bundles, we now get four new lovely vintage equalizers to complement all the vintage compressors that were introduced in the previous chapter. These expand our sonic palette quite nicely. We also get the LoAir plug-in, which can be thought of as the evil twin of MaxxBass.

API 550A/API 550B Finally, we get some plug-ins that use the actual name of the original manufacturer. Instead of calling them something like “Apey Eye,” Waves is able to market authorized plug-in versions of the API hardware using the name and the same front panel look. You’ll notice that the API equalizers all share the same distinctive vertical format. This is because the hardware they are modeled on consists of all 500-series modules, which fit into a standardized slot that API used on their consoles. Later the “lunchbox” frame

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came along that provided the power supply and terminals for multiple modules so they could be used as outboard gear. Since there was no need for an internal power supply, each module was less expensive to manufacture and allowed for a lot of processing in a small space. After the third-party market exploded for these 500-series modules, API started the VPR Alliance to bring official technical standards to the format. All of the filters in the API units use what they call Proportional Q, which causes the bandwidth of the filter to change depending on how much boost or cut is applied. With small changes, each filter has a low Q so it affects a wider bandwidth. As you increase the boost or cut, the Q increases so that you affect a narrower band. In many cases this is exactly what you want a filter to do in a musical context, so these filters are very usable even without Q controls. The API 550A (see Figure 8.1) and 550B (see Figure 8.2) are similar to the V-EQ3 and V-EQ4 in that they are 3-band and 4-band equalizers, they both use concentric controls, and the frequency selection involves choosing specific frequencies rather than

Figure 8.1 API 550A.

Figure 8.2 API 550B.

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being fully sweepable. They differ in that the API filters have a +/12-dB range, the boost/cut controls also use fixed values that are not fully sweepable, and the knobs are reversed such that the frequency selection is the inside knob and the boost/cut is the outside knob. The layout of the knobs follows the original hardware very closely, with the frequency knob rotating through the bottom range of the knob and the boost cut above. I find these controls to be even more difficult to manage than the V-EQ, so I always click on the labels to set the frequency and boost/cut. At the bottom of the API 550A plug-in there are some additional controls. The FLTR toggle engages an overall band-pass filter that will roll off 50 Hz and lower and 15 kHz and higher. The slope of the high-pass portion seems to be steeper, because I hear more attenuation in the low frequencies than in the high frequencies. Below the FLTR switch are the bell/shelf controls that change the filter type for the high and low bands. Between those is the filter in/out switch, which will bypass the filters but leave the harmonic distortion and noise. The API 550B differs from the 550A in that it drops the fixed band-pass FLTR, includes a fourth filter band, and provides seven frequency choices for each filter rather than five. The bell/shelf controls for the top and bottom bands are presented as small toggle switches next to the filter knobs. On the right side, both the 550A and 550B provide a Polarity invert switch (finally, a polarity control that’s not labeled “Phase”), an output gain control, and an Analog on/ off switch. Unlike the Mains and Analog controls on the other modeling plug-ins that I’ve covered so far, the Analog switch in the API plug-ins introduces some harmonic changes in addition to adding just a small bit of noise. With the API equalizers, I think the Analog mode is well worth using. It is nice to see plug-ins that look just like the original hardware, but in this case it’s a mixed blessing. The compact interface and concentric knobs are important for use in a console channel strip or a 500-series module, but in a virtual version they make the plug-ins a little tricky to work with.

API 560 For the uninitiated, the API 560 (see Figure 8.3) might look a lot like the graphic EQ that you added to your car’s stereo as a teenager, but in reality it’s a very different beast sonically. It took me a while to get over my aversion to graphic equalizers because I’d had many bad experiences with phase-shift issues that many graphic EQs are plagued with. Once I got past that, I found that the combination of the Proportional Q filters and the intuitive layout of the graphic EQ make for an equalizer that sounds great and is very easy to use.

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Figure 8.3 API 560.

Kramer HLS Channel The “HLS” in the Kramer HLS Channel plug-in name is another pseudonym, along the lines of CLA and PuigTec. In this case it’s a stand-in for Helios. It has the Kramer name because, like the PIE compressor, Eddie worked extensively with Helios gear back in Helios’ prime, and he helped Waves by selecting the existing channel that best characterized the Helios sound, which was then modeled. The HLS (see Figure 8.4) has three filters. The high-frequency filter is a fixed 10-kHz high-shelf filter that has a very gentle slope that reaches quite a ways down below 10 kHz. The only knob for this filter is the boost/cut control, which is pretty coarse and can only be changed in 4-dB increments. The mid-frequency filter is composed of a gain adjust knob, a frequency selection knob, and a boost/cut toggle switch. The frequency selection knob has eight frequencies you can choose from that range from 700 Hz to 6 kHz. The boost/cut is different from other filters in that you’ve got a single sweepable knob that sets the amount and a

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Figure 8.4 Kramer HLS Channel.

toggle that determines whether you’re boosting or cutting by that amount. The toggle is labeled PK for “peak,” which boosts, and TR for “trough,” which cuts. If you think the mid filter is odd, it’s just because you haven’t read about the bass filter yet. It consists of two knobs—one that’s a boost amount and another that performs multiple jobs, acting as a frequency selector, a high-pass cut amount, or a filter bypass. With the bass filter knob at the zero position, the filter is disabled. As you rotate it counterclockwise, you are selecting the boost frequency. Even when leaving the low band boost knob at zero, you still get a little EQ gain. At maximum gain you are getting 15 dB of boost at any of the four fixed frequencies. When you rotate the low band knob clockwise from zero, you are selecting how much attenuation you’re getting with the 50-Hz high-pass filter. In the high-pass mode, the gain knob has no effect. You might expect that a plug-in that uses the word “channel” in its title would include a compressor, as is the case with other channel strip plug-ins. I suppose Waves named it

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this way because the HLS has Mic/Line and Preamp gain controls. On a mic preamp, the Mic/Line switch and the Preamp gain knob would drastically change the input level, but with the HLS Channel, they’ve modeled the harmonic distortion that’s present at each setting and normalized the output levels. The distortion in Line mode is subtler than in Mic mode, which has more low-frequency emphasis. As you increase the Preamp gain, you will get more distortion in either mode. The Analog switch will simply inject noise into the output signal. The original unit was pretty noisy, so Waves also provides a Noise switch that can be set to Orig or Lo, where the latter attenuates the noise by 20 dB. I always leave the Analog switch off when I use the HLS. This plug-in also shares the same cleverly hidden VU calibration feature found in the Kramer PIE compressor.

LoAir As I mentioned in the beginning of this section, LoAir (see Figure 8.5) is like the evil twin of MaxxBass. Instead of moving low frequencies up, LoAir will generate subsonics below the input signal. It was initially designed as a surround tool to provide lowfrequency generation and routing for the LFE channel for 5.1 mixing, but many beta testers (including myself) requested a stereo version for music production purposes. Waves listened, so now we have another specialized tool for mixing.

Figure 8.5 LoAir.

The Range control sets the corner frequency for the high-pass filter that’s used to isolate the signal for processing. LoAir will create low-frequency enhancement in two ways—

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one is by boosting the selected frequency range using the Lo control, and the other by generating audio that’s an octave below the input signal using the LoAir control. The Direct control allows you to attenuate the direct signal. If you were to use LoAir as a send effect, you would want to turn this all the way down so you would have 100% wet output. The Align button is used to minimize phasing problems between the direct and effected signals. When engaged, it will cause the direct signal to be delayed by around 360 samples, which works out to just over 8ms in a 44.1-kHz sample rate project. You’ll have to decide whether the phase relationships between the direct and effected signals are more important than those between the track and the rest of the mix. You need to be careful when using this plug-in—it’s capable of creating all sorts of ugly, distorted muck that will cloud your mix. On the other hand, it can be pure gold when you’ve got an anemic loop, a weak kick, or a thin bass track—the judicious addition of sub-frequencies can really make a track nail the low end.

Dynamics Mercury gives us a pair of new dynamic plug-ins, which rounds out our current tool set nicely.

Kramer PIE Compressor In case you’re not familiar with the name, Eddie Kramer has engineered albums for a few artists that you may have heard of, such as the Beatles, the Rolling Stones, Led Zeppelin, Jimi Hendrix, David Bowie, and others. At the dawn of modern recording, there weren’t many companies making recording-specific equipment. Much of the gear was either custom-made or adapted from other industries. The Kramer PIE Compressor, shown in Figure 8.6, is a model of the venerable Pye Telecom compressor that was manufactured back in the 1960s for broadcast applications. The original Pye compressor was a solid-state device, so it didn’t include tubes, but it did have transformers that colored the sound in a pleasant way. As with the PuigTec equalizers, Waves has modeled the effect of the transformers as well as the harmonic distortion that the device introduced to the signal. This modeling provides a good reason for the PIE to have a 1:1 compression ratio—it allows you to add some analog-like goodness to a track without actually compressing the signal. The controls are all very straightforward and include Threshold, Output Gain, Decay Time, and Compression Ratio. There is one hidden control that is easy to overlook. The screw that’s directly below the VU meter needle can be used to calibrate the input and output meters. If you’re feeding a hot signal to the PIE compressor, you will likely see the VU meter pinned when monitoring anything but gain reduction. You can click and drag on that screw to change the calibration for the VU meter. This doesn’t have any effect on the sound, but it’s still a nice touch.

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Figure 8.6 Kramer PIE Compressor.

API 2500 The new dynamics plug-ins are rounded out with a plug-in modeled after the formidable API 2500 stereo bus compressor (see Figure 8.7). It has the normal Threshold, Attack, Ratio, Release, and Knee controls, but it also possesses controls not found on other compressors I’ve covered so far. The Ratio works a little differently in that it’s not sweepable and instead has seven discrete settings that range from 1.5:1 through 10:1—and at its highest setting, ∞:1. The Release control has six discrete settings from 50ms to 2 seconds, but it also has a seventh continuously variable mode that ranges from 50ms to 3 seconds. In the analog world, it makes sense to have discrete, perfectly recallable settings, but in the plug-in world, because you can recall anything, there’s not as much reason for this. I just leave the Release set to variable and adjust to taste.

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Figure 8.7 API-2500 compressor.

The Thrust setting controls the preset sideband filter. The Loud setting applies a tilt filter that starts with a 15-dB low-frequency cut and smoothly transitions to 15-dB high-frequency boost. The Med setting has a low cut, a flat midrange, and a high boost. Norm bypasses sideband filtering altogether. The Med and, to a larger extent, the Loud settings will reduce the amount of pumping caused by low frequencies and will result in a punchier sound overall. The compression Type setting changes the compression architecture from feed forward to feed back. A feed-forward compressor will send the input signal directly to the detector, and a feed-back compressor uses the output of the compressor to feed the detector. Each has its own compression characteristics, suitable for different material and different sounds. For comparison purposes, older compressors, such as the 1176 and the LA-2A, are feedback designs, whereas the popular dbx compressors use feed-forward architectures. The Link section is unique to the 2500. Most stereo compressors automatically link the channels or at most offer the option of delinking the channels. The 2500 gives you control over how much linking happens. When Link is set to IND, the detector on each channel will work completely independently. You may also link the channels from 50% linked to 100% linked. At each link setting, a percentage of the sideband signal for one channel will be mixed with the other so that the compressor will act in a more uniform fashion. When you are linking, you can also apply a sideband filter that will only apply to the signal being cross-mixed. This filter can be a low-pass, highpass, or band-pass filter. This can be useful if you have some hard-panned high- or

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low-frequency transients that could cause excessive compression in the opposite channel. You can use the sideband filter to mitigate this. The Analog switch on the API 2500 will simply add noise to the signal. The harmonic distortion effects are independent from the Analog switch setting. Even though this was designed as a mix or submix bus compressor, there’s no reason why it can’t be used on an individual track. In the virtual world, we are not constrained by things such as $3,000-per-instance price tags.

C6 Multiband Processor Here’s another plug-in reprise with a twist. You can find all the C4 controls present in C6 (see Figure 8.8). They might be in slightly different places, but they all work identically to those in C4 (see Chapter 4). The big news is that there are two additional compression bands, and instead of just splitting the signal into six bands, the two new bands can “float” independently at any frequency. There have been many times when I’ve had C4 set up on a track and then realized that I needed to squeeze the signal in a frequency range that I hadn’t anticipated. This usually means that I need to reconfigure the crossover points and move the bands around to get what I need. With C6, it’s as simple as enabling one of the floating bands and leaving everything else as is. The additional bands, as well as the larger plug-in window, make C6 a powerful upgrade to C4.

Figure 8.8 C6 Multiband Compressor.

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The sideband version of the C6 (see Figure 8.9) adds another dimension of dynamic control to C6. As with any plug-in that can accept an external sidechain, you can route any signal you like into the sidechain. The kicker with C6 is that you can selectively set each band to use either the main input signal or the sideband as the input to the compression detector. With this type of control, you can build things such as combination compressor/duckers and more.

Figure 8.9 C6 Sidechain Multiband Compressor.

The S/C Mode (sidechain mode) applies to both the internal and external signals used for compression detection in C6. It has two modes: Split and Wide. In Split mode the sidechain signal is filtered according to the bandwidth of each band. If you were to use a boomy kick drum as the sidechain input, Split mode would result in more compression in the lower bands and less compression in the higher bands because most of the energy in the kick is in the lower frequencies. In Wide mode that same sidechain signal is fed in its entirety to each band, so the compression in the high bands would be triggered by the same signal as the lower bands. Most of the time, Split mode makes the most sense for what I’m doing with C6.

Effects There is only one new effect in the Mercury bundle, and it’s not even completely new. It does offer quite a bit of power, though.

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IR-1 Convolution Reverb The IR-1 Convolution Reverb (see Figure 8.10) uses the same engine as the IR-L Convolution Reverb that we covered in Chapter 3. IR-1 can load all the same impulse responses and presets as IR-L, but IR-1 provides many more parameters that you can use to tailor and—if you so desire—mutilate the sound. Many convolutionbased reverbs don’t allow for the same kind of flexible parameter manipulation that algorithmic reverbs provide; IR-1 bridges that gap.

Figure 8.10 IR-1 Convolution Reverb.

The Size, Density, and Decorr controls work the same as the similar controls found in TrueVerb and RenVerb. The Damping EQ also works the same, but IR-1 generously provides two shelf and two bell filters. The Gain Envelope is the horizontal orange line that’s shown about a third of the way down from the top in the IR graph. With the Gain Envelope, you can subtly change the decay characteristics of an impulse, or you can change it into something that could not exist in nature. Double-clicking on the line will add a breakpoint that you can then position anywhere in the graph. The reverb produced with the Gain Envelope shown in Figure 8.10 is clearly non-linear and indeed very odd-sounding. Changes to the EQ, Gain Envelope, and Reverb Time are all reflected in the IR graph, so there is visual feedback for the changes you make. Another interesting control is the Reso slider, which will increase or reduce resonances found in the IR. You can use this to tame ringing that existed in the original impulse response or to create odd, highly resonant spaces.

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IR-1 provides extensive control over individual components of the reverb. When an impulse is recorded in a space, there will be an initial signal that will reach the measurement mics before any of the reflections. This is the direct signal, which is not exactly the same as the dry signal because it includes the effects of the speakers and microphones used to produce the IR. If you’re using IR-1 as an inline effect, you can choose to either use the wet/dry mix control or use the direct signal. When using IR-1 as a send effect, you should set it to 100% wet with direct turned off. You also have control over the relative levels and the pre-delay of the direct, early reflection, and tail portions of the signal. The small round button found between the parameters serves to link controls. In Figure 8.10, the levels of the early reflections and the tail are linked. The early reflections and the tail signals are extracted by analysis by IR-1. The ER/ TR-X control will allow you to override the division between the two by manually specifying the duration of what’s considered the early reflection. The ER Buildup control seems to allow you to extend and layer the early reflections. The Reverse button is kind of fun because it lets you reverse the entire impulse response. It’s not one of your everyday needs, but it’s fun nonetheless.

Other Plug-Ins The Mercury bundle returns the X-Noise and Z-Noise plug-ins to the fold and augments those with a pair of noise-reduction plug-ins designed for dialogue cleanup. We also get the full-blown version of Waves Tune.

W43 Noise Suppressor You wouldn’t think that noise-reduction devices would fall into the desirable “vintage” category, since there has been constant advancement in the technology, but the venerable Dolby Cat 43, on which the W43 is based, is still in use today. This was a groundbreaking device for which Dolby received an Academy Award for Technical Achievement in 1987. Devoid of any sort of metering, the W43 (see Figure 8.11) is most definitely a device that you use your ears to set up. The controls are really quite simple; as you move the Threshold fader up, you will get more overall noise reduction. As you move each of the frequency faders down, you will get more noise reduction in the respective band. It’s easy to see how useful a tool like this would be in the fast-paced world of postproduction.

WNS Noise Suppressor The WNS (see Figure 8.12) works very much like W43 with the addition of a number of features. WNS has six noise-reduction bands and also includes high-pass and low-pass

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Figure 8.11 W43 Noise Suppressor.

Figure 8.12 WNS Noise Suppressor.

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filters, which are applied to the output of the plug-in. The frequency selection for the highpass and low-pass filters will also change the center frequencies for each of the six faders. The most striking difference between W43 and WNS is the display. The white line on WNS shows how much noise suppression is occurring at any given moment. The lightblue line that borders the shaded blue area shows you the noise-reduction curve that’s been established by the frequency faders. The orange line is an instantaneous display of the detected noise profile. A quick way to set all the faders is to play the audio and click on the Suggest button that’s found to the right of the display. Clicking on that button at a point in between the dialogue phrases will produce an excellent starting point to dialing out the noise. In a typical music-mixing context, you probably aren’t going to have much use for either the W43 or WNS, but you never know when the next mix is going to include a noisy spoken-word intro that needs cleaning up.

Waves Tune The full version of Tune gives you all sorts of new features, although for some people the most important one is the ability to replicate the often-overused Auto-Tune effect. With Waves Tune, it’s pretty easy to get extreme pitch correction, as shown in Figure 8.13. You set the Note Tolerance to 100, the Speed to 4, the Note Transition to 0, the Ratio

Figure 8.13 Waves Tune.

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to 100%, and voilà—you now have the infamous Auto-Tune effect. To get all the proper notes to sound, you may have to select the proper key and limit the note choices. But please don’t do this unless it’s absolutely necessary. You’ll thank me later. If you’re using Waves Tune to correct rather than abuse, you’ll appreciate all of the new editing tools, such as the Line, Curve, Glue, Cut, Pencil, and Hand tools, which make life much easier when coaxing notes to the correct places. The new Vibrato section allows you to scale up or down the natural vibrato present in the vocal, or you can introduce synthetic vibrato if you like. The MIDI support is quite extensive as well. You can control the retuning note in real time via a MIDI input so you can in essence “play” the notes and have the vocal follow those notes. You do have to stick fairly close to the original pitch of the vocal, or it will sound strange and unnatural. MIDI control runs the other direction as well. If you’ve tracked the pitch of an entire vocal track, you can export a Standard MIDI File (SMF) and import that to a track so you can have a MIDI instrument play the same notes that the vocalist sang. With the MIDI options, there is a multitude of creative possibilities.

Dorrough Stereo The Dorrough meters have been longtime favorites among broadcast and audio postproduction engineers because of their accuracy and also because of the stringent level requirements that are imposed on audio for those applications. The Dorrough meters (see Figure 8.14) are beautiful, and their response is excellent. There are three different formats available—Arc, Horizontal, and Vertical—and you can display them in three different sizes. Don’t think these meters are just a pretty face, though; they include tools for counting sample overs, displaying L/R sum and difference (mid/side), and showing channel phase coherence.

Figure 8.14 Dorrough meter.

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The Mix Here we are, the last mix in the last chapter of the book. With all of the Waves plug-ins now at my disposal, certain options were opened up. The vintage equalizers, like the vintage compressors, each lend their own sound to the track by way of harmonic distortion and filter shape or behavior. As you will see, API gear is among my favorite for use on drums. These tools seem to add more punch to everything you pass through them. Because the Analog switch on the API equalizers provides harmonic distortion, I’ve left Analog engaged on all the API EQs. Because the Analog (or Mains, as the case may be) controls in all of the other vintage modeled plug-ins do nothing but add noise, I’ve left them turned off.

Table 8.1 Mercury Bundle Mix Track

Slot A

01-Kick-in

V-EQ3

02-Kick-out

API-550A

03-Snare-top

V-Comp

04-Snare-btm

API-560

05-Tom-hi

Slot B

Slot C

Slot D

Send1

Kramer PIE

C1 comp-gate

06-Tom-mid

RenEQ2

C1 comp-gate

07-Tom-low

RenEQ2

C1 comp-gate

08-Hi-hat

RenEQ4

09-Overhead

API-2500

10-Room

API-550B

Drum sub*

H-Comp

API-560

11-Loop1

API-550B

TransX Wide

12-Loop2

V-EQ4

MaxxVolume

13-Loop3

RenEQ4

L1

14-Bass

API-2500

Kramer HLS

15-Gtr-chords

PuigTec MEQ5

16-Gtr-picked

PuigChild 660

PuigTec EQP1A

17-Gtr-verse

MV2

GTR Amp

Kramer PIE

18-Gtr-solo

Z-Noise

GTR Stomp 2

Enigma (m->s)

21.5 dB 18.0 dB 25.5 dB CLA-2A

32.5 dB

Send2

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Table 8.1 Mercury Bundle Mix (Continued ) Track

Slot A

Slot B

19-Acoustic

CLA-3A

PuigTec EQP1A

20-Mel-synth

V-EQ3

21-Epiano

Kramer HLS

22-Synth-chorus

SuperTap2 (m->s) S1 Shuffler

PuigTec EQP1A

23-Lead-vox-verse

RChannel

CLA-2A

Vocal Rider

H-Delay (m->s)

CLA-76

Q4

Vocal Rider

24-Lead-vox-chorus Tune LT 27-Bvox-Mel-M1

Q2

28-Bvox-Mel-M2

Q2

29-Bvox-Har-F1

Q2

30-Bvox-Har-F2

Q2

Bvox sub*

RenDeEsser

33-Oooh-F1

Q2

34-Oooh-F2

Q2

Oooh sub*

RenChannel

Effects 1

IR-1 (m->s)

Effects 2

RenVerb (m->s)

Slot C

Slot D

Send1

Send2

29.0 dB 16.4 dB

CLA-76

MondoMod (m->s)

11.4 dB

RenChannel

*Drum sub is fed by Tracks 3 through 10, Bvox sub is fed by Tracks 25 through 32, and Oooh sub is fed by Tracks 33 through 36. The bolded plug-in names in the table above indicate plug-in or settings changes between the previous chapter and this one. The (m->s) at the end of the plug-in names denotes that a mono-to-stereo instance of the plug-in is to be used.

Drums The drum tracks were recipients of several of the new vintage modeled plug-ins. I got the drums to sound much punchier and more exciting, so they are now on par with the guitars. The first order of business was a makeover for the kick. Initially, I thought that LoAir might work on the inside kick track in place of the Renaissance EQ that I’d been using ever since the first mix. That turned out to be the wrong direction, so I instead looked to the outside kick track for the tone I was after.

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Figure 8.15 API-550A on the outside kick.

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Figure 8.16 Kramer PIE on the outside kick.

The outside kick track had been left alone until now. With the mix getting thicker, largely due to the guitars and vocals, I decided to work more on the sound of the kick beater to give the kick the right kind of impact. My first stop was the API-550A equalizer, where I dialed in 6 dB of boost at 3 kHz (see Figure 8.15). To give the attack some slap and consistency, I followed the API-550A with a Kramer PIE compressor (see Figure 8.16). The combination of the two kick tracks now sounded pretty good, but it still needed more bottom end. I added a 9-dB, 50-Hz boost to the outside kick track, which caused the PIE to give me a couple more dB of compression. The kick was just about there, but there was something still not quite right in the low end. I swapped the RenEQ on the inside kick track for a V-EQ3 and set up a similar low-frequency filter (see Figure 8.17). I was able to use less filter boost because the V-EQ gave me a little fuller tone by virtue of its Neve-like goodness. At this point I had a kick sound that was doing just what I wanted it to do.

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Figure 8.17 V-EQ3 on the inside kick track.

Figure 8.18 API-560 on the bottom snare track.

The next change I made was to the bottom snare track. So far, I’d just used Q1 to invert the polarity of the track, but now I turned to the trusty API plug-ins and replaced Q1 with an API-560 (see Figure 8.18). First, I hit the Polarity button to get the proper polarity, and then I did some low-mid and high-end contouring to give the snare more bite.

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The overhead track was another one that hadn’t gotten any processing until now. A little spank from the API 2500 tightened up the snare and made the overall drum tracks sound bigger, with a wider perceived stereo field (see Figure 8.19).

Figure 8.19 API 2500 compressor on the Overhead track.

I made another player substitution on the room mic track by swapping out the Renaissance EQ for an API-550B (see Figure 8.20). The RenEQ consisted of a single high-shelf boost at 5 kHz. I took some liberties with the API and did a high-shelf boost at 7 kHz, as well as bell-curve boosts at 3 kHz and 240 Hz and a whopping 9-dB boost at 30 Hz. There wasn’t a lot happening that low on those mics, but it did add a more visceral impact to the track. If you bypass this EQ and the 2500 compressor on the room mics, you can really hear the drums become much more subdued. To put the icing on the cake, I dropped in one more API equalizer on the drum submix track (see Figure 8.21). Because it was just icing and wouldn’t significantly change the dynamic character of the track, I went ahead and placed it after the Hybrid Compressor. I did just a little shaping on five of the bands. Because the boost and cut were minimal, the Q for the filters remained nice and wide. There’s just as much change in the sound coming from the harmonic distortion as there is from the filters. In the end, I have four different compressors acting on the drums, but the way I’m using them hasn’t squeezed the life out of them.

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Figure 8.20 API-550B on the Room track.

Figure 8.21 API-560 on the drum submix track.

Loops If swapping a RenEQ for an API EQ worked for the live drums, why not for one of the loops? Because this is a loop that I had shaped into a support track, the RenEQ I had set up dumped the low end and did some high midrange boosting. I created a similar sort of curve with the 550B, but I played around with the frequencies and boost amounts and came up with a sound that carved out its own space and benefited from the API mojo (see Figure 8.22).

Bass Over the course of the mix, I’d tried several different processing approaches to try to overcome the mediocre technique of the bass player. Each attempt met with a varying degree of success. I had the sneaking suspicion that some of the analog modeling plugins, such as the Kramer HLS (see Figure 8.23) and the API 2500 (see Figure 8.24), would be able to work some magic here.

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Figure 8.23 Kramer HLS on the bass track. Figure 8.22 API-550B on Loop1.

In the real world, you may not have an extra API 2500 sitting around that you can dedicate to a bass track, but fortunately in the world of DAWs, this isn’t an issue. Using some pretty heavy compression with the Thrust sideband EQ set to Loud really did a great job of smoothing out the dynamics while playing down the clicky finger noise. This caused the tone of the bass to point more toward growl than purr, so I followed the 2500 with a Kramer HLS. I gave it a little more bottom end at 250 Hz and a big helping of 1-kHz growl with the Mid band. The addition of a Kramer HLS gave the bass the presence it needed, and it finally found its place in the mix. It anchored the sound while still sounding clearly articulated.

Guitars The chorus guitars were still rocking from all the vintage goodness that I added in the previous chapter, so there wasn’t anything that I felt I needed to do to them—that is, other than properly clean up the noisy solo guitar track. Now that we have Z-Noise

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Figure 8.24 API 2500 on the bass track.

back in the Mercury bundle, I put that back in place of the C1 Gate that I had to revert to in the previous chapter. The guitar tracks all share the reverb I had set up on Send 1, which was initially set up using IR-L. Using the same impulse response, I swapped the IR-L for the more flexible IR-1 (see Figure 8.25). The biggest change I made was to introduce some pre-delay to both the early reflections and the reverb tail. Adding a small amount of space between

Figure 8.25 IR-1 on effect send 1.

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185

the dry signal and the reverb gave the guitars a little more definition. I also used the Gain Envelope to shorten the tail of the reverb. The first half of the reverb decay is unchanged, but you can see in Figure 8.25 that I’m causing it to fade out more quickly. The changes to the Density, Resonance, and Damping EQ also contribute to the reverb working better in the mix. The success I had using the PuigChild on the picked guitar track in the last chapter inspired me to try out a few vintage compressors on the verse guitar. I was not looking for overt compression, but just something to give the guitar a little vibe to get it to stand out a bit more. The Kramer PIE gave me exactly what I was looking for (see Figure 8.26). With only a few dB of compression and its modeled transformers, it gave the guitar a little more life. The combination of MV2, GTR Amp, and Kramer PIE provided excellent detail and presence.

Figure 8.26 Kramer PIE on the verse guitar.

Keyboards With the acoustic guitar track sounding more full, the melody synth track seemed to be falling into the background during the intro. It was also disappearing in the pre-chorus sections. To give it a little boost, I put V-EQ3 on it with a boost at 2.4 kHz and a

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high-pass filter at 300 Hz for good measure (see Figure 8.27). During the pre-chorus, this track blends with the background vocals so you don’t really hear it as a distinct part, but if you mute it, you certainly miss it.

Figure 8.27 V-EQ3 on the melody synth track.

Ever since Chapter 3, I’ve been using Q4 to take out some of the low end in the electric piano track. I changed that out for a Kramer HLS that I set up with a 6-dB cut at 50 Hz and a hefty boost at 700 Hz to give it a thicker tone (see Figure 8.28).

Figure 8.28 Kramer HLS on the electric piano track.

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I liked what the EQ was doing for the tone, but it also caused some of the chords to become significantly louder than others. What better time to break out another vintage compressor? I dropped a CLA-76 on the track and used the slowest attack it had to preserve as much of the initial transient as possible (see Figure 8.29). I set the release fast enough to smoothly follow the decay but not fast enough for the release to “chatter” and draw attention to the compression.

Figure 8.29 CLA-76 on the electric piano track.

The inline SuperTap followed by the S1 Shuffler was still giving the chorus synth track its own space out on the edges, but I thought it could use a little tonal shaping to fit into the mix better. A side effect of the extreme S1 settings was that the low end thinned out a little, but the track still had too much bottom and was a little dull. I followed S1 with a PuigTec EQP-1A with a substantial 100-Hz cut, a big boost at 4 kHz, and an output gain boost to make up for the overall loss of level due to the 100-Hz cut (see Figure 8.30). This gave the track some sizzle out on the edges that didn’t take away from the cymbals or the top end of the guitars.

Figure 8.30 PuigTec EQP-1A on the chorus synth track.

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Vocals Tune LT was doing everything I needed, so I didn’t see any reason to change that out for the full version of Tune. If you don’t have Tune LT installed, you can load up the same preset in Tune with identical results. When soloing the chorus lead vocal, I noticed (courtesy of PAZ) that there was substantial low-frequency buildup below 45 Hz, which was no doubt due to the freaky compression caused by the “all-button” mode on the CLA-76. I moved Q4 after CLA76 so the high-pass filtering in Q4 would kill the compression-induced rumble. I had originally inserted the high-pass filter prior to RenComp so that the low frequencies wouldn’t influence the action of the compressor. The lack of low-frequency filtering on the track didn’t seem to phase the CLA-76 in all-button mode—it hardly even noticed that they were there, so the plug-in order change was not an issue.

Wrap Up It was an interesting process for me to mix a song a little bit at a time over the course of a couple of months. Normally, I like to work quickly and not overthink the mix. To keep things fresh, I would write the first half of each chapter and then jump in and do the mix. I was able to keep some perspective, and I didn’t even end up getting sick of the song. The track isn’t finished, but I feel as if we ended up with a solid mix, which is always the goal I shoot for before delving into automation. There are a few rides that I would do with automation in the next mixing step—specifically, I think the background vocals could used some balancing between voices in each section to make them blend a little more. I would also probably bring down the drum overhead and room mic tracks 2 to 3 dB during the verses to tighten up the drums and create a little more dynamic contrast. Another thing I’d consider doing would be to remove the acoustic guitar entirely from the pre-choruses. The way that those sections ended up, you really can’t hear the acoustic guitar at all. Over the course of writing this book, I learned a few things about some of the Waves plug-ins, even though I’ve been using their products for 15 years and have been involved in beta testing them for nearly 10 years. There are features and nuances that I’d missed or rediscovered in the process of writing this book. I hope I was able to effectively explain the tools and pass on some solid technical info as well as some tricks of the trade. Thanks for reading.

Index

? command, 7

Musicians 2, 2–5 Platinum, 2–5 plug-ins by, 5 Silver, 2–5 See also plug-ins buttons ?, 7 A->B, 7 Load, 7 Preset Step, 7 Redo, 7 Save, 7 Setup A/B, 7 Undo, 7

A A->B command, 7 acoustic guitars Renaissance Axx on, 33–34 Renaissance EQ on, 34 See also guitars acoustics, relationship to RT60 value, 50 AlphaTrack fader, using, 6 AM (amplitude modulation), 47–48 Amp, GTR, 146–147, 153 amplitude modulation (AM), 47–48 API 550A/API 550B plug-in, 161–163, 179 API 560 plug-in, in Mercury bundle, 163–164, 180 API 2500 plug-in, 168–170, 181 using with bass, 182–183 using with drums, 181 API EQ, using with loops, 182–183 API-550A equalizer, using with drums, 179 API-550B, 181 API-560 plug-in, 180 audio controlling volume of, 16 displaying information about, 54–55 audio restoration, 127–130. See also noise reduction AudioTrack plug-in, 81–82 aux tracks, using effects on, 46

C C1 comp-gate, using with Gold bundle, 87–88 C1 Gate, using with Silver bundle, 61 C1 Parametric Compander, 42–43 C4 Multiband Processor, 71–72 C4 plug-in, 89 C6 Multiband Processor plug-in, 170–171 Center plug-in, 123–124 channel strip, AudioTrack as, 81–82 chorus effects, 73–74 chorus synth, livening up, 35. See also vocals Citron cabinet, using with guitars, 153–154 CLA-2A dynamics plug-in, 138–140, 154–156 CLA-3A dynamics plug-in, 140, 155 CLA-76 Bluey/Blacky dynamics plug-in, 140–142, 158, 187 using with keyboards, 187 commands ?, 7 A->B, 7 Load, 7 Preset Step, 7 Redo, 7 Save, 7 Setup A/B, 7 Undo, 7 comping, defined, 8 compression, upward, 72 compressors AudioTrack, 81–82 Fairchild 670, 137 features of, 19 in Musicians 2 bundle, 19

B bandwidth setting, using with filters, 16 bass in Gold bundle, 87–88 in Mercury bundle, 182–183 in Musicians 2 bundle, 30–31 in Platinum bundle, 110–111 in Silver bundle, 58–59 bell filter, using with equalizers, 15, 17 boost and cut setting, using with filters, 16 breath sounds, attenuating, 149–150 brick-wall limiters, 18–19 bundles Diamond, 2–5 dynamics plug-ins, 3–4 effects plug-ins, 4 equalizer plug-ins, 2 Gold, 2–5 Horizon, 2–5 Mercury bundle, 2–5

189

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compressors (Continued ) using with Gold bundle, 86 See also V-Comp compressor controls, changing values of, 6 convolution, use with reverberation, 50–51 Course PTR website, 13 crackle, defined, 129

D DAW (digital audio workstation) software, 6. See also Digital Performer DAW DeBreath dynamics plug-in, 149–150 DeEsser dynamics plug-in, 44–45. See also Renaissance DeEsser plug-in de-essing, 41 delay compensation, 103 delay time, specifying with Hybrid Delay, 74 Diamond bundle Center plug-in, 123–124 dynamics plug-ins, 3–4, 119–122 effects plug-ins, 4, 122–123 equalizer plug-ins, 2 guitars in, 134–135 L3 Multimaximizer plug-in, 121–122 loops in, 133 mix in, 132–133 Morphoder plug-in, 124–125 plug-ins, 5 Q-Clone plug-in, 125–126 SoundShifter plug-in, 122–123 Trans-X plug-in, 119–121, 133 X-Hum plug-in, 134 X-Noise plug-in, 127–130, 134 Z-Noise plug-in, 130–131, 134 digital audio workstation (DAW) software, 6 Digital Performer DAW Doubler 2 versions, 23 master fader, 12 pan values, 11–12 distortion occurrence with guitars, 32 related to master fader, 12 documentation, accessing, 7 Dolby Cat 43, 173 Doppler plug-in, 81 Dorrough Stereo plug-in, in Mercury bundle, 176 Doubler 2 using with keyboards, 34–35 versions in Digital Performer, 23 Doubler effects plug-in, 24–25, 34 downloading files, 12 drum tracks, panning in Digital Performer, 11–12 drums in Gold bundle, 84–87 in Mercury bundle, 178–182 in Musicians 2 bundle, 27–30 in Platinum bundle, 109 in Silver bundle, 57 ducker, using, 42

dynamics plug-ins API 2500, 168–170 by bundle, 3 C1 Parametric Compander, 42–43 C4 Multiband Processor, 71–72 C6 Multiband Processor, 170–171 CLA-2A, 138–140 CLA-3A, 140 CLA-76 Bluey/Blacky, 140–142 DeEsser, 44–45 in Diamond bundle, 119–122 Enigma, 48–49 in Gold bundle, 68–73 in Horizon bundle, 137–145 Hybrid Compressor, 70–71 IR-L Convolution Reverb, 50–51 Kramer PIE Compressor, 167–168 L1 Ultramaximizer, 43–44 L2 Ultramaximizer, 102–103 L3 Multimaximizer, 103–104, 121–122 L3-16 Multimaximizer, 142 Linear Phase Multiband (LinMB), 100–102 MaxxVolume, 105 in Mercury bundle, 167–171 MondoMod, 47–48, 62–63 in Musicians 2 bundle, 3–4, 18–22 MV2 Dynamics Processor, 72–73 in Platinum bundle, 100–106 PuigChild 660/670, 137–138 Renaissance DeEsser, 105–106 in Silver bundle, 41–45 Trans-X, 119–121 TrueVerb, 49–50 V-Comp, 69–70 Vocal Rider, 143–145 dynamics processors compressors, 19 expanders, 19–20 gates, 19 limiters, 18–19 Renaissance Axx, 21 Renaissance Compressor, 20–21 Renaissance Vox, 22

E editing, 8 effects plug-ins by bundle, 4 in Diamond bundle, 122–123 Doubler, 24–25 Enigma, 48–49 in Gold bundle, 73–78 GTR Amp, 146–147 GTR Stomp, 146 GTR Tool Rack, 147–148 GTR Tuner, 145 in Horizon bundle, 145–148 Hybrid Delay, 74–76 IR-1 Convolution Reverb, 172–173

Index

IR-L Convolution Reverb, 50–51 in Mercury bundle, 171–173 MetaFlanger, 73–74 MondoMod, 47–48 mono-to-mono configuration, 23 mono-to-stereo configuration, 23 in Musicians 2 bundle, 4, 22–25 phase and polarity, 46–47 Renaissance Reverb, 77–78 in Silver bundle, 45–51 SoundShifter, 122–123 stereo-to-stereo configuration, 23 SuperTap, 23–24, 63 TrueVerb, 49–50 UltraPitch, 76–77 electric guitars, PuigTec equalizers on, 111. See also guitars electric piano, using Doubler on, 34 Enigma dynamics plug-in, 48–49 EQ curve, controlling sharpness of, 16 equalization doppelganger, Q-Clone, 125–126 equalizer plug-ins API 550A/API 550B, 161–163 API 560 plug-in, 163–164 AudioTrack, 81–82 by bundle, 2 explained, 15 in Gold bundle, 67–68 Kramer HLS Channel plug-in, 164–166 Linear Phase EQ, 97–99 LoAir plug-in, 166–167 MaxxBass, 40–41 in Mercury bundle, 161–167 in Musicians 2 bundle, 2, 15–17 in Platinum bundle, 95–100 PuigTec EQP-1A, 96–97 PuigTec ME Q-5, 95–96 Q10 Paragraphic Equalizer, 39–40 Renaissance Bass, 99–100 Renaissance EQ, 17–18 V-EQ3/V-EQ4, 67–68 equalizers filter types in, 15 using to boost frequencies, 15 expanders DeBreath plug-in, 149–150 features of, 19–20 in Musicians 2 bundle, 19–20 using with Gold bundle, 87

F Fairchild 670 compressor, 137 filters bandwidth setting, 16 bell-curve, 17 boost and cut setting, 16 frequency setting, 16 high-pass, 15–16 low-pass, 15–16

191

Q setting, 16–18 shelf, 15–17 using with equalizers, 15 FIR (Finite Impulse Response) system, 98, 103 flanging effects, 73–74 FM (frequency modulation), 47–48 frequencies, boosting with equalizers, 15 frequency setting, using with filters, 16

G gates features of, 19 in Musicians 2 bundle, 19 using with guitars, 32–33 Gold bundle AudioTrack, 81–82 bass tracks, 87–88 C1 comp-gate, 87–88 C4 Multiband Processor, 71–72 C4 plug-in, 89 Doppler, 81 drums, 84–87 dynamics plug-ins, 3–4, 68–73 effects plug-ins, 4, 73–78 equalizer plug-ins, 2, 67–68 guitars, 88–90 Hybrid Compressor, 70–71, 85–87 Hybrid Delay effects plug-in, 74–76 keyboards, 90 loops, 87 MetaFlanger effects plug-in, 73–74, 90 mix, 83–84 MV2 Dynamics Processor, 72–73, 88–89 PAZ PsychoAcoustic Analyzer, 87 plug-ins, 5 Renaissance Reverb effects plug-in, 77–78, 92 TrueVerb dynamics plug-in, 92 UltraPitch effects plug-in, 76–77 V-Comp compressor, 84–85 V-Comp dynamics plug-in, 69–70, 84–85, 91–92 V-EQ3/V-EQ4 equalizers, 67–68 V-EQ4 equalizer, 87–88 vocals, 90–93 Waves Tune LT, 78–81 ground-loop noise, removing, 129–130 GTR Amp effects plug-in, 146–147, 153 GTR plug-ins, using with Horizon bundle, 154 GTR Stomp effects plug-in, 146, 154 GTR Tool Rack effects plug-in, 147–148 GTR Tuner effects plug-in, 145 guitar tracks, panning in Digital Performer, 11–12 guitars in Diamond bundle, 134–135 in Gold bundle, 88–90 in Horizon bundle, 152–156 in Mercury bundle, 183–185 in Musicians 2 bundle, 31–34 in Platinum bundle, 111–112 in Silver bundle, 59–62

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guitars (Continued ) using high-pass filters with, 32 using Renaissance Axx with, 33–34 using Renaissance Vox with, 32 See also acoustic guitar; electric guitars

H

Kramer HLS plug-in, 182–183, 186. See also HLS filters Kramer PIE Compressor plug-in, 167–168, 179, 185

L

hardware control surfaces, using, 6 high-pass filters using with equalizers, 15–16 using with guitars, 32 hi-hat mic, using with drum mix, 28 HLS filters, 164–165. See also Kramer HLS plug-in Horizon bundle CLA-3A dynamics plug-in, 155 CLA-76 Bluey/Blacky dynamics plug-in, 158 DeBreath plug-in, 149–150 dynamics plug-ins, 3–4, 137–145 effects plug-ins, 4, 145–148 equalizer plug-ins, 2 GTR plug-ins, 154 guitars, 152–156 keyboards, 156 mix, 151–152 plug-ins, 5 PuigChild 660 plug-in, 152 PuigTec EQP-1A, 152 PuigTec EQP-1A plug-in, 155 Renaissance Channel plug-in, 156–157 Vocal Rider dynamics plug-in, 157–158 vocals, 156–159 Hybrid Compressor, 70–71, 85–87 Hybrid Delay effects plug-in, 74–76

L1 Ultramaximizer, 43–44 L2 Ultramaximizer plug-in, 102–103 L3 Multimaximizer plug-in, 121–122 L3 Ultramaximizer plug-in, 103–104 L3-16 Multimaximizer dynamics plug-in, 142 LA-3A Leveling Amplifier, successor to, 140 latency, defined, 103 LFO (low-frequency oscillator), 24 limiter dynamics processor, 18–19 limiters L1 Ultramaximizer, 43–44 in Musicians 2 bundle, 18–19 See also peak limiter Linear Phase EQ plug-in, 97–99 Linear Phase Multiband (LinMB) plug-in, 100–102 Load command, 7 LoAir plug-in, 166–167. See also MaxxBass equalizer loops in Diamond bundle, 133 in Gold bundle, 87 in Mercury bundle, 182 in Musicians 2 bundle, 29–30 in Platinum bundle, 109–110 in Silver bundle, 57–58 Lord-Alge, Chris, 137–139 low-frequency oscillator (LFO), 24 low-pass filter, using with equalizers, 15–16

I

M

“icing” track, 35 IIR (Infinite Impulse Response) system, 97–98 inline effects, using, 45–46 IR-1 Convolution Reverb, 172–173, 184 IR-L Convolution Reverb, 50–51

master fader level of final output, 12 turning down, 12 using in Digital Performer, 12 mastering, 9 masters, making loud, 142 MaxxBass equalizer, 40–41, 58–59. See also LoAir plug-in MaxxVolume plug-in, 105, 109 Mercury bundle API 550A/API 550B plug-in, 161–163, 179 API 560 plug-in, 163–164 API 2500, 168–170 API 2500 plug-in, 181 API-550B, 181 API-560 plug-in, 180 bass, 182–183 C6 Multiband Processor, 170–171 CLA-76 Bluey/Blacky dynamics plug-in, 187 Dorrough Stereo plug-in, 176 drums, 178–182 dynamics plug-ins, 3–4, 167–171 effects plug-ins, 4, 171–173

K keyboards in Gold bundle, 90 in Horizon bundle, 156 in Mercury bundle, 185–187 in Musicians 2 bundle, 34–35 in Platinum bundle, 112 in Silver bundle, 62–64 using Doubler 2 with, 34–35 using SuperTap with, 35 kick mics, using with drum mix, 27 knobs, changing values of, 6 Kramer HLS Channel plug-in in Mercury bundle, 164–166 using with bass, 182–183 using with electric piano, 186

Index

equalizer plug-ins, 2, 161–167 guitars, 183–185 keyboards, 185–187 Kramer HLS Channel plug-in, 164–166 Kramer HLS plug-in, 182–183, 186 Kramer PIE Compressor, 167–168, 185 LoAir plug-in, 166–167 loops, 182 mix, 177–178 plug-ins, 5 PuigTec EQP-1A plug-in, 187 V-EQ3 equalizer, 179–180 vocals, 188 W43 Noise Suppressor, 173 Waves Tune plug-in, 175–176 WNS Noise Suppressor, 173–175 MetaFlanger effects plug-in, 73–74, 90 MIDI Learn option, using with Hybrid Delay, 75 mid/side miking, 123–124 mid/side processing, 52 miking, mid/side, 123–124 mix automation, 9 mix levels and processing, 8–9 mixes bass in Musicians 2 bundle, 30–31 bass in Silver bundle, 58–59 in Diamond bundle, 132–133 downloading, 13 drum loops in Musicians 2 bundle, 29–30 drums in Musicians 2 bundle, 27–29 drums in Silver bundle, 57 in Gold bundle, 83–84 guitars in Musicians 2 bundle, 31–34 guitars in Silver bundle, 59–62 in Horizon bundle, 151–152 keyboards in Musicians 2 bundle, 34–35 keyboards in Silver bundle, 62–64 loops in Silver bundle, 57–58 in Mercury bundle, 177–178 Musicians 2 bundle, 26 in Platinum bundle, 108–109 Silver bundle, 56 subgroups, 86 vocals in Silver bundle, 64–66 mixing automation, 9 editing, 8 levels and processing, 8 versus mastering, 9 mastering, 9 subgroups tool, 86–87 tracking, 7–8 modulation, defined, 24 MondoMod dynamics plug-in, 47–48, 62–63 Morphoder plug-in, 124–125 mouse, using, 6 Musicians 2 bundle bass tracks, 30–31 compressors, 19 Doubler effects plug-in, 24–25

193

drums, 27–30 dynamics plug-ins, 3–4, 18–22 effects plug-ins, 4, 22–25 equalizer plug-ins, 2, 15–17 expanders, 19–20 gates, 19 guitar chords track, 31–34 keyboards, 34–35 limiters, 18–19 loops, 29–30 mix, 26–27 plug-ins, 5 Renaissance Axx, 21 Renaissance Compressor, 20–21, 30 Renaissance EQ, 17–18, 27 Renaissance Vox, 22 Slots A and B, 26 SuperTap effects plug-in, 23–24 vocals, 35–37 MV2 Dynamics Processor, 72–73, 88–89

N Neve equalizers, 67–68 noise reduction W43 Noise Suppressor, 173–174 WNS Noise Suppressor, 173–175 See also audio restoration; X-Noise plug-in; Z-Noise plug-in

P panning “Running Aground” sample song, 11–12 parametric EQ, 17 PAZ PsychoAcoustic Analyzer, 54–55, 64, 87 peak limiter, L3-16 Multimaximizer, 142. See also limiters pedalboard, GTR Stomp, 146, 154 phase and polarity, 46–47 phasing effects, 73–74 piano, electric, 34 piracy, impact of, 1 pitch, controlling with UltraPitch, 76–77 Platinum bundle bass in Silver bundle, 110–111 drums in, 109 dynamics plug-ins, 3–4, 100–106 effects plug-ins, 4 equalizer plug-ins, 2, 95–100 guitars in, 111–112 keyboards in, 112 L2 Ultramaximizer plug-in, 102–103 L3 Ultramaximizer plug-in, 103–104 Linear Phase EQ plug-in, 97–99 Linear Phase Multiband plug-in, 100–102 loops in, 109–110 MaxxVolume plug-in, 105, 109 mix, 108–109 plug-ins, 5 PuigTec EQP-1A plug-in, 96–97 PuigTec equalizers, 111–112

194

W a ve s Pl u g- In s Wo r k s h op : M i xi n g by t he B u nd l e

Platinum bundle (Continued ) PuigTec MEQ-5 plug-in, 95–96 Renaissance Bass equalization plug-in, 99–100, 110–111 Renaissance Channel plug-in, 106–107, 116–117 Renaissance DeEsser plug-in, 105–106, 113, 115–116 Renaissance Vox plug-in, 115 vocals in, 113–117 plug-in processor, using, 6 plug-in settings, versions of, 7 plug-in values, changing, 6 plug-ins by bundles, 5 configurations of, 23 dynamics by bundle, 3–4 effects by bundle, 4 equalizers by bundle, 2 features of, 6 See also bundles polarity of signals, 47, 57 preset files, accessing, 13 Preset Step command, 7 presets downloading, 13 saving, 7 selecting and loading, 7 project sample. See mixes Puig, Jack Joseph, 137 PuigChild 660/670 dynamics plug-in, 137–138, 152, 185 PuigTec EQP-1A plug-in, 96–97, 152, 155, 187 PuigTec equalizers, using with guitars, 111–112 PuigTec MEQ-5 plug-in, 95–96

Q Q setting, using with filters, 16–18 Q4 high-pass filtering, 65 Q10 Paragraphic Equalizer, 39–40 Q-Clone plug-in, 125–126 Q-series plug-ins, flipping polarity in, 57

R Redo command, 7 Renaissance Axx features of, 21 using on bass tracks, 30–31 using with guitars, 33–34 Renaissance Bass equalization plug-in, 99–100, 110–111 Renaissance Channel plug-in, 116–117 using with Horizon bundle, 156–157 using with Platinum bundle, 106–107 Renaissance Compressor, 20–21 Smooth setting, 21, 30 using with Silver bundle, 58, 64 using with vocals, 36 Warm setting, 30

Renaissance DeEsser plug-in, 105–106, 113, 115–116. See also DeEsser dynamics plug-in Renaissance EQ equalizer, 17–18, 27, 31 Renaissance Reverb effects plug-in, 77–78, 92 Renaissance Vox plug-in, 21, 115 enhancing vocals with, 35–37 gates, 32–33 using with guitars, 32 using with vocals, 35 RenComp, using with Silver bundle, 58–59, 64–65 reverb effects, 46 IR-L Convolution Reverb, 50–51 Renaissance Reverb effects plug-in, 77–78 TrueVerb dynamics plug-in, 49–50 RT60 value, relationship to acoustics, 50 rumble, eliminating in tracks, 16 “Running Aground” sample song downloading, 13 master fader, 12 mixes, 10 panning, 11–12 performance credits, 10 preset files, 13 tracks, 10–11 See also songs

S S1 MS Matrix plug-in, 53–54 S1 Shuffler, 53, 63 S1 Stereo Imager, 52–54 sample project. See mixes Save command, 7 send effects, using, 46 Setup A/B command, 7 shelf filter, using with equalizers, 15–17 sibilance, reducing, 44–45 sidechaining, explained, 41–42 signals amplitude of, 18 changing pitch of, 122–123 clipping, 12 modifying with sidechains, 41 polarity of, 47 shifting phases of, 46 Silver bundle bass tracks, 58–59 C1 gate, 61 drums, 57 dynamics plug-ins, 3–4, 41–45 effects plug-ins, 4, 45–51 Enigma plug-in, 48–49 equalizer plug-ins, 2, 39–41 guitars, 59–62 IR-L Convolution Reverb plug-in, 50–51

Index

keyboards, 62–64 loops, 57–58 MaxxBass equalizer, 40–41 mix, 56–57 MondoMod plug-in, 47–48, 62 PAZ PsychoAcoustic Analyzer, 54–55, 64 plug-ins, 5 Q10 Paragraphic Equalizer, 39–40 Q-series plug-ins, 57 Renaissance Compressor, 64 S1 Stereo Imager, 52–54 TrueVerb plug-in, 49–50 vocals, 64–66 sine waves, looking at, 46–47 sliders, changing values of, 6 songs editing, 8 mastering, 9 mix automation, 9 mix levels and processing, 8 tracking, 7–8 See also “Running Aground” sample song sound. See audio SoundShifter plug-in, 122–123 stereo imaging, 52–54 stereo recording, mid/side technique of, 52 subgroups, using with mixes, 86–87 SuperTap effects plug-in, 23–24, 35, 63

T tracking, 7–8 tracks downloading, 13 in sample project, 10–11 transient control, use of, 119 Trans-X plug-in, 119–121, 133 tremolo and vibrato, generating, 47 TrueVerb dynamics plug-in, 49–50, 92 Tune. See Waves Tune LT plug-in Tune LT, using with vocals, 188 tuners, GTR Tuner effects plug-in, 145

U UltraPitch effects plug-in, 76–77 Undo command, 7 UREI 1176, revisions of, 140–141 user-interface features, 6

195

V V-Comp compressor, using in Gold bundle, 84–85. See also compressors V-Comp dynamics plug-in, 69–70, 84–85, 91–92 V-EQ3 equalizer, 67–68, 179–180 using with drums, 179–180 using with keyboards, 185–186 V-EQ4 equalizer, 67–68, 87–88 vibrato and tremulo, generating, 47 Vocal Rider dynamics plug-in, 143–145, 157–159 vocals in Gold bundle, 90–93 in Horizon bundle, 156–159 in Mercury bundle, 188 in Musicians 2 bundle, 35–37 in Platinum bundle, 113–117 reducing sibilance in, 44–45 in Silver bundle, 64–66 using Renaissance Compressor with, 36 using Renaissance Vox with, 35 using Tune LT with, 188 See also chorus synth vocoder effect, 124–125 volume, controlling, 16

W W43 Noise Suppressor, in Mercury bundle, 173 waveforms, cycles on, 46 Waves plug-ins. See bundles; plug-ins Waves toolbar, 6–7 Waves Tune LT plug-in, 78–81, 91 Waves Tune plug-in, in Mercury bundle, 175–176 WNS Noise Suppressor, in Mercury bundle, 173–175

X X-Click plug-in, 128–129 X-Crackle plug-in, 129 X-Hum plug-in, 129, 134 X-Noise plug-in, 127–131, 134. See also noise reduction

Z Z-Noise plug-in, 130–131, 134–135. See also noise reduction