AUDIO MASTERING

IN A PROJECT STUDIO

A PRACTICAL APPROACH

FOR A PROFESSIONAL SOUND

SIMON TAYLOR

FOREWORD BY KEVIN PAUL

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First published in 2022 by

Inform Inspire Publishing

Enquiries: www.simontaylorauthor.com/contact

www.simontaylorauthor.com

© Simon Taylor 2022

All rights reserved. No part of this publication may be reproduced or transmitted in any form or by any means, electronic or mechanical, including photocopy, recording, or any information storage and retrieval system, without permission in writing from the publishers.

All trademarks and copyrighted material is reproduced under permitted 'fair use' license for educational purposes.

British Library Cataloguing-in-Publication Data

A catalogue record for this book is available from the British Library.

ISBN 978-1-7391670-0-4

Foreword image: Courtesy of Kevin Paul

Table of Contents

Title Page

Copyright Page

AUDIO MASTERING IN A PROJECT STUDIO: A PRACTICAL APPROACH FOR A PROFESSIONAL SOUND

FOREWORD

INTRODUCTION

1 | ASSUMPTIONS AND CONCEPTS

2 | MASTERING ROOM

3 | MONITORING AND LOUDNESS

4 | CRITICAL LISTENING

5 | BASE TO TOP LEVEL

6 | MASTERING TOOLS

7 | MASTERING TEMPLATE

8 | TEMPLATE IN USE

9 | EXPORTING

10 | MASTERING AN ALBUM

IN CLOSING

ABOUT SIMON TAYLOR

GLOSSARY

FOREWORD

Having known Simon for many years and worked with him on his own music where I was mixing and mastering on his behalf, I was delighted to be asked to write a few words about this book. Initially I was not sure how or what to write, but after reading it became clear what kind of book Simon has written. ‘Audio Mastering in a Project Studio’ is quite simply the definitive book on how to approach setting up your studio, learning to listen, approaching your music and learning how to master it. Mastering is a forgotten process, I feel, but it is fundamental to making a record and I always appreciate the mastering process and mastering engineers for not only their ears, but their taste and professionalism. This book does more than talk about mastering and that is vitally important in today’s singular world. LUFS, RMS, Peak SPL, K-System and many other acronyms are detailed here in a concise and easy to understand manner, allowing the reader to approach their mastering in a professional way. Of course, what makes the difference between one engineer and another is not only taste, but experience, and this book will provide you not only with a starting point, but also allow you to refer to it time and again to keep yourself refreshed and to make sure you fully understand technically what is going on. If I was looking at this book at the start of my career, I would have found it invaluable. I suggest you start from the beginning and follow Simon’s detailed steps and see where it takes you. I think you will find that not only will the room you use sound better, but your own skills as a mastering engineer improve dramatically. Learning how to listen is probably the most important skill of a mastering engineer and Simon’s approach and tips will no doubt put you in a better place when it comes to your own work. It is full of actual settings and starting points that you take and then adjust to taste, and taste is what music is all about. I found the entire book engaging and presented in a way that makes it seem as if Simon is right next to you walking you through the steps. Highly recommended and Simon, thanks for asking me to contribute. Good luck everyone and happy music mastering.

Kevin Paul

www.kevin-paul.com

INTRODUCTION

‘There is nothing permanent except change’

Heraclitus, Greek philosopher

The music business is a vast bustling community, an exciting evolving metropolis that dominates the social and economical landscape. It generates unimaginable wealth for individuals and conglomerates, whilst feeding consumers with as much content as they can devour. The music industry’s lifeblood is songs, with each composition representing a unique moment of inspiration for both creator and listener. Hearing beauty in sound is a uniquely human experience; the euphoric feeling of joy as sound waves crash over our senses. And whenever music is recorded at a professional or semi-professional level, audio mastering is required. It is the cherry on the top, the final sheen that brings out the musical colours in all their glory, a tried and tested method of presenting music in its best possible light.

The problem is mastering is one of the hardest processes to get right and the easiest to screw up! If you are a project studio owner (i.e. any recording and/or mixing environment that is not a professional establishment) and the master of your own creations, or collaborating with friends, you will know how hard it is for production qualities to live up to your expectations. You will know it is tough to match up to the best in the business – the pros that seemingly produce perfect masters every time; a constant flow of balanced, fantastic songs that sound wonderful no matter what environment they are played in. To me this is the greatest challenge for all project studio owners: to match their final output to that of the professional mastering engineer. This is a challenge that we are all capable of rising to with the correct approach, knowledge and workflow.

Music has always been my method of artful expression and the driving passion in my life. At an early age I became enthralled with the process of capturing sound and started recording rudimentary compositions. I have always had the gift of ‘head tunes’ whereby songs appear to me in my imagination. Writing music has never been difficult for me. The problems started when trying to present them at a professional standard that was acceptable to listeners and peers alike, so I enlisted the help of a mastering engineer. My initial experiences were as different as night and day. Firstly I had the misfortune of employing a poor mastering service. The results were mediocre at best. I remember thinking “is this right?” “Is this really what mastering does?” Then I hired the best mastering engineer I could find and was blown away by the transformation of my demo sounding mix to a commercial quality release. I soon realised that mastering comes in many grades of quality. I wanted the best but could not afford the best on every project and budget mastering results were poor. So I set to work on the mastering system that appears in this book, a mastering system based on over twenty years experience that I would like to share with you – a method of smashing the average mastering service out of the park and knocking on the door of the professionals. I have never paid for mastering again.

I understand the problems you are facing. It may seem right now that you are standing at the foot of a mountain looking up at the incredible heights that one must climb to get close to professional sounding masters. Indeed, you may be frozen into inaction by the thought of all the learning and tasks that culminate to form one solid production. You may be looking for answers to questions such as:

How do I master to a level that is comparable with the commercial releases I hear on the radio?

How do I retain an ongoing consistency of excellence when mastering?

How do I get the level of songs as loud as commercial releases?

How do I turn average mixes into quality masters?

What tools do I need to master audio?

Can I make top masters without spending thousands on analogue outboard equipment and the finest plugins?

Do I need an acoustically perfect room in which to master?

Do I need years of training before my ears can begin to compete with the best?

If you are in any way questioning your ability, your tools or your resolve then the answer is ‘yes’!

Yes you can compete with the best.

Yes you can master your tracks at home.

Yes you can master on any computer with any quality of plugins available to you.

Yes you can transform your average mix into a commercial sounding release.

And no, you do not have to spend a decade practising and training your ears to hear every nuance and subtlety.

This book will help you progress. I will guide you through the process. I am guessing that you want to master without wasting time and you want your music to be polished to a professional level, then this book is for you. If you have a busy life and want to reduce the time spent in production and simply get your music out there, this book is for you. Maybe you are a songwriter, a hook-and-groove creator who needs to produce content fast, this book is for you. Within these pages you will find the system you need, a system that will empower you to master within your own project studio. All you need is the knowledge contained within this book.

To reap the very best rewards, it is recommended that you work through each chapter one-by-one starting at the beginning. This is because every step is an important part of the complete whole. You will never regret the time spent laying down the foundations for your new mastering system. The closer you follow the guidelines and the more you practise this system, the greater the improvements will be. All the ensuing information is written in a ‘set and leave’ format, so once read and acted upon it will never need doing again. We will work through the chapters with a balanced, practical approach providing only the information you need to progress rapidly; an easy, workable, repeatable system that is accessible through logical steps no matter how advanced you are. Simply follow the methodology laid out in an easy step-by-step fashion and you will soon have better sounding masters, achieving a level in mastering that you thought was beyond your reach. Yes, you can master audio, let me show you how.

1

ASSUMPTIONS AND CONCEPTS

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When we as musicians first dream of writing and recording songs, we learn to play an instrument and practise singing oblivious to the many techniques and processes that each studio recording goes through. As time passes we become aware that our efforts do not sound the same as the songs we hear on our favourite compact disc, digital music file or vinyl record. We then look for the reasons why. We all follow more or less the same path of exploration and discovery. Some voyagers may sprint to their destination while others amble along taking in the view. Speed is irrelevant. All that matters is that the journey is an enjoyable ride.

This book talks about mastering from a project studio viewpoint; my goal is to make your personal journey as stress free and easy as possible. I have made the following assumptions as we start to work together to raise the quality of your mastered recordings:

You own a DAW (digital audio workstation)

You are comfortable with the DAW’s basic functions and can set up projects, tracks, channels and buses

You know how to add effects and processors (plugins)

You are a project studio owner and either a musician, songwriter, producer, mix or mastering engineer

You already have a basic understanding of how to mix musical elements together and want to master audio

You have a fundamental knowledge of audio editing and source files on which to practise

If you are only interested in mastering then dive straight in. If however, you are still learning to mix, then the perfect companion to this book is ‘Template Mixing and Mastering’ a title I co-authored with ‘A-list’ mix engineer Billy Decker. It will answer your mixing questions and give you a mix template that you can rely on to give professional results. You will see as we progress deeper into this book that the mixing and mastering processes can be interlinked, so a flexible mixing system with instant recall and total control is recommended.

This book is predominantly written with independent musicians, songwriters and project studio owners in mind; those who are looking to master their own songs simply, quickly and professionally. I define an independent musician as an artist not currently signed to a major record label and who does not have access to a professional recording studio. If you purely want to

study the art of mastering and become a professional mastering engineer then you will still find the information useful, but not an all-encompassing source. This book will serve as a great grounding for future studies and instil core values, but it does not look to educate to the level of a professional mastering engineer. Instead, it looks at the problems independent musicians encounter and offers solutions as to how you can master at home to a consistent, professional level, quickly and efficiently.

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THE PAST, PRESENT AND FUTURE OF MASTERING

You may have seen the quotation at the start of this book by the Greek philosopher Heraclitus stating ‘there is nothing permanent except change’. This is especially relevant to this book as we independent musicians find ourselves caught up in the digital revolution where a bewildering amount of opportunities beckon. All of us are constantly changing. Life around us is forever changing. We must change with it or accept that we will stay where we are and life will go on without us.

The music business is not exempt from change, either. When faced with extinction it has always moved with the times and survived. For every question asked, every challenge faced, the music business has always found the answers by adapting and embracing new ways, new technology and meeting new public demands. If we want to be a part of these brave innovative adventures, part of the changing face of the music business, then we must adapt too.

So where are you on the curve of progress? You may have been recording music for many years and feel that the current level of quality that is demanded of you is now out of your reach. This is not true. You may be excited about all the possibilities, but with so many options and so many roads you could go down, where do you start? I will simplify your roadmap. You may feel that mastering is a ‘dark art’ only achievable after a lifetime of application. Again, not true, you can get very close to the high bar of professional mastering by following the guidelines in this book. You may be wondering what has changed in the music business that has caused me to write this book now? Why are the contents of this book relevant to you and your musical output? Why should you read this book and follow its methodology? How will I help you to succeed and realise your dream? Well, three key developments have aligned themselves in the musical galaxy over recent years.

Firstly, the music business faced probably the biggest fight of its life when digital files became the mainstream medium for listening to tunes. The new dominant business model that was born from this challenge (alongside surviving traditional revenues such as live performance) is one of subscription-based music streaming services. Listeners can now gain access to a vast pool of content for a monthly fee. Today’s listener has become accustomed to a continuous feed of new content. A streaming service’s biggest fear is that subscribers will become bored and terminate the payments that drive their business. To overcome this, they have developed complex computer algorithms to ensure that each listener is kept entertained and engaged. The algorithm continually refreshes material and supplies new music daily to suit individual tastes. As an independent recording artist, to remain relevant and be of interest to your fan base, it is essential to adapt to listeners demands. This means keeping listeners interested and engaged with regular new material. It is imperative to release new recordings often and these recordings should be produced to a quality that is on a par with the commercial recordings released by major labels.

This brings us to the second development - advances in technology. How can you now produce commercial sounding recordings that would previously have required an army of studio employees and tens of thousands worth of recording gear to engineer? The answer of course is that computers, music technology and shared information have now developed to an amazing level at an astonishingly low cost. The tools required are now available to everyone. Everyone has a computer. Anyone can download a free or reasonably priced digital audio workstation and record, mix and master music. Indeed some of the same digital audio workstations that are used by professional engineers are available in free or entry priced ‘lite’ versions. These are perfectly usable with the right approach; and there is now an abundance of information to help the beginner or advance the intermediate creator. We are living in a glorious age of knowledge sharing, an arrangement that benefits all.

The third development is the upsurge of online algorithmic mastering services. If you are not familiar with online algorithmic mastering, it basically involves a remote computer mastering your work. The computer sits on the internet and waits for you to upload a mix. Once it receives your mix through a dashboard GUI, your file is mastered online without any human interaction. The computer does this through an algorithm (an algorithm is a set of instructions to perform a certain task). The algorithm analyses your mix and compares it to the benchmark references it has gleaned from thousands of songs that it holds in memory. Its job is to shape your mix into what matches many of the technical characteristics of a professional master. Depending on the service provider there may be settings you can tweak to influence the outcome. You may, for example, be able to increase the overall loudness or rebalance the high, middle and low frequencies. There is always a preview process where you can hear the mastered results and choose whether to buy the master or trash it. Many people ignore these services and regard them as poor quality, but as you will see later, I feel they have an important part to play in the learning of audio mastering. I mentioned above the glorious age of knowledge sharing, which is what I would like to do now. This book will empower you to master your music at home in an easy-to-follow step-by-step format. All you need to add is desire, endeavour, practise and critical listening. Ultimately this is what will elevate you from the pack. Sticking to the proven system in this book and practising your craft every day. This is what the most successful sound engineers have always done. They have found a system that works for them and applied it time and time again, day after day, on every song they work on. Walk with me through this book and you will have you very own fast and effective, repeatable mastering system; a system that will bring you closer than ever before to the quality of commercially released master recordings. You will then be confident in your ability; and confidence, speed and a repeatable quality system equal professionalism. Professionalism is what you need to be taken seriously.

At this point it may be worth taking a brief step back to look at the role that mastering and the mastering engineer have played in the development of recorded music. Understanding audio mastering and a mastering engineer’s role will help you to visualise what is expected of you as you fill their boots!

Mastering has always been the process of preparing audio to suit the playback medium. This is very important to understand. Audio must be mastered differently depending on its intended replay device. Originally mastering was a simple task with no real allocation of duties, and indeed not known as mastering at all. Musical performances were captured via a single microphone and transferred in real-time to an acetate disk that would then be used to produce a ‘stamper’ for the production of vinyl records.

Over the years the role of the mastering engineer has emerged as an important stand-alone function that has evolved and changed considerably. Nowadays it is not unusual to see a mastering engineer involved in mixing and creative decision-making, and while it is true that the mastering engineer has become a complex animal, the key fundamentals that we will focus on still hold true:

Mastering is the preparation of audio to suit the intended playback medium.

Audio should be mastered differently depending on its final destination.

Mastering is the final layer of production, the last chance to correct any problems and polish a mix to make it as good as it can possibly be.

A mastering engineer can take a recording to another level or degrade it to the realms of the recycle bin.

There are many informative books on audio mastering. Most are aimed at those wishing to become a fulltime professional mastering engineer. Little information exists on how to fast-track the process and obtain the best quality results in the minimum time without fuss and too much detail. I intend to focus on only what is relevant to you, what you need to know to release your music as swiftly as possible to a consistently high level. Absorb the previous bullet points and open your mind to what can be achieved with minimal resources and the right knowledge. You are the future of mastering.

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CHALLENGES, PROBLEMS AND SOLUTIONS

We have established that there always has been, and always will be, a mastering process to finalise recorded music; and that the music industry is constantly changing and with it the role of mastering. In order for us to output as much music as is now asked of us we must somehow perform the task of mastering and become mastering engineers ourselves. Accepting our situation is a big part of overcoming what lies ahead. It is now necessary for us to self-master our productions. No longer is the release of one album a year enough to satisfy demand, and independent artists simply cannot afford expensive professional mastering on every song. It is basic economics. Self-mastering is the answer.

Having said that, let us be absolutely clear, the best and easiest solution to master your music is to employ an ‘A-list’ mastering engineer with a proven track record for amazing results. In fact, I recommend that you do hire the finest mastering engineer money can buy at least once. This will enable you to hear how good your work can sound. You will learn a lot and it will be money well spent. But herein lies problem number one - professional mastering is very expensive, with an average three-minute song costing hundreds for a world-class engineer. And be aware not even professional mastering will save a bad mix. Even if you can afford professional mastering you may be throwing good money away if you have not thought how your mix translates to mastering. A respectable mastering engineer will tell you your mix is not master ready and inform you that to realise a great master you need a great mix, but some less scrupulous engineers may not.

The second problem I see is the quality of information. Online sources post extremely fragmented snippets of knowledge designed to keep the viewer coming back for more, whilst most self-help books focus on a professional mastering engineer scenario where it is assumed the reader is looking for a full-time career. As an independent artist you are left wondering how glass masters, high end mastering processors, DAWs costing thousands, PQ encoding, coloured book standards and error checking relate to your latest Spotify release. The answer is that they do not. This book only details the information you need.

I address audio mastering as a standalone process. This is something I have always done personally. Although it is possible to mix and master within one seamless workflow, I believe keeping the mix and master separate gives improved results; a chance to change your mindset and listen with a fresh perspective.

Finally, you may have heard that mastering is all in the ear of the mastering engineer. It is true that at a professional level all a mastering engineer does is listen to two-track finished mixes and masters all day every day; their ears have become accustomed to aligning music to the benchmark sound in their minds eye. But all they are doing is placing the song before them into a pre-formed framework. I will show you how you too can learn to place your mix in the context of other great songs, learn how to listen critically and to react to what you are hearing. I will give you a framework that will free your mind to listen, not worry about complex terminology and processes you will never use. Go with the flow, do your best at all times and be confident that this book will bring out the best masters in you.

2

MASTERING ROOM

Arguably the most important link in the mastering chain is the room that you work in. Professional mastering rooms have been meticulously set up to give the best possible listening experience; an environment that is optimised to ensure masters sound exactly as intended. When a mastered song is played outside of the mastering room it is referred to as ‘translating’ or ‘travelling’. Translation can be good or bad. If a master is said to ‘travel well’ this means the translation from mastering room to earbuds, car audio, home stereos and even dance venues has been successful, i.e. the copies of the master sound similar to the original when played. I say similar because there will always be a compromise. A mastered song cannot play perfectly everywhere. Think of the differences between a mobile phone that utilises a minute speaker incapable of reproducing low frequencies and then, at the other end of the scale, the full frequency PA systems in nightclubs that rumble super deep waves through your body. Acceptance is the way forward. Accept that your master will sound different on each new replay system. The target is to find the middle ground so that translation is fine on the majority of devices.

How you hear your music within your room will determine how well your master travels. Think of the studio as a mirror image of the audio master. If you hear an excessively loud bass response in your room, then the natural response is to lower the bass frequencies. This means they are now quieter than they should be and will result in a bass light master. This mirroring effect applies to all frequency ranges. All frequencies that sound louder in your room will be quieter in your master and all quiet frequencies will sound louder elsewhere; and the more extreme the deviation, the worse the translation. This also applies to reverberation. If, for example, your room is unnaturally dead, then you will find yourself adding too much artificial reverberation. When your track is then played back in a livelier environment it will be awash with all the sparkling reflections that were suffocated by your dampened room. This prompts the question ‘how much reverberation is the right amount for a mastering room’? Well, there is an industry standard for measuring reverberation and it is called an RT60 calculation. RT60 represents the time it takes reverberation to fall by -60 decibels in any given space. We will look at how to measure RT60 later on in this chapter.

Try not to be daunted by the word ’mastering room’. It describes any room that you will master in. This could be a project studio control room in a rented facility or a laptop on a coffee table in your lounge. It is easy to imagine that you must have a super fabulous space or your masters will sound awful; this is not the case. There are many successful mastering engineers all around the world making do with less than perfect rooms. They adapt to their environment and know exactly how their wonderful masters translate to the waiting world. Any four walls can serve as a mastering room with the right approach. The purpose of this book is to optimise your mastering space, whatever that might be, whilst developing your knowledge and ears so the quality of your final mastered product improves to a commercially accepted standard. You may surprise yourself with how much progression you can make in a short space of time.

Let us look at your mastering space and what improvements you can make. In this chapter we will walk through the process of optimisation from the initial choice of a room and the placement of monitors, through to adding acoustic treatment and controlling natural reverberation. Even if you already have an existing room that you are using please stay with me on this chapter to double-check that your room is as good as it can be.

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CHOOSING A MASTERING ROOM

The very first step in setting up a room is to assemble a selection of ‘monitor reference’ songs that you know well and have listened to in multiple environments over many years. At this point, 8-10 tunes are sufficient. Preferably chose commercial releases, but maybe add a song of your own and include one mono vocal recording. Place these songs on a flash drive in .wav file format at the highest possible bit depth, or burn them to a CD. You will listen to these tracks to evaluate the steps you take in this chapter, so go ahead and find your favourite ‘monitor reference’ songs.

Then, proceed with the assessment of an empty room, assuming you are starting from scratch. Preparation and thought at this stage will save wasted time later. It is tempting to dive straight in and start mastering in a room straight away, but a wrong decision taken in haste will mean starting all over again later. If you only have one possible choice, that is fine, any room can be improved and made suitable for our purposes. Remember, no room will ever reach perfection. Even the finest mastering rooms in the most lavish commercial recording studios have imperfections. Acceptance of imperfections whilst making our room as good as it can possibly be is the goal.

Assuming you have multiple rooms available, which one is best? To help you decide, here are a few universally accepted guidelines:

Where possible, avoid square rooms or room measurements that are simple divisions of each other. For example, a room that is 20 feet long, 10 feet wide and 10 feet high would not be a good choice. The simple 1:1 and 2:1 ratios mean an exaggerated unbalanced response across the full spectrum as sound waves longer than the room itself (a 50Hz wave is 22 feet long!) double up or cancel out in a noticeable way. A room 16 feet long, 12 and a half feet wide and 10 feet high would result in a fine room as recommended by successful mix engineer Bruce Swedien

Generally the more head height the better as long as the height is not a simple ratio of the length and width

Symmetry is good. If the left and right hand side of your room are a perfect symmetrical mirror image of each other then this will help provide a balanced stereo image. However, symmetry with unparallel dimensions is even better as not only will there be a solid stereo image but also the sound waves will be spread in a more complex way and lessen the build up of problematic frequencies

One simple trick to evaluate a room is to clap or make a sharp percussive noise and listen to the response. Listen to how long the reverberant tail is; is the room dead or live sounding? Listen to the frequency response, especially a ringing sound as high frequencies bounce about. It will probably be the case that all rooms will be ringing and too reverberant without treatment, but who knows, you might get lucky. The room with the least problems to start with would be the logical choice.

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PLACEMENT OF MONITORS

Having chosen your best empty room, it is now time to grab those beating beasts of sound – the monitors. The first task is to find their optimal placement. Where does the manufacturer recommend you place them? Some monitors are designed to be placed up against a wall whilst some should be situated in an open space, so check the manual to be sure. Many studio monitors will be able to adapt to ‘wrong’ positioning by way of built-in switches that change their response. If your monitors have any such switches set them to their default state and go with the manual’s recommendations where your room allows. Your goal is to find the best place for them within your room. This will be unique to every room. Your evaluation will involve placing the monitors and listening to your reference songs to assess the results; then moving the monitors and listening again. A change in monitor position might be to another area in the room or a matter of millimetres in situ, but each time you move your monitors ask yourself ‘has the change improved what I am hearing or made it worse’? If your alterations have made matters worse, try other positions until you cannot improve on your best placement.

Even if you think you know where your monitors will sound best it is essential to try other areas to experiment and explore. You will soon realise that small movements can bring about large changes. Expect your room to do unpredictable things at this stage. It is difficult to predict how a room will react, but be confident that with each stage of this and every subsequent process you will be taking an empty room and improving it toward a stable working mastering room. Where to begin? Again, sticking to universally accepted convention, the first position to try is with the monitors centred along the narrowest wall, at a distance from the wall as specified by the monitor’s manual, and with the monitors and your ears forming an equilateral triangle. The tweeters should be firing an imaginary laser beam to your ears at a height that is level with your ears. Your seating position should be at 38% of the length of your room. This is a theoretical best seating position and should be viewed only as a start point. The size of your imaginary equilateral triangle will increase or decrease as you move your monitors closer or further apart. To find your 38% measure the length of your room and mark where 38% falls. As an example, if your room is 3 metres long you would divide 3 by 100 to get to 1%, then times the result by 38 to get to 38%; the result in this example is 1.14 metres (3/100=0.03 then 0.03x38=1.14).

It is worth repeating that this should be viewed as a starting point. It maybe the best placement for your monitors, it may not be. The only way to know it to compare this position with other locations around your room so be sure to experiment. The better your monitors sound at this early stage, the less work will be needed later. Start playing your reference songs and listen very carefully to the stereo imagery and the overall balance of frequencies. Look for the sound to become more open and detailed in the mid and high range and more solid and focused in the lows? Can you close your eyes and visualise all the different instruments in their own spaces? Are all the frequencies represented equally? Start to listen really critically in a way that the monitors are telling you the answers and not the other way around; sit back, relax, clear your mind and let the sound waves hit your ears. Go with your gut instincts and your first reactions. The goal here is to find the best balance possible at this early stage simply from monitor placement, not a finished room. We will look at sound treatment shortly, but do not cut corners here and settle for the wrong positioning. If the current position of your monitors is lacking in bass response, try every possible position to rectify that loss. How will you know what your best position is unless you exhaust all possible scenarios? It may well be that your first position was the best, but without trying all options how will you know? Until you are sure you know the answer doubt will remain in your mind and months down the line that nagging voice in your head will say “maybe the monitors are in the wrong place”! Now is the time to experiment and clear any doubt. There should be a smooth transition from the bass through the mids up to the highs where all areas are equally represented? Try to reach as close as you can.

Once you have settled on the position that gives you the most balanced sound with an open stereo field, it is time to check and optimise the left, right and centre image. To check the lateral response, listen to the mono vocal recording. Is the voice perceived to be central within the monitors as if the narrator was standing in front of you? It may sound a little to the left, for example. If so this means that the sound from the left monitor is reaching your ears a fraction earlier than the right monitor. A slight adjustment is needed to bring the left and right monitors in line. In this example you would move the left speaker back a fraction to delay the signal slightly (or move the right speaker forward). Small corrections are all that is needed. You can always undo any moves that prove detrimental. At all times think “is it better or worse?” You will know when the recorded voice is central. Human brains can locate sounds, especially in the mid and high frequencies, with great precision. We do it continuously every day without thinking. So clear your mind, close your eyes if that helps and listen critically, if the source feels off centre then it is, adjust the monitors until the feeling is right. Your brain will only place the central phantom image centrally if it receives sound information from both monitors at the exact same time.

To aid the task of monitor placement there are various commercial test discs available that provide test tones and guides. Most test discs can easily be found via an internet search. There are also a couple of more obscure tests which I have found useful. The first is the ‘Sound on Sound: Bass Staircase Test’ which is an mp3 file downloadable from

https://www.soundonsound.com/techniques/room-improvement-audio-file.

The second is a series of LEDR tests downloadable from audiocheck.net. The Bass Staircase Test is particularly useful for testing the bass response of a room. It comprises of a series of bass tones that increase by one semi-tone at a time. Crucially all the tones are of equal loudness. When playing the mp3 file you will see that the DAW’s master meter stays fixed at a constant level for each step. This confirms that all tones play at the same level, but listening via your monitors you will hear differences as you move up the bass staircase, the bigger the fluctuations in level, the worse the problem. Use this test to find the best bass response placement where the differences in volume are the least. Regarding the audiocheck.net tests; LEDR test stands for ‘Listening Environment Diagnostic Recording’ test. This is a way to subjectively evaluate the accuracy of stereo image reproduction. It specifically focuses on the stereo imagery. To find out more visit audiocheck.net, look out the information and download the test files to have some fun. All that matters is that you do the best you can within a reasonable time. Do not sell yourself short but also try not to stress; this should be an enjoyable experience. Stay positive, believe in the process and be assured that everything will come together by the end of this book.

Once you have settled on the best position for your monitors it is time to add some acoustic treatment to further improve your mastering room.

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ACOUSTIC TREATMENT

What is acoustic treatment and why should you care? Acoustic treatment is the process of adding sound absorption and reflection panels to a room in order to balance the response of the room. As we have seen so far in this chapter, careful planning and the considered choice of room and monitor location can go a long way to smoothing out problems. But for a room to be used as an audio mastering suite, some form of acoustic treatment will always be a good option. Adding acoustic treatment will give clarity to sound, making it crisper, less muddy and more reliable as errant sound waves are tamed. It is important to note that you can do significant damage by overtreating a room. The best way to approach acoustic treatment is by adding a little, evaluating the changes and then adding more if needed. This way you will add exactly the right amount without going too far into ‘dead sound land’ where sound is lifeless.

There are a variety of ways to attack acoustic treatment. These range from using your own ears and making your own traps and panels to employing a professional consultant. Obviously the easiest way to fine tune a mastering environment is to employ a professional designer to measure and treat the room. However, you may not have the funds required or perhaps you would prefer to tackle the job yourself for the fun and learning involved. Assuming you are not using a professional studio designer, what is involved? Firstly, there has to be a method of measuring the response of your room as it is now and as it will be after the changes. The method of measurement may be as simple as your ears. Many studio owners prefer to trust their ears and shy away from measurements, and you may be no different. I would advise writing down your thoughts at each stage. Assuming you wish to measure the frequency response and reverberant qualities in a more detailed way, you will need a piece of software that is capable of generating test pulses and at the same time recording the results of the pulses within your room. Luckily there are some great applications that do just that. Do your research and choose your room acoustics software tool. There are two main variations of software to be aware of. The first will simply measure the frequency response within a room; the second will measure and then add a corrective layer of equalisation to your listening chain to flatten the response. Which one you choose will come down to personal preference. However, to measure your room you will need to be 100% confident in your ability to use the tools and interpret the results, so do your research well and read any manual thoroughly. Wrong measurements are no better than no measurements at all and could potentially make your room sound worse.

Once you have your software loaded it is as simple as measuring the room’s response with a microphone and highlighting the problem frequencies. You then can choose to use corrective monitor equalisation and/or add acoustic treatment to compensate, and then re-measure to check the correction has been successful.

There are various opinions regarding monitor equalisation. Some purists demand that rooms be adjusted only by acoustic treatment. It must be said, however, that these purists usually work on specially-designed and purpose-built rooms. But recently more and more engineers and producers are embracing monitor equalisation and/or corrective software solutions such as Sonarworks ‘SoundID Reference’. A rule of thumb, if equalising manually, is to tune mostly be cutting rather than boosting in the third-octave bands and not to cut more than 6db or boost more than 3db in any band.

When treating, addressing certain areas will reap greater benefits than others. For example, areas of particular concern are the early reflections around the listening sweet spot and the corners of a room. For the early reflections, ask a friend to hold a mirror on the side walls between you and your monitors and where you see the monitors in the mirrors is where the early reflection trapping is needed. Do not forget to address early reflections from the ceiling and floors too, if possible. The corners of the room are notoriously problematic as all frequencies build up here, so look to add trapping. Each time you add acoustic treatment measure to see the results. The goal is to get as close to a flat linear response as possible throughout the frequency spectrum. There are many companies that specialise in manufacturing acoustic traps and dispersion panels if you prefer to buy them ready made, but a little DIY can go a long way and save money. The subject of acoustic treatment could fill a book on its own, so I will leave you to do as much or as little research as you wish. As a minimum I would recommend buying a piece of ‘room measurement’ software and making or buying a few traps and dispersion panels within your budget and time constraints. You will be surprised how quick and easy this can be to install and how good the results can be.

Having said that, remember no room will ever be spot on perfect. Yes, the desired result is to achieve a flat frequency response across all frequency ranges with a controlled RT60 reverberation response. And it is always good to have a target to aim for, but in the real world this is rarely, if ever, achieved. There will always be peaks and troughs in the frequency range and as I will explain in a moment, RT60 is very difficult to measure. It will always be a case of making the room sound the best you can and then adapting your mastering decisions to reflect its inadequacies. This is a reality that everyone goes through. Over time you will learn the sound of your room and be able to trust your room totally. You will adapt, but perfection unfortunately is an unreachable, ever moving goalpost. Console yourself with the fact that many great records have been made in rooms that are less than perfect!

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REVERBERATION CONTROL

If you want to get really scientific, there is the possibility of measuring the reverberant quality of your room to assess whether too much or too little reverberation exists. The industry standard for measuring reverberation within a room is called an RT60 reverb calculation. RT60 is the time it takes reverberant energy to fall by -60 decibels. Whereas acoustic treatment measures and balances the level of frequencies in relation to each other within your room, RT60 measures the time it takes these frequencies to decay. Too much reverberation will colour the sound you hear from the speakers and blur your ability to make critical judgements, whilst too little reverberation will result in an exaggerated ‘liveliness’ to your master when played out as you add too much artificial reverberation to compensate.

As we now learn how to analyse and measure reverberation, it will help to break the life span of sound waves down into three phases. They are as follows:

Direct Sound - The sound waves that reach the ear first. As is implied by the name tag, direct sound takes the quickest, direct path to your ears which is why it is first to arrive

Early Reflections – The sound waves that reach the ear after the direct sound. These are the sound waves that bounce off nearby surfaces. Early reflections are generally thought of as any reflection that arrives up to 30 milliseconds after the direct sound

Reverberation – Sound waves that arrive at the ear after the direct sound and early reflections

These three phases describe the life cycle of a sound wave as it bounces around a room. Sound energy in the form of waves explodes from the monitors in many directions; we first hear the direct sound, then the early reflections which we perceive as part of the same direct source, and then the remaining build up of reverberant clutter as noise bounces off all surfaces until all the power is lost and the sound fizzles out to silence; and it is this transformation of sound to silence over time that the RT60 calculation measures.

To take an RT60 measurement you will need to buy or borrow a Sound Pressure Level Meter (abbreviated to SPL) that is purpose built to take reverberation measurements and follow the instructions the manufacturer provides. These specialised meters are expensive to buy and I do not recommend their purchase to merely set up your room. However, you may know a friend who owns one or you may have a continual need for one outside of this task. The process, should you procure the tool, is to fire a very short loud burst of sound energy into the room. The complete sound and decay is recorded with the RT60 being the time the room sound takes to reduce by -60db after the initial pulse. A full in-depth analysis of RT60 measurements are beyond the scope of this book, but I encourage you to do some research and take multiple RT60 measurements if you can as any acoustic treatment you add will naturally affect the RT60 reverberation decay time. Acoustic treatment and reverberation decay are linked; it is a balancing act to stabilise levels of frequencies across the frequency spectrum and control how quickly they disappear over time.

If you want to make an unscientific approximation of RT60 it is possible to create your own experiment by recording an exploding balloon or a loud hand clap into your DAW and record the time it takes for the room sound to reduce by -60db. The problem comes when tracking the reduction as the sound will need to be loud and the resolution of the DAW deep. To overcome this there are workarounds named RT15 and RT30. For an RT15 calculation track the time the room sound takes to reduce by -15db and multiply it by four; for an RT30 calculation track the time the room sound takes to reduce by -30db and multiply by two.

For an even rougher approximation of reverberation levels in your room simply clap or make a short loud vocal explosive sound and listen to the results. The first thing I see producers do when entering a new space is to clap their hands and listen. You will be amazed how your intuition will tell you whether a space is too reverberant or too dead sounding; your mind carries a data bank of many thousands of spaces and is constantly analysing the reverberant space around you every moment of every day, even though you are rarely aware of this happening. Clap or shout away and trust your judgement. If it sounds too live then it is. If it sounds too dead then it is.

Fantastic, you have now chosen your room, placed your monitors in their optimal position for clarity definition and focus, added acoustic treatment to the space you will master in whilst giving consideration to its reverberant qualities – a huge step in the right direction in only the first chapter! Now let us look at the monitor levels within this finely tuned mastering room.

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MONITORING AND LOUDNESS

The level at which music is played is vitally important and a crucial part of our mastering system, so much so that it is worthy of a chapter in its own right. While mixing engineers stress the need to listen at different levels, every piece of music in my studio plays back at the same average level when I am mastering. I use the integrated loudness of a track as a guide in the same way streaming websites do. I strongly recommend you do the same. The control of loudness is one of the most important aspects of modern production. Music should always play at a consistent and comfortable level for any listener through a given medium; that was the premise which initially gave birth to the role of a mastering engineer. But more than ensuring a pleasurable listening experience, playing music at the correct consistent level gives the best perspective from which to make critical mastering decisions.

This chapter will explain how to listen at the optimum level every time you press play. By the end of this chapter you will literally hear music in a better light. Once you apply this system of listening you will never go back to random working practices again. It is time to take control of wild replay levels. Recorded music is a precious joy that should be cherished and played at optimum quality wherever possible.

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PERCEPTION OF SOUND

Have you ever wondered how the level at which you listen affects your perception of sound? It is a scientific fact that humans hear music differently depending on the replay level. Let us explore why this is and how different monitor levels will affect your mastering decisions. Your hearing system has two important characteristics to be aware of:

Ears are more sensitive to certain frequencies than others

The degree of sensitivity changes depending on how loud the sound source is

We do not hear all frequencies at an equal loudness and, more confusingly, the relative levels of frequencies we hear differ depending on how loud music is played. For example, at low listening levels, bass and treble frequencies require more actual sound level than mid frequencies in order to be perceived as being at the same level. This gives the illusion of mid frequencies being accentuated in relation to the bass and treble frequencies. However, raise the volume to a very loud level and the bass and treble frequencies will now seem more prominent than the mids even though the tonal balance remains unchanged. This has obvious consequences for mastering, for example, master at a low level and your bass and treble decisions will be too hot for the listener when they crank up their amp to the max.

Audio engineers have been studying this phenomenon for years. The results of how humans perceive frequency response as loudness changes were first documented by Harvey Fletcher and Wilden A. Munson. They published a paper in 1933 entitled ‘Loudness, its definition, measurement and calculation’. The graphs that resulted from Fletcher and Munson’s experiments are now known as ‘equal-loudness contours’. Consumer stereo equipment in the 1970s often had a loudness switch that implemented these contours for low-level listening. Although Fletcher and Munson’s curves have been superseded, they are still the most widely known equal-loudness contours and clearly illustrate the problems we face.

EQUAL-LOUDNESS CONTOURS

‘Equal-loudness contours’, of which the Fletcher-Munson curves are one example, show how perception of frequency levels change with loudness. It is worth repeating this as it is really important to understand. The actual level relationship between frequency bands do not change when we turn our monitors up or down, the tonal balance remains the same, it is only our perception of them that changes.

If you study the Fletcher-Munson curves diagram you will see the sound pressure level (dB SPL) vertically on the left and then running from left to right horizontally are the frequencies of human hearing. The crucial point to note is that the blue curves represent the adjustment in levels needed for the listener to perceive a constant loudness across the whole frequency range. The Fletcher Munson experiment involved many listeners who were presented with test signals containing the full range of frequencies. All frequencies started at an equal level. But as we have learned above, even though the levels were technically the same, due to our hearing characteristics, each listener would have perceived the frequencies as being unequal. The test investigator then boosted or cut any frequency requested by the listener until all frequencies were perceived to be equal. What we see in the graph is the average amount of boost or cut that was needed to satisfy the listeners that frequencies were indeed equal. The graph clearly shows that the amount of gain compensation needed to flatten the frequency range is greatest at lower listening levels. This backs up our earlier example of bass and treble frequencies being perceived quieter than mids at low level listening. Listening levels not only affect frequencies but other aspects too, for example, the amount of audible reverberation and the impact of compression. Turning the monitor gain down really low will allow transients to be heard more clearly, accentuate how compression shapes initial bursts of sound and nullify decaying tails and ambience.

In light of all the above, and the realisation that however you present a master it will never sound the same to others as it does to you, you may be thinking “what is the point of trying too hard?” My answer would be to always do your best no matter what. Control what you can control and let go of the rest. Soak up knowledge so you will be better informed to make the right decisions at the right time. The more knowledge you store and the more practice you get, the greater your improvements will be. Information works its way into your subconscious through repetition and influences your moves without you being aware. You will be more competent simply by gaining knowledge. For me, the realisation that I could not control every aspect of my master was liberating. I felt a relief that it was okay simply to do my best and let go, knowing that my finished master would vary depending on how and where it is played. The takeaway here is that it is normal for music to sound bass and treble light at lower listening levels and it is normal for music to sound different depending on the environment and the replay system – there is nothing you can do to change this. Mastering is all about balance. Aim for the middle ground and take decisions that will show your song in the best possible light across multiple replay systems; that is all we can do as mastering engineers.

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SOUND PRESSURE LEVEL AND LOUDNESS

As we move on to set the optimum monitor level and control the integrated loudness of each song, it will be beneficial to define the following: how sound level is measured; how loudness is measured and perceived; and how we accommodate both within a monitoring and metering system. Luckily for us a mastering engineer called Bob Katz has already done all the hard work and developed such a system. It is called the ‘K-System’ and addresses both monitor gain and metering protocol. We will set up our K-System at the end of this chapter, but for now let us define the two key terms of ‘sound pressure level’ and ‘loudness’.

The level of sound itself is straightforward enough. Its principles are firmly seated in mathematics and measured in decibels. Decibels are often written as ‘db’ and are named after its inventor Alexander Bell who also invented the telephone. Bell’s sound measurement system is clearly represented by a scale that expresses a change in value, for example, +1db or -1db signifies a shift of one unit whereas +6db equals a shift of six decibel units in an upward direction. The decibel scale is logarithmic rather than linear; an increase of 3db doubles the sound. Sound level can be accurately measured using an SPL (Sound Pressure Level) Meter. All straight forward enough, but when we talk of loudness it is more confusing. We could measure a sound source that is say 80db and label anything over this threshold as being loud. The trouble is each human being hears sound differently. Each has an individual threshold of loudness tolerance so no single figure would work for all people. Some would comment “wow that is loud” while others would remark “that is not loud at all”. However, there are a few sounds so high in level that no-one would disagree with them being called loud, for example, a jet aeroplane engine. But more often than not the term loudness is used in an ambiguous way that reflects the listener’s perception of what is loud and not a specific figure in decibels. Human hearing and loudness are perceptual. Loudness lies in the ear of the beholder.

Loudness has only one true meaning which is ‘the way we interpret sound pressure that stimulates our auditory senses’, but as the description highlights, loudness is an interpretation unique to each individual. It will be helpful to think of loudness in two separate ways:

The factual true level of sound in decibels recorded by an electronic device

The level of sound perceived by human hearing

The easiest way to show the difference between the two contexts is with the following examples:

“That jet engine is loud. It is measuring 120 decibels on this SPL Meter, wow that is loud!”

This is a factual statement. The sound level is measured at 120 decibels. Everyone would agree that it is indeed loud and the term loud can, therefore, be applied to a very high factually recorded decibel level which is universally agreed as being loud. However, consider the following conversation:

“Turn that music down, it is far too loud”

“No way, it is not loud at all, it is just right”

In this second scenario loudness is expressed as a personal opinion and not a definitive measurement. Neither person is right or wrong. They both have a valid internalised perception of what is loud, and herein lays the problem. There is no universally accepted point at which quiet becomes loud. Perceptual loudness relies on the interpretation of the listener.

HISTORICAL SOUND LEVEL MEASUREMENT SYSTEMS

Over the years there have been many accepted methods of representing ‘perceived’ sound levels against a scale. A simple ‘quiet to loud’ set of written instructions works well for classical music. The varying degrees of loudness are simply noted as instructions for the performer with ‘piano’ being the musical term for quiet and ‘forte’ the term for loud (the full name of the musical instrument we know as a piano is ‘Pianoforte’ meaning quiet to loud). Sound engineering too has many ways of registering the sound levels that pass through electrical equipment. Past technological developments of note include the gorgeous glowing backlit ‘VU (volume unit) Meter’, most prominent on analogue recording consoles. These return a humanistic musically sounding result which takes perception into account. An ‘RMS (root-mean-square) Meter’ is another example of a meter that is in tune with the way our ear perceives sound levels. In fact these meters worked pretty well for many years. It was only with the development of digital production that things went askew. Metering then moved away from perception to precise mathematics. New dazzling standalone digital recorders and computer audio workstations incorporated ‘Peak Meters’ which only measured the loudest peaks. At the inception of digital 16 bit recording this made perfect sense. Peak Meters would notify a user of any digital distortion, a horrible noise that once recorded cannot be undone (digital distortion is not like the pleasing analogue distortion we know and love). But as computers developed and higher bit rate recording became the norm, the problem of digital distortion all but disappeared (unless recording or rendering to 16 bits). However, the metering remains and the problem with Peak Meters is there is no representation of integrated loudness and human perception. Poor metering can be problematic, but the correct metering is a vital tool that should be used copiously and relied on at all times to help sculpt professional masters.

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MOVING FORWARD BY LOOKING BACK

It became clear that something was badly wrong. Commercial music was becoming more and more degraded as mastering engineers competed to make each song louder than the last. The thought process at the time was “louder is best”, and tests have shown that listeners tend to perceive the louder of two tunes as better. The main weapon engineers have to increase perceived loudness is to reduce dynamic range, i.e. reduce the loudest moments so that the whole song can be nudged upwards in level. This process involves the use of compression, clipping and limiting, all of which diminish or distort the loudest peaks. Master after master was released with a crushed and lifeless sound that was hard to listen to by audiophiles and the humble Peak Meter had a lot to do with this degradation. Engineers were used to working audio to a benchmark of 0db on analogue meters. The trouble was 0db analogue and 0dbfs digital are two completely different entities. On analogue VU Meters 0db appears nearer the centre of the meter with +3db of headroom to the right. An engineer could average around the 0db mark with a degree of safety above for peaks to shine through unharmed. If pushed beyond the +3db headroom an incremental degree of analogue distortion would result, but even then this would not be the end of the world and often thought of as desirable depending on genre. 0dbfs digital has no such luxury of headroom or pleasurable distortion (when referring to 16 bit processing or rendering, e.g. a CD). On a digital Peak Meter 0dbfs appears at the top of the scale with nothing above. If an engineer attempted to push audio levels past 0dbfs they would be met with a harsh wall of screeching digital distortion (*clipping). But engineers persisted on working towards the benchmark of 0dbfs. They resorted to ‘brick-wall limiting’ as a way of increasing loudness while preventing digital distortion. And the more they pushed, the more dynamic range they ate up, the more the musical quality and principles of purity, dynamics and liveliness suffered. Something had to be done.

Enter Bob Katz, mastering engineer, in 1999. Bob recognised the need to take loudness out of the equation and began a campaign against the loudness war. He theorised that if there was no benefit to be gained in making music louder the music establishment would return to producing the highest quality audio, not the loudest. He proposed that each piece of music should include a loudness reference number. This would allow replay systems to automatically turn up or down each track according to its rating. Each song would then be played at the same level regardless of loudness and purer unprocessed music would no longer be heard as inferior. The result was the Bob Katz K-System, a method of monitoring and metering that borrowed from the past as well as introducing new principles. Bob recognised the benefits and limitations of both the RMS Meter (perceptual) and the Peak Meter (actual) and designed a brand new ‘K-Meter’ that incorporated both. Furthermore, he introduced the idea of headroom within the digital domain by moving the 0db standard downwards on the meter. This he concluded would overcome the urge to clip and over-compress peaks in order to fit the program material to the medium. Compression, he suggested, could now be used for creative purposes only. But more than just a metering system, Bob also saw the importance of monitoring at a consistent and best level. He proposed 83db SPL as the optimum monitoring level. This benchmark was in line with the film and broadcast industry who had long established a set of standards. Within this industry 83db is regarded as providing the best listening experience across all frequencies and the closest humans will get to a flat frequency response with a satisfying feeling of power of bass energy. As mentioned, Bob recognised the need to combine both monitoring and metering together in one system. To do this he presented a way of calibrating studio monitors to K-Meters, a process that we will walk through shortly.

To a large degree Bob Katz’s vision came true. K-Meters appeared on many new products and the principles he instilled were widely accepted as the way forward. Today most music played from streaming Web sites is played back at an automatically adjusted level so that every song streams at roughly the same perceived level. This is achieved by analysing the average loudness of a song and assigning a loudness rating. Music is then turned up or down just as Bob envisaged. The challenge now is to make music that sounds better in other ways, rather than simply being loud. Having said that, the years of exposure to loud music has become ingrained in consumer ears with many people accustomed to the sound of limiting and a narrow dynamic range. Many listeners have grown up with this being the norm, so the battle to change perception still rages on in certain genres. Even so, the K-System forms a crucial part of our mastering system and paved the way for the developments that control loudness today. The integrated loudness of program material is now considered a vital part of audio mastering. The most widely accepted and implemented method of measuring loudness is currently the LUFS. LUFS stands for ‘loudness units relative to full scale’ and calculates the loudness of a complete piece of music taking into account all the highs and lows from start to finish. We call this measurement ‘integrated loudness’. We will look at the specifics of LUFS later when outputting our master, but for the moment our task is to set up a K-System in our mastering room.

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K-SYSTEM

To calibrate your monitoring and metering I recommend you use the K-System devised by Katz. Although the K-System dates back to the turn of the 21st century I still consider it to be the most complete system for metering and monitoring sound level and loudness. The K-System will ensure that you monitor at 83db, the optimum level for hearing as close to a flat response across all frequencies as is possible. There are two parts to the system that work hand in hand - metering and monitoring. Before the K-System was introduced there was no recognised standard for monitoring levels. Mixing and mastering engineers all over the world listened to music at widely varying levels. With the K-System we all have the choice of calibrating to a benchmark knowing that we are all listening at the same level and hearing music in a similar sonic quality as one another.

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K-METERS

There are three versions of a K-Meter (K-20, K-14 and K-12). Each meter takes the digital ceiling level of 0dbfs and moves it downwards by -12db (K-12), -14db (K-14) or -20db (K-20). This change happens only within the meter itself and the monitor level is always adjusted to maintain 83db SPL at the listening position. To further understand this shifting scale, consider the following. As we know a standard DAW Peak Meter displays an upper limit of 0dbfs; the digital ceiling before clipping. On a K-Meter ‘zero’ is moved down the scale to either -12, -14 or -20db depending on the K-Meter of choice. The space above the new zero is renumbered +12, +14 or +20db which represents the restored safe headroom where peaks can exist

without clipping. It is important to note that the zero on a K-Meter is not the same the 0dbfs ceiling which still exists 12, 14 or 20db above the new ‘K zero’. 0dbfs will still have an important part to play in optimising our finished master, but for the moment we can put any thoughts of 0dbfs aside.

K-Meters, therefore, are labelled and referred to by the amount of offset applied to zero. A K-12 meter has a 12db offset, a K-14 meter has a 14db offset and a K-20 meter has a 20db offset. It is usual to be able to select all three K-Meters within one, although some meters might be set to a predefined level, for example a K-14 meter, and cannot be altered. Bob Katz recommends that a K-20 meter be used for mixing natural, acoustic, classical or jazz music and a K-14 meter used for mastering the same. Other styles of music will need a different combination of metering, for example, pop and rock should be mixed at K-14 and mastered accordingly. A true K-Meter should show both peak and RMS values. A final point to note is that a K-Meter should include a red coloured section of 4db above zero, a nod to the red zone in old analogue meters. This ‘hot’ section of the meter is to be used creatively by pushing the musical content into the red zone as and when it suits for an emotional kick. This is a powerful and often overlooked function. If used with care and thought the red section can result in dynamic movement as levels are pushed into the heat when the music swells.

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K-SYSTEM CALIBRATION

It is now time to calibrate your K-System. You will need the following:

Pink noise reference files (available from the Bob Katz website - see below).

An SPL Meter.

K-Meter plugin (or an offset to your master meter – see below).

I recommend you go to the Bob Katz website (www.digido.com) and download his purpose made pink noise calibration files. The full link is www.digido.com/downloads/#general. Here you will find three pink noise audio files, one for the left channel, one for the right channel and one stereo. Use the left and right separate files if your monitoring system has the ability to control each channel independently, if not, the stereo file is the way to go. If you do not own a K-Meter plugin there is a workaround. Most DAWs will allow the user to offset the top figure of zero on the master bus meter. Assuming your DAW does and it displays Peak and RMS results then offsetting 0dbfs by -12db for a K-12 meter, -14db for a K-14 meter and -20db for a K-20 meter is the same. Also, before you buy, check the settings of the meters you own already as they may be configurable as a K-Meter.

WARNING: Please be careful when setting your monitor levels as they are loud enough to potentially cause damage to equipment and ears if used in the wrong way. Be mindful at all times. Always start with the monitor gain low and increase the level cautiously.

Firstly we will calibrate a K-20 meter/monitoring system. Please follow the instructions below whilst paying careful attention to the warning above. It is assumed that you will be using the stereo pink noise file. If you have independent control of left and right monitor channel levels, use the left and right files and follow the instructions twice – once for the left channel and once for the right.

Start with a blank new session and load the Bob Katz stereo pink noise calibration file (-20dbFS RMS pink noise stereo 44.1 WAV file) into a new track/channel on your DAW. Do not alter any levels or pan positions, the files are designed to be played as they load

Press play and confirm the presence of pink noise via your monitors

Insert a K-Meter into your empty master bus and set the meter to K-20. When the pink noise is played the meter should register zero (-20db lower than 0dbfs)

Take your SPL Meter and set the ‘weighting’ to C, the ‘response’ to slow and arm the meter to accept a level of 83db

Place the SPL Meter at the listening point of your monitoring equilateral triangle at the height and position of your ears

Whilst playing the pink noise, adjust the monitor gain (i.e. the amplifier that controls the studio monitors, not the master bus fader) until the SPL Meter registers a level of 83db

Once a level of 83db is recorded on the SPL Meter, make a note on the monitor gain dial and label it ‘K-20’

Your studio monitors are now calibrated to play K-20 audio at a level of 83db. Now let us calibrate a K-14 system.

Switch your K-Meter to K-14

Increase the level of the pink noise wav files by +6db using the clip gain tool

Reduce the monitor gain (i.e. the amplifier that controls the studio monitors) by -6db

Press play and the pink noise passing through the K-Meter should again register zero (-14db below 0dbfs)

Check your SPL Meter to confirm a level of 83db has been retained

Make a note on the monitor gain dial and label the position ‘K-14’

Your studio monitors are now calibrated to play K-14 audio at a level of 83db. All that remains is to calibrate a K-12 system.

Switch your K-Meter to K-12

Increase the level of the pink noise wav files by +2db using the clip gain tool

Reduce the monitor gain by -2db

Press play and the pink noise passing through the K-Meter should once again register zero (-12db below 0dbfs)

Check your SPL Meter to confirm a level of 83db

Make a note on the monitor gain dial and label the position ‘K-12’

Warning: Do not play any actual music until you have completed the ‘Measuring Integrated Loudness’ section below. Commercially mastered tracks could be up to 14db louder (or even more) than our K-20 default monitoring level is calibrated to receive. This means that if your monitor gain is set to K-20, a commercial master may play at 97 decibels! All music must be measured and adjusted before playing.

You should now have three different positions marked on your monitor gain control K-20, K-14 and K-12. You may be wondering what all the fuss is about. You moved the pink noise wav file up in level and the monitor gain down to compensate – big deal, it all sounded the same. That is exactly the point. When mastering, every piece of music should be played at the same level. That was the goal we set ourselves. By measuring and compensating for integrated loudness on every music file, we can control the level of program material in the same manner as we have done with the pink noise, i.e. measure loudness and adjust the clip gain to match a K-System number – measure, adjust, play; measure, adjust, play. One final point; depending on your room size and style of monitoring (e.g. close nearfield monitors), 83db can sound louder than intended. You may feel the need to reduce your monitor gain by a few decibels, but stick with the 83db benchmark for the time being; reserve judgement until after the next chapter when we spend time listening to commercial reference tracks adjusted to a K-System scenario.

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MEASURING INTEGRATED LOUDNESS

The integrated loudness of all music must now be measured and adjusted to fit the K-System before playing. This may seem a chore at first, but I assure you that once you are used to the quick and easy method of ‘measure, adjust and play’ (which literally takes seconds) and you hear the benefits of level matched listening, you will never go back.

As we learned above, the integrated loudness of music is the average loudness of a complete work from start to finish. It takes into account psychoacoustic properties and other technical matters. To measure this you will need an offline Loudness Meter. Measurements can be taken offline or online. Online means in real-time, so to analyse a three minute song would take three minutes which is not ideal for our purposes. Offline processing on the other hand will analyse a music file and return a result in seconds. Personally I use the ‘TT Dynamic Range Meter’ by The Pleasurize Music Foundation. This meter has been around for many years and is no longer available, but you will find others online and it is not important which you decide to use; the important point is to only use one meter for this task. We want every song to play in a uniform way and using different meters may cause inconsistencies. The reading needed is the RMS or LUFS figure ignoring the peak and dynamic range figures that will become significant later on. Both RMS and LUFS are pretty much the same and close enough for this exercise, so either is fine, do not think too deeply about how the figures are generated or how one meter compares to another, choose one measurement scale and stick with it. Let us look at how we level all music files.

I suggest you use the K-14 System as the default setting for your studio unless you only work with classical, acoustic, jazz or similar natural sounding genres; then a K-20 meter would suit. I use a K-14 benchmark in my studio and this is how I perform the task every day. In this example the goal is to play back three commercially mastered songs at the same integrated loudness level. The process is as follows with diagrams on the following pages:

Set your monitor amp to the previously calibrated level for a K-14 meter

Open your DAW and create a new blank session

Import three commercially mastered songs from your monitor reference set to one track/channel

Leave all files as they import and do not change any levels at this stage

Export the first song to your desktop and call it ‘temp’

Process the ‘temp’ file with the Offline Loudness Meter to determine the current RMS/LUFS rating

Back within your DAW adjust the clip gain of the song in question so that its new RMS/LUFS value equals -14db. For example, if the current RMS/LUFS rating is -11db then you need to decrease the clip gain by -3db (-11 + -3 = -14); if the current rating is -15db then you need to increase the clip gain by +1db (-15 + 1 = -14). If your meter returns fractions of decibels (e.g. 14.9db) then you can adjust more precisely

Now export the gain adjusted song to your desktop for a second time overwriting the existing ‘temp’ file

Finally, re-run the offline Loudness Meter process to confirm your calculations and adjustments have been successful and repeat the above steps for the next two songs

Always write a temporary file to the desktop and delete it after any adjustment: this way you will not accumulate hundreds of unnecessary wav files on your hard drive. Once you get used to the technique you will be surprised how quick it is. In Reaper, my DAW of choice, the task takes thirty five seconds from start to finish.

Now press play and sit back in amazement as all three songs play at exactly the same perceived loudness as they would on a compilation album or the radio. Congratulations, you have now overcome one of the biggest problems in audio production – the influence of loudness and sound levels on perception. It will soon become second nature to adjust every piece of music that flows through your monitors. From now on, whenever you are mastering, any target level and dynamic range figure can be calculated against K-14. We will learn how as we progress.

If you are unable to find an offline Loudness Meter, a close approximation can be achieved by playing the loudest section of each song through a K-Meter, LUFS Meter, or a VU Meter with a -14db offset and adjusting the clip gain until the music plays at -14db.

You now have a mastering space you can be confident in. You have chosen your best room and placed your monitors in their ideal location. You have stabilised the frequency response and stereo imaging with acoustic treatment. You have calibrated your monitor gain and metering to K-System standards. With all these key elements in place you can now begin to analyse audio and listen critically whist training your greatest tools – the ears. Let’s go listen!

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4

CRITICAL LISTENING

In this chapter we will expand on the principles previously introduced and show how level-matched listening within context is crucial. We will establish a set of guidelines to aid critical listening and begin ear training exercises. In subsequent chapters we will explore algorithmic masters and how to beat them, look at the tools of mastering, set up a mastering template and work through a step-by-step method of use; output a master and learn additional tricks useful in mastering an album, but this chapter is all about listening and forming the mental images that will help you to work your magic.

All professional mastering engineers are experts in listening and have a picture in their mind as they start each project. Try and do the same to make each project’s roadmap as clear as it can be. The phrase “putting right all that is wrong in a sympathetic way” sums up the task of preparing a master for export. The hardest aspect of mastering is to know what is wrong and how to put it right. How can you make corrections if you do not know what is wrong? A process of thinking and analysis is called for, one that will highlight all issues. The following will help guide you in organising your thoughts and building a framework for critical listening.

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TOP 50

Your first task is to augment your existing ten ‘monitor reference tunes’ that you collated in ‘Chapter 2: Mastering Room’. Find another forty songs that you know and love, and import them into your DAW. This is so important - a key element and cornerstone to our mastering system; an aid that will be used on every mastering session you complete to give a constructive viewpoint that places your master in context. Please do not skip this stage; you should end up with a total of 50 songs. If you do not have time to complete the task, set aside an evening or weekend to be alone with your favourite tunes. Look back at the songs that first inspired you as well as your latest muses and most importantly, enjoy the process; relax and make it fun. Grab a stack of CDs, vinyl, cassette tapes or trawl through digital files and find your favourite songs to use as an ongoing reference. You may find it easier to list your favourite artists and then pick one or two tracks by each. It is important to choose the songs you know best, the vivid songs that have been the roadmap to your life. Try to mix up genres, tempos and eras if you can and be sure to include masters you admire for their production qualities along with a selection of recent releases. What made your pulse race and your imagination spin as you decided to take your first magical steps into music production? I would not recommend that you reference aging cassette tapes or vinyl for modern sonic qualities, but there could be songs in that format which you admire for the arrangement, performance and mix attributes. Indeed the sound of cassette tape or vinyl maybe your thing – that is great. There is no right or wrong in music, if recreating the sound of cassette tape with your masters is your goal then of course you must reference cassette tape masters. The best and most useful Top 50 songs will span different genres and different eras. Some people feel you should only references the latest highest quality masters – I disagree. I think that this approach is unbalanced and not representative of reality. It can lead to disappointment and discouragement if you are not able to instantly match masters of the highest level. Instead, by choosing a wide variety of material, it will become apparent that audio masters vary a great deal in sound properties and quality. You will see the bigger picture. Masters come in all shapes and sizes, all you have to do is produce one that fits within the boundaries of historical quality control and be confident you are sitting in good company. Through repeated success your confidence will grow and grow.

Once you have selected your Top 50 songs it is time to organise them. If your music is already on your hard drive as digital files, then create a new folder at a convenient location, name it Top 50 and copy all the files across so that they are grouped together in one place. If your songs are in vinyl, cassette tape or CD format you will need to *rip/record the songs to a digital format so that they can be used within your computer’s DAW. Either way, the goal is to have a folder called Top 50 containing fifty professionally mastered songs.

*Please respect each recording artist’s copyright and intellectual property. Only rip/copy songs that you have purchased and have a legal right to make a copy for personal listening.

Assuming you have your Top 50 folder full of inspirational masters, it is time to create a level matched listening session. Each track will need to be adjusted so that it plays with an equal loudness. I will assume you are using a K-14 monitoring/metering system as detailed in the previous chapter. The process for creating a K-14, Top 50 session, is as follows:

Open your DAW and create a new blank session. Use the ‘save as’ function to save the session within your Top 50 folder naming it Top 50. It is also advisable to create a shortcut to the session and place it on your toolbar – this is one session you will be opening again and again

Locate and import all fifty masters into one single DAW track. They should all be lined up one after another horizontally

*Export the first song to your desktop, naming it ‘temp’

Run the ‘Offline Loudness Meter’ to establish the current integrated loudness of the song

Use the integrated loudness figure to calculate the adjustment required to equal -14db

Back in your DAW, adjust the clip gain of the song so that the new integrated loudness equals -14db

Export the track once more overwriting the existing ‘temp’ file and confirm the integrated loudness is now -14db by running the ‘Offline Loudness Meter’

If correct then move on to the next song. If incorrect, redo until correct

Repeat instructions three to eight for the remaining songs (backup the session when finished)

* Points three to eight are covered in depth in the previous chapter ‘Monitoring and Loudness’.

Excellent, you now have fifty reference songs all level matched to a consistent integrated loudness within a K-14 calibrated system. Notice how wildly all the masters vary in level and waveform shape. Press play and enjoy level matched listening. Submerge yourself in all the wonderful sounds, rhythms, melodies and sonic soundscapes that evoke magical moments and memories. Listen, listen and listen some more. Sit back in your sweet spot and let the waves wash over you. I cannot overemphasise the importance of referencing commercial masters. If possible, listen to a couple of your Top 50 songs before any mastering session. This will ingrain the successful properties of a great master deep in your subconscious. Repetition is the key. The more you listen the clearer your picture will become until you are able to see the end product in your mastering mind’s eye upon first audition of a mix, exactly as a professional does.

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CRITICAL LISTENING

It may sound obvious to state that listening is one of the key skills in audio mastering. But it is easy to forget that the first act in music production should always be to listen, especially when a busy mind is full of all possibilities. Rather than first absorbing a mix totally, it can be the case that due to a keenness to begin mastering, problems are invented or unnecessary actions performed through habit. Try to become aware of when you should be listening and when to let go of all analytical thought and feel the music. It is this switching between thought and instinctive action that will eventually happen without your knowledge. If in doubt always listen to evaluate, then envisage, change and re-evaluate.

There is a right and wrong way to listen. If you focus in on one element, that particular aspect will be highlighted artificially in your thoughts and perceived a touch louder than it is. Instead, try to see the whole picture and let the sound waves hit your ears rather than projecting yourself forward into the monitors as if reaching out for answers. When you do feel yourself moving forward, either literally or mentally, straining your ears as if being pulled towards the monitors like magnets, stop – literally press stop. Sit back, take a deep breath, relax and start again, let the waves of sound wash over you once more. There is only one continual wave of frequencies at any one time emanating from your monitors. We talk of separation in audio mixing as though each sound has its own unique set of frequencies – not so. Everything affects everything else. Whether mixing or mastering, the end combination of frequencies equals one wave; one glorious wave full of all frequencies within and beyond our hearing. The laws of psychoacoustics allow us to manipulate sound to fool the brain into thinking there are separate elements, but the truth is there is only one instance of, for example, 250Hz coming out the monitors at any one time. There is only one instance of 12kHz. There is only one instance of any and all frequencies. All instruments present in a recording share the same frequency spectrum, so sit back, relax and let that one beautiful sound wave wash over you and remember changing anything affects everything.

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FREQUENCY SPECTRUM

As you begin training your mastering ears it will be helpful to have a frequency chart detailing all the various bands in hertz, corresponding musical notes, and any words that represent how music is felt. The frequency spectrum is measured in hertz (Hz) and kilohertz (kHz). One thousand hertz is equal to one kilohertz. Not everyone can hear all frequencies, but a range of 20Hz to 20kHz encompasses all possible human listening potential however good the ear. It will be beneficial to learn the frequency spectrum inside out, including how musical notes equate to frequencies. Practise boosting and cutting each range of frequencies and listen carefully to how the changes affect individual instruments as well as complete mixes. There are many good ear training software programs, websites and CDs available to assist you, some free and some not; search out the one that you feel most comfortable with and make ear practice part of your daily routine. Expect the path to be a long and gradual one, but taking the first steps now without hesitation means you will reach your destination soonest and once learned, the frequency spectrum will never change; 2kHz will always be 2kHz; 250Hz will always be 250Hz, and all the rest of the frequencies will be as they are forevermore.

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EAR TRAINING

There is one very simple exercise I recommend. If possible practice for 15 minutes a day making it part of your daily routine. It is a good one to start your session with as the frequencies will stay with you during the mix; it goes like this:

Open your Top 50 session

Insert an EQ plugin into the relevant channel

Play back a song and boost or cut (whichever you are practicing) EQ gain by around 12db, slowly sweep through each octave band back and forth and internalise the sound of each frequency change

Really listen to each master and internalise how the sound is affected by each frequency boost and cut. It is important to know when a song is lacking in certain frequencies or has too much. Also, randomly move around the frequency spectrum. Spend time jumping from octave to octave as if you were learning where the notes are on an instrument. Think of a frequency number and the musical zone, think how it sounds and the frequency range it lives in and try to jump straight to that frequency. Make it your goal to be able to instantly jump to any one frequency and know where it lies regarding musical regions i.e. bass, mid, presence etc. This way you can hear the changes you need to make before you act and begin to control tonal balance.

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TONAL BALANCE

The process of altering tonal balance will come later in this book, for the moment it is really important to focus on the basics. One key feature we established with the Top 50 songs exercise is referencing commercial masters. Begin playing yours and listen critically. Put yourself in the headspace of a mastering engineer. Listen to the sonic quality of each recording. Focus your attention on the overall balance of complete sound. Then hone in on the bass frequencies pounding out of the woofer and then the glassy treble frequencies coming from the tweeters. Feel the energy and the beauty of sound. The sparkling highs, the pulsing lows and focus in on the cutting mid frequencies too. Visualise where the different energy sources come from. Note the balance from top to bottom, how the frequency range level is perfect. Adding or subtracting gain to a frequency would bring instruments forward or set them back and cause an imbalance. Form a picture in your mind of all the instruments, the complete ensemble. Listen to how the sonic qualities are perfectly balanced for each genre and file the impressions away in your memory for recall later. For now the task is to listen, listen carefully and with purpose. Be aware of how nothing sounds wrong. All the songs sound perfectly right in isolation, yet jumping around and playing snapshots of each song shows that all tonal balances are very different. The key is to establish the correct tonal balance for any song you master.

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PANORAMA

Now pay attention to the left to right panorama. Again listen carefully to how the elements are balanced from left to right. Make mental notes and build lasting pictures. Where are the instruments panned? How wide are the extreme boundaries of sound? Does the music stay within or sit outside of the monitors? Is there space in between the elements or a connected wall of sound? Does the master have stereophonic width or monophonic narrowness? The width is certainly an aspect which is often manipulated in mastering. Internalise how the width influences power. Lower frequencies (think bass and kick drum) are not as well localised by the listener; see if you can place the kick drum along the panorama as easily as higher frequency elements. The more mono-sounding a master is the more solidification and power is perceived. The more stereo, the more lush the impression and space, but too much space and the master quickly falls apart.

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DYNAMIC RANGE

As you continue listening, focus on the dynamic range of the music. Dynamic range is essentially the difference between the quietest and loudest parts of a piece of music. But more than that, as the reduction of dynamic range involves tools such as compression and limiting, a master with a narrow dynamic range will sound more ‘produced’, ‘polished’ and ‘fatter’; having been shaped by studio tools and trickery. A wide dynamic range is commonly found in classical music and natural sounding recordings. Music with a narrow dynamic range can be thought of as commercial hip hop and chart music. The euphoric feeling of a dynamic recording that suddenly lifts into a huge crescendo is one of music’s greatest pleasures. However, in the less-than-perfect listening conditions of day-to-day life, the noise floor is so high that the delicate low-level sounds needed to create such an impact would be lost in the sonic trash of noise clutter. The mastering engineer must sometimes act to close the gap between the quiet nuances and pounding explosives by reducing dynamic range.

Dynamic range is also greatly influenced by the musical arrangement, musician’s performances and the mix engineer. As you listen, try to gauge the dynamic range of each recording. Where do you think the mastering engineer has had an impact and where is it simply a matter of production choices? It may help to compare the cleanest, most open sounding classical or folk recording against a super loud pop or hip-hop track to highlight the differences. Another tip, the Offline Loudness Meter you previously used should return a dynamic range figure. Use the meter to determine wide and narrow dynamic range recordings and listen to the difference. Hear how open and clean wide dynamic range recordings are, how the initial transients of each instrument are beautifully reproduced and how lively the feeling of interaction between players is. Compare this with a very narrow dynamic range recording which is all about sustained power: claustrophobic, relentless ‘in-your-face’ energy. Neither is right nor wrong. Each is true to the values of the song, the genre and expectations of the listener.

There are various reasons why dynamic range reduction is necessary or desirable. We will look deeper into this subject later, including the tools for measuring dynamic range, but for the moment soak up the sounds and try to differentiate between the varying levels of processing. Try and listen for compression, clipping/limiting and distortion as a result of over-processing. These effects can be hard to pick up at first, but you will soon develop a greater understanding of, and sensitivity to, the perils of processing as you progress. Do not judge at this stage, simply listen, listen and listen some more.

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PROCESSORS

Although all plugins fall under the umbrella of ‘effects’, it is helpful to break them out into two subcategories - ‘effects’ and ‘processors’. An effect adds to an existing sound whereas a ‘processor’ processes the whole sound. Think of a food processor. All the food enters the machine is processed and comes out totally different. The same applies to sound processors. Sound enters the plugin or hardware unit, is processed and exits in a new form, anywhere from a subtle shaping to a huge transformation. The processors used in mastering, and how you use them, can mean the difference between a demo or pro sounding product, the main reason being that they bring control and a finishing sheen.  Professional audio is a highly controlled product. Controlled music sounds professional and out of control music sounds amateurish. Examples of a processor could be a limiter, compressor or equaliser; indeed these are the most common in mastering and come in single or multiband versions, ‘multiband’ meaning they are capable of processing different parts of the frequency spectrum independently. Probably the most influential processor in modern mastering is the ‘brickwall’ limiter; try to hear its effects. This type of limiter stops any peak whatsoever from passing through a set threshold. It was the processor responsible for instigating the loudness war and has the effect of pinning music back at a certain level so that no transient peaks can pass, allowing the average level to rise. Play some of your Top 50 songs once more and try to pick out the masters that have a definite brickwall sound, where the music hits a threshold and cannot pass. The waveforms of these masters will appear very flat visually with all instruments held back at a certain level and never any louder. Then pick out the tracks that have very little limiting and notice the difference as instruments spring forward as the musician plays louder.

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EFFECTS

As mentioned above, an ‘effect’ adds to an existing sound. In use, a portion of a sound is fed to the effects unit. The return from that unit is known as the ‘wet’ sound. The original version stays clean and unaltered and is known as the ‘dry’ sound. The two signals can then be blended together using either an integrated mix knob or channel faders to control the amount of effect to original sound ratio. Examples of effects could be reverberation, delay, flanger or a phaser. We will cover mastering effects in detail later on, but for the moment let us carry on listening. The two effects of particular interest are ‘reverberation’ and ‘delay’. It is fair to say that the majority of decisions regarding effects have already been taken at the tracking and mixing stages of production and sealed into the stereo mix beyond our reach in mastering. Generally the only consideration at mastering would be whether to add more reverberation or delay. In fact, it is common to add additional reverberation at the mastering stage and little else in the way of effects. For now it is enough to listen to how much reverberation and delay exist in your Top 50 songs. Again, these can be difficult to pick out at first without experience, especially subtle delays used to add ambient thickening. It is fair to say that few sources sound right without ambience of some kind. To fully understand this, load or record into your DAW a totally dry mono sound source and play. Then add a very short reverberation effect (any preset labelled ‘ambience’ should do) and lower the level of the reverb so that you can barely hear it in relation to the original sound source. Now turn the effect on and off to hear the difference between total dryness and ambient dryness.

Going back to your Top 50 songs, try and gauge the level of reverberation and ambience present in these commercial masters. Getting used to how much reverberation is expected in a recording will help flag when a mix is lacking in the future. Listen for the sound of the room, the tail of a reverberation decay or ambient stereo delay. Does the music sound live as though it has been recorded in a concert hall or is it dead sounding as though electronic sounds are confined to a computer’s box with no air? Insert a reverberation plugin to hear what too much reverb sounds like. Switch between presets and try to accustom your ears to the sound of a room, plate, hall and ambience; these are the main four algorithms and it would be good to be able to differentiate between them. Play away and see how much reverberation you can hear. All this helps to add to your ever growing databank of mental mastering images.

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MID-SIDE LISTENING

Now that you have spent time listening to commercial songs as they are designed to be heard, let us look at how we can break them apart and listen in a unique way. Mid-side listening is an audio magic trick that will help you hear stereo reverberation and other side elements in an interesting way. In recent years, thanks to digital processing, the use of mid-side processing at the mastering stage has become accessible to us all. Many plugins now implement mid-side processing as standard. This manipulation of the mid and side channels can nudge your master closer to the bar of professionalism when used correctly. Now is the time to establish a listening technique that will help you judge the properties of mid and side channels. To be able to fully understand the stereo replay system it will help to break the medium down into its basic components. This may seem obvious, but I will document them for completeness.

A stereo replay sound system is made up of two channels feeding two speakers. These channels are commonly known as left (L) and right (R) or A and B. For the purpose of this exercise we will refer to the channels as A and B where A supplies one monitor speaker and B the other. If we think about the sound waves flowing out of the two monitors there are four possible scenarios:

Sound waves come from channel A only – this would happen when an instrument is panned 100% left (assuming channel A feeds the left monitor)

Sound waves come from channel B only – this would happen when an instrument is panned 100% right (assuming channel B feeds the right monitor)

Identical sound waves come from both A and B channels in equal proportions – this happens when an instrument is panned centrally. Audio is output from both channels at the same level and time delivery. The source is perceived as being central between the monitors even though sound comes from the left (A) and the right (B). This phenomenon is known as the centre channel

Sound waves come from both A and B channels in unequal proportions, i.e. different levels, time differences or uneven tonal balances – this happens when an instrument is panned somewhere between 100% left or right and centre or one channel is delayed in time or equalised differently to the other. Audio is still output from both channels but the balance between the two channels determines where in the stereo soundscape the instrument appears

The ‘phantom centre’ channel’s secret lies in psychoacoustic laws. The human brain determines where a sound is located mainly by the time difference between sound waves reaching the left and right ear (although other factors come into play). If a sound source is located to the left of the listener then sound waves from the source will reach the left ear before the right. This information tells the brain that the sound source is located to the left. If a sound comes from the right of the listener then sound waves hit the right ear before the left and the sound is perceived to be located to the right. If, however, there is no time delay between the left and the right ear then the brain’s conclusion is that the sound source can only be directly in front – there simply was no other possibility when the brain evolved all those years ago and there was no technology to deceive it.

The upshot of this is that the two-speaker stereo system offers possibilities of manipulation which can aid our critical listening studies and improve our masters that were not possible with one speaker mono replay. As we learned above, when two identical sound waves are output from speaker A and B the phantom channel is perceived. It is this phantom channel that is referred to as the ‘mid’ channel in the mid/side terminology. Any instruments that are panned centrally when mixing will make up the mid channel. Think of kick drums, bass guitars and lead vocal. These are nearly always panned centrally and perceived as being in the centre channel due to identical sound waves being fed to both channels A and B in equal proportions. The knowledge that these sound waves are identical means that we can cancel them out to silence by inverting the polarity of one of the channels and summing to mono. It is important to note that if there is any difference at all between channels A and B they will never cancel to silence, for example, if an instrument is panned centrally but the channel A is equalised differently to channel B. If there is perfect cancellation to silence we call this a ‘null’ result. You may have come across engineers referring to the null test, a powerful test in sound engineering to determine whether two sounds are identical. The null test is as follows: to determine if two sounds are identical, place them on two separate tracks exactly in time and pan them both centrally. Then invert the polarity on one track. If they are identical they will null to silence. If there are differences, sound will be audible.

If the ‘mid’ information (anything equal in both A and B channels) is silenced then all that remains is the ‘side’ information, i.e. sounds that are not identical in both A and B channels. It is the ‘side’ content that we want to listen to. The ‘side’ information can be thought of as the ‘stereo’ part of the program material. This, as we will see later, can be equalised, compressed, limited or the level raised and lowered in the same way as any other channel (we can also process the mid channel similarly in isolation too). You will see the possibilities for enhancing the ‘stereo’ properties of a track or making it more ‘mono’; or bringing out the overhead cymbals by equalising the sides or tightening the bass frequencies, but again, for now the exercise is to listen to the magic that is mid-side listening.

The easiest way to listen to the ‘side’ information is to insert a plugin that does all configurations for you at a click of a button. Two free plugins I would recommend are Bob Perry’s Listener plugin available from www.bobperry-audio.com or Voxengo’s MSED plugin available from www.voxengo.com. To set up ‘side’ listening manually in your DAW, you will need to split your Top 50 stereo track into left and right channels. Then pan both tracks centrally and reverse the polarity on one track. Press play and what you are now listening to is anything that is different in channel A to channel B, for example, panning, time delay or equalisation. Play away and listen to all the wonderful elements, some of which you may be hearing for the first time!

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PHASE

Finally to expand on phase and the null test mentioned above; phase is a vitally important subject in audio production, more so in recording and mixing than mastering. Phase can be defined as ‘the complete cycle of one sound wave’. When two sound waves are combined they can be ‘in phase’ or ‘out of phase’ partially or wholly. Two identical ‘in phase’ sound waves will combine to create a perfect louder version of itself. Two identical ‘out of phase’ sound waves cancel each other out to total silence. There are many other possibilities in between these two outcomes which we know as ‘phase cancellation’. Identical wave lengths partially cancel and partially enhance as they bounce around a room or are deliberately nudged backwards or forwards along the timeline as in ‘time based effects’ such as flangers and phasers.

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5

BASE TO TOP LEVEL

You may be aware of, or even used, online mastering services. These companies will master your mix in a matter of minutes using automated processing. A few trade names you may be familiar with are LANDR, CloudBounce and eMastered. They use an algorithm that is designed to analyse your mix and change it in a good way, making it more like a professional master. I call this ‘base level mastering’. Let us look at how this works.

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BASE LEVEL MASTERING

The online algorithm compares and shapes the uploaded mix to references held in its memory. The more references the algorithm has to call on, the better the results will potentially be. Many, but not all services, allow you to tweak the interface settings giving you some control over the outcome. It is fair to say the professional audio community was underwhelmed by these services when they first appeared. The thought of a computer mastering the expressive human emotions played out through sound wasn’t too appealing, and the early results were nothing to get excited about. Depending on your viewpoint, they may still be lacking the missing element that only a human can impart. Certainly most mastering engineers feel that way.

My thoughts regarding online mastering are that they have an important role to play in the learning of, and the journey toward mastering. I think of algorithmic mastering as a base level of mastering that is available to everyone instantly. It does not even have to cost you a penny, as all the service providers will let you audition their mastering prior to buying. I see it as a no-lose situation. It is free to use and audition the results. By listening to what a computer thinks is wrong and how it changes a mix, you can learn some valuable lessons as you try to beat it. Rather than dismissing online mastering, I recommend you sign up with at least one service provider (the more the better as they all run different algorithms). There is no need to pay a subscription; their free level is all you need to listen and learn. View it as another tool in your development armoury. Think of online mastering as your base level mastering, the lowest quality of mastering available to you and a way of producing a master in a matter of minutes at the click of a button. The more important question is, how do your masters compare to this base level? I assure you they will improve over time as we work through this book, but where are you right now?

The following experiment will help you answer honestly, it assumes you are already performing your own mastering to some degree and can output a master to act as a comparison in the experiment. If you are new to mastering, go through the process of signing up and uploading a mix anyway and begin listening to your base level online masters. Try this experiment to assess your masters against algorithmic mastering:

Take a mix and master it using your usual techniques. Output a final stereo audio file to your desktop calling it ‘master one’

Take the same mix in its original form prior to mastering and upload it to your chosen online mastering service

Audition the online mastering result and record the output to your DAW

Export the online mastering clip to your desktop and name it ‘master two’

Open a new blank session in your DAW and import both masters to two new separate tracks

Level match both files for K-14 (-14db integrated loudness) listening using the Offline Loudness Meter technique as per the previous chapter

*‘Blind test’ both level matched masters switching between one and then the other

Ask yourself, “which master is better, why is it better, which do I prefer and why”? Write down your answers

*The plugin company ‘HOFA’ have developed a fantastic tool for performing blind tests. The plugin is called ‘4U + Blind Test’. There is a free version which allows the user to compare up to three sources. Download the free plugin from https://hofa-plugins.de/en/plugins/4u/

Be honest and listen critically. Listen to the key areas we spoke about in the previous chapter. As you compare the two versions, visualise and trace the overall frequency spectrum in your mind. Asses the tonal balance among bass, mid and treble. What about the stereo panorama - which is wider or narrower? And the dynamic range? Has either been crushed to within a db of its life, or are both still an open and honest representation of the mix? And finally, how about the level of reverberation and have any effects been applied? Assess all these subjective aspects and listen to your inner body’s quality control system – your gut instinct.

What was your conclusion? If your master was better than the online master then perfect, you already have a good understanding of what is required and a solid foundation on which to improve. Maybe you found that your master was not quite as good, then that is great too because we have established a starting point from which to grow. Before long you will be beating the base level master every time and no longer have a need for the comparison, only to look back occasionally to see how far you have come. Either way, I recommend you repeat the above test on every master you perform to monitor the quality of your output, it takes only a few minutes and acts as a constant quality control and reality double check.

If your master was lacking in some way against the base level, do not feel disheartened; this is the start point on your mastering journey. We are yet to construct a mastering template and work through guidelines on mastering proper. Remember too that a great master is dependent on a great mix. Always bear in mind that a mix may need attention to improve the master. One last thought on our base level concept is this, and it applies to all the areas of music production that you may be involved in: do not use anything you know to be of an inferior quality. This is so important because as humans we are influenced by many forces when faced with decision making and do not always make the correct one. For example, it would be easy to think “yes, the online master is better, but I’ll still use my master anyway as I have put a lot of effort into it”. Please do not be tempted. Always present yourself and your work in the best possible light, wherever the light may shine from. If the online master is better than your master then you need to bring your master up to scratch. Later, I will show you a technique to match your master to the online version, but if you prefer, you can always purchase the online master and accept that on this occasion you have been beaten. Mark it down to experience and make a mental note to get closer each time until you leave the algorithm for dust and race away with your new skills and template driving you to the top level.

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TOP LEVEL MASTERING

I define ‘top level mastering’ as the work performed by professional mastering engineers. This is the highest level of expertise and excellence. Your goal, and the whole purpose of this book, is to move you as close to top level mastering as is possible with the tools and resources at your disposal, thus closing the gap between where you are now and that of professional mastering.

I would advise you to employ a professional mastering engineer to master at least one song, if you only ever do it once, do it now. This will give you a fabulous insight as to your destination and what is achievable with your music (assuming you are mastering your own productions). Be mindful that there are good and bad mastering engineers. Shoot for the best you can afford; a poor mastering engineer is no better, and possibly worse, than no mastering at all. Look to the mastering engineers that have shaped your favourite Top 50 songs, you may be surprised how affordable these engineers are compared to the ‘wanabees’. In my experience there is not a huge price difference for one song. There are great mastering engineers working for a very reasonable rate and poor mastering engineers asking inflated prices. If you do take the plunge, spend some time doing research and aim high, you will not be disappointed by the best unless the source material is lacking. But then a respectable mastering engineer will not take on a sub-standard mix but will instead advise on how to put errant decisions right, so you may get some production advice thrown in. If you are unsure where to start and would like a recommendation, I can, from personal experience recommend Kevin Paul. He is a top guy and amazing talent who will take the time to listen and talk through your vision. Find him at www.kevin-paul.com.

The top level of quality is, therefore, professional mastering. This is the standard set by every one of the songs you have chosen to appear in your Top 50. That is why I asked you to collate your favourite fifty masters, to give you a top level benchmark. Add this to your base level mastering concept and you now have two definitive, measurable top and bottom boundaries showing what is achievable. You can access the base level with the click of a button and the top level already exists in your Top 50 session. There is a gap between the two where you can comfortably sit. You know you can achieve base level already simply by uploading to an online provider and matching or buying their product; all you need to do is keep pushing upwards, learning and practising as you elevate your skills.

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HOW TO BEAT THE ALGORITHM

The algorithm is a great learning tool which can be surpassed by setting up a mastering template and using this template on every mix you master. It will soon become apparent that consistency equals professionalism as you establish a repeatable system within a framework that uses definable benchmarks and references to up your game. Before too long you will progress past the algorithms by practice and adhering to this new workflow and very soon you will be comparing your masters to the professionals with little difference. But remember this is a continual journey. This system will get you in the ballpark and mastering with the pros in your sight, but the final 10 percent is down to your sustained listening and practice. This system will elevate your masters above the base level algorithms, how high you go is down to you. Be sure of one thing – by the end of this book you will have achieved a better mastered product.

LEVEL MATCHING

As you move onto the construction of your mastering template I would like to end this chapter by reiterating the importance of listening to level-matched audio. There are many examples of top engineers being fooled into thinking one master is better than another in a blind test when the only difference is a fraction of a decibel boost. This is a massive factor in mastering. Always listen at an equal loudness when comparing audio or you too may be deceived by the illusion of loudness. Do not be tempted to cut corners and skip the level matching step or you will naturally be drawn to the louder source. If you previously bypassed the task of compiling your fifty favourite songs into a level matched Top 50 session, then I would ask you to please do that now, it is key to this system. If you already have them ready to roll, fantastic, let us look at the tools for our template.

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MASTERING TOOLS

In this penultimate chapter before the construction of your mastering template I will detail the recommended tools to use. It is important to review these before blindly diving into treacherous waters. To know you have the right equipment and means to reach your destination without drowning is the key. Our musical toolkit comes in the form of plugins, individual software programs that slot into a host program. I previously introduced the idea of separating plugin effects into two categories – effects and processors (see chapter 4). In this chapter I will ask you to consider a third type too, that of analysis tools. Analysis tools are exactly as they sound; tools for analysing audio. This may be in the form of meters or spectrum analysers, for example. As we will see, they play an important role in ongoing consistency. Regarding plugins, I recommend reading each plugin manual carefully. I like to download them to my phone so that the PDF files are always to hand.

With so many effects, processors and analysis tools available, it is difficult to know which the best ones for our mastering template are. I will list my plugins of choice and suggest you start with these in the same configuration. This is by no means exhaustive. Feel free to do further research and add or take away any plugins that do not match your preferences. No one setup suits all, and it is absolutely fine and expected to experiment and adapt. I am not including the manufacturer or model as I feel it is easy to waste energy stressing about the pros and cons of different plugins that all perform the same action. Use what you have and learn to get the best out of it. The difference between digital plugins of each type is not huge today. I will simply itemise each type of plugin, give a brief background and list the common controls. Then, after you have built your template, we will look at exactly how to use them and what to listen for. Here are my chosen plugins for mastering:

VU Meter.

Compressor.

Parametric Equaliser.

High Pass and Low Pass Filters.

EQ Matching Software.

Multiband Harmonic Exciter.

Multiband Dynamics (limiter, compression, expansion).

Stereo Imager.

Mastering Reverb.

Mastering Limiter/Loudness Maximiser.

Spectrum Analyser.

Phase-Correlation Meter.

Loudness/Dynamic Range Meter.

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VU METER

Meters are an essential but often overlooked part of audio production. They are considered one of the less glamorous tools as they simply measure a passing signal and return a value. There are various types of audio meters, a VU (volume unit) Meter being one. The humble VU Meter has been around for many decades. Anyone who loves analogue gear will relate to the inner warmth and happiness of a backlit VU Meter gently ticking away. In relation to other meters a VU Meter responds slowly to the passing audio. This means that it misses most of the initial transient information (the very first burst of musical sound) and instead returns results based on the secondary, more constant part of a sound source. We interpret this as being musical and feel more connected to the meter as it shares similarities to how our ear and brain system perceives sound. Put simply, the needle of a VU Meter appears to move with the music. VU Meters have a long proven track record of giving excellent results in a useful way and are still around to tell the tale today. This is why I think VU Meters are still important in audio production. VU Meters have always guided sound engineers in the right direction and I do not see any reason to cast them aside now. A VU Meter differs slightly to peak and Loudness Meters, in that it has a slow response time and reacts neither fast enough to measure peaks nor is complex enough to calculate averages. Each meter has a unique response time much the same as other audio devices. Each has a predetermined rate of attack and decay for the needle or light-emitting diode (LED) display to rise and fall. We call this the meter’s ‘ballistics’. The ballistics of a meter can often be altered by the user to personalise its response. The best advice I can give is to try a few different manufacturer’s meters to find one that works for you. They all differ slightly; the right one will jump about and react to sound in a way that reflects the shape of music as you hear it.

COMPRESSOR

A compressor is one of the most used processors in modern music, especially in popular genres such as pop, rock, metal, country, hip-hop and each subcategory that colours our wonderfully diverse sonic tastes. Some genres limit the use of compressors or do not use them at all, so be aware of the requirements for each genre. If you are recording and mixing your own music you will no doubt have used many different compressors already. If this is your first time of meeting the ubiquitous compressor, let me explain the hows and whys.

A compressor was born out of the need by radio stations to automatically control the level of recorded instruments when broadcast to the public. Unpredictably loud blasts of music could annoy listeners. The problem was that the performance of music is a very dynamic act, often shifting between quiet and loud sections. When music is very quiet it is not easy to hear above the ambient noise of a family listening room, car or bustling social space. Imagine the scenario of uncontrolled radio levels. Your favourite song is played over the airways but it is mastered to a low level so you turn the radio up to feel the power as you move your body and dance away. The next song is mastered extremely loudly, but you cannot know that until it plays. You turned the radio up to a high volume for the previous track and now this new song is playing so loud that it shakes and rattles those unsuspecting tiny speakers. You jump up and lunge for the volume dial, quickly turning the radio down before it explodes, but as the next song plays you realise you have turned it down too far for this new piece of music, aargh! The answer was to establish a controlled replay system where the loudest peaks are reduced and the quietest parts are raised so that a more consistent overall level is reached – enter the compressor.

Compressors soon moved into recording studios too. Level control was, and still is, possible to achieve using a mixing desk fader to manually even out those pesky peaks. Simply move the fader up when the music is quiet and down when it is loud. Some studio engineers prefer to do this, and still do, but there was obviously a need to automate the process. That automation is performed by a compressor. The compressor replaces the finger on the fader of a mixing desk and turns down the loudest peaks when they become too loud and increases the quietest lows when they become too quiet. The exact method of bringing highs and lows closer together will depend on the type of compressor used, but they all perform the same task, i.e. reduce dynamic range and bring more control to the dynamics of the source (the side effects of which may be desirable or detrimental depending on your opinion). As the loudest peaks are lowered, this creates unused space above to nudge the audio up in level and thus make the overall performance sound louder. Compression is used in live performance, recording and mixing, but can also be used in mastering to control the micro-dynamics that exist within every stereo recording.

The common controls that you will find on a compressor are:

Input and output gain: controls the level of audio entering and exiting the compressor.

Threshold: tells the compressor when to compress and when not to. A threshold is set using a minus decibel figure relative to 0dbfs (the highest point of digital audio), for example, a threshold level of -15db means the point at which the compressor starts working is fifteen decibels below 0dbfs. If the audio level is below the threshold the compressor lets it pass unaltered. Only when the audio becomes louder than the threshold does the compressor spring into action and turn the music down. It lowers the level by the amount specified in the ratio setting. Using the analogy of our finger on a mixing desk fader, the threshold would represent the level in your mind that tells you when a source is too loud and must be lowered. As soon as the sound source seems too loud you would move your finger downwards to lower the channel’s output (i.e. the level has passed your threshold). If the music sounds fine then you leave your finger where it is and do nothing as it is below your threshold of action, and so it is with a compressor. Some compressors have a non-adjustable, fixed threshold that you breach by raising the input level.

Ratio: can be thought of as the amount of gain reduction taking place. Ratio describes the input to output relationship of sound passing through a compressor and is expressed as a ratio between input and output levels with the input figure always displayed first. For example, a ratio of 2:1 means that for every two decibels of sound entering the compressor only one decibel is output. Ratio figures of up to 8:1 are considered ‘compression’ settings and a ratio of 8:1 or above are thought of as ‘limiting’. Limiting is an extreme form of compression. Using our finger on the mixer fader analogy again, the ratio would be the distance you move your finger down when the music becomes too loud and has passed your threshold of action. If you were to move your finger down slightly the output level would only be mildly reduced, an action comparable to a low 1.5:1 setting, for example. The further the distance you move your finger the more dramatically you are changing the output compared to the input, more in line with a 3:1 or 4:1 ratio perhaps. Ratio is usually set at a fixed level for the complete song, but can be automated for creativity.

Attack: controls the amount of time in milliseconds that passes between the input level pushing through the threshold and the compressor starting to work. For example, an attack time of 35ms means that when audio passes the threshold there is a delay of 35 milliseconds before the signal is reduced by the ratio setting. Relating this to our finger on the fader once more; if you were to ride a fader manually and move your finger down too quickly you may kill the transient of the sound that you want to shine through. Before too long you will realise that delaying your moves by a fraction is the answer. Your finger now reacts a little later thus letting the all-important transient through – you have added in an intuitive attack time. An attack time lets the initial burst of sound through unaltered. An example of this could be the human voice. With vocals it is important that a listener hears the initial burst of words spoken or sung in an unaltered state as this contains vital information for which to decipher our language. Setting a fast attack time of say 5ms will dull the crucial portion of sound and make the voice more unintelligible. A longer setting of 30ms will let each transient through before the compressor acts and clamps down on the sound, giving a more comfortable listening experience as words are more easily heard.

Release: is the time that a compressor takes to stop working and move from a compressed state to an uncompressed state once the threshold has been breached in a downwards motion. Without a release time, a compressor would switch from compressing to not compressing instantly each time the input level drops below the threshold. This would produce awkward results as the compressor snapped in and out. The release setting overcomes this problem. A real-world example using our finger on the fader scenario would be: imagine your mixing desk fader is set to -10db and you have pulled it down by 3db to control a loud peak, once the peak has passed you could instantly throw the fader back to -10db or slide the fader up gradually over time. The time you take to slide your finger back to the pre-reduced level is the release function. To throw the fader back almost instantaneously would equate to a very fast release time and to slide the fader up gradually a slow release time. Release times are displayed in milliseconds and seconds and are typically longer than attack times.

Knee: controls how the compressor transitions between uncompressed and compressed signals. Knee settings range from ‘hard knee’ to ‘soft knee’. With a ‘hard knee’ selected the switch from uncompressed to compressed signal occurs exactly on the threshold point. However, different knee settings change the shape of compression and create a curve before and after the threshold. Softer knee settings give a smoother and more gradual transition than a ‘hard knee’. Using our finger on the fader analogy for one last time; the threshold level is the point at which you bring a signal down in level, but suppose that threshold is not set in stone, you might want to start gradually reducing gain very gently at a low level and increase the amount of gain reduction when the sound becomes louder. This is what altering the knee setting can bring to the table.

Make-up gain: is the control for raising the overall output level of the compressor. It is usual to set the make up gain to compensate for the amount of gain reduction taking place so that the before and after compression levels are at the same (but with a reduced dynamic range). For example, if you are seeing a -3db reduction in peaks then adding +3db of make up gain will restore the pre-compression level. Switching the compressor on and off should not result in a noticeable difference in level, only less dynamic range.

Meters: there will always be a selection of meters available to track an audio signal through the compressor at various stages. This might be one switchable meter that displays one source at a time or separate meters for the different stages, either way, expect metering for the input and output levels as well as the amount of gain reduction taking place. Gain reduction meters usually rest at zero and deflect downward as compression increases.

Mid-Side processing: some compressors may include mid-side processing. This allows the mid and side channels to be compressed independently. This is especially advantageous for mastering.

It is through the careful manipulation of the above controls that sound can be moulded into something bigger, better, fuller and more powerful when used correctly. If applied incorrectly, broken, ruined, weak, dull and distorted results can ensue. However, it is not always necessary to play by the rules. Techniques such as ‘pumping’ have been developed by compressor misuse so experiment and have fun. You will not always find all controls on all compressors. Many have certain controls that are pre-set in manufacturing such as a fixed attack or release (or both). Some compressors may be so minimal that they only contain a threshold. All compressors though reduce dynamic range by bringing the lows and the highs of level closer together.

All of this explains what a compressor does, and sounds much more complicated than it actually is. It all comes down to spending time with a compressor, playing with the controls, listening and experimenting. As you do, really listen hard to all the subtleties of a sounds envelope - the attack, decay, sustain and release of a sound - and find a method of setting a compressor that works for you. The most helpful method of setting a compressor up for consistent results every time that I have found comes from the document prepared by the Solid State Logic company (SSL) to accompany their Duende X-Comp plugin; the following extract is taken from the X-Comp manual, but can be used on any compressor of choice:

It can be difficult to set a compressor effectively due to the complexity and interdependency of the controls and the subtle dynamic changes that can be difficult to hear. Here is a suggested method which allows you to judge the effect of each control at every step of the way without being distracted or confused by the others.

Start with a fresh instantiation of X-Comp.

Set the Ratio to maximum (50:1).

Release to 20ms.

Knee to hard (0db).

Set the attack to any setting.

The threshold should be set to a point where about -6 to -18db of gain reduction is occurring. Firstly, concentrate on the attack. Whilst adjusting this listen to the effect it has on the leading edge of the transients. Try to ignore the pumping and/or distortion that may be occurring after the attack. You can think of the attack as the control for the ‘size’ of the sound

Next look at the release control. This affects how quickly the signal comes back after being reduced in level. Try to make this as slow as possible whilst still retaining control of the signal

Now move to the ratio. So far this has been set to maximum, so should now be reduced (or not!) to a point where the effects that you have created with the attack and release controls are still occurring, without the sound being obviously squashed

Finally use the threshold control to a point where X-Comp is not compressing all of the time, which will avoid flattening the sound

When using a compressor for mastering it is no longer possible to process individual sounds, but the controls and the process remain the same.

PARAMETRIC EQUALISER

Within audio production, an ‘equaliser’ (EQ) balances the level of frequencies in relation to each other to make the frequency spectrum more equal (or unequal depending on the desired result). That is all they do; turn frequencies up or down, but an equaliser along with the compressor are probably the two most commonly used tools in sound production. An EQ is vital for shaping a professional recording or live soundscape. Without an equaliser it is very difficult to stop certain frequencies building up over many tracks. If recording or balancing live instruments, a great awareness of the different types of microphones and how microphone placement alters sound would be needed to have any chance of a successful sonic blend. For the most part equalisation is essential.

A parametric equaliser works by splitting the frequency spectrum into a number of bands which function independently. Each band encompasses a range of frequencies that can be boosted or cut. An EQ may contain as few as two bands or as many as six or more. Within each band a central frequency is displayed and referred to when boosting or cutting. For example, a cut of -1db at 250Hz means that the central focus of the selected band is 250Hz with a one decibel cut applied. But a parametric EQ is not simply a vertical line affecting a single frequency. Although it appears that way, the reach of an equaliser can be much wider taking in many frequencies. Its shape when boosting or cutting looks like a bell and in fact another name for a parametric equaliser is a ‘bell shaped equaliser’; the wider the shape of the bell the bigger the range of frequencies being influenced. The controls for a parametric equaliser are mostly universal across all makes and models. They are:

Input and output gain: controls the level of audio entering and exiting the equaliser.

Gain: controls the individual level of each frequency band. There is one gain control per band which boosts or cuts the bell curve’s centre frequency in decibels.

Frequency: moves the central focus point of the bell curve up and down the frequency spectrum. Moving to the left selects a lower frequency, moving to the right a higher frequency. Some equalisers allow the crossing over of bands and freedom to roam the frequency spectrum whilst others have fixed boundaries that cannot be passed.

Q: alters the range of frequencies that are affected, i.e. the frequencies that sit either side of the centre frequency. Q defines the shape of the bell curve. A smaller Q number will select a wider range of frequencies and a larger Q number will select a smaller range. For example, a Q setting of 0.5 encompasses a wide range of frequencies whilst a Q setting of 10 equals a narrow band. On some equalisers the shape of Q is pre-set to a particular shape whilst others may have different ‘flavours’ of heritage shapes to choose from, each reflecting prized EQ models throughout history. Also, look out for equalisers where the shape of Q changes depending on the amount of gain boost or cut applied. Generally a gentle boost or cut results in a wide Q which narrows as more gain change is applied.

Meters: there will always be two separate meters showing input and output levels.

Graphical Display: most digital EQs include a graphical display depicting the shape of the bell curve and the amount of boost or cut applied. A few EQs will also display a real-time analysis of the frequency spectrum to show how changes are altering frequency response. Many engineers find this extremely helpful.

Mid-Side processing: may be included which allows the mid and side channels to be equalised independently. This is a great tool to have within mastering.

Although an equaliser is described as being a ‘parametric EQ’, particular bands may have the ability to function as other types of equalisers such as shelving equalisers or high pass and low pass filters. As described above, but worth repeating as this is extremely important, equalisation will always affect not only the central band that appears on the read out but neighbouring frequencies too. When we talk of applying a boost of say +3db at 1kHz, what is actually happening is that 1kHz is the central frequency and given the biggest boost, but a boost is also being applied upward from 1kHz to as much as 10kHz and downwards as low as 100Hz depending on the Q setting. Be aware of exactly which frequencies you are affecting by knowing your EQ inside out and reading the manual. Keep in mind when mastering that it is impossible to EQ individual frequencies of isolated sounds. The ability to equalise mid and side elements separately is the most flexible option available. The key point to remember when equalising (or any processing) is that changing something changes everything; careful small steps are called for.

One final point to be aware of is that most equalisers create a ‘phase shift’ during processing. This will change the sound in a very subtle way and is a technical subject that falls outside of the remit of this book. For our purposes it is enough to know that it is happening and that there is an alternative called a ‘Linear Phase EQ’ that does not change phase relationships at all and is ideal for mastering. But as phase shifting EQ’s have been the mainstay of audio production for many years, we are used to hearing this phenomenon and generally accept it as being normal. And linear phase EQ uses more CPU resources and may introduce additional delay. All I would say is; if you have a linear phase EQ in your arsenal of tools then try it alongside a standard EQ and see which you prefer, but do not worry, millions of hit records have been mastered for decades without a linear phase EQ. Use whatever you have at your disposal, a standard parametric EQ will be fine.

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HIGH PASS AND LOW PASS FILTERS

Another form of equalisation is a filter. A filter turns down unwanted frequencies whilst letting other frequencies pass unaltered, hence the term ‘pass filter’. The most common filters used in audio production are the ‘low pass’ and the ‘high pass’ filter. As the names suggest, a ‘low pass’ filter lets low frequencies pass while cutting highs and a ‘high pass’ filter lets high frequencies pass while cutting lows. This can be a bit confusing to start with as to cut out low frequencies you will need a high pass filter. If you were to place both filters on the same channel you would create a ‘band pass filter’ which lets a certain range of frequencies pass while attenuating the rest. High and low pass filters are usually included within a parametric equaliser and the common controls for the filter sections are:

Frequency: selects the central reference frequency for the filter.

Q: sets the shape of the filter’s curve, i.e. defines how steep or gentle the slope of a curve is. A filter’s Q is represented by a reduction in decibels per octave. An attenuation of -48db per octave will result in a steep curve whereas a -6db per octave reduction will equal a gentle curve.

The shape of a filter’s curve resembles one half of a bell-shaped EQ, which may be preset by the manufacturer and not changeable, although some manufacturers allow the user to select different curves. As with all EQ’s the frequency that you see on the readout for the filter is the central focus point. This does not mean that a high pass filter set to 250Hz will not turn down frequencies above 250Hz - it will. The amount of reduction above the central frequency will depend on the shape of the slope. A gentle slope will usually affect a greater number of frequencies than a steep slope. For example, a gentle -6db/octave slope on a high pass filter set to 250Hz may attenuate frequencies as high as 1kHz. Even with a steep slope of -48db/octave frequencies up to 500Hz and beyond may still be affected to some degree. Filters are predominantly used to reduce unwanted frequencies but can also be used creatively to shape sounds during recording. When mastering, the role of the filter is reserved for controlling the frequencies that are at the outer edges of human hearing - extreme lows and highs.

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EQ MATCHING SOFTWARE

Continuing the theme of equalisation; ‘EQ Matching’ software is a fantastic tool for learning, one that will propel you closer to a professional standard and is key to the quick repeatable system in these pages. It is possible to match EQ curves manually to a certain degree, but a purpose-built plugin will save time when comparing your work to the Top 50 and base level masters. If you already have an EQ matching plugin you will know all about it, but if the technology is new to you I will explain the concept behind matching EQ curves and how to use the software.

Computer processing power has given us the ability to perform functions that were not previously achievable. EQ matching is one such task. The matching EQ is a digital ‘linear phase’ EQ with the ability to use thousands of bands across the frequency spectrum for very precise matching. It is usual to find EQ matching technology within high-end EQ plugins. I do not know of any plugins dedicated to the job. There are a few well known brands that handle the process extremely well and performing a quick internet search for ‘EQ matching software’ will highlight these. EQ matching works by analysing two recordings and generating an averaged EQ curve for each. The two recordings in question are generally referred to as the ‘source’ (the mix you want to master) and the ‘target’ (a professionally mastered guide track you wish to emulate sonically). The greater the portion of material analysed the better the results will be. Once both recordings have been analysed your ‘source’ master is transformed to sound like the ‘target’ by manipulating its EQ curve. The amount of EQ change applied is controlled by various sliders and buttons which vary elements such as the amount of equalisation applied and the smoothness of the curve. It is also possible to make changes to the EQ curve manually in the usual way.

EQ matching is a process that some people frown upon. I have heard it said that it is cheating or that it cannot possibly work because each song is unique. My thoughts are that it is a great tool to highlight what differs in your mastering when compared to professional masters and software algorithms. Do not knock it until you have tried it. If you do give it a go, which I hope you will, ask yourself one simple question – is your master better or worse for EQ matching? Often there will be an improvement, sometimes there will not, but whilst going through the process you will be learning and critically listening in a level matched, blind test environment, which in itself will bring about progression. The usual controls that appear on an EQ Matching plugin are as follows:

Matching: switches the EQ module to matching mode.

Capture: begins taking a snapshot of either the source or target track, whichever is playing through the plugin. There will be a separate section both source and target and one must be captured after the other. The snapshot process might be automatic or you may have to manually click an icon, either way by the end of each song an averaged EQ curve will be displayed.

Clear: allows you to discard the captured curve and try again if you are unhappy with the results.

Matching: controls the amount of EQ Matching taking place, usually by way of a slider. The more the slider is increased the greater the amount of the matching taking place.

Smoothing: irons out the peaks and troughs of an EQ curve, usually with a slider ranging from 0 to 100 and the higher the number the smoother the curve.  

6db Guide: is an alternative EQ shape to sampling a ‘target’ track. Many commercial masters closely resemble the 6db guide curve. Matching to this reference will bring your master in line with the majority.

Pink Noise: similar to the 6db guide, pink noise is a brighter more modern shape for matching.

The method of use is straightforward. Capture the source track’s EQ curve, capture the target track’s EQ curve then match and smooth to taste.

EXCITER

A ‘Multiband Harmonic Exciter’ works by duplicating an audio source which is then split into separate bands. Two to four different zones are standard. The duplicated version is then processed to add excitement and mixed back with the original source, all of which happens within the effect itself. ‘Aphex Electronics’ were one of the first companies to design an exciter back in the mid 1970’s. It is rumoured they discovered this effect by analysing a defective playback channel of a cassette recorder. They marketed it as the ‘Aphex Aural Exciter’ with the techniques involved clouded in mystique and secrecy. The only way an engineer could access the sound of an ‘Aphex’ was to rent the magic box on a payment per final recorded minute basis. As time passed other manufacturers developed harmonic exciters and today there are many exciting choices. The range is varied from rack mounted hardware to computer plugin versions and although Aphex, amongst others, have developed and re-fashioned different types of harmonic exciters, the core process remains pretty much as it did years ago. Subtle harmonic content is created and added to the source through careful use of dynamic equalisation, phase manipulation and harmonic synthesis. It is important to know that harmonic exciters generate new additional synthesised harmonics based on the fundamental frequencies present in an individual sound or mix of instruments. It is an additive process which can easily damage a production if overused. Less is definitely more here as it is easy for your ears to become accustomed to the hyped sound and overdo it.

Exciters work best on higher frequencies. Indeed the original Aphex Aural Exciter only manipulated high frequencies until they developed their ‘Big Bottom’ technology to address the low end. As with many tools in audio production, just because you have something it does not mean you have to use it, just because you can does not mean you should, so if you have four bands of harmonic excitement do not feel you have to use any or all. Bear in mind also that when the concept of aural exciters was first conceived back in the 1970’s there was a need to replace the high end that was lost during the process of recording to magnetic tape. With today’s full frequency digital recording we are able to track as much as we like without any degradation. Having said that, it should still be a tool in your armoury ready for the right track, the right moment, when a touch of magic dust brings a dull mix back to life. The controls available that you should expect are:

Band Select: as exciters split audio into multiple bands there will be a way of choosing which band to edit.

Tune: moves the crossover points (the outer edges of each band) to widen or narrow the range of frequencies affected.

Solo: allows the user to hear only the selected band in isolation.

Bypass: mutes the selected band.

Amount: controls the level for the selected band and thus the amount of harmonic excitement that is mixed back with the original source.

Character: many digital exciters present the effect in different flavours; for example, ‘tape’ ‘tube’ or ‘warmth’ each of which aim to reproduce an analogue sound.

Mix: there is often a global function that controls the total excited sound for all bands that is mixed back with the original source.

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Delay: often a ‘phase delay’ slider is included. This offsets the phase in milliseconds for the selected band relative to the other frequencies in other bands.

Mid-side processing: the ability to excite bands in ‘mid-side’ configuration.

Meters and graphic display: expect separate metering and crossover imagery for each band as well as a graphic display showing the full frequency spectrum in real-time.

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MULTIBAND DYNAMICS

A ‘multiband dynamics’ plugin is a processor that affects the dynamics of a recording. It will usually be comprised of a compressor, expander and limiter. We looked extensively at the standard compressor earlier in this chapter. The knobs and dials are the same here; with a ‘limiter’ being a ‘compressor’ with a high ratio and fast attack; all that is new is the all important multiband functionality and expander section. Any effect described as ‘multiband’ splits the audio into separate frequency ranges, each of which can be processed separately and an expander can be thought of as the opposite to a compressor. Whereas a compressor reduces peaks above the threshold by an amount specified in the ratio setting, an expander increases peak levels above the threshold by the ratio amount. In doing so an expander increases dynamic range, i.e. moves the quietest and loudest sections further apart. The controls you should expect are as follows:

Band Select: as this is a processor that splits audio into multiple bands, there will be a way of choosing which band to edit.

Crossover: moves the crossover points of each band to widen or narrow the range of frequencies affected.

Solo: allows the user to hear only the selected band in isolation.

Bypass: mutes the selected band.

Mix: specifies the amount of processed sound that will be mixed back with the original source for the selected band.

Threshold: the point at which processing starts.

Ratio: The amount of reduction or expansion applied.

Attack: the time that elapses once the threshold has been passed in an upward direction before processing begins.

Release: the time that elapses once the threshold has been passed in a downward direction before processing stops.

Make-up gain: changes the output level of individual bands. There might be a selectable ‘auto gain compensation’ button which attempts to keep the output at a constant level.

Global make up gain: a global control that changes the output level for the complete effect.

Global mix: There is often a global dial that controls the amount of processed to unprocessed sound (the total of all bands) so that once each band is optimised it is possible to control the overall balance of effect to original source.

Metering and displays: As well as the usual input and output meters expect a visual display showing individual bands along the frequency spectrum. Also, it is usual to see a ‘level histogram’ which is a graph charting the history of levels passing through the plugin. This is useful to see how the audio is being reduced or increased in level over time.

Mid-side processing: multiband dynamics processors will generally have the ability to process mid and side elements separately.

All the above controls are applicable whether you are compressing, limiting or expanding. For a more detailed insight into the workings of each function, see this chapter’s ‘compression’ section.

STEREO IMAGER

As the name suggests, a stereo imager allows the user to manipulate the stereo soundfield. A stereo imager comes in multiband or single band versions. Either way the stereo imager uses classic stereo processing techniques that have been around for many years and can be used to correct faults within a stereo recording or to enhance an existing stereo image. It is important for mastering purposes that your choice of imager offer the highest level of mono compatibility and induce a minimal amount of side effects. Past imagers have been known to impart weird ‘pseudostereo’ effects that sound ‘phasey’ and strain the listening ear. Find the most transparent stereo imager you can, i.e. one that offers a way of altering the stereo field without colouring the original sound and keeping the master true to the mix tonally. Stereo imagers come in all shapes and sizes; generally the more you pay the more advanced the options and quality, but this is not always the case so try the many free plugins too. You may already have an ‘imaging’ module as they often appear within a mastering multi-effects processor. The most basic stereo imager will simply have one or two gain controls to rebalance the phantom centre channel against the side elements (see the ‘mid-side’ section for a breakdown of how this works). Conversely, the most advanced stereo imagers have controls to not only rebalance but move and alter specific frequency ranges making it possible to perform rudimentary remixing and produce a totally different sounding master – not that I am suggesting that would be a good idea! The standard controls you should expect to find are as follows (these may be labelled differently or be simplified depending on the manufacturer).

Input and output gain: there may be a control to alter the level of sound entering and exiting the processor.

Width: will widen or make narrower the sound stage. At the minimum setting the image will be summed to mono (left and right channels added together) and at the maximum setting the stereo image will be transported way out beyond the confines of the loudspeakers.

Asymmetry: alters the relative level of the left and right ‘side’ elements whilst keeping the ‘mid’ soundstage central. This is useful for realigning out of balance mixes that have one side louder than the other, but where the centre image is perfect. Using a conventional balance control would affect the whole stereo soundstage including key central sources such as lead vocal, kick, snare and bass guitar, which would not be desirable.

Rotation: rotates the entire sound stage in much the same way as a pan control does for mono sources. The entire sound stage can be moved left or right without the relative levels of instruments changing.

Shuffling: increases stereo width within the bass frequencies. One of the many psychoacoustic phenomena is that bass frequencies are perceived as being more mono than high frequencies. Adding shuffling to the lows can give the perception of a more spacious mix (or a less solid punchy mix if overused or used incorrectly).

Frequency: wherever there is a shuffling option it is usual to find a frequency control to specify the point in the frequency spectrum where shuffling should begin. The ‘frequency’ knob controls the top frequency at which shuffling and thus stereo widening to bass frequencies happens thereafter in a downward direction.

Bass trim: the widening of bass frequencies through shuffling may result in a lower perceived bass level. A ‘bass trim’ control increases bass levels to restore lost power.

Metering and display: As well as the standard input and output meters expect to find a visual display of the frequency spectrum and the composite bands if the plugin is a multiband type. Other visual interfaces vary depending on manufacturer, but it is not unusual to see a representation of the stereo field in other ways, for example, a ‘correlation meter’ or ‘vectorscope’ which we will look at in a moment.

If used wisely a subtle beauty can be added with the deftest of touches from a stereo imager that can shout out ‘pro master’.

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MASTERING REVERB

A reverb will not generally be labelled as a ‘mastering reverb’ unless found within a ‘mastering multi-effects’ plugin. Any reverb plugin will work if the correct preset is selected and adjusted appropriately. As mentioned previously, reverb decisions are generally taken at the recording and mixing stages of music production, but there are times when a mastering reverb will be needed. If you are lucky enough to have a specific mastering reverb in your toolbox then it will have been designed to add a consistent sense of depth and space whilst preserving the vibe created by the original mix. The idea is to add a final gloss and enhance the song as a whole, to add to the character already present whilst never smearing or muddying the mix. You may have one or two presets to choose from, for example a room or plate, or the options may be pretty limited. There is a very good reason for this; it is not the time for major dark tunnel presets with a swirling twisting decay! Subtlety with a sympathetic room is the way to go. The controls will generally be the same as for any other basic reverb patch with the most common as follows:

Reverb: selects the types of reverberation available, for example, a plate or room.

Mix: controls the ratio between the wet reverb signal and the dry original signal.

Room size: specifies the size of the room simulation. This will appear as a representation of an actual room size in metres cubed (length x width x height = m3) or a simple rising numeric scale with larger numbers equating to a bigger room.

Room dampening: cuts brightness from the room sound by lowering the level of high frequencies present in the reverberation tail. The more high frequencies the livelier, brighter and closer a sound appears. Conversely, the more dampening applied the more distant the effect with the sounds moving backwards in perception.

Pre-delay: introduces a delay in milliseconds between the original signal and the beginning of the reverb. In the natural world sound consists of a direct impulse and reflections which bounce off nearby objects to build into what we know as reverberation. Pre-delay is a way of delaying the onset of reflections. Adding a pre-delay can have the effect of making the original sound source appear clearer.

Room width: there may be a control to alter the stereo spread of the reverb signal which affects how wide the reverb effect is perceived between and beyond the loudspeakers.

Solo and mute: solo provides a way of hearing the reverb in isolation without the original source. Mute allows the original mix to be heard without the mastering reverb. Sometimes it is only by taking away and then reintroducing an effect that you can truly evaluate its impact.

Low and high frequency cut-offs: reduce the amount of high and low frequency content contained within the reverberated signal (see the ‘High Pass and Low Pass Filters’ section within this chapter).

Metering and display: the standard input and output metering system will be present and there may be an integrated phase-correlation meter to help visualise what impact the additional reverb effect is having on the stereo image (see ‘Phase-Correlation Meters’ later in this chapter).

Mid-side processing: depending on how advanced the effect is there could be a way of manipulating the effect in ‘mid-side’ mode.

When considering the use of mastering reverberation, be true to the original concept of the production, always be careful never to break the mix and make it worse – as always, less is more.

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LOUDNESS MAXIMISER

A loudness maximiser is a super fast limiter with ‘lookahead’ properties and an automatic make-up gain. The lookahead function allows the maximiser to analyse each waveform before deciding how to process it - a useful advantage over conventional limiters. Although the lookahead feature implies magic powers that can see into the future, this of course is impossible and is actually a smoke and mirrors trick. Rather than looking ahead, the maximiser delays the audio in order to investigate and react. We do not hear any delay because modern computers align every piece of audio and plugin delay impeccably via a system called PDC (plugin delay compensation). It is fair to say that the sound of modern music has a lot to do with compression, limiters and loudness maximisers (also known as mastering limiters or clippers). Today’s engineers find great difficulty in competing with loud integrated levels without a loudness maximiser. The ‘Waves L3’ is probably the most famous of them all and the one often credited with kick starting the loudness revolution (or war depending on your viewpoint). Though their impact is great, a loudness maximiser is generally easy to use with only a few controls. The key to their successful application will lie in your critical listening skills. I will explain all as we walk through the ‘Template in Use’ chapter, but for now let us examine the common controls you can expect to find on a maximiser:

Threshold: controls the level at which the plugin starts and stops limiting. As audio crosses the threshold in an upward motion the output is limited and as it drops back below the threshold, limiting stops with unprocessed play resumed.

Ceiling Output: the maximum level at which sound is output and the point at which no audio can pass, no matter how hard the plugin is pushed. It is essential that the maximiser limits the output on or below the digital ceiling of 0dbfs as any peak higher will be clipped (distorted) when rendered to 16-bit resolution, which not only sounds terrible but can potentially distort a playback-system. Many mastering engineers use -0.1dbfs or -0.2dbfs.

Attenuation: displays how much limiting is occurring.

Quantise: reduces the bit-depth of audio from a higher number to a lower one, for example, from a 24-bit mix to a 16-bit master. There are various methods of transforming one bit-depth to another including ‘truncation’, for example, which simply discards the least important bits in order to match the requirements of the new resolution. Quantisation happens every time a digital audio process changes the original data, which is basically ninety nine percent of everything performed within a DAW including every time a fader is moved!

Dither: adds low level noise to disguise the unpleasant side-effects of quantising which affects audio in a negative way as the full beauty of a waveform is distorted through mathematics. The degradation is very subtle but is there nonetheless in the form of low-level digital noise. Dithering attempts to cover up ‘quantising’ with a more acceptable analogue style hiss. Dithering should be performed when exporting from your DAW to a lower bit-depth.

Noise shaping: places dither noise in the least distracting section of the frequency spectrum. The exact position is debatable with each noise shaping method settling on slightly different techniques and frequencies, but in general dither noise is moved out of the region at which humans are most sensitive to a high of above 15kHz or so.

Release: the time a maximiser takes to stop processing once audio passes the threshold in a downward direction. Release might be controlled automatically, manually or include a button to switch between the two (attack time is always automatic as to add in an attack time would allow unwanted peaks to slip through in an unlimited state and cause distortion).

Gain: controls the level of audio at the output stage of the plugin. This could be automatic or manually set. If automatic the function will be called ‘auto gain’ which continually maximises the audio to its loudest potential.

Mastering ‘maximiser’ limiters are generally classed as ‘brick wall limiters’ meaning that at no point will the level of program material be higher than 0dbfs, the digital audio ceiling point. Brick wall limiters have a fixed ‘hard knee’, meaning that the threshold point is the exact point that limiting starts and stops. More advanced mastering limiters may even include two staged limiting with transient controls for more varied manipulation.

SPECTRUM ANALYSER

A spectrum analyser will measure the real-time level of frequencies within a piece of music or a single instrument and display the ongoing results on a graphical display. Spectrum analysers have been around for many years and usually come in third octave versions. They are used by audio engineers who find it useful to ‘see’ as well as hear the frequency spectrum over the timeline of a song. A spectrum analyser is not an essential tool, and indeed there are still those who never use a spectrum analyser, instead relying on their ‘golden ears’. But for our purposes, and certainly for the most part of your progression into professional masters, I would recommend that you use a spectrum analyser as a major part of your workflow. It will help guide you into making better decisions as you analyse both your work and the work of professional engineers. Using a spectrum analyser during your critical listening practice will help build an image of what a professional master looks like in any given genre. You will start to notice certain patterns, certain frequencies that are consistently lower than others and EQ curve shapes that crop up time and time again. When analysing your own work, a spectrum analyser can highlight areas in the frequency spectrum that are out of balance. Over time your goal is to be able to predict the findings of a spectrum analyser simply by listening and so lessen the dependence on it, moving instead to feel and instinctive decisions. The common feature set of a third-octave spectrum analyser is as follows:

Analysis graph: the main bulk of the visual display will be taken up with the analysis graph that shows decibels vertically and frequencies horizontally. Each third octave division of the frequency spectrum rises and falls with the music and can be thought of as many individual level meters, each a third of an octave width. More advanced analysers may include a piano style keyboard to show the musical note that equates to any given frequency.

Level meter: shows the level of audio passing through the analyser. This plugin should not change the sound in any way, so one level meter is all that is required. It is often the case that the level meter is switchable to function as different meters, for example, a Peak Meter, RMS Meter or a Loudness Meter.

Hold: there will always be a function to hold the peak levels of an EQ curve either momentarily or indefinitely. The time will be set in the settings and held as a snapshot in time, a time dependant refresh state, or both.

L/R: expect to be able to change the channels being analysed. It is usual to be able to switch between left, right or left + right.

Phase Meter: more advanced spectrum analysers incorporate other metering to show the stereo field and phase relationships between left and right channels, for example, a ‘goniometer’ which shows the panning position of instruments or ‘Phase Correlation Meter’ which measures in and out of phase.

An internet search will point to some really good free spectrum analysers that perform the basics well. Pay more and expect more advanced features. Spend time trying demo versions of different brands to find your perfect analyser as each looks and feels slightly different. Look for an analyser that contains all the features that are important to you, one that mirrors the way you hear music and tells you the truth about the pulsing sound waves flying out of your loudspeakers. And, as important, one that is easy to use – this is one tool you will be using a lot so you want to feel comfortable with it.

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PHASE-CORRELATION METER

A ‘Phase-Correlation Meter’ is used to monitor the phase relationship between left and right channels of a stereo input, basically a real-time monitoring system that shows the stereo properties of a two-channel signal. The Phase-Correlation Meter was originally developed in order to aid the process of cutting vinyl records. Any out-of-phase issues had the potential to affect playback and make a record player’s stylus leap out of the disc’s groove. Although the technology is old, Phase-Correlation Meters are still very much a part of the mastering engineer’s toolbox and can be found integrated into mixing desks, DAW metering and analysis plugins. These meters come in many different designs. The most common and the simplest of which can be found in the form of a horizontal rectangular bar containing a moving needle and numerical scale. On the extreme left hand side is minus one, in the centre is zero and on the extreme right is plus one. There may be other intermediate numbers but these are the three important ones. When a stereo signal is passing through the meter in a totally mono state (i.e. the right and left hand signals are exactly the same) the needle will point fully right to plus one. The greater the stereo effect within the signal (i.e. more variation between left and right) the closer the needle points to zero with the centre point being the fullest degree of natural stereo. A meter reading of a negative figure to the left of centre means that the left and right channels are out of phase in some way with fully left being totally out of phase. It is not unusual to see occasional fleeting jumps into minus numbers which can be ignored, but these exceptions should be brief. If the needle’s move left of centre is sustained this signifies an out-of-phase state that would be disconcerting for the listening ear. For the most part, in standard stereo recordings, the needle should be dancing around between plus one and zero. As mentioned previously, Phase-Correlation Meters are often built into other plugins and come in different forms, so check to see if you already own one. If not there are many great free and paid options.

DYNAMIC RANGE METER

‘Dynamic Range Meters’ are a fairly new system of metering within the lifeline of music production. The first to gain popularity was the ‘TT Dynamic Range Meter’ designed by ‘Pleasurize Music Foundation’ which recorded dynamic range against a measurable scale. The ‘TT’ Meter was born out of a desire to reintroduce pleasing dynamics to recorded music. By making the engineer aware of the dynamic range of a piece of music it was thought this would counteract the loudness war that was raging at the time. We covered the principle of dynamics and dynamic range in the ‘Critical Listening’ chapter, dynamic range being the difference between the loudest and quietest parts of a piece of music. A Dynamic Range Meter, therefore, measures and shows us the gap between the two, i.e. how dynamic our master is. Dynamic Range Meters have come a long way since the TT Meter and have been superseded in many studios by the Loudness Meter. There are very few controls or ways of interacting with a Dynamic Range Meter. It will simply monitor the passing audio and return a real-time readout. As mentioned previously, one important feature you will want is the ability to analyse audio in an offline state for faster than real-time processing. This function is extremely important for the speed and efficiency of our mastering system. The standard controls may include the following:

Dynamic Range: returns a visual indication of passing dynamics.

Peak: it is usual to measure and return peak levels of passing audio.

RMS: it is also usual to measure and return RMS levels of passing audio.

Metering: a standard Level Meter is included which often shows Peak, RMS and Dynamic Range levels in a combined visual way, a Correlation Meter may be included too.

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LOUDNESS METER

As mentioned above, the ‘Loudness Meter’ has in many ways superseded the Dynamic Range Meter. In today’s streaming world it is more important than ever to gain control over the loudness of your music. A Loudness Meter is the tool that will measure the integrated loudness of a piece of music and return a ‘LUFS’ and ‘true peak’ figure which are so important in online delivery. ‘LUFS’ stands for ‘loudness units relative to full scale’ and is the latest attempt to unify the levels of program material. I have already covered why this is important and will expand further when outputting a master. For now it is enough to know that a Loudness Meter is the way to go regarding loudness units or ‘LUFS’. A Loudness Meter can look extremely complex at first glance with all sorts of metering, scales of measurement, K-weighting, targets, short term levels, long term levels, ranges and so the list goes on. But each has its purpose and for the most part you will simply be selecting a target and pressing play. The important point is to know why you are using it and what it will bring to the table. All of which will become clear as we output our master in ‘Chapter 9: Exporting’.

That concludes the overview of all the plugins I have in my mastering template and which you may cross paths with on your mastering journey. If you are able, I recommend you include them all in your template in the exact same order as I lay them out in the next chapter. You may not have access to all of them, choose not to include some or indeed there maybe additional plugins you would like to add too. Remember many of these plugins exist as multi-effects plugins which group similar units together or as free plugins without a fee. Download demos of as many different manufacturers as possible to find your happy workflow. It is amazing what a difference it makes having a GUI (Graphical User Interface) that you are comfortable with. The sound may not be much different, but the freedom of working without thinking nor fighting a plugin or head scratching your way through a maze of menus cannot be overstated. The ease and speed in which you interact with a plugin will influence the decisions you make. Let’s go build our mastering template!

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MASTERING TEMPLATE

Having laid down a foundation of skills, tools and knowledge to enable you to begin mastering successfully with confidence, the next step is to gather all you have at your disposal into one reusable template system. All audio must enter, be processed and exit this template.

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WHY A TEMPLATE?

One bonus of digital audio is the ability to instantly recall every conceivable system configuration. All the settings, levels and routings for any given project can be stored and recalled at the click of a button. The name for files that contain all this information is a template. I absolutely love templates; they save me many hours through not having to repeat the same tasks over and over on every project. The extra free time can be better spent on pursuits such as critical listening, ear training and mastering more songs!

The mastering template is separate from the Top 50 session that we set up earlier. I like to keep my Top 50 session and my mastering template separate, importing and exporting between the two as is necessary. This works for me and suits my ethos of clarity and separation between processes, which is important to me. But once you have learned the complete system you may prefer to merge the two by importing your Top 50 into your mastering template. Indeed, there are many approaches to organising all the different elements that make up your template with no right or wrong way. I have spread my processors and effects over different channels but theoretically they could be condensed into one or two channels. My advice is to run with the system as I propose and then have some fun making it your own by adapting the layout to suit your own personal values. So let us create some magic to conjure up a spirited master.

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MASTERING TEMPLATE

Start by creating a new blank session in your DAW. Then use the ‘save as’ function to place the file in an easy-to-find location on your hard drive naming it ‘Mastering Template’. It is important to start with a new blank template that contains no previous routing or pre-formatted configurations. I will ask you to create a number of tracks/channels. When tracks are created they appear on the editor screen of DAWs with a subsequent channel within the mixer screen. You cannot usually have a track without a channel. Tracks hold the audio clips which feed each mixer channel with sound. The music then passes to the master bus (via auxiliary buses if required) and finally to your monitors. All tracks/channels are stereo as we will be handling stereo mixes. If you are ever faced with a mono mix to master, stereo configuration is still fine; it simply means that the left and right side of the audio will be identical. Go with me on the channel names until you are familiar with their function. During setup leave all the channels routed to the master bus in their default state. Once all the tracks/channels have been created I will provide a routing grid to show which outputs feed which inputs. Regarding plugins, do not set any specific settings at this stage; we are simply creating our template – we will look at specifics in the next chapter ‘Template In Use’. Once each effect has been inserted dis able each plugin so that none are functioning. In the initial state upon ‘start up’ we want all plugins to be turned off and only activated in a thoughtful way when needed.

CHANNEL 1: TIMELINE

With your new blank mastering template open, insert the first stereo track/channel and name it ‘Timeline’. This channel is blank upon ‘start up’. You will use this track to create blank items along the project timeline, thus saving time setting the start and end points of each export. The channel strip is as follows:

Insert Fx

n/a

Sends and Returns

n/a

Panning

Centre

Fader

0db

Output

Master

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CHANNEL 2: REFERENCE

Now create a second stereo track/channel and name it ‘Reference’.

Insert Fx

n/a

Sends and Returns

n/a

Panning

Centre

Fader

0db

Output

Master (mute)

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This track/channel is where our reference tracks belong, so, go ahead and open your Top 50 session and copy and paste the fifty, K-14 level matched song clips across. There is no need to move the source files for the tunes on this occasion, as your Top 50 session files will always be in the same location for evermore.

There are no plugins to insert on this channel.

There are two important points to note:

The output from this channel should never run through any plugins. Only route directly to the master bus if the master bus is free from plugins, which it is in this template. If you later choose to place plugins on the master bus then you will need to route the reference track out to your monitors via a separate pair of outputs and use a switching box or mixing desk to change sources. Alternatively, there are plugins available that sit on the master bus and allow the user to load reference tracks directly to its interface and switch between sources within the plugin, for example, Sample Magic’s ‘Magic A/B’. Such a plugin would replace the ‘Reference’ channel

Any further commercial reference tracks that you add will be mastered at loud levels that are much hotter than the start level of the mix to be mastered. This means that any additional reference masters must be level matched to an integrated loudness of K-14 prior to playing or it will sound far too loud for your K-14 monitoring system. This is important! If you have calibrated your monitoring to K-14 83db, playing a normalised master with an integrated loudness of -6db will play 8db louder than it should. That equates to a playback level of 91db which will make you jump out of your skin and leap out of your comfy mastering chair in a blind panic!

CHANNEL 3: ORIGINAL MIX

Insert a third stereo track/channel and call it ‘Original Mix’. This is the channel where you will place the all important mix in need of your mastering finesse.

Insert Fx

n/a

Sends and Returns

n/a

Panning

Centre

Fader

0db

Output

Master

––––––––

No plugins to insert on this channel, keep this one clean as this is the original mix you have been entrusted with or exported yourself, any processing will be performed further down the channel chain.

CHANNEL 4: PAR COMP

Create a forth channel and call it ‘Par Comp’. This channel is the parallel compression channel that can be blended with the original mix when required. If you are not familiar with any terms such as ‘parallel compression’ all will become clear in the next chapter ‘Template In Use’.

Insert Fx

Compressor

Mastering Multi-effects

Sends and Returns

n/a

Panning

Centre

Fader

-14db

Output

Master (mute)

––––––––

I suggest you insert two different ‘flavours’ of compressors, i.e. two different brands that perform differently. Look for an analogue and a digital compressor, for example, or do as I do and place one dedicated compressor and a multi-effects plugin that will smash the sound with more than a simple parallel compression giving two extremes. As mentioned above, do not concern yourself with specific settings yet, we walk through these as we look at the template in use. Note that this channel should be muted to start with. As the parallel compression signal is mixed with the ‘Original Mix’ signal, if this channel is not muted when it is not in use, you will hear the output from both channels. Therefore, only un-mute it when you want to apply some parallel compression.

CHANNEL 5: PRE-MASTER

Insert a fifth track/channel and call it ‘Pre-Master’. This channel will receive the output from the ‘Mix’ channel and the ‘Par Comp’ channel combined.

Insert Fx

Matching EQ/Mastering Effects

Stereo Imager

Limiter

Sends and Returns

n/a

Panning

Centre

Fader

-1.5db

Output

Master

––––––––

I think of this channel as my ‘tricks’ channel. Parallel compression should always come first as we want any subsequent processing to affect a parallel compressed signal. Then comes any EQ matching and stereo imagery trickery! Note the first appearance of a limiter at the end of the chain as we start to gently bring down the peaks so that the level as a whole can be increased – little and often are the key.

CHANNEL 6: LIMITER

Insert a sixth track/channel and call it ‘Limiter’. This channel adds a second layer of limiting should it be required.

Insert Fx

Limiter

Sends and Returns

n/a

Panning

Centre

Fader

0db

Output

Master

––––––––

You may question whether this channel is necessary. It is better to use several dynamics processors in sequence with each doing very little, rather than have one processor doing too much. I prefer to dot limiters about my template and use different models for maximum choice and colour. Remember, we do not need to use any or all of the limiters, each can stay deactivated and switched in and out to compare and utilise. You could add this limiter to the plugin chain of the previous channel, but I prefer to separate every process out and label it clearly. This helps me to consciously think of the impact every channel and plugin has and what its purpose is. Then, rather than clicking away blindly in a mindless fashion lost in a long chain of effects, I instead think, would it help to have a touch more limiting? If so I will go to my ‘Limiter’ channel.

Insert your limiter of choice.

CHANNEL 7: FINAL MASTER

Create a seventh track/channel and call it ‘Final Master’. Again, insert and disable all the plugins; start with them off and use them only when there is a reason to do so.

Insert Fx

Parametric Equaliser

Linear Phase Equaliser

Three band Equaliser

Multiband Dynamics

Multi-Effects

Maximiser 1

Maximiser 2

Sends and Returns

n/a

Panning

Centre

Fader

0db

Output

Master

––––––––

This channel offers the last chance to fine tune your master. Think of it as the final dymanic/EQ control with a bit more finesse than standard operation. Here you will find multiband, mid/side functionality and dynamic EQ as well as two different ‘flavours’ of a final loudness maximiser. If you do not have the ‘bells and whistles’ versions of plugins, pick the best you have and go with it. A small range of plugins that you know inside out will outperform a trillion top flight plugins that are too numerous to know how each performs best.

MASTER CHANNEL:

No plugins on the master channel as all the processing is happening on the individual channels.

Insert Fx

n/a

Sends and Returns

n/a

Panning

Centre

Fader

0db

Output

DAW monitor output

––––––––

You may have been expecting to see a ‘dither’ plugin at the very least, but dithering is handled within our last maximiser plugin. As long as dither is the very last process it matters not that it is performed on a mixer channel rather than the master bus. You can prove this to yourself by running a null test, a great tool to check that two signals are identical. It goes like this; import a piece of music to the ‘Mix’ channel that is of 24-bit or 32-bit resolution and then activate a dither plugin on the ‘Final Master’ channel. Set the dither plugin to 16-bit and export a master naming it ‘Test 1’. Then deactivate the dither plugin and instead activate one on the ‘Master Bus’, exporting a second test master calling it ‘Test 2’. Now close the template and import both test masters into a fresh blank session, invert the polarity on one track and press play. The result will be total silence signifying that both tracks are identical. This confirms that the dithering effect is the same whether placed on the master bus or the final channel as long as it is the very last process.

––––––––

ROUTING GRID

Until now we have left all the channel outputs routed to the master bus. Having set up the individual channels we can now complete the final connections by following the routing grid on the next page.

Within a DAW almost any output can be routed to any input. A great way of showing all the ins and outs is by way of a visual overview called a routing grid. You will note from our grid that only the ‘Final Master’ channel in our processing chain remains routed to the master bus. Although the ‘Timeline’ and ‘Reference’ channels are routed to the master bus too, these are not part of our processing. The ‘Timeline’ channel will never contain any audio and the ‘Reference’ channel remains muted until called upon, at which stage it will override all other channels. So, go ahead and route all the outputs to the relevant inputs. The point of connection is denoted by an ‘X’ and I have provided you with an example to show the process. When you have finished, be sure to save and backup your template.

8

TEMPLATE IN USE

Fantastic, now all your hard work is about to pay off; the prep work and one-off setting-up tasks that will serve you well for master after master has been accomplished. It is now time to run through the process of using your template for the first time as we bring the critical listening, Top 50 references, K-System monitoring, metering tools and plugins together into one complete system. You will soon see how everything flows seamlessly. If you have not done so already, please back-up your template and store it safely in a location away from your main computer.

Now create a new session by opening your mastering template. In some DAWs you create a new session based on your template and give it a new name. If you are actually opening your template, immediately rename the session with the title of the mix you are mastering using the ‘save as’ function. Always make this your first task to prevent overwriting the original template file. It is important to know the basics of your DAW and your computer inside out; how they name and store files, when they copy source files or merely reference their location, how to copy all source files into the session folder for archiving and where files go when you delete them. All of this is really important if you are to avoid losing program material now or in the future if a project needs to be restored.

The steps that follow describe the complete system in an easy-to-follow logical format. As a quick overview so you know what to expect, the process is as follows:

First produce a master yourself using my guidance notes and your reference tracks and template, whilst working quickly and intuitively. Follow your instincts and do your best at all times.

Then, when you have finished your master you will compare it to the base level ‘algorithmic’ mastering by taking the original mix and uploading it to your chosen online mastering service provider. If yours beats it, then great - good work. If not, you will use the new information to bring your master in line with the base level.

You will then finalise your master, confirming that it is of a professional standard by placing the master within the context of your Top 50 songs.

Finally, you will export your master to your chosen LUFS number ready for uploading to an online hosting site.

Just do your best and trust in the process and your best will quickly get better and better through practice. As you become more familiar with this system the process will all become automatic, seamless and swift – it really takes very little time once all the processes are second nature. Any new learning curve can seem daunting at first but with methodical, repeat application everything will fall into place.

There is one important point to bear in mind: be aware that the mix may not yet be ready for mastering. You will become a better judge of what is and what is not master-ready through experience. A great master comes from a great mix, but not every mix can be mastered successfully. A poor mix will always result in a substandard master. Sometimes there is no alternative other than to remix a song to see the project through to completion. If the track is your own mix you are mastering, then it is easy to jump between mix and master in a matter of moments with modern technology. Take advantage of this boon and do just that. Thinking of both mixing and mastering in tandem will be of great benefit to your development. One influences and improves the other and vice versa; you will soon be thinking of mastering as you mix and mixing when you master. When mastering another artist’s work, the decision is more difficult. If you feel a mix is not quite ready to be mastered, be prepared to talk to your client and point out the shortcomings in a positive and constructive way. It is far better to have a conversation right away if you have concerns, rather than sit on your fears and spend hours trying to make a broken mix slightly less broken. You will never mend a flawed mix and will only be able to achieve a poor master that will reflect negatively on you the mastering engineer. Be decisive and confident even if in reality you are unsure. It is better to make decisions even if they are wrong and learn from your mistakes rather than sit wondering, frozen by indecision. Success in music production lies not only in knowledge but also in confidence, belief and attitude. Start every session with a positive attitude believing that you will nail it no matter what.

Let’s master this mix!

STEP ONE: IMPORT

The first task is to import the mix to be mastered into the template. When importing a mix file ensure that a copy is placed in the session folder. Some DAWs will only reference the file from its original location by default and not copy the actual data across. This can be problematic if you need to rework a master at a later stage and the source audio has been deleted. You will be faced with an ‘item missing’ warning and no mix to master. It is good practice to manually copy and paste each file into the base session folder and import from there so you are certain all is correct. Go ahead and import the mix into your template to the ‘Original Mix’ track, press play and check all is good. Each plugin should be deactivated, the mixer faders should

be in their default state and the only channel routed to the master bus should be channel seven ‘Final Master’. If you have set up each channel’s routing correctly, as you play the mix the meters on channels three through to seven will be responding to the audio and you will hear music through your monitors and headphones.

STEP TWO: CHECK THE MIX IS READY

Play the mix through a few times to get a feel of the production style and vibe. Ask yourself ‘is this mix ready for mastering’? Do you feel the mix is of a good enough standard to be mastered? Can you envisage the finished master and see ways to improve the mix making it taller, wider, deeper, brighter and fuller or is there a stumbling block that can only be remedied by an alteration to the mix at the mix level? Initially, if mastering your own mix, it can sometimes be hard to critique your own work honestly and constructively; you could be too involved and too close to your work. It is a great skill to be able to step back and listen as though it is the first time you have ever heard music you are familiar with. Giving yourself space between mix and mastering sessions may help or seeking out a knowledgeable friend who you trust to tell you the truth in a positive way rather than what they think you will want to hear.

The mix will seem a long way off professional master standard at this stage but are the fundamental musical building blocks laid securely on which to build? There should be no elements widely out of balance unless for deliberate artistic reasons. If the answer to the question is, no the mix is not ready, then it is back to the original session to put right everything you feel is wrong before rebooting the mastering process once more. If the answer is yes the mix is ready to be mastered then proceed to step three.

STEP THREE: CHOOSE A REFERENCE TRACK

The beauty of this system is that you now have fifty professionally mastered reference tracks within your mastering template to guide you. Chose one song that is closest to the finished master vision you have for the mix. You can slide the mix along the timeline to see how it compares to any given reference tune by toggling the solo on and off. You may settle on a couple of songs, for example, you might choose one song for its low end and one for its loudness and overall sonic qualities. The song you choose should share similar sonic characteristics to the mix and the end master that you have in mind. It should be one that you admire and wish you could master to the same skill level, but not wildly different in terms of context. It is no good picking a hip hop track as a reference for a slow country ballad. Be realistic as to how far you can bend and shape a mix towards another master. Always remember that less is more, so the closer the reference to the mix the better.

STEP FOUR: LEVEL MATCHING

As we have established, all references tracks, and indeed any music played over your monitors, should now be level matched to your calibrated monitoring system so they play at an integrated loudness level of -14db. This is a vital step and necessary to hear all music equally. The fifty reference tracks you copied into the template were all level matched when we set up our Top 50 session; therefore, all that remains is to set the mix to -14db too. To achieve this, run the ‘Offline Dynamic Range Meter’ process as detailed earlier and adjust the clip gain of the mix so that it plays at -14db. Try and form the good habit of double-checking your calculations. Inserting a K-14 meter will give a less detailed but perfectly useable result. Remember it is not the dynamic range figure we are interested in yet but the RMS number - the musical average used in the K-System.

Both your references and mix should now be playing at an integrated loudness of -14db at 83db. You may be surprised how much better the mix now sounds compared to the reference track and we have not even started mastering yet. Simply by bringing the two tracks in line level wise takes away any loudness advantage one may have. It may appear to you that the mix has fewer problems now and indeed there may not be much wrong at all. Without this vital step that many people pass by, problems can be perceived and acted on incorrectly. Be aware that the mix will fluctuate in level as you apply mastering processes. This will be a constant moving target to check regularly while you work, but for now it is only by level matching both mix and reference that you can get a true picture of the differences and a valid start point.

Jump about the timeline with your new level matched mix, listen to brief snapshots of each song and note how they differ to each other and your mix. Start to write down differences and similarities between your references and mix. Check that your chosen reference tune from step three is still the best.

You should now have your reference track and your mix playing side by side and a method of switching between the two. You never want to hear both tracks simultaneously nor hear the reference track coloured in any way. Enable ‘solo overrides mute’ in your DAWs settings, if it supports this function. This saves time in un-muting muted tracks, but also the carnage of both reference and mix playing at the same time as you juggle the mute and solo buttons. With this elegant solution you simply keep the reference track muted at all times and press the solo button to engage it and silence the mix.

STEP FIVE: LISTEN

Listen carefully, listen critically. Use all you have learned from the ‘Critical Listening’ chapter. Remember there were four key areas (excluding final loudness maximisation): tonal balance, stereo image, dynamic range and effects.

Switch between mix and reference track and continue to compile the list of the differences between the two and problems that you need to solve. Also, if there are areas that are fine and best left alone, make a note not to mess with these. This list will provide the roadmap to travel from mix to master. Work quickly and follow your instincts. What instantly jumps out at you as being wrong? Switch often so that your ears do not adapt and compensate for bad balancing, which they will given the chance. Work in short bursts of 20 minutes at a time before taking a short break to refresh your ears. Okay, with a list of ideals, let’s transform this mix to a pro master as we now progress through the key areas one by one.

STEP SIX: DYNAMICS

The first area I zone in on is the dynamics of the mix as any dynamic processing will affect the tonal balance. Mastering, though, is a juggling act among all processes, so remain flexible and open to the fact that nothing is set in stone until the final export. When assessing dynamics we are talking about the loudness envelopes of musical instruments, alone and collectively, i.e. the way individual performances push, pull and blend, thus contributing to the song envelope as a whole. Listen carefully to the dynamics that already exist in the composition you are mastering. If the mix sounds less produced and less professional than your reference track the answer may well lie in the amount of dynamic control exercised. There are, of course, other influences such as quality of recording environment and gear used to capture sound, along with the ability of the performer; but whether a listener knows it or not, within commercial popular music a certain degree of well-applied compression equals professional perception. Consider the overall dynamic envelope of the complete piece of music, i.e. the loudest and quietest parts and the difference between them. Are the dynamics intentionally creative and therefore in need of preserving for the integrity of the piece, or can the gap safely be closed? Much will depend on the genre of the track in question. Do your research so that you are aware of the listener’s expectations within any given genre.

Use your Top 50 songs to compare the dynamic curves of established and accepted masters throughout the history of recorded music. Focus, too, on the individual instruments that make up the mix. Do they sound overly processed? Have they been compressed and squashed during tracking (the process of recording in a multitrack format) or do they sound clear, clean and pristine as if captured in their full naked glory? These are the signs to look out for and the clues that point to the intention of the creators. Try putting yourself in the shoes of the mix engineer to discover what direction was envisaged. If the mix you are mastering is your creation, pretend that you are polishing an unknown musician’s song. This shift in mental approach can prompt questions about your work that would not normally have been broached. When addressing dynamic content, I always ask myself these three questions:

Has the mix been overprocessed already to the extent that the life is squashed out of it? We are looking here for mixes that are too flat with minimal definition between instruments and a lack of a three-dimensional imaging, possibly with distortion on the loudest sections and dulled transients

Is the opposite true that not enough control has been applied and the music is too live for the genre? I look out for mixes that are unsettling and uncomfortable to listen to as instruments jump about in a distracting manner and lack coherent musical glue

Is the mix weak and thin, lacking in power for the genre? We’re watching here for mixes that do not stack up against references and sound more like a demo than a professional master

Now, consider your answers.

If the answer to all three questions is “no”, consider whether you need any additional dynamics processing at all.

If the answer to any question is “yes”, some form of additional dynamic control is called for.

If you are still unsure, refer to the notes you compiled when first listening. Did you highlight any dynamic issues? Listen again to the reference tracks in your Top 50 with the above questions in mind. Note the strong unified sound of the reference tracks, the way that professional masters pin the levels of instruments back and do not let volume spikes burst forward to distract the listener, but still sound open yet glued together. Close your eyes and imagine the mix as a performance on stage. See how the performers stay in their positions. Is that how your mix sounds - stable and cohesive, or does it feel as if each musician is lunging towards you from the stage? Assuming there are dynamic issues to resolve, let us work through the scenarios one by one. Should you need to refresh your memory on the plugins

used, namely the compressor, expander, noise gate and limiter/loudness maximiser, check back to ‘Chapter 6: Mastering Tools’.

Scenario A: if the answer to question one was “yes”, too much dynamic processing has been detrimental to the mix, then a re-mix could well be the answer. If, however, there is the slightest hint of a pulse to work with, an expander may rescue the mix from the dead, bringing it back to life and restoring dynamics. Expanders are controlled in exactly the same way as compressors, i.e. they have the same set of knobs, the difference is, whereas a compressor turns peaks down above a threshold, an expander turns them up. This is reflected in the ratio settings. The first digit (the input level) will be less than the second digit (the output level) signifying that the signal will be louder at output than input. As an example, a setting of 0.50:1 means that for every 0.5db registered at the input stage 1db will be output (this is a simplified example assuming the complete signal is processed). Without getting too technical, enable an expander and select a mastering preset; pull down the threshold until the gain reduction meter ticks away and listen. Has the mix improved or not? Has some of the life been restored or does it now sound worse? Do not be afraid to push a plugin beyond its limits before reigning it back in. Only by going too far will you know you have pushed too hard. Less is more in mastering generally, and yes you will be applying many small gentle nudges rather than one huge sledgehammer blow, but it does not hurt to know what ‘too much’ sounds like before settling on what works. Do not shy away from using presets either. With expanders, always try a preset first as nine times out of ten it will be good enough. Remember an expander works by increasing dynamic range so it will be turning up any peaks above the threshold, widening the gap between quiet and loud passages or micro dynamics and restoring some liveliness. Check the result by enabling and disabling the expander or export two versions, one with the effect applied and one without and then performing an A/B level matched blind test. If you cannot improve on the mix and the damage has been already been done, suggest a re-mix or revisit your mix if the work is yours and ease off the dynamic processing before starting to master again.

Scenario B: if you answered “yes” to question two then the mix is out of control and distracting. At this stage of production, light compression is the tool that will tame some of those wild peaks (keep in mind that a limiter/maximiser will increase the effect prior to exporting). Within our template we have various types of compressors, all of which are disabled awaiting action. There is a single band compressor, a multiband compressor, parallel compression and a limiter/maximiser (a very fast compressor) - which one should you use? A standard compressor is the go-to tool. It is great for pulling a mix together and adding what many people refer to as ‘glue’, a word that describes the unification of elements. The type of standard compressor designed for mastering is often labelled as a ‘bus compressor’. These are optimised to process groups of instruments and full mixes, although there are no rules with many stock compressors being successfully used by mastering engineers. Let your ears be the judge. Go ahead and enable your compressor of choice and try the following setting as a start point; a 2:1 ratio with the slowest attack and fastest decay. This lets as much of the attack through as possible before the compressor clamps down. If sounds are still jumping out too much, shorten the attack and lengthen the release. Moving the ratio higher to 3:1 or 4:1 will give a tighter sound. As for the alternatives, a multiband compressor is best reserved for localised problems where the lack of control can be tracked down to one area within the frequency spectrum. These compressors are designed to house multiple compression units in one plugin, each of which can be tuned to a specific range of frequencies and function independently. And as mentioned, a limiter/maximiser will be the final stage of production prior to exporting.

Scenario C: if “yes” was the answer to question three then the mix is thin sounding, more demo than pro. Parallel compression might be the perfect cure for this ailment. I have to say ninety nine times out of a hundred parallel compression is my first move and one that I rarely master without. Once experienced, it is an effect that is hard to ignore, so un-mute the parallel compression channel and enable a compressor. Parallel compression works by taking a feed from one channel and routing it to a second. Compression is then applied to the second channel which is then mixed back with the original first channel. The unprocessed signal is referred to as ‘dry’ (our ‘Original Mix’ signal) and the processed signal ‘wet’ (our ‘Par Comp’ signal). Compressors that feature a ‘mix’ control can handle parallel compression within one plugin, the mix function blends the wet audio with the dry, 0 percent being totally dry with no compression and 100 percent being totally wet with the compressed signal only. However, I prefer to separate my parallel compression out to two channels, this way I am totally aware of what is happening rather than have the effect hidden. Parallel compression is similar to a process called ‘upward

compression’ which also increases low level content and is more transparent than standard downward compression, whilst still reducing dynamic range.

A great start point for parallel compression are the following two settings from Bob Katz amazing ‘Mastering Audio: the art and the science’ book, one that I thoroughly recommend if you wish to take the next step into comprehensive detail and technical learning. Bob presents two variations on the parallel compression theme, one transparent and one with attitude. They are as follows:

Transparent Parallel Compression

Threshold: -50dbfs

Attack: 1ms (or less using the fastest setting available)

Ratio: 2:1

Release: 250 – 300ms

Output level: -15 to -5db (to taste)

Although a ratio setting of 2:1 is given, in reality it is the level of the parallel compression channel that determines the amount of dynamic range reduction. The low threshold means the compressor is nearly always compressing so the parallel compression channel contains a highly compressed version of the original with a very narrow dynamic range. By increasing the output level of the parallel compression channel we are introducing controlled low-level signals which reinforce the original channel in a transparent way. Loud passages cause the compressor to really clamp down on the parallel compression channel, reducing the output of this channel when it is needed less. Whereas, during quieter passages when the original source needs a boost, the parallel compressor does not compress as much, causing more level to be added to the original source. This increases the overall combined signal during quiet passages of music. The end result is that quieter sections become louder while loud sections remain pretty much unchanged. The more the output from the parallel compression channel is raised, the more the dynamic range is squashed and the more the effect is audible. As this technique is designed to be transparent and not heard, start with the parallel compression fader low and gradually raise the fader upwards in small amounts whilst switching the compression plugin in and out. The result you are looking for is thicker more solid sound, but not so much that the parallel compression is apparent as a sound in its own right. Less is more, only enough to have impact. If, when switching the compression plugin in and out you cannot hear any change, then the fader is too low; if the audio changes drastically then the fader is too high.

Attitude Parallel Compression (more attitude and punch)

Threshold: 5-7db gain reduction on the medium loudest parts of music

Attack: 125ms

Ratio: 1.5:1 to 3:1

Release: to taste to achieve maximum rhythm and punch

Output Gain: -6db max

Experiment, have some fun and say a big thanks to Bob Katz.

There is a third alternative that is true to my simple quick ideals. I often enable a simple one-knob mastering multi-effects processor on the parallel compression channel and drag the threshold down so that the source is deeply squashed whilst bringing up the channel fader to taste. This gives an additional quick colour to your sound pallet. Try whatever you have at your disposal. Different compressors or multi-effects will result in different flavours of compression, so stack the various musical colouring plugins up and enable them one at a time to hear how each compressor reinforces the mix in a unique way.

One final quick trick that will add a professional sheen is to use a multiband compressor. As mentioned previously, multiband compressors are useful for affecting a certain range of frequencies whilst leaving others untouched. This is perfect for tightening up the bass and treble areas, which alone can result in a smoother, more polished sounding master. It works like this:

Enable a multiband compressor and solo the low end (the module furthest left) dragging the right ‘cut off’ handle left until the passing audio sounds like music coming from inside a building or a car in the distance, the familiar ‘boom, boom, boom’

Then lower the threshold and raise the ratio until -3db gain reduction is showing (attack and release settings should be fine in their default state)

With a -3db gain reduction applied, raise the make-up gain by +3db to restore pre-compression levels

Next, un-solo the low end and instead solo the high end (the unit furthest right). This time drag the left ‘cut off’ handle right so that only the extreme highs are being processed. Try to aim for somewhere above the sibilance of the vocal so that it sounds like a really tinny small speaker or the percussive sounds you can make when hissing through your tongue pressed up against your teeth

Then use the exact same process as for the low end, i.e. -3db reduction and +3db make-up gain

Un-solo the high end and listen to the overall difference by switching the compressor in and out

This suggests a timely reminder that all plugins should be disabled when not in use and only enabled to test a process. If that process is unsuccessful the plugin should be disabled once more to keep a clean overview of active processing.

You may have read the above and think “hold on”, raising the gain of one section of a multiband compressor is the same as equalisation boost, i.e. a band of frequencies is increased. You would be absolutely correct - there is no major difference between the two. You could raise a compressor’s gain higher than is needed to return to pre-compressed levels and this would be the same as adding extra equalisation to the frequency band. As a crude EQ unit this method functions well in certain circumstances and may negate the need for additional equalisation.

As previously mentioned in ‘Chapter 6 – Mastering Tools’, the easiest way to learn compression is to enable a compressor and move the controls to hear how each affects the waveform. Pay careful attention to how the compression affects the first transient burst of sound in the attack; also listen to the decay phase and how the loudest sounds influence the level of the quieter ones when applied to a complete mix. This is one of the processors where the more time you spend using it, the more you will discover and hear the wonder of its powers. Continually switch between your level matched reference track and your master and see if you can decipher the magic that will move your track closer to the professionals.

If this is a learning curve for you do not worry, each and every one of us resides in a constant state of learning. We can never know everything and with more knowledge comes more questions. Work quickly and try to see the bigger picture whilst focusing in on the different parts of the jigsaw puzzle. Enjoy the process of doing, learning and progressing as you practise and improve. Think of every master as practising; always practising. Free yourself from the results. Have no expectations and let go. It will be what it will be. If you apply yourself, you will learn and progress no matter what. You have the back-up ‘algorithmic’ base level master to fall back on. You will produce a master and that is all that matters, outputting work to a consistent standard quickly. Never give up on a project or release sub-standard work that will reflect badly on you.

One final processor that includes a compression effect is ‘tape saturation’. Analogue emulations of desirable outboard gear have become numerous in recent years with the promise of analogue warmth and colour. Although saturation plugins are normally used to add ‘warmth’ (harmonic distortion) to individual tracks in a mix, they do work like a compressor by softening peaks. Personally, I am not a huge fan of tape recreations or any other analogue modelling, especially where mastering is concerned. I would not recommend you use any such device as the colouration is too strong. Mastering should impart as little deviation from the original mix as is needed to satisfy standards. In my book this does not include adding noise and subtle distortion. But never say “never”. If you find the right song with the right flaws, analogue emulations and tape saturation might be the missing element that elevates the track. Also, I have purposely omitted the superfast mastering limiters and maximisers here, which although technically are dynamics processors which work to reduce dynamic range, we will leave these until last to bring the overall level up to match the output target.

You will now have a more lively mix if the dynamics needed resuscitating; a more stable ‘glued together’ musical image if the mix was out of control; or a more powerful, punchier pro-sounding mix if parallel compression was applied. With dynamics in the bag let us turn our attention to the tonal balance.

STEP SEVEN: TONAL BALANCE

Let the mix play and mentally scan the frequency spectrum from top to bottom and bottom to top. It may help to close your eyes so there are no visual distractions. Think of the frequency spectrum in three different parts – the bass, middle and treble. How balanced are they? Maybe there is too much bass energy, too little treble brightness or dominating harsh mids. This is where knowing your frequency spectrum and where each individual instrument sits will help (Chapter 4: Critical Listening). If you are not sure what is out of balance by listening to the mix, switch to the reference. How does this compare? Assume the pro master is right and work towards the same degree of balance. Visualise sound level on a front to back basis. Interpret louder pulses as being more forward and quieter ones set back. Write down your thoughts. Which of bass, mid and treble is farthest forward and which is the most backward? There may be little difference in each if style calls for a tight production and the mix engineer has performed well, we might be talking of fractions. Also listen for any problematic frequencies that are way out of proportion to others, or where chunks of frequency levels are low in comparison causing an unbalanced whole. Once you have exhausted your investigations enable the spectrum analyser plugin. Analyse the mix to see if technology backs up your conclusions or whether it highlights other areas of the frequency spectrum that are too loud or too quiet. Use everything at your disposal; your ears, technology and your knowledge of the genre in question. You have subconsciously built up years of passive listening. Through musical relaxation, without any effort or conscious endeavour, you have burnt into your mind a set of guidelines to work to. Your instinct will tell you if something is wrong. You will feel it in your gut and you will hear it in the questioning voice in your mind. Always listen to your instinct. Ninety-nine times out of a hundred it is right. Your instinct does not lie or make things up; for it to be triggered something is awry. Now analyse your chosen reference track again. How do the two EQ curves differ? Note down the obvious variations between the mix and reference, i.e. the frequencies and degree of difference in decibels. If you have any questions regarding equalisers, what they do or how they work, refer back to the earlier overview in ‘Chapter 6: Mastering Tools’. The idea is to turn down any frequencies that are too loud and turn up any that are too quiet and in doing so sculpt an altogether more balanced frequency curve.

Of course, not every mix is out of balance. You may be sitting in front of your monitors thinking “I cannot fault this mix tonally. Nothing is jumping out at me, nothing sounds wrong, my instinct is enjoying the ride and my head is relaxed and empty, chilled at what it is hearing”. Great, the hardest decision is the decision to do nothing at all. The more experience you gain the more you will come to realise that less is more. Always stay true to the mix. The worse result for a mastering engineer is a master that is in some way a lesser entity than the mix itself. Never break a mix or degrade it in any way because you felt “you had to do something”.

Assuming there are tonal discrepancies how do you go about correcting them? There are four tools within our template that will help; a three-band equaliser, a parametric equaliser, dynamic equalisation and EQ matching. I feel it is right to attempt to EQ using ears first rather than jump straight to a spectrum analyser or EQ matching software, which can prevent progression if relied on too heavily. As humans we often take the easiest route to our destination, which is not always in our best long-term interests. Let us look at our options one at a time and delve into their workings through the following scenarios.

Scenario A: the three tonal areas of bass, mid or treble are out of balance in a broad sense. Then the three band equaliser, the simplest of all options may be all that is needed. Think of a three band EQ as similar to the bass and treble dials on a consumer hi-fi, with an added mid control. Imagine yourself playing the music through a hi-fi unit. Is there any aspect that you would correct using these basic controls? Better still, output a rough master and play it over your home hi-fi outside of the studio. Are any of the three bands too strong or too weak? If so, cutting is always better than boosting, for example, rather than boosting bass and treble, cutting the mid section will have the same result. A small amount will have a big impact. Always err on the side of caution and shoot for the middle ground. Compare your master to your reference in the Top 50 songs and even if no remedial equalisation is called for, enable a three band EQ anyway and practice for a few minutes. Hear what ‘too much’ sounds like by giving the bass a healthy boost and take a deep cut from the mids to hear what a lack of mid frequencies sounds like. All this helps to build pictures in your mind’s eye for when you encounter such a scenario.

Scenario B: there is an out of balance frequency or narrow band of frequencies that can be pinpointed within the frequency range, one that is too loud and harmful to the master. The parametric equaliser is your friend here. With your EQ enabled, use the knowledge gleaned from your prep work. Try a few boosts and cuts in the appropriate regions whilst switching the EQ in and out to check progress. Also switch between the reference master and your work from time to time, both of which should stay approximately level matched. This is difficult to sustain whilst in the moment of intuitive boosts and cuts, but bear in mind that louder sounds better so try to roughly keep the overall RMS level around -14db by keeping a K or RMS Meter visible. You may have heard the phrase ‘find the pain, reduce the gain’. This is generally accepted as the best way to find and rebalance loud peaks, especially when honing in on a problem. To do this apply a +10db boost with a narrow Q and sweep up and down the problem area until the loud frequency becomes louder still. When the issue is at its most painful you have found the source of pain. Then simply widen the Q setting to ensure all of the painful frequencies are affected and none are escaping. Then reset the gain back to 0db and let your ears settle for a moment. Once the ear jarring has subsided, reduce the painful frequency by 0.5db to start and listen to the effect by disabling and re-enabling the equaliser. Remember you are affecting the whole ensemble of sounds. If more reduction is needed try small half-decibel steps one at a time. As always with equalising, cut when you can rather than boost. There may be situations where a boost is necessary. Do not overthink the situation, and accept the boost as the way to go.

An interesting note is that it is much harder to hear when frequencies are too quiet as opposed to too loud, but you can perform the opposite to the above and apply a -10db cut and sweep away. If your ears pick up on lacking areas then a parametric boost is the answer. Boosting often requires a wider Q setting and a more gentle approach than cutting, which can be more surgical with a narrower Q and a deeper notch. It may help to think in descriptive terms such as boomy, tinny, harsh, honky and nasally until you get a hold of the exact frequency range responses. As with every change you make, without exception, switch the plugin in and out to check you have improved the mix and compare it to your level matched reference track to ensure you are heading in the right direction.

An alternative way to identify an errant frequency is to hum or sing the exact note and then find the pitch on a piano or guitar. It is then easy to translate the musical note into hertz using a conversion chart. This works well within the frequency range accessible to the human voice and is amazingly accurate, taking you straight to the heart of the problem.

Scenario C: a certain musical note is too loud whenever it appears. Scenarios A and B relate to static ‘set and leave equalisers’ that are always cutting and/or boosting by a specific amount. But a permanent gain adjustment may not always be the best solution. There may well be occasions when a problem only occurs occasionally during the timeline of a piece of music, when a certain note is played on one instrument perhaps. If a parametric equaliser is used then the frequency in question will be turned down permanently for the whole of the track. This would solve the issue but would also have a negative effect for all sounds all of the time. The answer then is a ‘dynamic equaliser’ which only alters frequency levels when they get too loud, leaving quieter frequency levels to play as they should. ‘Dynamic equalisers’ work in a similar way to compression in that they have a threshold, gain and Q control. In fact you can use a multiband compressor to perform a similar function. The key is setting the threshold at just the right level so that acceptable frequency levels pass through unaltered until the problem occurs when sound passes through the threshold and is reduced.

That covers three of the main scenarios you will face on a regular basis. Let us now look at other ways of controlling and manipulating tonal balance.

High and low pass filters: Make a habit of enabling a high pass and a low pass filter on every master. Set the high pass filter to around 30Hz and the low pass filter to 20kHz. These settings top and tail the audio to optimise the bits available to human hearing, keeping inaudible energy out of your master.

As stated previously, it is good practice to try to solve problems with your ears and experience first, but having given it your best shot without assistance there are two tools which may help further, they are a spectrum analyser and matching EQ.

Spectrum Analyser: Firstly disable any active EQ plugins for now and enable a spectrum analyser. Watch the EQ curve dance about and get a feel for the song’s response in this ‘dry’ state, taking a snapshot if you can. Secondly, re-enable all the EQ plugins and note the differences. Again, it will be easier if you have the facility to take a snapshot. Now compare the two ‘dry’ and ‘wet’ EQ curves. Did your equalisation decisions flatten the EQ curve or have the opposite effect? When looking at the ‘dry’ curve with this new visual information, are there any new frequency areas that you missed that would benefit from attention? If so, try altering theses ranges by an amount that will flatten the curve somewhat and make it more equal. Use gentle corrections first and apply more to taste until the changes become too audible or have a negative impact. Has the direction the analyser taken you improved the overall sound further? Compare the results in a level matched way to your reference tune. Take a snapshot of your reference EQ curve – what can you learn, how does it differ? Write down your thoughts. If yes, stick with the results; if no, undo and revert back.

We are so lucky to have a colourful and diverse history of unbelievable songs to call on. Songs that span generations fill your musical filing cabinet, fifty of which are now in your own personal reference library. Playing a few songs that fall within the same genre as your mix should give an idea of the general recognised EQ shape. Analyse many and assume these professional masters are correct. Do not rely on the ingrained EQ curve in your memory banks. It took me a long while to realise that my perception of a good EQ shape for my genre was actually wrong. Being predominantly a singer/songwriter and guitarist, I would accentuate the frequencies around the vocal area and produce mid dominant masters without the subtleties that a complete full frequency professional master demands. Song after song I would work to the same sound I had in my head not realising that it was out of line with listeners’ expectations. For a long time I did not compare my work to references. My masters sounded great to me at the time, but now when I compare them to my Top 50 it is evident they were not. Putting them up against references really highlighted that fact and showed with clarity that all my masters had followed the same wrong shape that I had in my mind. I was living in ignorance until I developed the methods contained in this book. I now know the tricks which match up to professional standards every time. I have rebooted and reprogrammed my mind replacing what I thought was right with what I now know to be correct. Be aware that you may have to go through this same process of discovery and resetting your mind.

Now, having used your ear and the spectrum analyser to realise your best equalised master, output a rough master and call it ‘Manual EQ’. With the master exported, disable all equalisation plugins so the template is back to the pre EQ state prior to tonal balancing. We are about to see how an ‘EQ matching’ plugin compares and we do not want any colouration added to the base mix.

Matching EQ: As we learned earlier in ‘Chapter 6: Mastering Tools’ EQ Matching software analyses the tonal make-up of a mix and reference track, applying the reference track’s frequency curve to a mix. So go ahead and enable your matching plugin, if you have one, snapshot your reference then your mix and have some fun. The power of this type of plugin lies not only in the sampling of tonal characteristics but in the control you have over how much tonal balance change is applied. Two of the most common controls are the amount of change in percent and a smoothing slider to even out peaks and dips. Varying the amount and the smoothness of the curve can give an extraordinary wide pallet of results. The majority of the time you will want to impart a subtle, positive correction without extreme settings or an unnatural outcome, but how hard you have to push to reach the optimum point will vary each time. Close your eyes if it helps to focus your ears. Start by moving the ‘amount’ control until you hear a tonal shift towards your envisaged master. The changes will generally result in less mud and brighter results through compensatory boosts and cuts in each of the bass, treble and mid areas depending on how the mix is balanced. The less correction needed, the better the original mix. Once you have settled on a favourable amount, turn your attention to the ‘smoothing’ control. Again, close your eyes and let your ears guide you.

When you are happy with the results turn the plugin on and off so that you can hear the difference EQ matching has made. Are you happy with the result? If not, try again making small changes and checking the results each time by disabling and enabling the effect. Having settled on a final match that sounds good to you, export a rough master and call it ‘Matched EQ’. This will now be compared to the ‘Manual EQ’ file you output earlier following your manual equalisation. Now save and close your mastering template, open a new blank session and insert both your ‘Manual EQ’ and ‘Matched EQ’ temporary masters to compare. Level match each as before, using your ‘dynamic range plugin’ to establish a -14db playback level and perform a blind test to see which you prefer. You will know by now the importance of level-matched listening, but also when performing an A/B comparison it is vitally important not to know which master is which because preconceived ideals will influence your decision. I have given a nod to HOFA previously for their wonderful free ‘4U + Blind Test’ plugin. If you have not already done so, download the free version in which you can compare up to three tracks. Once you have decided which to go with, reopen your saved session and enable either the EQ plugins or your matching plugin, depending on which won the ‘battle of the blind test’!

As mentioned previously, you may think that using EQ Matching is cheating and does not progress your development. I would say the opposite is true. We use technology in all aspects of music production and this is no different. EQ Matching is one of many tools to empower you to reach a goal quickly and efficiently. I want to give you the fastest repeatable system with the highest quality results. The learning that can be gleaned from regular use of this plugin will shape your future decisions to a point where you may no longer need it. Do not knock EQ Matching until you have tried it; this is an incredible learning tool.

Finally, I include one more useful trick that I use alot. It concerns using a multiband compressor as a quick double-check to see that all the useful ranges of a master are balanced. Simply enable a multiband compressor as your last plugin and choose a mastering preset. Each band should tick away gently in the same proportion to each other. You may need to reduce the threshold a touch to see some activity, so make sure you reduce all the thresholds equally as one. There is usually a master control to move all bands simultaneously. Once one band starts to register a reduction all bands should respond also. If one band is banging away on its compressor unit and the others are not moving then it is an indication that your master is out of balance and the band in question is too loud (taking creative preferences within genres into consideration).

That concludes the tonal balancing. Hopefully you now have a more pleasing master that is another step closer to professional standards through equalisation. Either way, accept your results, do not try to re-work, re-visit and dwell on every tiny detail. To go over and over again putting all your efforts into making one particular master perfect is fruitless. Perfection is a moving target that can never be realised. Instead, be happy and positive that you have learned new processes and skipped across another stepping stone on this long journey of continual improvement – you are now firmly on your way. There will always be improvements to be made, as even the best are continually learning. World champions of all sports employ coaches to add another dimension and perspective to their game, forever probing. Look to up your game too with each and every master. A small improvement on one master may not seem revolutionary, but tiny improvements on each of seven masters starts adding up to a much bigger step. And if you can manage seven masters in a week, suddenly in a month all those tiny stuttering moves forward add up to one giant leap. So if you have not gotten as close as you would like, do not worry, leave it and move on to the next step regardless. We have another trick up our sleeves later on when we match the ‘algorithmic master’. I cannot overstate the benefits of working quickly and finishing masters fast. Thinking negatively or overthinking certain issues whilst spending too long on one area will only drag you down and cause you to lose perspective. As long as you have done your best with the tools you have, that is all you can ask of yourself. Now, leaving tonal balance behind, move on to the next step and add some lovely stereo imagery.

STEP EIGHT: PANORAMA AND STEREO IMAGERY

Press play once more on your dynamically-controlled, tonally-balanced master and allow your mind to focus on the panoramic soundscape. Switch back and forth a few times between your master and reference. Make a note of your thoughts. Is the pro master generally wider or narrower? Mentally scan each of the low, mid and treble ranges. Look out for width in the mids and treble but a narrow focused central power in the low bass area. This is common and although is to a large degree mix dependent, it is possible to re-shape individual ranges separately with a stereo imager plugin. Pay careful attention to the bass area, has it been artificially widened? We are used to hearing a spread in the mid and treble areas but usually bass instruments are panned centrally and offer little width. Import your master in the Top 50 session and compare it to the references. Do your initial thoughts still hold true when averaged over fifty tracks? These references hold the clues to look out for and point to corrections that may improve your track.

As we learned earlier in ‘Mid-Side Listening’, panorama only exists in stereo playback because of the differences between signals coming from two speakers. This difference of sonic content creates the panoramic left to right sound canvas. Take away all time, level and sonic variation and sound sources will revert to mono - there is no panorama in mono. Therefore stereo equals difference; and it is these dissimilarities that we can manipulate to make the panorama wider or more narrow. A frequent observation of mastered tracks is that they are wider than the basic mix. The stereo imager plugin may well be the answer. A mix can certainly be widened to appear as if it comes from outside the confines of the humble monitor speaker, but remember a mix can be made narrower too or left alone without processing at all. The decision is yours and is purely subjective. The best way to fully appreciate a plugin is to use it, so enable your stereo imager and go with your instincts using the references to guide you. Tread carefully with this effect or the results may be distracting at the least, or even lead to a total breakdown in connection between instruments that the mix engineer has laboured hard to achieve. A powerful cohesive ‘monocentric’ mix, or in fact any mix, can lose its impact if spread too wide. Alternatively, narrowing the panorama can re-inject solidity and punch back into an over ambitious mix that is too ‘spacey’ and weak. But do not feel you have to change anything at all if the mix already sounds spot on. Listen and listen some more. Act with clear intent or leave be. A stereo imager tool uses mid/side processing and psychoacoustic techniques that are beyond the scope of this book. For our purposes it is enough to know the following:

Raising the level of side content in relation to the mid will give a more stereophonic experience.

Decreasing the side content level in relation to the mid will have the effect of narrowing the stereo sound field and making a master more mono.

These are the most fundamental examples of mid/side manipulation which are available in even the most basic of stereo imagers and can give pleasing results. They could be combined in one width control or similar. Go ahead and rebalance the mid and side content. Make your master more stereo and then mono. Although you want subtle settings in your final version, pushing to extremes when evaluating can help you to hear the degree of change that is possible. However, merely adjusting the balance between mid and side content will not push the audio into the ‘super wide’ zone. If you wish to push the music outside the monitors you will need a plugin manufactured specifically for the task of stereo widening. Be aware that tricks such as stereo widening can put a smile on the faces of engineers, but as with so many gimmicks the gratification is short-lived. You soon realise that it was probably better without the meddling once the magic dust settles, and indeed the results can distract and fatigue the listener. As always, think carefully and compare your master with the masters of your peers and then make small, considered moves, checking your progress constantly.

If a ‘shuffler’ function is included you have the option to widen bass frequencies. The human listening system perceives bass frequencies as being narrower than high frequencies. A ‘shuffler’ re-addresses this disparity. Side-affects induced by this process will vary between plugins with the biggest issue being unwanted phase errors between the stereo channels. The best plugins provide phase compensated playback to limit this problem.

A good starting point, assuming you have all the functions available, is a touch of widening and a trace of shuffling to give extra width to the lows; and then nudge up the dials in small increments whilst checking results by switching the effect in and out. Test the extremes by setting the controls up to their top value and experiment with different bands if available. Absorb how the bass frequencies respond differently than the treble; close your eyes and visualise the sounds in the stereo image; then move and manipulate them. One final point to note is that all stereo imagery effects disappear when summed to mono. If mono playback is important to you check your result by clicking the mono button. Here are some common scenarios that may help:

Scenario 1: the mix is not as wide as the reference. The answer is to increase the level of side content by upping the width control of a stereo imager or increasing the level of the side content using a mid/side processor.

Scenario 2: the mix is not as narrow as the reference. The answer is to reduce the level of side content by lowering the width control of a stereo imager or decreasing the level of the side content using a mid/side processor.

Scenario 3: one specific range of frequencies is wider on the reference master. Use a stereo imager’s multi band function to only affect the certain frequencies. If the range in question is the bass area then apply shuffling.

Scenario 4: one side of the stereo field is louder than the other. Look for an ‘asymmetry’ function that alters the relative level of the left and right side elements whilst keeping the mid soundstage central.

For further exploration into the effects of mid/side content ratios I would recommend that you spend some time listening to your Top 50 masters and assess how the mid and side content has been balanced in each. To do this you will need a VU Meter that measures mid and side levels. Place the VU Meter on the master bus and check that your Top 50 songs are hitting around 0db on the mid VU Meter needle. Remember we have already level matched our Top 50 to all play back at a K-14 dynamic level, so if your songs are not hitting 0db then open your VU meter settings and set a -14db offset.

As you play back each song, make a note of the relationship between mid and side content. You will soon pick up on the patterns that exist. For me, a good average balance is when the side content registers around -10db to -7db when the mid content is hitting 0db to +3db on the loudest parts of a song. I constantly see this mid/side relationship on songs that appeal to me. In my opinion, this relationship gives the most pleasing balance between mono power and stereo lushness. Any deviation towards a lower level of side content I perceive as being too mono and an increase of side level I feel is a step too far towards being ‘lost in space’. You may have different values. What are your preferences? It is important that you know what end goal you are working to. Once you have a clear reference point you can align musical content to this. Musicality always wins. Work towards finding the answers to the questions that the music is asking. Move with the vibe and do not try to impose a different feel that fights what already exists. If the mix is power orientated and crying out for a master that leans towards mono, it would be unwise to crank up the side dial and move it towards stereo heaven and vice versa. Some music needs a more powerful cohesive mono feel, some music needs space. Try and internalise the difference.

With this in mind, insert a stereo widener on your Top 50 reference channel and alter the state of the professional masters. See what impact changing the mid/side balance makes. Move the levels up and down to hear what your favourite songs would have sounded like if the balance had been different. Go complete mono. Go to the edge of spacey stereo then back to normality and every degree between. Experimenting in this way will help solidify your mind’s eye image for future masters. But just to reiterate, when working for real, do not feel you have to impart any stereo widening at all; and if you do consider a little magic dust will aid travel, evaluate very carefully the processed and unprocessed versions to make sure you have improved your master.

STEP NINE: EFFECTS

In this step we will consider effects before moving on to the final stage of loudness maximisation. As before, press play and switch between master and reference with effects in the forefront of your mind. Jump around your Top 50 songs for an averaged overview. Initially listen to all effects, how they have been applied and where they sit in the mix; but for the bulk of this step I want to focus on the two main effects used in mastering, first reverberation and then delay.

Reverberation: With that in mind, how dry or reverberant is your master when compared to the reference track? Is there an upfront ‘in your face’ feel or a three dimensional quality? Your goal is to assess whether ample effects have been used already or is something missing? Is there more room for spatial spice? If the answer is yes, the solution is to add to taste until you are happy with your musical delicacy!

There are two main ways to use reverberation when mastering, A) enhance the existing room sound or B) add ambience where there is none. The reason we have left reverb until now is because any compression would have raised its level. As we know, compression works by lowering the loudest peaks so that the music as a whole (including low level reverberation) can be raised. This means, if you have applied compression, the perceived reverberation level will have changed since auditioning the basic mix and you could now be satisfied with the balance. If you still feel additional reverb is required here are a few scenarios to help. Note that the reverb plugin has a mix control to vary the balance between wet and dry audio. Set the mix to 100 percent wet if your plugin is on a separate channel from the main signal and use the two channel faders to blend. If all of your audio passes through the reverb unit, as is the case in our template (my mastering reverb module is contained within a mastering multi-effects plugin), the mix control is used to vary the wet to dry ratio.

Scenario 1: the mix is too dry for the musical genre. Additional reverberation is required. Having decided to add some room noise you have two options, match the existing sound and enhance what exists or go for a totally different sound that is sympathetic to the music. Try both to find your preference. Enable your mastering reverberation plugin and step through the presets one by one until you find one that fits your vision. Then blend only enough to fulfil the task and satisfy the ear. Check your decision by switching the effect in and out. Adding too much reverb can quickly muddy and blur a clean mix so pay particular attention to the low end and reduce low frequencies within the reverb plugin if problematic.

Scenario 2: The reverberation tail has been cut short or no reverberation exists. Throughout music production there are many tasks to occupy the mind. Inevitably an occasional mistake will filter through such as cutting a reverberation tail short or deleting silence that was not thought to contain anything worth keeping. This generally happens when a mix engineer is tidying up clips of recorded sound. Sometimes important information is lost to the digital equivalent of the cutting room floor. Every room contains noise, all of which helps to add realism and authenticity to a recording if kept at a low enough level. Even though you may be sat in silence, the truth is you are surrounded by noise. Try recording the sound of a room and analyse the sound wave. You will hear buzzing of electrical appliances, whirling of distant fans and extraneous activity bleeding through walls. Silence in the natural world and total silence on a digital recording through the deletion of all content are very different; one is natural, the other unnatural and disconcerting. Check any moments where instruments fade to silence to ensure that the reverb tail is complete, consistent and natural sounding. Reverberation that comes to an abrupt halt or where reverberation/ambient noise is missing will distract the audience. The answer is to replace the missing reverberation. In this scenario it is vital to match the sound of the original reverb or the result will be as unsettling as before. Being able to distinguish between the standard reverb sounds of hall, room and plate will help, but it is usually as simple as enabling your mastering reverb plugin and finding a suitable preset by trial and error. You will know you have succeeded when the two different reverb sounds are present, but you as the engineer cannot differentiate between them. The only complexity is that you only want the new reverb to sound during one or two brief moments when filling gaps, not all the time. The easiest solution is to either automate the on/off state of the plugin so it comes alive when needed. If you prefer not to use automation, create a new channel and route the mix through this channel, whilst inserting a new instance of reverb. Then, when you have the sound to your liking, export the channel to an audio file and import it back into the session to a new track. Delete the unwanted temporary reverb channel and you will now have the new reverb for the complete song and can simply cut out the sections you do not want.

The very start of the song is an interesting point to note. For most genres mixing engineers want total silence before the song starts. But occasionally the preceding emptiness can cause the song to feel as if it starts too abruptly; it just does not sound right. The answer is to add a moment of ambience before the song begins to help set up the music and draw the listener in to the illusion that a real event has been captured. Of course reverberation is caused by sound waves bouncing around an environment after a source is plucked or hit, so a little ‘out of the box’ thinking is required to flip the process and have the reverb sounding before any music. There are a couple of easy ways to achieve this with a modern DAW and a few simple editing moves. Firstly you could get creative and add some ‘reverse reverb’ to the start. Here is how.

Reverse the mix audio in processing settings; the start of the song will now be the end and all audio will be heard backwards

Play the reversed version through a reverb plugin and record (export, bounce or render) the output of the reverb channel

Now return the mix to the correct playing direction and import the ‘reverse reverb’ wave file. The section you want to use is at the end of the imported sound file, so reversing the clip will move the end to the start! Delete all but the very first section prior to the mix start

The result is a gorgeous, slow building, lush reverb tail that stops exactly as the song starts. You have created an imposing effect that is not always appropriate to use, but shortening the clip to a few milliseconds could be the answer. If you do not like the reverse reverb sound, try moving the end of the clip to the start without reversing, shorten to taste so that only noise is heard and add a short fade in. Other solutions to try include literally recording the sound of the room you are in and placing a short edit of the sound file at the start, you may get lucky and the sound may suit. Also, try sampling the hiss and crackle at the start of an old analogue recording for a retro vibe. Adding this to the start might be the perfect texture to engage the ear. Moving on to the delay effect:

Delay: In most circumstances, delay should be used sparingly if at all in mastering. Time delay effects such as flangers, phasers and chorus, or indeed any other high impact effect that distorts or modulates the audio is not a good idea at the mastering stage and would change the feel of the mix considerably. These should be avoided unless a client has asked for a specific effect or the music is your own work and you envisage adding a new idea. The only delay effect that is generally used is a 30 milliseconds or less, stereo delay. The science behind the magic 30ms threshold is that when hearing two identical sounds separated by a time delay, a human interprets the results differently depending on the amount of delay. When one sound is followed by a second identical sound 30ms or less after the original, the human brain groups the two sounds together to form one thicker sound. However, when one sound is followed by a second identical sound hitting the ear after 30ms, the brain interprets as two separate instances of the sound.

So, a delay time of less than 30ms thickens and enhances the source material without causing the sensation of hearing two separate sounds. It is this thickening effect that you may wish to try. The effect is achieved by creating a new channel and inserting a stereo delay plugin. Then send a feed from the original mix to the stereo delay channel and blend the 100 percent wet signal with the original signal by using the channel faders. Experiment with the delay times. Any setting between 7 and 30ms is best. The effect should add not only a thickening effect but also a subtle spaciousness. The one important point to note is that a different delay time should be set for the left and right channel. For example, try 15ms on the left channel and 25ms on the right channel. This will mean that the stereo audio will pull slightly to the left channel as the left side is heard before the right; the answer is to raise the level of the right channel (or lower the level of the left channel) within the stereo delay plugin until the audio appears balanced centrally again. The more wet effect you blend in with the dry source material the more spacious the result will be.

This effect also works well when used in tandem with a small amount of a long reverberation, a preset such as a canyon or a cathedral. Place the two effects in series and change the order to see which you prefer. The key here is to feel the effect rather than hear it. Push the effect level up as far as you think works and then mute the effects channels and re-enable them. Bringing the effect back in should result in a feeling of additional spaciousness and thickening and not a noticeable addition of sound and change of tonal character.

STEP TEN: LOUDNESS MAXIMISATION

All that remains is to utilise your loudness maximiser (aka mastering limiter) to set the final output level and dynamic range. Jump back to ‘Chapter 6: Mastering Tools’ if you need to refresh your memory regarding the specific controls. As mentioned previously, the purpose of a maximiser is to reduce the loudest peaks so the remaining sound wave can be increased in level, ideally without too much signal degradation. Some genres like hip-hop, pop and rock are more forgiving to the side-effects of loudness maximisation; other genres such as jazz, acoustic and classical music, not so. If you are working in these genres you may not choose to add any loudness maximisation at all. Depending on your plugin of choice, your maximiser might be a simple one knob/slider processor or a complex beast that requires a complete manual dissection to work out what is going on. Either way the end goal is the same – to reduce dynamic range and increase the overall level. In this section we will look at how to use a maximiser and what to listen out for. Then in the next chapter ‘Exporting’ we will discuss specific target numbers of streaming sites.

Let us look at the maximiser in practice. For the purposes of a fast and efficient system the simplest loudness maximiser will suffice with the bare minimum controls. A threshold, ceiling control and automatic make-up gain will do. By all means dig deeper into further controls and settings as you progress, but for now keeping matters simple is a good way to focus your mind on how your maximiser is changing the sound. To get the cleanest results use the following method:

Enable the loudness maximiser and start to lower the threshold whilst playing back the loudest portion of the song

Continue lowering the threshold whilst watching the ‘Attenuation Meter’. Stop when a gain reduction of -1 to -2db is reached

If make-up gain is not automatic, raise the make-up control to restore lost level

If the audio registers 0dbfs on the master bus on the loudest peaks then lower the ceiling control to -0.3db

Set the dithering output to suit the end playback medium, i.e. 16 bit for a compact disc or the preferred resolution for the intended online distribution service (CDBaby, TuneCore etc)

If you only have one loudness maximiser to call on then the choice is easy, but if you are lucky enough to have two, three or more which do you use? Every brand of mastering limiter/maximiser affects audio differently; therefore each will produce slightly varying results. Assuming you have multiple maximisers, your task is to become familiar with each one; to know exactly how each will affect your master and thus give you the knowledge to pick the right tool for the job. To hear and evaluate the sound of a maximiser will require pushing the boundaries beyond what is normally acceptable. Only then will you hear with clarity its true essence. So slowly pull the threshold down and induce some serious limiting whilst listening across the complete frequency spectrum and all instruments. Listen for harshness, a gritty sound, distortion and for a changing balance between sounds. Move the threshold further and listen more. Write down or memorise your results. Which produced the smoothest transparent results? Which imparts a vintage sound? Which makes the vocal more present? Which kills the snare? Which brings the guitars up in level or reduces bass energy? These are a few examples to observe, and you may find more. It will help to load all your maximisers onto the same channel with each one following the next, then disable all of them and switch one on at a time comparing and assessing the results. Having the knowledge that one brand of maximiser brings the vocal forward while another is totally transparent means you are equipped to handle each situation, for example, if the vocal of the master you are working on is buried slightly, reach for the maximiser that enhances the human voice. Conducting this exercise is a valuable listening workout and will give you a deeper insight to your maximisers, but do not feel you need more than one. If you only own a single processor stick with it and get the best out of it.

You have now experienced both ends of the loudness maximisation scale, from a gentle brushstroke of clipping colour (-1db to -2db of gain reduction on the loudest peaks) to a heavy fisted slam on the defenceless waveform, squeezing it to within an inch of its life (driving the threshold down lower than any sane person would go). But these two examples represent the outer limits of what is possible. Learn to hear degrees of change in between. Also, judging by listening alone is great, but you also must control the specific loudness number of the master for the intended use. For example, you or the client may want to output the song to a dynamic range setting of DR8. Therefore, it is important to be able to accurately measure an audio master to an exact loudness/dynamic range if you are to meet professional standards. The answer is to activate your loudness meter and start monitoring how the loudness rating changes as you apply more limiting in readiness for the next chapter.

One final thought on the subject of limiting, clipping and loudness maximisation; it is worth noting that people differ in their sensitivity to digital distortion and clipping. You may be surprised how much clipping you can get away with before the listener asks “is that distorted”? In fact, the average listener is not aware of the concept of clipping and distortion. Even at the height of the loudness war when degradation peaked, the main voices of discontent were that of audiophiles, not the bulk of the buying public. I am not suggesting you use excessive limiting; use only what you need to get the desired results, but you may be surprised how much you can, and need to, push. You are now fully equipped to output your master!

9

EXPORTING

The act of exporting a master (also known as rendering) can be as simple as selecting the export function from a menu and with one swift click of the mouse the deed is done. However, as with most tasks in audio production, the devil is in the detail. Consideration should be given to a variety of settings in order to produce the correct master. I will run through the most common options before moving onto output levels.

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START AND END POINTS

Firstly, the computer will need to know what section of audio you wish to export. This is achieved by specifying a start and end point. DAWs handle this by the use of markers that are user-definable moments in time. There could be two specific markers called ‘start’ and ‘end’ or your DAW may use a ‘time selection’ function; consult your DAW manual for the exact terminology. When setting start and end points, it is important to include enough silence before and after the music so as to give the listener the best possible experience. If your start marker is too near the first beat of sound they will be surprised, caught off-guard and distracted from the first few seconds of the intro; not the best first impression you want to create. Listen to examples of commercial songs and set a comfortable length of silence to match.

Remember our ‘timeline’ channel? I use this for the very purpose of reproducing start and end points exactly and suggest you do the same. With my DAW of choice Reaper, the start and end points are created by specifying a ‘time selection’. The trouble is; ‘time selection’ is used for many other functions also. I found that often I fine tuned my start and end selection and then used the ‘time selection’ for another process, which meant having to set the start and end points all over again! The solution is to create empty items along the ‘timeline’ channel that are the exact length you wish to export. Then, whenever you have lost your original start and end points, you can simply click on the empty item and use the function ‘time selection to item’ (or start and end points) and the export parameters are instantly restored.

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SOURCE

Modern DAWs are capable of rendering audio from a variety of sources or points in the audio chain. This might be one individual track, a selection of tracks (also known as ‘stems’) or a complete master bus mix. To complicate matters further, sources can include or exclude effects. For our mastering purposes we only need one source and that is to export a stereo file exactly as it is heard through the master bus. Therefore, select the ‘master output’ or ‘master mix’ output from within the boundaries set above. I say output the master exactly as heard, but one caveat applies: if you have inserted any monitoring plugins on the master bus then these should be removed or disabled first.

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DESTINATION DIRECTORY AND FILE PROTOCOL

Regarding file naming and location for your progressive and final master, always export into the main project folder and find a system of labelling that you are comfortable with that you will stick with forevermore. That way in ten years’ time you can open an archived project and know exactly where every file is from a quick scan. Personally, I like to create a separate folder for each different type of media within the main project folder. I have one folder for images, one for text documents and one for audio. When naming an audio master, my convention is to label each work-in-progress ‘NAME X mst Y’ where ‘NAME X’ equals the project title and ‘Y’ is the current version. Below is an example:

The first version of a song called ‘Destination Protocol’ would be labelled ‘Destination Protocol mst v1’.

If the master is then revised an alphabetic suffix is applied, i.e. ‘Destination Protocol mst 1a’.

If the work-in-progress master is scrapped and a completely new version started from scratch, then the end number is increased by one and any suffix dropped, i.e. ‘Destination Protocol mst v2’.

It is also vital that each project’s session carries the same name as the exported master to inform of its source. This way you will never be unsure of where you output the master from. Make renaming each new version of a work your first task. And finally, label your final master in a unique way. This ensures there will be no mistake uploading the wrong version! I like to label my final master with only capital letters followed by a lowercase ‘mst’ suffix. In our example above this would result in the file name ‘DESTINATION PROTOCOL mst’. If the master is exported at a higher resolution than 16 bits for HD Audio, then I add a ‘HD’ to the title to give ‘DESTINATION PROTOCOL HD mst’. This may seem overly complicated and unnecessary, but trust me, it saves many hours of stress and head scratching when you have to revisit an archived master and you are faced with a long list of undocumented project sessions and stereo audio files.

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BIT DEPTH AND SAMPLE RATE

There will always be the option to set a sample rate and bit depth for the exported file. I have grouped these two settings together as they are nearly always mentioned in the same breath, but in your DAW they will appear separately. For many years the only useful option for independent musicians was 16-bit 44.1kHz audio. This combination was established by the CD format invented in 1979 and has been the dominant digital consumer resolution ever since. However, with the advent of high definition audio streaming, more and more listeners are becoming aware that a higher quality of audio exists and is accessible. It would therefore be wise to output both a standard 16-bit 44.1kHz version and a high resolution version. In fact the advice for decades within music production circles has been to output a HD master to future-proof your work. But what high resolution is best? After all the term ‘high resolution’ applies to anything above the base 16-bit 44.1kHz consumer CD grade. For a long time 24-bit 48kHz would have been considered high-definition audio, a standard within most independent musicians’ reach. Today’s interpretation of HD audio would probably be 24-bit 96kHz. If you have ever recorded and mixed music at 24-bit 96kHz resolution you will know that the increase in file sizes and processing power required is considerable. Many musicians have shied away from working this way because of the extra demands. All I would say is master at the highest resolution that works for you and your system and output a HD master at your chosen rate calling it a HD version, whether that be 48kHz, 96kHz or 192kHz. If you are happiest working at 16bit 44.1kHz throughout do not worry, a fantastic song in this standard definition will always outshine a mediocre song in high definition. Indeed many listeners, including music industry personnel, cannot tell the difference between resolutions when blind tested.

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CHANNELS

The choice of channels to output will include one mono channel, standard two channel stereo, two channel stereo where each channel is output to a separate file and multi-channel formats. The stereo setting is the go-to choice for our purposes.

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RENDER SPEED

This is an interesting one. The render speed is the rate at which your computer processes the export of the master. There has been much discussion over whether faster render speeds affect the quality of audio. The options are to process in real-time or to render at the fastest setting. There may have been an issue with older systems, but I can assure you there is no need to render masters in real-time anymore. That is unless one of your plugins disagrees and causes glitches in the audio. Some plugins are more sensitive than others and results will be unique to each setup. If you do come up against a problem, the answer is to play safe, accept it, export in real-time and go make a coffee or check your emails while the machine whirls away. If you have a curious mind and want to check for yourself that both render rates are identical you can use the ‘null test’ diagnostic tool described previously to prove this. To do this, export one version in real-time and one version at the fastest render rate. Then import both files into a new empty project, inverting the polarity on one version and, when played, the result should be silence.

DITHER AND NOISE SHAPING

The option to apply ‘dither’ and ‘noise shaping’ will often appear as a render setting within the export menu. In practice you would only use this to create a master that is at a lower bit rate than your mix, e.g. for a CD. If you dither, my advice would be to use the ‘dithering’ function within your final mastering plugin if possible. This should be of superior quality and give better results. But again, if you are in any doubt, export different versions using each alternative dithering unit and compare. Which do you prefer? Dithering should always be the very last process.

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METADATA

Metadata can be defined as “data that provides information about other data”. Within the world of music production metadata is a chance to log additional information regarding the song and the artist. This is an area that is becoming increasingly important. Within the lightening-fast global infrastructure where we all connect, you could upload a song today and hours later that same file could be downloaded and shared in multiple locations around the world. As the artist who created the masterpiece, or the mastering engineer representing that work, it is important to know that name and ownership is locked in. Metadata is necessary for the artist to be paid for their music. Metadata appears as simple text fields that need inputting. The fields are numerous and include amongst others the title of the song, the musical key, the date of recording and musicians involved. Probably the most important piece of metadata is the ‘ISRC code’ (international standard recording code), which is a unique number assigned to each individual song, much the same as an ISBN number in book publishing. If you do nothing else, ensure that the correct ‘ISRC code’ is registered against each master. If you are releasing your music via an online distribution service, you will not be able to complete the submission without an ‘ISRC code’. A code can usually be generated by the distributor, but you may wish to create your own if you plan on releasing a sizable quantity of content. Either way, it is good practise to try and enter as much metadata as you can at the time of exporting a file. You will not always remember and not all information will be available to you, but it is a good habit to develop. Many DAWs do not permit exporting extensive metadata. If yours is one of them there are separate programs that can attach metadata to your master.

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FORMAT

The final decision is which format to render in? Each DAW will have the ability to export audio in a variety of formats. Professional masters are always printed in an uncompressed format (WAV for Windows or AIFF for Apple Mac) and at a bit depth of at least 24 and a resolution of 48k or a multiple thereof. The end publisher is then able to convert the master to whichever format is needed. Below is a list of common file types.

WAV – ‘Waveform Audio File Format’ was developed by IBM and Microsoft as a way of storing and playing back uncompressed audio on a Microsoft Windows computer. The important point to note is that no data compression is used. The audio file shared should look and sound exactly as it does on your hard drive.

AIFF – ‘Audio Interchange File Format’ was developed by Apple Inc and is the Apple Macintosh computer version of an uncompressed audio storage and playback medium.

MP3 – ‘MPEG-1 Audio Layer III’ is a file format that compresses digital audio and reduces the file size. Basically the encoding system lessens the accuracy of components of sound that psychoacoustic laws consider beyond the capabilities of human hearing. The end result is a file size that is a 75-95 percent reduction of the original, depending on which level of quality you choose. The MP3 format became prominent when the internet was much slower and memory more expensive. Storing and transfer of uncompressed digital audio was more difficult than it is now. Data compression allowed for swift transfer times and light files which made up for the loss of quality in millions of end users’ eyes. The MP3 format is still one of the top ways to share music, but its prevalence is waning. MP3 files can be created by anyone from a WAV or AIFF. 

FLAC - ‘Free Lossless Audio Codec’ is another consumer audio coding format that can reduce a file size by 50 percent or more. Whereas MP3 compression imparts damage on the audio that can never be undone, FLAC compression is reversible by way of a decompression process that restores an identical version of the original file. This format is perfect for storage of vast amounts of audio. Less storage space is needed and a perfect copy is always brought back to life, exact in every way, by uncompressing.

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This concludes the main settings that will influence the quality of your master. You will always find more options unique to each DAW, but these fall into the category of personal workflow choices, such as the ability to output two formats simultaneously, import the mastered track back into the project and so on. Awesome, now you have a handle on all the settings needed to export the right way first time, what about level?

LEVEL OF AUDIO

When talking of the level of audio within mastering, and our requirements for this system, there are three all important forms of level we need to monitor. They are:

Peak Level (DB).

Dynamic Range Level (DR).

Loudness Level (LUFS).

Peak level is exactly as it sounds and represents the absolute peak of a sound wave. Peak level is measured by a Peak Level Meter, the type that appears on each channel in your DAW. Peak Level Meters have various settings whereby they can constantly update and return a level in real-time, but also can hold the maximum level that is reached. It is common to display both and it is the held peak level we are interested in here to establish the loudest peak in the program material. Imagine waveforms as a range of mountains, all at different heights, the highest mountain peak of them all (Mount Everest) is the one that is recorded as the peak level for the entire range. We need to know which sound wave peak is the Mount Everest and the hold figure gives us that information. Peak levels are recorded in decibels relative to 0dbfs, so if the loudest peak in a song is -3db below the digital ceiling of 0dbfs then the held peak recording will show -3db. The peak level is important because although theoretically you could output a master that hits the maximum peak level of 0dbfs, it is generally accepted good practice to lower the peak level a fraction to say -0.2db to avoid inter-sample distortion. This form of distortion happens when actual true levels of passing audio are a fraction louder than the levels displayed.

Dynamic Range Level is the degree of dynamic range that exists within a recording. This generally is as simple as the difference between the loudest and quietest moments of music, although some meters may incorporate more complex psychoacoustic methods of measurement. Dynamic range is recorded on specific Dynamic Range Meters but can also appear within other metering systems. In general music production it is not necessary to put a dynamic range figure on a piece of music – the old adage ‘if it sounds good, it is good’ works well; but within mastering, and more specifically within our system, it is a good measuring stick to check that we are in line with our Top 50 professional references. We have already established the method for measuring the dynamic range of a master, so make a habit of comparing every export. Dynamic range measurements are represented by the letters DR then a number. DR8 means there are eight decibels of dynamic range between the loudest and quietest moments. If the peak was measured at -5db then the trough would be -13db. The most important point to realise is that dynamic range figures never change until you apply some dynamic range processing. The act of turning music up or down with a volume control or fader will never change the dynamic range rating. A volume control change will alter the peak level figure and the loudness rating, but never the DR number.

Loudness Level is a comparatively new concept and is measured using a Loudness Meter which returns a number relative to 0 decibels full scale (0dbfs), hence the LUFS unit name of ‘loudness units relative to full scale’. As well as being recorded by specific Loudness Meters, LUFS measurements are now appearing within many mastering analysis plugins and often feature at the export stage of a DAW. Such importance is placed on this figure because streaming sites are now measuring and rating the LUFS of each song and adjusting playback level to match their LUFS benchmark. For example, if their benchmark loudness level is -14 LUFS and you upload a recording with a loudness level of -8 LUFS, then the hosting site will turn your song down by -6db. LUFS data is usually shown in two forms, an averaged figure derived from the complete song or a momentary figure giving short bursts of information. The number that interests us is the averaged LUFS readout for the complete song.

It is important to understand how these three measurements interact with each other.

If the peak reading is below 0dbfs then there is available headroom to exploit. Increasing the peak level to nearer 0dbfs will make the recording sound louder. This will also increase the LUFS loudness rating, but will not change the dynamic range number.

Once 0dbfs is reached, or if you have a recording that already carries a peak reading of 0dbfs, then the audio cannot simply be turned up anymore to be made louder as the maximum digital ceiling level has already been reached.

To increase loudness (LUFS) once 0 dbfs is reached, one or more of the following needs to happen - the quietest parts must be made louder, the loudest parts made quieter so that the recording as a whole can be turned up, or a combination of the two. Either way we are talking about dynamic range reduction, i.e. limiting, upward or downward compression or both combined. This will change both the loudness and dynamic range rating, and the peak level too if downward compression or limiting without auto gain makeup is used.

Recordings with a lower dynamic range rating sound louder than recordings with a wide dynamic range, but only when each is normalised to peak at 0dbfs. If these recordings are level matched then the loudness advantage is neutralised and in fact, more often than not, the wider dynamic range recording will sound subjectively better.

DIGITAL HOSTING SITE UPLOAD LEVELS

And now to the all important question; what level of loudness is right for each specific online hosting service, for example Spotify or YouTube? Well the first point to note is yes, each streaming service will have their specific benchmark level of playback. But the reality is you are not going to have the level of control over your uploaded music that you might be envisaging. You will only get the opportunity to upload one master at one level to a distribution service such as ‘DistroKid’ or ‘CDBaby’. It is the distribution company’s role to liaise with the hosting sites. Companies like Spotify and YouTube will not work with individuals on a one-to-one basis, so you will never have the opportunity to customise your master to suit each major hosting website. Even if you could, would you really want to? At the time of writing (and taking the random selection of CDBaby as an example - this is not an endorsement of any kind, all distributors are great), CDBaby will pass your music on to nearly forty hosting sites and that is only the majors that pay creators – there are many, many more if you wish to gift your music for free. Would you really want to be able to research each of the forty plus hosting companies and tailor each upload to suit? What we do have control over is the power to optimise one master for the best results across all sites, which if you cast your mind back to the start of this book, is the very essence of mastering; to be able to find the optimum middle ground for all replay systems. So accepting that one loudness level is the best solution for an uploaded digital single (compact disc and albums differ – see the next chapter ‘Mastering An Album’) which loudness level should you choose? The most widely accepted answer to this question is a LUFS level of -14db, i.e. fourteen decibels below full scale digital zero (0dbfs). Which happens to be the exact level at which we have calibrated our monitoring system! Although we have used the K-System and referenced K-14, K numbers and LUFS are basically the same; for K-14 read -14 LUFS. Therefore, if you master a track to a K meter reading of K-14 you will have a loudness level that matches the recognised best average for all streaming websites.

However, I will offer one alternative view, an approach that I prefer to take, and it is this; gain the knowledge of how to measure and manipulate DR, LUFS and peak levels so that in the event you have to output a master to a specific requirement you can, but do not let any preconceived loudness levels dominate your thoughts or influence your musical moves. Instead use the system within these pages to present the master in its best light, with loudness being one part of the jigsaw. Assess loudness as a comparison to your Top 50 references rather than an ‘end goal’ figure. Only measure the DR and LUFS once you have completed your first version of the master and view the figures as secondary in importance to the vibe of the music. Whatever the measurement comes out at, within reason, upload the master as it is, assuming it falls within the boundaries of the professional references, any master with a -14 to -7 LUFS will be fine. The reason I take this approach is if the master is too loud for any given streaming site all that happens is the controlling system turns the track down. No problem, it will still sound as it should. There will be no other change in the quality, no alteration except for a reduction in level and I am happy knowing that I have made it the best it can possibly be and not compromised. Therefore, make it as loud as it can be without inducing any degradation. Never prioritise loudness over quality, but neither let any concerns over loudness cloud your view on what is best for the music.

Now enable your Loudness Meter if you have not already done so, export your final master and measure the LUFS. How does it sound compared to your references? What is the LUFS rating? Could you push the loudness more? Always act as your own critique as well as asking trusted sources. Even though our Top 50 songs are embedded within our template, I still prefer to make the final master comparison within my clean Top 50 session. Transferring work from the mastering template to the Top 50 session takes literally seconds and provides a fresh reality check without all the distractions of working plugins and channels. I recommend you do the same; you will be surprised how much more focused your mind becomes.

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BACK TO BASE LEVEL

It has been an amazing journey since the opening chapter when I first introduced the concept that you can and should master your own work. Do you remember earlier in the book I mentioned we would return to the online algorithms and base level mastering? Well here we are back full circle. The whole idea of using the base level online mastering algorithms was to compare your work to a level of mastering available to all and is achievable at the click of a button. You need to confirm that your mastering is better than the base level. This process will also serve as a chart of your progression. Having exported your best final master let us compare it to the base level algorithmic master in a blind test. The process is as follows.

If you have not already done so, upload the original un-mastered mix to an online mastering service and record the preview clip they provide for auditioning purposes into your DAW on track one.

On a separate second track load in your best final master.

Align both masters as close as you can by zooming in and dragging one along the timeline to match.

Load the free ‘4U + blind test’ plugin from Hofa-plugins or route both tracks to separate outputs on a mixing desk.

Ensure both are playing in sync and are level matched (‘4U + blind test’ has a level matching function which you should use).

Then skip from one track to the other comparing all aspects of each master and write down your thoughts.

Which sounded better? If your master sounded the best, wonderful, sign off the project, backup your work, upload the track to your chosen distributor and start all over again on the next song! Did I mention practise, practise, and practise? If your master sounded second best, no problem, next time you will get closer and before long you will beat it every time. Use this experience to learn from the algorithm. Why does it sound better? With this in mind, I do not recommend going back to the start and trying to master your project again with a view to trumping the remote wizard. In my experience this only wastes time as you shoot for a moving target over and over. You will improve with each project. Your energy is better served by correcting your master to bring it in line, and if you cannot reach an acceptable match, put it down to experience and pay for the online master with a view to beating it next time.

The process of matching your master to the online version is the same process as you learned earlier except this time you are using the online master as your reference and not a professional Top 50 track. Disable all plugins and import the online algorithmic master audition clip (level matched to K-14). Start by EQ Matching your master to the algorithm and then level match using your maximiser or mastering limiter to find the same level of loudness. Now scan the two files mentally as you have done all the way through and work on each step, one at a time. Work fast and intuitively. Once you have completed the steps export the new version of your master and run the blind test comparison again. How close did you get? If your master is now an acceptable match, job done, archive and upload. If you still did not quite get to the base level do not feel despondent, stay positive and safe in the knowledge that you have achieved a massive amount. You now have all the tools to smash it out the park and reach bigger heights; it is now only a matter of time. You will get there. Be excited about your next project, your next master and the many more fabulous creations that will materialise now you have the knowledge and empowerment. Each master will be better than the last until you reach a level where you no longer need to look down at the base level, you will have transcended multiple layers of quality and you will smile to yourself and remember this day and marvel at how far you have come. Each day is a chance to learn and progress, look up, stay positive and practice every moment you can.

10

MASTERING AN ALBUM

Until now the focus has been on mastering one song to be released as a digital single. However, when mastering a collection of songs for a full album or shorter EP the process is slightly different. Consideration must now be given to how each song relates to the other, in addition to how they sit in the wider audio universe. The good news is that all the skills you need have already been learned in the previous chapters. A logical step by step process is all that is needed to avoid any mishaps and achieve the best results. The process in brief is as follows:

Copy all songs into a project folder and import them into the album’s session.

Listen to the songs and choose one best song that is the closest fit to the sonic quality you envisage for the album and optimise if needed.

Balance the remaining songs to the best song in both sound and level.

Add fade in and fade outs whilst setting the correct gaps between sound files.

Set the peak level for the complete album.

Export.

Two points to note before moving onto ‘The Album’:

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ONE: MONITORING LEVELS AND OUTPUT TARGET LEVEL

All through this book we have worked to a studio monitoring level of K-14 (-14dbfs at 83 decibels). The reason, we learned, was that 83 decibels is widely regarded as the best optimum monitoring level and -14dbfs the most quoted upload benchmark for auto-levelling mechanisms on streaming websites. This all makes perfect sense for single songs whose destination is a streaming website. However, we are now talking about an album so it must be assumed that these songs could end up on a physical format such as a compact disc. Therefore the output level should meet compact disc standards, i.e. the program material must hit 0dbfs at some point during playback. This will probably mean you will be mastering to a louder integrated loudness. DR12 to DR7 are common with DR4 or DR5 not unheard of, so adjust your monitor gain accordingly to the loudest track.

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TWO: COMPACT DISCS

And a brief word about compact discs. There are two ways to produce an audio CD - either professional CD replication or short run duplication. CD replication generally starts with orders in the 10,000 to 100,000 range and requires a front-end process not covered in this book. CD replication resides in the realms of record companies and pro mastering engineers. Short run CD duplication is a service offered to independent musicians who can order very small batches duplicated from a standard CD master that you can burn at home using any media player interface. The assumption is made that you will be producing short run batches produced by yourself or employing a short run production company. Anyone involved in a CD replication run of tens of thousands of units would, I am sure, use a professional mastering engineer. But, even if the purpose of producing a CD is for your own listening pleasure, to share with friends and family or sell at a gig, you still want it to sound and play like a commercial album, as there is no way of knowing where the project may take you. Right now it may be a collection of songs sitting on your hard drive, but amazing results happen when you commit energy and persistence to a cause, next year who knows? The truth is no-one, not even the best in the business can guarantee hit after hit without any misses and neither can you predict how your work will be received by a wider audience. Therefore, always do your best at all times and approach every task as practicing and waiting so you are ready when your big opportunity arises. Okay, let us focus on each step in detail.

THE ALBUM

The absolute best results for mastering an album will be obtained by working from an un-mastered mix for each song, i.e. not an already mastered version. Having said that, if the majority of the songs on the album have already been mastered as singles, then the time and effort in re-mastering all of these from the core mix, in my experience, is not worth the trouble. As long as you have a HD version of the master, which has not had any dithering applied, try using that first. For me this is the best trade-off between quality and time. You may have to re-master from the mix if one particular song is way off the sound of the album, but the majority of time a subtle re-balance gets you there.

When mastering an album there are three possible scenarios to consider:

The songs appearing on the album have already been mastered and released as singles

None of the songs have been mastered and are still in mix form

A combination of A and B where both mixes and masters exist

The steps that follow below apply to Scenario 1, i.e. all songs have previously been mastered and released as singles. This is common in today’s singles-led market as individual songs are released and then compiled as an EP or album later. Scenario 1 is the quickest and easiest route as most of the hard work has already been finished. A simple balancing of level and sonic response is all that is needed. For Scenarios 2 and 3, I suggest you run separate sessions in tandem – one session for the album as a whole and a separate session for each un-mastered track. Work each un-mastered song individually as no two songs are alike and each will require different settings. Treat each un-mastered track as if it were a single and export each to your album session when you feel you are close to the album sound. Then proceed as below from ‘Step Two’ as you finalise the complete album session.

STEP ONE: COPY AND IMPORT

Open a new blank session; save the project using the ‘save as’ function and name the album. You are starting with a blank canvas and not your mastering template as all songs have already been mastered at different times. However, the techniques and plugins used within your mastering template will still be applied. Locate the source file for each song and copy them into the project folder. For best results use a 24-bit or 32-bit master and not the 16-bit dithered version. If you only have 16-bit masters available you may want to contemplate returning to the source mixes and re-mastering. Once you have all the songs in the project folder, import them to separate tracks. Then move each song along the timeline so that they will play in the order they appear on the play listing. Giving each song its own track means you can add unique plugins to each individual song and spacing them along the timeline avoids the need to constantly solo or mute channels.

STEP TWO: CHOOSE ONE ‘BEST’ SONG

Start listening to all tracks by clicking into each song and listening to a small snapshot of each. There is no need to listen to each song in its entirety; a few seconds of the loudest section will suffice. Which song has the closest properties to the vision you have for this album? If you do not have a fixed sound in mind, which song stands out as being the best match to your Top 50 references? Pick the one that would make you smile if all the others matched its EQ, dynamics and spatial properties, because that is the plan, to bring all the other songs in line with the best. It is so important to have a consistent vibe running through an album. Even if the individual songs are very different the product as a whole should give the listener a sense of oneness. They should perceive all songs as being from the same source, from the same artist in one finely tuned easily consumed package. Leave nothing to distract them from their enjoyment and lead them to question authenticity. A listener has to believe in the product. Once you have settled on your best track, ask yourself “Is the master as good it could possibly be?” “Is there anything extra I can add to further enhance the song?” Compare against your references and if you feel the answer is “yes, I think I can add a sprinkling of extra magic”, then work through the same process you applied earlier to single songs, inserting any relevant plugin from your mastering template onto this song’s channel. Feel free to move back to your mastering template for this one important song if you wish, but as each song in this scenario is already mastered the alterations should be minimal. If you envisage making many changes then a re-master from the original mix is the way to go. Assuming you are happy with your choice and the song is as good as it can be then move onto the next step bearing in mind that nothing is set in stone, you can always tweak away later.

STEP THREE: BALANCE ALL SONGS IN SOUND AND LEVEL

As mentioned previously, an album should have a consistent sound and the way to achieve this is to align all other songs to your best track; that is the goal here. The process is very similar to mastering a single song in ‘Chapter 8’, only now you are using your chosen best song as the benchmark, not your Top 50 references; and corrections should be small and few in comparison. Use all that you learned previously to carefully bring all the other tracks in line with the benchmark best. Start by finding the optimum level for each track. Raise or lower each channel’s fader until a consistent sound level is obtained across all tracks. This must be done by instinct alone; no form of technology can perform this task. Once you think you have nailed it press play and click into each song in sequence at the loudest section playing a few seconds – do they all sound at the same level? Fantastic, now start with the first song and go through the same process as established in ‘chapter 8’ (step five onwards) for each song. Listen, react and process adding only the few plugins you need. There should be no need for major change, most of the work will be subtle EQ shaping, a touch of additional dynamic control and the occasional reverb patch if the odd track is too dry. But do whatever it takes to find a uniformed sound and align all other tracks with the best song. No-one listening to the end product will know or care how many plugins you have used or how you have changed a song; only you know what each song was originally like. Pay careful attention to the transitions from the ending of one song to the intro of the next, they should flow without erratic inconsistencies.  Finally, double-check the levels you set initially. You may have been compensating as you progressed, but even so, with the extra tonal and dynamic processing the subjective levels may now be out of balance. Work through the process of balancing levels once more in exactly the same way as you started this step. It may seem strange to be levelling twice but as we now know assessing audio at the same level is one of the most important factors in distinguishing differences. Without the initial levelling you cannot be totally confident in what you are hearing when comparing each song to the benchmark best. After levelling all songs a second time move on to step four.

STEP FOUR: FADE IN, FADE OUT AND SET GAPS

Shift your attention to the start and end of each track and the silence in between. Starting with the fades; are the existing fade-in and -outs working within the album context? It is easy to shorten a fade that is too long by applying a new curve to the audio file. The only way to extend a fade is to first extend the music. The usual way to do this is by applying a short burst of reverberation; just enough to accommodate the fade out, but any out-of-the-box method will suffice. Again, pay special attention to the transitions between two songs and not each song in isolation. Does the fade-out of a song work with the fade in of the next song if there is one; many songs do not have an audible fade-in, but even songs with an instant musical start contain some sound in the preceding few milliseconds, check this ‘noise’ is pleasing and contains no distractions. Each DAW will provide a number of different shaped curves to use as fades. Familiarise yourself with the different results each shape gives. Often the default diagonal line is not the best.

Moving on to the silence between sound, or more specifically the time between one song finishing and another starting; this is another task that can only be accomplished by feel alone. Position the DAW’s cursor four or five seconds before the end of the first song and let it play through the start of the second song. How did it feel? Were you waiting too long for the second song to start? If so, close the gap between songs one and two. Were you surprised how quickly song two started after song one ended? If so extend the gap between the two. Carry on through the complete album taking one gap at a time until you are satisfied with the way the album plays. Do not move on until you can sit back and enjoy the album in its entirety. If your stomach turns or you feel a negative emotional response, your instinct is telling you it is still not right.

Having said that, it is enough to do your best in any given time frame; set time limits and do not stress the small stuff. If you fail to set time limits you will fail to produce finished work. There is one final method of setting gap length that I always use and that is to count the beats of the preceding song and continue counting the beats to the tempo of the song through the silence and up to the first downbeat of the second song. Try both feel and counting beats to see which works for you.

STEP FIVE: SET A PEAK LEVEL FOR THE COMPLETE ALBUM

With all the creative decisions made, all that remains is to optimise the peak level and export the music files. As previously mentioned, one of the standards of CD production is that the program material must at some point hit 0dbfs. This is as easy as finding the loudest peak and reducing or increasing the DAW master fader (or using a gain plugin) to suit. If you are not sure where the loudest peak is, let the album play through while you have some lunch and the peak level will be held for you upon your return. Remember, it is advisable to set the actual fader to register slightly below 0dbfs at say -0.2db to allow inter sample peak headroom. Once the overall level is set all that is left is to export the music.

STEP SIX: EXPORT

The process of exporting an album for uploading to streaming and download hosting sites and indeed for short run CD duplication is straightforward enough. Earlier I highlighted the difference between short run CD duplication and professional CD replication; you may have read articles on setting all manner of PQ encoding and producing glass masters, all of which apply to professional replication. For our purposes it is enough to export standard audio files in the same way as any other file (see the previous chapter ‘Exporting’). Each file will need to be 16-bit 44.1kHz unless uploading to a high definition hosting website. The process of outputting is therefore to create a folder to hold the final files and export each song one at a time. Although you may have the ability to output all song clips in one automated pass, this method, although quick gives unsatisfactory results with gaps that are longer or shorter than are ideal. One trick I always use to aid this process is to add a ‘timeline’ track between each song track and create an empty clip that spans the length of each song including silence spaces. This will be different to the actual audio clip length and will represent the real start and end points you want to achieve. Once all ‘timeline’ empty items are created it is as simple as clicking on the relevant one and using the ‘time selection to item’ function to move start and end points in place. The regions to export for each song are always correct this way and are saved with the song, so if you ever have to re-visit the project and export each file once more, you do not need to re-set all the individual start and end markers. With the bit and sampling rate set and any dithering plugin enabled (if needed) name the track and export away. One final trick with regards to track labels; adding a number before each song, for example, 01 before the first and 02 before the second saves time when dragging all the files into a playlist when creating a CD track listing or similar. Having the tracks numbered in their correct order means the songs fall perfectly in position rather than being arranged alphabetically, which would then need re-ordering or having to load each song individually in sequence. A small timesaver but small moments saved wherever possible add up to minutes and hours over the duration of many projects. Fantastic, export away!

Before uploading and setting your creations free, listen to the album the next day, or better still a week later, to ensure you are happy with the results. Play it out in as many settings and on as many playback-systems as you can. There is no way the music will be identical on each system, look for an average middle ground. Any problems should make themselves known, but do not assume any fault is with your mastering, the playback-system and environment might be the cause so always double-check on another system. The more you know about the sound qualities of a replay device the better your diagnostics will be, so listen to music you are familiar with too; if it suffers from the same issues you can be sure it is an artefact of the hi-fi or boombox etc. When listening with fresh ears, whether driving in your car or exercising with ear buds, are you satisfied that you did your best and the album sounds like an album should? That is all that matters. Personal satisfaction; be happy!

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IN CLOSING

Sadly this is where our journey ends as far as this book is concerned; but feel free to write with your experiences. All that remains is for me to thank you for buying and reading. I hope you have enjoyed this title and that you are now seeing the act of mastering at home in a new informed, enjoyable and positive ‘can-do’ light. The big take-away is, yes, you can be a mastering engineer; and yes, do use modern technology to give you an advantage. But whilst technology is great, there is no substitute for the spark of brilliance that can only be injected by a human creative mind. Yes, do use software that analyses your master and double-check that the decisions you have made are in line with recognised best practices; but compliance does not always make the best, most exciting masters. Go to the edge of what is possible and look over the abyss. When analysing commercial tracks there will almost certainly be some elements that are technically wrong. Maybe it is the ‘good wrongness’ that makes them stand out. Once you have mastered the rules of mastering take chances; it is okay to break the rules and go beyond accepted limits if no audible damage results. So go forth, create wonderful audio that is alive and spitting with sparks of energy. Push the boundaries and make music with your own signature sound. Make beautiful masters that you are happy with and can be proud of. Practise and practise some more. Learn to trust and listen to your inner voice, it is rarely wrong. I wish you all the luck in the world on your journey. Choose where your path is leading and be the master of your own destiny.

ABOUT SIMON TAYLOR

I was born in Bath, England on 14th October 1969 and raised a few miles east in the historical market town of Bradford-on-Avon. I have obtained industry certification in the subjects of music technology and advanced recording techniques, passing with distinction and have been an active member of the music industry for over thirty years as a songwriter, mix/mastering engineer, as well as providing private tuition to the next generation of Chris Lord-Alges and Billy Deckers. In 2020 I co-wrote the best selling music technology book ‘Template Mixing and Mastering: The Ultimate Guide to Achieving a Professional Sound’ with Billy Decker. If you want to speed up your mixing and take it to the next level I wholeheartedly recommend it.

It would be an absolute pleasure for me to read your thoughts and hear your sparkling masters. Feel free to write to me any time using the contact form on my website http://www.simontaylorauthor.com the place for all my publications. I will leave you with a small selection of the press reviews for ‘Template Mixing and Mastering: The Ultimate Guide to Achieving a Professional Sound’ to whet your appetite.

‘This book is a gift of gratitude to anyone that wants to take a step into out-of-this-world sounding songs.’

Rodney Atkins, multi-million selling recording artist

‘This book is accessible and engaging, with plenty of pithy one-liners that make serious and thought-provoking points. Essential reading for anyone interested in improving their mixing skills, and streamlining their working methods.’

Paul Tingen, Sound on Sound magazine

‘Some absolute gems of advice to pick up here. The brilliant and clear way everything is presented and the actual (hallelujah!) settings for your template laid out for you, make this essential for any mix engineer, or anyone striving to be one.’

Andy Jones, Computer Music Magazine

‘The book is also an excellent reference for identifying frequencies a pro mixer will often cut or boost on certain instruments; I always nodded my head at Billy’s choices. Throughout the book there are a lot of thoughts and practices that will help a mix engineer keep their focus, searching for ‘feel’ as opposed to ‘perfection.’ There really is a lot of information packed into these 126 pages.’

Larry Crane, Tape Op Magazine

‘The book is fantastic! I methodically went through and copied the template. Results for me have been amazing and my mixing clients are thrilled!’

Matt Boudreau, Working Class Audio

GLOSSARY

Absorption – When music is played within a room sound waves bounce around, reflecting off all surfaces. Not all of the sound is reflected, however; some becomes absorbed into the materials found within the room. ‘Absorption’ is the term used to describe this loss of sound energy and is found within ‘reverberation effects’ and ‘room measurement tools’ to specify the amount of sound that becomes absorbed.

Acoustics – The scientific study of sound. Also a term used to describe the natural quality of a musical room, i.e. a ‘live’ room used for recording can be said to have ‘good acoustics’.

AD/DA Converter – Is a digital device that takes sound from the analogue world and digitizes the audio so that computing devices may process the music before returning it to the analogue world. ‘A’ stands for analogue and ‘D’ stands for digital. ‘A/D converters’ convert analogue to digital. D/A converters convert digital sound back to analogue.

ADSR Envelope – Describes the complete cycle of any one sound from its initial burst of energy back to silence. ADSR stands for ‘Attack, Decay, Sustain and Release’ and describes how sound changes over time. The whole movement from initial transient to silence is often described as an ‘envelope’. The phrase is most common among synthesizer and keyboard sound design, and in drum transient shaping plugins.

Air – A word to describe the uppermost highs of the audio spectrum. Increasing the level of this range of frequencies can be said to give a sound more ‘air’.

Amplifier – Is a device that increases the power of an audio signal and is often referred to simply as an ‘amp’. Amplifiers can be found in many forms within a recording studio from the ‘device’ that ‘amplifies’ an electric guitar to the ‘amplifier’ that links the mixing desk to the monitor speakers. Devices such as monitors that have amplifiers built in are referred to as ‘powered’ or ‘active’.

Amplitude – The size of a vibration that results in a sound wave. The bigger the vibration is, the deeper the wave and the louder the sound.

Analogue – In sound engineering terms ‘analogue’ relates to the audio signal that exists as one unbroken physical sound wave and the equipment used to process this wave of frequencies.

Attack – The very first initial burst of sound energy.

Attenuation – A technical word that means to ‘turn down’ or ‘reduce’ levels of sound.

Audio – Can be used to describe any sound in general, but is more often used where sound is recorded and broadcast.

Audio Interface – allows sounds from outside a computer to be processed digitally by the computer. Interfaces use ‘analogue to digital’ converters to digitise analogue signals and ‘digital to analogue’ converters to return the signal back to the analogue world. Often interfaces can accept signals in many formats and also pass on ‘MIDI’ data.

Auxiliary Send and Return – Often referred to simply as ‘aux’ inputs and outputs; auxiliary connections are generally configured as additional ‘sends and returns’ on an audio device (analogue or digital). The most common uses are to divert sound out of, and back into, a mixing desk to an ‘effects unit’ in an ‘effects loop’ or to feed the headphones of a recording artist (send only).

Balance – Can have different meanings relating to mix engineering. Firstly, a balance dial can be found on a mixing desk to simultaneously control the levels of both the left and right channels in a stereo signal.  Balance can also be used to describe the overall composition of a mix, i.e. the ‘balance’ among all sounds. If a mix sounds wrong it is referred to as ‘out of balance’ whereas a good mix is a ‘balanced mix’.

Band – When relating to ‘equalisation’, a band is the range of frequencies that one particular control affects. Equalizers may have many ‘bands’, e.g. a graphic equalizer, or as few as two or three. A ‘band’ may also be used to describe a group of musicians.

Bass – The lowest region of the frequency spectrum. ‘Bass’ can be represented by frequency numbers, e.g. 64Hz, or spoken of in subjective terms such as ‘punchy’ or ‘thumping’. ‘Bass’ also relates to the instruments that form the foundation of music, such as a ‘bass guitar’ or ‘bass drum’.

Bass Enhancer – A ‘bass enhancer’ is one particular type of effect that creates artificial bass frequencies to enhance existing ones. It uses psychoacoustic technology to fool the brain into thinking that bass frequencies are present when they are not. This is useful when playing music through small speakers.

Bell Shape – Is a type of equalisation curve; an equalisation curve being the visual shape that is formed when a range of frequencies are either boosted or cut. A ‘bell shaped EQ’ is also referred to as a ‘peaking EQ’ or ‘parametric EQ’.

Bit – The smallest unit of information used in computing, a bit is derived from the words ‘binary digit’.

Bit Depth – When relating to digital audio ‘bit depth’ means the number of ‘bits’ of information that each sample contains (a sample being a digital snapshot of analogue sound at any given moment in time). CDs are able to accommodate 16 bits per sample, for example.

Boost – Is used to denote an increase in the level of a signal or range of frequencies.

Brick Wall Limiter – A certain type of ‘limiter’ that restricts an audio signal to an absolute maximum level. The maximum level will never be breached and is controlled by the ‘threshold’ setting. A ‘brick wall limiter’ is most commonly placed on the master bus to reduce dynamic range and safeguard against digital distortion.

Bus – Or ‘buss’ as the term is sometimes written, is a mixer channel or circuit that accepts more than one source. It can be thought of as a vehicle for audio, transporting multiple sounds together from one destination to another in one direction.

Bus Compressor – A type of ‘compressor’ that has been designed to work with a mixture of sounds blended together as one. As the name suggests a ‘bus compressor’ works best when placed on a ‘bus’.

Cents – A measurement system for dividing the smallest musical interval, a semitone, into further subdivisions. There are 100 cents to a semitone and although it is impossible to hear the transition from one cent to another, cents still have an important role to play in sound design.

Channel – A ‘channel’ represents the path an audio signal takes when travelling from one location to another, commonly on a mixing desk and within a digital audio workstation. A channel carries one source in one direction. Mono is a one channel format. Stereo is a two-channel format carrying a left and a right signal.

Channel Strip – Is a collective word to describe the various components that sit along a mixing desk’s ‘channel’. From top to bottom, a ‘channel strip’ may include an input level control, an equalizer, dynamics processors, a fader and output options. A channel strip is duplicated many times on a mixing desk, giving the impression that it is a complex beast, but closer inspection will show that much of the desk is one channel strip replicated again and again.

Chorus – Can be a musical section within a song (‘Don’t bore us, get to the chorus!’ is a well-known phrase in songwriting). Alternatively a ‘chorus’ can be a type of effect, one that takes a signal, duplicates it and then intentionally alters the duplicate sound so that it is fractionally out of time and out of tune, before mixing it back in with the original sound. When performed correctly the effect ‘thickens’ sounds in a pleasing manner.

Clip – When used within the context of a ‘digital audio workstation’, a ‘clip’ is the part of recorded audio that is actually played back. Not all the sound that is recorded has to be heard. By specifying that only one section is replayed you are creating a ‘clip’. Alternatively, to ‘clip’ audio is a term given to the act of pushing sound through the limits of the carrying device, resulting in distortion.

Clipper – A type of sound processor that intentionally ‘clips’ a sound source in a controlled manner to reduce the dynamic range of a signal. A clipper works much the same as a compressor or limiter.

Compression – Can be the act of reducing the ‘dynamic range’ of a sound source; ‘dynamic range’ being the difference between the quietest and loudest parts. ‘Compression’ reduces the loudest peaks of a sound. The whole signal can then be boosted back up to pre-compression levels. The result is a more controlled, stable signal. Alternatively, ‘compression’ can be used when discussing the reduction of the quantity of data when changing audio to a compressed format. MP3, for example, is a data compression format.

Compressor – An audio processor that reduces the dynamic range of a sound source (See also ‘compression’).

Console – Another word used to describe a mixing desk. Within a ‘digital audio workstation’ context the term ‘console emulation’ is often used to describe a plugin that claims to recreate the qualities of a physical mixing desk.

Crossover Frequency – Is the dividing point at which the frequency spectrum is split, for whatever purpose. All speakers with more than one cone have a crossover frequency and ‘equalizers’ contain crossover frequencies as different bands interact.

Cut – Another word for ‘attenuation’ or ‘reduction’, or alternatively used to mean ‘delete’ or ‘mute’.

DAW – Is an abbreviation of ‘digital audio workstation’, the all-in-one ‘studio’ in a computer that musicians and engineers use to create musical compositions. Generally, but not always, a ‘digital audio workstation’ will have two main windows; a ‘tracking/editing’ window that replicates a multi-track recorder and a ‘mixer’ window that re-creates a physical mixing desk.

DBFS – An abbreviation of the term ‘decibels relative to full scale’; ‘decibels’ is a unit of measurement for the level of sound, ‘full scale’ being the maximum peak level of a digital audio system.

Decay – The part of a sound that reduces after the initial attack; ‘decay’ is a parameter commonly found in synthesizer sound design and effect plugins.

Decibels – A unit of measurement used to document the power of sound.

De-esser – An audio processor designed to reduce the ‘sibilant’ consonants in a vocal performance. Although designed with vocals in mind, a ‘de-esser’ can be used to reduce any problematic frequency that is within the range of the plugin.

Delay – To ‘delay’ a signal or recorded track is to move it backwards in time. This could be for the purpose of aligning one sound recording to additional takes, or perhaps to create a time-based effect.  A ‘delay’ effect is a plugin that automatically records a signal and then plays it back at a later point in time. The delayed signal may be one ‘echo’ or many repeated ‘echos’.

Detune – A process that moves the pitch of a sound downwards. The amount of ‘detuning’ applied can range from a few ‘cents’ to many musical pitches or octaves.

Diffusion – Is to affect sound in a way that spreads the sound energy evenly across any given environment, be it real or virtual. ‘Diffusion’ is a term used in ‘room analysis’ tools and ‘reverberation’ plugins, for example.

Digital – Digital audio can be defined as sound that has passed through an analogue-to-digital converter or has been created synthetically by a digital sound engine. Basically ‘digital’ means digits have been used in the process - e.g. a computer uses the digits 1 and 0.

Digital Audio Workstation – See ‘DAW’.

Distortion – This happens as a result of pushing the level of sound beyond the limits of its replay system. Music can be ‘distorted’ intentionally or by mistake. There are many ‘distortion’ effects that change the sound of instruments such as guitars, but accidental digital distortion is generally to be avoided.

Dithering – The act of reducing the number of ‘bits’ within a digital recording. The compact disc format only allows a maximum of 16 bits. Most music is created in 24- or 32-bit resolution. This means that during mastering a process must happen to allow for 16-bit CD playback (assuming a 16-bit format is the requirement). Dithering is that process.

Double Track – Is a production technique whereby one musical part is performed and recorded twice in a near identical fashion. The performance of both recordings must be as accurate as possible as any differences will cause distractions to the listening ear. When executed properly ‘double tracking’ has a ‘thickening’ effect on the audio.

Doubler – Also known as a ‘harmonizer’, is a type of plugin that simulates the ‘double track’ production technique (see ‘double track’). One or more additional layers of ‘doubling’ are automatically added without the need to perform and record each separate take. Created at the behest of John Lennon.

Drive – A form of ‘distortion’ associated with analogue emulation plugins. The effect ranges from subtle to extreme. Also a control found on guitar pedals and amplifiers to ‘overdrive’ the signal.

Drum Replacer – A plugin that automatically replaces one sound with another. Designed for use with drum performances, a ‘drum replacer’ can actually play any sample loaded into its interface as a replacement for the incoming audio. There is generally an option to blend both source audio and triggered samples together.

Dry – A term used to describe a signal that is unprocessed by any effect or processor.

Dynamic Range – A phrase used to describe the difference between the loudest and the quietest part of a sound recording.

Dynamics – Relates to the differences in volume of a sound or note when compared to other sounds or notes. Dynamics processors are processing modules that affect the dynamic range.

Early Reflections – Sound bursts forth from studio monitors at all angles and bounces off whatever lies in its path on its journey to the eardrum. The earliest bouncing waves that arrive in approximately the first 30 milliseconds (this varies per listener) are called ‘early reflections’ and are perceived by the listener as being part of the original sound, not a separate echo.

Ears – Are the most important piece of equipment in any audio chain, or to be more accurate, the brain and ears working together as one. Hear sound incorrectly and you will make the wrong decisions, you will hear problems that are not there and miss problems that are. Training your ears should be your priority.

Echo – Is a time delayed copy of an original sound that is repeated one or more times. The terms ‘echo’ and ‘delay’ can both be used to describe a time delayed signal. In earlier times the word ‘echo’ was also associated with ‘reverberation’ in an era when ‘echo chambers’ existed in recording studios to generate natural ‘reverberation’.

Effect – Can be thought of as any plugin that alters the state of a sound. More precisely, an ‘effect’ adds a second altered ‘wet’ copy of an original sound back to the original in a blend of the two. If the whole of the sound is changed by the plugin then this should be considered a ‘processor’.

Effects Chain – It is possible to send the output of one effect to the input of another. When this is done one or more times, the result is a chain of effects all linked on the same signal path and is commonly known as an ‘effects chain’.

Emulation – The act of trying to copy or imitate. Many plugins are designed as ‘emulations’ of physical hardware. They can never be the exact article, but that does not mean they should be avoided. Use your ears and follow your instinct to find the ones that best suit your style.

Equalizer – Is an audio processor that alters the levels of specific frequencies within a sound or mix of sounds and can be thought of as a more detailed variation of the basic tone controls found on hi-fi systems. The word ‘Equalizer’ is often abbreviated to ‘EQ’.

Equilateral Triangle – Is a triangle in which all three sides are of equal length. Often used to form the best monitoring solution whereby the two monitors and the listener are placed on each of the three points of the imaginary triangle.

Expander – Increases ‘dynamic range’ making quiet sounds quieter and louder sounds louder; the opposite of compression.

Export – Is the act of saving audio as a new file under a new name. This could be a ‘mixdown’ of a multi-track recording or individual tracks that are ‘exported’ to be shared or used in other projects, for example.

Fade – A ‘fade’ can be either a ‘fade-in’ or ‘fade-out’ and is the act of reducing or increasing the level of a sound over time. Fades can be automated within a digital audio workstation with different shapes of fading applied.

Fader – A control device found on a mixing desk channel to alter the sound level of the signal passing through the channel.

Feedback – This occurs when an audio output is returned to the same audio input, thus creating a ‘feedback loop’. This can be heard by the accidental ‘screech’ of a live PA system or by design as a guitarist controls the musical ‘wailing’ of their guitar feedback.

Filter – A device that turns down or totally removes certain frequencies whilst passing other frequencies unaffected. The cut-off point for determining which frequencies are affected is set by the user. A ‘high pass’ filter will cut low frequencies letting high frequencies pass, whilst a ‘low pass’ filter will cut high frequencies letting low frequencies pass.

Flanger – A type of time delay effect which also introduces a reoccurring phase shift element.

Frequency – When relating to sound, ‘frequency’ is the rate at which an object emitting sound vibrates. One vibration causes one sound wave. Measuring ‘frequency’ is achieved by recording the number of sound waves per second. This measurement is returned in ‘hertz’ or ‘kilohertz’ (1000 hertz = 1 kilohertz).

Frequency Spectrum – Is a term used to describe both the range of frequencies that a certain sound contains and the full scope of human hearing.

Gain – Refers to the loudness of audio, and in particular when loudness is increased or decreased. A gain device is often the first device after an input and is used to regulate the incoming signal.

Gain Reduction – Is simply the reduction of sound level. When a dynamics processor such as a ‘compressor’ reduces the peaks of a passing signal it is said to reduce the gain or apply ‘gain reduction’.

Gain Staging – A method of increasing or decreasing gain to ensure that each output passes the correct ‘expected’ level to the following input stage of an audio chain.

Gate – Or ‘noise gate’, is a sound processor that reduces the level of a signal below a user set threshold. The threshold is the main setting by which the ‘gate’ opens to let sound through or closes to reduce levels, although other controls are available to influence the response of the gate.

Glue – A term given to the effect certain mastering plugins have as they ‘glue’ instruments together to produce a good cohesive balance associated with professional mixes.

Hardware – Physical equipment that resides in the real world. Hardware can stand-alone or be integrated into a computer where it interacts with software.

Harmonizer – A type of effect that duplicates a signal and then moves the copied sound both in pitch and in time to create new ‘harmonies’. The degree of movement in pitch and time can be subtle or drastic.

Harmony – Pleasing tones that are added to a melody to form more complex music.

High Definition Audio – Is generally thought of as any audio with a specification that improves on the CD format (sampling rate greater than 44.1kHz and a bit depth of more than 16). ‘High Definition Audio’ is usually referred to simply as ‘HD audio’.

Hertz – The unit of measurement for sound frequency, i.e. the number of sound waves per second. ‘Hertz’ is abbreviated toHz.

High End – Is a term that is used to describe the highest frequencies within the frequency spectrum which can also be referred to as ‘top’ or ‘top end’.

High Frequencies – The frequency spectrum can be divided into the three manageable bands named ‘low’, ‘mid’ and ‘high’ (each of which is often then subdivided for additional accuracy). ‘High Frequencies’ refers to the uppermost band.

High Pass Filter – Is a device that reduces low frequencies whilst letting high frequencies pass unaltered (see also ‘Filter’).

Hold – A common control found within plugins which ‘holds’ the current state of a device for a predetermined time.

Import – Is the act of bringing material into a project from an outside source. This may be a previously recorded instrument, a sample of a drum hit, a programmed MIDI part or a video clip, for example.

Input – The point at which audio enters a device. Inputs may be real inputs on physical hardware and instruments such as guitars, or virtual inputs on plugins, virtual instruments and digital audio workstation’s channels and buses.

Insert – To ‘insert’ a plugin (or physical effects unit) into a signal flow is to place the effect directly in the audio stream so that the whole of the signal is processed and no dry unprocessed signal remains. An ‘insert point’ is a physical connection on a mixing desk that allows the engineer to route a signal through outboard hardware.

In-the-box – A term that describes the performing of audio production tasks within a computer.

kilohertz – A unit of measurement for sound frequency, i.e. the number of sound waves per second. 1 kilohertz is equal to 1000 hertz and can be abbreviated to kHz.

LCR – Stands for left, center and right and refers to the three panning positions used in a style of mixing called ‘LCR mixing’.

Limiter – Is a type of dynamics processor that reduces dynamic range in much the same way as a ‘compressor’ does. Traditional ‘limiting’ is basically a compressor with an extreme setting. However, with the digital age came more advanced ‘brick wall limiters’. It was this style of limiting that became an integral part of what we now refer to as the ‘loudness war’.

Limiting – The act of reducing dynamic range by the use of a ‘limiter’.

Loudness – Is a term used to state the ‘perceived’ level of sound. This is a subjective term that varies from person to person. One listener may think of a sound as being too loud while to another it may not appear loud enough!

Loudness War – A phrase used to describe the battle that was fought over many years to produce the loudest masters. The driving factor was the notion that louder songs were perceived as better and would result in more sales. However, greater loudness led to degradation in quality as dynamics were squashed and audio became distorted and lifeless.

Low End - Is a term that is used to describe the lowest frequencies within the frequency spectrum.

Low Frequencies - The frequency spectrum can be divided into three manageable bands known as ‘low’, ‘mid’ and ‘high’ (each of which is often then subdivided for additional accuracy). ‘Low Frequencies’ refers to the lowest band.

Low Pass Filter - Is a device that reduces high frequencies whilst letting low frequencies pass unaltered – see also ‘Filter’.

Makeup - Is a control that increases ‘gain’ (sound level), and is most often found on dynamics processors which work by reducing the loudest peaks of the audio signal. ‘makeup’ gain can then be applied to raise the signal back up to pre-processed levels.

Master Channel – Also referred to as the ‘master bus’ or ‘2-bus’; the master channel is the stereo channel on a mixing desk to which all other channels are routed.

Mastering – A term that has become blurred over recent times; ‘mastering’ can be thought of as the final stage in audio production that takes a final mix, maximizes its impact and prepares it for the medium on which it will be replayed. Traditionally mastering has always been a separate process, but is becoming increasingly integrated into the mixing workflow.

Maximizer – A type of mastering ‘brick wall’ limiter that ‘maximizes’ sound levels.

Melody – A sequence of single notes that form the bare bones of a musical tune.

Metre – Within a musical context, ‘metre’ relates to the expected number of bars and beats that a musical piece contains. ‘Metre’ is written at the start of a musical composition and at any point where the ‘Metre’ changes. ‘Metre’ is expressed by two numbers, one above the other; the first number being the quantity of notes within each bar and the second being the length of note.

Microphone – A device for recording analogue sound waves. The sound that is captured is converted into an electrical signal which can be recorded and processed or simply amplified for live music.

Mid Frequencies - The frequency spectrum can be divided into the three manageable bands known as ‘low’, ‘mid’ and ‘high’ (each of which is often then subdivided for additional accuracy). ‘Mid Frequencies’ refers to the middle of the three bands.

Mid Range - Is a term that is used to describe the middle frequencies within the frequency spectrum, often referred to simply as ‘mid’.

Mid/Side – Often written as ‘M/S’, is first and foremost a recording technique for producing spatial stereo recordings. However, ‘Mid/Side’ processing has become popular in recent years as more and more plugin developers add ‘Mid/Side’ manipulation tools to their designs giving the user independent control of the center channel (Mid) and the left and right channels (Side).

MIDI – Stands for ‘Musical Instrument Digital Interface’ and is an all encompassing technical ‘industry standard’ for a communications protocol between digital musical devices including all the processes and connections that are involved.

Milliseconds – A measurement of passing time used in musical production. A millisecond is a sub-division of a ‘second’ and is usually shortened to ‘ms’. There are one thousand milliseconds to a second.

Mix – Can be the act of blending multi-track recordings into a stereo two track format (or multi-channel surround sound). A ‘mix engineer’ performs a ‘mix’. Alternatively it may describe the actual article that is produced prior to the final mastering stage. A ‘mix’ of a song is sent to a ‘mastering engineer’.

Mix Engineer – Is a specialist in the ‘mixing stage’ of music production. It is a ‘mix engineer’s’ job to blend the multiple tracks of a live performance or recorded song and balance them in a professional manner in preparation for the final ‘mastering’ stage.

Mixdown – To ‘mixdown’ is virtually the same as an ‘export’, and is in fact the act of ‘exporting’ a mix to a stereo (or surround sound) format for use outside of the song’s session. The purpose of this could be to pass the mix onto a mastering engineer or to act as review copies for production staff and artists. Whereas the term ‘export’ can be applied to any single track, ‘mixdown’ is only used when a complete mix is exported.

Mixing Desk – A device for accepting multiple sound sources and blending them via channels, buses and groups to a single stereo output (or surround sound multi-channel format).

Monitor – Has multiple meanings regarding sound production. It could relate to a loudspeaker found in a recording studio; these are generally called ‘monitors’ within pro-audio situations and ‘speakers’ in consumer ‘hi-fi’ speak. Or it may mean the visual display unit that a ‘digital audio workstation’ is viewed upon. The term ‘to monitor’ is also a phrase used when listening to a sound source whilst recording.

Mono – One channel played via one loudspeaker.

MP3 – Is a coding format for digital audio that reduces the amount of data required to reproduce a song. Compression techniques are used to discard bits of data that have the least impact upon playback. When encoding audio to MP3 format, quality is sacrificed for convenience.

Mud – A term that describes the build-up of frequencies centered around 250Hz that ‘muddies’ a mix or a sound source.

Multi-effects – A type of plugin (or hardware) that offers a chain of effects within one easily manageable box of tricks.

Multi-track – Simply means multiple tracks are involved. For example a ‘multi-track’ recording is a recording that contains many separate tracks all time aligned in perfect synchronization. A ‘multi-track’ recorder is a device that is used to capture those multiple recordings.

Outboard – A collective name given to any hardware outside of a computer.

Output – The final connection within an audio device that passes the signal onto the ‘input’ of the next unit in a chain. Every ‘output’ is routed to the following ‘input’ until the destination is an ‘exported’ file or a loudspeaker. Outputs may be real or virtual and are found everywhere on channels, buses, groups, effects and processors.

Panning – To ‘pan’ a sound source is to move it around the stereo sound field using a device called a ‘pan pot’ (an abbreviation of ‘panning potentiometer’). The ‘pan pot’ is located on a mixing desk’s channel strip. ‘Panning’ is actually a form of level control. When sounds are moved to the left it is because the level feeding the right channel is being reduced. Likewise, moving sounds to the right is achieved by reducing the level of the left channel.

Parallel Compression – Is a compression technique that mixes a second ‘compressed’ signal in with the original source. parallel compression is also known as ‘New York Compression’ due to its origin.

Parametric Equalizer – Is an ‘equalizer’ that generally contains between 2 and 6 bands. It allows control over the width of the frequency range it is affecting, as well as the level. The control for width is known as ‘Q’. A parametric equalizer is also known as a ‘peaking’ or ‘bell shaped’ equalizer.

Peak – The loudest momentary level of a sound or piece of music passing through electrical equipment. A ‘Peak Meter’ is used to show the peak values.

Peak Meter - Is a measurement tool for recording the ‘peak’ level of a signal passing through audio equipment. A ‘Peak Meter’ returns the peak value at any given time.

Phantom Image – When audio is replayed via two speakers a stereo sound field is created and a phenomenon called the ‘Phantom Image’ occurs. A ‘phantom image’ happens when audio from the left speaker and the right speaker both contain identical sounds. Due to ‘psychoacoustic’ laws the listening brain will place that particular sound centrally between the speakers.

Phase – In relation to music production, ‘phase’ can be thought of as the complete cycle of one sound wave. When two or more sound waves are combined, the sound waves can be said to be ‘in phase’ or ‘out of phase’ depending on the shape and timing of the individual waveforms.

Phase Cancellation - When two sound waves are the exact opposite of each other no sound will be audible as both waves cancel each other out due to a phenomenon known as ‘phase cancellation’. This is an example of total cancellation, but ‘phase cancellation’ is happening all the time in varying degrees as multiple tracks fight against each other and produce patterns of ‘phase cancellation’ at specific frequencies.

Phaser – Is an effect that takes a duplicate signal and ‘filters’ notches in the sound wave, thus altering the shape of that wave. The altered sound is then modulated to create a sweeping effect as it is added back to the original signal.

Pink Noise - Is random noise that contains an equal amount of sound energy per octave.

Pitch – Musical ‘pitch’ is the label given to each musical note to denote how high or low each note sounds. Musical ‘pitch’ is generally written using a letter, e.g. C D E F G A B are the pitches that make up the C Major scale. However, every pitch also carries a unique frequency number. For example, the ‘A’ above ‘middle C’ by which an orchestra tunes is 440Hz.

Pitch Shift – To ‘pitch shift’ a sound, or combination of sounds, is to move the ‘pitch’ of the sound in question up or down. The degree of shifting can vary from ‘tonal’ shifts (the pitches of a musical scale), or minute shifts that our ears cannot distinguish as pitches; for example shift in pitch of only a few ‘cents’ is often used in certain type of effects plugins such as ‘harmonizers’.

Plugin – Is a collective name given to ‘effects’ and ‘processors’ that are ‘plugged in’ to the audio signal path. The name comes from the use of ‘outboard’ effects units that were indeed physically ‘plugged in’ to hardware mixing desk channels.

Polarity – Is the state of having two opposite aspects. An electrical audio signal, therefore, has ‘polarity’ (the two opposite aspects being the peak and dip of a sound wave). On a mixing desk channel strip, ‘polarity’ can be switch so that the peaks become dips and the dips become peaks. This is useful for aligning one sound wave with another and, therefore, reinforcing the sound.

Post Fader – Is a term used to describe anything that sits after a mixing desk channel ‘fader’, for example, an auxiliary send or an audio processor.

Pre-delay – Is a control generally found on a ‘reverberation’ effects unit. The ‘pre-delay’ control adds an amount of time before which the effect is heard. This is useful to add perceived separation between the dry signal and the reverberation signal.

Pre Fader – Is a term used to describe anything that sits before a mixing desk channel ‘fader’, for example, an auxiliary send or an audio processor.

Presence – Is a word that describes how certain sounds appears more forward, or ‘present’ in the stereo field than others. To give a sound more ‘presence’, therefore, is to bring it out front in the mix. One way of achieving this is to boost a certain range of frequencies, which is why the range of frequencies in question is sometimes known as the ‘presence’ range. These are centered around 4kHz or ‘octave 8’ of the musical scale.

Processor – Any plugin that ‘processes’ the complete sound, i.e. when ‘processing’ the whole of the signal passes through a processor and no dry sound remains.

Psychoacoustics – The area of science concerned with how human beings perceive and react to sound.

Q – Is a name given to the control that varies the width of a frequency band within a ‘parametric equalizer’. The wider the ‘Q’ the more frequencies selected.

Range – A word given to the number of frequencies that an individual equalizer ‘band’ is affecting. Alternatively, a singers ‘vocal range’ is the range of notes they are able to sing comfortably.

Ratio – Is a control found in dynamics processors such as ‘compressors’. The ‘ratio’ setting varies the degree of dynamic range reduction taking place; the higher the setting, the more compression. ‘Ratio’ appears as two numbers, a before and after compression figure, with each number representing a level of decibels in relationship to each other. For example, a ratio setting of 2:1 means a signal of 2 decibels before compression will be reduced to 1 decibel after compression.

Record – Could mean to capture a sound or performance onto a recording medium such as tape or hard disc. There is also a ‘record’ button to achieve this. The word ‘Record’ is also used to describe an analogue ‘vinyl’ disc, a sound recording medium that is played on a ‘record player’.

Recording Studio – A building where sound recordings are captured. The term ‘Recording Studio’ is generally applied to the large commercial studios, many of which have now closed down as musicians choose to record their own compositions in smaller ‘project studios’ at home.

Release – The decay of a sound and the final portion of an ‘attack, delay, sustain, release’ envelope found in sound design and synthesizers. Also, a ‘release’ setting can be found on many effects units where it is used to smooth the transition of a signal that moves from being processed to un-processed.

Render – To ‘render’ a record part, or a selection of many parts, is to export them to a new file. In many digital audio workstations the new ‘rendered’ file will automatically replace the old ones. ‘Rendering’ is useful for combining fragmented recordings into one easy to handle waveform or for freeing up processing power by ‘rendering’ audio with active effects plugins. The plugins can then be removed after ‘rendering’.

Reverberation – Can be thought of as the prolonging of a sound within a given environment. ‘Reverberation’ effects units simulate the effect of playing a sound within a responsive environment, with many ‘virtual’ environments to choose from.

Rhythm – When playing or creating musical notes there are choices to make, and the decisions regarding how hard a note should be played and for how long determine the ‘rhythm’ of a piece. Short and long notes mix with stressed and unstressed notes to form repeated patterns that we know as ‘rhythm’.

Roll Off – A term given to the reduction of frequencies. For example, to ‘roll off’ the low end is to reduce low frequencies.

Routing – The act of diverting a sound signal from one place to another. To ‘route’ a signal from one channel to a ‘bus’, for example, is routing. All of the signal may be sent or a signal may be split to feed multiple destinations.

Routing Grid – Found within a ‘digital audio workstation’, a ‘routing grid’ is a means of displaying as an overview, the complete ‘routing’ paths of all signals within a project.

Sample – To ‘sample’ can mean the act of ‘sampling’, e.g. the re-recording of a clip of music or a musical instrument for the purpose of playing back the sample via a ‘sampler’. Electronic musicians today often ‘sample’ other artists’ work. Alternatively, digital audio is also represented in ‘samples’, one sample being a digital snapshot of an analogue signal.

Sample Rate – The rate at which a digital device ‘samples’ per second; a measurement that is expressed in ‘kilohertz’. For example, the compact disc format plays audio at 44,100 samples per second.

Sampling – Is to record a clip of music or a musical instrument for the purpose of playing back via a ‘sampler’.

Saturation – See ‘tape saturation’.

Semitone – The smallest musical interval between two notes. To move a finger one step up or down the fret board of a guitar, or to play the next note up or down a piano keyboard is to move one semi-tone. Two semi-tones equal a tone.

Send and Return Loop – When applied to the analogue world, a ‘send and return loop’ is a form of routing. A signal is sent out of a mixing desk to be processed by outboard equipment and then returned back into the mixing desk via an unused channel or an auxiliary return. The same principle applies to virtual ‘send and return loops’ found within a ‘digital audio workstation’.

Session – The term given to a recording ‘session’. Back in the day, recording artists booked time slots in recording studios. These were called ‘recording sessions’. The word ‘session’ has been carried forward to the ‘digital audio workstation’ era with each new recording phase or project being labelled a ‘session’.

Shelf – A ‘shelf’ or ‘shelving equalizer’ is a device that increases or decreases a range of frequencies equally below or above a certain point. A ‘shelving equalizer’ will be either a ‘high’ or ‘low’ shelf as they affect only the highest and lowest areas of the frequency range.

Side-chain – Many effects require a second ‘side-chain’ signal to function properly. This second signal, not the original signal, is the one used to ‘trigger’ the effect. The ‘side-chain’ signal may be a copy of the original or a totally independent source. The act of injecting a ‘side-chain’ signal in the effect is called ‘side-chaining’ and the point that the signal arrives at is referred to as the ‘side-chain’ input. Many plugins will employ ‘side-chaining’ without the user’s knowledge as everything is handled internally.

Software – Is the name for the computer programs found within a computer; a ‘digital audio workstation’ being a piece of ‘software’. ‘Software’ interacts with ‘hardware’ (the physical devices found outside of, or integrated into, a computer).

Soundcard – A small interface that can reside within a computer.

Sound Engineer – The person who is responsible for optimizing the quality of sounds at source during a studio recording session or live performance.

Sound Field – The term given to the spatial illusion created by ‘stereo’ sound recordings and productions within the ‘stereo’ format.

Speaker – Is the part of a hi-fi system found in domestic homes that is responsible for playing recorded sound and is also known as a ‘loudspeaker’.

Spectrum Analyzer – A sound production tool used to visually display the frequency content of a sound or mix of sounds.

SPL – Stands for ‘sound pressure level’ and is the level of sound pressure measured in decibels.

SPL Meter – Is a device for measuring sound pressure levels.

Stereo – A format for reproducing sound recordings using two channels.

Submix – Is the ‘pre-mixing’ of individual sounds spread across multiple mixing desk channels. The channels in question are balanced against each other and then routed to one new ‘submix’ channel (two channels for stereo). The ‘submix’ channel is then used as a controller. This is useful, for example, when mixing a drum kit. Many microphones can be used to record a drum kit; this can become unmanageable when trying to balance the drums against the rest of the mix. The answer is to ‘submix’ them and control the level of all the drums with one fader.

Sustain – The third phase of an ‘attack, decay, sustain, release’ envelope found in sound design and ‘synthesizers’. ‘Sustain’ can be used to describe any part of a sound that retains a certain level over a given time. To ‘sustain’ a sound is to prolong its level.

Synthesizer – Often referred to simply as a ‘synth’, a ‘synthesizer’ is an electrical musical instrument used to generate sound and can be analogue or digital.

Tape – Relating to ‘analogue tape’, ‘tape’ is a sound recording and archiving medium mostly reserved for prestige productions these days, having been superseded by cheaper, more user-friendly digital solutions.

Tape Saturation – When using an ‘analogue tape recorder’ to capture a sound source, it is possible to increase the input signal to a point where ‘saturation’ of the tape is reached. Any further increase in the level past the saturation threshold will result in a squashed sound similar to ‘compression’. Recording sound to analogue tape is a creative process in itself and a skilled ‘sound engineer’ will feed the perfect amount of signal to tape to achieve ‘tape saturation’ if required. There are now many ‘tape saturation’ plugins available in an attempt to bring analogue ‘warmth’ (usually odd-harmonic distortion) to digital recording.

Template – A means of saving the complete set up of a ‘digital audio workstation’; a ‘template’ can then be recalled at a later date and used as a starting point for a new production, thus saving much time.

Tempo – The speed at which music is played. When relating to sound recording and mixing, ‘tempo’ is represented by a ‘beats per minute’ figure which can be fixed for the duration of the song, increased or decreased at certain points along the timeline or disregarded altogether.

Threshold – A control used on audio ‘processors’ to change the processing state from passive (not processing) to active (processing). A signal is ‘dry’ when its level is below a threshold and ‘wet’ as soon as it passes through the ‘threshold’ level.

Timeline – The ‘ruler’ type display found in all digital audio workstations. The ‘timeline’ starts at zero and counts upwards in time. All audio is placed along the timeline and ‘time stamped’ with a start and end time. A ‘timeline’ can also display other means of counting passing events such as beats and bars, timecode and video frames.

Tone – Controls found on amplifiers and audio equipment that are used to alter the balance of frequency bands, e.g. a bass and treble control are collectively called ‘tone’ controls. Alternatively, the term ‘tone’ is also used by musicians such as guitarists who place a great deal of emphasis on perfecting their sound or ‘tone’.

Track – To ‘track’ is to record a musical performance to a computer hard disc (or tape). Each performance is usually, but not always, recorded to a separate ‘track’ within a ‘digital audio workstation’. The term ‘track’ remains from the days when recording to tape was the ‘industry standard’. A tape recorder was known by the number of tracks it was capable of recording, e.g. a 2-, 4-, 8-, 16- or 24-track recorder.

Transient – The very first spike of sound energy produced by a musical instrument. ‘Transients’ contain a great deal of information that is used by our brains to recognize words or place sounds within an environment.

Transient Designer – Is a device that enables the user to shape the response of a sound’s ‘transients’ (see also ‘transient’).

Transistor – A device used to amplify electronic signals. These have increasingly been replaced by integrated circuits that incorporate thousands of individual transistors.

Translation – Describes how a master sounds outside of the mastering room. A master that sounds true to the original is said to translate well.

Transparent Audio – The term used to describe audio that passes through a processing device unaltered. Also used to describe music that sounds clean and clear.

Treble – The term used to describe the upper frequency range of music, usually within consumer ‘hi-fi’ speak.

Trim – Can be a name for a device that controls sound level (see also ‘gain’ and ‘gain staging’). Alternatively, to ‘trim’ an audio waveform is the term applied to the shortening of a recorded clip in length.

Valve – Is a component found in analogue equipment which subjectively colors the passing sound in a musically pleasing way. Also called a ‘vacuum tube’.

Vinyl – When relating to music, ‘vinyl’ is a term given to an analogue sound storage medium established many years ago. The ‘vinyl record’ appeared to have gone out of fashion with the emergence of the ‘compact disc’ and a certain death was predicted, but in recent times a new generation of music listeners have valued ‘vinyl’ for its ‘warm’ analogue sound and has created resurgence in the medium.

VU Meter – VU stands for ‘volume unit’. A VU Meter is a tool for measuring the volume units (signal level) in audio equipment. A VU Meter returns an averaged sound level over time, which is a more musical method than purely registering the peak value at any given time.

Warmth – Is a subjective term used to describe a certain quality of audio. ‘Warmth’ is seen to be a good attribute and is generally used when referring to analogue signals. It is thought that ‘warmth’ is a by-product of analogue components such as valves and the subtle distortion introduced by the analogue processes. Analogue ‘warmth’ is often compared to digital ‘coldness’.

Wav – Is a shortened version of ‘Waveform Audio File Format’ which is probably the most common format for storing audio on computer. A ‘wave’ file is generally referred to as a ‘wav’ due to its file extension of .wav.

Waveform – The shape or ‘form’ a sound wave takes when converted to an electrical signal, or when generated electronically. The term ‘waveform’ can also be used to describe the visual representation of recorded audio in a digital audio workstation ‘clip’.

Wet - A term used to describe a signal that has, or is being, processed by any effects or processors. Often used in describing the amount of reverb applied.

White Noise – Is random noise that contains many frequencies of equal intensity.

Workflow – Is a modern word that encapsulates the whole process of music production from start to finish, but more often than not is applied to the various stages of digital ‘in-the-box’ music production, e.g. ‘tracking workflow’, ‘mixing workflow’ or ‘mastering workflow’.