Home Studio Mastering

Home Studio Mastering is a step-by-step manual that gives you all the tools to professionally master your music yourself.

It demystifies the subject in a hands-on way for those working in a home studio and provides comprehensive guidance, from buying equipment and applying acoustic treatment, to using different audio applications and mastering plug-ins.

The book is accompanied by five mastering plug-ins (VST/

AU/AAX for Mac and PC), to facilitate your personal mastering sessions from start to finish.

J.D. Young began producing music in his home studio at the age of fifteen, and later on started producing music professionally. From the beginning of his career onward, he always felt drawn to the mysterious art of mastering. He researched the topic extensively and over the years he managed to optimize his mastering workflow, overcoming the challenges of working in a home studio. He wrote everything down in his book Home Studio Mastering, eager to share his experiences with other home studio producers. Besides writing his book, he also professionally develops audio plug-ins for different companies around the world, and is the founder and lead-developer of his own small software company that creates plug-ins for mastering.

Home Studio

Mastering

J.D. Young

First published 2018

by Routledge

711 Third Avenue, New York, NY 10017

and by Routledge

2 Park Square, Milton Park, Abingdon, Oxon, OX14 4RN

Routledge is an imprint of the Taylor & Francis Group, an informa business

© 2018 Taylor & Francis

The right of J.D. Young to be identified as author of this work has been asserted by him in accordance with sections 77 and 78 of the Copyright, Designs and Patents Act 1988.

All rights reserved. No part of this book may be reprinted or reproduced or utilised in any form or by any electronic, mechanical, or other means, now known or hereafter invented, including photocopying and recording, or in any information storage or retrieval system, without permission in writing from the publishers.

Trademark notice: Product or corporate names may be trademarks or registered trademarks, and are used only for identification and explanation without intent to infringe.

Library of Congress Cataloging-in-Publication Data Names: Young, J. D., 1982– author.

Title: Home studio mastering / J.D. Young.

Description: New York, NY : Routledge, 2018. | Includes index.

Identifiers: LCCN 2017059725 | ISBN 9781138747074 (hardback : alk. paper) |

ISBN 9781138747081 (pbk. : alk. paper) | ISBN 9781315180328 (ebook) Subjects: LCSH: Mastering (Sound recordings) | Computer sound processing. |

Sound—Recording and reproducing—Digital techniques. | Popular music—Production and direction.

Classification: LCC MT723.Y64 2018 | DDC 781.49—dc23

LC record available at https://lccn.loc.gov/2017059725

ISBN: 978-1-138-74707-4 (hbk)

ISBN: 978-1-138-74708-1 (pbk)

ISBN: 978-1-315-18032-8 (ebk)

Typeset in Times New Roman

by Apex CoVantage, LLC

Visit the companion website: www.homestudiomastering.com

Contents

Preface

ix

Acknowledgments

xi

Part I

Setting Up Your Home Studio

1 Mastering

3

1.1 Can you Master Your Music in a Home Studio?

3

1.2 What is Mastering?

5

1.3 A Bit of History

8

1.4 Managing Expectations

11

2 Your Own Mastering Studio

15

2.1 Make Smart Choices

15

2.2 Room Characteristics

16

2.3 Arrangement

19

2.4 Don’t Panic Please

21

3 Room Acoustics

23

3.1 Common Acoustic Problems

23

3.2 Acoustic Treatment

24

3.3 Standing Waves

27

3.4 Comb Filtering

30

3.5 Flutter Echo

32

3.6 Reverberance

34

4 Hardware

37

4.1 A Basic Digital Setup

37

4.2 Computers

38

4.3 Audio Interfaces

40

4.4 Monitor Sets

42

4.5 Headphones

44

5 Software

47

5.1 Audio Editor or Sequencer?

47

5.2 Mastering Plug-Ins

49

5.3 Restoration

49

5.4 Tonal Balance

51

5.5 Width and Depth

54

5.6 Dynamics

56

5.7 Loudness

58

v

vi

Contents

6 Mastering Workflow

61

6.1 Evaluating Your Setup

61

6.2 Choosing Reference Tracks

64

6.3 The Art of Listening

65

6.4 General Workflow

67

Part II

Home Studio Mastering Techniques

7 Preparing Your Session

73

7.1 Finalizing Your Mix

73

7.2 Choosing Samplerate and Bitdepth

75

7.3 Setting Output Volume and Headroom

78

7.4 Checking Mono Compatibility

80

7.5 Checking Render Section and Fades

81

7.6 Setting up Your Mastering Project

82

8 Restoration

85

8.1 Removing DC Offset and Unwanted Noises

85

8.2 DC Offset

86

8.3 Ground Hum

88

8.4 Noise and Hiss

88

8.5 Clicks and Crackles

90

8.6 Sibilance and Plosives

92

9 Tonal Balance

97

9.1 Frequency Distribution

97

9.2 Frequency Analysis

100

9.3 Mastering Equalization Basics

102

9.4 Optimizing Tonal Balance

105

9.5 Low End Control

106

9.6 Corrections and Improvements

108

9.7 Match Your Tracks

109

9.8 Enhance Overall Timbre

111

10 Width and Depth

115

10.1 Stereo Image and Reverberation

115

10.2 Stereo Imaging Techniques

117

10.3 Narrowing the Low End

118

10.4 Overall Mid-Side Stereo Enhancement

120

10.5 Reverberation Techniques

121

10.6 Algorithmic Reverb

123

10.7 Convolution Reverb

126

11 Dynamics

129

11.1 Dynamic Range and Crest Factor

129

11.2 Mastering Compression Basics

131

Contents

vii

11.3 Optimizing Dynamics

135

11.4 Compression for Punch and Impact

136

11.5 Compression for Balance and Glue

138

11.6 Multi-Band Compression and Dynamic Equalization

141

12 Loudness

143

12.1 Track Order and Individual Volume

143

12.2 Maximizing Loudness

146

12.3 Loudness by Equalization

148

12.4 Loudness by Compression

150

12.5 Loudness by Brickwall Limiting

152

13 The Final Master

157

13.1 Housekeeping

157

13.2 Rendering Your Tracks

158

13.3 Creating a CD-Montage

160

13.4 Creating a Master-CD or DDP-File

163

Part III

One Hour Mastering Workshop

14 One Hour Mastering Workshop

167

Download the Mastering Bundle

167

The Plug-Ins and 5-Step Workflow

168

Prepare Your Mastering Session

170

Step 1: Monomizer

171

Step 2: Equalizer

175

Step 3: Compressor

180

Step 4: Convolver

184

Step 5: Limiter

187

In Conclusion

190

Appendix A—Mastering Dictionary

193

Index

209

Preface

Mastering is the final step in the production process of a song or album, where you make sure it is ready to be presented to the world. Traditionally, this is the job of dedicated mastering engineers in specialized studios, but the audio industry is changing.

Since about a decade or so ago, it has been possible to set up a DAW-based home studio on a relatively modest budget, which is why increasing numbers of musicians now produce their music at home. For most home studio owners mastering has remained a challenge, but over the past few years the quality of digital audio plug-ins has reached a level that is sufficient for professional mastering. This gives musicians a fighting chance in an extremely competitive industry, as well as a way to be completely self-reliant producing and releasing music.

Home Studio Mastering uncovers the secrets of the contemporary mastering engineer. It is a simple and practical guide for musicians and producers working in a home studio who are ready to raise their productions to a professional level. The book is divided into three parts. The first part of the book describes the criteria a mastering studio must meet and provides ways to set up your home studio accordingly. The choice of equipment is discussed, along with the basic principles of room acoustics and acoustic treatment. The second part of the book provides an analysis of all possible mastering techniques for optimizing the tonal balance, width and depth, dynamics, and loudness of your music. The third part of the book is a mastering workshop that guides you through the whole mastering process step-by-step in about one hour, using the bundle of five mastering plug-ins that was specifically developed to accompany the book. In addition, the book contains an appendix that provides you with a concise definition list of the ix

x

Preface

most commonly used mastering related terms, to help you navi-gate the subject more easily.

The chances are you will read this book with the goal of preparing a specific song or album for release. However, mastering can be much more than a means to an end. Understanding the role mastering plays in the production process, and gaining experience with different tools and techniques used for mastering, can really put the whole audio production process in perspective. By learning the art of mastering, you will develop unique listening skills that are extremely beneficial for anyone creating music, whether you are a musician, a songwriter or an audio engineer. In addition, if you have been producing music for many years, trying out different song structures and mixing techniques, you will find that listening with the ear of a mastering engineer will help you regain the ability to experience a song as a whole. In that regard, mastering can take you back to the essence of the listening experience.

Acknowledgments

I would like to thank Mart van Hamersveld, Bas Bonthuis, Yuri Guerin and Ewan Cameron, for offering their continuing technical advice and feedback while writing Home Studio Mastering, as well as Erik de Jong and Acle Kahney for their generous support and for believing in the book from the start.

In addition, I would like to thank Dennis Kruithof, Nicolette Cameron, Marta Klement, Pepijn Oldenburg, Rob Stegeman, Mandy Boere, Dounia Bouzoubaa and Dick Provoost for all their help over the years, as well as Sil de Graaf for assisting me with preparing the final manuscript and my friends and colleagues Acle Kahney, Ahmed Samadie, Anthony Jones, Bas Bonthuis, Daniel Kvak, Eric Coen Versteeg, Ewan Cameron, Gilles Ketting, Jeroen Dieks (Collision Course Music), Kevin Stunnenberg, Lodewijk and Olivier Schutte, Marc Baronner, Marco Riebeek, Mart van Hamersveld, Poul Sven de Haan (Faboem), Roman Trachta, Thomas Beer and Zack Hartmann, for sharing photos of their home studios.

Finally, I would like to thank Rob Meere, Gertjan Koning, Pieter Offringa and Oscar Meijers for teaching me the basics of audio production, when I first started out producing music many years ago, as well as Nathan Hayman, Theo Niessink, Stefan Mat-thijssen, Will Pirkle and the team behind the launch of the original Noiz-Lab Mastering Suite in 2015, who generously helped me to pursue my goal of becoming an audio plug-in developer.

xi

Part I

Setting Up Your Home Studio

In the first part of Home Studio Mastering, you will learn how to turn your bedroom, garage or attic into a home studio suited for mastering. It is very important to create an environment that is optimized for critical listening, because your ability to create great-sounding masters is limited by the accuracy of your playback system. The main factors involved are your studio room, its arrangement, and its acoustic treatment, as well as your choice of software and hardware. It is essential that every component of your playback system is of high quality, for in mastering as in most other disciplines, you are only as strong as your weakest link.

Chapter 1

Mastering

1.1 Can you Master Your Music 1.3 A Bit of History

in a Home Studio?

1.4 Managing Expectations

1.2 What is Mastering?

1.1 CAN YOU MASTER YOUR MUSIC

IN A HOME STUDIO?

Mastering music in a home studio can be done. Moreover, if you know what you are doing, it shouldn’t be that difficult. It is true that you probably shouldn’t master a top-10 billboard commercial album in your home studio, and that you might encounter some challenges with genres like Hip-hop or EDM that need an extremely tight low end, but generally speaking, it is just another skill you can learn if you are willing to spend time and effort on it. It might sound contradictory, but mastering is easy once you know how to do it. The difficulty lies in understanding the mastering process and developing a mastering workflow that suits you.

The initial challenges are creating a neutral playback system, and becoming aware of certain acoustic and psycho-acoustic principles to compensate for working in a less than perfect home studio.

Furthermore, the mastering process itself requires well-developed listening skills, clever decision-making and a creative and musical mind, which fortunately are all traits common to many musicians and producers to begin with. If you are able to produce music 3

4

PART | I Setting Up Your Home Studio up to the point of delivering a finished mix, the chances are you are already familiar with most tools needed for mastering a song.

You just need to know how to use these tools in the context of mastering.

“The technical and creative decisions you need to make during the mastering process require a high level of objectivity.”

Although generally speaking it is possible to master music in a home studio, various challenges arise when mastering your own music specifically. The technical and creative decisions you need to make during the mastering process require a high level of objectivity, and usually, after producing a song up to the point of mastering, you will have lost this objectivity completely. Fortunately, with the help of reference tracks and a well-defined mastering workflow, you can work around this problem, especially when FIGURE 1.1 The Author’s Home Studio—Leiderdorp, The Netherlands (1)

Mastering

5

you take a few days off between mixing and mastering. Nonetheless, mastering a song you have produced yourself will probably take you a great deal longer than mastering a song you hear for the first time. It is not uncommon to get overly concerned with tiny details in your track, and go back and forth between mix and master many times. This is not only frustrating and time consum-ing, but it also prevents you from staying in the specific state of mind that is needed for mastering—which is fundamentally different from mixing. Once you get more familiar with the mastering process, though, these issues will become less problematic or even completely disappear. However, for the purpose of learning the art of mastering, it is generally a bad idea to use your own tracks.

1.2 WHAT IS MASTERING?

It is still somewhat of a mystery, what exactly takes place when an album gets mastered—especially when it comes to the industry veterans who master big commercial releases in studios that seem to resemble the cockpit of an airplane. Not even those who make regular use of their services—like musicians sending off their album to be mastered, or mixing engineers who are often eager to leave the mastering to a dedicated professional—know what the work of a mastering engineer entails exactly. The consensus is, however, that when an album gets back from mastering, it sounds better, even when it is sometimes hard to tell why. The following description is meant to take away any confusion regarding the difference between mixing and mastering, and to provide a carefully defined definition for the purpose of this book: The post-production process of an album can be divided into two parts: mixing and mastering. During mixing all tracks recorded for a particular song are combined to form a balanced whole, which is done for all songs on the album consecutively. After mixing, every song is carefully saved as a single file in preparation for mastering.

6

PART | I Setting Up Your Home Studio During mastering, the overall tonal balance, width and depth, dynamics and loudness of all songs on the album are optimized and matched to create an album that sounds great, translates well to a variety of audio systems and has a natural flow to it when listening to it from start to end. Finally, the songs are prepared for online distribution or replication, either in the form of single releases, or as an official master-CD or a DDP-file (Refer to: 13.4

Creating a Master-CD or DDP-file).

● A song’s tonal balance can be described as the way the song’s frequency content is distributed along the audible frequency spectrum. As a result of this distribution, your track may for instance have a bright or dull timbre, or may sound muddy in the low end or harsh in the high end.

Most popular genres of music benefit from a relatively transparent sound with clearly audible vocals.

● A song’s width can be described as the way different elements in the song are placed in the stereo field. A song’s depth can be described as the acoustic information or reverb it contains. Your song may for instance sound narrow or wide, or nearby or distant. The appropriate width and depth vary across different genres of music.

● A song’s dynamics can be described as the differences in loudness and intensity of different sections of the song, and the amount to which a song or particular parts are allowed to vary in loudness. Your song may for instance sound uncontrolled, or lifeless. The appropriate dynamics vary greatly across genres, but in most popular genres of music, the dynamics sound controlled without the liveliness being negatively affected.

● A song’s loudness can be described by the way you acoustically and psycho-acoustically experience its loudness. Because the human experience of loudness is biased in many ways, due to our specific physiology, there are

Mastering

7

a few techniques to trick the brain into thinking a song is louder, with minimal changes to the amplitude of the actual waveform that makes up the song. This can be a difficult concept to grasp, but in practice optimizing loudness is actually quite an easy thing to do.

Creating a professional-sounding album calls for optimization of all four of these aspects. Besides making your songs sound good, this optimization also ensures that your songs will translate to the broadest possible range of audio equipment, from tiny laptop speakers to huge PA installations. In order to make meaningful adjustments to a song during mastering that accomplish this, it is vital that you know what a professionally produced song sounds like. This might sound obvious, but production conventions can vary greatly depending on genre. In addition, it is often surprising how different a song sounds when recalled in memory as opposed to while critically listening with the ear of an audio engineer—

especially songs from the period before you started producing, and that you haven’t heard in a while. The emotional impact of such a song might have greatly influenced the way you perceived its sonic qualities with untrained ears. Critically listening is a true art, and it is easy to fool yourself when chasing a dream instead of working towards a well-defined goal.

“There is no standard approach to mastering music.”

A mastering studio should act like a magnifying glass, because every single detail should be clearly audible, in order for you to make the right adjustments to your song. Mastering is a highly goal-oriented activity. You listen, you detect problems and you solve them. You listen again, you detect points of improvement and you perfect them. Since every song usually requires a unique approach, you should refrain from using standard presets. Don’t be led astray by certain presets that make your tracks sound better,

8

PART | I Setting Up Your Home Studio because carefully selected settings almost always lead to better results. Reverting to presets is indicative of insufficient knowledge or experience. Ideally, you need to know the options provided by each individual plug-in, as well as the effects of every control on its interface. So remember, there is no standard approach to mastering music. You are on your own a lot of the time, and you should learn to accurately judge what a song needs by yourself. In that regard it is good to be opinionated when it comes to some controversial mastering related topics, like best practices for samplerate and bitdepth settings and optimizing loudness, and to develop a recognizable mastering style of your own.

1.3 A BIT OF HISTORY

Originally, music was recorded straight onto vinyl records, but when the first analog tape recorder became commercially available in 1948, this started to change. As tape recorders became more popular, the need arose for a dedicated transfer engineer to carefully transfer the tape recordings onto vinyl records. With the arrival of the multitrack recorder in the late 1950s, the production process changed for good, and the era of the mixing engineer began. Nonetheless, transferring the finished mixes of these mixing engineers was still the job of transfer engineers, who gradually started to optimize the recordings as a whole. As they started to expand their activities, they inadvertently founded the profession of mastering engineer. They became responsible for optimizing and matching the sonic aspects of different songs on a record, as well as correctly delivering the master-disc to the CD-plant for replication or duplication. This is officially called pre-mastering, since the traditional meaning of mastering is producing the actual glass master that is used in CD-plants as the mold to produce retail CDs.

Over time, the music industry became heavily dependent on the mastering engineer. This was caused by a discovery in the late 1980s, when it became evident that listeners favored loud tracks over more quiet ones. This seemingly harmless conclusion led to a

Mastering

9

fiercely competitive battle between record labels. From that point onwards, all labels strove to produce louder CDs than those of their competitors, or at least equally loud. This is what became known as the Loudness War. As the years progressed, the Loudness War gave rise to a huge increase in the loudness of CDs, which in turn started to negatively affect the dynamics and overall quality of commercial music a great deal. When the first DAW software became commercially available, more and more people started setting up home studios throughout the 1990s. However, the ever increasing loudness of commercial albums mastered by dedicated professionals with access to high end gear, essentially made it impossible for home studio owners to realistically compete with the releases of big record labels. Fortunately, as the Loudness War drove album loudness to an extreme, many mastering engineers became reluctant to go along with this trend. Initially, from a commercial point of view, many big record labels were not prepared to compromise. But then, the Internet changed the rules once more.

Since about a decade or so ago, many people have been listening to music online, using services like Spotify and iTunes.

Because these services provide easy interfaces for quickly skipping through countless albums produced in different decades, differences in loudness between songs can create a bumpy listening experience. Therefore, ways of automatically playing every song at the same perceived loudness were developed, referred to as: equal loudness functionality. This is a true game changer, because loud songs that used to sound better at the expense of dynamics and overall quality are now played at the same perceived loudness as songs with far more dynamics and less distortion. The result of this is that songs that are mastered more conservatively sound much better when played through most online services, but still have a disadvantage when played on a stereo, an mp3 player or on many radio stations. This can complicate the decision-making a great deal during a mastering session. However, it is fairly safe to say that in the long term the arrival of the equal loudness functionality will render the Loudness War obsolete.

FIGURE 1.2 Conservative and Competitive Loudness

Mastering

11

1.4 MANAGING EXPECTATIONS

Before diving into the good stuff, it is important to know what you can realistically expect from setting up your own mastering studio and mastering your own music. As mentioned before, you probably shouldn’t master a top-10 billboard commercial album in your home studio. Most people in that particular kind of situation will get their album professionally mastered. However, a lot of people who wish to master their own music nowadays are simply independent musicians and producers working in a home studio, who just want to get their music out there. And with the Internet having changed the music industry for good, not many people like that are going to pay a big fee for mastering every time they want to release a demo online or email the first version of a new song to friends or band mates. For this purpose, as well as for doing independent or medium-sized album releases in general, the quality of masters that can be achieved in a home studio is more than sufficient. For bigger releases, there are many aspects to consider that may contribute to the decision to master an album yourself or to send it off to a professional mastering studio, all of which will be discussed throughout this book. But nothing is impossible, especially in a time when it is getting harder and harder to distinguish a home studio from a professional studio, and where the experience and ingenuity of the engineer often has more influence on the final product than anything else.

“The potential you have for creating great-sounding masters depends on how well you are able to solve any issues that arise when setting up your studio for critical listening.”

The actual quality of the masters you will be able to create depends in many ways on your specific situation. As gear can be upgraded over time, and new mastering techniques can be learned, the thing you have the least control over when working in a home

12

PART | I Setting Up Your Home Studio studio is the choice of your studio room. And in order to be well equipped for the task at hand, you need a studio that is optimized for critical listening above anything else. The following chapters are dedicated to helping you set up such a studio, or to helping you upgrade your existing home studio if necessary. Certainly, not everybody agrees on the exact specifications for such a studio, but for decent performance, especially in a home studio situation, most people will agree a mastering studio must at a minimum comply with the following requirements:

● The appropriate shape and size.

● Appropriate building materials for the walls, floor and ceiling.

● An appropriate arrangement and a correct setup.

● Decent acoustic treatment.

● A professional monitor set.

● The appropriate additional hardware and software.

FIGURE 1.3 The Author’s Home Studio—Leiderdorp, The Netherlands (2)

Mastering

13

The potential you have for creating great-sounding masters depends on how well you are able to solve any issues that arise when setting up your studio for critical listening, as this will form the basis for your decision-making during mastering. Fortunately, as you will read in the remaining chapters of the first part of this book, optimizing your home studio doesn’t have to be a complex task, and you will probably be able to quickly create decent- sounding masters on a set of decent near-field monitors and a quality pair of headphones. In the long term, after developing a mastering workflow over time that fits your needs, you can become totally self- reliant and have complete creative freedom over your music. So, let’s get started.

Chapter 2

Your Own Mastering Studio

2.1 Make Smart Choices

2.3 Arrangement

2.2 Room Characteristics

2.4 Don’t Panic Please

2.1 MAKE SMART CHOICES

Most musicians or producers do not have many options, if any, when it comes to picking a room for setting up their home studio.

The space you choose plays a huge factor in setting up a neutral playback system, though, because you are subject to its particular acoustic characteristics. Most bedrooms, garages and attics are not built with critical listening in mind and can have a direct impact on your judgment. If you are not aware of this, you will most likely exaggerate the sonic qualities in your master that the room lacks, and suppress the sonic qualities that the room has in abundance.

For example, in a very dry sounding room you might add more reverb than appropriate, or in a very bright sounding room you might cut the high frequencies too much. Besides the space itself, the way you arrange your studio, especially the placement of your speakers and your listening position, can influence your judgment as well. For example, if you put your speakers too far apart you might narrow the low end stereo image too much during mastering. In this chapter you will learn how to select the right room and how to arrange your studio for critical listening.

15

16

PART | I Setting Up Your Home Studio FIGURE 2.1 Mart’s Home Studio—Sassenheim, The Netherlands (1)

2.2 ROOM CHARACTERISTICS

Traditionally, the recommended size of a mastering studio is a minimum of about 1,400 cubic feet. With a regular ceiling height of between 8 to 10 feet, this would mean you need a room with a minimum surface area of about 140 to 175 square feet. Not many people will have a spare room that big in their house. Fortunately, in many cases smaller spaces will suffice, as this recommendation is based on the usage of monitor sets much bigger than the near-field speakers that are specifically designed to perform well positioned closely to the engineer, as seen in most home studios.

Besides that, today’s market offers top-grade materials for optimizing the acoustics of smaller spaces. Still, as a general rule for home studios, you should pick a bigger room over a smaller room, as long as you don’t have to compromise too heavily in other areas.

Your Own Mastering Studio 17

“Every dimension in your room will cause a specific frequency in the low end to be amplified.”

The shape of a room can be critical as well, when it comes to accurate playback. The most predictable option would be to choose a rectangular space where all dimensions—height, width and length—differ from each other, without being exact multiples of one another. More specifically, you want the dimensions to be as unrelated as possible, to minimize problems with an acoustical phenomenon called standing waves ( refer to: 3.3 Standing Waves). Basically, every dimension in your room will cause a specific frequency in the low end to be amplified, and you don’t want these standing waves to accumulate. To evenly disperse the standing wave frequencies, the ideal ratio of the dimensions of your room is approximately 1.618, which is a number known as the Golden Ratio. Ideally, this means you want the result of the biggest dimension divided by the middle dimension, as well as the result of the middle dimension divided by the smallest dimension, to equal this ratio. As the ceiling height is the most constant dimension for regular homes, and this dimension is usually the smallest one, here are three examples with a ceiling height of 7, 8

and 9 feet.

Example 1:

A space measuring 7 x 11 x 18 feet (height x width x length).

Although the ceiling is low, and the room may be on the small side from a traditional point of view at 1,386 cubic feet, the dimensions differ without being multiples of one another, while the ratio of dimensions closely approaches the Golden Ratio: 18 / 11 = 1.636

11 / 7 = 1.571

18

PART | I Setting Up Your Home Studio FIGURE 2.2 Different Room Sizes That Approximate the Golden Ratio Example 2:

A space measuring 8 x 13 x 21 feet (height x width x length).

At 2,184 cubic feet, you are well above the traditional minimum.

In addition, the dimensions differ without being multiples of one another, while the ratio of dimensions closely approaches the Golden Ratio:

21 / 13 = 1.615

13 / 8 = 1.625

Example 3:

A space measuring 9 x 15 x 24 feet (height x width x length).

At 2,898 cubic feet, you are well above the traditional minimum.

In addition, the dimensions differ without being multiples of one another, while the ratio of dimensions closely approaches the Golden Ratio:

24 / 15 = 1.600

15 / 9 = 1.667

Your Own Mastering Studio 19

You will notice that to conform to the Golden Ratio even a room with a ceiling height of just 7 feet needs to be quite spacious already. Because not many people have such a spare room at their disposal, a lot of home studios suffer from issues with standing waves. Besides that, not all rooms are rectangular shaped. Rooms that are irregular or non-symmetric in shape can behave in different ways when it comes to the dispersion of the standing wave frequencies. However, in general the same guidelines apply: if you have a choice between several non-ideal rooms, try to avoid rooms with two or more dimensions in common, and if possible rooms that have dimensions that are exact multiples of each other as well.

Apart from the size and shape, the building materials used for the walls, floor and ceiling can have a huge influence on the way a room sounds, especially when it comes to its frequency response.

For example, a room with hard surfaces like concrete or brick will sound a lot brighter than a room with soft surfaces like plasterboard or wood. To get a feel for how a room sounds, close all doors and windows and make a loud sound, like a high-pitched yell or a handclap. Ideally, no specific acoustic phenomena will stand out and it all sounds natural and warm. However, especially in mostly empty rooms with hard surfaces, you might hear a number of problematic artifacts, like sharp repetitive echoes, a metallic resonance or an overly bright reverberance, that can seriously affect your decision-making during mastering. Although most of these acoustic problems can be solved with acoustic treatment, you should try to avoid rooms that suffer from excessive unnatural sounding artifacts.

2.3 ARRANGEMENT

Besides the characteristics of the room itself, the positioning of your desk, monitors and listening position within the room are vital as well. You start by choosing which side of the room is best suited for accurate playback. In a rectangular room, your best option is to set up your desk and monitors symmetrically along

20

PART | I Setting Up Your Home Studio one of the shorter walls, so they project lengthwise into the room.

As long as the walls to your sides are constructed from the same building material, this will guarantee an accurate representation of the stereo image. In an irregularly shaped room you want to opt for the most symmetric side, where you can set up your speakers along a straight wall in front of you, and preferably at equal distance from two walls to your sides. If possible, keep any irregular-ities, including doors and windows, behind the listening position, especially sharp corners or the alcove in L-shaped rooms, as these can introduce annoying resonances and reflections.

“For critical listening, your head should form an equilateral triangle with the monitors, with your ears at the height of the tweeters.”

After deciding on the best side of your room to set up, you need to correctly position your speakers. For critical listening, your head should form an equilateral triangle with the monitors, with your ears at the height of the tweeters. As the high frequencies projected from the tweeters are highly directional, it is very important to rotate the speakers to the point where the sound travels to you in a straight line (a 60-degree angle) to ensure a realistic stereo image and high end frequency response. The point where you have the most accurate playback is called the sweet spot. Due to differences in the tweeter projection, however, the size of the sweet spot varies between monitor sets. Some sets confine listeners to a very tight spot, whereas others allow greater freedom of movement without losing definition of the high frequency range.

Most home studio owners work with near-field monitors, which are meant to be positioned symmetrically in the room, about 40

to 80 inches apart. When choosing the exact distance between speakers, make sure the stereo image has a well- defined center in the low frequency range, as well as good stereo placement

Your Own Mastering Studio

21

FIGURE 2.3 Listening Position and Speaker Placement in the mid and high frequency range. If your room suffers from excessive unnatural sounding artifacts, you might want to put your speakers a little closer together, even if the manufacturer recommends a specific distance between them. This limits the influence of the room, as the signal that reaches you directly from the speakers will become louder relative to the indirect signal bouncing around the room. In addition, to prevent a build-up of the low end, it is extremely important to keep your speakers away from any walls, preferably by about 40 inches or so. This is especially true if your speakers have bass ports on the back-side. If possible, it is also beneficial to conform to the Golden Ratio once more when deciding on the distance between the speakers and side walls and the distance between the speakers and the wall in front of you.

2.4 DON’T PANIC PLEASE

The criteria discussed in this chapter might sound strict, but don’t be discouraged. Instead, use the information provided to make

22

PART | I Setting Up Your Home Studio informed decisions while choosing a room, determining the best side of the room for accurate playback and positioning your speakers. By doing this, you will lay the foundations for optimizing your studio for critical listening. The next step is to tackle any remaining problems with the appropriate acoustic treatment.

FIGURE 2.4 Mart’s Home Studio—Sassenheim, The Netherlands (2)

Chapter 3

Room Acoustics

3.1 Common Acoustic

3.4 Comb Filtering

Problems

3.5 Flutter Echo

3.2 Acoustic Treatment

3.6 Reverberance

3.3 Standing Waves

3.1 COMMON ACOUSTIC PROBLEMS

In a broad sense, acoustics is the science of sound. In the context of mastering, however, you are mainly concerned with the way sound behaves in a room, and how this behavior changes what comes out of your speakers before it reaches your ears. As discussed in the last chapter, home studios can suffer from a range of acoustic problems that can seriously affect your decision-making during mastering. Acoustic treatment is the process of solving these problems and optimizing your studio room for critical listening. This should not be confused with soundproofing a room, which basically means insulating the walls, floor and ceiling to keep sound in or out, not optimizing the acoustic properties of the room.

“Determine to which degree your room suffers from acoustic problems, and implement specific solutions to reduce them.”

23

24

PART | I Setting Up Your Home Studio With different frequency ranges come different acoustic problems. Low frequencies behave very unpredictably. They are omnidirectional (traveling in all directions), they will travel through thin surfaces essentially undisturbed, and their intensity can vary greatly depending on where you stand in a room. The behavior of higher frequencies, ranging upwards from about 100Hz, is considerably easier to predict. They are directional (traveling in a single direction), they are absorbed by soft surfaces, and they are reflected by hard surfaces. These general behaviors of sound result in the following four common acoustic problems found in home studios:

● Standing waves

● Comb filtering

● Reverberance

● Flutter echo

Each of these acoustic problems can be reduced or eliminated with acoustic treatment, which usually means strategically placing acoustic panels on the walls and ceiling of your studio room, and optionally placing bass traps in the corners. These panels and traps need to have very specific qualities to be effective, so don’t buy into the egg-carton myth, or go about gluing random leftover foam to your walls. Instead, determine to which degree your room suffers from these four common acoustic problems, and implement specific solutions to reduce them. In the remaining part of this chapter we will discuss how to do this.

3.2 ACOUSTIC TREATMENT

There are three main kinds of acoustic panels: absorption panels, diffusion panels and bass traps. Absorption panels absorb sound waves and are usually made of mineral wool, fiberglass or foam with very specific properties. They are usually either 2 or 4 inches thick and have a smooth surface. Diffusion panels, on

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25

the other hand, disperse sound waves and have a very recognizable and often artistic look due to their unusual surface patterns.

They are usually made from a series of wooden blocks or boards, or from synthetic materials. Ideally, in order to achieve balanced results, absorption and diffusion panels would be equally effective over the whole audible frequency range. However, the effectiveness of most acoustic panels decreases when approaching the low end of the frequency spectrum. Therefore, to solve low frequency problems, you use bass traps. Bass traps are either big and heavy broadband absorbers, or meticulously built constructions with membranes or air chambers that target specific frequencies.

As a reference, when buying or creating absorption panels or broadband bass traps, you want to find products that have a table or graph with the absorption coefficients for different frequencies.

As a general guideline, a coefficient of 0.00 means 0% absorption, and a coefficient of 1.00 means 100% absorption. However, there are different ways for measuring and calculating these coefficients, FIGURE 3.1 Basic Acoustic Treatment

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PART | I Setting Up Your Home Studio sometimes with an outcome above 1.00, which makes these tables and graphs a somewhat unreliable source for forming an objective judgment. You should therefore mainly use them to compare different panels or materials, and to determine how consistently a panel or material performs over the audible frequency range. Especially, averaged values, such as the NRC value ( Noise Reduction Coefficient) you find in most tables for absorption panels, only give you a very general idea of a panel’s quality. In general thicker panels absorb lower frequencies. A regular 2-inch absorption panel might perform well ranging down to about 500Hz, while a 4-inch absorption panel still has acceptable values at 250Hz or even 125Hz. The performance of diffusion panels can vary somewhat based on their design, but is more consistent compared with absorption panels.

Diffusion and scattering coefficients can be measured for diffusion panels, but the standards for these measurements are still under debate. In any case, not many companies show tables or graphs for diffusion panels on their websites, which can make it difficult to compare diffusion products. However, because of their relative consistency in performance, any proven design will do the job.

TABLE 3.1

Absorption coefficients of high quality absorber—2-inch thick 125Hz

250Hz

500Hz

1,000Hz

2,000Hz

4,000Hz

NRC

0.17

0.86

1.14

1.07

1.02

0.98

1.00

TABLE 3.2

Absorption coefficients of high quality absorber—4-inch thick 125Hz

250Hz

500Hz

1,000Hz

2,000Hz

4,000Hz

NRC

0.84

1.24

1.24

1.08

1.00

0.97

1.15

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3.3 STANDING WAVES

Depending on the exact dimensions of your studio room, one or more low end frequencies will naturally get amplified. Sometimes this is a subtle phenomenon, but it is also possible that a specific bass note in a song suddenly fills your room with a burst of sub that causes all kinds of objects to start resonating. These frequencies are called standing waves. To understand why standing waves appear, and why they have different frequencies in different rooms, you need to know a little bit more about the way sound behaves.

The pitch or frequency of a sound wave is determined by its number of vibrations per second, which is expressed in hertz (Hz).

Human hearing can process frequencies ranging between about 20Hz and 20kHz, which are respectively the lowest and highest frequency we are able to hear. Besides the specific frequency usually used to describe a musical note, a sound wave also has a wavelength, which is the distance it travels through the air until it repeats itself. A sound wave’s wavelength is inversely proportional to its frequency, which means that low musical notes have long wavelengths, and high musical notes have short wavelengths. For example:

● a 20Hz tone has a wavelength of about 56.3 feet;

● a 200Hz tone has a wavelength of about 5.63 feet;

● a 2,000Hz tone has a wavelength of about 0.563 feet (approximately 6.75 inches);

● a 20kHz tone has a wavelength of about 0.0563 feet (approximately 0.675 inches).

As you can tell from these distances, the wavelengths of certain frequencies will correspond with the dimensions of your studio room. Whenever one of these frequencies occurs, or an exact multiple, it gets amplified by your room and creates a standing wave. Standing waves will be more problematic if several dimensions of your studio space are the same size, as this causes them to accumulate. This is why it is preferable to choose a room with dimensions that conform to the Golden Ratio ( refer to: 2.2 Room

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PART | I Setting Up Your Home Studio Characteristics). In addition, standing waves are usually more evident in smaller spaces.

To solve issues with standing waves, or to adapt your mastering workflow to them later on, it is essential to know at what frequencies they occur. In a rectangular room it is not hard to calculate the standing wave frequencies yourself: you take the speed of sound, which is approximately 1,125 feet per second and divide it by the dimensions one at a time. So say your studio room has a width of 9 feet, this would cause a standing wave at 1,125 / 9 = 125Hz, which is approximately the musical note B2. In an irregular room it can be harder to calculate the standing wave frequencies, but as a general rule you can expect one between any two parallel walls. As an alternative approach you could use a tone generator in your DAW

to sweep through the low frequency range and listen for changes in loudness and resonance. Even just playing the low notes on a bass guitar or synth can help you pinpoint standing wave frequencies.

Once you have identified the standing waves in your studio room, there are several things you can do to reduce or eliminate them:

● You can reduce the standing waves in your studio room by placing velocity-based bass traps in the corners behind your speakers, and perhaps in other corners as well. These kinds of bass traps are essentially big chunks of highly absorbent material, sometimes referred to as broadband absorbers because they perform well over the whole audible frequency spectrum. Pay close attention to their absorption coefficients when buying them, though, as you want to make sure they absorb well enough in the low end to eliminate the particular standing waves in your studio room. You can relatively easily create velocity-based bass traps yourself as well, by applying absorption panels to the corners of the walls at 45-degree angles, and filling up the space behind them using highly absorbent materials. The bigger and deeper you build these custom bass traps, the more effective they become in the low end of the frequency spectrum.

● If you have trouble controlling specific standing waves with velocity-based bass traps, you can try pressure-based bass traps,

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29

also known as tuned traps. Tuned traps are designed to tackle problems at specific frequencies, which make them highly suitable for solving problems with standing waves. Unfortunately, tuned traps are very expensive and hard to manufacture yourself. In some cases they might be your only option though—

apart from moving your walls, that is.

● If you are unable to eliminate the standing waves in your studio room by placing bass traps, there is one more thing you can try.

When you move around your studio room you will notice that in some spots the standing waves will appear to be louder than in other spots. You can take advantage of this, by changing the position of your monitors and adjusting your listening position, until you have found a spot with an acceptable representation of the low end. Remember, however, that it is very important to preserve the symmetry in your arrangement and setup, in order to have an accurate representation of the stereo field.

FIGURE 3.2 Bas’s Home Studio—Zoeterwoude, The Netherlands (1)

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PART | I Setting Up Your Home Studio

3.4 COMB FILTERING

In contrast to low frequencies, the general behavior of frequencies ranging up from about 100Hz is fairly predictable: they are reflected by hard surfaces, absorbed by soft surfaces and dispersed by irregular surfaces. In rooms that are not acoustically treated, especially with concrete or brick walls, reflections can cause all kinds of obvious problems, like echoes, resonance and reverberance. However, there is another less obvious issue caused by reflections that needs to be addressed first: comb filtering.

Comb filtering is a phenomenon that happens when an audio signal converges with a delayed version of itself. When the delay is very short, and falls below something called the echo perception threshold, you won’t be able to distinguish the two signals from each other, but you will instead experience quite radical changes in the overall tonal balance. You can easily hear the effects of comb filtering for yourself, if you load up an audio file in your DAW, duplicate it and use a delay plug-in to delay one of the tracks. If you start with a delay time of 0ms and then increment the delay time by steps of 1ms, you will clearly hear the timbre of the audio change with every step. These changes in the tonal balance are the result of the phase differences introduced by the delay, which will boost some frequencies and cancel out others. When increasing the delay time, at some point you will pass the echo perception threshold, where you will be able to distinguish the two tracks from each other. Depending on the transient content of the track, this threshold might be as low as 10ms or 15ms for drums and percussion, or as high as 50ms or more for vocals and synth pads.

To understand why comb filtering is relevant for creating a neutral playback system, we need to look at how sound bounces around in small studio spaces.

“To make sure you can accurately judge the tonal balance of the tracks you are mastering, you have to prevent problematic comb filtering.”

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31

When you sit at your desk, the direct sound waves from your speakers will reach your ears first. A fraction later, delayed versions of the same sound waves, reflected by the walls, floor and ceiling, will follow. Especially, the reflections that reach you after bouncing off the walls or ceiling just once before they reach you can be problematic, as they will only have very short delay times.

For example, if the sound from your speakers has to travel 4 feet to reach you directly, but the sound reflected by the walls to your sides has to travel 6 feet to reach you, the 2 feet difference will account for a delay of 1.78ms. As you have been able to hear while delaying one of two duplicate tracks in your DAW a similar amount, converging two identical signals with a delay of 1.78ms has a big impact on the tonal balance of your signal. Although, in contrast with the DAW experiment, the volume of real-world reflected signals is much lower than the volume of the direct signal from your speakers, the effects of this kind of comb filtering will still be significant in many cases. Therefore, to make sure you can accurately judge the tonal balance of the tracks you are mastering, you have to prevent problematic comb filtering.

You can solve most comb filtering problems by creating a so called reflection-free zone, or RFZ, at your listening position. To do this, you will need to determine the first reflection points on the walls between your monitors and your listening position. Use the location of the tweeters in your monitors as your point of depar-ture. At these reflection points you will need to apply absorption panels of about 3 by 3 feet or bigger to the walls. For the purpose of creating a reflection-free zone, you should opt for high quality 4-inch absorption panels that perform well over the whole frequency spectrum. If you have 2-inch panels instead, it can be beneficial to leave some space between the panels and the walls, and potentially fill these up with leftover absorbent materials, as this will absorb additional lower frequencies as well. When such a floating construction is applied to the ceiling, it is called a cloud.

Sometimes a thick sheet instead of a panel is used for creating a cloud, because it is easier to attach to the ceiling. For the reflection

32

PART | I Setting Up Your Home Studio point on the wall in the back of your studio room, you can either use absorption or reflection panels. If you feel like your studio room is starting to sound overly dry, you might want to opt for a diffusion panel, or a nice big open bookcase with many different sized books, to preserve the liveliness of the room. Otherwise another absorption panel is probably your best bet. Some people treat the floor of their studio with acoustic materials as well, but in most cases—at least when you don’t have a bare concrete floor—

this is not necessary because the human ear seems less sensitive to acoustic effects caused by the floor than to those caused by the walls and ceiling. Once you have created a reflection-free zone to reduce comb filtering, you have laid a solid foundation for solving any remaining high frequency problems.

3.5 FLUTTER ECHO

If you firmly clap your hands in a small room with hard parallel surfaces, you will hear a fast, rattling echo effect. This acoustic phenomenon is produced by high frequency sound waves reflecting between two parallel surfaces, and is called: flutter echo. In rooms with concrete or brick walls the echoes might be really loud and bright. On the other hand, in rooms with wooden or plasterboard walls, the echoes might be less noticeable or lower in pitch, or, if you are lucky, they may not be there at all. Without clapping your hands you might not notice that your room suffers from flutter echo in any obvious way. However, flutter echo can make the transients in your tracks lose their power and level of detail. This will make it hard to accurately judge the dynamics of a track when it comes to the drums and other percussive instruments.

“Flutter echo can make the transients in your tracks lose their power and level of detail.”

Room Acoustics

33

For the purpose of optimizing your studio room for critical listening, you first of all need to eliminate any flutter echo around the listening position and speakers. So, sit down at your desk, and firmly clap your hands at different positions around you. Depending on the building materials used for the walls, floor and ceiling, and on whether you have already created an effective reflection-free zone, it is possible that no further acoustic treatment is necessary. If you do hear any remaining flutter echo, though, you should symmetrically place additional absorption panels on the walls and ceiling around the listening position, at roughly the height of your speakers and ears. Although high quality 4-inch panels would guarantee a more balanced-sounding room overall, you could opt for 2-inch absorption panels for this specific purpose, because of the high-pitched nature of flutter echo.

Usually, eliminating flutter echo around the listening position and speakers will suffice for critical listening. However, in smaller rooms with concrete or brick walls, the echoes can be so loud that more acoustic treatment is needed. Be careful, though, that you don’t cover the greater part of your walls with absorption panels in an attempt to eliminate the problem entirely, as this can either result in an overly dry and unnatural sounding room or in an extremely dull sounding room if you use low quality absorption materials that only absorb high frequencies. Fortunately, there is no need for this, as only one of two parallel surfaces needs to be acoustically treated to reduce flutter echo. This means that you can alternately place absorption panels on opposing walls at speaker height, as long as you make sure this doesn’t disturb the symmetry of the room too much. If you feel your room is starting to sound overly dry, use diffusion panels instead of absorption panels to preserve the liveliness of the room. In more extreme cases like this, it is important to use your ears to find an acceptable compromise, in which the transients in your tracks are powerful and clear, without the room sounding unbalanced or unnatural.

34

PART | I Setting Up Your Home Studio

3.6 REVERBERANCE

Because sound repeatedly reflects off the walls, floor and ceiling of a room, it can bounce around for a while before it dies out. This acoustic phenomenon is called reverberance. In large buildings like churches or concert halls the reverberance is obvious because the decay time of the reverberance is very long. In small rooms it is far less obvious, but using the trusted handclap again, it is easy to get a feel for the length and character of the reverberance in a home studio as well. Generally, the reverberance in rooms with concrete or brick walls has a more bright and metallic character than the reverberance in rooms with wooden or plasterboard walls.

And, of course, big rooms will generally have a longer decay time than small rooms. In any case, reverberance with a long decay time or unnatural character can seriously affect your decision-making when it comes to the overall clarity and tonal balance of your tracks.

FIGURE 3.3 Bas’s Home Studio—Zoeterwoude, The Netherlands (2)

Room Acoustics

35

If you have strategically placed acoustic panels and bass traps after reading the last few paragraphs, the reverberance in your studio room might already sound subtle and controlled. If you have a relatively big room, however, you might want to shorten the reverberance decay time some more by placing several additional absorption panels around the room. You don’t want your room to sound dull and unnatural, so it is very important to use high quality 4-inch panels that perform well over the whole audible frequency spectrum. If, on the other hand, you have a small room with hard surfaces, the reverberance might have a nice short decay time, but still have an annoying bright and metallic character. In this case you can either place a few more 2- or 4-inch absorption panels around the room, or use diffusion panels if your room is starting to sound overly dry. Remember, though, that although diffusion panels will treat frequency-specific issues like metallic resonances, they will not treat the reverberance itself. Don’t give up if you don’t get the intended results right away, because positioning acoustic panels in order to reduce reverberance will always be a matter of trial and error and personal preference.

Chapter 4

Hardware

4.1 A Basic Digital Setup

4.4 Monitor Sets

4.2 Computers

4.5 Headphones

4.3 Audio Interfaces

4.1 A BASIC DIGITAL SETUP

In this chapter we will go over the specifications for each hardware component of your mastering studio. It probably won’t come as a surprise that many of these subjects are heavily debated, which can make it difficult to find hands-on advice about things like selecting monitor sets or determining which quality convertors you need.

Remember, however, that the most heated discussions about mastering gear are usually between people who don’t share the same motives. For example, an industry professional mastering big album releases in a high end studio will certainly have a different view on the requirements of monitor sets and convertors than a songwriter recording music in a home studio, looking for the best way to share a new song on Soundcloud or a similar online service.

This book, of course, targets the latter, and presents a purely digital approach to mastering. This means that, if you are already recording and mixing music on a computer with a decent audio interface and monitor set, you probably won’t have to change much to your hardware setup for the purpose of mastering. This makes a digital approach particularly well suited for educational purposes, as you 37

38

PART | I Setting Up Your Home Studio don’t have to make any big investments and can start right away.

Moreover, because digital audio technology has properly matured over the past decade, there is no longer any reason for dismissing digital mastering. This doesn’t mean the analog versus digital discussion is over though. Fortunately, more and more engineers these days are combining the smooth sound of analog gear and the flexibility of digital editing and processing in so-called hybrid studios. So although this book focuses on setting up a basic digital studio for pragmatic reasons, nobody is stopping you from expanding your setup with analog components and finding a good balance between the virtues of both analog and digital mastering.

But first things first, of course. These are the required hardware components for a basic digital mastering studio:

● a computer

● an audio interface

● a set of monitors

● a pair of headphones

Because your ability to make correct decisions during mastering is limited by the accuracy of your playback system, it is very important to make sure all parts of your hardware setup are of sufficient quality for critical listening. Therefore, we will go over the required specifications for each hardware component in the remaining paragraphs of this chapter.

“Because digital audio technology has properly matured the past decade, there is no longer any reason for dismissing digital mastering.”

4.2 COMPUTERS

The foundation of your DAW setup is your computer. There isn’t any convincing reason for using a Mac over a PC or vice versa these days when it comes to audio production, except for when you want to make use of specific software that is not cross-platform

Hardware

39

supported. Also, for the purpose of mastering, you do not require a particularly advanced computer. You don’t need a lot of RAM

for mastering as you will only utilize processing-based plug-ins, and no virtual instruments that require loading thousands of samples in memory. And if you are already mixing multitrack projects without glitches during playback, your CPU will probably suffice as well. However, during mastering you will mainly use high quality plug-ins, which can potentially become a strain on your CPU depending on your specific workflow. If you take a more traditional approach and you just process and edit a single stereo track at a time, you will certainly stay out of the danger zone. Then again, if you take full advantage of modern multitrack software and have a twelve-track album loaded up with a number of high quality mastering plug-ins on each track, you might just push your CPU to its limit.

FIGURE 4.1 Eric Coen’s Studio—Leiden, The Netherlands (1)

40

PART | I Setting Up Your Home Studio If you plan on purchasing a new computer, you might want to keep in mind that developers of third-party audio software sometimes need time for fixing compatibility issues whenever a new version of Windows or macOS is released. If you plan on buying a brand new machine, you might want to do a little research to make sure the software you are using or plan to use is stable on its OS.

If you plan on buying on older machine on the other hand, and you might want to make sure your RAM or CPU is up to the task at hand with regard to a specific software you want to use, it can’t hurt to check the minimum system requirements for this software on the manufacturers’ website. Besides checking if a new system will be compatible with your preferred software, you should also make sure the new machine’s cooling system isn’t overly noisy, a common disadvantage of more powerful computers. This might sound trivial, but a fan constantly producing noise can mask potential noise in your track and can become very tiring if you have to be highly concentrated for a few hours in a row.

4.3 AUDIO INTERFACES

Although virtually all consumer computers come with a built-in sound card, these usually lack flexibility and do not meet the technical requirements for high quality audio processing. Instead, most home studio owners nowadays use a dedicated external audio interface for recording and playing back music. Because a lot of commercial brands compete for the attention of the still-growing target group of home studio producers and musicians, it is not hard to find a good quality external audio interface for a reasonable price. And although for the purpose of recording and mixing you might easily get lost deciding between a wide range of products, the specifications for mastering are actually quite straightforward:

● left and right mono outputs for your monitors

● a stereo headphone output

● decent quality convertors

Hardware

41

● support for a samplerate up to 96kHz

● support for a bitdepth up to 24bit

For the sole purpose of mastering, you only need outputs for your monitors and headphones. In addition, because changes in loudness can affect your perception of sound, it is very helpful to have physical volume controls on the audio interface for both your monitor set and your headphones. This way you can adjust the playback volume quickly after applying processing techniques that affect the volume or perceived loudness of the track you are mastering. Furthermore, your main concern is the quality of the conversion from the digital signal (bits) inside your computer to the analog signal (voltage) that goes into your monitors. An inaccurate DA-conversion will negatively affect the reproduction of your music, which may lead to poor decision-making throughout the mastering process. Although the quality and thus the price of convertors can vary enormously, decent quality convertors are getting more and more affordable. Companies that develop audio interfaces specifically for home studio usage are therefore able to offer fairly advanced models that support up to 96kHz and 24bit for very reasonable prices. Because, when it comes to mastering, you don’t have to worry about latency, about the amount of inputs and outputs, or if your audio interface connects to your computer with USB, FireWire or Thunderbolt; the chances are that your current audio interface will already meet the above requirements. Why exactly you need support for samplerates up to 96kHz and bitdepths up to 24bit, specifically, will be discussed in Chapter 7. For now, you only need to know that even though CD audio is 44.1kHz and 16bit, there are certain practical and mathematical advantages to converting and processing audio at a higher resolution.

“An inaccurate DA-conversion will negatively affect the reproduction of your music, which may lead to poor decision-making throughout the mastering process.”

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PART | I Setting Up Your Home Studio

4.4 MONITOR SETS

A high quality monitor set is essential for mastering. Because every decision you make throughout the mastering process will be based on the sound produced by your monitors, there is no way around that. However, for the purpose of home studio mastering, regular active near-field monitors, like you might already own if you produce music, are usually still the preferred choice. Bigger and more professional types of monitors like mid-field monitors or main monitors might be the obvious choice for big and fiercely acoustically treated studios, but besides being total overkill they do not function very well when placed too closely together in small spaces. Near-field monitors, on the other hand, are specifically designed for use in small spaces. Although they might not meet the traditional standard for mastering, you can certainly use them for home studio mastering, as long as you are aware of their limitations and adjust your workflow accordingly. However, not just any active near-field monitor set will do. These are the basic requirements near-field monitors should meet for home studio mastering:

● a frequency response as close as possible to 20Hz—20kHz

● an ‘unhyped’ linear frequency response

● an accurate reproduction of transients, dynamics and details

● a big enough sweet spot

● decent speaker cables

Ideally, you want a monitor set that is able to produce frequencies from 20Hz to 20kHz, which is approximately the audible frequency range for human beings. However, most near-field monitors don’t go lower than about 40Hz, which means they are unable to reproduce the lowest octave we can perceive. Not being able to hear what’s going on in the low end is a serious problem when it comes to mastering, and can be especially frustrating if you are producing Hip-hop or EDM, or any other music genre where deep kicks or bass lines play a central role. Although you probably won’t find

Hardware

43

any regular near-field monitors that go all the way down to 20Hz, the closer they get to 20Hz the better. This means you should look for near-field monitors with woofers of about 6 to 8 inches. Be aware, though, that many near-field monitors utilize bass ports to increase the low end frequency response. If you are working in a small room and you are forced to place your speakers close to any walls, it is best to have these bass ports on the front of the monitors to prevent unwanted low end build-up caused by the walls.

Although in many situations it is enough to double-check the low end on a pair of headphones that does go down to 20Hz, you can use an additional subwoofer to extend—not boost!—the low end.

For mastering it is also very important that your monitors have a linear frequency response, which means that they give an accurate representation of your music and are not deliberately hyped to sound better, like many hi-fi speakers. For the purpose of mastering, you want them to be as honest as possible. In this regard, monitors with three drivers (3-way systems) are usually preferred over monitors with two drivers (2-way systems), as the crossover frequency of monitors with just a woofer and a tweeter is often right in the intelligibility area of speech. However, 3-way systems are much more expensive, and whenever you see them being offered in the same price range as a comparable 2-way system, this usually means the manufacturer has used cheap parts. And as you require an accurate reproduction of transients, dynamics and details for the purpose of mastering, you don’t want to buy a product for which the manufacturer has deliberately cut corners. For the same reason, make sure you connect your monitor set with decent balanced cables, as cheap cables will gradually pick up more noise and hum, and might even lose high end if they are really long. So make sure they are short and preferably of the same length.

As there is a large range of different brands and products available when it comes to active near-field monitors, researching and comparing different product specifications can pay off. Also, many music stores have a listening room where you can play your favorite music and skip between monitor sets to compare them.

44

PART | I Setting Up Your Home Studio FIGURE 4.2 Two-Way and Three-Way Speakers This often tells you more than any list of technical specifications, as taste plays a big role as well. Don’t forget that even the best monitor set in the world will not be able to compensate for acoustic problems that your home studio may have. Therefore, investing in an overly expensive monitor set is pointless if the rest of your setup is flawed.

4.5 HEADPHONES

Traditionally, using headphones for mastering is not recommended for several good reasons. First of all, because the left and the right signals do not interact, phase and masking issues can remain unde-tected. This makes analyzing a track’s frequency balance very difficult. And second, because headphones will exaggerate the stereo image and eliminate the natural acoustics of a treated studio, which makes judging the width and depth of a track impossible. Nonetheless, there are four important reasons why you need a quality pair of headphones for mastering in a home studio:

● for double-checking the low end of your tracks;

● for listening to your track unhindered by any acoustic problems;

● for analyzing undesirable sounds like noise, clicks, crackles and pops;

● for double-checking edits and fades.

Hardware

45

For double-checking the low end of your tracks, it is vital that your headphones have a frequency response that goes down to 20Hz.

Furthermore, your headphones should have a linear frequency response and should not be deliberately hyped to sound better. In this regard, you should also favor an open pair of headphones over a closed pair, as these will not affect the frequency response as much. Fortunately, contrary to monitor sets, an affordable pair of headphones that meets these criteria is not hard to find.

FIGURE 4.3 Eric Coen’s Studio—Leiden, The Netherlands (2)

Chapter 5

Software

5.1 Audio Editor or Sequencer?

5.5 Width and Depth

5.2 Mastering Plug-Ins

5.6 Dynamics

5.3 Restoration

5.7 Loudness

5.4 Tonal Balance

5.1 AUDIO EDITOR OR SEQUENCER?

When it comes to digitally producing music, there is a distinction between sequencers dedicated to recording and mixing, like Cubase, ProTools, Logic, Ableton, FL Studio and Reaper, and audio editors dedicated to mastering, like Wavelab, Soundforge and Triumph. In practice, however, both types of software usually share a lot of the same functionality when it comes to processing audio and editing waveforms, especially now most audio editors support multitrack, and most sequencers incorporate advanced wave editing tools. This means that you don’t necessarily need to learn how to use a dedicated mastering audio editor to get started, regardless of your specific mastering workflow—you can just use any professional sequencer with which you are already familiar.

“It is up to you if you want to invest time in learning to use a dedicated mastering audio editor and have everything you need in one place, or to use a sequencer with which you are already familiar.”

47

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PART | I Setting Up Your Home Studio That is not to say that audio editors dedicated to mastering don’t have any advantages, as they will usually come with a bundle of high quality mastering plug-ins and will be equipped with advanced metering and tools for preparing an actual CD release. It is not hard to integrate the same functionality in a sequencer-based workflow, with a few third-party plug-ins and a separate standalone application for creating an official master-CD or DDP-file to send off to a CD-plant. It is up to you if you want to invest time in learning to use a dedicated mastering audio editor and have everything you need in one place, or to use a sequencer with which you are already familiar, and if needed a standalone CD and DDP-creator like Hofa CD-Burn and DDP or Sonoris DDP-creator. If you find it hard to make a decision about what software to use at this point, don’t worry. For most of the mastering techniques that will be discussed in this book, any application that lets you edit audio files and load audio plug-ins will suffice.

FIGURE 5.1 Gilles’s Home Studio—Brighton, The United Kingdom (1)

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5.2 MASTERING PLUG-INS

During mastering you are processing your entire mix as a whole, so you need high quality plug-ins to prevent the potential overall degradation of your audio. This doesn’t mean the plug-ins you use for mastering should be expensive or have many features, but just that care has been taken by the developer to avoid or minimize any unwanted artifacts in the processing algorithms. Audio plug-ins are digital, and although some of them seem to possess analog or even magical qualities, from the programmer’s point of view they follow predetermined logic and fully predictable patterns. That being said, certain types of effects, like de-noisers and pitch-shifters, will have a degrading effect just due to the nature of their algorithms. That is why for the purpose of digital mastering, you benefit most from making use of basic and proven processing techniques like equalization, compression and saturation. On the one hand, you can use these techniques surgically to make specific corrections and improvements with the help of multi-band and mid-side processing, and, on the other hand, you can use them to apply an overall pleasant character or timbre to your masters. In the next paragraphs you will learn what kind of plug-ins you need for any restoration your tracks might need, and for improving the tonal balance, width and depth, dynamics, and perceived loudness of your tracks.

5.3 RESTORATION

Sometimes a track contains undesirable sounds like noise, clicks, crackling, pops or sibilance. As unwanted details buried in a mix have the tendency to surface during mastering, it is often wise to see if you can remove or reduce them at the start of your mastering session. For this purpose you use restoration plug-ins like de-noisers, de-clickers and de-essers. Restoration plug-ins can produce serious unwanted artifacts, though, and because you will

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most likely apply these plug-ins on your mix as a whole during mastering, using a professional restoration bundle can make a difference, especially in extreme cases. However, if you are making clean and professional recordings, or if you are producing music in-the-box exclusively, the chances are that you will only incidentally encounter situations that require restoration plug-ins.

5.4 TONAL BALANCE

There are two main reasons for adjusting the tonal balance of your tracks during mastering. You either want to make corrections or improvements to a specific frequency area, or you want to enhance the overall timbre of your track. When making specific corrections or improvements, you want to take a transparent approach, making sure nobody listening to the final track will notice there was an issue in the first place. When enhancing the overall timbre of your track on the other hand, you want to add as much character as possible. Both can be done with a basic equalizer plug-in, but choosing an equalizer that is specifically designed to do one of these jobs can make a significant difference.

For making corrections and improvements to a specific frequency area, you want to use a parametric equalizer. Contrary to ( para)graphic equalizers with fixed frequency bands, parametric equalizers give you full control over its parameters, allowing you to be very precise in targeting specific frequency ranges. Although most advanced parametric equalizers have an integrated spectrum analyzer that shows you the frequency content of your track, you shouldn’t rely on them too much, especially in the low end where they are usually not very accurate. Because your goal is making clean and surgical adjustments, you want an equalizer that has a transparent character and can switch between processing stereo, left, right, mid and side. In addition, it is useful to be able to switch between regular and linear phase processing. Don’t blindly buy into the linear phase hype, however. With regards to mastering, the advantages of linear phase equalizers over regular equalizers

52

PART | I Setting Up Your Home Studio FIGURE 5.3 The Noiz-Lab LE Equalizer Plug-In are subtle and come at a price. Yes, linear phase equalizers will produce more predictable and coherent results when you want to combine copied tracks that are filtered differently, or maybe during mastering when you want to independently equalize the left and right, or the mid and side channel. However, at the same time, they might cause the transients in your track to lose definition. So, the choice between using a regular or linear phase equalizer is not as straightforward as some marketeers would like you to believe.

In addition to making transparent, surgical adjustments, you sometimes want to make more general adjustments that enhance the overall timbre of your track. For this purpose you want an analog-style equalizer specifically designed to add character.

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FIGURE 5.4 Gilles’s Home Studio—Brighton, The United Kingdom (2) Often these kinds of equalizer plug-ins are carefully modeled after famous analog equalizers, which means they usually have only limited features and incorporate some kind of saturation scheme in their algorithms. Equalizers meant for transparent processing usually sound very similar to each other, but the sound of analog emulations can vary significantly. Having a few different analog-style equalizers to choose from can be very helpful.

Besides analog-style equalizers, you can also use saturation and convolution plug-ins to add character to your tracks. Saturation plug-ins mimic the soft-clipping properties of analog components, like tape, tubes or circuitry, and can be used to add warmth and smooth out harsh transients before any distortion becomes audible. In addition, convolution plug-ins can be used to subtly add the character of existing high-end recording studios or concert halls to your masters, which can be especially useful when it comes to music that is produced in-the-box exclusively, and lacks natural acoustics.

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5.5 WIDTH AND DEPTH

When adjusting the width of your track, you shouldn’t use just any stereo imaging plug-in. Many of them are not appropriate for the purpose of mastering, because they work by delaying the left or right channel, or by directly affecting the phase relationship between the left and right channel. When applying these techniques to a full mix, they can cause serious balance and phase correlation issues. Instead, you should mainly use plug-ins that simply sum the left and right channel for narrowing the stereo image, or plug-ins that use mid-side processing for general stereo enhancement. Many stereo imaging plug-ins for mastering are multi-band plug-ins, and although this is useful for narrowing the stereo image of a specific part of the frequency range, when it comes to mid-side-based stereo enhancement you can achieve much more subtle and balanced results by using a standard parametric mid-side equalizer, where you have full control over the shape of your filters. If you want to make sure the stereo imaging plug-in you are using is suitable for mastering, or if you want to check whether your adjustments to the stereo image don’t cause any phase issues, you can use a phase correlation meter or vectorscope.

For adding more depth to your tracks, you use a reverb plug-in.

Reverb plug-ins come in two varieties: algorithmic reverbs, that internally use a combination of delays with feedback to mimic room reflections, and convolution reverbs that use an impulse response or IR to add the natural reverberance of an existing room or hall to your track. Although algorithmic reverbs are very flexible, they often sound artificial and sterile, which makes them not very suitable for general mastering purposes. Convolution reverbs on the other hand, although relatively inflexible, sound much more natural. To make the most of using a convolution reverb, it really pays off to create an extensive IR library with many different rooms and halls from which to choose. Many professional reverb plug-ins for mastering come in the form of multi-band plug-ins,

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PART | I Setting Up Your Home Studio sometimes with additional mid-side support. However, just like in the case of mid-side based stereo imaging plug-ins, this actually limits your options. You have much more control by using a mid-side equalizer instead, to shape the tonal balance of the reverb output.

“To make the most of using a convolution reverb, it really pays off to create an extensive IR library with many different rooms and halls from which to choose.”

5.6 DYNAMICS

For making adjustments to the dynamics of your tracks, you will mainly use one or more compressors. Similarly to adjusting the tonal balance, you either want to make corrections or improvements to the dynamics of a specific element, or you want to enhance the overall dynamics and character of your track, making all elements in your mix glue together subtly. For making specific corrections or improvements, you want a compressor with a transparent character that can switch between peak mode and RMS mode and has a wet-dry mix function for parallel compression. In addition, you want a multi-band version with mid-side support for making surgical dynamic adjustments. For enhancing the overall dynamics and character of your track, on the other hand, you want an analog-style compressor that is specifically designed to add character and glue your mix together. There are many compressor plug-ins out there that emulate well-known analog models like VCA, FET, Opto and tube compressors. They will all respond differently to your tracks’ dynamics, so it is definitely helpful to have a few different emulations lying around.

Whatever compressor you use, though, make sure it has a gain reduction meter, so you have a visual reference of how much compression is applied.

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FIGURE 5.6 The Noiz-Lab LE Compressor Plug-In In addition to compressors, there are several other types of plug-ins that can be used to influence dynamics: dynamic equalizers, transient designers and saturators.

● A dynamic equalizer will only cut or boost a certain frequency range when this frequency range exceeds a certain volume threshold. This makes it possible to surgically adjust an element in your track based on volume and frequency combined.

Although you might rarely use a dynamic equalizer, in some cases they can be extremely useful.

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PART | I Setting Up Your Home Studio

● A transient designer is a plug-in that can increase or decrease the attack and sustain of the transients in your tracks. Although transient designers are not often used in mastering, because they can degrade the overall quality of your track, a multi-band transient designer can be carefully used to bring out percussive instruments that are buried in a busy mix.

● A saturator can be viewed as a dynamic processor as well, as it will react stronger to louder signals. Besides that, many saturators apply some sort of smoothing that is very similar to the release function of a compressor. Therefore, they can be used to smooth out transients or to subtly reduce the overall dynamics of your tracks.

5.7 LOUDNESS

Maximizing loudness is done with a brickwall limiter. This type of limiter increases the volume of your tracks, and intelligently rounds off the edges of any peaks that are cut off by exceeding the digital zero dBFS limit. Because of the extreme loudness standards caused by the Loudness War, brickwall limiters are essentially trying to make the best of a bad situation. Over the years plug-in developers have come up with all kinds of elaborate limiting algorithms that increase loudness without causing unwanted artifacts like distortion or the typical breathing-and-pumping. All of these brickwall limiters have their own strengths and weaknesses. Therefore, if you want your masters to be loud, you want to try as many as you can, so you can pick a brickwall limiter that works well specifically for your music. To get a general idea of the damage a brickwall limiter is doing, it is important that it has a gain reduction meter. In many cases, however, a brickwall limiter alone won’t do the trick. Fortunately, to achieve true competitive loudness in this day and age, there are several equalization and compression techniques, based on psycho-acoustic principles, which can be used to increase the perceived loudness of a track.

Also, because many brickwall limiters will take the power out of

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Although loudness is best judged by ear, there are times advanced loudness meters can be useful. There are three loudness meters that you might need, depending on your situation: a true peak meter, an RMS meter and a LUFS meter.

● A true peak meter can be used to check that your track won’t have any intersample peaks that exceed the zero dBFS limit after conversion to analog. Although in most cases the subtle distortion caused by intersample peaks will not be noticeable,

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PART | I Setting Up Your Home Studio in the case of extreme brickwall limiting you can use a true peak meter to make sure you are not doing any serious damage.

● An RMS meter shows the average loudness of a track, which is much closer to the way humans perceive loudness compared to peak volume. If you know the approximate RMS values of the music genre you are working on, you can use an RMS meter as a reference while using a brickwall limiter to maximize the loudness of a track.

● A LUFS meter usually shows three different loudness values that are commonly used when preparing audio for broadcasting: integrated loudness, short-term loudness and momentary loudness. Because LUFS meters are considered to give a more accurate and comprehensive representation of perceived loudness than RMS meters, they are slowly becoming more common.

Chapter 6

Mastering Workflow

6.1 Evaluating Your Setup

6.3 The Art of Listening

6.2 Choosing Reference Tracks

6.4 General Workflow

6.1 EVALUATING YOUR SETUP

Now that you have read the first five chapters of this book, you are almost ready to start mastering. Let’s quickly recapitulate what we have discussed so far, and prepare for your first mastering session. The first chapter explained that a mastering studio has to meet certain criteria to be suited for critical listening. The subsequent chapters outlined the guidelines that enabled you to make well-informed decisions about room arrangement, acoustic treatment, hardware and software, in order to create a neutral playback system:

● Room arrangement: based on the shape, the dimensions and the building materials of the walls, floor and ceiling, you have selected the most suitable room available for your mastering studio. Preferably, this is a rectangular room with dimensions that approach the Golden Ratio, with relatively soft surfaces.

● Acoustic treatment: you have determined if your studio room suffers from four common acoustic problems: standing waves, comb filtering, flutter echo and reverberance. If so, you have strategically placed bass traps, absorption panels and diffusion 61

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PART | I Setting Up Your Home Studio panels to reduce or eliminate these problems, without creating an overly dry sounding room.

● Hardware: first of all, you have an audio interface with decent convertors that supports up to 96kHz and 24bit. Second, you have a high quality monitor set with a linear frequency response and an accurate reproduction of transients, dynamics and details. And finally, you have a high quality pair of headphones with a frequency response that goes all the way down to 20Hz.

● Software: you have decided if you want to use an audio editor that is dedicated to mastering so you have everything you need in one place, or that you want to use a regular sequencer with which you are already familiar, and a standalone CD and DDP-creator for delivering your album to a CD-plant when the time comes. In addition, you have assembled a collection of plug-ins suitable for mastering.

As mentioned in earlier chapters, it doesn’t have to be a problem if your setup has a few flaws, as long as you find a way to work around them. In this regard, it is vital that you are aware of the strengths and weaknesses of your setup. To familiarize yourself with the sound specific to your studio, you should spend some time sitting in front of your speakers, carefully listening to well-produced music you know inside and out. Please choose CD-quality audio for this purpose, and not any compressed formats like MP3. For different psycho-acoustic reasons we will discuss later in this chapter, it initially might take a while—up to 10 to 15 minutes—before you can judge sound objectively. So relax and listen to a few of your favorite tracks at a modest level, while being aware of the details in the music becoming more prominent, and the overall listening experience becoming more natural. After a while you should feel comfortable comparing different tracks to each other and identifying subtle differences in tonal balance, width and depth, dynamics and loudness. At this point you should be able to identify any remaining issues or characteristics from which your setup might suffer, that you should take into account

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when mastering, or to which you should adapt your mastering workflow. The most common things to take into consideration in a home studio are:

● Does the low end in your studio sound tight and consistent over different frequencies?

● Does your studio sound overly bright or dull, or colored in any way?

● Does your studio sound overly dry or wet, when it comes to reverberance?

● Is the stereo image in your studio balanced?

● Do the transients in your studio sound tight and direct?

● How much can you move before you are out of your monitors’

sweet spot?

Knowing the answers to these questions can help your decision-making during the mastering process and help you determine when you should double-check your settings on your headphones, FIGURE 6.1 Kevin’s Home Studio—Haarlem, The Netherlands (1)

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PART | I Setting Up Your Home Studio taking your monitor set and room acoustics out of the equation. In

Part II of this book, when actually applying processing techniques, we will discuss in detail how to work around any specific issues your setup might present.

6.2 CHOOSING REFERENCE TRACKS

After identifying the sonic challenges specific to your setup, you need a way to preserve your objectivity, both for the extent of a long continuous mastering session, as well as over the course of different mastering sessions spread out over time. For this purpose, you should select a number of high quality and well- produced reference tracks to which you can compare your masters during a session. Because the sonic conventions for commercial music can vary greatly between music genres and era of release, it is important to choose contemporary reference tracks, released by bands or artist from the same music genre you are mastering. In addition, because even within a specific music genre there can be great diversity in sound, you should choose reference tracks that have a particular sound you like very much. Make sure, however, not to confuse tracks you like from a musical perspective, with tracks that are well-produced. Reference tracks are only meant for referencing the overall tonal balance, width and depth, dynamics and loudness of a track, and nothing else.

“Because the sonic conventions for commercial music can vary greatly between music genres and era of release, it is important to choose contemporary reference tracks, released by bands or artist from the same music genre you are mastering.”

To get a general idea of the sonic conventions of a particular music genre, three to five reference tracks should be enough. Make sure you have CD-quality versions of them: WAV or AIFF files of 16bit and 44.1kHz. Compressed formats like MP3 are not suitable

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for detailed referencing. To make sure your reference tracks are professionally produced and mastered, listen to them on as many audio systems as possible, from big PA installations, to boomy car systems, to hyped hi-fi installations, to tiny mobile phone speakers. If they translate well across all these systems, they will make a good reference and provide insight into how a professional production sounds. Next, you should carefully listen to them in your own studio and familiarize yourself with how they translate to your own setup. From then on, you can use your reference tracks to calibrate your brain at the start of a mastering session, or to fall back on during a mastering session when you feel you are losing your objectivity. In the next paragraph you will learn why this is essential for creating a good master, and how your brain can fool you if you are not careful.

6.3 THE ART OF LISTENING

Although casually listening to music is generally a fairly intuitive experience, as a result of human evolution your brain is constantly changing your perception of sound without you knowing it. With regards to music production, and mastering in specific, this can cause all kinds of seemingly inconsistent results. For example, a master that sounds balanced and powerful upon completion might sound thin and weak a day later. You may question your skills or judgment in such a situation, but usually it is just your brain playing tricks on you. The human perception of sound is described in the scientific study of psycho-acoustics, and because all decisions you make during mastering are based on how your brain interprets sound, it is essential to be aware of the following three psycho-acoustic principles: tonal balance normalization, loudness bias and ear fatigue.

● Tonal balance normalization: over the course of a few minutes, your brain will make the tonal balance of any track sound neutral. Because your brain is constantly normalizing your perception in this manner, it is extremely hard to objectively judge

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PART | I Setting Up Your Home Studio the tonal balance of a track, or to make meaningful adjustments with an equalizer. To work around this problem, you can use reference tracks to calibrate your brain at the start of a mastering session, and to compare your track to while setting an equalizer.

● Loudness bias: your brain will change your perception of the tonal balance of a track depending on the playback volume.

A rough guideline: when you turn up the playback volume, the low and high range will get relatively louder compared to the mid range. This means that a track will sound more powerful and clear when you turn up the volume, and less powerful and clear when you turn down the volume. (This is the psycho-acoustic principle that led to the Loudness War). To work around this problem, it is vital that you compare different settings and different tracks to each other on the same perceived loudness, judged purely by ear.

● Ear fatigue: in the course of a long mastering session, you might notice you are becoming less perceptive to the details in the music, especially after listening back at a high playback volume for a while. It is very important to recognize these moments, as it will become increasingly more difficult to apply any meaningful processing until you take a break. To work around this problem, it is best to listen back at a constant and modest playback volume while mastering, except when making adjustments to the low end of your track, as low frequencies are best judged on a high volume.

“The key to producing a great-sounding master, is using reference tracks and comparing different tracks and settings at the same perceived loudness.”

Your brain can easily fool you, with all kinds of problems as a result. Keeping these three psycho-acoustic principles in mind, however, will enable you to maintain control over the mastering process

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FIGURE 6.2 Fletcher-Munson Curves for Equal Loudness and produce consistent results. The key is using reference tracks and comparing different tracks and settings at the same perceived loudness. In addition, some audio engineers recommend a specific playback volume of 83dB SPL, or sometimes 89dB SPL, because at these levels your perception of tonal balance is supposed to be most accurate (as described in the Fletcher- Munson Curves). However, listening back at such a high volume can cause ear fatigue quickly and is not necessary when consistently using reference tracks. Just like compensating for any issues from which your setup might suffer, we will discuss dealing with psycho-acoustic principles in Part

II of this book as well, when actually applying mastering techniques.

6.4 GENERAL WORKFLOW

Mastering a single track for online release or a full album to send off to a CD-plant requires quite different approaches. In case of a single track, you follow several consecutive and predictable steps

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PART | I Setting Up Your Home Studio that can quickly lead to an optimized and great-sounding master.

However, mastering a full album—or any release with more than one song for that matter—is far less straightforward. Besides just optimizing all tracks individually, you now have to create a sonic unity between them as a whole as well, so the album has a natural flow when listened to from start to end. Creating such a sonic unity requires you to match all tracks in terms of tonal balance, width and depth, dynamics and loudness. In many cases this will limit your options for optimizing the individual tracks. In addition, because processing techniques like equalizing and compression influence the actual and perceived loudness of a track, and because setting the relative volumes between tracks influences your perception of the tonal balance, creating a balanced master can be a challenge.

Especially when your release consists of a large number of songs, you might find yourself going back and forth between tracks for quite a while, before arriving at a suitable compromise between the optimization of the individual tracks and the overall album flow.

General workflow for mastering a single track:

● Step 1: restoring any unwanted artifacts like noise, clicks and crackles.

● Step 2: optimizing the tonal balance, width and depth, dynamics and loudness.

● Step 3: rendering the final master, applying start and end fades and uploading it.

General workflow for mastering a full album:

● Step 1: restoring of any unwanted artifacts like noise, clicks and crackles.

● Step 2: creating an initial volume mix for the relative volumes between tracks.

● Step 3: optimizing and matching the tonal balance, width and depth, dynamics.

● Step 4: creating the final volume mix for the relative volumes between tracks.

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● Step 5: maximizing the overall album loudness.

● Step 6: rendering the final masters and applying start and end fades.

● Step 7: creating an album montage with the appropriate spacing and crossfades.

● Step 8: adding ISRC and CD text and creating the master-CD

or DDP-file for the CD-plant.

For the purpose of learning the art of mastering, it is best to start practicing with single tracks. This way you can get used to working around issues in your setup, using reference tracks and dealing with psycho-acoustics principles. In Part II of this book, a wide range of mastering techniques will be discussed, for you to explore and combine in different ways. Of course, once you feel comfortable with mastering a single track, you can make the transition to EPs and albums.

FIGURE 6.3 Kevin’s Home Studio—Haarlem, The Netherlands (2)

Part II

Home Studio Mastering

Techniques

In the second part of Home Studio Mastering, you will grow familiar with the various techniques you can apply while mastering your tracks. If you have prior experience with mixing, you will already know many of the plug-ins used, such as equalizers, compressors and limiters. During mastering, however, these plug-ins are used far more subtly compared to mixing. Even so, a series of subtle adjustments can amount to a huge difference in quality between the mix and the master. Chapters 7 through 13 will take you through the various mastering techniques in what can generally be considered the appropriate order for any basic mastering session. Keep in mind, however, that many of these techniques are both used for optimizing single tracks, as well as for matching the sonic character of multiple tracks that are part of the same release to enhance album flow.

Chapter 7

Preparing Your Session

7.1 Finalizing Your Mix

7.5 Checking Render Section

7.2 Choosing Samplerate and

and Fades

Bitdepth

7.6 Setting up Your Mastering

7.3 Setting Output Volume and

Project

Headroom

7.4 Checking Mono

Compatibility

7.1 FINALIZING YOUR MIX

Before you can begin the mastering process, your mix needs to be ready. First, this means that you must be 100% happy with your track from a musical and creative production perspective. And second, mix-technically all sonic problems should be solved, and all parts of your mix should be well balanced, at least relative to each other. During mastering you are working with a single stereo WAV

or AIFF file, so you essentially want your mix to sound as close to release-ready as possible, but without unnecessarily reducing overall dynamics, maximizing loudness or applying any kind of potentially destructive processing to your track as a whole. If you are a beginner, this means it is best to leave the master bus of your mix empty. Some more experienced engineers mix with certain effects inserted on their master bus, but unless you have very specific reasons for doing so, this is considered bad practice, especially in the case of aggressive compressors and limiters for maximizing 73

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PART | II Home Studio Mastering Techniques loudness. Maximizing loudness is a destructive process, and will limit the effectiveness of other mastering techniques. Therefore, it should be the very last thing you do during your mastering session.

If your mix has effects inserted on the master bus—which is unfortunately standard with some modern DAW template projects—it is usually best to remove them. Apart from that, if you are happy with your mix both creatively and technically, you are ready to render it as a stereo WAV or AIFF file for the purpose of mastering.

Although this might sound like a trivial thing, there are actually a few much-debated tasks that come into play here:

● choosing samplerate and bitdepth

● setting output volume and headroom

● checking mono compatibility

● checking render section and fades

In the next four paragraphs of this chapter, you will learn how to approach these tasks for the purpose of home studio mastering, so FIGURE 7.1 Tom’s Home Studio—Mannheim, Germany (1)

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you can make informed decisions about how to render your mix.

In addition, in the last paragraph you will learn how to properly set up your mastering project.

7.2 CHOOSING SAMPLERATE AND BITDEPTH

It is very important that the stereo WAV or AIFF files you use during mastering are rendered at the appropriate samplerate and bitdepth, and that you set your DAW mastering project to the same rates. Although regular CDs used for the commercial distribution of music contain audio data with a samplerate of 44.1kHz and a bitdepth of 16bit—referred to as the Red Book Standard, there are benefits to using higher rates during the mastering process itself. There is a lot of debate about which exact rates are appropriate for digital mastering, but fortunately, for the purpose of home studio mastering, the answer is fairly straightforward. Using a samplerate of 44.1kHz and a bitdepth of 24bit will ensure good quality masters without putting too much strain your computer’s CPU. If you own a fast computer, however, you might want to consider using a higher samplerate of 88.2kHz for increased precision (or 96kHz, if 88.2kHz is not available). Even if you have recorded or mixed your track on lower rates, it is still beneficial to render your mix on these recommended higher rates for the purpose of mastering, as you will be applying processing techniques to your mix as a whole. This is especially true for the bitdepth, as most DAW software will internally process audio at 32bit or 64bit, until a file is rendered. To understand why these rates are appropriate you need to know a little bit more about the intricacies of digital audio.

“Using a samplerate of 44.1kHz and a bitdepth of 24bit will ensure good quality masters without putting too much strain your computer’s CPU. If you own a fast computer, however, you might consider using a samplerate of 88.2kHz for increased precision.”

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PART | II Home Studio Mastering Techniques Inside your computer, an audio file is essentially a string of values that are referred to as samples. These samples form a waveform that your audio interface can convert to the continuous voltage signal needed to feed your speakers. The precision of a digital waveform determines the accuracy of the conversion to the analog realm, and is dependent on its samplerate and bitdepth.

● An audio waveform’s samplerate can be viewed as its resolution over time, as it tells you the number of samples used to represent one second of audio. For example: a samplerate of 44,100Hz (or 44.1kHz) means that for every second of audio playback, 44,100 samples are sent to your audio interface for conversion.

● An audio waveform’s bitdepth can be viewed as the resolution of each individual sample, as it tells you by how many different amplitude steps or values a sample can be represented. For example: a bitdepth of 16bit means that there are 65,536 steps (2^16 = 65,536) that can represent one sample.

Besides having to conform to available technology, the standard samplerate of 44.1kHz used for audio CDs was chosen because reproducing frequencies up to 20kHz, which is approximately the upper limit of human hearing, requires a minimum samplerate of 40kHz: twice as high. (The explanation for this is beyond the scope of this book, but is described in detail in the Nyquist-Shannon sampling theorem). To leave some headroom FIGURE 7.2 Sampled Waveform

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for the slopes of the conversion filters, however, a slightly higher samplerate was needed, and the number 44,100 has properties that provide some mathematical advantage. Although some people claim they can distinguish between different samplerates higher than 44.1kHz in studio rooms with superb acoustics and high-end audio installations, the question whether using higher samplerates actually results in audible differences remains a topic of discussion. It is true, though, that when actually processing audio digitally on a relatively low samplerate of 44.1kHz—as opposed to just reproducing it—some equalizers might be less accurate in the top end, and some saturators and limiters might produce more aliasing. However, these problems can be considered algorithmic artifacts, rather than being the direct result of a lack of precision.

To ensure these particular plug-ins perform equally well on lower samplerates like 44.1kHz, professional plug-in manufacturers will prevent these artifacts by upsampling and downsampling the audio signal within the plug-in, a process usually referred to as oversampling. When it comes to mastering plug-ins, oversampling is used very often to conveniently provide high resolution processing on any samplerate. However, repeatedly upsampling and downsampling can theoretically degrade your audio. Whether this degradation is actually audible remains a topic of discussion as well, but if you have a computer that can handle it, you might want to consider mastering at a higher samplerate. Choosing a samplerate that is a multiple of the target samplerate of 44.1kHz, like 88.2kHz, will ensure accurate upsampling and downsampling. In any case, please remember that if you aim to optimize the quality of your masters, there are much more influential factors involved than the samplerate, as long as you master on a minimum samplerate of 44.1kHz and use decent plug-ins.

The standard bitdepth of 16bit used for audio CDs was chosen because it is relatively precise and conformed to available technology. In contrast to the samplerate, the difference between bitdepths can be clearly audible. A bitdepth of 16bit sounds fairly clean.

However, if you reduce the bitdepth—something you can easily

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PART | II Home Studio Mastering Techniques do with a Bitcrusher plug-in—the precision of the sample values will decrease, and you will notice the audio signal getting distorted ( refer to: 13.2 Rendering Your Tracks). In turn, when working on a higher bitdepth of 24bit, the precision of the sample values will be far greater. A 16bit sample can be represented by 65,536

steps (2^16 = 65,536), but due to the exponential increase in precision, a sample of 24bit can be represented by 16,777,216 steps (2^24 = 16,777,216), which is significantly more precise. In an acoustically treated home studio with decent near-field monitors, you might be able to distinguish between the two, especially when applying more extreme processing techniques or when you are dealing with low-level files. The 24bit signal will sound less gritty. The difference might be subtle, but nonetheless significant in the context of mastering, where you apply processing techniques to your mix as a whole. However, due to the exponential increase in precision, the audible difference between 24bit and 32bit is negligible at best, and therefore there is no real need to go higher than 24bit. Of course, you will have to return the files to 16bit eventually, when creating the actual master-CD or DDP-file, but there are ways to preserve the character of the 24bit files throughout this conversion ( refer to: 13.2

Rendering Your Tracks).

7.3 SETTING OUTPUT VOLUME AND HEADROOM

The volume at which you render and save your mix can have a direct effect on your track’s precision, especially when you can’t avoid performing a 16bit render instead of a 24bit render. You can view the bitdepth of a WAV or AIFF file as the maximum number of bits available to represent the samples in the audio signal. However, because signals that are lower in volume utilize fewer bits, the actual amount of bits used to represent a waveform is eventually determined by the file’s volume. Unintuitively, as your DAW

will internally process audio on 32bit or 64bit, you might only notice this after playing the actual rendered audio file. You can easily verify all this. First, create a 16bit render of a track with the

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volume turned way down, say −72dB. Then re-import this file in your project and turn up the volume again by 72dB. The decrease in the precision of the sample values will make the track noisy and distorted just like a bitcrusher plug-in. Some soft passages might even disappear completely from the signal. To ensure sufficient precision, it is therefore important to render your file at the appropriate volume.

“You can view the bitdepth of a WAV or AIFF file as the maximum number of bits that is available to represent the samples in the audio signal. However, because signals that are lower in volume utilize fewer bits, the actual amount of bits used to represent a waveform is eventually determined by the file’s volume.”

As a general rule you can say that every time you lower your signals volume by about 6dB, you lose one bit of precision. This is because lowering the volume of a waveform by 6dB decreases the waveform’s amplitude by approximately half, and that lowering the bitdepth of a file by one bit decreases the precision of the sample values by half. Some people like to leave a little headroom when rendering their mixes to compensate for the volume increase caused by certain mastering techniques. However, when it comes to 16bit audio files, you do not have a lot of bits to spare. Besides that, in the modern digital domain you can just turn down the track’s volume fader in your mastering project to create some headroom, as the resulting DAW volume processing will be done at high precision internally. In any case, if you hold to the recommendation in paragraph 7.2 and render your mix at 24bit, you probably don’t have to worry too much about volume or headroom. As a reference, a 24bit file with 48dB headroom has a precision equivalent to a 16bit file. Even leaving 12dB to 18dB

headroom, for example, is certainly acceptable, as it will only decrease the precision by two or three bits. On the other hand,

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PART | II Home Studio Mastering Techniques if somehow your only choice is rendering a 16bit file, it is best to render your mix as loud as possible, without exceeding zero dBFS on the master bus. This will ensure your file gets saved with maximum precision, while you can safely turn down the track in your mastering project if you need any headroom. If you do so, be aware that channel volume faders generally adjust the volume after any channel effects have been applied. If you are using mastering plug-ins that emulate analog behavior and react differently to your track depending on its volume, it might be better to turn down the actual waveform, which will lower the track’s volume before any channel effects are applied.

7.4 CHECKING MONO COMPATIBILITY

Although stereo playback is standard in most audio devices these days, some online services and mobile phones still play audio in mono. Because a stereo-to-mono conversion can result in lowering the volume of certain elements of your mix, or even make them disappear completely, it is important to listen to your track in mono before starting your mastering session. This is especially true when you have used stereo widening techniques in your mix that have changed the phase relationship between the left and right channel, or have introduced a delay between the left and right channel. When a stereo signal is converted to mono, the left and right signals are summed and the result is divided by two. Because of summing the channels, any mix elements that are out of phase will lower in volume or cancel themselves out, and any mix elements that are delayed will experience changes in tonal balance due to the effects of comb filtering. This can potentially disturb the balance in your mix. Switching between mono and stereo playback should give away most phase and mono compatibility issues immediately. For more subtle issues, or for double-checking your findings, you can use a correlation meter or a vectorscope to analyze the phase relationship between the left and right channel.

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“Because a stereo-to-mono conversion can result in lowering the volume of certain elements of your mix, or even make them disappear completely, it is important to listen to your track in mono before starting your mastering session.”

7.5 CHECKING RENDER SECTION AND FADES

Finally, it is important to check if there is sufficient headroom and tailroom (a brief moment of silence or ambient noise) at the start and end of the actual rendered WAV or AIFF file, and if no fade-in and fade-out have been made prematurely. Having no headroom at the beginning of a track can sound very unnatural, especially in the case of recordings that contain live-recorded instruments or a lot of acoustic information or ambient noise. Therefore, when rendering your mix, make sure to leave some space for creating a FIGURE 7.3 Tom’s Home Studio—Mannheim, Germany (2)

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PART | II Home Studio Mastering Techniques short fade-in at the final stage of the mastering session. This will introduce a less instantaneous transition from silence to music, and prevents accidentally cutting off the initial transients of live-recorded instruments. The same goes for the tailroom at the end of the track, so make sure to leave enough space for creating a smooth fade-out at the final stage of the mastering session. This is especially important when your track ends on a note or chord that slowly dies out, because after adjusting the dynamics and loudness of your track during the mastering process, a prematurely created fade-in or fade-out may turn out to be too short. Creating a fade-in and fade-out can make the ending of your track sound rushed or unnatural, so it should be done during the final stage of the mastering session. In addition, if your track contains a lot of steady noise you might want to remove, you should leave a few seconds of noise intact at the start or end of your track, for noise reduction purposes ( refer to: 8.4 Noise and Hiss).

7.6 SETTING UP YOUR MASTERING PROJECT

To recapitulate, this chapter recommends the following specifications for rendering your tracks for the purpose of home studio mastering:

● A samplerate of 44.1kHz. If you have a fast computer, you can try 88.2kHz, or 96kHz if 88.2kHz is not available in your DAW.

● A bitdepth of 24bit. No reason to go higher.

● Minimal headroom for 16bit renders, to preserve precision. For 24bit renders leaving headroom is fine, but not necessary. We are in the digital domain, and you can safely turn your track’s waveform or volume fader down in your mastering project.

● Preferably solve any mono compatibility issues.

● No fade-in at the start of your track, and no fade-out at the end of your track.

● Leave enough headroom at the start of your track and enough tailroom at the end of your track, especially if you plan on doing noise reduction.

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After rendering your track, it is time to create your mastering project. As discussed, you can either do this in a DAW or in an audio editor ( refer to: 5.1 Audio Editor or Sequencer? ). Whichever one you choose, create an empty project and set it to the same samplerate and bitdepth you used for rendering your track. Then, first load up your reference tracks, on different channels. If any of your reference tracks have a different samplerate than your project, they need to be converted to the project samplerate. Most DAWs and audio editors will ask you if they should convert them for you automatically, but in some cases you will have to do this manually.

When the reference tracks are loaded in, create a group channel and route them to it, so you can control their volume with just one fader. As your reference tracks are probably already mastered, it is best to lower their volume by about 12dB or so, to create some headroom. Then, use the tracks’ individual volume faders to match their loudness, purely by ear. The best way to do this is to instantly switch between them during playback, using the solo buttons on the track’s channels. In some cases matching the loudness of your reference tracks is easier said than done, either because of the psycho-acoustic relationship between volume and frequency ( refer to: 6.3 The Art of Listening), or simply because they are very different from each other. Don’t rush this process, as reference tracks that are well-matched in loudness will make your life a lot easier throughout the mastering process. Besides that, spending some time critically listening to your reference tracks will calibrate your brain to the sonic conventions of the music genre you are working on, which is a great way to start any mastering session.

If you feel your reference tracks match in loudness, you can load in the track or multiple tracks you want to master. If you have just one track, load it on a new channel and adjust the volume of your reference tracks to match the loudness of your track. Apart from the psycho-acoustic relationship between volume and frequency, this can be difficult, as you are comparing your track to reference tracks that are probably already mastered. In that regard, don’t be discouraged if your reference tracks sound much more

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PART | II Home Studio Mastering Techniques controlled and clean than your own track. You can expect your track to measure up to them more and more, while you go through the remaining chapters of this part of the book. If you are mastering multiple tracks that are part of the same release, things get more complicated. Initially, load them up on different channels in the track order of the release. Create a group channel for them, so you can both apply processing on the tracks’ individual channels, as well as overall processing on the group channel. Then, match the loudness of your tracks by ear, taking into account the intended album flow of your release. For example, if your release contains four tracks played with a full band, and one quiet ballad with just piano and vocals, it is probably appropriate to turn the ballad down a little. The levels you choose at this point are only meant to provide an initial balance, and you will continue to tweak them throughout the process. Finally, use the reference track group channel to match the volume of the reference tracks to the tracks you want to master. When mastering multiple tracks, it is usually easier to pick one of them—the track that sonically represents your release best—to compare to your reference tracks instead of all of them. Matching the loudness of your reference tracks to your own tracks will enable you to make a relatively objective assessment of the sonic character of your tracks within the context of its genre, and will bring you in the right mindset to start your session. When everything is set up, you are ready to proceed.

Chapter 8

Restoration

8.1 Removing DC Offset and

8.4 Noise and Hiss

Unwanted Noises

8.5 Clicks and Crackles

8.2 DC Offset

8.6 Sibilance and Plosives

8.3 Ground Hum

8.1 REMOVING DC OFFSET AND UNWANTED NOISES

The first thing you will do in your mastering session is checking for DC offset and unwanted noises. This should be done at the start of the session, because the restoration techniques for removing DC

offset and unwanted noises in the mastering stage will become increasingly less effective as you apply more processing, especially in the case of plug-ins that influence dynamics, like limiters, compressors and saturators. In the restoration phase, you generally address the following five issues:

● DC offset

● ground hum

● noise and hiss

● clicks and crackles

● sibilance and plosives

In the remaining paragraphs of this chapter, we will discuss the appropriate restoration techniques to remove DC offset and unwanted noises. However, keep in mind that many restoration plug-ins introduce degradation and artifacts themselves, especially 85

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PART | II Home Studio Mastering Techniques when used in more extreme ways. Therefore, if any of these issues are present in your track, your best option is to go back to the mix and fix them there. This way you can just process the relevant parts and pieces in your mix, and avoid applying degrading processing techniques to your track as a whole. In case you are providing mastering as a service, you should ask for a remix and provide the mixing engineer with specific instructions for reducing the DC

offset or unwanted noises. Only if going back to your mix or asking for a remix is not possible, you should apply the restoration techniques described in this chapter during the mastering stage.

If this is indeed the case, make sure you only process the parts of your track that actually suffer when removing unwanted noises, as sometimes these kinds of issues are only present for a short dura-tion, or just in the left or right, or mid or side channel of your track.

“Keep in mind that many restoration plug-ins introduce degradation and artifacts, especially when used in more extreme ways.”

8.2 DC OFFSET

A track has DC offset when the average of all its samples amounts to a non-zero value. This phenomenon can be caused by many things, including low quality hardware, cheap soundcards, digital synthesizers that feature modifiable wavetable oscillators, and asymmetric distortion and saturation plug-ins. You can use the analysis tools in your DAW or audio editor to measure your track’s DC offset. A small amount of DC offset is usually unavoidable and generally not an issue, but if the amount is bigger—say a few percent or more—two problems may arise:

● Your track’s displacement from the zero line will result in less headroom, forcing you to push any plug-ins with a threshold harder than necessary. This may limit the maximum loudness you can achieve using a brickwall limiter.

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● You may hear a digital click when pressing play or stop in the middle of your track. These clicks are easiest to detect with good quality headphones, during a more quiet passage of your track.

If you experience one or both of these problems, there are two tools you can use to remove the DC offset from your track: a DC offset removal plug-in or a simple high pass filter. Which one you should use depends on whether the DC offset in your track is constant over time. If the DC offset is constant over time, this means that your track is shifted as a whole, which is usually obvious when you look at the positioning of the waveform. In this case you can best use a DC offset removal plug-in that compensates for the displacement by shifting the center of the waveform back to the zero line. However, because your track is a rendered mix which consists of many parts and pieces that may have had DC offset problems of their own, the chances are that your track’s DC offset is far from constant over time. In this case you can best use a high pass filter, set to a value just below the human audible range—say 20Hz or just below. This will compensate for the DC offset in a more dynamic and balanced way over the course of your track. Be careful though, if the sub-frequencies are important for your track, not to accidently remove these because of the roll-off curve of the high pass filter.

FIGURE 8.1 A Waveform With DC Offset

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8.3 GROUND HUM

Grounding issues in your home studio can cause a low frequency ground hum to be picked up during the recording stage. The exact frequency of the hum depends on the electric potential difference specific to your region. In Europe it will be a 50Hz tone, and in the United States it will be a 60Hz tone. If a ground hum was present in tracks that are processed with plug-ins during the mixing stage, it is possible that there are also harmonics (sometimes called overtones) present in your track, related to the hum frequency. In order to remove the ground hum, you can use a dehummer plug-in.

Most dehummer plug-ins are actually equalizer plug-ins with a series of steep peak filters, called notch filters. Usually, they feature standard presets for eliminating a 50Hz or 60Hz hum, and although using presets during mastering should be avoided, as an exception to this rule, dehummer presets are a good place to start.

However, as with all restoration plug-ins, in the mastering stage a dehummer should be used with care. First, to preserve the overall tonal balance of your track, you want to use as few notch filters as possible—ideally just one. And second, to avoid resonances, you don’t want the notch filters to cut too deep. To minimize side-effects, you should aim to reduce the hum until it blends in with your track, instead of trying to remove it completely.

8.4 NOISE AND HISS

As the result of using live-recorded instruments, or noisy gear or plug-ins, some tracks contain a steady amount of audible noise or hiss. Because many home studios have a high noise floor due to relatively poor insulation and computer fans, it is best to use a good quality pair of headphones for checking your track for this type of noise. If you find your track contains audible steady noise, you can use a de-noiser plug-in to reduce or eliminate it. The basic controls on a typical de-noiser plug-in are:

● learn or analyze

● reduction

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● threshold

● attack and release

● quality or resolution

● difference

For the best results using a de-noiser plug-in, you need a short sample of the noise specific to your track, so the plug-in can create a noise profile. If your track has a few seconds of headroom or tailroom, you might be able to find a short noise sample there, but sometimes you will have to go back to the mix. If you have found a suitable noise sample, you can use the de-noiser plug-in to generate a noise profile for you. This is usually done by playing the noise sample and then briefly turning the learn or analyze function on the plug-in on and off. As a last resort, if you were unsuccessful in retrieving a suitable noise sample, you can select one of the standard built-in noise profiles, or if such a feature is available, have your plug-in intelligently extract a noise profile from your track as a whole. Plug-ins that offer this extraction feature sometimes have an adaptive mode as well, that tweaks the noise profile over time based on the audio that’s going through the plug-in. Be somewhat conservative with this feature, however, because although it can be helpful, it can also lead to unpredictable results. Once the noise profile is set, usually represented as a graph that shows the frequency content of the noise, the best way to get a feel for the plug-in and its potential degrading effects is to set the reduction to its maximum value and then move the threshold slowly from its lowest to its highest value. First off, you might not hear any difference, but when moving the threshold up, you will gradually hear the noise reduction getting more extreme, until finally the degrading effect of the plug-in gets noticeable as well. If attack and release controls are available, you can use these to set how fast the plug-in reaches full noise reduction, and how fast it goes back to a neutral state. Choosing your actual settings will always be a compromise, when it comes to de-noising during the mastering stage. In that regard, your goal is not to eliminate the noise entirely, but to reduce it until it blends in with your track. Set the

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PART | II Home Studio Mastering Techniques FIGURE 8.2 Poul’s Studio—Leiden, The Netherlands (1) quality or resolution of the plug-in to the highest setting available, be conservative when setting the threshold, and aim for only a few dB of reduction. Many de-noiser plug-ins feature a difference control that lets you listen to the part of the signal that gets removed.

By using this feature you can make sure the de-noiser doesn’t do excessive harm. In addition, if the particular de-noising plug-in you are using has any features that let you control the amount of degradation and artifacts it is causing, it oftentimes pays off to play around with these. If you don’t get the results you are looking for without causing damage, the best thing might be to go back to the mix, or even re-record any noisy parts of your track.

8.5 CLICKS AND CRACKLES

Due to inconsistencies in the waveform that makes up your track, or software or hardware that has malfunctioned during the recording or mixing stage, your track may contain clicks and crackles.

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Similarly to detecting noise or hiss, it is best to use a good quality pair of headphones to ensure these sounds are not obscured by the noise floor of your home studio. Typically, digital clicks are short and high-pitched sounds with an intrusive character. Sometimes they are caused by an inexperienced mixing engineer who has not applied proper fades or crossfades when splitting or combining tracks in the mix, in which case it is especially helpful to go back to the mix. If the clicks are caused by malfunctioning hardware or software, your only choice might be to use a de-clicker plug-in.

Crackles are different in nature, and are usually the result of bro-ken instruments, amplifiers or microphones that were used during the recording stage. In addition, audio captured from a vinyl record or subtle distortion caused by accidentally clipping the signal path may be similar in nature. Both crackles and subtle distortion can be reduced with the use of a de-crackler plug-in. The basic controls on typical de-clicker and de-crackler plug-ins are:

● reduction

● threshold

● difference

Similar to using a de-noiser plug-in, the best way to get a feel for a de-clicker or de-crackler plug-in and its potentially degrading effect, is to move the threshold slowly from its lowest to its highest value, with the reduction set to its maximum. In addition to the overall degradation, you should pay special attention to the transients in your track, as these can suffer as a side-effect of the algorithms used, making your track sound weak. For your actual settings, be conservative when setting the threshold, aim for only a few dB of reduction and use the difference function to check that the plug-in does no excessive harm to your track. If you are just removing a few random digital clicks or sporadic crackling, process only the pieces of your track that need fixing. Some de-clicker plug-ins offer a click width feature or a special mode for periodic clicks that are evenly spread out over time. In addition, just like de-noiser plug-ins, both de-clicker and de-crackler plug-ins might

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PART | II Home Studio Mastering Techniques offer other features to minimize the degradation or artifacts caused by the plug-in. Playing around with these additional features can make a significant difference. Other more advanced restoration plug-ins will display an editable spectrogram, in which you can manually remove or reduce specific noises. Although these plug-ins can be a true lifesaver when it comes to sound design and broadcasting audio, use them with great care in the context of mastering, as they can cause obvious degradation and artifacts, especially in the case of tracks that contain many elements. If you do feel the need to use a spectrogram-based restoration plug-in, make sure you set its quality or resolution to the highest value possible, and reduce unwanted noises until they blend in with your track, instead of removing them completely. This will cause the least amount of degradation and artifacts.

8.6 SIBILANCE AND PLOSIVES

Recording and mixing vocals is one of the more difficult tasks when producing a song. There are two common problems that sometimes remain unaddressed or even get aggravated by less experienced vocalists and recoding and mixing engineers: sibilance and plosives. The term sibilance is used to describe speech sounds with a hiss-like character, such as: “s,” “sh,” “z” and “zh,”

which can disturb the balance of your vocal mix if they are too loud, and distract the listener. Plosives are speech sounds that produce a small burst of air, such as: “b” and “p,” which can lead to a sudden low end boost in a vocal track. It can be difficult to record a problem-free vocal track, but vocalists with a lot of studio experience will often make an effort to control these sounds during a recording, and when the recording is done in an acoustically treated room with the use of a pop-filter, it is certainly achievable.

However, even if care has been taken during the recording stage, and no excessive sibilant and plosives sounds are present in the raw recordings, these sounds can still surface when misusing compressors and equalizers during the mixing stage. If this is the case,

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it is best to go back to the mix and fix any issues there. However, if this is not possible, there are a few techniques that might reduce sibilance and plosives during the mastering stage.

“Even if care has been taken during the recording stage, and no excessive sibilant and plosives sounds are present in the raw recordings, these sounds can still surface when misusing compressors and equalizers during the mixing stage.”

Reducing sibilance is done with a de-esser plug-in, which is usually some type of compressor or limiter dedicated to detecting and reducing sibilant frequencies. However, many ordinary de-esser plug-ins are not suited for the purpose of mastering. They will be able to react to sibilant sounds by applying equalization to their internal side-chain, but they will still apply attenuation to the track as a whole. This scheme works fine when just processing a single vocal track, but during mastering you want a de-esser plug-in that not only reacts to just the sibilant frequencies, but also reduces just the sibilant frequencies, causing a minimal disturbance of the overall tonal balance of your track. So instead of a de-esser plug-in based on a regular compression algorithm with an internal side-chain, you want a de-esser plug-in based on either a multi-band compression algorithm or dynamic equalization algorithm.

The basic controls on a typical de-esser plug-in are:

● side-chain

● frequency

● range or reduction

● threshold

To set a de-esser plug-in, turn on the side-chain mode to listen to the internal detection signal and use the frequency control to isolate the sibilant sounds in your track, which will occur approximately between 3kHz and 7kHz. Then, turn off the side-chain mode again, set the range or reduction to its highest possible value,

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PART | II Home Studio Mastering Techniques and move the threshold up and down until the plug-in catches any disturbing sibilant sounds. You can decrease the range or reduction for a more subtle effect. If you don’t have a de-esser plug-in suited for mastering at your disposal, or if you are not sure about the inner workings of the de-esser plug-in with which you are familiar, you can recreate the effect yourself as well with a multi-band compressor or dynamic equalizer. First, use a band or filter to isolate the frequency range where the sibilance occurs, and set both the attack and release to an initial value of 0ms to make sure you will catch the sibilant sounds. Second, set the ratio to the highest value available and move the threshold up and down until the sibilance gets attenuated up to the point where it blends in with your track.

Finally, slightly increase the release to smooth out the gain reduction and possibly decrease the ratio for a more subtle effect. If you feel removing the sibilance with global processing has a negative effect on your track’s tonal balance or dynamics, your best option FIGURE 8.3 Poul’s Studio—Leiden, The Netherlands (2)

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95

is to reduce the sibilant sounds one by one, by automating the gain control of a high frequency peak or shelving filter.

Reducing plosives during the mastering stage can be more difficult than reducing sibilance, because the low end build-up that plosives produce occurs in the same frequency range as the kick-drum and bass guitar. Of course, it can’t hurt to try a multi-band compressor or dynamic equalizer first before going through them one by one, but the chances are that these plug-ins will react to other mix elements in the low end as well, affecting the dynamics and tonal balance of your track in a negative way. If this is the case, your best option is to automate the gain control of a low frequency peak, approximately between 100Hz and 200Hz. Obviously, in a mix that contains a kick-drum and bass guitar, this technique can disturb the low end of your track to a great extent. Therefore, go through the plosives one by one, and aim for an acceptable compromise between the volume of the plosives and the interruptions in the low end.

Chapter 9

Tonal Balance

9.1 Frequency Distribution

9.5 Low End Control

9.2 Frequency Analysis

9.6 Corrections and

9.3 Mastering Equalization

Improvements

Basics

9.7 Match Your Tracks

9.4 Optimizing Tonal Balance

9.8 Enhance Overall Timbre

9.1 FREQUENCY DISTRIBUTION

A song’s tonal balance can be described by the way the song’s frequency content is distributed along the audible frequency spectrum, and is sometimes referred to as the color of a track. As a result of this distribution, your track may, for example, sound dull or bright, or may have another characteristic tonal timbre like a muddy low end, a nasal mid section or a harsh high end. Although the appropriate tonal balance for a track very much depends on the track’s musical content as well as the artistic vision of the musicians and engineers involved, most popular genres of music benefit from a relatively transparent sound with clearly audible vocals.

To aid in discussing the subject of tonal balance with other musicians and engineers, there are a few different ways you can refer to specific frequency areas: frequency ranges, timbre and formants.

● First, the human audible frequency range from 20Hz to 20kHz can be subdivided into smaller frequency ranges.

Although these ranges are not officially established, you can 97

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PART | II Home Studio Mastering Techniques use Table 9.1 as a guideline (using values from the ISO stan-

dard for 1/3 octave equalizer).

● Second, you can describe the timbre that characterizes certain frequency ranges (see Table 9.2). In many cases the same fre-

quency range can represent both a positive or negative quality, depending on the production. Just like the frequency ranges, these specific descriptions and ranges are not officially established, and may vary per instrument and mix.

● And third, you can reference certain frequencies by using formants: frequencies that represent certain vowel sounds, regardless of their actual pitch. Table 9.3 lists both the first, second and third formant of various vowel sounds.

TABLE 9.1

Section

Range

Frequency

Low end

Sub low

20Hz - 80Hz

Low

80Hz - 200Hz

Mid section

Low mid

200Hz - 500Hz

Mid

500Hz - 2.5kHz

High mid

2.5kHz - 5kHz

High end

High

5kHz - 10kHz

Top high

10kHz - 20kHz

TABLE 9.2

Description

Frequency

Full vs boomy

20Hz - 200Hz

Warm vs muddy

250Hz - 500Hz

Nasal vs thick

750Hz - 1500Hz

Clear vs harsh

2kHz - 10kHz

Airy vs bright

15kHz - 20kHz

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TABLE 9.3

Vowel

Example

Formant 1, 2 and 3

/ow/

bought

570Hz

840Hz

2,410Hz

/oo/

boot

300Hz

870Hz

2,240Hz

/u/

foot

440Hz

1,020Hz

2,240Hz

/a/

hot

730Hz

1,090Hz

2,440Hz

/uh/

but

520Hz

1,190Hz

2,390Hz

/er/

bird

490Hz

1,350Hz

1,690Hz

/ae/

bat

660Hz

1,720Hz

2,410Hz

/e/

bet

530Hz

1,840Hz

2,480Hz

/i/

bit

390Hz

1,990Hz

2,550Hz

/iy/

beet

270Hz

2,290Hz

3,010Hz

In a professional production, generally the full human audible frequency range is represented. As you approach the high end of the spectrum, the frequencies gradually decrease in volume, which ideally results in a relatively smooth downwards curve, going from the low end to the high end of the frequency spectrum. It is the curve’s general slope, as well as its specific peaks and dips that give a track its unique tonal balance. If the curve’s general slope deviates too much from the conventions of the track’s music genre, however—meaning it might sound overly bright or dull, for example—it might not blend in well with other popular tracks in the genre, even if it does have a nice gradual shape. In addition, if there are peaks that rise too high or dips that run too deep along the curve, the tonal balance of your track may sound unnatural and have a resonant quality. Therefore, if you want your track to translate well to a wide variety of audio installations, you need to optimize your track’s tonal balance, while holding to the conventions of the track’s music genre. Because of the psycho-acoustic principles discussed earlier—tonal balance normalization, loudness bias

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PART | II Home Studio Mastering Techniques and ear fatigue ( refer to: 6.3 The Art of Listening)—this can be a challenging task. In the next paragraph you will learn how to make an objective analysis of your track with the use of a spectrum analyzer and the reference tracks you have selected.

“If you want your track to translate well to a wide variety of audio installations, you need to optimize your track’s tonal balance, while holding to the conventions of the track’s music genre.”

9.2 FREQUENCY ANALYSIS

By using a spectrum analyzer, you can get an objective but general representation of the frequency distribution of your track. These days, most mastering equalizer plug-ins have a built-in spectrum analyzer graph that smoothly reacts to the audio that is fed through the plug-in in real-time. (There is one included in the Noiz-Lab LE

Equalizer plug-in that comes with this book). The graph’s X-axis shows the human audible frequency range from 20Hz to 20kHz, and the graph’s Y-axis shows the volume range up to 0dB “full scale”, which is where audio clips and distorts in the digital realm.

Watching the graph while playing your track can give you a general indication of the frequency distribution and potential problems with the tonal balance. Also, it can be very insightful to compare the graph of your track to those of your reference tracks. However, you shouldn’t rely too much on spectrum analysis for three reasons:

● Spectrum analyzer graphs are not very precise in the low end, which is unfortunately where you need the most help when it comes to mastering in a home studio.

● Tonal balance conventions differ greatly across and even within music genres, so there is no such thing as an ideal distribution curve for which you can strive.

● Apart from certain technical considerations and conventions, you should mostly base your decision-making on taste and artistic vision, something with which a spectrum analyzer won’t help you.

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If the real-time spectrum analysis of your track doesn’t show any obvious flaws, like gaps or very steep peaks or dips, it is time to put your track to the test for real, using the reference tracks you have selected. First though, listen to the differences in overall tonal balance between just your reference tracks, by instantly switching between them during playback. Make sure you have matched their loudness by ear, otherwise this exercise will only complicate the mastering process ( refer to: 7.6 Setting Up Your Mastering Project). If you listen carefully, you might notice that your reference tracks have distinctive differences in tonal balance, even though they are from the same music genre. This can have many reasons, varying from a specific character or timbre imposed on the production by a certain instrument, amplifier or microphone during the recording stage, to technical considerations and creative choices during the mixing stage. Once you have identified the differences between the reference tracks, compare them one by one to your own track. Try to define any problems and points of improvements regarding the tonal balance, that prevent your track from blending in with the reference tracks. If the reference tracks are very different from each other, pick one of them as a main reference, instead of aiming for a tonal balance that falls somewhere in the middle. After objectively defining the problems and points of improvement with the help of a spectrum analyzer and your FIGURE 9.1 The Spectrum Graph of the Noiz-Lab LE Equalizer

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PART | II Home Studio Mastering Techniques reference tracks, you are ready to optimize the tonal balance of your track. However, because an equalizer is used somewhat differently during mastering than during mixing, it is useful to know what general good practices to hold to, regarding mastering.

9.3 MASTERING EQUALIZATION BASICS

For most basic equalization tasks during mastering, you will use a parametric equalizer. This type of equalizer gives you full and continuous control over its parameters, allowing you to make surgical corrections and improvements. Of course, you can also use them for more general adjustments to enhance the overall timbre of your track. However, for this task engineers often use much simpler equalizers modeled on analog gear. But these emulations usually have analog-like limitations that are not required in the digital realm. Using a parametric equalizer with analog model-ing, or with the option to switch between transparent-style and character-style equalization, gives you much more control. Therefore, in general make sure you think twice whenever you notice you are choosing one plug-in over another just because the user interface looks more vintage. In default mode, most parametric equalizers show a graph with a straight line in the middle, with several handles. The graph’s X-axis shows the human audible frequency range from 20Hz to 20kHz, and the graph’s Y-axis shows the volume in dB, relative to the zero line in the middle. If you drag the handles around, the graph shows you how the plug-in affects the frequency distribution. Many parametric equalizer plug-ins also show a spectrum analyzer graph in the same display, reacting to the output signal of the plug-in, so you can see in real-time how the changes you make affect the frequency distribution of your track. Remember that in this case, oftentimes the graph’s Y-axis just shows the volume relative to the zero line in the middle, and not the dB full-scale range of the spectrum analysis. (This combination of equalizer and spectrum analyzer is also found in the Noiz-Lab LE Equalizer plug-in that comes with this book).

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Every handle of the equalizer represents a filter that can be either one of the following types:

● high pass (low cut)

● low pass (high cut)

● band pass

● peak (bell)

● high shelf

● low shelf

For the purpose of adjusting the tonal balance with a parametric equalizer during the mastering stage, you will mainly use peak and shelving filters, because these filter types allow you to create very gentle, natural sounding slopes. In addition, especially when it comes to home studio mastering, you might use one high pass filter to control the low end of your track in the sub region. Most filter types will give you control over one or more of the following three parameters: frequency (Hz), quality (Q) and volume (dB):

● For peak filters the frequency defines the center frequency of the filter. For shelving filters it usually defines the frequency halfway up or down the slope, and simultaneously the point around which the curve changes its steepness as the quality is changed. Most filters let you set their frequency within the bounds of the human audible frequency range of 20Hz to 20kHz. Both for spectrum analyzers and equalizer graphs this range is divided into octaves so it is relatable from a musical perspective. For this reason the range is scaled logarithmically instead of linearly, which results in any octave—a doubling of the frequency—occupying the same width on the graph.

● The quality defines the steepness of a filter. For peak and shelving filters, the higher the quality value, the steeper the filter is, causing them to have a more resonant character. For peak filters the quality value is usually inversely related to the bandwidth that is affected by the filter, which is defined as a number of octaves: quality = 1 / bandwidth. Be aware though, that not

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PART | II Home Studio Mastering Techniques all equalizers use the same Q calculations. Being able to see the shape of the filter’s frequency response can be extremely helpful in that regard. For high pass, low pass and band pass filters, the quality is not always available. If it is, however, it controls the resonance of the filter, which is not of much use for mastering purposes. In some cases you can set the order of a high pass, low pass or band pass filters, which is another way of setting their slope’s steepness in decibel per octave.

● For peak filters the volume defines the amount in decibel by which the center frequency of the filter is cut or boosted. For shelving filters it defines the target height of the shelf. High pass, low pass or band pass filters do not have a volume feature.

When setting the parameters of any filter type, you should keep in mind that during mastering you are processing your track as a whole. Therefore, your approach should be subtle. During mixing, you might make big changes to the tonal balance of single mix elements, to get rid of unwanted frequencies or combine them with other elements in the mix. However, during mastering it is best to avoid steep filter curves and big boosts or cuts. If your track is reasonably balanced, cuts of 0.5dB to 1dB will generally be enough. For less balanced mixes you occasionally might need to go up to 2dB or maybe even 3dB, but if bigger cuts or boosts FIGURE 9.2 Filter Controls

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105

are necessary it is probably best to go back to the mix. In addition, instead of applying static equalizer settings to your track as a whole, you might find that it sounds more natural to automate the equalizer’s settings over the course of your track, or to only apply the equalization to the mid or side, or left or right channel of your track. As a rule of thumb: when fixing specific problems, use transparent-style, steep and surgical filtering, mostly subtractive; when enhancing overall timbre, use character-style, gradual and global filtering, mostly additive. Let’s give it a try.

9.4 OPTIMIZING TONAL BALANCE

Optimizing the tonal balance of your track is one of the more difficult tasks you face when mastering your own music, especially when you are mastering multiple tracks that are part of the same release. Because of the psycho-acoustic relationship between volume and frequency, it is easy to fool yourself: changing a track’s tonal balance will change your perception of the track’s volume, and changing a track’s volume will change your perception of the track’s tonal balance. For this reason, it is very important to keep level matching your tracks as you optimize and match them in terms of tonal balance. In any case, whether you are mastering a single track or a full album release, to keep this undertaking manageable, it is best to divide the equalization process into several distinct steps:

● Low end control

● Corrections and improvements

● Match your tracks (if more than one)

● Enhance overall timbre

In the next four paragraphs we will discuss these steps in detail.

Remember, however, that it is sometimes more effective to apply equalization multiple times at different places in your mastering chain, instead of trying to fully optimize the tonal balance of your track, before applying any other processing. For example, fixing

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PART | II Home Studio Mastering Techniques serious issues gets increasingly more challenging as you apply more processing, so this is best done early on in the process. This will also guarantee that any compressors and limiters you might use later in the chain will not react to any frequency content you don’t want in your track in the first place. On the other hand, many types of plug-ins will influence the general tonal balance of your track in some way, and it is common practice to use an equalizer right after to compensate for any unwanted coloration, even multiple times in your chain. Also, with many plug-ins influencing the tonal balance, enhancing the overall timbre of your track is usually more effective at the very end of the mastering process, just before maximizing the loudness of your track. Try to think logically about all this, and base your processing on an as objective as possible assessment of what your track needs.

“Changing a track’s tonal balance will change your perception of the track’s volume, and changing a track’s volume will change your perception of the track’s tonal balance.”

9.5 LOW END CONTROL

Professional studios that are acoustically treated and equipped with high-quality speakers usually have a well-defined low end frequency response. This makes it possible to accurately analyze the low end and to make specific optimizations. In a home studio, however, to a certain degree you are flying blind. Therefore, especially for novice engineers, it is sometimes better to be safe than sorry, and to use a transparent-style high pass filter to remove a part of the sub-frequencies. Professional mastering engineers might only apply a high pass filter in this fashion to solve DC offset problems ( refer to: 8.2 DC Offset), and consider this bad practice. However, working in a home studio, this technique can save you some trouble, at least when you choose your filter settings carefully. Use a good pair of headphones that go down to 20Hz, to

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107

compare your track’s low end to that of your reference tracks. Turn up the volume, as this will make the low end become relatively more pronounced ( refer to: 6.3 The Art of Listening). If you notice that your track contains frequencies that extend all the way down the spectrum, and your reference tracks do not, you can use a high pass filter to compensate for this. To get a feel for this type of filter, choose the steepest filter slope available, and slowly move the frequency up, starting at 20Hz. On decent headphones you should clearly hear the low end gradually disappear. If you are working on a genre that depends heavily on its sub- frequencies, like for example Hip-hop or EDM, you probably want to set the frequency somewhere between 20Hz and 30Hz, preserving the low end as much as possible. On the other hand, if you are working on music where the low end is less important, like a singer-songwriter track, you might want to set it a little higher, to about 40Hz or even 50Hz.

Make sure, though, that you preserve the warmth of your track.

After finding the appropriate frequency, you can choose a more FIGURE 9.3 Marc’s Home Studio—Leiden, The Netherlands (1)

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PART | II Home Studio Mastering Techniques gradual slope for a more subtle roll-off. Carefully double-check your settings on your speakers with the volume turned up. Be careful with this technique, though, for two specific reasons. First, it is possible that both your headphones and speakers do not go low enough to make an accurate judgment about the low end, in which case you can’t really know what you are doing. And second, it is possible you mistake the low end just being louder or less controlled, for the low end actually extending further down the spectrum compared to your reference tracks. Don’t worry about that too much now, however, as we will get back to optimizing the low end of your tracks several times in the remaining chapters of this part of the book.

9.6 CORRECTIONS AND IMPROVEMENTS

To make specific corrections and improvements, you need to be able to identify and target specific frequency areas. In some cases this is straightforward. For example, if your track sounds overly bright or dull compared to your reference tracks, using a smooth and subtle low or high shelving filter will probably solve your problems quickly. However, sometimes it is hard to determine exactly why your track doesn’t blend in well with your reference tracks, and which frequency areas are to blame. In these cases you can use a technique called focusing. Begin by comparing your track to your reference tracks, and concentrate by ear on a specific frequency area or mix element that needs work. Then, use a transparent-style peak filter to create an extreme boost of about 10dB. Choose a relatively narrow Q, without the filter getting overly resonant. If you move this peak filter slowly to the left and right along the frequency spectrum, the specific area that needs work will get boosted at a certain frequency. Be aware, though, that a big boost can cause your signal to clip on certain frequencies and that the resulting distortion might make these frequencies appear to be problematic, while they are not. Once you have identified the target frequency, it is good practice to see if the issue is

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only present in the left or right, or mid or side channel, so you can be more surgical if needed. Then, set the Q to a value at which the relevant frequency area is affected sufficiently, without unnecessarily harming any mix elements around it. Then cut or boost the frequency by setting the filter’s volume. Be aware, however, that your brain quickly normalizes changes in the tonal balance, so the big boost you used to focus the filter has most likely caused some psycho-acoustic issues. Therefore, this is a crucial moment to fall back on your reference tracks, especially for novice engineers. So, put the filter’s volume back to 0dB, and compare your track to your reference tracks again. Then, set the volume to cut or boost the relevant frequency area. A cut or boost of 1dB or 2dB will usually be more than enough. If you feel you don’t get the intended results, although you are already cutting or boosting a lot, the problem might exist in harmonic frequencies related to the target frequency as well. Harmonic frequencies are usually multiples of the target frequency, but in this case it is possible that the target frequency you have selected is actually a harmonic frequency itself, making other harmonics difficult to find. Therefore, using the focusing technique and finding these related frequencies purely by ear is probably easier. If you find the issue is indeed spread over different frequencies, it is usually better to apply subtle filtering to all of these areas, instead of aggressive filtering to the target frequency alone. Using the focusing technique to make any necessary corrections and improvements to specific target frequencies and sometimes their harmonics as well, should make your track blend in well with your reference tracks, at least in terms of tonal balance.

9.7 MATCH YOUR TRACKS

If you are optimizing the tonal balance of multiple tracks that are part of the same release, it is usually easiest to start with the track that is most representative for the release. This is usually the track with the most complete arrangement in terms of the number of instruments recorded or layers in the mix, or the track that

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PART | II Home Studio Mastering Techniques is chosen to promote the release, perhaps in the form of a music video. Make the necessary corrections and improvements to this track, comparing it to your reference tracks, until you find a tonal balance with which you are happy. From that point on, use this first track as your main reference, only occasionally falling back on your actual reference tracks. Going through the remaining tracks one by one, you might notice that certain improvements you make to an individual track actually cause a loss of coherence in the release as a whole. How you approach this problem is entirely up to your vision in regards to the release. If it is an actual physical EP or album, and you expect your audience to listen to it from start to end on a quality hi-fi system at home, you probably want the release to be as coherent as possible and have excellent album flow. If there is one particular track used to promote the release, you might want to create a fully optimized master as well, for the sole purpose of releasing the track on its own. On the other hand, if it is an online release, and you expect your audience to skip FIGURE 9.4 Marc’s Home Studio—Leiden, The Netherlands (2)

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111

through it to their favorite track on a laptop or smartphone, you probably want to optimize the individual tracks to a greater extent, even if it means the release as a whole is less coherent. Either way, go back and forth between the different tracks, adjusting and level matching them, until you find a compromise with which you are happy. Especially for novice engineers, this can take a while.

However, as you gain more experience, the relationship between loudness and tonal balance will become more intuitive. When you have achieved a proper tonal balance that ties all your tracks together, while mostly maintaining their individual optimizations, you have completed one of the most difficult parts of the mastering process.

9.8 ENHANCE OVERALL TIMBRE

After you have optimized and matched the tonal balance of your tracks, there are three subtle processing techniques you can apply to enhance the overall timbre of your release: saturation, convolution and character equalization. If your tracks are very different from each other, it is probably best to apply these techniques to your tracks individually. However, if your release is very consistent, you can safely apply them to all your tracks at once, by using the group channel to which you have routed your tracks.

● In essence, saturation is clipping your signal in a controlled way. Basic saturation plug-ins, driven hard, will therefore make your track distort as if it was fed into a guitar amplifier. For mastering purposes, though, there are much more elaborate saturation plug-ins that emulate the subtle saturation of analog gear, such as tape machines and studio consoles. As most saturation plug-ins for mastering are carefully designed to have a specific flavor, it can pay off to try as many different ones as you can get your hands on. Be aware, though, that applying saturation will influence the volume of your track. If you do not level match your tracks by ear when doing a before and after comparison, it is very easy to fool yourself.

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PART | II Home Studio Mastering Techniques

● Although convolution is typically associated with reverberation ( Refer to: 10.7 Convolution Reverb), it is also a great technique to add a certain timbre or flavor to your track as a whole.

To apply this technique, you need a convolution plug-in that is able to load custom IRs (impulse responses)—for example, a convolution reverb plug-in. However, instead of using an IR

of a very reverberant space, you use an IR of a professional studio room with lively and natural acoustics. Such an IR will add the coloration and natural acoustics of the studio room to your track, which can make your track come alive, especially if you are working on a purely in-the-box production. It is important to listen to the fully wet signal when using this technique, to make sure it doesn’t add any unwanted frequencies. Some convolution plug-ins feature a high pass and low pass filter to shape the wet signal, which is especially handy for eliminating rumble in the low end from which many IRs suffer. If you want a more flexible way to shape the tonal balance of the wet signal, you can choose to load the convolution plug-in on a separate bus and place a parametric equalizer right after it.

● You can have full control over the coloration you add to your track using a character equalizer. However, although most character equalizers apply some type of saturation, the coloration it adds will usually be far less pronounced compared to the saturation or convolution techniques detailed previously. In that regard, a character equalizer is mostly useful if you are looking for a subtle effect, or if you just prefer its sound to a regular equalizer when making smooth adjustments to the overall tonal balance of your track. Don’t forget, though, that your brain constantly keeps normalizing the listening experience. So, for example, adding a little more highs will usually sound relatively better, whether your track benefits from it or not. Therefore, you might repeatedly subtly boost the highs over the course of the mastering session, making your track sound overly bright, without realizing it until the day after. To avoid fooling yourself like this, make sure you keep comparing your track to your reference tracks.

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As discussed earlier in this chapter, because many plug-ins you might use during the mastering process will influence the tonal balance of your tracks, it is usually best to apply these enhancement techniques in the final stage of the mastering session, just before maximizing the loudness of your tracks. Especially when using moderate to heavy compression in your session, these enhancement techniques will usually sound better and much more controlled when applied after reducing the dynamics of your tracks.

However, although the chapters in this part of the book follow a workflow that is generally considered appropriate for any basic mastering session, experimenting with the order and possible repetition of equalization and compression techniques can lead to very controlled and balanced masters.

Chapter 10

Width and Depth

10.1 Stereo Image and

10.4 Overall Mid-Side Stereo

Reverberation

Enhancement

10.2 Stereo Imaging

10.5 Reverberation Techniques

Techniques

10.6 Algorithmic Reverb

10.3 Narrowing the Low End

10.7 Convolution Reverb

10.1 STEREO IMAGE AND REVERBERATION

The human brain has evolved over time to accurately pinpoint the distance and direction of sounds, which is why the width and depth of a track play a big role in the listening experience. A track’s width can be described by the way different mix elements are positioned in the stereo image. Some tracks are fully mono, meaning there is no difference between the left and right signal, while others have many instruments positioned off-center to create room in the mix or just out of creative considerations. A track’s depth can be described by the reverberation or acoustic timbre it contains. Some tracks sound very direct and dry, while others sound very roomy or spacious. The amount of width and depth you create in your track should mostly be based on the artistic vision of the musicians and engineers involved. Do take in account, however, that the appropriate width and depth vary across different music genres to a certain degree. Therefore, it is wise to not deviate too much from your reference tracks.

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PART | II Home Studio Mastering Techniques To understand the relatively subtle role of mastering in enhancing width and depth, it is helpful to know that the stereo image and acoustic timbre of a track are initially created during the recording stage and usually optimized later on during the mixing stage.

If most instruments are close-miked and recorded mono in a dry room, the recordings will be centered and direct. It will be easy to control the width and depth later on in the mixing stage, but challenging to make the recording sound expressive and natural.

If, on the other hand, most instruments are recorded from a greater distance, in stereo or with multiple microphones positioned in a lively room, the recordings might already have a distinct stereo image and acoustical timbre. It will be much harder to control the width and depth later on in the mixing stage, but if the recordings are done properly, they will already sound sufficiently expressive and natural. In any case, the general width and depth of any track can best be optimized before the mastering stage, because while FIGURE 10.1 Lodewijk en Olivier’s Studio—Warmond, The Netherlands (1)

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117

processing your track as a whole you have limited control over its stereo image and acoustic timbre. However, there are several stereo imaging and reverberation techniques you can apply during mastering that will subtly enhance the overall impact of your track, in slightly different ways than you are able to do during mixing. If you are mastering multiple tracks that are part of the same release and you want to create good album flow, these techniques can also help to match the tracks’ width and depth to create a consistent listening experience.

10.2 STEREO IMAGING TECHNIQUES

In the context of mastering, stereo imaging is the general term for processing techniques that enable you to adjust the width of the stereo image. However, not all stereo imaging plug-ins are well suited for mastering, as they can cause serious balance and phase correlation issues when applied to your track as a whole ( refer to: 5.5 Width and Depth). Therefore, especially for novice engineers, it is best to stick to summing-based, and mid-side-based stereo imaging techniques. Stay away from any delay-based and phase-based stereo imaging techniques as much as possible, unless you have a very specific reason to use them and know how to check for potential issues with a correlation meter or vectorscope. In the next two paragraphs, we will discuss the following two techniques to optimize the stereo image during mastering:

● Summing-based stereo imaging, for narrowing the low end

● Mid-side-based stereo imaging, for overall stereo enhancement Although they are subtle, combining these two techniques can be very powerful. Remember, however, that if the stereo image of your track was already optimized during the mixing stage, it is easy to do more harm than good. After all, your brain constantly keeps normalizing the listening experience, so enhancing the stereo image a little will usually sound relatively better, whether it is the right thing to do or not. Therefore, especially with boosting

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PART | II Home Studio Mastering Techniques side channel frequencies, you might repeatedly make enhance-ments over the course of the mastering session that are far too extreme when combined, without you realizing. To avoid this, make sure you keep comparing your track to your reference tracks on a regular basis. In addition, while experimenting with these techniques, keep in mind that most professionally produced tracks have a stereo image that starts relatively narrow in the low end, and gets wider as you go up the frequency spectrum. Now, let’s try these two stereo imaging techniques.

”Stay away from any delay-based and phase-based stereo imaging techniques as much as possible, unless you have a very specific reason to use them and know how to check for potential issues with a correlation meter or vectorscope.”

10.3 NARROWING THE LOW END

When transitioning from fully stereo to fully mono, using a regular summing-based stereo imager, you can actually hear the stereo elements in your mix moving closer to the center of the stereo image. This movement is the result of the left channel of your track gradually being mixed in with the right channel of your track, and vice versa. With a multi-band summing-based stereo imager, you can independently narrow the stereo image of different frequency ranges. During mastering, this technique is mainly used to create a tighter and more defined low end, and has little use higher up the frequency spectrum. To determine if any low end stereo width adjustments are necessary, first compare your track to your reference tracks. Because you are working on the low end of your track, it is best to turn up the volume and use a decent pair of headphones to make sure you know what’s going on all the way down the audible frequency spectrum. If you feel like your track could benefit from a tighter low end, solo the lowest band of your multi-band stereo imager and slowly increase the band’s frequency, until

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you find the point at which the bass instruments in your track, like the kick-drum and bass guitar, transition into the lead instruments, like the guitar, piano or synth. This will usually be somewhere between 100Hz and 200Hz. Make sure you set the frequency of the band to only affect the bass instruments of your track, and not the lead instruments or vocals. Then, to get a feel for the plug-in, use the width control of the plug-in to gradually make the low end fully mono, and then back to fully stereo again. If you don’t hear much difference, the low end of your track is already practically mono, and you have no need for this technique. However, if the low end of your track contains a lot of stereo information, the effect will be obvious. In that case, you want to un-solo the low band of the plug-in and experiment with narrowing its stereo image. In many cases a little goes a long way, and just slightly narrowing the low end stereo image can tighten up your track and create a more defined low end. Taking it too far, on the other hand, can make your track sound weak, so make sure to fall back on your reference tracks regularly.

FIGURE 10.2 Stereo-to-Mono Graph of the Noiz-Lab LE Monomizer

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PART | II Home Studio Mastering Techniques

10.4 OVERALL MID-SIDE STEREO ENHANCEMENT

A tight and well-defined low end provides a great foundation for enhancing the mid section and high end stereo image of your track.

For mastering purposes, it is best to use a mid-side-based stereo imaging plug-in, to minimize the risk of doing any harm to your track. Such a plug-in will not so much move elements around the stereo image, but instead will allow you to independently control the frequency distribution of the mid and side channel of your track. Although the multi-band stereo imager plug-in you have used for narrowing the low end of your track might also have the ability to widen the stereo image of a band using mid-side processing, a parametric mid-side equalizer will produce much more subtle and balanced results because you have full control over the shape of your filters. To use a mid-side equalizer for stereo imaging, it is important to know what audio information the mid-side channels actually contain:

● The mid channel is equivalent to a mono version of your track: mid = (left + right) / 2. Any audio information in the left and right channels that is out of phase will be reduced in volume or canceled out. All remaining off-center panned audio information will move to the center of the stereo image.

● The side channel contains the audio information from the left and right channels that is different in volume or out of phase: side = (left − right) / 2. Usually, this means the side channel will contain parts of off-center panned and mid-side recorded elements, as well as stereo effects like reverbs and quadraphonic choruses and flangers.

To determine if your track needs mid-side stereo enhancement, compare your track to your reference tracks. In addition, it can be very educational to compare the separate mid and side channels of your track to those of your reference tracks, as this will tell you a lot about how the engineers that produced these tracks utilized the stereo image, and to which degree they made their tracks mono

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compatible. If you feel your track lacks a certain width or could be more engaging, experiment with boosting certain frequency areas in the side channel, with the goal to enhance overall warmth or clarity, or whatever timbre you feel can improve your track. Just like with general equalization during mastering, make sure you use peak and shelving filters with smooth curves, and only make subtle boosts ( refer to: 9.3 Mastering Equalization Basics). If you are unsure how to choose your settings, remember that most professionally produced tracks have a stereo image that starts narrow in the low end, and gets wider as you go up the frequency spectrum. Therefore, it might help to use subtle high shelving filters to cut some highs in the mid channel, and boost some highs in the side channel. This will exaggerate the high frequency content of any stereo effects and creatively panned high frequency elements, and might make your track sound wider and more engaging. In addition, you could do the opposite in the low end: cut some lows in the side channel and boost some lows in the mid channel. This might give your track a little more power and stability. If you like the sound of this technique, but you feel the overall frequency distribution of your track becomes less balanced, you can use a regular stereo equalizer to compensate for any unwanted coloration. It is also possible that your track does not contain any relevant side information, or even none at all. This will make mid-side-based techniques ineffective. If this is the case, it is best to use one of the reverberation techniques described in the next two paragraphs, to create new stereo information, as opposed to using a different kind of stereo imaging plug-in with a more destructive algorithm.

10.5 REVERBERATION TECHNIQUES

Reverberation is the result of sound decaying over time, while reflecting off the walls, floor and ceiling of a room. The character of the reverberation is determined by the size and shape of the room, and the building materials used ( refer to: 2.2 Room Characteristics). Reverberation can exist naturally in a production due

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PART | II Home Studio Mastering Techniques to the way it was recorded, or it can be artificially added in the mixing stage. Traditionally, in the late 1940s and 1950s, artificial reverberation was created by playing a production in a physical reverb-chamber and then re-recording the signal. Soon after, in the late 1950s, the more convenient plate reverbs and spring reverbs came into use. In the 1970s, digital hardware reverbs became available, and with the commercialization of DAW software, starting in the 1990s, the first algorithmic reverb plug-ins emerged.

Finally, made possible by the steady increase of CPU power over the years, the first real-time convolution reverb plug-ins hit the market around the year 2000. Nowadays, there are so many different reverb plug-ins available, it is sometimes hard to choose. As a starting point in the context of mastering, it is important to understand the difference between algorithmic and convolution-based plug-ins.

Algorithmic reverbs basically consist of a number of delays with feedback, working together in parallel to produce a reverb tail of arbitrary length. Although there are certainly high-quality algorithmic reverb plug-ins out there, they usually sound somewhat clean and artificial. Convolution reverbs, on the other hand, use an IR (impulse response), recorded on location, to impose the reverberation and character of an actual room on to your track. Convolution reverbs sound much more natural and lively than algorithmic reverbs, but are much less flexible. Both reverb types can be used for the purpose of mastering. However, because their inner workings are so fundamentally different, deciding which one you should use depends entirely on what you are trying to achieve. For example, if you are mastering a track that is produced entirely in-the-box, or recorded only with close-mike recording techniques, a convolution reverb might help to create a more lively and natural sound. On the other hand, if you are mastering a track that just needs a little more depth in a subtle and transparent way, a high-quality algorithmic reverb might be your best bet. In the next two paragraphs, we will discuss the

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following two techniques to optimize the acoustic timbre of your track during mastering:

● Algorithmic reverb, for creating depth in a transparent and surgical manner

● Convolution reverb, for creating depth in a natural and charac-terful manner

If you are mastering several tracks that are part of the same release, and some of them sound much drier than others, using a reverb plug-in can tie your release together. Adding additional reverb is easy to overdo, though, causing your tracks to lose power and definition. So make sure to fall back on your reference tracks regularly, especially if you apply the reverb to your release as a whole.

“If you are mastering a track that is produced entirely in-the-box, or recorded only with close-mike recording techniques, a convolution reverb might help to create a more lively and natural sound.”

10.6 ALGORITHMIC REVERB

If you feel your track, or any particular mix element in your track, lacks depth, you can use an algorithmic reverb plug-in to enhance its acoustical timbre in a transparent and potentially surgical way.

Make sure, though, that the reverb plug-in you use is of high quality and does not have a metallic or harsh character. In addition, if you want to surgically target a specific mix element in your track, you want a reverb plug-in with a built-in parametric equalizer, preferably with mid-side support. If your reverb plug-in lacks these features, you could put the reverb on a separate effects channel, like you would do in a mix, preceded or followed by an equalizer.

However, in a typical mastering project, oftentimes this makes it more difficult to manage your plug-in chain. After all, in most cases you want to send your track to the next plug-in in your chain,

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PART | II Home Studio Mastering Techniques fully processed, instead of sending the wet reverb signal directly to your master bus, bypassing the rest of your plug-in chain. If you do find a convenient way to set up a reverb effects channel in your mastering session, make sure to put any mid-side-based equalizers on your reverb channel before the reverb plug-in, as many algorithmic reverbs will cloud the phase relationship between the left and right channel of your track, making any mid-side processing afterwards far less defined. The basic features of an algorithmic reverb are: reverb time, room size, dampening, pre-delay and stereo width.

● The reverb time lets you adjust the length of the reverb tail. As the reverb tail of algorithmic reverbs decays exponentially, and theoretically never dies out completely, the reverb time is usually based on the RT60 standard, which means it indicates the time it takes for the reverb signal to drop 60dB. For the purpose of mastering, usually a relatively short reverb time is used.

● The room size in high quality algorithmic reverb plug-ins lets you adjust the distribution of the reverb’s early reflections, actually mimicking different room sizes. However, in many plug-ins of lower quality, the term room size is used interchangeably with the term reverb time, referring to the length of the reverb tail as well. For the purpose of mastering, usually a relatively small room size is used.

● The dampening reduces the high frequency content of the reverb over time. This is usually done by integrating a low pass filter in the feedback loop of the reverb’s delay elements. Little or no dampening mimics a room with hard surfaces like concrete or brick, while increased dampening mimics a room with soft surfaces like plaster or wood. Increasing the dampening on an algorithmic reverb can reduce the plug-in’s potential metallic and harsh character.

● The pre-delay lets you delay the reverb signal, allowing you to separate the wet and dry signal from each other. In practice, increasing the pre-delay enables you to apply reverb in a

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very subtle manner, even using longer delay times, as well as to retain the directness of your track and to preserve its transients.

● The stereo width lets you adjust the width of the reverb’s stereo image. By default, most algorithmic reverbs are fully stereo, both for stereo and mono source material, and let you gradually collapse the reverb signal to mono. For the purpose of mastering, it is usually best to keep the reverb fully stereo. However, if your track is mostly mono to start with or the reverb is distracting, it might sound more natural to narrow the stereo image somewhat, so the added reverb doesn’t become overly obvious.

Applying overall depth to your track is fairly straightforward.

Start by soloing and turning up the reverb channel, and setting the length of the reverb tail using the reverb time and possibly the room size control. For the purpose of mastering, you usually want to simulate a relatively small room, like a studio or small concert FIGURE 10.3 Lodewijk en Olivier’s Studio—Warmond, The Netherlands (2)

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PART | II Home Studio Mastering Techniques hall. Use the dampening to make the reverb sound nice and warm, and the plug-in’s built-in equalizer to shape the reverb’s overall tonal balance or to remove any distracting resonances. If you are happy with the length and character of the reverb signal, un-solo it, but keep the reverb turned up. If you feel like the reverb is making your track sound too distant or washy, or that it makes the transients in your track lose definition, increase the pre-delay a little to pull the dry signal to the front. Then, turn the reverb channel all the way down and slowly mix it in again, to add the intended overall depth to your track. If the reverb sounds distracting, use the stereo width to narrow the stereo image, or experiment with applying it only to the mid or side channel.

Applying reverb to a specific mix element on the other hand, like the snare-drum, lead vocals or lead guitars, can be a little more challenging. Start by determining whether you want to affect the entire track or just the mid or side channel. For example, the snare-drum and lead vocals are usually panned dead-center, so if you want to be surgical in your approach, you might only want to process the mid channel. In addition, you might want to use the built-in equalizer to remove some low frequencies, and possibly some high frequencies as well, to make sure the reverb is mostly affecting the snare-drum or lead vocals, or any specific mix element you are trying to target. The lead guitars, on the other hand, might be hard-panned to the sides. In this case you might only want to process the side channel and remove some high frequencies from the reverb signal, to stay away from the lead vocals.

In general, when applying reverb to specific mix elements during mastering, it is important to listen carefully to any reverberation already present in your track, as you want the reverb you add to blend in naturally.

10.7 CONVOLUTION REVERB

Because a convolution reverb imposes the acoustic properties of an actual space onto your track, it is ideal for enhancing the overall

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acoustical timbre of your track in a very natural way. Therefore, this type of reverb works great for bringing mostly in-the-box or close-miked recorded tracks to life. However, you are fully reliant on the quality and type of the impulse responses you use, which can vary a great deal. In that regard, it can pay off to try as many IR libraries as you can find and to put together your own collection of favorite IRs. For the purpose of mastering, these should mainly be high quality IRs with relatively short tails like high-end studios, small rooms and concert halls. In addition, it is very helpful to have both mono (one channel) and stereo (two channels) IRs. Mono IRs are great when you just want to add character and depth without affecting the stereo image too much. Stereo IRs, on the other hand, are great if you want to add a natural and lively acoustic timbre to a track, or to create a mono-to-stereo effect for tracks that are mostly mono. Besides regular stereo convolution, some convolution reverbs offer true stereo convolution as well (four channels), which more closely mimics actual stereo room reflections, and can sound very pleasing. However, during mastering, reverb is usually added in such a subtle way that it may be difficult to discern between stereo and true stereo convolution.

In general, convolution reverbs are not very flexible. However, over the years more and more features have appeared that cleverly simulate the inner workings of algorithmic reverbs, offering a much more intuitive interface. Apart from finding good quality FIGURE 10.4 Basic Impulse Response (IR)

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PART | II Home Studio Mastering Techniques and suitable IRs, this makes setting a convolution reverb very similar to an algorithmic reverb. For example, nowadays most convolution reverbs offer familiar features like reverb time, dampening, pre-delay and stereo width. In addition, many convolution reverbs allow you to shape the IR’s volume envelope. It can be useful to use this envelope to add some attack if the reverb sounds too aggressive or direct, or to add some decay if the reverb needs a smoother tail. In addition, because IRs sound so authentic and lively, many modern convolution reverbs feature an integrated parametric equalizer to shape the reverbs’ tonal balance. This is very helpful, as you most likely need to cut some low rumble and dampen the high frequencies to get a more controlled reverb signal. The strength of convolution reverb is adding a natural flavor to productions that have a “digital” sound, which makes it a powerful tool for producers and musicians working in a home studio. Most of the time, however, adding just a little convolution reverb is all you need.

Chapter 11

Dynamics

11.1 Dynamic Range and

11.5 Compression for Balance

Crest Factor

and Glue

11.2 Mastering Compression

11.6 Multi-Band Compression

Basics

and Dynamic Equalization

11.3 Optimizing Dynamics

11.4 Compression for Punch

and Impact

11.1 DYNAMIC RANGE AND CREST FACTOR

Optimizing the dynamics of your track or release is an important step in creating a modern sound that is both powerful and controlled. When discussing dynamic processing techniques, however, it is important to distinguish between a track’s dynamic range and its crest factor, as these terms are often used interchangeably while they actually have different meanings. A track’s dynamic range can be described by the differences in loudness and intensity between different sections of the song. For example, in the majority of more traditional commercial tracks, a chorus will be louder and contain more expressive playing and singing than a verse. Also, many modern electronic music genres make use of extended build-ups that get louder and more intense over time, to create tension and anticipation. A track’s crest factor, on the other hand, can be described by the difference between a track’s peak volume and average or RMS volume, at any given point in the track. It is sometimes called 129

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PART | II Home Studio Mastering Techniques FIGURE 11.1 Jeroen’s Studio—Aalsmeer, The Netherlands (1) short-term dynamic range. Essentially, it gives you an indication of the number of peaks or transients in your track, which are necessary for creating a powerful sounding master. Because the transients in a track oftentimes get squashed by using compressors and limiters for maximizing loudness, the crest factor can tell you if any destructive processes have been used prematurely, or if the techniques you are using have the intended effect.

In most popular music genres, the dynamics sound controlled without the liveliness of the tracks being negatively affected. However, the dynamic range as well as the crest factor of commercial releases have been greatly reduced over the past decades, due to the Loudness War ( refer to: 1.3 A Bit of History). More traditional music genres have suffered a great deal from this development, as most tracks featuring real musicians require sufficient dynamics to convey the details and nuances in their playing. On the other hand, because of the demand for loud music—and specifically for tracks that maintain competitive loudness from start to end—many electronic music genres have evolved to create dynamic range mainly by varying intensity and rhythmic complexity, instead of by varying volume and expression. In that regard, even entirely new electronic music genres have appeared based on this principle, usually

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taking a relatively minimalistic approach. In any case, it is safe to say that the reduction in dynamic range and crest factor has only widened the gap between traditional and modern music genres.

Because of this, even though incorporating equal loudness functionality is slowly becoming the standard, mastering requires you to be opinionated and have a clear vision for the dynamics in your releases. Keep that in mind, while we go through the different techniques in this chapter to optimize the dynamics of your track.

“It is safe to say that the reduction in dynamic range and crest factor has only widened the gap between traditional and modern music genres.”

11.2 MASTERING COMPRESSION BASICS

For most dynamic processing techniques during mastering, you will use a compressor plug-in. As discussed earlier, similar to approaching equalization and reverberation, you should differenti-ate between surgical and transparent processing for making specific corrections or improvements, and processing to enhance the overall character of your track ( refer to: 5.6 Dynamics). For applying full-band transparent and surgical compression, you want a compressor plug-in that sounds clean and gives you full and continuous control over the following features: threshold, ratio, attack time, release time, soft knee, peak and RMS mode, mix and make-up gain. In addition, although dangerous in the hands of novice engineers, having a transparent sounding multi-band compressor with mid-side support, or alternatively a dynamic equalizer, can help you to target specific problems in your track. For adding overall character to your track, on the other hand, you want a compressor plug-in that is specifically designed to add flavor, or that emulates a well-known analog compressor design. Such character compressors will usually only provide limited control over a subset of the above features. Whichever compressor plug-in you use though, it is very important to have a good understanding of a compressor’s

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PART | II Home Studio Mastering Techniques basic features, as applying compression during mastering is quite a delicate business:

● The threshold lets you adjust the volume level in dB, above which the compressor kicks in. This means that as long as your track’s volume stays under the threshold level, no compression is applied. For this reason, among others, most compressor presets are useless, as your plug-in will not take into account your track’s volume.

● The ratio lets you adjust how much any peaks that exceed the threshold will be reduced in volume. The higher the ratio, the more the signal gets attenuated. For example, if the ratio is set to 3:1, and a peak exceeds the threshold by 6dB, it will be attenuated by 4dB. The peak that exceeds the threshold is scaled down by dividing it by the ratio: 6 / 3 = 2. Then, the attenuation is calculated by taking the before-and-after difference of the peak: 6– 2 = 4, resulting in 4dB of attenuation.

● The attack time lets you adjust the time it takes to achieve full attenuation, after a peak exceeds the threshold. Increasing the attack time allows you to preserve the transients in your track.

Be aware that the same attack time can have a different result depending on the compressor plug-in you are using. Oftentimes, analog-style compressor plug-ins respond about 5 times as slowly as digital-style compressor plug-ins.

● The release time lets you adjust the time it takes for the attenuation to revert back to zero, after a peak returns below the threshold. Increasing the release time allows you to smooth out the compression over time for a more subtle effect. As with the attack time, remember that analog-style compressor plug-ins respond about 5 times as slowly as digital style compressor plug-ins.

● The soft knee lets you adjust how gradually the compressor reaches full attenuation, as your track’s volume approaches and exceeds the threshold. For example, with a soft knee of 10dB, the attenuation starts fading in 5dB under the threshold, until it

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reaches full attenuation 5dB above the threshold. Increasing the soft knee can make the compression sound a little smoother.

● The peak and RMS mode lets you adjust the way the compressor reacts to the input signal. In peak mode, the compressor will react instantaneously to any peak it encounters, and lets you target the transients in your track. In RMS mode, the compressor will react to the real-time RMS volume of your track. As RMS volume is meant to resemble the perceived loudness of a track, this mode lets you target your track in a more musical way.

● The mix control, sometimes called wet-dry, lets you adjust the balance between the compressed and uncompressed signal.

This feature makes it possible to apply parallel compression techniques without over-complicating your plug-in chain. In addition, it gives you yet another option to make the compression sound a little smoother.

● The make-up gain lets you adjust the output volume of the compressor. Because compression changes your track’s volume, a make-up gain feature makes it easy to do a before-and-after comparison on the same volume, avoiding any psycho-acoustic issues.

FIGURE 11.2 A Compressor’s Gain Graph and Attack and Release Response

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PART | II Home Studio Mastering Techniques Besides these basic compressor features, there are a few additional features that can help you out in specific situations. Some compressors feature an auto-release control, that continuously adapts the release time to your track in real-time. This feature is especially helpful for compressing minimalistic tracks featuring live musicians in a smooth and natural way. Keep in mind, though, that there is no standard way of implementing auto-release in a compressor, so its behavior will vary between plug-ins and can potentially lead to unpredictable results. Also, some compressors feature a high pass control that lets you filter out a part of the low end of the compressor’s detection signal (sometimes called the internal side-chain signal). This feature is very helpful when compressing electronic tracks with heavy kick-drum and bass parts, because you can make the compressor ignore the low end as it calculates the necessary attenuation. This approach is usually preferred over using a multi-band compressor, as it is more predictable. And last, but most certainly not least, some compressors feature a stereo link control that lets you either process the left and right channel independently, or apply the same amount of attenuation to both channels. For the purpose of mastering, it is important to make sure the compressor you use is 100% stereo-linked. If it is not, the center of your track’s stereo image may move sideways every time there is a significant difference in volume between the left and right channel. In general, when using any compression technique during mastering, keep in mind that you are processing your track as a whole. This not only makes the compression process potentially very destructive, but it also means the transients are buried in the track and are therefore difficult to target. You essentially have no choice but to be subtle, or use parallel processing. For most mastering compression techniques, this means that you will seldom generate more than a few dB of attenuation.

It can take some experience, however, to accurately perceive the workings of a compressor. In that regard, especially for novice engineers, it is important that your compressor plug-in features

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a detailed attenuation meter, so you have a visual reference of what’s going on as well.

11.3 OPTIMIZING DYNAMICS

How to approach optimizing the dynamics of your track is entirely situational. In the case of a properly recorded and well-mixed track that has controlled but lively dynamics, applying additional dynamic processing during the mastering stage might be totally unnecessary. On the other hand, in the case of a track that has uncontrolled dynamics or you need to match the dynamics of different tracks that are part of the same release, there are a few different full-band techniques you can try:

● Compression for punch

● Compression for impact

● Compression for glue

● Compression for balance

Although there are many more ways of using a compressor during mastering, the techniques listed here are often used because they are relatively subtle and produce predictable results. This is helpful, especially for novice engineers, as the workings of a compressor can sometimes be difficult to detect, which increases the risk of over-compression. Once you gain more experience in optimizing dynamics, though, and you understand why these techniques are effective, you will be able to use compressors in a much more purposeful and intuitive way, including the following two more advanced multi-band techniques:

● Multi-band compression

● Dynamic equalization

Whatever compression techniques you use, it is important to preserve your track’s transients, as these give your track its power and definition. Also, make sure that you do before-and-after comparisons on the same perceived loudness, and fall back on your

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PART | II Home Studio Mastering Techniques reference tracks often. Compression not only influences volume and tonal balance at the same time, by compressing louder frequency ranges harder, but it also has the ability to influence the perceived loudness of your track in another unique psycho-acoustic way. In real life, when you listen to extremely loud sound sources, your ears will naturally compress these as a biological safety mechanism. Consequently, sounds that are compressed artificially before they reach your ear are interpreted by your brain as being louder than they actually are. So, even more so than other processing techniques you might use during mastering, compression can mess with your perception, and thus with your objectivity.

On a final note, before we actually discuss these techniques: try a few different compressor plug-ins. There are many configurations for implementing compression, and some might work better than others for your particular tracks.

“Whatever compression techniques you use, it’s important to preserve your track’s transients, as these give your track its power and definition.”

11.4 COMPRESSION FOR PUNCH AND IMPACT

More powerful music genres, like rock, Hip-hop and EDM, rely on a strong rhythm section, which either consists of a drummer and bass player, or a combination of electronic samples, loops and bass parts. If your track has such a rhythm section or a lead instrument like a guitar or synth that has a similar role in your arrangement, but it somehow lacks power and unity, there are two compression techniques that increase punch and impact, respectively. Be aware, however, that if the transients in your track have accidently been reduced somehow during the mixing stage, these techniques will be ineffective and probably make your track sound worse.

● Compression for punch is a relatively subtle technique to give the transients in your track a little boost. For this technique you

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need a flexible and digital style compressor that is able to react fast to any peaks in your track. You can experiment with character compressors as well, as long as you keep in mind they will often react much slower while set to the same attack and release times. Start by making sure that the compressor is 100%

stereo-linked to avoid destabilizing the center of your track’s stereo image, and that the compressor is in peak mode, as RMS

is too slow for this technique. In addition, as we don’t want parallel processing, make sure the compressor’s mix control is at 100%, so it is not letting through any of the unprocessed signal. Then, for the compressor’s initial values, set the attack time to 15ms, the release time to 50ms and the ratio to 1.2:1.

Play the loudest part of your track and pull down the threshold until you get a maximum of about 1dB to 2dB gain reduction.

You can adjust the attack to let through more or fewer of the transients, and adjust the release to control how smoothly the attenuation fades away. Everything, except for the transients in your track, will be brought down a little. As this will change the volume of your track, use the make-up gain to perform an objective before-and-after comparison. This technique is meant to be subtle, so don’t overdo it.

● Compression for impact is a parallel compression technique based on the New York compression principle, and is perfectly suited to give loud and powerful music more impact and a modern and produced sound. For this technique you need a flexible and digital style compressor that has a mix control for easy parallel processing, as well as an RMS mode. Start by making sure the compressor is 100% stereo-linked, in RMS mode and that the mix control is at 100%. Then, for the compressor’s initial values, set the attack to 15ms, the release to 100ms and the ratio to 8:1. Play the loudest part of your track and pull down the threshold until you get an overly compressed sound where the transients begin to sound explosive. You can adjust the attack to let through more or fewer of the transients, and the release to control how smoothly the attenuation fades away.

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PART | II Home Studio Mastering Techniques Turn the mix control down to 0% and begin to slowly add the processed signal to the original. Depending on how extreme your settings are, you probably want to set the mix control to about 20% to 40%. Use the make-up gain to perform an objective before-and-after comparison. This compression technique in particular can be used effectively to create psycho-acoustic loudness. However, it is easy to get carried away with it, for that same exact reason. So make sure to regularly fall back on your reference tracks.

Combining these two techniques can be especially effective. Not only can this make your track sound more powerful and controlled, it might also allow you to choose more conservative settings, achieving the intended results with a less destructive approach.

In addition to using these two compression techniques, try experimenting with transient designing plug-ins as well. Although transient designing plug-ins are not often used in mastering because they can degrade the overall quality of your track, subtly boosting your track’s transients before using compression for punch or impact can help the rhythmic parts of your track to cut through the mix. For boosting specific elements in your track, like the kick-drum or snare-drum, you can experiment with a multi-band transient designer, but stay on the lookout for any artifacts and degradation.

11.5 COMPRESSION FOR BALANCE AND GLUE

More subtle music genres, like singer-songwriter, folk and more classically oriented styles, that make use of expressive acoustic instruments, require a much more subtle approach when it comes to compression during the mastering stage. For these kinds of tracks, there are two compression techniques that respectively balance out the overall dynamic range in a natural sounding way, and subtly glue the parts and pieces of your track together. In addition—no matter what music genre—these techniques can also be used to add subtle warmth to a track and create a more controlled sound,

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or to very effectively even out the differences in volume between multiple tracks that are part of the same release.

● Compression for balance is a parallel compression technique that will bring out the softer parts of your track in a very natural way, which can be very helpful if you want to subtly even out the dynamic range of your track. It is especially useful for guitar music and other acoustic genres. For this technique you need a flexible and digital style compressor that has a mix control for easy parallel processing, and if possible an RMS mode. You can experiment with character compressors as well, as long as they have the ability to set fast attack time, preferable 0ms. Start by making sure the compressor is 100%

stereo-linked. Then, for the compressor’s initial values, set the attack to 0ms, the release to 250ms and the ratio to 2:1. Play the loudest part of your track and pull down the threshold until you get a maximum of about 5dB gain reduction. If you feel that the compression is too obvious, you could set the compression style to RMS and adjust the threshold until you have about 2dB

or 3dB of gain reduction. This might sound more subtle. You can adjust the release to control how smoothly the attenuation fades away. Turn the mix control down to 0%, and begin to slowly add the processed signal to the original. Depending on how extreme your settings are, you probably want to set the mix control to about 30% to 60%. Use the make-up gain to perform an objective before-and-after comparison.

● Compression for glue is a technique to glue your track together and add some subtle fullness and warmth. If you hear someone talking about general mastering compression, this technique is probably close to what they mean. For this technique you need an analog-style compressor that sounds smooth and has a pleasing character. Start by making sure the compressor is 100%

stereo-linked to avoid destabilizing the center of your track’s stereo image. Then, for the compressor’s initial values, set the attack time to 30ms, the release time to 300ms and the ratio

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PART | II Home Studio Mastering Techniques to 2:1. Play the loudest part of your track and pull down the threshold until you get a maximum of about 2dB to 3dB gain reduction. Be aware that some analog-style compressors will apply some kind of processing even with the threshold at 0dB, in which case you might want even less attenuation. You can adjust the attack to let through more or fewer of the transients, and the release to control how smooth the attenuation fades away. If available, turning on the soft knee will smooth the compression a little, giving the compressor a more subtle character. Use the make-up gain to perform an objective before-and-after comparison.

Both of these two techniques are meant to be used in a subtle way. If you feel like you need to reduce your track’s dynamics in a less subtle way, it is probably best to use the compression for impact technique described in the last paragraph first, before applying compression for balance and glue. Or, alternatively, to subtly apply these techniques multiple times in different places in your plug-in chain. As suggested earlier, experimenting with the order and possible repetition of equalization and compression techniques can lead to a very controlled and balanced track ( refer FIGURE 11.3 Jeroen’s Studio—Aalsmeer, The Netherlands (2)

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to: 9.8 Enhance Overall Timbre). In addition to using these two compression techniques for creating a warm and controlled sound, try experimenting with mastering saturation plug-ins as well, especially the ones that are modeled on analog tape machines. Many of them will subtly smooth out the transients in your track and reduce dynamics in a very natural way, while at the same time adding warmth and presence in a way that is difficult to achieve with most compressor plug-ins.

11.6 MULTI-BAND COMPRESSION AND

DYNAMIC EQUALIZATION

In some cases you only want to target the dynamics of a specific element in your track. For example, maybe the kick-drum in your track lacks punch or varies in volume too much, or the lead vocals sound harsh whenever the singer sings louder words.

In these cases you can use a multi-band compressor or a dynamic equalizer, preferably with mid-side support. Common problems that can be reduced with multi-band dynamic plug-ins are: weak kick-drums, unbalanced bass guitars and unstable sub in the low end, unbalanced lead instruments like guitars and piano in the mid range, and harsh sounding vocals and solo instruments in the high end. All these issues can potentially be solved, or at least reduced, by making educated decisions while setting a multi-band compressor or dynamic equalizer. Keep the following rules in mind:

● Use mid or side processing if appropriate, but don’t unnecessarily risk destabilizing your track’s stereo image.

● Use peak mode if you want the plug-in to react instantaneously to any peaks, and RMS mode if you want to target perceived loudness.

● Use a short attack time if you want to target the transients, or a long attack time if you want to let them through.

● Use a short release time if you only want momentary attenuation, or longer release times if you want the attenuation to fade away smoothly.

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● Use a low ratio and low threshold for subtle attenuation over a relatively large dynamic range, and a high ratio and a high threshold for more extreme attenuation over a relatively small dynamic range.

● Use the plug-in’s attenuation meter as a visual reference, while setting the threshold.

● Use the make-up gain to compensate for the plug-in’s attenuation, as multi-band processing will not only change your track’s volume, but also its tonal balance.

In general, having to solve dynamic problems during mastering that require such precision tools is probably a sign you need to go back to the mix. If you know how to use these tools properly, they can be real life-savers from time to time. However, as it is very easy to do more harm than good using a multi-band compressor and dynamic equalizer plug-ins, only use them when you have a specific reason, and definitely stay away from their preset section.

Chapter 12

Loudness

12.1 Track Order and

12.4 Loudness by

Individual Volume

Compression

12.2 Maximizing Loudness

12.5 Loudness by Brickwall

12.3 Loudness by Equalization

Limiting

12.1 TRACK ORDER AND INDIVIDUAL VOLUME

After optimizing the tonal balance, width and depth, and dynamics of your track or release, you have completed the most challenging tasks of your mastering session. You have made many corrections and improvements and, in the case of a full release, maintained a level of coherency between the tracks that is needed to achieve the intended album flow. That means it is time for the final processing step in the mastering session: maximizing loudness. In the case of mastering a single track, this is a fairly straightforward process.

However, when mastering multiple tracks that are part of the same release, there are two tasks left to complete before you can maximize the loudness of your release as a whole: deciding on the track order and setting your tracks’ individual volumes.

So far you have kept your tracks on different channels, to be able to skip through them instantaneously and to accurately judge the differences between them. For finalizing the track order and individual volumes, though, it is usually much easier to have your tracks in order on a single channel. Not many sequencers and audio editors support these two set-ups simultaneously, at least 143

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PART | II Home Studio Mastering Techniques not without breaking your channel routing and individual plug-in chains. Therefore, at this point in the mastering session, it is often most practical to save your tracks including all processing on the same samplerate and bitdepth as your mastering project, and to re-import them again on a single channel in your master project.

This way, you can put them in order, fully processed, and tweak their individual volumes to create good album flow. Only if your release is very balanced overall and consists of tracks that are very much alike, this in-between rendering stage might be unnecessary, and you can just use the group channel to apply the processing techniques in this chapter. Consider the following when deciding on the track order of your release and setting your tracks’ individual volumes:

● Deciding on the track order of your release can be a bit of a puzzle. Some composers and songwriters approach writing an album more conceptually and will create an outline of the type and order of songs beforehand, but most will just write and produce a number of songs and then make a selection out of them to form an album. Whatever approach you have chosen, there seem to be certain reoccurring conventions for putting together a commercial album that you could use as a guideline. Many albums start with a more powerful up-tempo song to immediately draw as much attention as possible, and end with a more modest down-tempo song to conclude the album in a subtle way. In addition, similar to a live set where certain songs might be played consecutively and others might be followed by a short break where the singer talks to the audience, many albums are divided in subsections. Oftentimes these subsections start and end with a more powerful up-tempo and a more modest down-tempo song as well, followed by a relatively long pause.

● Setting your tracks’ individual volumes is best done by ear.

Play the ending of one song and listen how it transitions into the next. Generally, more powerful songs need to kick in with force, while more modest songs call for a subtle beginning.

Because of this, more modest songs are usually set to a lower

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volume than more powerful songs. In addition, when you have two powerful songs in sequence, it is often beneficial to lower the first song slightly to have the second song kick in just a little louder. Of course, due to the relationship between loudness and tonal balance, tweaking your tracks’ individual volumes relative to each other might disturb the overall balance of your release.

If you have level matched your tracks consistently throughout the mastering session, these differences are probably negligible.

However, in case you do have to make considerable volume adjustments, you can subtly automate the shelving filters of an equalizer to compensate. In general: when turning turn down a track, increase the highs and lows a little, and when turning up a track, decrease the highs and lows a little. Be very subtle.

After finalizing the track order and the tracks’ individual volumes—

perhaps with the help of an automated filter, you should be able to randomly skip through the songs without being distracted by any significant sonic differences. The listening experience should be consistent. At this point any individual processing is completed and all further processing for the purpose of maximizing loudness will be applied to your release as a whole.

FIGURE 12.1 Tsjah’s Home Studio—Leiden, The Netherlands (1)

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12.2 MAXIMIZING LOUDNESS

Loudness is a controversial and heavily debated subject, especially in the mastering community. This is understandable, as for many mastering engineers it has become common practice to carefully shape a track to sound close to perfect, after which they have to crush it to pieces to hold to a commercially driven and inordinate loudness standard. That is not to say optimizing loudness is a bad thing, though, as for most tracks there is certainly a sweet spot where it sounds both powerful as well as controlled. However, the Loudness War has taken it too far, indubitably putting audio quality second to loudness. Fortunately, as discussed earlier, it seems the Loudness War is slowly coming to an end, due to many popular online audio services like Spotify and iTunes implementing equal loudness functionality ( refer to: 1.3 A Bit of History). As a result, tracks that are competitively mastered in regard to loudness will sound relatively flat and degraded when played at the same perceived loudness than tracks that are conservatively mastered.

However, although equal loudness functionality is being implemented more and more, many record labels seem hesitant to go along with this trend and continue to release albums that have been mastered at competitive loudness. The music industry has not reached a consensus yet when it comes to loudness, so it is up to you to form a well-defined opinion about current loudness trends and to choose the best approach for the loudness of your release.

To help you choose an approach, consider the following:

● How loud are your reference tracks mastered? In many cases, commercial albums within a specific music genre follow the same trends over time when it comes to loudness. Therefore, if you have chosen high quality and contemporary reference tracks that are from the same music genre as your release, it is probably best to use these as your main reference. In addition, if you see a clear increase or decrease in loudness comparing well-known releases over the years, you might want to try to stay ahead of the game, by anticipating this trend.

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● Are you working on a more traditional or a more modern music genre? More traditional music genres have suffered most from the Loudness War and will benefit from a more conservative approach. On the other hand, many more modern music genres have adapted to the demand for tracks that have a competitive loudness from start to end and can therefore be mastered more competitively without doing excessive damage ( refer to: 11.1

Dynamic Range and Crest Factor).

● In what setting do you expect people to listen to your track or album? If you plan on an online release on Spotify or iTunes with equal loudness functionality, or you expect people to listen to your album from start to end on a high quality hi-fi system at home, it is probably best to take a more conservative approach. On the other hand, if you expect people to mostly listen to your release on an MP3-player or radio station without equal loudness functionality, most likely alternating between different artists and albums, it might be best to choose a more competitive approach.

Once you have decided on using a more conservative or competitive approach, there are several processing techniques you can use to maximize the loudness of your release. Don’t just use a single brickwall limiter, as these can cause serious degradation and artifacts, especially if you are aiming for a competitively loud master and you need to push it hard ( refer to: 5.7 Loudness). By intelligently applying the following three processing techniques succes-sively, you will be able to maximize loudness without doing any excessive harm to your track:

● Equalization

● Compression

● Brickwall limiting

If you are working on a single track, you can just apply these techniques at the end of the channel’s plug-in chain. If you are working on multiple tracks that are part of the same release, it is best

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PART | II Home Studio Mastering Techniques FIGURE 12.2 Rounded Peaks by Intelligent Limiting to apply these techniques to your release as a whole. This way, if you take the loudest section of your release as your reference point while maximizing loudness, any overall degrading effects caused by the limiting process will be minimal. If you have put your tracks in order on a single channel for finalizing the track order and individual volumes, you can use the chain of this new channel. If you have kept them on separate channels, you can use the chain of the group channel to which you have routed them. In the next three paragraphs, we will go through the different techniques for maximizing loudness one by one.

12.3 LOUDNESS BY EQUALIZATION

Although by this point you might have carefully sculpted the tonal balance of your tracks to match that of your reference tracks, there are a few subtle equalization tricks that can help maximizing loudness, both perceived as well as actual loudness. Remember, when you turn up the playback level, due to psycho-acoustic principles the low and high end will get relatively louder compared to the mid range, making your track sound more powerful and clear ( refer to: 6.3 The Art of Listening). This also works the other way around, meaning that subtly boosting the high and low end will increase the perceived loudness to a greater extent than the actual loudness.

Boosting certain high frequencies that increase psycho-acoustic loudness can be done effectively without running into trouble, as long as you don’t diverge too far from your reference tracks.

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Depending on how transparent you want the processing to be, you can either use a regular clean equalizer or a character equalizer for this task. The low frequencies, however, are usually the loudest in modern commercial music and will therefore strain compressors and limiters used for maximizing loudness much more than the mid range and high end. Boosting the low end for maximizing loudness can therefore also be counter-productive, because increasing psycho-acoustic loudness this way will decrease the actual loudness you will be able to achieve with a compressor and limiter before causing degradation and artifacts. In fact, for music genres that don’t rely too much on the low end, it can actually be beneficial to reduce the low end. In any case, for both reshaping the high end and low end to maximize loudness, it is important to keep comparing and level matching your tracks to your reference tracks, to make absolutely sure your adjustments have a positive effect. How exactly you judge this, is again up to you and your vision for your release. So go through the following equalization techniques and decide for yourself if your tracks benefit from them:

● Use a high pass filter to remove all frequencies below about 15Hz to 20Hz. This way, you remove any inaudible frequencies that might limit the effectiveness of the compression and brickwall limiting techniques in the next two paragraphs. Be careful if your track relies heavily on the low end, as you don’t want to lose any valuable sub-frequencies due to the roll-off curve of the filter.

● Use a spectrum analyzer with good low end resolution to see where most of the energy in the low end is concentrated. Use a peak filter to reduce that frequency area by 1dB to 2dB. If your track doesn’t rely on the low end too much, you can alternatively use a low shelving filter to reduce the low end by 0.5dB to 1dB

starting at the same frequency. Be careful not to lose any warmth.

● Use a wide peak filter to boost the frequency range around 1.8kHz by about 1dB to 3dB. This frequency is often boosted in rock and metal productions for a loud and edgy sound. Make sure your track doesn’t start to sound overly thin compared to your reference tracks.

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● Use a moderately wide peak filter to boost the frequency range around 5kHz by about 1dB. Human beings are highly sensitive to this particular frequency, because this is where the intelligibility of speech is located. Alternatively, use a high shelving filter to boost the high end by about 0.5dB starting just below 5kHz. Make sure your track doesn’t sound harsh or overly bright compared to your reference tracks.

Instead of using an equalizer, it is also possible to use a saturation plug-in for boosting certain frequencies that increase psycho-acoustic loudness. Any type of mastering saturator, enhancer or exciter might do the trick, as long as it gives you control over the tonal color of the saturation. Remember, however, that because saturation plug-ins will react stronger to louder signals, they might not produce consistent results applied to your release as a whole, especially if your tracks considerably vary in volume. If you do use a saturator plug-in, make sure you use the loudest part of your track or release as your main reference when setting the saturator, so you are sure the plug-in will not audibly distort your track.

“Remember, that because saturation plug-ins will react stronger to louder signals, they might not produce consistent results applied to your release as a whole, especially if your tracks considerably vary in volume.”

12.4 LOUDNESS BY COMPRESSION

Besides controlling dynamics, compression can also be used to maximize the actual loudness as well as the perceived loudness of your track or release. Ideally, you want to use the techniques explained in this paragraph in an extremely subtle way, especially when you apply them to your release as a whole. You want to preserve the transients in your tracks as much as you can and make sure you do not introduce any dynamic volume changes that sound unnatural. For a competitively loud master, you might have to do

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some damage at this point. Try out the following three techniques, and see if they raise the loudness of your tracks without causing problems:

● Use the compression for impact technique to raise the average volume of your track or release and to create more body and a more dense and controlled sound ( refer to: 11.4 Compression for Punch and Impact). This technique is most effective when applied to tracks with a heavy rhythm section. Use a fast digital style compressor in RMS mode and initially set the attack to 15ms, the release to 100ms and the ratio to 8:1. Play the loudest part of your track and pull down the threshold until you get an overly compressed sound where the transients begin to sound explosive. Adjust the attack and release to preserve the transients and smooth out the attenuation and carefully add the processed signal to the original with the mix control.

For the purpose of maximizing loudness, just a few percent should do the trick. This style of compression also roughly mimics the way your ears naturally compress extremely loud sounds as a biological safety mechanism ( refer to: 11.3 Optimizing Dynamics), making your track appear louder psycho-acoustically. Be careful not to overdo this technique, especially if you already applied it while optimizing the dynamics of your track or release.

● Use the compression for balance technique to bring out the softer parts of your release, and to create consistent loudness from start to finish. Use a fast digital style compressor in peak mode and initially set the attack to 0ms, the release to 250ms and the ratio to 2:1. Play the loudest part of your release and pull down the threshold until you get a maximum of about 3dB to 5dB. Adjust the release to smooth out the attenuation and carefully add the processed signal to the original with the mix control. If your release has a limited dynamic range, about 10% to 25% should suffice to bring out the softer sections. On the other hand, if your release has an extended dynamic range, you might

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PART | II Home Studio Mastering Techniques have to go up to a maximum of about 50%. This style of compression can help to create album flow in a very natural way.

● In addition to the two previous techniques, there is a strategical way to use a compressor to aid the brickwall limiting process explained in the next paragraph of this chapter. Use a digital or analog-style compressor in peak mode with the soft knee enabled, and initially set the attack to 100ms, the release to 250ms and the ratio to 1.5:1. Play the loudest part of your release and pull down the threshold. Your aim is to reduce the loudest sections of your release by about 2dB to 3dB, without significantly affecting the softer sections. If the softer sections do get reduced too much, raise the ratio a little and try again until you find a suitable balance where the loud sections are reduced slightly but the transitions between sections still sound natural. This technique will help a brickwall limiter achieve the same results with less effort, minimizing limiting artifacts and degradation.

Because compression influences actual loudness, perceived loudness and tonal balance at the same time, make sure to fall back on your reference tracks often. It is not difficult to fool yourself into thinking you have created more impact due to the sound of the compression itself, only to later discover you have accidentally squashed all the transients in your track, doing just the opposite. In addition, because compressors react differently depending on the volume of the source material, listen to a few other sections with different volumes before committing to your settings.

12.5 LOUDNESS BY BRICKWALL LIMITING

The final and most well-known technique for optimizing loudness is brickwall limiting. In essence, brickwall limiting is clipping your track’s peaks at a fixed threshold, so you can raise its volume due to the headroom created. However, differently from just pushing your track over the digital zero dBFS limit, which causes extreme distortion, a mastering brickwall limiter plug-in will intelligently

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round off the clipped peaks, reducing distortion artifacts. There is no standard way of implementing this intelligence, though, and over time many schemes have been developed to meet the demands brought on by the Loudness War. Some brickwall limiter plug-ins only implement a form of adaptive release, but others include elaborate schemes to preserve transients and the frequency ranges responsible for creating psycho-acoustic loudness. However, in most brickwall limiting plug-ins for mastering, this functionality is kept hidden from the user, or is only partly available on the interface. Most of them do feature the following three controls though: ceiling, gain and auto-release.

● The ceiling lets you adjust the maximum output value of your track. Usually, the ceiling is set just below zero, for example to −0.3dB. This will help prevent intersample clipping caused by converting back to analog and compensates for the potential slight volume increase that can occur when converting your track to a data-compressed format like MP3.

● The gain lets you boost the volume of your track up against the ceiling you have set. On some limiters the gain acts as a volume control that goes from 0dB upwards, but on other limiters it acts as a threshold that goes down from 0dB, while automatically applying the appropriate make-up gain behind the scenes to hit the ceiling. However, both implementations have the exact same effect.

● The auto-release lets you adjust how the release time adapts to your track. Because there are many different ways of implementing adaptive release—based on the crest factor, or the amount or persistence of the attenuation generated, for example—it is best to set this control purely by ear.

Before you can apply your limiter, you need to make some preparations. So far, you have level matched your reference tracks to the track you are mastering, to help you stay objective and avoid psycho-acoustic confusion. However, for this final step of maximizing loudness with a brickwall limiter, you need an actual

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PART | II Home Studio Mastering Techniques real-world reference instead. Therefore, it is essential to put your reference tracks back to their full volume, so you can judge their absolute differences in loudness. If you are aiming for competitive loudness—although you should mostly rely on your ears and reference tracks—it might be helpful to use a full-featured loudness meter as a visual reference. It should feature a true peak meter, so you can check if your track indeed doesn’t exceed zero dBFS, an RMS meter, to give you an indication of the loudness the way humans perceive it, and preferably a crest factor meter to give you an indication of the negative effects the limiting process has on your track’s transients. If you are aiming for conservative loudness, on the other hand, and you want to make the most of the equal loudness functionality used by Spotify and iTunes, for example, you should use a LUFS meter instead, to hold to the most current LUFS

recommendation used by these services. Remember, however, that although at this point the goal is making an absolute comparison with your reference tracks, you should still do a before-and-after comparison of the limiting process applied to your track on the same perceived loudness to avoid psycho-acoustic confusion.

In order to apply the limiter, first compare your reference tracks to each other in terms of absolute loudness. If your reference tracks FIGURE 12.3 Tsjah’s Home Studio—Leiden, The Netherlands (2)

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are equally loud, they provide a clear absolute reference that can help you hold to the loudness conventions of the music genre. If your reference tracks significantly differ in loudness, on the other hand, you might want to think more carefully about your approach ( refer to: 12.2 Maximizing Loudness) and pick one of them as a main reference. When you have defined the approximate target loudness for your track or release by ear, make sure the channel you use for maximizing loudness, as well as the master channel of your project, is set to 0.0dB, and load up your limiter at the very end of your plug-in chain. Start off by setting the limiter’s ceiling to -0.3dB. Then, to get a clear idea of the limiter’s destructive nature, play the loudest part of your track or release and gradually increase the gain to its maximum value. Some limiters feature a control that lets you limit your track without actually maximizing its loudness, which can be convenient for this purpose. Otherwise, compensate for the volume increase manually by turning down your monitor level. Try to get a good feel for the degrading effects caused by the limiter when pushed hard, as these are the artifacts you want to minimize when choosing your actual settings. If your limiter has additional features for reducing artifacts, it can be helpful to try them out as well, because their workings will be much more audible when the limiter is pushed hard. If you know what to listen for, you are ready to choose your final settings.

“Whatever combination of plug-ins and techniques you end up using for maximizing loudness, remember that it usually sounds better using a combination of moderate techniques that work well together, than putting all the strain on one particular plug-in, especially when it comes to brickwall limiters.”

Play the loudest part of your track or release and gradually increase the gain, trying to match the absolute loudness of your main reference track. Watch the limiter’s gain reduction meter and keep in mind that you can generally achieve a maximum gain

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PART | II Home Studio Mastering Techniques reduction of about 3dB to 6dB without causing audible limiting artifacts. In the case of tracks that only consist of a few instruments or layers, achieving competitive loudness without causing limiting artifacts shouldn’t be that hard, especially if you have applied the equalization and compression techniques described in this chapter.

However, in the case of more dense arrangements with a lot of instruments and layers, the limiting process might result in distortion or breathing-and-pumping before reaching your target loudness. These artifacts can sometimes be minimized by adjusting the auto-release. If the limiter causes breathing-and-pumping, you want to decrease the average release time. If the limiter causes degradation or even distortion, on the other hand, you want to increase the average release time. Take into account, however, that increasing the average release will prolong the limiters attenuation over time and may significantly limit the potential maximum loudness.

So if you are aiming for competitive loudness, you want the average release time to be as short as possible, without causing distortion. If you are aiming for conservative loudness, on the other hand, you want the average release time to be a little higher, to prevent any degradation of the audio. If adjusting the auto-release doesn’t help, you can try to use more extreme equalization and compression for loudness settings and try the brickwall limiting technique again. Remember that most mastering brickwall limiters have their own particular scheme, and some of them might be significantly more effective on certain tracks. Therefore, you should try as many as you can, until you find one you like. In extreme cases, where you can’t get your track loud enough no matter what limiter or technique you use, there is one last trick to try: insert a clipping plug-in before the brickwall limiter and experiment with pre-clipping your track in different ways to give your limiter a little more breathing room. Whatever combination of plug-ins and techniques you end up using for maximizing loudness, remember that it usually sounds better using a combination of moderate techniques that work well together, than putting all the strain on one particular plug-in, especially when it comes to brickwall limiters.

Chapter 13

The Final Master

13.1 Housekeeping

13.4 Creating a Master-CD or

13.2 Rendering Your Tracks

DDP-File

13.3 Creating a CD-Montage

13.1 HOUSEKEEPING

At this point in your mastering session, all processing is completed. From a purely sonic point of view, you have optimized and matched your tracks to sound great and to ensure they will translate to the broadest possible range of audio equipment, from tiny laptop speakers to huge PA installations ( refer to: 1.2 What Is Mastering? ). It is now time to prepare your tracks for online distribution or replication, either in the form of single releases, or as an official master-CD or a DDP-file. In the case of single releases, this can be done quickly. However, in the case of an EP or album, there is some more housekeeping to be done:

● Rendering your mastered tracks

● Creating a CD-montage

● Creating a master-CD or DDP-file

Although these tasks might sound predominantly technical, there is certainly creativity involved in regard to creating the album montage and inserting the PQ codes. So although your master is near completion, it is important to maintain focus just a little 157

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PART | II Home Studio Mastering Techniques FIGURE 13.1 Ewan’s Studio—Leiderdorp, The Netherlands (1) longer, going through these final tasks and finalizing the album flow of your release.

13.2 RENDERING YOUR TRACKS

To prepare your tracks for release, you need to save them to be compatible with the Red Book Standard used for creating audio CDs. In practice this just means rendering your tracks as standard WAV or AIFF files with a samplerate of 44.1kHz and a bitdepth of 16bit ( Refer to: 7.2 Choosing Samplerate and Bitdepth).

If you have followed the recommendations of this book, you will have to bring down the bitdepth from 24bit to 16bit. In addition, if you have used a higher samplerate of 88.2kHz or 96kHz, you need to bring down the samplerate to 44.1kHz as well. Bringing down the samplerate can usually be done safely by just rendering your track to 44.1kHz, as most sequencers and audio editors have high quality built-in samplerate convertors that will handle this

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for you automatically. Bringing down the bitdepth, on the other hand, needs to be handled with care. If you reduce the bitdepth of your track from 24bit to 16bit by just discarding 8 bits—a process called truncation— you risk causing quantization distortion. As one bit approximately equals 6dB of dynamic range ( refer to: 7.3

Setting Output Volume and Headroom), truncating an audio track to 16bit basically means silencing all samples that go under −96dB.

Signals with an average level around −96dB will therefore get distorted, as they are essentially abruptly turned on and off again, at an extremely rapid pace. Although the −96dB quantization distortion that can occur with 16bit tracks is very low in level, it has an intrusive character and can actually be clearly audible in soft passages, like the end of a long and gradual fade-out, for example. Fortunately, quantization distortion is easy to prevent by adding a dither signal to your track. Dither is essentially just low-level noise that randomizes the quantization process of the least significant bit. It prevents quantization distortion very effectively, sounds much less intrusive and in most cases is virtually inaudible. Applying dither needs to be the last process you apply before saving your track, which is why many mastering brickwall limiters include a dither function. If your brickwall limiter does not have this, you can use a separate dither plug-in, or possibly the built-in dither functionality of your sequencer or audio editor. In most cases you can just activate the dither, set it to quantize your track to 16bit, and you are done. However, if you are working on a low volume track and the dither noise is clearly audible, you might want to experiment with the dither’s noise type and noise shape settings, to see if you can make the noise blend in better with your track. If you have applied dither, you can then render your tracks one by one as Red Book compatible stereo 44.1kHz and 16bit WAV or AIFF files.

After rendering your files, re-import them to manually trim the silences and create fades at the start and end of the tracks ( refer to: 7.5 Checking Render Section and Fades). Even for tracks that do not require an actual audible fade-in or fade-out, it is still helpful to create brief fades to make sure the tracks start and end at a

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PART | II Home Studio Mastering Techniques sample value of zero, to prevent digital clicks. After saving your changes, your individual tracks are done.

“Dither is essentially just low-level noise that randomizes the quantization process of the least significant bit. It prevents quantization distortion very effectively, sounds much less intrusive and in most cases is virtually inaudible.”

13.3 CREATING A CD-MONTAGE

For releasing your tracks as an album, the next step is to create a CD-montage that holds to the Red Book Standard. If you plan on having your album physically duplicated or replicated by a CD-plant, it is best to use a professional audio editor or a standalone CD and DDP-creator for this purpose. Create a new CD-montage project and load in your tracks in the proper track order. Three things need to be done before creating your actual master:

● Setting the track spacing

● Inserting PQ codes

● Adding ISRC codes and CD text

By carefully setting the spacing between the different tracks, you can optimize the album flow of your release. In most professional applications for creating CD-montages you can drag the tracks around, and use the zoom function to have a detailed look at the transitions between tracks. However, sometimes you are only presented with a basic form-like interface, where you have to fill in the pauses between the tracks manually. Keep in mind that such a basic interface usually doesn’t allow for creating crossfades between tracks. Setting the track spacing is best done by ear, especially in the case of tracks that end on a chord that slowly dies out over time. Although in the case of more abrupt endings, some engineers like to determine the pause by adding a certain number

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of beats or bars in the same tempo as the track. A pause of one bar seems to be common. However, if you have consciously divided your album into subsections, you want to make sure to choose longer pauses between subsections, than between tracks within a subsection ( refer to: 12.1 Track Order and Individual Volume). As a general rule, remember that the longer the pause you choose, the more anticipation you create with the listener. In the end, though, the best way to get a feel for a transition between two tracks is just to relax and play a good portion of the ending of the previous song, to experience what an actual listener will hear. With setting the track spacing you have finalized the album flow.

Next is inserting the PQ codes. The P and Q codes on a Red Book compliant CD store the start and end points of the tracks on your album, respectively, which enables CD players to discern between different tracks, and lets the user skip through them. In most professional applications for creating CD-montages, inserting the PQ codes can usually be done automatically. However, there are certain situations in which you might want to insert them manually, to ensure that the random-play mode and skipping through tracks is handled properly: continuous live-recordings and albums that contain crossfades. In the case of continuous live-recordings, you should carefully place the P codes at the start of the tracks, making sure you don’t hear any distracting instrument noises or applause when skipping through the tracks. In addition, you will also have to choose strategic locations for the Q codes at the end of the tracks, to avoid strange transitions in random-play mode.

Crossfades between tracks also force you to be creative in placing the PQ codes. You should strategically place the P code to ensure the second track has a well-defined starting point when skipping between tracks, as well as strategically place the Q code to make the first track end as naturally as possible in random-play mode.

Finally, you have the option to add ISRC codes and CD text to your album. ISRC stands for International Standard Recording Code and is used as the global identification code for music recordings (comparable to the ISBN code for books). One is assigned

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PART | II Home Studio Mastering Techniques FIGURE 13.2 Ewan’s Studio—Leiderdorp, The Netherlands (2) to each of the tracks on your album. Adding ISRC codes to your album is not mandatory, but for bigger releases it usually does make copyright registration and royalty collection a lot easier. An ISRC code consists of four parts separated by hyphens, for example: US-A01–18–00001):

● The two-character ISO country code. This would, for example, be ‘US’ for an American release.

● The registration code. This is a three-character combination you will have to apply for yourself, if you are doing an independent album release.

● The last two digits of the year in which the ISRC code is issued to a particular track.

● A unique designation. This is a five-digit number you can assign to your tracks yourself.

In addition to adding the ISRC codes to the tracks on your album, you can also add certain meta-data to your album as a whole,

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as well as to your individual tracks, by adding CD text to your album. CD text is an expansion of the Red Book Standard and can, among other things, be used to store the album name, as well as the track names, composer and performers, physically on the CD. This information will be visible on certain CD players. However, because CD text is added to the album as a whole, and not to the individual tracks, in general it will not show up in any audio players on your computer or online that play the individual tracks.

Most applications, both desktop as well as web-based, use certain online databases to retrieve any album information they show in their players. Therefore, if you want your album name and track titles to show up in commonly used audio players, you need to find out which databases they use for retrieving their album information and register your album to as many of them as possible.

Keep in mind that the way most audio players identify an album is relatively primitive, which can result in inconsistent behavior.

13.4 CREATING A MASTER-CD OR DDP-FILE

After creating your CD-montage, you are ready to create the final master of your album, which you can send to a CD-plant for replication or duplication. There are two generally accepted ways to create an official master:

● Burning a physical master-CD, to send in the post.

● Rendering a virtual DDP-file, to send over the Internet.

Traditionally, digital masters are created by using a high quality CD-burner, capable of burning at low speeds, to burn your Red Book compliant CD-montage on a high quality audio CD. However, even though nowadays most consumer CD-burners and audio CDs are of sufficient quality, the burn process can introduce different kinds of writing errors: C1, C2 and Cu errors. As most CD-plants will only accept master-CDs within a certain error margin, it can be helpful to ask them for an error margin specification and scan your CD yourself with an error-scanning application,

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PART | II Home Studio Mastering Techniques before sending it to them. Of course, by sending your master in the post, you risk physical damage to the CD as well. Therefore, to overcome all these issues, a virtual format has been developed, called a DDP-file (Disk Description Protocol). A DDP-file is a virtual format that contains your Red Book compliant CD-montage, which you can send to a CD-plant over the Internet. This eliminates the risk of write errors and physical damage to the master-CD. In addition, because a DDP-file is a virtual format, it is possible to create a hash-value or checksum of your DDP-file, and send this along with the master to the CD-plant. This enables the CD-plant to do a data-integrity check, which is accurate up to a single bit.

If the CD-plant accepts DDP-files, this is the preferred method of supplying them with your final master.

“A DDP-file is a virtual format that contains your Red Book compliant CD-montage, which you can send to a CD-plant over the Internet.”

Part III

One Hour Mastering Workshop

In the third part of Home Studio Mastering, you will put theory into practice. With the book comes a bundle of five mastering plug-ins in VST, VST3, AU and AAX format for Mac and PC: the Noiz-Lab LE Mastering Bundle. This plug-in bundle was specifically developed by the author to provide you with a basic real-world mastering chain. This section of the book guides you through the whole mastering process in about one hour, using only these five mastering plug-ins. Keep in mind that although the plug-ins can certainly be used to produce a professional master, the Noiz-Lab LE Mastering Bundle has been created to be a learning tool. In that regard, the One Hour Mastering Workshop can best be viewed as a basic interpretation of the book’s recommendations, providing a foundation you can expand on as you gain more mastering experience.

Chapter 14

One Hour Mastering Workshop

Download the Mastering

Step 1: Monomizer

Bundle

Step 2: Equalizer

The Plug-Ins and 5-Step

Step 3: Compressor

Workflow

Step 4: Convolver

Prepare Your Mastering

Step 5: Limiter

Session

In Conclusion

DOWNLOAD THE MASTERING BUNDLE

Before you can start the One Hour Mastering Workshop, you need to install the Noiz-Lab LE Mastering Bundle. You can download this software bundle from the book’s companion website, for which the website address can be found on the back of the book. Depending on your system, download either the PC or Mac version and unpack the zip-file. Then, make sure your sequencer or audio editor is closed, and run the installer. You will be presented with a standard installation dialog that will automatically install the five Noiz-Lab plug-ins in the appropriate folders on your system. Although the installer allows you to choose custom VST folders on PC, and allows you to deselect certain plug-in formats, it is best to just stick to the default installation, unless you have specific reasons not to do so. If the installation is successful, the plug-ins will be added to the plug-in menu in your sequencer or audio editor, either listed alphabetically or in a developer subfolder called “Noiz-Lab.” If the installation is unsuccessful somehow, or the plug-ins do not show 167

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PART | III One Hour Mastering Workshop up in your sequencer or audio editor, read the FAQs on the companion website, or use the website’s contact form to get support.

THE PLUG-INS AND 5-STEP WORKFLOW

In order to quickly and easily create a professional master, the five plug-ins in the Noiz-Lab LE Mastering Bundle are designed to work together in a very specific way. Before starting your mastering session, therefore, it is important to understand the role each plug-in plays in the Noiz-Lab LE 5-step Workflow: 1 Monomizer:

The Noiz-Lab LE Monomizer is a stereo imaging plug-in that lets you narrow the low end stereo image of your track. It gives you full control over the low end frequency range you want to affect, as well as over the curve and amount of the stereo-to-mono effect. In addition to its stereo imaging functionality, the Monomizer plug-in also features a filter that will reduce DC

offset problems dynamically over time. This plug-in is first in the chain, because its stabilizing effect on the stereo image, and possibly on the tonal balance as well, will make the remaining processing more effective. Removing DC offset problems is also done at the start of the mastering chain, because the asymmetry introduced by more extreme DC offset can negatively affect the workings of dynamics plug-ins that make use of a threshold, like the Compressor and Limiter further up in the chain.

2 Equalizer:

The Noiz-Lab LE Equalizer is an analog-style minimal phase equalizer, for optimizing your track’s tonal balance and stereo image. The plug-in features six filters that can be individually set to have a low pass, high pass, low shelf, high shelf or peak response, and to process in stereo, mid or side mode. It is possible to switch the filter type and processing mode of all filters individually, allowing you to quickly and easily try different equalization approaches. In addition, the Equalizer features a real-time spectrum analyzer for visual feedback, and lets you

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listen back to just the mid or side channel of your track, which allows you to analyze and adjust the tonal balance and stereo image of your track in more detail. The Equalizer plug-in is second in the chain to ensure that the tonal balance of your track is optimized before using any dynamics plug-ins, like the Compressor and Limiter further up in the chain, which might otherwise react to unwanted frequencies.

3 Compressor:

The Noiz-Lab LE Compressor is a dynamics plug-in that lets you apply four commonly used mastering techniques for controlling dynamics and adding character: compression for glue, balance, punch and impact. As these compression styles depend on a large number of specific parameters and specific values, most functionality is taken care of behind the scenes. This makes it possible to quickly and intuitively compare compression styles, using just the threshold control to vary intensity, and the gain reduction meter for visual feedback. In addition, the compressor features a saturation control to add character to your track, and a make-up gain control for doing an easy before-and-after comparison on the same perceived loudness.

The Compressor plug-in is third in the chain, after the Monomizer and Equalizer, to ensure it won’t react to unwanted frequencies, and before the Convolver and Limiter, that will both react more naturally to a track with more controlled dynamics.

4 Convolver:

The Noiz-Lab LE Convolver is a convolution plug-in that lets you impose the tonal balance and acoustic properties of different real-life rooms onto your track. You can choose between a studio, hall and church response, which will all create a certain depth and add a specific timbre or character to your track. You can mix the convolved signal in with the amount control, and use the make-up gain control for doing an easy before-and-after comparison on the same perceived loudness. In addition, you can shape the tonal balance of the wet convoluted signal with a high pass and low pass filter, and choose between a mono and stereo

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PART | III One Hour Mastering Workshop impulse response. The Convolver plug-in is fourth in the chain, after optimizing the tonal balance and dynamics, to ensure the convolution process produces a balanced and subtle result, and before the Limiter, which needs to be the last plug-in in the chain.

5 Limiter:

The Noiz-Lab LE Limiter is a brickwall limiting plug-in that lets you maximize the loudness of your track. Although most of the trickery involved in reducing limiting artifacts—like distortion and breathing-and-pumping—is kept behind the scenes, you can choose between a fast, medium and slow auto-release, and between different modes to boost the transients in your track.

With the maximize control turned on, turning down the limiter’s threshold will maximize your track’s volume up against the ceiling value. However, with the maximize control turned off, the volume will not be automatically boosted. Preventing loudness bias like this will make it easier to hear any degradation caused by the plug-in. The Limiter plug-in is fifth and last in the chain, after all other optimizations have been done, to minimize artifacts and degradation, and to ensure your track will not clip on the master bus.

PREPARE YOUR MASTERING SESSION

Once you have familiarized yourself with the Noiz-Lab LE plug-ins and workflow, it is time to render your track and set up your mastering session. If you want to keep to the default recommendations of this book, remove all plug-is on the master bus of your mix project, make sure your track doesn’t clip, and render your track as a stereo WAV or AIFF file on a samplerate of 44.1kHz and a bitdepth of 24bit. Then, open a new project in your sequencer or audio editor for your mastering session, and set this project to 44.1kHz and 24bit as well. Import your track and reference tracks on different channels, stacked vertically for doing instant comparisons using the solo option. If your reference tracks have a different samplerate and bitdepth than the project, it is fine to let your

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sequencer or audio editor convert them to the project settings. Create a group channel and route your reference tracks to it, so you can control their volume with just one fader. As they are probably already mastered, lower their volume a little to create some headroom, say by about 12dB. Then, use the reference tracks’ individual faders to match their loudness to each other by ear. And finally, use the reference tracks’ group channel fader to match their volume to the track you are about to master. Although your mastering session is now set up, after repeatedly adjusting the volumes of your reference tracks, the chances are you have lost your objectivity. Therefore, before you actually start your mastering session, it is a good idea to first take a short break, and then re-calibrate your brain by listening to your reference tracks for about 10 to 15

minutes at a comfortable volume. After doing that, let’s get started.

STEP 1: MONOMIZER

Goals:

● Removing DC offset dynamically over time

● Tightening up the low end stereo image Load up the Monomizer plug-in on your track’s channel, making sure it is the first plug-in in the mastering chain. You are going to use the Monomizer plug-in to remove potential DC offset, and to tighten up the low end stereo image of your track. Removing DC offset is usually not necessary, except when the offset is more extreme, say a few percent or more. Most sequencers and audio editors have an analysis tool that shows you the DC offset of your track. If the analysis shows a significant offset, you can simply activate the Monomizer’s DC offset filter to compensate for this.

Tightening up the low end of your track, on the other hand, can be more of a challenge, because the low end generally poses the most issues for home studio users. For the purpose of this first step, therefore—and for doing any low end processing during this mastering session—keep the following recommendations in mind:

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PART | III One Hour Mastering Workshop FIGURE 14.1 Noiz-Lab LE Monomizer plug-in

● Find out the range or frequency response of your speakers, so you are aware of any limitations that could cause trouble. If you followed the default recommendations of this book, you are using decent near-field speakers with 6- to 8-inch woofers.

Such speakers will usually go down to about 40Hz, which is certainly manageable if you use the remaining workarounds in this list as well.

● Double-check all low end processing you do on a professional pair of headphones which has a frequency response down to 20Hz (the low limit of the human audible frequency spectrum).

Using a pair of headphones like this might not give you a highly

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accurate response in the sub-low end, but it will clearly reveal issues otherwise obscured by acoustic problems in your room or by the limitations of your speakers.

● When working on the low end, turn up the volume. Listening at a higher volume will exaggerate the low end (and high end) in a psycho-acoustic manner, which makes it easier to analyze the low and sub-low frequency range. If you are only concerned with the low end, you can insert an instance of the Equalizer plug-in after the Monomizer and filter out the high frequencies with a low pass filter. This will make analyzing the low end much easier, especially in the sub-low range.

Before choosing your settings for the Monomizer, it is a good idea to compare the low end stereo image of your track to that of your reference tracks, so you have a clear reference. Primarily use your headphones for this with the volume turned up, but double-check your findings on your speakers as well. For a more detailed analysis, experiment with a temporary low pass filter to isolate the low end for playback. In general, commercial tracks featuring drums and bass will be mostly mono in the low end from about 100Hz, and fully mono approaching the low limit of the human audible frequency range of 20Hz. That is, when no stereo effect is used to deliberately widen the bass or bass-synth parts. On the other hand, commercial tracks that do not feature drums and bass, or put less emphasis on the low end, will usually roll off the low end, to clean up the track. If this is the case, it still is a good idea to analyze the low range of any lead instruments like guitar, piano or synth, as their fundamental frequencies might be low enough to make it worthwhile to slightly narrow their low end stereo image. Go through the following steps:

● To get a feel for the Monomizer effect, turn up the volume of your track, set the low frequency control to 20Hz and the high frequency control to 320Hz, and turn the make mono knob slowly to 100%. You will notice that any mix elements in the low end gradually move closer to the center of the stereo

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PART | III One Hour Mastering Workshop image, due to a summing-based stereo imaging algorithm. Skip through the different shapes to hear how they affect the stereo-to-mono transition along the frequency spectrum. The differences between shapes are subtle, and therefore best detected by using headphones and a low pass filter to isolate the low end.

● Now, determine if your track actually benefits from narrowing the low end stereo image. If the low end already is predominantly mono, this might not be the case. However, if the low end has a lot of stereo information, or the fundamental frequency ranges of any lead instruments spill down to the low end, your track might benefit from a tighter low end stereo image, as this will stabilize your track’s overall stereo image and possibly the tonal balance as well. In that case, proceed with the next two steps.

● If you already like your track’s low end stereo image compared to your reference tracks, or you have trouble determining the best course of action, it is best to take a conservative approach.

Set the low frequency knob to 20Hz, the high frequency control to 100Hz and the shape to S-curve. Then slowly increase the make mono control up to a maximum of about 50%, for a subtle stereo-to-mono effect. As frequencies below 100Hz are generally perceived as mono anyway, this will add a little more stability to your file, without the risk of harming your track.

● If the low end of your track sounds a bit messy compared to the reference tracks, raise the low frequency to 40Hz, raise the high frequency to about 160Hz and raise the make mono control to a maximum of about 75%. In addition, experiment with the different shape settings to find the best stereo-to-mono transition along the frequency spectrum. Be careful not to take it too far, though, as this might cause your track to lose power and punch.

For more extreme settings like this, make sure you carefully compare your track to your reference tracks.

As mentioned at the start of this step, the low end generally poses the most issues for home studio users. However, if you are

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uncertain about your Monomizer settings at this point, don’t worry too much. We will revisit the low end several times during the remaining steps of this mastering course. In addition, the remaining steps will require you to listen to your track in different ways, gradually revealing the best general approach to mastering your track. In that regard, the chances are that you will go back and forth a few times between plug-ins before reaching the end of this mastering course.

STEP 2: EQUALIZER

Goals:

● Optimizing the tonal balance

● Improving the stereo image

Load up the Equalizer plug-in on your track’s channel, making sure it is inserted in the mastering chain after the Monomizer.

Because the Equalizer plug-in can process in stereo, mid and side mode, you are going to use it for both optimizing the tonal balance of your track, as well as improving its stereo image in a very natural way. The biggest challenge with making these optimizations in a home studio, as explained in the previous paragraph, is controlling the low end. Particularly when it comes to sub-heavy genres like Hip-hop and EDM, you need to take special care, because it is usually the well-balanced sub-bass that gives these genres their powerful sound. As controlling the low end will make all other processing techniques to come more predictable, and therefore easier to perform, that’s where you will start. Before touching any of the plug-in controls, take some time to compare the low end of your track to that of your reference tracks again. As with the Monomizer, primarily use your headphones for this with the volume turned up, but double-check your findings on your speakers as well. Then, go through the following steps:

● Select the outer-left filter, set the filter type to high pass and the processing mode to stereo. If you move the frequency up

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PART | III One Hour Mastering Workshop FIGURE 14.2 Noiz-Lab LE Equalizer plug-in and down the spectrum, either by dragging the node or the frequency control itself, you will gradually hear the low, then the mid and finally most of the high frequency content in your track disappear. Using your headphones, set the frequency to 20Hz, and listen carefully to how the low end timbre changes, when you very slowly raise the frequency to a maximum of about 100Hz, and see if you can match the timbre of your reference tracks.

● If you are somewhat experienced, and have a nicely treated room and a good set of speakers or headphones, I recommend being conservative at this point, especially with sub-heavy genres. Set the frequency between about 20Hz and 30Hz, but

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make sure you are not losing any power due to the roll-off of the filter. This will clean up the low end slightly, and increase the potential maximum loudness of your track by removing any sub-low frequencies that may have very high amplitudes.

● If you are not that experienced and don’t know what to listen for exactly, I recommend being a little more aggressive to be on the safe side. Set the frequency between about 30Hz and 40Hz, to make sure you won’t encounter any problems when playing your track on bigger systems. For tracks without (kick-)drums and bass, like a singer-songwriter song or piano ballad, you probably want to go a little higher, up to about 50Hz. Make sure that you are not losing any warmth. Don’t worry about any remaining uncontrolled low end in comparison to the reference tracks at this point, as you are going to have a few more opportunities to compensate for this.

Next, to ensure your track will translate well to the broadest possible variety of audio systems—from tiny mobile phone speakers to huge PA systems—you will need to make sure your track’s overall tonal balance matches your reference tracks and has no significant inaccuracies. First, take some time again to compare the tonal balance of your track to that of your reference tracks, primarily using your speakers. Do make sure that your reference tracks are accurately level matched to your own track, to avoid any bias caused by the psycho-acoustic relationship between loudness and tonal balance. Start by comparing the tracks in regular stereo mode, to determine if the overall tonal balance differs from the reference tracks in any significant way, and if actual corrections and improvements to the overall tonal balance are needed. Then, use the solo button on top of the equalizer plug-in to listen to the isolated mid and side channel of your track. To compare your track’s mid and side channels to that of your reference tracks, insert a temporary instance of the Equalizer plug-in on the reference channel’s group channel, to switch between playback modes for all reference tracks at once. Comparing the mid and side channels will

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PART | III One Hour Mastering Workshop give you a good indication of how the stereo field of the reference tracks transitions along the frequency spectrum. Just as with the overall tonal balance, try to determine how your track’s mid and side channels differ from that of your reference tracks, and if there are opportunities to enhance the stereo image. To create an initial balance, go through the following steps and make the necessary adjustments:

● Compare your track’s overall tonal balance to that of the reference tracks in regular stereo mode, and listen to the low and high end. If your track sounds overly dull or bright in comparison, use a stereo low and high shelf to create a general balance. Make sure the curves of the shelving filters are smooth to avoid any resonant characteristics. If your mix is decent, you shouldn’t need to boost or cut the stereo shelving filters more than about 1dB to 2dB.

● Listen to the mid channel of your track, and use the high and low shelf filters to make sure there is enough low end present, and no excess of high end present. In addition, listen to the side channel of your track, and make sure the opposite is true: enough high end, and no excess of low end. If your mix is decent, you shouldn’t need to boost or cut the mid or side shelving filters more than about 2dB to 3dB. Apart from improving the tonal balance, the combination of these techniques can improve your track’s stereo image in a very natural way.

● Carefully level match and compare your track to the reference tracks again, both on your speakers as well as on your headphones, to make sure your settings have the intended effect.

Although you might need to revisit your settings later on, this initial general balance will provide you with a solid foundation for making any other adjustments to the tonal balance.

Now that a general tonal balance has been established, determine if your track contains any unwanted or resonant frequencies, or lacks certain important qualities. For example, your track might sound overly muddy or boxy, or nasal or harsh in comparison to

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your reference tracks, or may lack power, warmth, fullness or definition. For every problem you define, go through the steps below.

In case you have used most of the Equalizer’s filters already, it is best to insert an additional instance of the plug-in, to keep the different equalization phases discussed in this paragraph separate and accessible.

● Find out if the problem is in the stereo, mid or side channel of your track. If the problem is in the mid or side channel, solo the channel so you can more accurately hear what you are doing.

Select one of the filters and set its filter type to peak. In addition, make sure its processing mode is set to the channel you want to affect. Then, create a boost by raising the node of the peak filter to the top of the window. Move the node slowly to the left and right until you find the target frequency.

● Once you have identified the target frequency, set the Q-value with the goal to isolate the problem area without affecting the rest of the channel. As you want your filter to have a smooth curve to avoid any resonant qualities, a Q-value of 1.0 or less is generally considered appropriate for mastering. If you have a very specific problem, however, you can try going up to a maximum of about 3.0 for a more surgical approach.

● After setting the Q-value, decide on how much you want to boost or cut the target frequency. After boosting the filter to find the frequency, it is easy to fool yourself. Therefore, take a moment to listen to your reference tracks again, before applying the final gain setting. To avoid moving the handle sideways, and losing your focus on the frequency, it is easiest to drag down the dB edit box control instead of the node itself. If your mix is decent, you shouldn’t have to boost or cut more than a few dB. Actually, peak filter adjustments of only about 0.5dB

to 1dB are probably the most common in mastering.

After matching your reference tracks’ tonal balance, you might feel you are doing harm to your track, when you do a before-and-after comparison. If this is the case, think carefully about the goal

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PART | III One Hour Mastering Workshop of the mastering session. Your goal is to create a master that translates to a wide variety of audio systems, which requires a tonal balance without significant inaccuracies that holds to the conventions of its genre. In that regard, doing a before-and-after comparison is often irrelevant, as it tells you nothing about the objective balance of your track. A before-and-after comparison only tells you something about the relative balance of your track compared to an earlier version, which makes your assessment of the effectiveness of your settings extremely vulnerable to psycho-acoustic bias. Therefore, during this mastering workshop, try to mainly focus on the objective balance of your track in relation to your reference tracks, but without compromising on your artistic vision. In addition, keep in mind that this is only the second step of our mastering session, and that you are comparing your track to fully-mastered reference tracks. Therefore, don’t worry if your track doesn’t match up to your reference tracks just yet.

STEP 3: COMPRESSOR

Goals:

● Controlling dynamics

● Adding character

Load up the Compressor plug-in on your track’s channel, making sure it is inserted in the mastering chain after the Equalizer or multiple Equalizer plug-ins you have used during step 2. You are going to use the Compressor plug-in to control your track’s dynamics, and to add character by subtly applying the Compressor’s built-in saturation feature. The Compressor contains four compression styles that are discussed below. Try them all, and determine if they benefit your track. Keep in mind that both the compression and saturation effect will influence the volume of your track, and in many cases the perceived loudness as well.

Make sure therefore, to use the make-up gain on the plug-in to compensate for any volume loss. This will enable you to do an

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FIGURE 14.3 Noiz-Lab LE Compressor plug-in objective before-and-after comparison to see that your settings don’t cause distracting volume changes or have a negative effect on your track’s transients, as well as objectively to compare your track to your reference tracks.

● Compression for glue:

This technique will help to glue your file together and add some subtle fullness and warmth. It is a very basic technique, but it can add a nice bit of stability to your file. If you hear someone talking about general mastering compression, this technique is probably close to what they mean. Set the mode to glue. Play the loudest part of your file and pull down the threshold until you get a maximum of about 2dB to 3dB gain

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PART | III One Hour Mastering Workshop reduction. Experiment with the different saturation settings to soften the transients and add a little warmth to your track. Adjust the make-up gain and do a careful before-and-after comparison.

● Compression for balance:

This is a parallel compression technique that will bring out the softer parts of your file, which can be very helpful if you want to subtly reduce the dynamics of your file without making it sound unnatural. It is especially useful for guitar music and other acoustic genres, like singer-songwriter and classical music. Set the mode to balance. Play the loudest part of your track and pull down the threshold until you get a maximum of about 5dB gain reduction. Experiment with the different saturation settings to even out the dynamics a little more. Adjust the make-up gain and do a careful before-and-after comparison.

● Compression for punch:

This is a subtle technique to add some more punch to your track by giving the transients—like the kick and snare—a little boost.

It works very well for pop and rock music with a drum-heavy mix. Set the mode to punch. Play the loudest part of your track and pull down the threshold until you get a maximum of about 1dB to 2dB gain reduction. Everything except the transients will be brought down a little. Experiment with the different saturation settings to add a little grit to the signal. Adjust the make-up gain and do a careful before-and-after comparison.

● Compression for impact:

This is a parallel compression technique based on the New York compression principle. It is far more vigorous than the previous compression techniques and is perfectly suited to give loud and powerful music more impact and a modern and produced sound. Set the mode to impact. Play the loudest part of your track and pull down the threshold until you get an overly compressed sound where the transients begin to sound explosive. Experiment with the different saturation settings to even out the dynamics a little more. Adjust the make-up gain and do a careful before-and-after comparison.

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After performing objective before-and-after comparisons to make sure your compression settings don’t do any harm to your track, as well as objective comparisons with your reference tracks to make sure you hold to the particular genre’s conventions, you should have a clear idea if any of these four techniques are beneficial to your track. If more than one technique gives a positive result, load multiple instances of the Compressor in your chain and experiment with the order of compression techniques. Combining compression techniques like that is common practice, because applying several subtle techniques consecutively usually sounds more natural and controlled than applying one technique in a more extreme way. In addition, inserting an Equalizer instance after a Compressor to subtly compensate for the compressors coloration can help in achieving a controlled and balanced master. In general, for more powerful music genres, like rock, Hip-hop and EDM, you probably want to predominantly use the compression for impact or compression for punch technique, or a combination of both. On the other hand, for more subtle music genres, like singer- songwriter, folk and more classical-oriented styles, you probably want to predominantly use the compression for glue or compression for balance technique, or a subtle combination of both.

As you have noticed, so far you have just been making small improvements with every step, which is typical for a standard mastering session. There are several reasons for this. First, because you want your processing to be as transparent as possible, meaning that you want to get the intended results without the plug-ins themselves being noticeable. Second, as discussed previously in the context of compression, several subtle adjustments will almost always sound more balanced than one big adjustment. And finally, applying all these different techniques and listening in different ways will make it gradually more clear what the best approach is for mastering your track. After completing this step, your track should start to sound relatively balanced, both in terms of tonal balance as well as dynamics. This will make the result of the remaining two steps predictable and stable.

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STEP 4: CONVOLVER

Goals:

● Creating depth

● Adding character

Load up the Convolver plug-in on your track’s channel, making sure it is inserted in the mastering chain after the Compressor or multiple Compressor plug-ins you have used during step 3. You are going to use the Convolver plug-in to create depth and add character to your track by convolving different impulse responses (IRs) of real-world spaces with your track. The Convolver contains three different IRs, of a recording studio, a concert hall and a church, respectively. All three IRs are deliberately shortened to be FIGURE 14.4 Noiz-Lab LE Convolver plug-in

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suitable for mastering, and can be used in mono as well as stereo mode. In most cases you want to use the mono IR version, to preserve your track’s stereo image. However, if your track is mostly mono itself, using the stereo IR version can create a very natural stereo image in a way that is difficult to achieve with any other processing technique. Go through the following steps:

● To get a feel for the Convolver effect, set the mode to studio, set the high pass frequency to 20Hz and the low pass frequency to 20kHz. Then, press the solo effect option and turn up the amount to 100%. You are now hearing the fully wet convolved signal, as if your track was played in the actual room in which the IR was taken. Skip through the studio, hall and church mode, and switch between the mono and stereo versions of the IR to get a good feel for the different characteristics they provide, both in terms of tonal balance and reverberation. Use the filters to shape the tonal balance of the wet convolved signal, allowing you to blend in the wet signal with your track more seamlessly.

● Now, determine if your track actually benefits from using the Convolver. Because the convolution process imposes the acoustic properties of an actual space onto your track, it is ideal for enhancing the overall acoustic timbre of tracks that are purely produced in-the-box or recorded exclusively with close-miking techniques. Any of the three modes might provide a suitable character, and add the appropriate depth to your track.

For tracks that already have an outspoken character or sound appropriately spacious, you might want to try the studio mode, as it is a subtle IR that can add a nice subtle flavor or help with creating a loud master.

● If you feel one of the IRs benefits your track, solo the wet convolved signal and use the filters to shape the wet signal’s tonal balance. In most cases when using raw IRs, you want to take out some of the low end to avoid possible low rumble. Removing up to 100Hz should solve the low rumble, but if you are aiming for a specific timbre, you might want to raise the cutoff

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PART | III One Hour Mastering Workshop frequency to taste. The same goes for the high end. If you want the convolved signal to blend in nicely with your track, you probably want to take out some of the high end to make the IR

less bright, or lower the cutoff frequency to taste.

● If you have shaped the wet convolved signal to your liking, set the amount to 0% and un-solo the effect. Then, slowly mix in the wet convolved signal with your track. Because adding the convolved signal will raise the overall volume, make sure you compensate for this with the make-up gain. In addition, try both the mono and stereo version of the IR. In most cases the mono version will give the most stable results, but if your track has a narrow stereo image, the stereo version might give it a more spacious sound in a very natural way.

Because adding the wet convolved signal has changed the tonal balance of your track, you might want to compensate for this with a new instance of the Equalizer. Usually, applying a subtle low or high shelf filter, or both, should suffice. Only if the IR introduces some kind of specific coloration or resonance frequency, making a subtle cut with a peak filter might be appropriate. Make sure you keep primarily comparing your track to your reference tracks, instead of doing a before-and-after comparison with just your own track. In addition, in preparation of maximizing the loudness of your track during step 5, you can use the Equalizer to boost certain frequencies that are important for psycho-acoustic loudness as well. Be careful, though, not to disturb your track’s tonal balance too much. Go through the following steps:

● Select a peak filter and make a boost at 1,800Hz of about 1dB to 1.5dB, with a Q-value of about 0.5. This frequency is often boosted in rock and metal productions, for a loud and edgy sound. Make sure that your track doesn’t start to sound overly thin and sharp.

● Select a peak filter and make a boost at 5,000Hz of about 0.5dB

to 1dB with a Q-value of about 1.0. Humans are very sensitive to this frequency range, because this is where the intelligibility region of speech is found. Make sure that your file doesn’t start to sound overly bright.

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● If the low end is not very important to your track, select a low shelving filter and make a cut at 250Hz of about 0.5dB to 2dB

with a very smooth curve. While this doesn’t actually do anything at this stage, it will give the Limiter in step 5 a little more headroom, as low frequencies have high amplitudes and usually reach the threshold of compressor and limiter plug-ins first.

STEP 5: LIMITER

Goals:

● Maximizing loudness

● Preserving transients

FIGURE 14.5 Noiz-Lab LE Limiter plug-in

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PART | III One Hour Mastering Workshop Load up the Limiter plug-in on your track’s channel, making sure it is inserted last in the mastering chain. You are going to use the Limiter plug-in to maximize the loudness of your track, while at the same time preserving transients and minimizing limiting artifacts like distortion and breathing-and-pumping. The limiter has three modes that control the speed of its auto-release scheme: fast, medium and slow. If you are aiming for a competitive master, you want the auto-release to be as fast as possible without generating distortion artifacts. However, achieving this can sometimes be difficult when large amounts of gain reduction are necessary.

In that case you might have to compromise between a grittier but loud master, or a smoother but less loud master—either by choosing a slower auto-release, or by manually taking back the threshold a little. On the other hand, if you are aiming for a more conservative master, and therefore less gain reduction is needed, a medium or slow auto-release is usually more appropriate. This will require a slightly lower threshold to compensate for the slower auto-release, but will overall produce a smooth-sounding master without any distortion. Remember, however, that in addition to decreasing loudness, an auto-release that is too slow might cause breathing-and-pumping, especially in case of high amounts of gain reduction. Go through the following steps:

● To get a feel for the Limiter effect and its artifacts, set the mode to fast, the ceiling to 0dB and turn off the maximize control. Gradually lower the threshold from 0dB to −48dB, while watching the gain reduction meter. As the maximize control is turned off, the plug-in will essentially stay inactive, until the threshold gets low enough to catch the first peaks of your track. The meter might show a few dB of gain reduction, but the audible difference will be negligible. When lowering the threshold further, causing the plug-in to generate more and more gain reduction, you will start to hear increasing distortion and breathing-and-pumping artifacts. Skip through the fast, medium and slow mode, and switch between the different

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transient settings, to get a good feel for the Limiter’s inner workings, and to clearly identify the artifacts the plug-in produces with different settings when pushed hard.

● To have a real-world loudness reference, put your reference tracks back to their full volume. Take a moment to compare your track to them, to get a good idea of the target loudness you are after. Then, put the Limiter’s threshold back to 0dB, turn on the maximize control and set the ceiling to -0.3dB to compensate for intersample clipping and the potential volume boost created by converting to MP3 later on. Then, play the loudest part of your track, and slowly lower the threshold. As the maximize control is turned on, the volume of your track will now increase automatically. Try to reach your target loudness, without causing audible distortion and breathing-and-pumping.

Experiment with the different auto-release modes and transient options to find a good compromise between loudness and quality, while preserving or even boosting your track’s transients.

Looking at the gain reduction meter, you will probably find that you can get away with about 3dB to about 6dB of gain reduction without doing too much harm to your track. If you are aiming for a conservative master, this is usually enough. Generating more gain reduction, however, will quickly cause audible artifacts. If you are aiming for a competitive master, but you can’t reach your target loudness without doing harm to your track, you can make a few strategical tweaks to your mastering chain:

● Step 1: use the Monomizer to tighten up the low end stereo image some more, creating a more stable foundation for the other steps.

● Step 2: use the Equalizer to cut some lows with a shelf, creating more headroom for the Compressor and Limiter.

● Step 3: Use the Compressor to even out the dynamics of your track some more with the compression for impact technique, setting the saturation to hard. You might have to do some subtle

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PART | III One Hour Mastering Workshop damage at this point to give the Limiter in step 5 the necessary headroom.

● Step 4: Use the Convolver plug-in to add more of the wet convolved signal, increasing the density of your track, and the additional Equalizer to boost the frequencies important for psycho-acoustic loudness.

● Step 5: Lower the threshold to see if you can achieve your target loudness after tweaking the other steps in the chain. If you are still short a few dB, insert an additional instance of the Limiter before the current instance, set its mode to fast and use it to pre-clip the highest peaks in your track.

When aiming for a competitive master, usually you can’t avoid a compromise between loudness and quality. However, using a combination of techniques that increase both psycho-acoustic as well as actual loudness (and perhaps trying a few different brickwall limiter plug-ins) will generally get the job done. Once you have chosen your Limiter settings, you are ready to render your master as a stereo WAV or AIFF file of 44,1kHz and 16bit. However, if you are working on 24bit, it is advisable to first add a dither signal to your track, to avoid any low-level quantization distortion, caused by lowering the bitdepth. Such a dither function is standard in most sequencers and audio editors. After rendering, trim the silences and create fades at the start and end of your track, and you are done mastering your track.

IN CONCLUSION

Having gone through the whole mastering process, it is a good idea to review your approach and the results you have achieved in order to learn as much as you can from the experience. This is best done by playing your mastered track on many different audio installations to see if it holds up. Remember to always start by playing your reference tracks first, to calibrate your brain to a new room and playback system. It is deceptively easy to fool yourself if you don’t. Hopefully, you are happy with your track, and you now

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realize that mastering is not some kind of dark art, but instead a set of specific skills that can be studied and practiced. In that regard, the biggest challenge in mastering your own music in your home studio is understanding the mastering process and developing a mastering workflow that suits you. You need this understanding and methodology to be able to trust your ears and make clever decisions during any mastering session, but don’t forget that good taste and artistic vision are just as important. Once you have found a suitable workflow and know how you want your music to sound, mastering can quickly become second nature. So, learn and experiment all you can, become opinionated and develop a workflow of your own. Good luck mastering!

Appendix A

Mastering Dictionary

2-way system: A 2-way system is a term used to refer to a monitor with two drivers, usually a woofer and a tweeter.

3-way system: A 3-way system is a term used to refer to a monitor with three drivers, usually a woofer, a mid range driver and a tweeter.

absorption coefficient: Absorption coefficients give an indication of the effectiveness of an absorption panel or bass trap, usually defined for different frequencies along the human audible frequency spectrum.

absorption: Sound absorption is the result of sound waves being absorbed by relatively soft surface areas and being dampened.

absorption panel: An absorption panel is a type of acoustic panel optimized for absorbing sound waves.

acoustic treatment: Acoustic treatment is the practice of treating a room to optimize its acoustic properties. When a room is acoustically treated to make it suitable for critical listening, this usually involves placing bass traps, absorption panels and diffusion panels.

acoustics: Acoustics is the science that describes the behavior of sound.

ADC: ADC is the abbreviation of analog to digital convertor, and is the component in an audio interface that handles the conversion from analog to digital.

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AIFF: AIFF is a digital format for storing uncompressed audio, similar to WAV.

algorithmic reverb: An algorithmic reverb is a type of reverb that makes use of a number of delays with feedback, working together in parallel to produce a reverb tail of arbitrary length.

aliasing: The term aliasing refers to the creation of new frequencies when the frequency content of a signal exceeds the Nyquist frequency.

anti-aliasing: The term anti-aliasing refers to schemes that are developed to prevent aliasing when the frequency content of a signal exceeds the Nyquist frequency.

audio editor: An audio editor is a software application used for editing and processing audio, and a term often used to refer to DAWs specifically developed for the purpose of mastering audio.

bandwidth: Bandwidth is the unit for specifying the width of a filter.

bass port: A bass port is the port in a monitor used for increasing its low end frequency response.

bass trap: A bass trap is a type of acoustic panel or construction optimized for absorbing low frequency sound waves.

bitdepth: An audio waveform’s bitdepth can be viewed as the resolution of each individual sample. It tells you by how many different amplitude values a sample can be represented.

breathing-and-pumping: The term breathing-and-pumping is used to refer to the typical erratic volume changes that occur when misusing compression and limiting techniques.

broadband absorber: A broadband absorber is a type of acoustic panel optimized for evenly absorbing all frequencies of the human audible frequency spectrum.

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checksum: A checksum is the hash string that results from using hashing algorithms like MD5, sometimes used for doing a data-integrity check with a received DDP-file.

cloud: A cloud is an absorption panel or construction that is applied to the ceiling above the listening position.

comb filtering: Comb filtering is the coloration that occurs when an audio signal converges with a delayed version of itself.

convertor: A convertor is the component in an audio interface that handles the conversion between digital and analog.

convolution: Convolution is a mathematical technique used to convolve two signals with each other. In plug-in development convolution is used, for example, for creating convolution reverbs, cabinet emulations and linear phase filters.

convolution reverb: A convolution reverb is a type of reverb that makes use of an impulse response recorded on location, to impose the reverberation and character of an actual room on to your track.

correlation: Correlation is mathematical technique used to determine the similarity between two signals. In plug-in development this technique is used, for example, for creating phase correlation meters and FFT implementations.

correlation meter: A correlation meter is an analyzing tool that shows the phase relationship between the left and right channel of a track, usually in real-time.

crest factor: The term crest factor—sometimes referred to as short-term dynamic range—is used to refer to the real-time difference between a song’s peak volume and its average or RMS

volume.

critical listening: Critical listening is the practice of objectively listening to music and analyzing audio, requiring a neutral playback system as well as basic knowledge of psycho-acoustics.

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DAC: DAC is the abbreviation of digital to analog convertor, and is the component in an audio interface that handles the conversion from digital to analog.

dampening: The term dampening is used to refer to the absorbent quality that soft surfaces have on high frequencies.

DAW: DAW is the abbreviation of digital audio workstation. This term is traditionally used to refer to a complete digital studio, but nowadays is often used to refer to just the sequencer or audio editor software.

DC offset: The term DC offset is used to refer to the waveform displacement that occurs when the average value of a waveform’s samples amounts to a non-zero value

DDP-creator: A DDP-creator is a software application that lets you create a DDP-file from a set of mastered tracks.

DDP: DDP is the abbreviation of Disk Description Protocol, and is a virtual format that contains a Red Book compliant CD- montage, that can be used to send a mastered album over the Internet.

Decibel: Decibel is the unit used for specifying relative loudness while doing a peak or RMS loudness measurement.

delay compensation: Delay compensation is a DAW feature that allows you to automatically compensate for plug-in latency to ensure synchronization and prevent comb filtering.

depth: The term depth is used to refer to the acoustic information or reverb a song contains.

DFT: DFT is the abbreviation of Discrete Fourier Transform, and is a mathematical technique for analyzing a signal’s frequency and phase content. In plug-in development this technique is used, for example, for creating spectrum analyzers, vectorscopes and pitch-shifting and phase-shifting effects, as well as for implementing fast convolution. Usually, the FFT implementation of the DFT

is used to save CPU power.

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diffusion: Sound diffusion is the result of sound waves reflecting off irregular surface areas and scattering in different directions.

diffusion coefficient: Diffusion coefficients give an indication of the effectiveness of a diffusion panel, usually defined for different frequencies along the human audible frequency spectrum.

diffusion panel: A diffusion panel is a type of acoustic panel optimized for diffusing sound waves.

directional: The term directional is used to describe a sound wave that travels in a single direction.

Disk Description Protocol: The Disk Description Protocol is the protocol used for creating a DDP-file: a virtual format that contains a Red Book compliant CD-montage that can be used to send a mastered album over the Internet.

dither: Dither is a low-level noise signal that randomizes the quantization process of the least significant bit, and is added to a track when converting it from a higher to a lower bitdepth to avoid quantization distortion.

downsampling: The term downsampling is used to refer to the practice of internally lowering the samplerate of the audio processing within a plug-in, which is usually only done after upsampling first.

dynamic range: The term dynamic range is used to refer to the differences in loudness and intensity between different sections of a song.

dynamics: The term dynamics is used to refer to the differences in loudness and intensity of different sections of a song, and to the amount to which a song or particular parts are allowed to vary in loudness.

ear fatigue: The term ear fatigue is used to refer to the human brain becoming less perceptive to details in a track over time, especially when listening back at a high playback volume.

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EBU R128: EBU R128 is a loudness norm for broadcast audio that makes use of the LUFS standard.

echo perception threshold: The echo perception threshold is the minimum amount of delay that is needed to distinguish a sound wave from a delayed copy of itself.

equal loudness curves: The equal loudness curves are a series of measurements, taken by Fletcher and Munson, which show how sensitive human beings are to different frequencies on different volumes. The curves represent the adjustments that should theoretically be made to compensate for these differences.

equal loudness functionality: Equal loudness functionality is the functionality used by services like iTunes and Spotify to automatically adjust for loudness differences between different songs.

FET: FET is the abbreviation of field effect transistor, and is used to refer to FET-based analog compressors.

FFT: FFT is the abbreviation of Fast Fourier Transform, and is a fast implementation of the DFT used for analyzing a signal’s frequency and phase content.

Fletcher-Munson curves: The Fletcher-Munson curves—sometimes referred to as the equal loudness curves—are a series of measurements that show how sensitive human beings are to different frequencies on different volumes. The curves represent the adjustments that should theoretically be made to compensate for these differences.

flutter echo: Flutter echo is the rattling echo effect of high frequency sound waves reflecting between two parallel surfaces.

formants: Formants are specific frequencies that represent certain vowel sounds.

Fourier analysis: The Fourier analysis is a mathematical technique for analyzing a signal’s frequency and phase content. For digital signals the DFT version is used, which in plug-in development is usually implemented with the FFT algorithm to save CPU power.

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frequency response: The term frequency response is used to refer to the particular way a hardware device or plug-in changes the frequency balance of a signal going through.

full-band: The term full-band—sometimes called wide-band—is used to refer to plug-ins that affect a track as a whole, as opposed to the term multi-band which is used to refer to plug-ins that affect different frequency ranges separate from each other.

glass master: A glass master is the mold CD-plants use to produce retail CDs.

glue: The term glue is used to refer to the pleasant way different elements in a track glue together as the result of using certain plug-ins.

Golden Ratio: The Golden Ratio is a ratio with interesting mathematical properties that is traditionally considered to have a pleasing and artistic quality. In the context of audio production, the Golden Ratio can be used to calculate the ideal dimensions of a studio room with evenly dispersed standing wave frequencies.

Hertz: Hertz is the unit for specifying frequency as the number of sound vibrations per second.

host: Host is a term used to refer to an audio software application that is able to load audio plug-ins.

hybrid studio: A hybrid studio is an audio production studio that combines the smooth sound of analog gear and the flexibility of digital editing and processing.

impulse response: An impulse response is the response captured when sending a single peak value through a device, or generating a loud and transient sound in a room. For audio production IRs are usually stored as regular audio files that can be loaded in convolution plug-ins to impose a specific character on to your track.

integrated loudness: Integrated loudness is the LUFS value taken over an entire recording.

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intersample clipping: Intersample clipping is a form of clipping that can occur when an intersample peak larger than 0dB is generated by the interpolation filter that is used during DA-conversion.

intersample peak: An intersample peak is a peak that occurs between two digital sample values, due to the interpolation filter that is used during DA-conversion.

IR: IR is the abbreviation of impulse response, and is the response captured when sending a single peak value through a device, or generating a loud and transient sound in a room. For audio production IRs are usually stored as regular audio files that can be loaded in convolution plug-ins to impose a specific character on to your track.

ISRC codes: ISRC is the abbreviation for International Standard Recording Code, a code that is used for globally identifying music recordings

K-weighting: The term K-weighting is used to refer to the specific internal filtering applied in a LUFS measurement.

latency: The term latency is used to refer to the delay a plug-in introduces to the signal going through, expressed in milliseconds or samples.

least significant bit: The term least significant bit is used to refer to the specific bit that represents the lowest binary position in a binary number. The least significant bit is the first bit to be truncated when converting from a higher to a lower bitdepth.

linear phase filter: A linear phase filter is a filter that produces an identical phase shift over the entire frequency range while producing the intended frequency response, which is the equivalent of applying a regular time domain delay.

listening position: The listening position is the engineer’s position relative to the monitor set. Ideally, the engineer’s head forms an equilateral triangle with the monitors, ears at the height of the tweeters.

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listening triangle: The listening triangle is the equilateral triangle the engineer’s head should form with the monitors for critical listening.

LKFS: LKFS is the abbreviation of Loudness K-weighted relative to full scale, and is synonymous to LUFS.

loudness: The term loudness is used to refer to the way you acoustically and psycho-acoustically experience a song’s loudness.

loudness bias: The term loudness bias is used to refer to the human brain changing the perception of the tonal balance of a track when adjusting its playback volume.

loudness unit: The loudness unit, or LU, is the unit used for specifying relative loudness while doing a LUFS loudness measurement.

Loudness War: The Loudness War refers to the competitive battle between record labels to produce louder CDs than their competitors, after the discovery that listeners favored louder tracks over more quiet ones.

LSB: LSB is the abbreviation of least significant bit, and is used to refer to the specific bit that represents the lowest binary value in a binary number. The LSB is the first bit to be truncated when converting from a higher to a lower bitdepth.

LU: LU is the abbreviation of loudness unit, the unit used for specifying relative loudness while doing a LUFS loudness measurement.

LUFS: LUFS is the abbreviation of Loudness units relative to full scale. The LUFS standard was developed to provide a loudness measurement technique that closely resembles the way human beings experience loudness, with the goal of developing equal loudness functionality for broadcasting purposes.

mastering: Mastering is the final stage in the audio production process, where the overall tonal balance, width and depth, dynamics, and loudness of all songs on an album are optimized and

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matched to create an album that sounds great, translates well to a variety of audio systems and has a natural flow to it when listening to it from start to end.

mastering engineer: A mastering engineer is the engineer responsible for optimizing and matching the sonic aspects of different songs on an album, as well as correctly delivering the master to the CD-plant for replication or duplication.

MD5: MD5 is a hashing algorithm to generate a checksum, sometimes used for doing a data-integrity check with a received DDP-file.

mid-side: Mid-side is a technique to separate a stereo signal into a mid signal that is equivalent to a mono version, and a side signal that contains the audio information from the left and right channels that is different in volume or out of phase.

minimal phase filter: A minimal phase filter is a filter that produces only the necessary phase shift in order to produce the intended frequency response.

mixing: Mixing is the stage in the audio production where the various recorded instruments for a song are combined and processed to form a balanced whole.

mixing engineer: A mixing engineer is the engineer responsible for combining the various recorded instruments for a song to form a balanced whole.

momentary loudness: Momentary loudness is the LUFS value taken over a 400ms window.

mono compatibility: Mono compatibility describes the extent to which a track stays intact when converted from stereo to mono.

MP3: MP3 is a digital format for compressing and storing audio.

MS: MS is the abbreviation of mid-side, and is a technique to separate a stereo signal into a mid signal that is equivalent to a mono version, and a side signal that contains the audio information from

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the left and right channels that is different in volume or out of phase.

multi-band: The term multi-band is used to refer to plug-ins that affect different frequency ranges separate from each other, as opposed to the terms full-band and wide-band that are used to refer to plug-ins that affect a track as a whole.

neutral playback system: A neutral playback system is a playback system that is optimized for critical listening.

noise reduction coefficient: The noise reduction coefficient is a single value that gives an indication of the general effectiveness of an absorption panel.

NRC: NRC is the abbreviation of noise reduction coefficient, and is a single value that gives an indication of the general effectiveness of an absorption panel.

Nyquist frequency: The Nyquist frequency is the frequency above which aliasing occurs and can be calculated by dividing the samplerate of a signal by two.

Nyquist-Shannon sample theorem: The Nyquist-Shannon sample theorem states that an analog signal can be accurately captured digitally, when sampled at a samplerate that is at least twice as high as the highest frequency in the signal.

omnidirectional: The term omnidirectional is used to describe a sound wave that travels in all directions.

Opto: The term Opto refers to the light sensor detection scheme used in Opto based analog compressors.

oversampling: The term oversampling is used to refer to the practice of internally raising and lowering the samplerate of the audio processing within a plug-in, to increase a plug-in’s accuracy or prevent anti-aliasing.

peak: The term peak is generally used to refer to the visual peaks in an audio waveform, but specifically to refer to the highest sample value of all samples that make up a waveform.

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phase: The term phase is used to refer to the normalized position within a cyclical waveform and can be expressed in degrees or radians.

phase correlation meter: A phase correlation meter is an analyzing tool that shows the phase relationship between the left and right channels of a track, usually in real-time.

phase response: The term phase response is used to refer to the particular way a hardware device or plug-in changes the phase balance of a signal going through.

playback system: A playback system is the total of components that make up your listening environment and influence the listening experience. In a home studio set up for critical listening, this usually means a studio room, acoustic treatment, a computer, an audio interface and a monitor set.

plosives: The term plosive is used to refer to the sudden low frequency boost that occurs with speech sounds producing small bursts of air, such as: “b” and “p.”

PQ codes: The PQ codes store the start and end points of the tracks on a Red Book compliant CD.

psycho-acoustics: Psycho-acoustics is the science that describes how the human brain processes sound.

Q: Q is short for quality, and is the unit for specifying the steepness of a filter.

Quality: Quality is the unit for specifying the steepness of a filter.

quantization: Quantization is the practice of quantizing sample values to the resolution determined by the bitdepth.

quantization distortion: Quantization distortion is a form of low-level distortion caused by truncating a signal to a lower bitdepth.

Red Book Standard: The Red Book Standard is the standard that defines the specifications for creating audio CDs.

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reflection-free zone: A reflection-free zone is created by applying absorption panels at the first reflection points on the walls and ceiling around the listening triangle to reduce comb filtering at the listening position.

reverberance: Reverberance is the effect of sound slowly dying out, by repeatedly reflecting off the walls, floor and ceiling of a room.

RFZ: RFZ is the abbreviation of reflection-free zone, and is created by applying absorption panels at the first reflection points on the walls and ceiling around the listening triangle to reduce comb filtering at the listening position.

RMS: RMS is the abbreviation of root mean square, which is a method to calculate the average volume of a signal. For the purpose of audio production, usually the mean is calculated over a 30ms window, which produces a measurement that resembles the way human beings perceive loudness.

room acoustics: Room acoustics is the science that describes the behavior of sound in enclosed spaces.

RT60 standard: The RT60 standard is a standard that defines the reverb time of an algorithmic reverb as the amount of time it takes for the reverb signal to drop 60dB.

sample: A single measurement in an audio waveform.

sample theorem: The sample theorem, as defined by Nyquist and Shannon, states that an analog signal can be accurately captured digitally, when sampled at a samplerate that is at least twice as high as the highest frequency in the signal.

sample value: The value of a single measurement in an audio waveform.

samplerate: An audio waveform’s samplerate can be viewed as its resolution over time. It tells you the number of samples used to represent 1 second of audio.

206

Mastering Dictionary Appendix A

scattering coefficient: Scattering coefficients give an indication of the effectiveness of a diffusion panel, usually defined for different frequencies along the human audible frequency spectrum.

sequencer: The term sequencer traditionally refers to a device or software application that is able to record and playback a sequence of sounds, but is nowadays used to refer to DAWs specifically developed for the purpose of recording and mixing audio.

short-term dynamic range: The term short-term dynamic range—

sometimes referred to as crest factor—is used to refer to the real-time difference between a track’s peak volume and its average or RMS volume.

short-term loudness: Short-term loudness is the LUFS value taken over a 3,000ms window.

sibilance: The term sibilance is used to refer to the hiss-like character of speech sounds, such as: “s,” “sh,” “z” and “zh.”

soundproofing: Soundproofing is the practice of treating a room to make it soundproof.

spectrogram: A spectrogram is an analyzing tool that shows the frequency content of a track over a specific time period.

spectrum analyzer: A spectrum analyzer is an analyzing tool that shows the frequency balance of a track, usually in real-time.

standing wave: A standing wave is a waveform with a wavelength that matches one of the dimensions in a room and therefore gets amplified.

stem: A stem is one of the separately rendered layers of a mix used for stem mastering.

stem mastering: Stem mastering is a style of mastering whereby a mix is rendered in separate layers, called stems, to enable the mastering engineer to apply more specific or surgical processing.

Mastering Dictionary Appendix A 207

tonal balance: The term tonal balance is used to refer to the way a track’s frequency content is distributed along the frequency spectrum.

tonal balance normalization: The term tonal balance normalization is used to refer to the human brain normalizing the perception of the tonal balance of a track over time.

transfer engineer: A transfer engineer is the engineer traditionally responsible for transferring tape recordings on to vinyl records.

transient: The term transient is used to refer to the initial attack of sounds with a percussive quality.

true peak: The term true peak is used to refer to a peak measurement that includes the intersample peaks that might occur due to the interpolation filter that is used during DA-conversion.

truncation: Truncation is the practice of naively discarding bits when converting from a higher to a lower bitdepth, risking quantization distortion.

tuned trap: A tuned trap is a type of bass trap that is designed to reduce specific low frequency sound waves.

upsampling: The term upsampling is used to refer to the practice of internally raising the samplerate of the audio processing within a plug-in.

Vari-Mu: The term Vari-Mu refers to the variable gain scheme used in analog tube compressors.

VCA: VCA is the abbreviation of voltage controlled amplifier, and is used to refer to VCA-based analog compressors.

vectorscope: A vectorscope is an analyzing tool that shows the phase balance of a track, usually in real-time.

WAV: WAV is a digital format for storing uncompressed audio, similar to AIFF.

208

Mastering Dictionary Appendix A

wavelength: The wavelength is the distance a cyclical sound wave travels through the air until it has completed a full cycle.

wide-band: The term wide-band—sometimes called full-band—is used to refer to plug-ins that affect a track as a whole, as opposed to the term multi-band which is used to refer to plug-ins that affect different frequency ranges separate from each other.

width: The term width is used to refer to the way different elements in a song are placed in the stereo field.

zero phase filter: A zero phase filter is a filter that produces no phase shift while producing the intended frequency response, which is not possible in real-time without using the DAW delay compensation feature.

Index

Term:

Paragraph:

Page:

2-way system

4.4

43

3-way system

4.4

43

absorption coefficient

3.2

25

absorption panel

3.2

24

acoustic treatment

3.1

23

acoustics

3.1

23

AIFF

6.2

64

algorithmic reverb

10.6

123

aliasing

7.2

77

analyze function

8.4

89

artificial reverberation

10.5

122

attack time

11.2

132

attenuation meter

11.2

135

audio editor

5.1

47

audio interface

4.1

37

auto-release

12.5

153

balance

11.5

139

band pass filter

9.3

103

bass trap

3.3

27

bell filter

9.3

103

bitcrusher

7.2

78

bitdepth

7.2

75

breathing-and-pumping

5.7

58

brickwall limiting

12.5

152

broadband absorber

3.3

28

C1 error

13.4

163

C2 error

13.4

163

CD text

13.3

160

ceiling

12.5

153

character equalization

9.8

111

checksum

13.4

164

clicks

8.5

90

209

210

Index

cloud

3.4 31

comb filtering

3.4

30

compression

11.2

131

convolution reverb

10.7

126

CPU

4.2

39

crackle

8.5

90

crest factor

11.1

129

Cu error

13.4

163

DA-conversion

4.3

41

dampening

10.6

124

DAW software

1.3

9

dBFS limit

5.7

58

DC offset

8.2

86

DDP-file

13.4

163

decay time

3.6

34

delay time

3.4

30

de-noiser

5.3

49

depth

10.1

115

diffusion panel

3.2

24

directional

3.1

24

Disc Description Protocol

13.4

164

dither

13.2

159

downsampling

7.2

77

dynamic equalization

11.6

141

dynamic range

11.1

129

dynamics

11.1

129

ear fatigue

6.3

66

echo perception threshold

3.4

30

equal loudness functionality

1.3

9

equalizer

9.3

102

fade

7.5

81

feedbacking delay element

10.5

122

FireWire

4.3

41

Fletcher-Munson curves

6.3

67

flutter echo

3.5

32

focusing technique

9.6

108

formants

9.1

98

frequency range

9.1

97

frequency response

4.4

42

Index

211

gain reduction

5.6

56

glass master

1.3

8

glue

11.5 139

Golden Ratio

2.2

17

ground hum

8.3

88

group channel

7.6

83

hash-value

13.4 164

headroom

7.3 78

high cut filter

9.3

103

high pass filter

9.3

103

high shelf filter

9.3

103

hiss

8.4 88

hybrid studio

4.1

38

impact

11.4 137

impulse response

5.5

54

individual volumes

12.1

143

internal side-chain

11.2

134

IR

5.5 54

ISRC codes

13.3

161

learn function

8.4

89

least significant bit

13.2

159

linear phase

5.4

51

listening position

2.3

19

loudness bias

6.3

66

Loudness War

1.3

9

loudness

12.2 146

low cut filter

9.3

103

low end control

9.5

106

low pass filter

9.3

103

low shelf filter

9.3

103

lsb

13.2 159

LUFS meter

12.5

154

make-up gain

11.2

133

master-CD

13.4 163

mastering engineer

1.3

8

mastering workflow

6.4

67

memory

4.2 39

mid-channel

10.4

120

212

Index

mid-side based stereo

imaging

10.4 120

mid-side

10.4 120

mixing engineer

1.3

8

monitors

4.4 42

mono compatibilty

7.4

80

MP3

6.2 64

multi-band compression

11.6

141

multi-band

5.2 49

near-field monitor

4.4

42

neutral playback system

2.1

15

noise profile

8.4

89

noise reduction coefficient

3.2

26

noise shape

13.2

159

noise type

13.2

159

noise

8.4 88

NRC

3.2 26

Nyquist-Shannon sampling

theorem

7.2 76

omnidirectional

3.1 24

over-compression 11.3 135

oversampling

7.2 77

paragrahpic equalizer

5.4

51

parallel compression

5.6

56

paramatric equalizer

5.4

51

peak filter

9.3

103

peak mode

5.6

56

phase correlation

5.5

54

phase difference

3.4

30

pitch-shifter

5.2 49

plate reverb

10.5

122

playback system

1.1

3

plosives

8.6 92

pop-filter

8.6

92

PQ codes

13.3

161

pre-delay

10.6 124

pre-mastering

1.3 8

psycho-acoustics 6.3 65

punch

11.4 136

Index

213

Q

9.3 103

quality

9.3 103

quantization distortion

13.2

159

RAM

4.2 39

ratio

11.2 132

reference tracks

6.2

64

reflection-free zone

3.4

31

release time

11.2

132

render section

7.5

81

resonance

2.2 19

reverb time

10.6

124

reverberance

3.6 34

RFZ

3.4 31

RMS meter

5.7

60

RMS mode

11.2

133

roll-off

.2

87

room size

10.6

124

RT60 standard

10.6

124

sample

7.2 76

samplerate conversion

13.2

158

samplerate

7.2 75

saturation

5.6 58

sequencer

5.1 47

shelf filter

9.3

103

sibilance

8.6 92

side-chain

8.6

93

side-channel

10.4

120

soft knee

11.2

132

soft-clipping

5.4

53

soundproofing

3.1

23

spectrogram

8.5

92

spectrum analyzer

9.2

100

spring reverb

10.5

122

standing wave

3.3

27

stereo imaging

10.2

117

stereo link

11.2

134

stereo width

10.6

125

summing-based stereo

imaging

10.2

117

sweet spot

2.3

20

214

Index

tailroom

7.5 81

threshold

11.2 132

ThunderBolt

4.3 41

timbre

9.1 97

tonal balance normalization

6.3

65

tonal balance

9.1

97

track order

12.1

143

transfer engineer

1.3

8

transient designer

5.6

58

transient

11.1 130

true peak meter

5.7

59

truncation

13.2 159

tuned trap

3.3

29

upsampling

7.2 77

USB

4.3 41

vectorscope

5.5 54

WAV

6.2 64

wavelength

3.3 27

width

10.1 115

zero line

8.2

86

Document Outline

Cover

Title

Copyright

Contents

Preface

Acknowledgments

Part I Setting Up Your Home Studio

1 Mastering

1.1 Can you Master Your Music in a Home Studio?

1.2 What is Mastering?

1.3 A Bit of History

1.4 Managing Expectations

2 Your Own Mastering Studio

2.1 Make Smart Choices

2.2 Room Characteristics

2.3 Arrangement

2.4 Don’t Panic Please

3 Room Acoustics

3.1 Common Acoustic Problems

3.2 Acoustic Treatment

3.3 Standing Waves

3.4 Comb Filtering

3.5 Flutter Echo

3.6 Reverberance

4 Hardware

4.1 A Basic Digital Setup

4.2 Computers

4.3 Audio Interfaces

4.4 Monitor Sets

4.5 Headphones

5 Software

5.1 Audio Editor or Sequencer?

5.2 Mastering Plug-Ins

5.3 Restoration

5.4 Tonal Balance

5.5 Width and Depth

5.6 Dynamics

5.7 Loudness

6 Mastering Workflow

6.1 Evaluating Your Setup

6.2 Choosing Reference Tracks

6.3 The Art of Listening

6.4 General Workflow

Part II Home Studio Mastering Techniques

7 Preparing Your Session

7.1 Finalizing Your Mix

7.2 Choosing Samplerate and Bitdepth

7.3 Setting Output Volume and Headroom

7.4 Checking Mono Compatibility

7.5 Checking Render Section and Fades

7.6 Setting up Your Mastering Project

8 Restoration

8.1 Removing DC Offset and Unwanted Noises

8.2 DC Offset

8.3 Ground Hum

8.4 Noise and Hiss

8.5 Clicks and Crackles

8.6 Sibilance and Plosives

9 Tonal Balance

9.1 Frequency Distribution

9.2 Frequency Analysis

9.3 Mastering Equalization Basics

9.4 Optimizing Tonal Balance

9.5 Low End Control

9.6 Corrections and Improvements

9.7 Match Your Tracks

9.8 Enhance Overall Timbre

10 Width and Depth

10.1 Stereo Image and Reverberation

10.2 Stereo Imaging Techniques

10.3 Narrowing the Low End

10.4 Overall Mid-Side Stereo Enhancement

10.5 Reverberation Techniques

10.6 Algorithmic Reverb

10.7 Convolution Reverb

11 Dynamics

11.1 Dynamic Range and Crest Factor

11.2 Mastering Compression Basics

11.3 Optimizing Dynamics

11.4 Compression for Punch and Impact

11.5 Compression for Balance and Glue

11.6 Multi-Band Compression and Dynamic Equalization

12 Loudness

12.1 Track Order and Individual Volume

12.2 Maximizing Loudness

12.3 Loudness by Equalization

12.4 Loudness by Compression

12.5 Loudness by Brickwall Limiting

13 The Final Master

13.1 Housekeeping

13.2 Rendering Your Tracks

13.3 Creating a CD-Montage

13.4 Creating a Master-CD or DDP-File

Part III One Hour Mastering Workshop

14 One Hour Mastering Workshop

Download the Mastering Bundle

The Plug-Ins and 5-Step Workflow

Prepare Your Mastering Session

Step 1: Monomizer

Step 2: Equalizer

Step 3: Compressor

Step 4: Convolver

Step 5: Limiter

In Conclusion

Appendix A—Mastering Dictionary

Index