Pro Audio Mastering Made Easy:

Give Your Mix a Commercial Sounding Finish Without Buying More Gear

David S Eley

Pro Audio Mastering Made Easy

By David S Eley

Copyright © David S Eley 2013

The right of David S Eley to be identified as Author has been asserted in accordance with the Copyright, Design and Patents Act 1988.

All rights reserved. No part of this book may be reproduced, stored in or introduced into a retrieval system, or transmitted by any means without the prior written approval of both the author and publisher of this publication. This book is offered subject to the condition that it shall not be lent, sold, hired out, or otherwise circulated by any means without the publisher's prior consent, in any form or biding or cover other than that in which it is published and without the same condition being imposed on the subsequent user. This book's entire contents is protected by Copyright House Ltd.

Who this book is for:

audio engineers looking to pursue a career in audio mastering;

DIY producers wanting to know how the professionals get a commercial sound;

student audio engineers whose studies include audio mastering;

anyone recording their own music who would like to know how to prepare their recordings for radio or replication.

Introduction: How best to use this book

Thank you for purchasing my book. My name is David Eley, I am a mastering engineer who carries out a wide range of mastering projects for TGM Audio, a UK based analogue and digital mastering facility. I have been mastering since I was a teenager and I've run my own mastering business since my early twenties. I am also a professional live sound engineer and have taught Audio Operations at a college in North West England.

I have always loved the mystery surrounding the subject of mastering, how it was some 'dark art' that only the professionals knew how to do. Well this certainly isn't the case any more. Recent advances in technology mean that anyone can do a professional job of their own audio mastering – that is if you know what you're doing. Websites like YouTube, SoundCloud and ReverbNation are fuelling an explosion in the amount of DIY producers uploading their music to the internet. This is great for audio mastering engineers as it means there's plenty of work to go around, but some artists struggle to afford to pay for professional mastering every time they want to upload a new track for their fans to listen to. We're now beginning to see a growth in artists learning this final stage of the production process for themselves.

When I was teaching, I saw a problem in that many of the books and literature available on this subject were heavy going and not quite suited to the modern age of consuming information quickly. This is how I came to create MasteringTuition.com, a website totally dedicated to teaching anyone the audio mastering process. On the first page of MasteringTuition.com I pose the question 'techniques or equipment – which is more important?' I'm sure most would agree that techniques are substantially more important than the tools you choose to use. If you don't know how to get the best from what you have, there's no point in upgrading until you do. I started out with fairly basic mastering tools and spent years perfecting my techniques before I upgraded to what some might consider 'high end' mastering equipment. I set about running my own audio mastering business and made a living using standard plugins and low-end hardware. I did this by using clever techniques to get a professional sound. Customers liked what I did and stuck with me.

Over the years I have been putting together a comprehensive, yet easy to follow guide on how to achieve a commercial sounding finish using the simple plugins you have as part of your DAW (Digital Audio Workstation like Cubase, Logic or Reaper), or plugins that are free to download from the internet. The book you are reading right now is the combination of the foundation course set out on my website and the advanced techniques book which accompanies the course.

I've studied, read many books and have mastered thousands of tracks for artists all around the world. I can tell you exactly what you need to know, giving you a head start. As with my audio mastering course, this book is split into two main parts. The first part lays the foundations consisting of step by step tutorials made up of text, diagrams and videos (links to my YouTube channel). I will explain about the videos shortly. I call this part 'The Audio Mastering Blueprint'. It's a set of rules that you will find yourself sticking to almost all of the time. Here you will gain a deep understanding of why mastering is so important. By the end of this part you will be a master of your EQ, single-band compressor, multi-band compressor and limiter. You will have the skills to realise a commercial sounding finish with these tools alone. It's very important that you become a master of these four tools before you move on to the advanced techniques found in part 2. It is for this reason that I don't reveal what's in part 2 until you get there. Part 2's contents page is at the end of part 1 (there is also a global contents page at the very end of the book for easy navigation). Please don't be tempted to read ahead unless I'm covering something you already know. And even then I'd advise you to keep reading in the order I have written it, as you may discover easier ways of doing the things you already do.

I have called part 2 'The Secret Notebook of a Mastering Engineer'. Unlike the tutorial format of part 1, this second part is presented how an actual book would read. It is full of clever techniques that many top engineers are using to give their masters the edge. As the title suggests, it is very much a book of all my findings over the years from my experience of mastering tracks for people all around the world. Once you have the solid foundation gained from part 1, part 2 then explains how to make your masters sound bigger, louder, more spacious, dynamic and much more.

Due to the course like fashion of part 1, I have laid out the contents page with teaching in mind. One of the first things I learnt as a teacher was to tell the student 'what' you are going to teach before you teach it. Part 1's contents is equipped with brief pieces of information about what each chapter and subchapter is about. This will help you learn as it shows you the path before you are asked to walk it.

Important point: This book provides you with the knowledge and skill to get your masters sounding professional but there will always be more for you to learn as the industry is always changing. As a bonus for purchasing this book, I have included a link to a hidden section in my website, MasteringTuition.com. Only readers of this book will know where to find it. Here you will find articles on subjects that I may not have included, or have only touched upon in this book.

If you're reading this as an eBook, you can follow this link to go there now (I will include this link a few more times throughout the book so you don't have to visit it right now).

http://masteringtuition.com/index.php/furtherlearning

If you're reading the hard copy and have a smartphone or tablet, use the QR code above to navigate there (again, I will include this a few more times throughout). Otherwise you can simply type the url into your browser at any point you like.

To conclude the introduction:

Part 1 is tutorial based, step by step instructions with accompanying videos. Think of it like being in class (a fun class with a great teacher).

Part 2 is presented more like a book. There's loads of tips and techniques showing you how to take what you've learnt in part 1 to a higher level.

Are you ready to become a master of audio?

Praise for MasteringTuition.com

MasteringTuition.com has been featured twice in MusicTech Focus Magazine's 'Best Mastering Tuition Roundup' (volume's 3 and 4).

Feedback via email from Earle Holder, Chief Mastering Engineer at HDQTRZ Mastering Studios and creator of the Har-Bal Harmonic Equaliser:

“Hello David

I have always believed that one is never too old to learn something new. I purchased your material because it appeared you have a real passion for your craft as do I.

I thought your tutorials were well written and easily understood. The area that I read where you stated that you were mastering a project for an individual over a three year period was priceless. I was easily able to relate to your dilemma of constantly improving your craft and the need to go back and redo the previous masters because you were constantly learning and becoming  more proficient.

There is plenty of business to go around so I support my fellow mastering engineers who appear to be honorable.

Both of your books were a breath of fresh air and I will be sure to tell others who are getting started in this wonderful field to purchase your books.

Cheers

Earle Holder

Chief Mastering Engineer

HDQTRZ Mastering Studios”

Some other readers comments: “Just wanted to say thanks for a great website. Lots of great tips here and all very useful. I'm a composer/producer and always struggling with EQ, compression and mastering. I'm getting better at it and your information helps. I also teach at the ArtEZ Conservatory in the Netherlands and think my students will benefit a lot too! Thanks!” Erwin Steijlen. “What a FANTASTIC resource this has been for me and a huge help as I'm planning my first mastering session soon. I like the way you cut a wide bandwidth at about 450Hz to allow the vox to stand out more!” Phil Davies.

I invite all readers to leave feedback either on my website or on Amazon, depending on where you purchased this book. I will provide a means to do so when you reach the end of the book.

Part 1: The Audio Mastering Blueprint

Introduction to Part 1: The Audio Mastering Blueprint

The first thing I'd like to point out about Part 1 is that some of what I teach has accompanying videos that can be found on my YouTube channel. They have been made using screen capture software. There is also a downloadable piece of music for you to practice one of the techniques with. The videos will improve your learning experience, however it is not mandatory that you watch them – they are not tutorials, they are simply demonstrations of the techniques I describe in this book. Whenever you come to a video, you will see a link to my YouTube channel. If you are reading this as an eBook you can simply click the link or copy it into a browser. For those of you reading a hard copy, you can simply type the url into a browser. I highly recommend when watching the videos you have access to at least fairly good quality speakers or headphones as the audio in each video plays the most important role.

There are 8 chapters that make up Part 1 (29 when you include subchapters). They are as follows:

The Definition of Mastering

Guide to a Better Listening Environment

Calculating Standing Waves

Tonal/Spectral Processing (9 subchapters including videos)

Dynamics Processing (14 subchapters including videos)

Audio Suitability for Mastering

Live mastering session (mostly videos)

Advanced Mastering Taster

As I've mentioned, the chapters and subjects are written in a specific order, each subject follows on from the last. They all lead up to the live mastering session which is a series of videos that can be found on my YouTube channel. I advise spending some time with all the subjects before going to the live mastering session, as this will greatly assist your understanding of what is happening and the decisions being made.

All the tutorials revolve around the idea that you have a finished mix, bounced down to a stereo wav file (or similar). The wav file is dropped into the first channel of an empty project inside your DAW. Processing is then applied by inserting plugins into this channel strip.

Below is a diagram of a basic master chain. Part 1 revolves around this chain and moves on to more advanced ideas later.

There is no magic order to arrange your processors, although as you practice you will start to see patterns develop. It's largely down to what you want to achieve. I can tell you that your limiter will almost certainly be at the end of the master chain but as for other processors, you'll have to decide that for yourself once you fully understand how they work and affect each other. When you get to Part 2, you'll find a chapter dedicated to how processors affect each other as part of a working master chain.

It's important to know that the main objective of this tutorial is to focus on the techniques and methods used to get good results – the art. All the techniques discussed and demonstrated are completely transferable to any other plugin or processor. I stick to using the controls/parameters needed for mastering which are found on all processors and plugins. I personally use Logic for mastering and so most of the video demonstrations and pictures include Logic's own plugins. I demonstrate that a professional finish can be obtained using a DAW's standard plugins. This applies to any DAW, not just Logic. Or you can apply these techniques to your favourite plugins like WAVES, PSP, T-Racks, Ozone or Sonnox etc – anything.

On the next page you will find Part 1's table of contents. Please read through to the end before moving on to the first chapter. As I said, it helps to see the path to the end before you walk it.

Part 1 Table of Contents

1. Definition of Audio Mastering: Explains why mastering is so important. Begins to paint the picture of what it is we are trying to achieve when mastering music.

2. Guide to a Better Listening Environment: Provides a detailed, step by step guide for improving the accuracy of your listening environment.

3. Calculating Standing Waves: Leading from the previous chapter, this chapter explains how to calculate the frequencies at which room resonance occurs.

4. Tonal/Spectral Processing: This chapter will explore the use of EQ – a tool of great power in making a track sound pro, and will describe the things you should be listening for when using your EQ.

4.01 Introduction to Tonal/Spectral Processing using a Linear Phase EQ Plugin: A brief introduction to the EQ plugin being used for demonstration – Logics Linear Phase EQ.

4.02 EQ, Tone & The Frequency Spectrum: Explores the idea of shaping the frequency spectrum using EQ, equipping the track with a tonal balance that will translate well to different music playing systems.

4.03 EQ Parameters: A look at the various parameters found in Logics Linear Phase EQ accompanied by a video.

4.04 EQ Decisions: Looks at the various situations where EQ may be required and reveals one of the fundamental purposes for EQ in mastering.

4.05 Problem Frequencies: Explores further the various situations where EQ is necessary, introducing a clever technique for identifying problematic frequencies with a downloadable piece of music to try the technique on.

4.06 Technique for smoothening out the mix using a linear phase EQ: Step by step guide of the technique for identifying and dealing with problem frequencies.

4.07 Problem Frequencies Continued: Explores other types of problem frequencies and explains why we perceive them as problematic.

4.08 Linear Phase EQ Demonstration: Link to a video demonstration of the technique mentioned using the downloadable piece of music.

4.09 Final Word: A few brief words summing up the chapter.

5. Dynamics Processing: This chapter explores three of the most common forms of dynamic processing in mastering – single-band compression, multi-band compression and limiting. As with the previous chapters, this chapter seeks to help you realise the very purpose of the mastering process, as well as revealing some great tips when using your dynamics processors.

5.01 Introduction to Dynamics Processing: Brief introduction to compressors and limiters, discussing their role in the mastering process.

5.02 How Compressors Work: An in-depth look at the controls/parameters on a compressor, discussing the way different settings affect the sound in mastering situations.

5.03 Circuit Types: A brief explanation of some of the different types of circuits found within analogue compressors giving them their character, of which our digital counterparts are direct copies of.

5.04 Setting Up Your Compressor: A detailed guide on setting up a compressor in mastering situations.

5.05 Demonstration of Compressor Technique: Link to a video demonstration of the technique described in the preceding subchapter.

5.06 Fatness and RMS: Explains what RMS is and how it relates to loudness and fatness. Includes further discussion into compression techniques that can fatten a mix on another level, all the time helping you realise your goal when mastering your music.

5.07 Limiters: Leading on from the previous subchapter, this talks about limiters and RMS, providing some useful guidelines on using limiters in mastering situations.

5.08 Limiters & RMS: Leading on from the previous subchapter, this describes in great detail how a limiter ties in with the subject of RMS giving you a greater understanding of the very purpose of the limiter and how best to use it.

5.09 After the Limiter: Answers the question of whether you can process further after using a limiter.

5.10 Demonstration of a Limiter: Link to a video demonstration of a limiter being used in a mastering situation.

5.11 Multi-Band Compression: An introduction to a very useful tool. Discusses the major differences in operation from single-band compression.

5.12 Multi-Band Compression Situations: A look at the various situations where multi-band compression comes into play.

5.13 Demonstration of Multi-Band Compression: Link to a video demonstration of multi-band compression being used in one of the discussed situations from the previous subchapter.

5.14 Final Word: A few brief words summing up the chapter.

6. Audio Suitability for Mastering: A detailed check-list for preparing a mix for mastering.

7. Live Mastering Session: Section made up entirely of videos (links to videos on my YouTube channel). Using screen recording software, I capture the mastering of a track in full, using the techniques discussed up to this point.

Part 1 – Listen

Part 2 – EQ

Part 3 – Multi-Band Compression

Part 4 – Single-Band Compression

Part 5 – Limiting

8. Advanced Techniques Taster: 'Ultra Depth' – a step by step guide to creating a depth enhancing effect of which will be examined more closely in Part 2.

Chapter 1: Definition of Audio Mastering

The term 'mastering' as we know it today derived from an older term known as 'premastering' which is the process of preparing a finished mix to be pressed to vinyl disk. The preparation will typically involve EQ and compression aimed at balancing tone and controlling levels, equipping the track with ideal sonic characteristics for pressing to vinyl. The term 'premaster' referred to the recording immediately prior to having the 'master disk' cut from the cutting lathe. Therefore the term premaster would assume all mastering processes like EQ, compression and limiting had been applied. Nowadays the term premaster is more commonly used for a finished mix before mastering processes have been applied. A mix engineer may bounce down a finished mix to a stereo file and then refer to it as the 'premaster' as it is about to be sent off to the mastering house for further processing.

So what is the purpose of  premastering; why does further processing have to applied to a mix before it can be pressed to vinyl?

Playing a record has its physical complications. The kind of processing required is determined partly by the limitations of the consumer's player, but also by the limitations of the vinyl disk itself. For example, if there is too much happening in the extreme low frequencies, the needle can literally rattle itself out of the groove of the disk. Too much stereo information can also cause the same thing to happen.

Fortunately you don't have to worry about this. The book you are reading is aimed at preparing and enhancing a mix for digital reproduction like CD duplication, or internet download where there are fewer limitations.

Premastering is a purely technical process spanning back many years. An enormous amount of skill is required to obtain the most out of the dynamic and tonal limitations of a vinyl disk. Preparing the mix is only part of the battle, there's also the operation of the cutting lathe used to create the master disk. Nowadays, down to the current way we consume music, and thanks to advances in plugin technology, the premastering – or 'mastering' process – has become accessible to the individual artist/producer. In fact, mastering has become very much an art in itself, playing a major role in the musical attributes to making a record. Let me explain a little why the mastering process is still so important, despite not having to worry anymore about the physical complications of pressing to vinyl.

As mentioned, the limitations of the vinyl disk were only part of the reason that mastering processes like EQ and compression would need to be applied. The other reason – which is the reason that concerns you – is the limitations of the consumers player, and the environment the music is being played in.

Think about when you're recording your own music, producing for a band or mixing for a friend. The likelihood is that your listening environment is free of noise and distraction, and your speakers are of at least fairly good quality. The problem is that when the recording reaches its audience, the music player and listening environment are likely to be very different compared to that of when it was produced, which can dramatically change how the mix sounds. Did you know that your environment may be leaving an impression in your mix? And so might your good quality speakers. The trouble is that you would never notice this fact until you play your mixes on another, perhaps inferior system somewhere else. When I was a kid, I used to take my own recordings to my friend's house and play them on his middle of the road hi-fi, but they never sounded as good as they did at home. They were un-clear and sounded small, yet when playing a commercial CD, everything sounded big and crystal clear, at much lower volumes. This is because the music on the commercial CD had undergone professional mastering making it sound its very best on that particular hi-fi and all other music players too.

Tone

Even fairly expensive speakers are not always one hundred percent true. By 'true' I mean reproduce every frequency at the same level. If your speakers are a little bass heavy, you will naturally compensate in your mixing, and vice versa. It goes without saying that the flatter your speaker's frequency response, the more suitable they are for mixing or mastering. In most cases, a 'studio monitor' will be designed to have a relatively flat frequency response. However, what's of more concern is how your room itself can alter the tone of what you are hearing, especially in the low frequencies. One of the targets of the mastering engineer is to correct the tonal imbalances derived from an imperfect listening environment. For this reason, it is of extreme importance that your listening environment be as neutral as you can get it. The next chapter contains a guide on how to do just that. Your listening environment is a crucial factor in creating a well balanced master.

Loudness

Why did I need to turn my mixes up so loud in order to hear all the details?

My listening environment at home where I produced the music was free from any noise or distraction – I could hear the detail just fine there. At my friends, there might have been people talking, moving around, playing computer games and who knows what. The subtle quieter details would be lost unless I cranked it up. Mastering combats this problem. By the use of dynamics processors like compressors and limiters, mastering is able to surface every little detail – the quiet bits become louder but don't sound like they have been turned up. Achieving this to the greatest effect is very much an art of mastering. Apart from background noise, you're also up against the quality of the speakers the music is being played on. My friend's hi-fi wasn't bad, you could hear the kick drum and bass line at least. But what if I played my recordings on some laptop speakers? They wouldn't have stood a chance.

Mastering equips a finished mix with the power to sound correct on whatever system it is played on. It does this by manipulating the dynamic and tonal content. At this point I won't go into detail on how this is done, all will be explained as you proceed through the book. I will just say now that by the clever use of compression and EQ, an entire mix can be distinguished on a speaker that only reproduces around 75% of the frequencies inherent in the mix.

Just mentioned are the more practical reasons for mastering. There are many more exciting reasons which are of a more musical nature – things like fattening, adding sparkle, creating loudness, tightening a loose bass-line or adding extra depth and space. As you practice the techniques in this book, you will realise the immense power you have in creating a more exciting musical experience.

Before we start looking at the techniques and methods, it's important to ensure our listening environment is as neutral as possible. The next chapter will explore this further.

Chapter 2: Guide to a Better Listening Environment

Studio acoustics is a large and rather complicated subject but you don't need to be an expert to do a good job of your mixing or mastering. I'm going to go through the main points and show you how to easily and quickly achieve a professional listening environment.

When mastering, it is vital that we are able to accurately monitor the sounds that we are working on. A major part of this is the build quality of the speakers and how flat – and therefore accurate – their frequency response is. However, an occasionally overlooked yet vital element is the design of the room within which we’re working. A fantastic monitor speaker response would be practically wasted in a room which unduly coloured the sound they produced. As an extreme example, imagine setting up your speakers at one end of a tunnel, then trying to produce an accurate mix from the other. As you can imagine, the physical manipulations of the tunnel will warp the direct speaker sound, completely changing the way we perceive it. Although less obvious, an unsuitable mastering environment can also be quite harmful to the end result. Depending on its size, shape and reflective characteristics, certain frequencies – or bands of frequencies – can be perceived at different levels from others. If the room happens to reflect more higher frequencies than others, then the sound you are hearing may appear brighter than it really is. If the room exaggerates the low end, then the speaker sound will appear to have more bass than is actually in the mix. Similar to how the speakers have a 'frequency response', the room's response to different frequencies will vary also. The more 'neutral' our room's response, the more accurate the sound reaching our ears will be. So before we discuss factors like monitor placement and acoustic treatment, let's first look at one of the most fundamental aspects which influences the way we perceive sound within the room – the size, shape and raw surface materials of the room.

Size and Shape

The worst case scenario for a room would be a perfect cube with flat walls, floor and ceiling. Apart from the walls, floor and ceiling allowing the sound waves to reflect around the room, changing the way we perceive the sound (which will be discussed shortly), the major problem is the cube shape itself as it causes a specific set of frequencies to appear to be louder compared to the speaker sound, particularly in the lower frequencies. This is due to a phenomenon known as 'standing waves' – sound waves that literally become trapped between parallel surfaces such as walls, bouncing back and forth, overlapping with each other perfectly, reinforcing themselves every time. Rooms with parallel walls will 'resonate' at certain frequencies. The frequency at which resonance occurs is referred to as the room's 'mode' and is directly related to the distance between the walls (a sound wave's frequency is directly related to its physical length in metres).

No matter what distance the parallel walls are apart, a sound wave of some frequency will be able to fit exactly into that space, then flip over when it hits a wall and go back the other way, perfectly in time with its immediate following sound wave (at that same frequency). Theory states that when you allow two identical sound waves to add together, they double in size.

Interestingly, half that frequency can do the same thing – it hits the wall, flips, then goes back, meaning it fits itself in between the walls after two trips. This adding also occurs when the frequency is doubled, quadrupled, octupled and so on, as they all still fit perfectly into that space.

When this reinforcement happens, a doubling of sound pressure will occur for that particular frequency, which can work out to a 3dB rise in audible level in the main area of the room. Quite a dramatic difference where audio accuracy is concerned.

What you will also find is that depending on where you are sat, you will experience differences in the way the room resonates. Two identical waves overlapping will increase the audible level of that frequency. However when two waves overlap whose wave cycles are opposite (anti-phase), a cancelation will occur giving the perception of a reduction in that particular frequency. The overlapping of identical waves is known as 'constructive interference', whereas the overlapping of anti-phase waves is known as 'destructive interference'. Some areas of the room may cause constructive interference, while other areas may cause destructive interference.

Where possible, choose a room that deviates from a cube, ideally one that is fairly symmetrical, with slopes, uneven features and details that help to redirect and diffuse reflections and lessen the impact of resonances. If this is not possible, don't worry as I'll be showing you how to lessen the impact of resonances using other methods.

Golden Ratios

Some of you may have heard of the term 'golden ratio' when referring to studio design. This is the ratio between the height, width and length of a room. If you are lucky enough to be in a position to 'choose' the dimensions of your room, then you may consider using the golden ratios to do so. Originating from ancient Greece, these ratios have been applied in many subjects and practices; from architecture to classical music, and even to book design. They also occur frequently in nature which is how they were first discovered. In regards to studio design, the golden ratios are a proven way to obtain a more accurate listening space as they allow for a uniform distribution of resonant frequencies around the room (yes, resonance still occurs). As for how they are calculated, the level of mathematics is probably too advanced for this book. In any case, the purpose of this book is to show you how to achieve a professional finish by utilising what you already have. So rather than have you build an extension on the side of your house, let's move on to how you can transform an existing room into a reliable listening environment.

Surface Material

A surface material will have different reflective characteristics depending on the frequency of the sound wave hitting it. A soft surface material such as carpet will absorb much of the higher frequencies preventing them from being reflected back into the room. However, a low frequency will pass straight through the carpet to the reflective solid surface beneath. A heavy drape may provide control of midrange and higher frequencies but again, the lower frequencies find their way through. Controlling the lower frequencies requires more elaborate efforts than the use of things like drapes, or carpet; their long wave lengths require the use of much larger objects.

Just before we reach the discussion on how to control these troublesome reflections, I'd like to point out one more issue you will face – the 'hanging around' effect of the lower frequencies caused by the solid surfaces beneath the soft furnishings reflecting the sound waves back into the room, and the resonating 'trapped' lower frequencies which subsequently take longer to disperse their energy. The result is a lack of definition in the low end as the sonic information literally starts to smear.

So we know that the room's reflections alter the perceived speaker sound. Should we attempt to 'remove' the reflections from the room? Fortunately it's okay to have some reflections in the room, we just need to 'neutralise' them the best we can so we can trust the speaker's sound.

Room Contents

Before you begin budgeting for acoustic treatment, there's a lot that can be done using the everyday things around you. As well as audio gear, the room will most likely contain some furnishings which will aid the absorption of reflections, as well as irregular surfaces that serve to break up and scatter incoming sound waves, preventing them from building up and appearing louder in certain areas of the room. As mentioned, lower frequencies are more of a problem as they can travel straight through thin coverings like curtains and carpets which are effective against high frequencies. Large furniture with soft surfaces such as beds, sofas and padded chairs all soak up a certain amount of the lower frequencies because their material is porous but quite dense, so although the powerful low frequency energy is able to enter the material as vibration, a lot of the energy is dispersed as friction and heat. This is similar to how professional 'bass traps' work – the vibrations cause the fibres inside the dense mineral wool filling to rub together converting kinetic energy into heat.

Half-full bookshelves, angled sofas, a laden coat stand, an open wardrobe, anything that is either absorptive or uneven – and may help to divert and fragment the direct speaker sound – is a worthwhile consideration.

Acoustic Treatment

You may consider the use of acoustic treatment. Wall-mounted tiles made of acoustic foam placed strategically around the monitoring position is a proven way to achieve a professional monitoring environment. It is also worth considering mounting bass traps around the room and in upper room corners where bass frequencies tend to build up. These can be made from mineral wool (as mentioned) or acoustic foam. Another effective form of acoustic treatment can be the use of 'diffusers' – precisely shaped solid objects designed to break up desired frequencies, stopping them from building up. It's likely you might require a combination of all three.

Acoustic tiles:

These absorb the reflections of high mid to high frequencies.

Bass traps:

These absorb the low mid and low frequencies.

Diffusers:

These are shaped to fragment reflections. The one pictured is known as a 'Quadratic-residue Diffuser', or a 'Schroeder Diffuser' named after Manfred Schroeder who developed the formulae for calculating its dimensions. Their size and dimensions can be tailored to deal with certain frequencies.

Before we look at where is best to place the furniture/acoustic treatment around the room, let's first look at speaker placement with regards to the room's surfaces.

Speaker Placement

An important factor you need to be aware of is the time between the direct speaker sound and the first reflections from the walls. When this time is too short, the reflections colour the direct speaker sound through a process known as 'comb filtering'. This is when the interaction of the direct sound waves with the slightly 'out of phase' reflected sound waves causes an undesirable filtering effect extending up the frequency spectrum. Increasing the initial delay of the first reflection lessens the impact of comb filtering. So it's okay to have 'some' reflections; the key thing is to try our best to prevent them from reaching our ears too early, and that there isn't an imbalance of the 'tone' of the reflections, such as too much low frequency compared to the rest.

The walls directly behind and beside the speakers are likely to cause issues as the reflections off these will be very short. Also, flat surfaces in front of the speakers like a table or a mixing desk can introduce reflections with very short delay times. Depending on the size of the room, there's a good chance that the reflections off the walls either side of the listening position will be troublesome too. I will illustrate this shortly.

Putting the discussion on reflections to one side, another important aspect to consider with speaker placement is the distance between the speakers and the distance from your ears. This is essentially what creates the 'sound stage'. You will most likely be dealing with stereo content – some sound will come from the left, some from the right and some from the middle. The balance between these three areas is what you must get right. If the distance between your speakers is much greater than the distance to your ears, the 'fantom centre image' can become weak and the 'stereo width' too wide – as if you're wearing headphones. Drawing the speakers closer together, shorter than the distance to your ears, the fantom centre can become too strong and the stereo width too narrow. A tried and trusted way to arrive at suitable speaker locations, with respect to the listening position, is to create an 'equilateral triangle' between your ears and the speakers. This is known as 'true stereo' or 'true 60 degrees' (the angles within an equilateral triangle are each 60 degrees).

Recap

aim to neutralise the sound of the room – balance the tone of the reflections;

parallel walls cause standing waves – the room will resonate at certain frequencies;

different surface materials have different reflective properties;

everyday room contents can be used as acoustic treatment when placed correctly;

direct speaker sound is coloured by the immediate reflections from the nearby walls;

the effectiveness of the perceived sound stage is a result of the distance between the two speakers and your ears.

Putting It All Together

Studio design is a large and tricky subject; there are many different ways of looking at how one can be designed. As mentioned, the aim of this book is not to show you how to build a state of the art mastering studio, but how to achieve a professional sound using mostly what you already have. Assuming you only have access to a rectangular room, as most people reading this book will, here's a simple way you can achieve a professional listening environment without breaking the bank. This method is loosely based on a concept by Don Davis of Synergetic Audio Concepts where you have a 'dead end' to the room and a 'live end' to the room.

This first diagram gives a rough idea of how to arrange the furniture/treatment within your room:

Notice the arrangement of the speakers and the space behind the listening position:

Using this arrangement of furniture/treatment will help neutralise the sound of the room, thus allowing you to have more trust in what your monitors are telling you:

Some more tips:

Use a mirror to determine which areas may require acoustic tiles – if you can see your speakers in the mirror whilst facing forward, you may need to treat those surfaces.

Don't cover all your walls in carpet (it's amazing how many times I have seen this). You have little chance of ever stopping the low frequencies from bouncing around anyway, so you should allow for an amount of reflections across the rest of the spectrum in order to balance the tone of 'all' the reflections.

Shock mount monitors to prevent vibrations travelling through the materials of the room, which can reach your ears as sound before the direct speaker sound!

Also consider mounting absorption/diffusion 'above and below' the listening position.

The larger the room, the better. Low frequency sound waves need space to develop down to their long wavelengths. A tiny room will reduce your accuracy in monitoring the low end.

For those of you who would like to delve a little deeper, the next chapter explains how to calculate the frequencies which will resonate with respect to the distance between the walls. If you're happy with your listening environment and want to move on to the techniques, feel free to skip ahead to Chapter 4.

Chapter 3: Calculating Standing Waves

As we've discussed, whenever mastering (or carrying out other forms of music production) we’re at the whim of the acoustical effects of the walls, floor and ceiling. Left unchecked, these surfaces reflect sounds back into the room and towards our ears. These reflections cause us to hear additional information that may lead to a misinterpretation of the direct speaker sound because of their interaction with each other, the direct speaker sound and the dimensions of the room.

Sound waves travel through the air in alternating stages of high pressure and low pressure. Once a sound wave has begun travelling through the air (and let’s imagine a simple sine wave for this), it reaches its peak power of high pressure, dips to a pressure of zero then begins a period of low pressure, which again returns to zero (back to the starting point). The distance across the air from the sound source that it takes the wave to complete this full cycle is known as the wavelength. Low frequencies have a longer wavelength than high frequencies.

It’s known that the speed of sound through air at room temperature is 344 metres per second. The number of full wavelength cycles that can happen per second at a given frequency (or put another way, how frequently the full wave occurs per second) is measured in Hertz (Hz). We’re able to measure the dimensions of a room in metres, and then calculate the frequencies that may be resonant, causing them to sound louder than they should.

A resonant frequency (in terms of this guide) is a frequency that is amplified by the dimensions of a room. Let’s imagine we have two flat parallel walls that are 4 metres apart. Any distance, including 4 metres, must be able to perfectly contain a sound wave frequency. This is the formula to discover which particular frequency it is:

Speed of sound (in metres per second) / distance between walls (in metres) = frequency (in Hertz)

So for the above scenario: 344m/s ÷ 4m = 86 Hz

This tells us that one full wavelength at 86 Hz exists between our two walls, bouncing back and forth and reinforcing the energy from the speakers more so than other frequencies and making it around 3dB louder than it should ideally sound. However, this isn’t the full story. As well as 86 Hz, half of this measured frequency is also able to reinforce, as will doubling 86 Hz, quadrupling, and so on. This means that as well as 86 Hz, we will be incorrectly hearing 43 Hz (half the frequency, twice the wavelength), 172 Hz (twice the frequency, half the wavelength), 344 Hz (four times the initial frequency, quarter the wavelength), and upwards.

Chapter 4: Tonal/Spectral Processing

In this chapter:

Introduction to Tonal/Spectral Processing using a Linear Phase EQ Plugin: A brief introduction to the EQ plugin being used for demonstration – Logic's Linear Phase EQ.

EQ, Tone and the Frequency Spectrum: Explores the idea of shaping the mix using EQ, equipping the track with a tonal balance that will translate well to different music playing systems.

EQ Parameters: A look at the various parameters found in Logics Linear Phase EQ accompanied by a video.

EQ Decisions: Looks at the various situations where EQ may be required and reveals one of the fundamental purposes for EQ in mastering.

Problem Frequencies: Explores further the various situations where EQ is necessary and introduces a clever technique for identifying problematic frequencies with a downloadable track to try the technique on.

Technique for smoothening out the mix using a linear phase EQ: Step by step guide of the technique for identifying and dealing with problem frequencies.

Problem Frequencies Continued: A discussion into the other types of problem frequencies and an explanation about why we perceive them as problematic.

Linear Phase EQ Demonstration: Link to a video demonstration of the technique mentioned using the downloadable track.

Final Word: A few brief words summing up the chapter.

Introduction to Tonal/Spectral Processing Using a Linear Phase EQ Plugin

The chapter 'Definition of Mastering' should have given you a rough idea about what we are trying to achieve when mastering music. The next step is to explore the idea of shaping a track from a spectral point of view. Throughout this chapter you will realise some of the very purposes of the mastering process. You'll be presented with an idea of where to start, and you'll begin to clearly imagine the final sound of which you are trying to achieve when mastering your own, or other artist's music.

The process of tonal adjustment is mostly achieved using an EQ, something you're probably familiar with already. EQ is possibly the most important tool in the box. Whether you mix or master, operate live sound or configure PA systems, EQ is an absolute necessity to do your job. As you can probably guess there are many different types – parametric, graphic, linear phase, passive and active. Essentially they all have the same goal of attenuating or boosting a specific band of frequencies chosen by the user.

Below is an image of the EQ we shall use for demonstration – Logic's own Linear Phase EQ. Please remember that these tutorials are transferable to any hardware or software processor. My DAW of choice is Logic and so many of the techniques I talk about will be aided by illustrations and demonstrations using Logic's own plugins. In terms of this book, they are no different to other plugins. Getting a commercial sounding finish is all about technique. I stick to using common parameters found on all plugins.

Side note: An explanation of the term 'linear phase' is probably a little beyond the level of the mastering tutorial at this early stage. For now, let me just say that adjusting the controls on any EQ will project an artefact across the rest of the spectrum and can be perceived in the form of distortion. In simple terms, linear phase means the process that causes this artefact has been adjusted to generate less distortion, making it more suitable for mastering.

I will assume that you are familiar with what the frequency spectrum is, and have at least some experience of altering the tone of a sound using EQ.

The controls from one equaliser to another do not vary much. You have a choice of bands, each set at a different frequency. For each you can adjust the gain or slope, the frequency and the Q. The gain will boost or attenuate the selected frequency. The frequency is pretty obvious, it's the point where all the action is happening. The Q can be used to narrow and widen, or to shape the affected area of the spectrum. Q actually stands for 'quality factor' and so it follows that the higher the quality, the more precise the affected area of the spectrum will be. The slope will adjust the gradient at which the gain rolls off when selecting either the 'shelf' or 'hi/low cut' option (this will be illustrated further on).

EQ, Tone and the Frequency Spectrum

As I've mentioned, EQ is possibly the most important tool you are ever likely to use. It is also one of the hardest to grasp. Even the slightest adjustment can have a detrimental impact on the whole mix. The skill of detecting what is required using EQ will come as you practice the techniques found in this book.

EQ in mastering has more than one use. Generally it is used to bring tonal balance to a mix, although it can surgically correct problem frequencies like resonance. It can also play a musical role by exposing a hidden gem in a mix, like the lush high mid of a synth pad, or the meaty low mid of a lead guitar. The trick is to know what you are looking for.

In most cases the fullest, or richest sounding music tends to have an even spread across most of the frequency range. This is very typical of pop music. In fact, the actual frequency curve for commercial pop, when viewed on a frequency analyser, can appear to take the shape of a bell curve.

Have a look at the following two images, one shows peaks and one shows RMS. These are two snap shots taken from my analyser when playing a typical commercial pop song in Logic:

Side note: Peak and RMS are explained further in the Dynamics Processing chapters, for now let me just say that peak is the highest level that the audio wave reaches, and RMS is an average of the various levels making up the waveforms. The RMS level is closely proportional to the ear's perception of loudness. For this reason you will probably find yourself monitoring RMS levels more closely than you will peak levels. On the Level meters to the right of each analyser plugin, the dark shade indicates the current peak level and the light shade indicates the current RMS level. 

It's a good idea to have a frequency analyser plugin (or a 'multi-meter' as the one I am using is called) inserted after all other processing as an aid to help you with what you are listening for. Voxengo offer a fantastic one for free called 'Span', see below:

Span is available to download from this address: http://www.voxengo.com/product/span 

Can you see what is meant by the bell curve?

The analyser in the snapshot above is showing an even spread of frequencies across the spectrum. When a mix has a rather low amount of mid-range frequencies compared to the highs and lows, it can sound thin, flat or hollow to the ear. On the analyser it may actually look as though there is a hole in the mix. EQ can help, to an extent. You can plug the hole with a gentle EQ lift in the specified area with a medium to low Q setting, producing a nice, smooth curve. Boost too far and the mix will sound un-natural. This leads to an important point: EQ in mastering must be used with great care. Large adjustments can sound harsh and damaging, especially when boosting an area of frequencies. Did you know that hearing a reduction with EQ can be somewhat more pleasing to the ear than hearing a boost? If you want to hear an increase in one group of frequencies, it can sometimes be more effective to reduce the frequencies around it. It's like a balancing act. If you reduce the lows, the mix can sound like you have turned up the mids and highs, reduce just the highs and the mix can appear to have more mid and low end. When using standard EQ plugins, it's good to try and stick to using reductions if you can as they perform a reduction more easily, thus preserving sonic quality. The higher the quality of the EQ, the better it will be at performing a boost.

The job of creating this even spread of frequencies that the human brain is so fond of is, in all fairness, largely down to the mixing engineer. The mastering process should be seen as more of a time to fine tweak an already good balance across the spectrum.

If, after looking at the two analysers again, I were to ask you to guess where the sonic energy was greatest, the low, the mid or the high frequencies, what would your answer be?

I think it's clear to see that the most is happening in the mids. This has a great advantage when it comes to being played on a typical hi-fi or kitchen radio. Here's why:

The frequency response of the consumer's music player is a large factor. Your finished master needs to still sound full and exciting when it is squeezed through a cheap kitchen radio's frequency response of about 100Hz to 14kHz. For this to happen, the mids need to sound clear and defined with plenty of energy. A big hole in the midrange of a mix will be a lot more damaging when played on such a system as it's mostly midrange that the speaker can reproduce. Let's be honest, the consumer is not likely to have a set of studio quality monitors set up in their kitchen to listen to whilst they are cooking.

A small kitchen radio will struggle to produce low frequencies.

The extreme highs will struggle too.

This leaves mostly midrange to get your point across. An even bigger challenge are laptop and smart phone speakers as these struggle to produce the lower mids too.

Side note: Getting clear and defined mids must be accomplished during the mixing stages. Mixing is subject in itself and a large one at that, (discussed in part 2) but this really needs to be said right now. A commercial sound is a joint effort between the recording, mixing and mastering engineer, which in many cases is the same person!

We all love to hear a powerful bass-line and a sparkly top end, but it's the mids that have the biggest responsibility. Have a look at the following three diagrams illustrating how an even curve, and an un-even curve translates to a set of studio quality monitors compared with a small kitchen radio:

The first illustration shows how the reproduction capabilities of a kitchen radio compares to a set of studio quality monitors. The audio displayed on the analyser has good midrange content, meaning the kitchen radio has a rich source of audio of which it can reproduce.

Let us see how the same two sets of speakers deal with poor midrange content. The next illustration shows how the studio monitors would be forgiving in such a scenario as they are easily able to reproduce the lows and highs.

However, the kitchen radio will not be so forgiving. In this scenario, much of the audio will be lost resulting in a thin and rather unprofessional sound.

Side note: I have exaggerated the two illustrations with low midrange content to help portray my point. The analyser is an 'aid' to what you are hearing; it should not be relied upon as a way to make decisions.

I hope that you can now see how the tonal balance should not just be a matter of taste, but also a step towards equipping the track with the elements required to sound great on a wide variety of audio systems.

EQ Parameters

If you're already familiar with all the parameters on your EQ, feel free to skip to the next subchapter.

On Logic's Linear Phase EQ there are buttons at the top with a little symbol in each. These buttons activate, or deactivate that band of frequencies. The symbol illustrates the shape in which the EQ cuts or boosts. They are fairly self-explanatory, the way the symbol looks is a reflection of how it will look on the EQ when you adjust that particular band. This is a link to a video demonstrating the controls on Logic's Linear Phase EQ and follows the explanation written beneath – the video does not have any audio:

http://youtu.be/a9_Uvrk4m7U

The first symbol is known as a 'low cut'. This band cuts everything below the chosen frequency. For this band, the Q will adjust the shape and gradient of the cut. Unlike most of the other bands, this first adjuster has a 'slope' as opposed to 'gain'. This is measured in decibels per octave and will also adjust the shape and gradient of the cut. Taking Q up past 1 starts to create a lift at the selected frequency. Adjustment of the slope can give you a long drawn out curve, or a steep drop.

The next symbol is known as a 'shelf'. Again, the way it looks is a reflection on how the EQ will look when using this control. The gain adjuster on this one will create a slope leading to a shelf like shape at the selected frequency which will extend down to the bottom of the spectrum, or until it is affected by another band. Like the first band (low cut), the Q will give you a change in the gradient and shape of the slope.

The next group of symbols are known as contours (sometimes referred to as 'bell'). You may agree they are simpler in the way they work. The gain boosts or attenuates the frequency selected in that band. The Q adjusts the width of the band. A low Q setting will equal a wide band width.

Toward the end you'll find another shelf, and then a 'high cut'. They are precisely the same as the first two, only they are facing the opposite way. The high cut will attenuate everything above the chosen frequency. The gradient and shape are dependent on the slope and Q settings.

EQ Decisions

Looking at the first EQ band – the low cut, why might you want to cut everything below a certain frequency? In mixing, you would low cut most channels except the kick and bass, but what about in mastering? We know the final master may be played on a crummy kitchen radio which won't produce the extreme lows anyway, but it may also be played on a very expensive hi-fi with a frequency response as wide as you need. One reason for this is that during the mixing stages the monitors used may only extend down to about 60Hz. Anything happening below that may have gone un-noticed. That's not so bad if the only thing down there is the extended bottom of the kick. But having reached the mastering stages, some unwanted rumble may be happening right down in the 30 to 40 Hz area. Only full range speakers, or a sub woofer will pick this up.

The speakers here at TGM Audio extend down to a useable 25Hz so we can spot anything that may have been missed. If your speakers will not produce frequencies so low, you can use headphones for this. A £150 pair of headphones will probably extend down as low as you need. If you have no way of telling what is happening down there, it might be a good idea to apply at least some kind of low cut. Most commercial pop records have little going on below 30Hz anyway. You may get some idea by the use of your analyser. A cut at 30Hz might give you peace of mind.

Low cutting some bottom end is also a way to concentrate more energy into the important midrange areas by use of compression and limiting (how this happens will become apparent soon). That's not to say you should just cut the low off everything – as I mentioned before, the extended bottom end of the kick might be down there, which on a quality set of speakers will sound very nice.

Side note: You will learn about how this energy is concentrated into the mids when you get to the dynamics processing chapters. You will also discover how having a kick with an extended bottom end can actually help it translate to a little kitchen radio/laptop speaker even though these extreme low frequencies will be lost.

The lows play an important role. What we perceive as 'warmth' is mostly the work of the lows and low mids so a low cut too far may take the warmth away. The following chart illustrates how the ear perceives various textures according to how the audio's tone is balanced:

If the track in question appears to have low or excessive amounts of any of the above, it may be possible to correct the balance by reducing or lifting the specified area of the spectrum, or the surrounding areas instead.

Looking at the last EQ band – the high cut, why would we want to cut away any of the highs? As said before about the balancing act, a reduction in the highs can give the perception of a gain elsewhere. In some cases this might add to the warmth, or it could give a more rounded sound. It may also allow for a little more energy to be concentrated into the mids (to be explained soon). On the other hand, the extreme highs may play an important part by providing air or sparkle.

The following two paragraphs are of utmost importance – please read carefully.

I mentioned earlier about the frequency response of a crummy kitchen radio or a set of laptop speakers. However I only stated where the frequencies roll off at the bottom, and where they roll off at the top – somewhere around 100Hz to 14kHz. But there is something to be said about the frequencies in between. Rarely are they reproduced evenly on any music playing system. The room acoustics can also contribute to an un-even frequency response at the final listening stage. Some hi-fi's have a built in EQ allowing the user to tweak the tone to his or her taste. But you need to be aware that the consumer's hi-fi or environment may well be about to boost, or reduce a group of frequencies and there is nothing you can do about it. I hope you can now see that the best fighting chance a master has to sound correct on different systems, and in different environments, from a tonal point of view, is to have an even spread of frequencies across the whole spectrum from the beginning.

An artist may love bass, a lot of people do, and the track may sound like it should contain a fair amount (think about reggae). But to produce a master with a heavy bottom end can be a risky business, knowing that the consumer's hi-fi, and room acoustics, by chance, might already be boosting the low frequencies. You will soon discover how creating a fat bottom end can be achieved by use of compression and limiting without actually raising the volume of the lows compared to the rest of the mix.

Problem Frequencies

The mix is usually made up of quite a number of different elements, each spanning its own individual group of frequencies. Not surprisingly, almost all these frequencies cross the midrange areas, even the kick drum can go right up into the high mids. When you consider the reverbs, delays and any other FX, you can appreciate that the midrange can become quite busy, which is one of the reasons why it is so important that good definition has been created here during the mixing stages.

There is a fantastic hand drawn chart illustrating the spanning ranges of various classical musical instruments called the 'Carnegie Hall Chart' which dates back to 1941. Mastering Engineer 'Bob Katz' has made it available to purchase from his website found here:

https://www.digido.com/products/carnegie-chart.html

The busyness of the midrange areas can contribute to an uneven build-up of layered sounds across the spectrum. Where there is a lot happening may cause an audible lift in that particular area, or it may sound like resonance. These areas, or specific frequencies can be problematic and can be known as 'problem frequencies'. Where there is little happening, you may notice a dip, or a 'hole'. Even the instruments themselves are not likely to produce each frequency at the same level, or there would be little expression in the way the musician plays. You can start to understand why the whole spectrum can end up being a little un-even when we get to the mastering stages.

We know that good translation to the consumer's hi-fi and environment has the best chance with a relatively even spread across the spectrum, but our goal isn't just to level it out like a bulldozer levelling an old garage. We have to have a clear view of the artist's or producer's intentions. The real art of EQ is very much being able to define the intentions of the artist or producer, then make an informed decision whilst being aware of the limitations, variations and imperfections of the consumer's player and environment. The track could be rich with varying tones and changing levels, expressing beautifully the feelings of the musicians. Levelling it all out might be damaging to the expressive nature of the track. On the other hand, what would be the point of playing the track if it can't be enjoyed, or even heard down to the challenging elements of the consumer's player and environment? We don't all sit in a quiet, neutral room and listen to music on good quality speakers like we do when we are recording and mixing.

I am going to show you a clever technique on how to obtain a more even frequency spread across the spectrum. This will help you distinguish the problematic areas. I'll also share with you a little tip in deciding whether these problem frequencies should be reduced or not, or by how much. This technique is very simple, anyone can do it. Although how easily you find it will depend a great deal on the functionality of your EQ. The next subchapter contains a step by step guide on how to apply this technique. You will also find a link to a video. The video is simply an aid to the written guide and has no audio.

Below you will find a link to a downloadable piece of music that is in need of some EQ. Having read the guide and watched the video in the next subchapter, try the technique using the downloadable track. In a short while you will arrive at a link to another video which demonstrates this technique using the downloadable piece of music, revealing where the adjustments needed to go.

Download the track from here:

http://www.masteringtuition.com/index.php/practice-music/category/1-eq-demo-track

Technique for smoothening out the mix using a linear phase EQ

Begin by listening to your mix with your eyes closed. See if you can distinguish where there may be resonance or problem frequencies in the spectrum. Then use the analyser to see if your suspicions show up visually. They won't always. The analyser can lend a helping hand but your ears are the best frequency analyser you will ever possess.

Below is a step by step guide to finding and dealing with problem frequencies, and with it a link to a video showing how to do it using Logic's Linear Phase EQ. As mentioned, this video has no audio – it simply follows the step by step guide below. Try this technique using the downloadable piece of music. When you arrive at the link to the video of this technique being applied, you will be able to see if you found the same areas to be problematic.

In the suspected area:

1. Choose a frequency band.

2. Boost the gain of the frequency band by a good few dB – you'll hear the lift in an area of the mix

3. Perhaps narrow the affected area a little using the the Q parameter.

4. Sweep the frequency up and down and listen closely.

5. You may notice that as you sweep across a certain area, you'll hear a slight resonance, or hear it become noticeably louder.

6. Narrow the EQ band some more and sweep again, focusing in more on the suspected area.

7. Consider applying a small notch like reduction at the point of the resonating frequency.

The video aid to this step by step guide can be found here:

http://youtu.be/5O0I30lhmwY

What you uncover with this technique may well be problem frequencies and the fate of the final listening experience depends on whether you decide to reduce them or not.

One thing to listen out for is the way the resonance, or increased level sounds. You may be sweeping across the resonating frequency of a single musical instrument, in which case the resonance will have a pure tone and will be in keeping with the particular chord playing at that time, or the key the music is in. It's fair to say that the more musical ear will be better at detecting the relationship between the resonance and the chord/key of the music. But there is another kind of problem frequency which usually requires more attention. Find out what is is in the next subchapter.

Problem Frequencies Continued

The resonating tones you have uncovered may well be problem frequencies when you consider the uncertainty of the consumer's player and environment. As mentioned, you should consider applying a small, notch like reduction at these areas.

A pure tone resonance found at the mastering stage could be caused by a bass amp reproducing one frequency louder than the rest. Perhaps the room the bass amp is in might be causing resonance. Or it could be down to microphone response or placement, picking up a certain frequency better than another. In a perfect world, all these instances would have been dealt with during the recording and mixing stages using channel EQ and compression (or better still, prevented in the first place with good recording techniques). But you must be prepared to deal with such nuisances during the mastering stages.

That aside, these 'pure' tone resonances are not always considered the most problematic. Covered next are the kinds of resonance we find more displeasing to the ear. It is these areas that require more attention.

Some resonant areas will appear to be an un-defined mixture of sounds.

In the true sense of the word these areas are not really resonance as they can be made up of more than one tone, but they take on the same resonant like characteristics when discovered using the sweeping EQ technique. They are a mixture of sounds all fighting for space in the mix. You will probably agree that this is more displeasing to the ear than pure tone resonance. This situation is caused by the overlapping effect of all the different playing instruments, reverbs and FX all happening simultaneously. There will be a muddy like texture to the area in question, which will most probably sound worse the further down the spectrum you find it.

Why does it sounds worse the further down the spectrum you go? To answer this I'll refer to a mixing technique – low cutting the mixer channels.

One of the reasons why you apply low cuts to most mixer channels is because low frequency audio doesn't mix too good. In a typical mixing situation, all the mixer channels will have low cuts except the kick and bass. This keeps everything out of the kick and bass-line's area, preventing it from becoming muddy. Here's a way to visualise this muddiness:

Imagine playing two notes simultaneously on a bass guitar, all the way down at the bottom of the bass guitar's range – a C and a D for instance. Will they sound nice? They simply won't sit right together. Play the same two notes but much further up, as high up the bass as they can go. They will sound much more pleasant to the ear than when they were played lower down. This is why muddiness can be more of a problem in the lows and low mids. Low frequencies simply don't blend together very easily.

Why can the kick and bass be together without sounding bad? In most cases the kick is not a 'tuned' sound. It doesn't have one particular frequency, so it doesn't clash with the note the bass is playing.

A good mix with appropriate low cuts on instruments, vocals and FX should reveal little or no muddiness in the lows or lower mids. Unfortunately, this is not always the case. The fact is, your consumer's player, or environment, may be about to boost a group of frequencies. What if the final master already contains a boost in the same area – in the form of a muddy like texture? The combined result will not sound too pleasant. This is where your attention should be focused. A reduction will certainly help with translation to the consumer's player. It will also help mask the unpleasant sound of a muddy like texture. But sometimes, the only solution for a problematic clump of overlapping sounds is to go back to the mixing stages and create better definition of the area in question using channel EQ.

When you're reducing a pure tone, a relatively high Q setting will suffice as you're homing in on a narrow banded area. When reducing an area of muddiness caused by many different sounds occupying the same space, the banded area may possibly be wider requiring a lower Q setting. When sweeping across the spectrum looking for problem frequencies, try to detect the width of the problem area. The further down the frequency spectrum, the wider the area is likely to be.

This technique is commonly used in mastering. Over time, your ears will learn to distinguish resonances, or other problem frequencies without the aid of a boosted EQ. Practicing this technique does more than find the areas that need attention, it also helps to build a solid relationship between you and the frequency spectrum. Coming up next is a live demonstration of this technique using the downloadable track. Use this link to download it now if you haven't already:

http://www.masteringtuition.com/index.php/practice-music/category/1-eq-demo-track

Linear Phase EQ Demonstration

This is a link to a video demonstration of the technique being applied to the downloadable piece of music:

http://youtu.be/8KnZP_sz-DA

See if you found the same areas to be problematic.

Tonal/Spectral Processing – Final Word

EQ's may differ in their looks and controls, but essentially they all perform the same job of boosting or reducing specific bands of frequencies.

The technique demonstrated in this chapter works great using Logic's Linear Phase EQ. However, it might not work so smoothly on all EQ's but the principle remains the same; boosting an area of frequencies can help you distinguish the problems happening in that particular area.

By now you should be starting to understand the purpose behind the whole mastering process. Through the use of EQ, we can literally mould a reliable shape into the tone of the music, enabling it to sound balanced and correct no matter where it is finally played.

This bring us to the end of the EQ chapter. The next chapter will explore the use of compression and limiting – extremely important tools in shaping a track for good translation to the consumer's player.

Chapter 5: Dynamics Processing

In this chapter:

Introduction to Dynamics Processing: Brief introduction to compressors and limiters, discussing their role in the mastering process.

How Compressors Work: An in-depth look at the controls/parameters on a compressor, discussing how different settings affect the sound in mastering situations.

Circuit Types: A brief explanation of some of the different types of circuits found within analogue compressors that give them their character, of which our digital counterparts are direct copies of.

Setting Up Your Compressor: A detailed guide on setting up a compressor in mastering situations.

Demonstration of Compressor Technique: Link to a video demonstration of the technique described in the preceding subchapter.

Fatness and RMS: Explains what RMS is and how it relates to loudness and fatness. Includes further discussion into compression techniques that can fatten a mix on another level, all the time helping you realise your goal when mastering your music.

Limiters: Leading on from the previous subchapter, this focuses on limiters and RMS, providing some useful guidelines on using limiters in mastering situations.

Limiters and RMS: Leading on from the previous subchapter, this describes in great detail how a limiter ties in with the subject of RMS, giving you a greater understanding of the very purpose of the limiter and how best to use it.

After the Limiter: Answers the question of whether you can process further after using a limiter.

Demonstration of a Limiter: Link to a video demonstration of a limiter being used in a mastering situation.

Multi-Band Compression: An introduction to a very useful tool and an explanation about the major differences in operation from single-band compression.

Multi-Band Compression Situations: A look at the various situations where multi-band compression comes into play.

Demonstration of Multi-Band Compression: Link to a video demonstration of multi-band compression being used in one of the discussed situations from the previous subchapter.

Final Word: A few brief words summing up the chapter.

Introduction to Dynamics Processing

Having read the previous chapters, you should have begun to understand the real purpose behind the mastering process; the complete uncertainty of the consumer's environment is what we have to overcome. Mastering aims to equip the track with the shape and sonic power to sound great on whatever system it is finally played on.

So where does compression and limiting come into all of this? In the chapter 'Definition of Mastering', I explained that when playing my recordings at my friend's house, I had to play them at higher volumes (in comparison to commercial CDs) to be able to hear any of the finer details. The fundamental difference between my recordings and the commercial records was the dynamic range. By that I mean the difference between the loudest parts and the quietest parts.

Because the commercial CD's had undergone professional mastering, their dynamic range had been considerably reduced enabling all the quieter details to be made louder, without sounding as though they had been turned up. Let's imagine our listening environment when recording and mixing. As previously discussed, we are free from distraction, background noise and have relatively good quality speakers. In such an environment, a large amount of dynamic range wouldn't be so much of a problem; if a track happened to have some really quiet details amongst some louder ones, they would be heard fairly well in this environment. The consumer's listening environment is not likely to be so perfect. There will probably be background noise and distraction, also the speakers themselves could be inferior quality – laptop speakers being a good example. If the dynamic range of the music is too large, as in the quiet bits are too quiet compared to the loudest, then an amount of the music will struggle to be heard in such environments. By reducing the dynamic range and raising the level of the quieter details, we give the music a better fighting chance of being distinguished in the uncertainty of the consumer's environment. The main tools for carrying this out are compressors and limiters.

Aside from the very practical purpose of reducing the dynamic range, compressors can help 'glue' a mix together by bringing it to a whole. They also play another role in the way they can fatten up the sound of the music. One reason why they give this fattening effect is down to RMS energy. You will remember from the previous chapter on EQ, that RMS levels are closely proportional to our perception of loudness. RMS is a subject in itself, and quite a complex one. Later in this chapter I will explain what RMS is in layman's terms.

Before we get to all that, let's explore in detail how compressors work.

How Compressors Work

Becoming familiar with the controls of your compressor is of extreme importance. You may already be familiar with how a compressor works in recording and mixing situations but when mastering, setting up a compressor can be a little different.

A typical compressor will have six main parameters/controls – threshold (dB), ratio (ratio), knee (hard-soft), makeup gain (dB), attack (ms) and release (ms or s). In truth, there is a chance there may be a few more options depending on the compressor you use, but for now it is only the above mentioned that we are concerned with, as they are the controls you will find on most compressors and are what you need to become familiar with for the purpose of mastering.

The compressor detects the level of the incoming signal. Whenever the signal reaches a decidable amount (threshold), the compressor responds by attenuating the signal – commonly referred to as 'gain reduction'. How much is dependent on how far over the threshold the signal has tried to go. The further past the threshold, the more attenuation/gain reduction. The relationship between how much attenuation is applied and how far past the threshold the signal has gone is determined by the ratio. A low ratio setting will equal a relatively low amount of attenuation in comparison to how far past the threshold the signal is trying to go. Increase the ratio and the attenuation will be greater.

Below is a diagram known as a transfer curve. This illustrates the relationship between the input and output of the compressor. A transfer curve can actually illustrate the relationship between the input and output level of any signal level processor, or even just an amplifier.

In the diagram above, notice the threshold is set to -40dB. Once the signal level passes this point, the compressor attenuates the signal at a ratio of 2:1. This means for every 2dB increase of input level, the output increases by 1dB. A 4dB increase of input level will equal a 2dB increase at the output. 8dB goes in, 4dB comes out, 16 in, 8 out and so on.

The speed at which the compressor responds is adjusted by the attack. A long attack will set the compressor to respond relatively slowly to the signal passing the threshold. A short attack and the compressor will respond quickly.

Once the signal passes back below the threshold, the compressor will return back to its idle state of zero attenuation at a speed decided by the release setting (sometimes known as recovery time).

How the attack affects the sound

The attack time affects the amount of attenuation. If the signal contains a very short burst of loudness, and the compressor's attack time is set to respond fairly slow, longer than the duration of the burst, then the loud burst will have passed before the compressor has had time to reach full attenuation. The slow attack in this scenario results in less attenuation than if the attack was set to be faster. This is an important characteristic when considering how much sonic information is contained in the short burst of sound at the point of when a musical instrument is struck – like the hammers on a piano, or a drum stick hitting a snare. These short bursts of almost instantaneous sound are known as the transients and play a vital role in the subject of compression in mastering. It is these short bursts of sound that contain all the punch. A very fast attack, causing the compressor to squash the transient hits can be damaging to the music.

On the other hand, controlling the problematic transient hit of a loud snare could be just what the track needed. Carefully setting the threshold can allow the compressor to be used as a tool in taming an undesirable loud percussive hit – more so with the use of multi-band compression, which will be covered in more detail shortly.

How the release affects the sound

The release setting will never affect the actual level of attenuation, but it can change the overall amount of compression over time.

A slow release will force the compressor to stay in its state of compression for longer. Choose a quick release and the compression time will be relatively shorter. Let's imagine the compressor's threshold has just been triggered by a burst of sound (transient) and has begun to attenuate the signal. If the threshold is triggered again, before the compressor has fully recovered from the last, then the process of attenuation will repeat before the compressor has returned back to its idle state, forcing the compressor to be in a constant state of attenuation. This would equate to an overall larger amount of compression over time.

Knee

The knee adjusts the smoothness of the transition from zero compression to full compression; or more technically, the curvature of the path to full ratio. A hard knee means the compressor will engage full ratio at the triggering point – no transition. A soft knee allows there to be a transition from a ratio of 1:1 (zero compression) to the ratio dialled in. Not all compressors are the same but the transition for a soft knee usually begins and leads up from about 10dB before the trigger point. Look at the two diagrams below:

As you can probably imagine, a softer knee tends to sound more pleasing to the ear in mastering situations as it's not so aggressive.

Makeup gain

The makeup gain is probably the simplest in terms of how it affects the signal – a post compression gain stage. Following compression, the likelihood is that the overall signal level will have been reduced. The makeup gain is used to raise the signal back up to a working level, which raises the uncompressed softer passages with it.

Can you use any compressor for mastering?

What determines whether a compressor is suitable for mastering is how transparently it carries out the process of attenuating signals. A vast amount of musical genre requires this to happen very smoothly, so much so that it becomes undetectable to the ear. Not all compressors have what it takes to deal with a whole mix passing through it. There is so much sonic information that a basic compressor, designed for a single sound source, may struggle to deal with such content and will produce distortion when driven hard. Compressors designed to cope with many elements, such as an entire mix, are commonly known as 'bus compressors' as they are typically inserted across a bus as opposed to a single mixer channel.

Some music will benefit from the obvious artefacts of a compressor being driven hard – most common in forms of dance music. I'm not referring to distortion here, I mean the way the compressor moves – the 'pumping effect' as it dips the level of the mix up and down in response to the music's rhythm.

In any case, the smoothness of a good quality compressor is crucial whether you intend to discretely glue a mix together, or aggressively pump the track in a rhythmical nature. The good news is that the standard compressor plugins you get with most DAW's are of a high enough quality that they can be used in many mastering situations (as long they are not driven too hard). I will demonstrate this soon.

Tone

Using compression in any situation will almost certainly affect the tone. The tone of any single sound, or a mix, is constantly changing over time. As the compressor works in the time domain by changing the level of a sound over time, you can appreciate that the tone of the sound being compressed will change with the compressor.

If the threshold triggering element of a mix is quite low in the spectrum, and the other elements which are not reaching the threshold are higher up, then the compressed mix may appear brighter than before as the compressor is acting on only the lower frequencies of the mix, sounding like you've reduced some low end.

Summary

By use of compression, the dynamic range of a piece of music can be reduced by attenuating the louder passages. Then the overall level can be lifted, transparently raising the softer passages and producing a better equipped master for the uncertainty of the final listening environment.

I hope by now you have begun to appreciate that the compressor is a somewhat mechanical device, constantly changing its state in relation to what the music is doing – the amount, the speed and the smoothness being completely dependent on what you dial in. This mechanical nature can become very musical with the right settings (explained soon), giving you that popular fattening effect or the perfect element for glueing a mix together.

Compressor Circuit Types

Before we get to using the compressor, it's worth briefly covering the subject of circuit types.

With a compressor plugin, you're likely to have the option of choosing the circuit type which will have an effect on the way the compressor sounds. With an analogue compressor, there must be a circuit that detects the level of the incoming signal, which in return gives instruction to attenuate the signal having exceeded a given threshold.

Most digital compressors (plugins) are copies of the way analogue compressors behave. One major factor of the behaviour, or character if you will, of an analogue compressor is the type of circuit it uses to determine the amount of attenuation with respect to the incoming signal. There are many types, however it wouldn't be practical to explain them all now. Following is a brief explanation of three circuit types – Opto, VCA and FET. These are all optional settings in Logic's own compressor plugin which is to be used for demonstration purposes.

Opto (Optical) – named accordingly. This circuit uses a light source and a photocell (phototransistor) or LDR (light dependent resistor). The level of the incoming signal is converted into light using a light bulb or an LED and then detected by the photocell/LDR. The amount of attenuation is determined by how much light the photocell is detecting. The result is a soft and smooth reaction to the incoming signal, which is often desirable in mastering applications. Optos can be a little slow in their response, so when a very fast attack is required, the Opto may not be suitable.

VCA – Voltage Controlled Amplifier. Probably the most common type down to their versatility. Attenuation is determined by detection of the incoming signal's voltage. They can cope with the two extremes very well – very fast, or slow and smooth.

FET – Field Effect Transistor. A transistor based circuit design with an aim to achieve a tube like sound (tube circuitry is often desirable in mastering applications as it can project its own flattering character across the mix). They are typically fast and smooth in their operation. The downside to FET is a high signal to noise ratio.

Compressor circuitry is another subject altogether but for now, don't worry too much. Just be aware that they differ and can each have a different sound. The best thing to do is to familiarise yourself by experimenting with each.

Setting Up Your Compressor for Mastering

Typically when mastering, we are dealing with varying levels of passages as opposed to something like the loose dynamic range of a bass guitar when mixing. In most cases, attack times should not be too short as not to risk damaging the important transient hits. They may start at around 20ms going up to as far as 100ms. Release times may be somewhere between 0.2s (200ms) to 2s (2000ms) sometimes as long as 5s. There is no real answer, it is completely dependent on the music at hand. You may, or may not be feeling confident at this point when determining your attack and release settings – it takes practice. The instructions in this subchapter will give you an idea of where to start.

The following technique assumes the track being mastered is of a typical commercial nature and has a kick and snare providing the rhythm. At this point I shall tell you that your compressor will most likely respond more to the kick than any other elements (providing the kick has been mixed at a proper level), you will soon discover why.

Can you distinguish the natural rhythm of the track? Can you nod your head back and forth with the rhythm? It's important that you can as this will help you decide upon attack and release times.

First choose a suitable ratio, no more than around 3:1. Relatively low ratios are best in mastering. Too much will result in a pumping effect of the compressor, unless that's what you're looking for. Some forms of dance music actually benefit from a little rhythmical pumping from the mastering compressor. For the technique I am going to demonstrate, the compressor will be set up in a relatively transparent way (undetectable to the listener).

Side note: The quality of the compressor being used may dictate how extreme you can go with your ratio.

To start with, set the attack time to be a medium amount, say around 30 to 40ms as we know that an attack too short may damage the transient hits and take away the punch of the music. The release time will be quite dependent on the rhythm of the track, for now just choose about 500ms. Select a medium to soft knee setting. A soft knee tends to give a more pleasant result in mastering as it sets the compressor to react in a less aggressive way to the threshold being triggered. Hit play and start to pull the threshold down; bring the compressor into an almost constant state of attenuation. You will notice a significant drop in volume. Use the makeup gain to bring the volume back up to a working level. Now it's time to use your ears to set the attack and release times.

If you shorten the attack too far, you will hear a reduction in the track's punch. Turn the attack too long and too much audio will have passed through before it has time to react. You will hear the compressor lagging behind, as if it can't keep up. It will really take away the transparentness of the compressor. There's a sweet spot for every track – with practice you'll feel it. Bear in mind that it may be after you've experimented with the release before you find the perfect attack.

The release time will really set the compressor swaying. Too short and you will hear the compressor aggressively jump back at you after every transient impact. Too long and you will feel the lagging behind effect again, along with a huge amount of attenuation. The real sweet spot is where the release begins to compliment the attack. Try to imagine it swaying back and forth to the rhythm. Again, there will be a certain amount of tweaking both attack and release together to find that sweet spot – try not to look at your compressor's meters, focus your mind on hearing the effects with your ears.

A little tip – if you're struggling to hear the effects of the attack and release, here's a way to visualise the attack and release in a mastering situation:

Imagine just a kick and hi-hat. The kick happens every beat, the hi-hat on the 8th's – all at about 100bpm. The kick's level is above the threshold and so it will trigger the compressor but the hi-hat's level is below. The kick triggers the compressor and can be noticed in the sudden reduction in the level of the hi-hats. As the compressor recovers, you would hear the hi-hat's level gradually rise back up. In such a scenario, if the hi-hats where to arrive back at their full level before the next kick triggers the threshold (the release time being shorter than the time between each kick), then the result would be an awkward jump back to the full level. The awkwardness coming from there being no real significance at the point between each kick where the release time stops and the compressor stops compressing (unless you get it bang on half a beat or something which is certainly a valid option). If the release was set to be only just longer than a beat, the compressor will still be in a state of compression when the next kick triggers the threshold. This takes away the awkwardness of the compressor jumping back and resting at no significant time, the result being smoother and more flattering to the music.

Obviously, the likelihood is that the music in question is probably much more complex than just a kick and hi-hat; there will be a whole array of different sounds triggering the compressor at various times, but this concept can still be applied. Having the release that tiny bit behind the natural rhythm can give you a smoother, more flattering result.

On your compressor, put the makeup gain back down to 0dB and go back to the threshold. At this point, there's a chance that we might be squashing the track a little too much for mastering purposes. Typical levels of gain reduction should not be more than about 3dB in most cases. More than that and the transparentness of the compressor may be taken away. Perhaps take the threshold back up a little to find a suitable amount of attenuation, then reset the makeup gain to compensate for the loss of level. Be aware that further adjustment of the threshold can affect your other settings – certain elements in the music may cease to trigger the compressor having raised the threshold.

The real transparency of this technique comes from the compressor moving in a natural way to the rhythm and so blends in with the musicality of the track. It's when your settings are too harsh, aggressive, too fast or too slow that the compressor falls out of sync and exposes itself as being an artificial element in the music.

Coming up next is a link to a video of this technique using Logic's own compressor.

I'm afraid there is no compressor setting that fits all. The attack and release times will vary massively from track to track. There's no real relationship between the BPM of the track and setting the attack and release times. It's all about the feel of the rhythm.

Demonstration of Single-Band Compression Technique

This is a link to a video demonstration of the technique described in the preceding subchapter:

http://youtu.be/8rCSTLMM4uY

Fatness and RMS

Why does compression give fatness?

I believe there are two reasons for this, one is quite technical and relates to the subject of RMS levels; the other is down to the way the compressor moves in response to the music.

Reason 1: RMS

As briefly mentioned in the previous chapter – EQ, Tone and Frequency Spectrum – the RMS level is closely proportional to our perception of loudness. Having set up your compressor using the technique discussed, you will increase the RMS level of the track without increasing the peak level. This increased perception of loudness sounds like we've fattened up the track. So what's happening here; why do RMS levels relate to loudness and why does compression increase the RMS level?

Imagine a single sound wave on a graph, any shape or form. The peak level is the very tip of the sound wave, the highest point that the wave has reached. One might think it would make sense that the higher this point is, the louder we shall perceive the sound – not necessarily.

RMS stands for root mean square; it's a way to measure the magnitude of varying quantities. It is the square root of the mean of the squares of the values. In simpler terms, our sensitivity to loudness is proportional to the average value of the various levels present in a given time. The RMS is a value given to the energy, or loudness of audio over time.

Imagine how a waveform looks inside your DAW. You can amount the RMS value to the actual amount of visual surface area that the waveform covers. Two audio files could peak at the same level but have massively different RMS values; this would appear obvious when looking at the two waveforms in an audio editor.

In the snapshot above: Two audio files of the same music track, the top one has not experienced mastering, the bottom one has. Below the waveforms are two level meters, one for each track. The dark shade indicates peak levels, the light shade indicates RMS. When comparing one against the other, the difference between the peak levels is much less than the difference between the RMS. The second track's RMS level has been significantly increased following the mastering process.

When compressing an audio track, we only reduce the level above a given threshold. It's as if we're squashing down just the tops of the sound waves leaving everything beneath untouched. Then, with this increased available headroom above the now lowered peaks, we can raise the overall volume of the track (makeup gain), so the peaks reach back up to their original level. Only now many more peaks will be reaching this level than before. Can you imagine how this increases the visual surface area when viewing the compressed waves in an audio editor? The result is an increase in RMS level with no increase in peak levels. It gives you a sense of the track being inflated almost, rather than just being louder. It has got fatter!

It's important to understand that this process is happening over the changing levels of different passages in the music, and not in relation to individual sound waves. The actual peak of each individual sound wave cannot be reduced without reducing the rest of the wave below it – unless you had a super fast compressor that can attack and release as quick as a single sound wave appears and disappears. There is actually a type of processor that does this naturally; there's a dedicated chapter about it in Part 2.

Our ear's sensitivity to RMS rolls off as we get closer to the bottom of our hearing range, so to hear the bass regions at the same level as the rest, more energy is required. Low frequency sound waves contain more energy than the rest. For this reason, low frequency audio will force the RMS level meters to jump higher in response to what you are hearing when compared to the rest of the spectrum. You need to consider this in mastering. The bass end of the music will always require more of the limited amount of available energy.

Side note: The idea of there being a limited amount of energy will be discussed shortly.

Reason 2. Compressor's movement

When referring to fatness in this way, I am referring to the fattening of a certain element in a mix, which in turn will give an overall perception of fatness. In particular the kick drum, which is most effective in dance and electronic styles, although this can be perceived in any genre.

When a kick drum is very prominent in a mix, it can really fatten up the track if the style is something like dubstep, drum 'n' bass, or some other dance style like various forms of house.

You should now have a good understanding of how the compressor responds to one sound being louder than any other – it basically dips in volume. So what will happen every time the kick passes through the compressor? We get a dip in the overall level. And with the correct compressor settings, it can be achieved in a flattering way, mostly for the instance of each kick drum. The clever thing is, we don't perceive this as a reduction of the kick itself, but rather the music around it. The kick, although being the main thing which is triggering the compressor, appears to be untouched by the reduction in level. The effect is as if the music is literally ducking out of the way for every kick. Think back to earlier in this chapter where I said that sometimes the pumping effect of a compressor can be desirable, well this is it. When set correctly, this can really fatten up your music. Remember how I said having an extended bottom end can actually help a track translate to a little kitchen radio, or some inferior laptop speakers? Well this is what I meant. By use of compression, the kick drum projects its imprint across the whole mix – right across the whole frequency spectrum. The way the compressor dips everything for the kick gives us a feel for the actual weight and solidity of the kick – it makes it sound fat.

The reason why this effect is most obvious on the kick is down to the RMS energy. The kick contains the most as it will likely extend down lower than anything else in the mix, and the lower the frequency the more energy is required for us to hear it. So the compressor will respond more to the kick than any other part of the spectrum. Try to imagine how the kick drum is made up of frequencies that extend right up into the high mids. Because the music can literally be forced aside for every kick, you can actually get a feel for the weight of the kick without the need to hear its bottom end. So it's possible to have a fat low end be perceived on speakers with no low end – magic.

This effect happens best when the bottom end of the kick extends down nice and low, lower than the bass line or anything else in the mix. That way the kick doesn't necessarily have to be mixed high in the mix to achieve the effect. There will be enough RMS energy in the bottom end to get those needles moving.

This audio effect is true to a whole variety of music. It doesn't just have to be dance music with a pumping kick. The effect can be slight yet still effective in all the same ways. And it's not just the kick drum; bass-lines can project their imprint across the whole spectrum too, helping with translation to inferior speakers.

By now you should be starting to realise the enormous power of your compressor when shaping a track to sound fantastic in whatever environment it is finally played in.

Limiters

The traditional purpose for a limiter is to catch stray peaks jumping out, preventing distortion or clipping when raising the mix to a more suitable loudness level. This is why you'll sometimes hear them being referred to as 'peak limiters'. In more recent times, the limiter's purpose has evolved somewhat. Rather than catching the odd stray peak, they are more often used as a solid ceiling to force the mix up against, gaining high levels of RMS. The whole subject of loudness and RMS energy ties in very well with the limiter.

Limiters are mainly found to be the last process in the master chain, as once you have used a limiter to full effect, the audio's condition is such that any further processing will not blend as well as when applying the same process earlier in the chain. In fact, further processing after a limiter can either harm the mix or un-do some of the earlier processing – you will soon understand why.

A limiter works in a very similar way to a compressor only much simpler. They typically have three parameters – threshold (dB), release (ms) and output/ceiling (dB). Just like the compressor, the limiter will attenuate the signal level at the point of it passing the threshold, only the amount of attenuation isn't dependent on the ratio as with a compressor. It simply limits the signal level from reaching any higher than the threshold, hence the name limiter. For this reason, there can be no adjustment of attack, the attack time is quite simply instantaneous. The release can be adjusted and, apart from the threshold, is the only way to change the way the limiter behaves and sounds.

Setting the limiter's release is similar to that of the compressor in the sense that too long will give a lagging behind feeling and a significant drop in perceived volume. However, fairly short release times are more acceptable as this is what the limiter is designed to do. So to get the limiter working in a transparent way, we don't necessarily have to have the limiter moving with the rhythm as with the compressor. The instant attack and a fast enough release (so that we don't feel it lagging behind) will provide a suitably transparent way to obtain more RMS energy. As with the compressor, there will be a sweet spot for the limiter's release; too quick will sound edgy, too slow will sound squashed or suffocated.

Warning – the instant attack will impact on the transient hits so go easy.

Limiters provide more RMS in the same way as compression; the reduced peak levels create more headroom above the peaks allowing there to be an overall gain without the risk of hitting the top and causing clipping. In most cases, the limiter performs the gain-increase automatically; however low you pull down the threshold, the limiter will increase the output gain by the same amount, so the output will always be the at the top of the ceiling.

Similar to most compression techniques, we usually want the limiting to be transparent; although as a limiter increases the RMS level of the mix, you can appreciate that the limiter can give a certain amount of fattening too, which is sometimes desired.

Side note: There's a fine line between pushing your limiter a little harder to achieve the fattening effect and over-cooking.

A limiter acts like a brick wall at the top stopping any stray peaks from triggering the reds, and with the right release setting, can do this in a transparent way. So much so that many peaks can be forced over the threshold to obtain the desired increase in RMS level, which in turn gives us the perception of increased loudness. You can generally achieve more RMS with a limiter than you can with a compressor.

As with the compressor, it is important to understand that as a peak is reduced, for that very instance in time, the whole signal's level is reduced, not just the tips of single waves. When a limiter is pushed too far, RMS heavy elements (like a kick drum) can sound as though they are literally punching a hole through the mix due to such a severe drop in level at that very instant. This relates back to the compression technique where the compressor's reaction to the kick drum can be of help when translating its weight and solidity to different systems. In the case of the limiter reacting in the same way, the effect is not always so desirable because of the enormous attenuation and aggressive attack, and should therefore be approached with caution. As long as you have your compressor setup correctly, it's unlikely you'll need to use a limiter for this effect, or risk of it happening unintentionally.

Limiters and RMS

The RMS level of a piece of music can only really go so far before you loose the punch and begin to create an unpleasant squashed effect. Always give it space to breathe. The point at which this occurs is different from one piece of music to another and is quite dependent on the quality of the mix – the better the mix the more RMS can be obtained before sounding unpleasant.

Side note: Part 2 includes further discussions on how the quality of a mix dictates how much RMS can be obtained, and what the 'quality of the mix' actually means.

After the compression stages in the master chain, the likelihood is you will have increased the RMS by a certain amount. Whatever room is left for more RMS, if any, is achieved by the limiter. It's as if the limiter fills in the gaps at the end of the processing with RMS energy, and then seals it off. It's usually the case that the less the limiter is required to do to achieve the desired loudness, the more punchy is the result.

With the idea that the limiter fills in any space left with energy, imagine now that we have just applied a low cut to our track and removed some very low end frequencies – say we've rolled it off at 30Hz. This removal of perhaps unnecessary RMS energy, will have effectively created some space for the limiter to fill in with RMS energy, and it does this by concentrating more across the remaining frequencies. Now let's imagine we have just rolled off the top a little, perhaps above 17kHz. This will effectively create more space for the limiter to replenish with RMS energy, and again will apply the energy to the remaining frequencies. Bear in mind that the top end doesn't carry anything like as much RMS energy as the lows.

Can you see how more apparent loudness can be obtained by concentrating the RMS energy into just the most relevant area of the spectrum?

We know from the chapter on EQ that the midrange carries the greatest responsibility, as in a lot of cases it is mainly midrange that will be heard. Well this is how we can ensure there is plenty of energy in the mids. If the track has a problematic huge amount of low end sub, so low that it would not even be heard on average speakers, then the limiter will not be able to give as much power to the rest of the spectrum. That is unless we adjust the low end by applying a shelf or a low cut, or multi-band compression which shall be discussed soon. But don't forget that a healthy amount of low can be put to good use with your compressor by fattening and helping with translation. Just be aware that the lows require the most RMS to be heard and so they take most of it up. The more low end, the less RMS can be applied elsewhere by the limiter and therefore lower perceived loudness on certain music playing systems.

In most cases, a low cut up to around the 35Hz area will still allow you to drive the single band compressor with the kick and bass, but it will let you push that tiny bit harder with the limiter before it sounds bad.

Limiters are very easy to dial in, but with ease brings carelessness. This leads on to a very important point:

Instant loudness, instant improvement? The brain's response to an increase in the loudness of music is usually positive – you turn up the radio when your favourite song comes on, not down. It's only really the transition from quiet to loud that gives us the feeling of the music sounding better. Over time loudness can cause fatigue and be rather unpleasant, even unbearable. When you start to apply the limiter, you may at first feel you have improved matters by increasing the apparent loudness. Be sure that you are really reacting to the limiter's positive effects and not just responding to everything sounding louder than it did before.

A good way to ensure your feelings are true is to compensate for the increased perception of loudness created by the limiter by reducing the output level to your monitors by roughly the same perceived amount. When I start mastering, I set the level of my monitors to a comfortable amount using the output control on my interface. As I proceed through the mastering processes, I constantly alter the output level to my monitors so the perceived level is always roughly the same as when I started out.

The fact is, pushing the limiter too hard is bad. The fast attack is extremely damaging to the transients, and it's the transients that contain the punch. There is a well known term that resonates throughout the mastering world... 'the loudness war'.

Do not fall victim to the belief that if one song appears louder than another, for some reason it is better. If you turn up the quieter of the two, you will probably discover that the quieter one packs more punch as it hasn't been squashed. Has your hi-fi got a volume control? Has your friend's hi-fi? You can always turn it up. We want a suitable amount of loudness to give a sense of body and weight. We also need to ensure all the details are audible amongst the distraction and background noise of the consumer's environment. Anything after that and you should be questioning the need for any more RMS.

That being said, there are ways to achieve high RMS levels without destroying the mix. All will be revealed in Part 2.

After the Limiter

Can we apply further processing after the limiter?

Strictly speaking, there should never be any plugins inserted after a limiter unless it's for dithering purposes (dithering is a process that helps the smooth conversion from higher word lengths to lower when bouncing a project down; 24-bit to 16-bit for instance). There is however one situation where a certain kind of technique can be applied after the limiter – when you plan to get the final result really loud but that's all to come in Part 2. For now it's best to always think of your limiter as the last process in the chain. Here's why:

Let's say you applied an EQ after the limiter. If you boosted any frequency bands, you will surely hit the top and trigger the red lights indicating you are clipping the audio (clipping off the tips of the waveforms). This is because the limiter has already pushed everything up to just before hitting the top. Doing so will probably produce audible distortion at the output. And if you reduce a band of frequencies, you will loose some of the loudness you just obtained by using the limiter in the first place.

If the EQ was to go before the limiter, there would be no risk of clipping when boosting a frequency band. Reducing a band's gain, which would lower the RMS level, could be replenished by pushing the limiter a little harder. For best results, put the limiter last in the chain. Try to think of a limiter as a final seal that cannot be broken.

Important point: Always set the ceiling to be about 0.2db below maximum level creating a tiny little gap between the top of the peaks and 0dB. If you are mastering loud, or 'hot' as some people call it, set it even lower, -0.4dB perhaps. There are a couple of reasons for this:

Reason 1

Some software's overload indictors (red lights) will trigger, indicating that clipping has occurred when a peak reaches 0dB regardless of whether it's actually clipping or not.

Reason 2

Inter-sample peaks. This subject veers off a little from the kind of tuition provided here but it's important to mention it. Basically, even though you have your limiter set to catch any stray peaks, some peaks may still reach higher than 0dB during the conversion process from digital to analogue within a music playing system. This is down to how the digital signal is converted from absolute numerical values to the smooth curvature of a sound wave. The smoothing process can cause peaks to jump higher than the actual numerical values they represent in digital form in order to find the smoothest curve. Setting your ceiling to be lower than 0dB helps prevent problems caused by inter-sample peaks.

The next subchapter is a link to a short video demonstration of the use of a limiter at the end of a basic master chain.

Demonstration of a Limiter

This is a link to a short video demonstration of the use of a limiter at the end of a basic master chain:

http://youtu.be/P08g8Gc-Q4w

Multi-Band Compression

Getting back to compression now, a multi-band compressor is essentially three, four, or possibly five compressors in one, with each compressor focusing on a specific band of frequencies. You will almost certainly find all the same controls as on a single-band compressor (that do all the same things), only multiple sets of them. But what really sets it apart is the types of things we use it for.

A multi-band compressor is not as much a tool for giving that layer of gel and bringing a mix to whole as is with the single-band, although it does have some similar benefits. It's more of a corrective tool for controlling elements within the mix that need some attention. A common use for a multi-band compressor is to tighten up the lower frequencies, such as a bass-line or kick drum. Other uses might be to control a stray loud percussive hit in the high mid. De-essing is a common use too, even in mastering. Another use might be to tighten an under-compressed vocal in the mids. Essentially, a multi-band compressor is a means to tap into the mix and tighten up, tidy up, or whatever is needed.

As the multi-band compressor can be set differently from one frequency band to another, a certain amount of spectral adjustment can be made too. As well as compressing, you can use multi-band for similar reasons as your EQ, such as to balance a particular area of the spectrum.

The only major difference in what you can dial in, compared to single-band, is the adjustment of the affected frequency bands – by that I mean where one compressor ends and the next one begins. With that in mind, it's quite astonishing just how different the two instruments are.

As with single-band compressors, there's likely to be a few extra options on your multi-band other than the ones we have already discussed, but they don't concern us right now. We're just going to examine the same parameters as we have with the single-band.

A major difference you will notice when using multi-band compared to single-band is how you set your attack and release times. Because we're tapping into the mix, it makes sense that the attack and release times are going to be closer to that found with the compressing of single channels, such as vocals or bass for example. The ratio may also be quite different with a resemblance closer to that of channel compression. In general, attack and release times may be shorter and ratios may be greater.

Multi-Band Compression Situations

Imagine you have received a finished mix for mastering but the hi-hats are a little over-powering, and they only appear at certain points in the track. EQ alone may tone them down to bring less attention to themselves, but the effects of the EQ will apply to the whole track, start to finish. EQ will permanently change the global texture of the track, which is often the desired effect, but by setting a band on the compressor to be focused on the area of the spectrum where the hi-hats are, you can set the compressor's threshold to be triggered only when the hi-hats become over-powering. It's as if you have access to the mix and have reduced the level of just the hi-hats.

Obviously there would be other stray peaks in the highs that would trigger the compressor too from time to time. You need to decide where to strike a balance with your threshold and ratio. The chances are the controlling of these other peaks would be desirable too. For this you will be looking at very fast attack and release times, something perhaps as low as 5ms for the attack, and release as short as 20ms. Set the attack too long and the hi-hat will simply pass by before the compressor has time to control it. Setting the release too long may sound like you've reduced the hi-hats using an EQ as it could stay in its state of compression until the next hi-hat arrives. The ratio will vary massively, it will depend on how much control you desire. As this effect reduces some top end, you can expect some of the same results as you would with the EQ. A more rounded sound may be obtained, or even warmth, depending on the affected area of the spectrum.

A very good and common use I find for the multi-band compressor is to tighten up a loose, or to control an overpowering low end. Slightly longer attack and release times will be useful here compared to the highs, 40ms attack, 100ms release perhaps. We want a certain amount of transient information to pass in this case as it will be carrying the low end's punch. This tightening of the lows does wonders for translation to the consumer's player as it improves the low end's punch, helping it to be identified on smaller speakers. In some cases, the tightening of the lows can actually allow for the final product to be played at higher volumes before a hi-fi's speakers produce distortion as the reduced dynamic range lessens the actual distance the speaker cone has to travel to produce the low end.

Referring back to the 'Limiters and RMS' subchapter, the use of multi-band across the lows can create a better conditioned mix for the application of limiting to gain higher levels of RMS. Multi-band compression is very effective when balance is needed in the low end.

On the next page is a video demonstration of a multi-band compressor in a situation where some control of the low end is required.

When applying multi-band compression to the midrange frequencies, great care must be taken as there is a high risk of taking the life away. The reason for this is quite simply that almost every instrument within the mix will pass the midrange area and makes up the life of the track. The punch of almost every sound is largely the work of the mids. Sometimes multi-band compression can be used to tidy up an under-compressed vocal, if it's sitting high enough in the mix. However, compression in this situation may take the impact away from the snare – be careful.

Whereas the single-band compressor is probably the best tool for providing a layer of gel and bringing the mix to a whole, multi-band can certainly be used for that too. In such a scenario, all compressor bands would be active and fairly light settings should be chosen. Attack and release times may be similar to what I've mentioned above but ratios will be quite low – similar to single-band, and should probably be the same on each band of frequencies.

Multi-Band Compression Demonstration

This is a link to a video demonstration of a multi-band compressor being used to control an over-powering low end:

http://youtu.be/l4NEIPx0xj0

Dynamics Processing – Final Word

Compressors and limiters come in all shapes and sizes, with controls/parameters varying from one to another.

The techniques and methods demonstrated in this chapter make use of the controls/parameters found typically on most dynamics processors.

The consumer's speakers and environment could be literally anything – laptop speakers, a noisy kitchen, inside a moving car, or a peaceful living room with an expensive hi-fi. Use of dynamics processors provide the control and sonic energy to cut through the noise and sound bigger than the speakers themselves.

Chapter 6: Audio Suitability for Mastering

You are now one step away from seeing the four tools discussed so far be put into action in a live mastering session. But just before we get to that, now is a good time to go through a checklist to ensure a premaster (finished mix) is ready for the final mastering stage.

1. Amplitude and Headroom – Is it clipping?

Ideally, the highest peaks should be approximately 3dB below the maximum level.

The digital realm, unlike analogue has very strict rules governing waveforms exceeding the allowed maximum level, or 0dB as it would appear on your master fader inside your DAW. Digital technology works in precise values. If more level is pushed into a digital system than it can handle, it will be unable to capture that part of the signal, resulting in a flat line rather than the natural curves of a waveform (clipping), which can sound quite nasty, and can even damage the high frequency drivers in speakers. A mix that clips before mastering should be re-bounced by the mixing engineer to fall within the allowed headroom before mastering can commence.

2. Signal-to-Noise Ratio and the Noise Floor – Is it too quiet?

Ideally, the highest peaks should be high enough to be approximately 6dB below the maximum level.

Recording systems, both digital and analogue, are not perfect systems. They often generate their own low-level distortions or ‘artefacts’ which can be caused by many things. For example the hum and whine of electronic components adding to the signal, or slight inconsistencies when an analogue input like a microphone is converted to digital information. This layer of undesirable information is known as the 'noise floor'. Tape hiss is a good example of a sound that makes up the noise floor.

The relationship between the noise floor and the audible signal that we actually intend to hear is known as the 'signal to noise ratio'. If we are dramatically raising the level of a song during mastering, then we are also raising the level of the noise floor, and the quieter the mix before mastering then the louder we have to make it. This means that the noise floor becomes ever louder. Ensuring that the song is at a good level before mastering gives us a good signal to noise ratio, and helps to mask undesirable noise.

3. Sample Rate and Word Length – Is there enough data to work with?

Check that the files are at least 44.1kHz sample rate and 24-bit word length (word length is sometimes referred to as bit depth).

In terms of a PCM Wave File or in fact any other professional digital audio file, the minimum quality required to produce a good master is a Sample Rate of 44.1 kHz (CD quality) and a word length of 24-bit (beyond CD). Hopefully the sample rate will be higher than this, since usually the more frequent the sample rate – say, 48kHz, 96kHz or even 192kHz as opposed to 44.1kHz – the more harmonic detail is present in the audio files and the better they will hold up to processing in the mastering chain.

16-bit files are okay, but since they have fewer memory addresses dedicated to dynamic range, and so contain a less detailed ‘map’ of the various loudness levels in the mix, they tend to respond less favourably to dynamics processors such as compressors and limiters.

4. Capture – Is the entire mix present in the audio file?

Check that there is a small amount of silence before and after the audio information begins and ends. This is to ensure the DAW that bounced the audio file was able to properly capture the beginning and end of the audio, which is a surprisingly common issue.

5. Master-Bus Processing – Was the mix hit with an overall compressor or limiter before reaching the mastering engineer?

Check that the song to be mastered has a good dynamic range and has not been over processed by master bus compression.

Some mix engineers prefer to add a slight amount of compression to the overall mix in order to add punch and/or cohesion before the mastering process. When the effects are very subtle, this should not be a problem. But if the audio is in an excessively compressed state before reaching additional mastering compression and limiting, it can lead to poor quality audio that has little punch or definition.

6.Tonal Balance.

Is the mix of a high enough standard for the mastering process?

You should now assess the mix. You must ensure that the individual parts that make up the mix are balanced and clearly defined. Is the bass synth too loud and masking the kick drum? Is the snare drum too quiet? Are the high frequencies of the strings section creating resonances with the high frequencies of the hi-hats? Is the overall mix too boomy in the low end or too brittle and harsh with treble? Although mastering is able to address some of these issues, it is preferable that they aren’t present in the first place. The fewer processes required at the mastering stage to achieve the desired sound, the better the end result.

If you put a good mix through the mastering process, you will produce a good master. On the flip side, if you put a bad mix through the mastering process then its sonic shortcomings become even more obvious.

In Part 2, there is a chapter dedicated to how a mix should ideally sound to produce a professional, and commercial sounding master.

This concludes Part 1, 'The Audio Mastering Blueprint'. Following this is a section made up almost entirely of videos. I will master a track in full before your very eyes using the techniques and methods discussed up to this point.

After the live mastering session is when it really gets interesting – this is where I reveal some of the most closely guarded mastering secrets.

Enjoy.

Chapter 7: Live Mastering Session

Welcome to the live mastering session. This chapter is made up almost entirely of videos. You are about to witness a track being mastered from start to finish using the techniques covered in the 'Audio Mastering Blueprint'. The track being mastered is called 'Breathe' from the album 'On Parade' and is courtesy of the band 'Unconscious Jungle'. In fact, Unconscious Jungle have generously donated the entire album for free download in its 'unmastered' form just for you practice your mastering skills on. To access it, simply go to the practice music downloads page at MasteringTuition.com.

The full album in its finished (mastered) state is available to purchase here:

http://unconsciousjungle.bandcamp.com

Unconscious Jungle are a Manchester based 5 piece with a sound combining elements of folk, psychedelia, 20th century dance themes and choral harmonies. It is a fantastic album which I highly recommend.

It's worth mentioning that the final mastering carried out on this whole album uses techniques that are somewhat more advanced than what is included in the Audio Mastering Blueprint (such techniques are discussed in Part 2). For this reason, on the finished album, the track 'Breathe' may sound slightly different to the master we produce at the end of this live mastering session.

Please note, I haven't sped up the mastering process in anyway by snipping bits out. These videos capture the actual mastering process from start to finish. The entire length is about an hour.

The order of the live mastering session is as follows, please read through before watching the first video:

Part 1:  Listen – A very important part of the mastering process. At TGM Audio, we make notes during the first couple of listens to a track. (3m 50s)

Part 2: EQ – Using the technique demonstrated earlier in the book, some problem frequencies are uncovered and dealt with. (17m 32s)

Part 3: Multi-Band Compression – Here you will see how multi-band compression can be used to process both dynamics and tone. (17m 17s)

Part 4: Single-Band Compression – Having tightened up the mix, single-band compression is used as gelling agent, bringing the mix to whole. (13m 8s)

Part 5: Limiting – The final part to this live session. Limiting is the final seal, giving more RMS, lifting the track up to a commercial volume. (11m 28s)

Total length: 62m 35s

Live Mastering Session Video Links

Part 1: Listen

http://youtu.be/e77VPI_oUhg

Part 2: Linear Phase EQ

http://youtu.be/1VsOjgm4og4

Part 3: Multi-Band Compression

http://youtu.be/qjQjNci6DsU

Part 4: Single-Band Compression

http://youtu.be/gX4tTOhuAcI

Part 5: Limiting

http://youtu.be/UJFDRXh1M3s

Chapter 8: Advanced Mastering Taster – Ultra Depth

You have nearly arrived at Part 2 'The Secret Notebook of a Mastering Engineer'. The techniques in this section will become your secret weapons – your golden bullets. But just before you get there, I'd like to first show you how to set up a depth enhancement effect inside your master chain using a few very basic plugins. I call this effect... 'Ultra Depth'.

I'm not going to explain why this process gives more depth, I'm simply going to show you how to set it up. I want to leave the explanation for Part 2. This will increase your curiosity for the kinds of things you're about to learn.

It's really quite amazing just what a few very basic plugins can achieve when arranged and programmed in a certain way.

How to add depth and space to a mix using psychoacoustic manipulation.

This technique is like building your own effect. It works by arranging a series of buses and some very simple plugins which you will probably have as part of your DAW. If you haven't, don't worry because I'll show you where you can download them for free.

What you're basically going to do is set up a matrix which splits your master chain signal in two. One signal will receive processing but the other will not. The two signals (one wet and one dry) will then be mixed back together. The new signal, comprised of both the wet and dry signal, will contain the Ultra Depth effect.

Wet signal: The wet signal will undergo a very short time delay, attenuation and EQ. It will also experience a left and right swap to become right and left, and finally a phase invert is applied to the right channel only. The signal is then routed to the output channel. I'll provide specific settings for the wet signal once I've explained how to set this up inside your DAW.

Dry signal: The dry signal receives no processing. The signal is routed straight to the output channel (mixing in with the wet signal).

The signal at the output channel will contain the original audio (dry signal), combined with a delayed and processed version of itself mixed in (wet signal). Almost like an echo.

Have a look at the following three diagrams:

The way the process works is very simple, although building the effect inside your DAW may require a little patience as there are many different ways you could set this up.

Before I show you how I build it inside my DAW (Logic), I'll first show you a generic block diagram of how the signal is routed and processed. It's important that you understand the signal routing and processing. Once you grasp this part, you will probably be able to come up with your own way of creating the matrix inside your DAW. Here it is:

In this block diagram I suggest using an auxiliary send as a way to split the signal.

Each part of the wet signal processing is achieved using very basic plugins which I will show you once I've explained how to create the matrix inside your DAW.

This is how I create the Ultra Depth effect inside Logic:

First of all, as with all the tutorials, I am using one stereo audio file dropped into track 1 in the arranger window like so:

Beneath the arranger is the mixer. The first channel (Audio 1) relates to track 1 in the arranger window (where the audio source is located).

Now look more closely at the mixer below. See how I've created a matrix which splits the signal, sends one to an auxiliary for processing before mixing it back in with the original signal:

Pay particular attention to the bus send. The first channel strip – Audio 1 (or 'MastChain 1' as I've named it) – uses a bus send to split the signal. Bus 1 is sent to an auxiliary channel – Aux 1 (or 'Processing' as I've named it) – for the Ultra Depth processing. The first channel's output (Audio 1) is set to send to the main output (Out 1-2), this is the dry signal. Aux 1 is also set to send to the main output. This means the main output channel will receive a mixture of the dry signal from the first channel strip (Audio 1) and the processed (wet) signal from the second channel strip (Aux 1).

Here's a break down of how the signal is routed inside my DAW:

The main output channel strip now includes the Ultra Depth effect (the original audio with a delayed and processed version of itself mixed in). If you wish to insert any plugins before the Ultra Depth effect, you would use the first channel strip (Audio 1). To insert plugins after the Ultra Depth effect, use the main output channel like so:

In the example above, I have inserted a linear phase EQ and a multi-band compressor (Multipressor) before the Ultra Depth effect. I've also inserted a compressor and limiter after the Ultra Depth effect.

The plugins and settings I have used to process the wet signal are as follows:

EQ: Low cut up to about 100Hz, possibly shelve out some highs or high cut the very highs (the settings on the EQ below are for demonstration purposes, set your own parameters to taste).

Delay: Between around 15 and 30ms (set to taste).

If you don't have a delay plugin, download Voxengo's 'Sound Delay' here for free:

http://www.voxengo.com/product/sounddelay

Left and right swap to become right and left. Phase invert right channel only.

If you don't have a plugin which switches left and right and can invert phase, use this one, it's free:

http://www.kellyindustries.com/stereo_tools.html

Effect level (Aux 1 fader level): Start at around -15dB (set to taste).

Tip: For dance music styles, stick to using short delay times.

Try it; as you bring up the effect level, listen to the extra depth suddenly appear in the mix. Don't forget that in mastering situations, we are only attempting to make very slight adjustments at a time. Ultra Depth's effect is impressive and subtle. There will not be a dramatic change to the way the mix sounds, as there never should be from a single mastering technique. The final sound comes from many different techniques all working together. When Ultra Depth is used as part of a working master chain, particularly with the techniques found in Part 2, you will be more aware of its positive effect on the final sound.

Ultra Depth, as amazing as it is, will simply not work on all material first time. In Part 2 there is a dedicated chapter all about Ultra Depth and how to modify it to make it more versatile.

The fascinating subject of psychoacoustics plays a significant role in the audio mastering process. I have become very involved in the idea of using psychoacoustic processes to improve the quality of the final master, particularly in the area of creating extra depth and space in the mix. I have read many studies on how this can be done. There are some more complex variations of the process I've described in this article, some can be found in papers that date back as far as 1983. From all my findings, I have simplified some quite complicated processes and have come up with this simple technique you can do to instantly add extra depth and space to your mix at the final mastering stages.

In Part 2, I have condensed down all the most important things you need to know about psychoacoustics to get your masters in line with the quality of the professionals.

Turn the page to begin Part 2.

Side note: Don't forget to read the bonus articles in the hidden section at MasteringTuition.com. Follow this link to go there now or scan the QR code:

http://masteringtuition.com/index.php/furtherlearning

Part 2: The Secret Notebook of a Mastering Engineer

Part 2 Table of Contents

Introduction

9: Stereo Thinking

10: Mid/Side Encoding and Decoding 

11: Stereo Widening, Or Is It? 

Widening the mix, how far can you go?

Narrowing the mix, how far can you go?

12: Mid/Side EQ and Stereo to Mono Compatibility 

Mid/side EQ

The lows

Sweetening the ambience

Focusing the vocals

Stereo widening

Stereo to mono compatibility

Midrange meat

13: Expansion

14: Mid/Side Dynamics

Compressing the mid channel

Compressing the side channel

Mid/side expansion

Mid/side processing conclusion

15: Parallel Compression

Parallel compression for attitude

Parallel compression for transparency

16: Mastering Reverb

17: Ultra Depth Continued 

Papers and patents about enhancing depth using psychoacoustic manipulation

Creating extra depth using psychoacoustic manipulation

Mono compatibility

When can Ultra Depth be used?

EQ'ing the effect

Other ways to improve versatility

Summary

18: Saturation in Mastering 

19: Exciters and Enhancers 

20: Independent Left and Right Settings 

Stereo EQ

Stereo Compression

21: The Mix

Mixing sounds together

Separation

Channel and bus compression

Summary

22: Loud Masters

Section 1. Why does one mix permit loudness and another doesn't?

Section 2. Tips for getting a little more RMS

Transient processors

23: The Master Chain (ordering your processors)

EQ and single-band compression

Multi-band compression

Multi-band expansion

Limiting

Mastering EQ

Mastering reverb

Ambience enhancement

Saturation and EQ

Harmonic excitement

Mid/side adjustments

Mid/side and Ultra Depth

Summary

24: Mastering Multiple Tracks for an Album or EP 

Consistency

Loudness

Where to begin

Arranging the tracks

Global levels

Gap lengths

Fade the silence

Overlaps

Gap length decisions

25: Final Word

26: About The Author and Some Friendly Advice 

Introduction

You have arrived at the most exciting part of the book… 'The Secret Notebook of a Mastering Engineer'.

Equipped with your deep understanding of the very purpose of mastering, and knowing how to get pro results from your basic tool kit (EQ & Compression), it's now time to take it to a higher level.

There are many techniques to get through, although the first few all have something in common, a simple process that gives you enormous power – mid/side processing, which is a means to tap into the stereo elements of a mix. So a good place to start is to teach you about this process. We're about to start looking at things from a whole new perspective, you know about tonal balance and dynamic control, it's now time to look at things in 3D – in stereo. By having the ability to tap into the stereo elements of a mix, you can really do amazing things.

Just before we get to mid/side processing, I'd like to get you thinking about the importance of the stereo image within a piece of music.

Chapter 9: Stereo Thinking

I'll assume that you know what it is that gives the sense of a stereo image as opposed to mono (the fact that you're hearing differences between the left and right channels). The stereo information within a mix is what gives you the sense of being involved in the music, it pulls you in and surrounds you. So why is this?

Because we have two ears, one on either side of our head, we're able to locate the source of a sound. Our brain knows the difference between what both our ears are hearing. Things like time delay and differences in spectral information are how we localise the position of the audio source. For most of the midrange frequencies and below, we rely on the time delay between our ears. From around 4k and above, we begin to rely more on level differences caused by the head shadowing one ear from the sound. Below about 100Hz, our ability to locate the sound source starts to disappear. This is down to the diffraction of sound waves around the head.

Imagine you are sat in a room in front of a band playing an acoustic set with your eyes shut. Using your ears you would be able to locate where all the different musicians are positioned and you would surely feel involved in the sound because you are literally sat in front of them playing. By having a left and right channel, we can recreate some of the same effects we get from hearing things in real life – by having differences between the left and right channel. Similar to how a 3D film provides visual depth and works by having a separate image for your left and right eye, stereo sound provides sonic depth and works by having separate left and right channels.

I'm going to depict a scenario to give you an understanding of what's going on when you perceive sound in stereo. Imagine you're stood in a rectangular room with flat walls facing a stage at one end. On the stage, there's a person stood with a percussive instrument, a snare let's say. The person strikes the snare and sound waves leave the snare in all directions. You can begin to imagine how the sound is going to bounce around the rectangular room.

First of all, think about which sound will reach you first. This will be the direct sound from the snare. If you're stood central between the side walls facing each other then the direct snare hit will reach both your ears at the same time, which is how your brain will detect the snare being in line with the way you are facing (apart from using your eyes too of course).

The next sound to reach your ears are the reflections from the walls. By the way, I'm not referring to reverb right now, I'm talking about the very next sound wave to hit your ear drum coming from another direction, milliseconds later. From which wall the immediate following reflection has come from is all dependent on where you're stood, but I'd like to draw your attention to the reflections coming from the two walls either side of you. These are possibly the most important in terms of feeling involved in the sound. It's these side reflections that create the biggest time difference between your ears. Coming in from the side, the sound has the longest distance to travel before reaching the ear on the opposite side of your head when compared to sounds coming in from other directions.

Your ears receive huge amounts of stereo information caused by the sounds reflecting off the surrounding walls then travelling around your head. Modern auditoriums designed for classical/acoustic performances make clever use of this concept. They contain cleverly angled walls either side of the audience to make good use of these side reflections, enhancing the listening experience. Basically put, the width of the stereo image comes down to how much difference there is between left and right, and the side refections are a contributing factor.

Early reflections are very important. There is a certain time delay between direct sound and reflections (from any direction) where the brain uses the reflections to 'enhance' the direct sound. Too short a delay and the brain will not detect the delayed sounds as being reflections. Too much of a delay and the brain will separate the direct sound from the reflections and will perceive it in the form of an echo. This certain time is typically somewhere between 12 and 30ms, depending on what you are hearing.

This 'fusing together' of the first wavefront and preceding sound wave is known as the precedence effect or Haas effect. It is a fascinating phenomenon that has been described in a handful of different papers dating back as far as the 1940's, but probably most famously described by Helmut Haas, hence the name 'Haas effect'.

Just before we move on to mid/side processing, let me give you one more thing to think about to help you appreciate how we perceive stereo sound.

A common way for many to create a stereo effect during the mixing stages is to make use of the pan pot, making a certain sound louder on one side compared to the other; you are attempting to move the sound to one side by making the sound louder for one ear compared to the other. By doing this, you have created a difference between the left and right and introduced some stereo imaging into the mix. But I want you to have a listen to this piece of audio. This is a link to a video. Make sure you're sat in an ideal monitoring position – evenly between the speakers or wear headphones:

http://youtu.be/t7WOXrM08go

I have made the sound source move from the middle to the right and back by simply introducing a tiny time delay into one channel – there is absolutely no difference between the level of left or right. You can have the brain think a sound is coming from one side without changing the actual level of the sound going to each ear. The brain hears the sound in one ear before the other, so naturally calculates that the sound must be coming from one side. This isn't necessarily a practical thing to apply in mixing as it affects stereo to mono compatibility which will be explained in the 'Mid/Side EQ and Stereo to Mono Compatibility' chapter.

How does all this tie into mastering, or more specifically, mid/side processing? The next chapter teaches you how to get a hold of this important stereo information, and just like in the auditorium scenario, enhance it for the audience's pleasure.

Please note, the next couple of chapters explain mid/side encoding and decoding, which at first can be tricky to understand. Please be patient because you need to take all of this in if you are to become a pro at mastering. I would strongly advise reading at least as far as the end of the 'Mid/side Dynamics' chapter before actually trying out any of the techniques. This is because mid/side processing isn't fully explained until then.

As with all mastering techniques, it's important to go easy, making very slight adjustments at a time. Mid/side processing is no exception, as you will soon discover.

Chapter 10: Mid/Side Encoding and Decoding

This chapter explains in detail what mid/side is and how to apply this way of processing inside your DAW.

A stereo left and right signal can be thought of as having mono information and stereo information. Anything contained within the two signals that happens to be identical between left and right is mono information. Anything else, which will be different between left and right, is the stereo information.

Mid/side processing is a means to separate the stereo information from the mono information contained within a stereo signal, isolate it in its own separate channel for processing, before converting the new processed stereo information back into a stereo left and right signal.

The process converts left and right into two new channels of audio – mid and side.

The side is quite simply all the stereo information.

One might then naturally assume that the mid is all the mono information – unfortunately not. Even though using this process, we are able to separate the stereo information and process it, at no point are we ever able to isolate the 'true mono' information on its own. But despite this, we are still able to process the true mono to some extent.

The mid channel is actually the sum of left and right – this is how the mid channel is created. If you add left and right together, what remains is a single channel comprised of all the audio from left and right. What also occurs is that any audio that happens to be identical in both left and right (true mono) will experience boost at a gain of 3dB. This 3dB boost is how we are able to tap into the mono information and process it (more will be revealed about how it is processed in chapters to come).

Left channel + right channel = mid channel.

The side channel is the clever bit. It is the difference between left and right. Only audio that is different in one channel compared to the other will end up in the side channel.

To create the side channel, in the same way as we create the mid, we add the left and right together, but we reverse the phase (polarity) of one of the channels before summing them. All the mono information is then cancelled out leaving only stereo information.

Left channel + negative right channel = side channel.

Or put more simply:

Left – right = side.

The side channel only contains stereo information; all mono information is silent, unlike the mid channel where you can still hear stereo information at a difference of 3dB lower than the mono information.

Incase you're not already familiar with the concept of adding signals, if you add two identical signals together, they double in size (gain increase of 3dB). So when we create the mid channel we get the 3dB boost of any audio that happens to be identical in both left and right – usually lead vocals, bass, kick and snare.

Side note: For anyone thinking 'hang on, I thought a doubling of signal size equates to a 6dB increase.' It does if you are measuring voltage or sound pressure level (SPL). When referring to gain (power), the doubling is measured as a 3dB increase.

If you add two identical signals but reverse the phase of one of them, the opposite happens – they cancel out. You're left with no signal, total silence, commonly referred to as a null.

Turning left and right into mid and side can be done inside any DAW using buses and a phase reverse plugin or switch. Here's a block diagram of how you would route the signals:

This is the first part of what's known as a mid/side matrix; this part is the mid/side encoder. Don't worry if this looks tricky because I'll soon explain how to do it automatically using a free plugin.

This process means you can have the mid coming down one channel and the side coming down another, allowing you to process the two channels separately. Exactly what processing I am referring to will be discussed in the coming chapters.

Once processed, we need to turn mid and side back into left and right. To do this we simply create another matrix, one that reverses the process. In the same way we created the mid and side, reversing the process is done by adding the signals together.

To re-cap, the mid is the sum of left and right, all in one channel. The side is the difference between left and right, all in one channel.

A simple way to visualise the process is to think of it algebraically:

L + R = M

L – R = S

To retrieve the left channel, we simply add the mid to the side, 

M + S = L

and just as before, we reverse one channel before adding them, this gives us the right.

M – S = R

So why does adding mid and side produce the left channel and adding mid to a reversed side produce the right? To help me explain, 'M' can be though of as (L + R), and 'S' can be though of as (L – R).

When you add the mid to the side, the result is just the left channel:

(L + R) + (L – R) = L

The positive 'R' in the mid channel and the negative 'R' in the side channel both cancel out, leaving 'L' on its own. Now we have the left channel.

To retrieve the right channel, we simply reverse the phase of the side channel before adding them together:

(L – R) becomes (R – L).

(L + R) + (R – L) = R

The positive 'L' in the mid channel and the negative 'L' in the side channel both cancel out, leaving the 'R' on its own. Now we have the right channel.

Here's a diagram of how to route the signals inside a DAW using busses and a phase plugin or switch:

This is the second part to the mid/side matrix, known as the mid/side decoder.

Don't worry because there's no real reason you'll ever need to manually set this up (unless you're a geek like me and just love experimenting!). You can set up a very simple matrix allowing you to process the mid and side separately using a free plugin called 'MSED' by Voxengo. In fact, there are lots of free plugins that allow you to do this. The reason why I've shown you how to build a manual mid/side matrix is so you fully understand what's going on when you start applying mastering techniques in this way. As I emphasised throughout Part 1, having a deep understanding of how the processors work, not just what they do, is key to becoming a master of your processors.

This is Voxengo's MSED:

You simply click the 'Mode' button, choose 'encode' and your left and right become mid and side (left becomes mid and right becomes side).

Now you can process mid and side before using the same plugin to turn mid and side back into left and right. Simply choose 'decode' to convert back to left and right.

Depending on what you're trying to achieve, the likelihood is that you'll need a processor – an EQ for instance – that can process left and right separately, so in a mid/side situation the EQ will then be processing mid and side separately.

PSP's 'Master Q' is an EQ which allows you to process either the left or right channel on its own. Here it is, inserted into a channel strip in Logic. The MSED plugins are set to encode to mid and side, then decode back to left and right:

Side note: Many EQ plugins now have the option of processing in mid/side already built in.

If you don't have any processors which allow you to process left and right separately, you can set up a matrix which sends the mid and side down separate channels allowing you to process them with any processor. Here's how it may look:

Notice how Audio 1 is split and sent down two new separate channels using sends to buses – similar to how I create the Ultra Depth effect. Now there is a duplicate signal in channel strips two and three – Audio 2 and Audio 3. Having inserted MSED into the duplicate channels set to encode, you'll need to mute the mid channel in one plugin and the side channel in the other. In MSED you simply click the 'mid mute' or the 'side mute' button:

By muting the mid channel for one channel strip, and muting the side for the second, you have effectively split mid and side into separate channel strips where you can use two of any processor to process the signals. In the example above I have used two of Logic's own LinearPhase EQs. One plugin EQ's the mid, and the other EQ's the side.

Mid and side are joined back together by routing both Audio 2 and Audio 3 to Bus 3. The fourth channel strip, Audio 4, is set to receive audio from Bus 3 which contains Audio 2 and Audio 3. The third MSED plugin is set to decode and converts the mid and side back into left and right.

DDMF offer a plugin called 'Metaplugin' whose sole job is to load in other plugins for the purpose of building effect networks. It's really easy to use and not too expensive. You can use Metaplugin to split mid and side for separate processing with any plugins you have in your DAW. It's worth checking out, find it here:

http://www.ddmf.eu/product.php?id=3

Before we start using mid/side processing creatively, the next chapter raises a few important points which you need to understand so as not to make a mess of things! Following that are a couple of chapters that give away lots of clever techniques for making use of this mysterious and hugely misunderstood process.

Chapter 11: Stereo Widening, Or Is It?

This chapter discusses a relatively simple effect in how it works but your understanding of what is really happening is of huge importance. The deeper your understanding, the better you'll be at using mid/side processing.

The first thing you can do to quickly get an effect using mid/side is to very simply boost the level of the side. This will give the effect of the stereo image becoming wider. It's a neat little trick. Not all material will be in need of it, some won't benefit from it at all, but sometimes it can work a treat. Even something as little as 0.2dB boost of the side channel can subtly enhance the stereo sound stage giving the impression of a bigger overall sound. When a mix really lacks stereo information, sounding a bit small and narrow, a suitable boost in the side could be just what the doctor ordered.

On the other hand, there are times when a mix might contain too much stereo information. This is not as often, but I do use a mid/side matrix to reduce the stereo width of a track from time to time by simply turning down the side by a dB or two. Too much stereo information can feel like the stereo field is stretched and can feel weak in the middle. As much fun as it is to widen the stereo field, it's important to maintain a clear centre image too.

Another situation where you might reduce the side might be when a mix is a little swamped out with reverb. Reducing the side channel can clear things up by revealing more of the centre image where the mono information is, like the dry vocals. Reverb typically exists most in the stereo field and therefore finds its way into the side channel more so than the dry vocal. It is important to consider this when boosting the side channel; in doing so you may be turning up the reverbs and FX to an amount that was not desired in the first place – take care.

Widening the mix, how far can you go? 

As you can see, having control over the level of the side channel can be useful; you have the capacity to actually increase, or decrease the apparent width of the sound stage.

When widening the mix, how far you can really take it; 2, 3 or even 4dB? Imagine how wide the stereo image would become if you boosted the side by 5dB.

Apart from the risk of stretching and blurring the centre image, or swamping the mix out in reverb, I'm afraid to tell you that anything above around the 2dB area begins to have an unpleasant effect for another reason. Whereas the maximum amount you can boost the side within a mix before it sounds bad depends a great deal on what's already happening in the stereo field, there is a point where you cannot really go any further no matter what the material is you're using. In my experience, I find the 2dB area to be the absolute maximum you can boost the side in any given situation. The reason is as follows.

You see, we're not really boosting the stereo width of the mix in the strictest terms, we're increasing the level of the side before we convert back to left and right. This gives an illusion of an increase in the stereo width by creating a larger difference between the left and right channel.

To widen the stereo image, you would naturally expect that the left channel information must move further to the left of the stereo image, and the right channel information must move further to the right, achieved by boosting the stereo information in each channel and leaving the mono untouched. Well, boosting the side does give that effect but... it also mixes some left channel in with the right, and some right channel in with the left. With the illusion of there being a wider stereo field comes a drawback. Having boosted the side before converting back to left and right, we are left with an undesirable artefact – cross-talk. So what's actually happening to cause this?

When converting back from mid/side to left and right, as previously explained, a cancelation occurs in the process. When getting the left channel back on its own for instance, the negative 'R' contained within the side cancels when added to the positive 'R' contained within the mid, thus we are left with just the left channel.

(L - R) + (L + R) = L

The problem is, when the side has been boosted, there can no longer be a perfect cancelation of the positive 'R' and the negative 'R' when adding the mid and side together. The remaining amount of right channel information from the failed cancelation (which has a reversed phase) is now in the left channel. Can you see now how the extra width is really an illusion? For every dB of extra stereo information you achieve in the left channel, you also get a dB of negative right channel information to go with it, and vice versa. We have succeeded in creating a larger difference between the left and right channel, but we haven't really widened the actual stereo image that was created during the recording and mixing. We've basically made the stereo information louder and spread it around.

As I have said previously, the material you are working on will dictate how much, if any, you can boost the side. But anything past the 2dB area will be a sure thing to start destroying the natural stereo image that was created in the first place.

Narrowing the mix, how far can you go? 

A reduction in the side before converting back to left and right does not produce the same problem, you simply get a reduction in the apparent stereo width of the track with no annoying artefacts so there's no limit to how far you can go. All that happens is, as you narrow the mix, the mono information creeps up in level. You can keep going until the mix is completely mono, at which point the mono information will have experienced a 3dB boost.

Referring back to the explanation of arriving back at the left channel by adding the mid to the side, when reducing the sides's level, in this scenario you still get the failed complete cancelation between the positive and negative right channel, only this time it leaves some positive right channel in the left as opposed to negative. Having some positive right in the left channel simply contributes to a lack of stereo width, as intended. The more you reduce the side, the more positive right moves to the left until eventually you get an equal amount of right and left channel information in the left. Obviously the same happens for the right channel – an equal amount of left and right in the right channel too. Having reached this point, there is now no difference between left and right; the stereo width has been reduced to nothing. Each channel now contains the sum of left and right.

So, to give the illusion of an apparent increase in the stereo width, there is a trade off. As the stereo width increases, so does the reversed stereo cross-talk. This is why you can only go so far before it sounds bad. But all is not lost, the next few chapters will show you how to use mid/side processing to achieve great things despite this annoying problem.

Chapter 12: Mid/Side EQ and Stereo to Mono Compatibility

In this chapter we start looking at mid/side processing in a more creative way. I'll be giving away some great tips for using EQ in mid/side situations. However, I'll also be using this chapter to raise your awareness of an important discipline – stereo to mono compatibility.

Mid/side EQ

All the same rules apply when EQ'ing in mid/side channels as discussed in the previous chapter. When EQ'ing, we're still boosting or reducing levels of mid or side, only we're applying the adjustments to a specific band of frequencies.

Using your EQ for processing the mid and side channels is when things get interesting. You can alter the width of particular elements within the mix. For example, you could increase the overall width of the mix by boosting the side level but then apply EQ to the side and dip out certain areas of the spectrum bringing them back in towards the centre.

Many EQ's have mid/side as an option these days, if yours does, then that's great. If not, refer to the mid/side encoding and decoding chapter for how to use a normal EQ within a mid/side matrix.

The first technique I'll cover is not so creative but something you must understand before we move on to the fun stuff.

The lows

It's rare that any low end will exist in the stereo field once mixed down. The bass and kick should more often than not be mixed dead centre. Sometimes I've received mixes where the bass has been right over to one side purposefully in order to give a vintage feel to the track. Years ago, instruments could be panned all over the place to give a sense of space in the mix, even the kick and bass. The Beatles are probably the most obvious band that springs to mind. But in reality, the very nature of low frequencies means that very little difference should ever exist between your ears down to their long wave lengths, so having the lows mixed to left or right only gives an un-natural effect. On headphones, the un-natural sense you get from the lows being to one side is even more obvious and almost unbearable. So it's commonplace these days to have your lows neatly in the middle, or at least very close to, just where nature intended.

Another reason I feel that the lows should be mixed centre is how they make up the foundation of the mix. It's as if the rest of the mix sits on top of the kick and bass, so the lows need to be solid and reliable. Referring back to Part 1, we used multi-band compression to help level out the low end, keeping it under control – it's more solid that way. Having it dead centre gives more of a solid feel to the lows too.

In fairness, there's no absolute rule about where to mix the lows but I'm referring to pop music styles or commercial if you will, that's when it becomes more important.

Using your mid/side matrix, you can mute the mid and listen to just the side to hear what's going on in the stereo field. Straight away you'll know if low frequencies are present. Using your EQ, you can apply a low cut to the side channel which will focus the lows into the centre. Somewhere around the 80 to 100Hz area will be adequate.

Please note that panning the lows to one side during mix-down is not the only way they find their way into the stereo field. Reverb can create low end information in the stereo field too. There may be a little reverb on the drums, or even on the bass. You'll know straight away when you listen to the side whether the low frequencies you are hearing are panned instruments, or reverb applied to the instruments in the lows. If you are using reverbs, it's usually best to apply low cuts to the wet reverb signal during mixing as not to take away definition from the lows and low mids. After all, these are the areas where definition is most important. More about definition within a mix will be discussed soon in a dedicated chapter.

Either way, some kind of low cut usually improves matters by helping to consolidate the low end, providing a more sturdy platform for the rest of the mix to sit on. Moving away from the low end, it's now time to get a bit more creative.

Sweetening the ambience

A simple effect you can achieve by applying EQ to the side channels is to sweeten the tone of ambience (reverbs and FX). The side channel can give you access to these elements as they usually exist mostly in the stereo field. You can use EQ to brighten, sharpen or soften the tone of just reverbs and FX.

An example of when this might be suitable is if you find the mix needs to be a little brighter in tone, but EQ across the whole mix is drawing too much attention to elements that are sat in the middle – the vocals for instance, which may already be a little prominent in the mix. Finding brightness by EQ'ing the side will sweeten the overall tone without affecting the 'prominent' vocal. It would also help towards the blending of the vocals with the rest of the mix.

At times, you may find that nuisance resonances exists mostly within the reverbs and FX. EQ'ing the side channel gives you better access and can be a more efficient way of dealing with such situations.

Focusing the vocals

There are times when a vocal can get a little lost in the mix. To combat this, a clever little technique is to use a reduction across the midrange of the side channel to bring more focus to the vocal. The narrowing effect in the midrange area raises the level of the dry vocal out of the surrounding reverb. It does this because you have effectively exposed more of the mid channel in which the dry vocal will be sitting 3dB higher. Remember that the dry vocal won't be the only mono element in midrange area of the mix. The dry snare will be close by too.

Stereo widening

In truth, the best time to widen the mix is when you're mixing; good stereo imaging within the mix wins every time. However, sometimes it can be nice to add that little hint of extra width during the mastering stages. Here's a way that doesn't involve boosting the entire level of the side channel.

A very simple and natural sounding way to widen the stereo image is to leave the side channel alone and place your EQ over the mid channel. Think back to the first technique in Part 1 – sweeping a boosted EQ band across the spectrum to find resonances in the mix – this is very similar. With the side muted, sweep a boosted EQ across the mid with quite a low Q setting (wide affected area) and listen for where the most is happening in the spectrum. As demonstrated in the Audio Mastering Blueprint, keep narrowing the Q to home in on frequencies that you can notch out.

Simply apply a notch like reduction in those areas then un-mute the side (how much of a reduction will depend on your findings when sweeping your boosted EQ). A/B between EQ and no-EQ to listen. Although a reduction in overall loudness will be noticed, the reduction in the mid channel will give the effect of more width overall. The reason why this gives a smooth, natural effect is because you may have discovered areas of the spectrum that possibly needed levelling a little anyway, but rather than EQ'ing the entire mix, you've removed it from the mid. When the mid and side channels convert back to left and right, the reduction in that particular area causes there to be a bigger difference between left and right than before, giving more apparent width.

Another reason why this can feel more natural is the simple fact that you're using reductions as opposed to boosts which, as discussed in Part 1, generally returns more pleasing results than applying boosts if you're using an average quality EQ. Please note, you might want to stay away from the lows with your cuts (150Hz and below). Introducing width into the lows is not usually that useful, and there's likely to only be a minimal amount of stereo content down there so notching out low end will not amount to much extra width anyway.

The more you EQ out of the mid, the wider the track can apparently become. You could use EQ to shelve out some highs in the mid channel giving more width to the high end (don't forget about what happens when there's too much side though).

Using headphones while you play around with the tone of either the mid or the side will give you the best idea of what exactly is going on, but be sure to use monitors for making final decisions.

EQ'ing the mid channels leads to a very important point – stereo to mono compatibility.

Stereo to mono compatibility

In some situations, your final master will be played in mono (sum of left and right). For instance, when a stereo FM radio struggles to receive a signal down to bad weather, the stereo information is lost first (FM signals are transmitted in mid and side as opposed to left and right because it's more stable and uses less band width). A single speaker radio might play your track in mono, so might a mono PA system in a night club. In fact, there are many times when your left and right might be summed to facilitate the environment it's being played in. For this reason, you must ensure that your final master sounds good in mono.

Every process you apply to the mid channel will expose itself more when your track is played through a mono system. Applying a big dip in the midrange of the mid channel, giving a lovely warmth of extra width to the stereo image, might not sound too good on a mono system. A mono sound system is like someone has muted the side channel in your mid/side matrix! To ensure you're keeping within acceptable mono capabilities, it's a good idea to insert a mid/side plugin at the very end of your master chain. Every now and again, mute the side channel on this plugin; this will allow you to hear how your master will sound on a mono system. With this in mind, care must be taken when EQ'ing the mid channel – don't go overboard.

So EQ'ing the mid channel affects the mono capabilities of the track, but does EQ'ing the side channel do the same?

EQ'ing the side channel will not make any change to the way the master sounds in mono, but interestingly, this very fact is why EQ'ing the side does affect the mono capabilities. If you achieve your desired overall tone by making significant adjustments to the side channel, as soon as your master is played on a mono system, all your tonal adjustments will have disappeared.

Changing the level of the side will not make any difference to the sound if the track is summed to mono at some point down the line because to get a mono version of the track, you must add left and right together. So any changes made in the side, like a boost for instance, causes there to be an equal but reversed boost in the opposite channel. Therefore, when the left and right are added together again for whatever reason, any alterations made to the side simply cancels itself out down to there being a reversed version of itself in the opposite channel.

Try it; insert a mid/side plugin at the end of your master chain and mute the side so you're hearing pure mono – a perfect sum of left and right – the same mono you get on a mono night club system or one of the other scenarios mentioned before. You can use a plugin with a mono switch if you have one. Now go back to your mid/side matrix and EQ, and start playing with the tone of the side channel, or the level. You'll hear nothing at the output, but if you change the tone or level of the mid channel, you'll hear it.

I have one last point to cover about mono compatibility. In the 'Stereo Thinking' chapter, I demonstrated how applying a time delay to either left or right will move the sound to one side. I also mentioned how this is not always a practical way to achieve stereo placement during mixing. Here's why:

Let us say you applied this technique to a guitar in your mix. If the final master happens to be played in true mono, the guitar may experience phasing issues or even some weird delay effects. As the guitar in one channel is slightly later than the other, when they both become one channel you no longer experience the guitar in stereo, you hear the guitar with a second version of itself, milliseconds later coming from the same place.

Okay, so now you get the point about stereo to mono compatibility. I've told you the rules but they can be broken. Most of the time your masters will be played on a stereo system so you might as well make at least some use of the amazing things that can be achieved in the mid/side domain, just remember to go easy.

The next trick I'm going to tell you about is how to get more width, and make it fat too. This involves processing the side channel, not the mid.

Midrange meat

In the same way that you used an EQ to boost and sweep across the mid channel, this time apply this to the side channel with the mid channel muted, particularly around the mids and low mids. Also consider applying a low cut to make sure the lows are centred. Use a couple of EQ bands to find the resonances in the side channel, or just to find where the most is going on. Apply reductions according to your findings.

Now that you've effectively reduced some of the midrange of the side channel by applying carefully selected dips to resonant frequencies and the like, un-mute the mid channel, then use a shelf to raise the entire low mid of the side channel.

Keep going and listen to how the midrange of the track begins to fatten up. By finding the resonances first, you've given yourself headroom to allow for a nice hefty boost in the lower midrange of the side channel. This trick can do wonders for heavy guitar styles. It's a way to beef up the low mids without an awkward EQ across the whole mix, and it works on many other styles too. However, it won't work on everything – a nice defined mix is needed in the first place. Mushy low mids will not sound good when doing this, but when it does work, it's very impressive.

One really good tip when making adjustments in the mid/side domain is to keep checking how it's sounding on a good pair of headphones. Whereas a nice hefty lift in the lower mids of the side giving some added meat to the guitars or synths may sound awesome on your monitors, your headphones may tell you something different. If it sounds like you've gone too far on headphones, then you probably have.

You should be getting the idea by now, that tapping into the stereo elements opens up all new possibilities. You can create a bigger sound before you've even switched on your compressor or limiter.

We're going to leave the stereo field for now and head back over to the world of dynamics, before coming back to the world of stereo to discuss how dynamics processors are used in mid/side situations too.

Chapter 13: Expansion

For those of you who are unfamiliar, expansion is the same as compression in the way you program the processor, but the opposite of compression in its effect. When the signal reaches the set threshold, instead of reducing the gain, the processor increases the gain. How much and how fast is determined in the same way as with compression – the ratio, attack and release. An expander will typically look the same as a compressor, or it might be the case that your compressor has the option of switching to expansion.

One way you could visualise the effect is like an 'un-compressor'. If you compressed a signal, you could then un-compress it again using expansion.

Expansion in mastering is usually carried out using a multi-band expander. For this reason, I will concentrate only on using multi-band expansion. Whenever I use the term 'expansion', I am referring to multi-band expansion.

WAVES offer a very useful multi-band compressor which includes the option of switching to expansion. It comes as part of their 'Masters' bundle. Rather than dialling in ratios as with normal compression, you simply have a fixed negative value. WAVES refer to it as 'RANGE'. The great thing is, if you input a positive value into the RANGE parameter, that particular band becomes an expander. You can have a mixture of compression and expansion inside the same multi-band processor.

Below is a snapshot of WAVES' multi-band compressor; the first two bands are set to be compressing, the third band is set to be expanding, and the fourth and fifth are set to be compressing. Notice the RANGE parameter for each:

One might think that the typical uses for expansion are the opposite of multi-band compression, well sometimes yes, but often no.

In Part 1, I gave the example of a mix having overpowering hi-hats and suggested the use of some multi-band compression to get them under control. Well, if the mix sounded like the hi-hats need some more life instead, the obvious answer might be to apply some expansion, however that could be the wrong decision. That's because you risk expanding other elements in the mix too, like vocal 'ess', or other percussive sounds like the crashes. These other elements hitting the threshold may not be a problem when compressing, but when expanding, the mix could start to sound really wild.

It's rare that you would ever need to use expansion in the highs, there are better ways of injecting more life up there like using EQ or may be even harmonic excitement (discussed soon), but there have been times when I've needed to apply expansion in the highs to revive over-compressed percussion. Expansion lives more suitably across the mids and lows; these are the areas that typically need more 'life' as opposed to 'level' when it comes to the mastering stages. More often than not, it's the drums that require attention.

One of the problems you may find when mastering is that the application of a single band compressor, and/or a limiter can sap the life out of the drums, particularly the snare drum. Even though we can use the very nature of a compressor's behaviour to bring attention to these RMS heavy elements, like a kick and snare, the fact is there's only so much compression you can apply without loosing the punch.

What can also affect the snare's punch is the application of notches using your EQ to deal with resonant frequencies. At times the resonance itself can be caused by other elements in the mix clashing with the snare. Cutting out certain frequencies might be scooping out important areas of the body of the snare. The drums are quite sensitive to processing, loosing their punch at the drop of a hat.

One way to get around this is to apply some expansion to the particular elements that are struggling to shine down to other mastering processes. For the snare you would apply the expansion across the midrange and set the threshold so it's just receiving the snare if you can. The attack would be set as fast as it can go, release pretty quick too. The ratio might typically be more than with compression – set to taste. Now every time the snare hits the threshold, we get a sudden lift in the snare's level – like it's snapping back. This can inject the life back into a depleted snare.

Sometimes you'll find the kick to be in need of the same treatment. But we can't blame the mastering compression as often for the kick as we can for the snare. In my experience, you can apply a considerable amount of compression before you loose the kick's power, providing your attack is not too short. When the kick is in need of expansion, it is usually down to the mix. The kick doesn't share the majority of it's space in the spectrum with anywhere near as many other elements as the snare (if the mix is good). Having its own space in the spectrum gives the kick an advantage when it comes to having impact; it's simply not got as many other things to compete with down there. As the snare is already battling with every other element across the midrange, it's easier for it to get lost in the midst. The main culprit for taking away the kick's power is the bass, and sometimes the guitars or synths if they haven't received appropriate low cuts during the mix down. Resonant frequencies in the bass can mask the kick's punch. For a kick to sound full, there needs to be a healthy amount of low frequencies, easily as far down as 40Hz – preferably further.

It may sound an unusual thing to do during the mix down but applying a very slight low cut to the bass can improve the impact of the kick without having to turn the kick too high up in the mix. This allows the very bottom end of the kick to extend down just that tiny bit further than the bass, improving separation. As long as your kick is filling in the very bottom end, you can actually get away with low cutting a tiny bit of the bass line's sub-sonics. I wouldn't recommend cutting any higher than around 60Hz though.

So with the possibility of the kick being impaired by the other elements it happens to be sharing the spectrum with, and the addition of mastering compression, it's sometimes the case that a little expansion will save the day.

In the same way as expanding a snare, you need to set the expander to respond to just the kick's area in the spectrum, quite low. Or perhaps you might apply it a little further up – the 80 to 150Hz area let's say – in search for the kick's punch. Program it to be triggered by each kick the best you can with a fast attack and release, ratio set to taste.

Aside from reviving the drums, expansion can put the spring back in an over-compressed lifeless mix. The midrange is the most common area for this, but you would probably set the expander to affect a wider area of the spectrum than you would for just the snare for example. You may find it useful to use more than one band for this. The settings might not be as aggressive as for reviving percussion but the attack and release would still need to be pretty fast.

Caution – the midrange is the most busiest area. Remember that the expander will lift all instances of the threshold being reached. You might have improved the mix during the first half of the track let's say, but take care to ensure there is nothing lurking down the line which might jump out like a jack in a box.

Just like the multi-band compressor allows you to tap into the mix and get things under control, the expander gives you all the same control but in the opposite direction. So when you find that your mastering techniques are affecting certain things in the mix that you wish they weren't (drums, stabs, any content that needs to have punch or demands dynamic range), expansion can help correct the affected elements in the mix without having to sacrifice the mastering techniques you are using.

In the next chapter, we stay with the dynamics theme, however we'll be looking at it from a mid/side point of view.

Chapter 14: Mid/Side Dynamics

This chapter outlines some of the problems that can occur when using multi-band compression or expansion, and explains how mid/side techniques can be used to overcome these problems.

Compressing the mid channel

In Part 1 we discussed that by the application of multi-band compression, overpowering hi-hats can be tamed down, and the lows can be tightened up, helping with the track's punch and translation across different systems. One of the points made was how you should take care when compressing the midrange areas. Whereas a healthy serving in the lows can consolidate the kick and bass, or some appropriate measures in the highs can get a grip on the hi-hats and catch the odd vocal ess, when compressing the midrange, you risk taking away the life of the mix. The mids are very important, they need to sound rich and dynamic – full of life. Too much compression across the midrange can result in an over-cooked sound and will probably take the punch away from the snare. However, using compression across the midrange is certainly doable, so long as you go easy and keep a close eye on other elements you might be affecting.

Probably the main reason you would need to apply multi-band compression to the midrange area is in the event of the lead vocal poking out a little too much. It might be under-compressed or just a little too high in the mix. This is where mid/side comes into play.

As lead vocals are usually mixed dead centre, they can be handled more efficiently in the mid channel of a mid/side matrix. Having converted left and right into mid and side, you simply insert your multi-band compressor across the mid channel. The effects of the compression are less harmful to the 'life' of the midrange areas because the ambience and stereo information contained within the side channel remain uncompressed. Keep an eye on the snare though, mid/side or not, compressing the vocal will most likely affect the snare's tone or punch.

On the subject of the snare, as described in the previous chapter, multi-band expansion can inject some life back into matters when we've lost the snare's impact. Let's now think about that same scenario but using the expansion on the mid channel of a mid/side matrix. Can you see how this would return even better results? Providing the snare has been mixed dead centre (not as often as the vocal but still most of the time), the expander can be focused more efficiently on the snare itself, again by not affecting elements within the side channel.

Compressing the side channel

Multi-band compression on the side channel can be used for widening or giving a feeling of more space. Instead of boosting the level of the side, you can instead compress it. With the inclusion of make up gain, you can add a little more weight to the side channel without raising the peaks. Depending on which bands of frequencies you affect, this can give more apparent width or a feeling of more space or air.

Or, you may use multi-band compression to calm down an over-dynamic element that exists mostly in the stereo field. Although rare, I do use multi-band for this purpose from time to time.

Mid/side expansion

Expansion can do quite a nice job of enhancing the width of the track by applying it in the side channel (without altering the gain like when compressing the side). This essentially gives more dynamic range to the stereo information. Care must be taken to ensure there aren't any nasty surprises lying ahead; you don't want the stereo field jumping out at odd times, over-emphasising reverbs and FX when it does.

Mid/side processing conclusion

In conclusion to mid/side processing as a whole, it should be clear now how this opens up an array of new possibilities for giving your mastering more of an edge. A carefully combined mixture of level, tone and dynamic adjustment in the mid/side domain can be a powerful recipe for achieving a bigger and more refined sound. As I keep saying though, mastering is the fusing together of many carefully chosen small adjustments that all add up to an overall major enhancement to the sound.

One last point (which I've already made), keep a close eye on the mono compatibility. Remember that any adjustments you make to the side channel will simply disappear when your master is played on a true mono system. Any adjustments to the mid channel will sound more exaggerated. When sculpting overall tone, be sure you achieve most of it in the left and right domain.

Now we're going to move away from mid/side processing and back into good old left and right processing. The following chapters contain some really clever techniques for achieving a commercial sound. As with mid/side processing, many of these techniques are not common knowledge for most, so if you want to always have the edge, keep them under your hat.

Chapter 15: Parallel Compression

This chapter will teach you about a powerful way of using a single-band compressor – parallel compression. Outlined are some of the downsides to using standard single-band compression and how parallel compression can overcome these. The chapter begins with what parallel compression is and how to set it up, then provides tips on how to get the best out of this incredible technique.

The compression technique demonstrated in Part 1 is technically known as downward compression. The compressor works by pressing down on the louder passages, creating more headroom above the peaks, in turn allowing for an overall increase in perceived level giving a fattened up effect.

The problem with compressing in this way is the very fact that the 'louder passages' are being pressed down, or squashed if you will, whereas the quieter passages remain relatively untouched. Just when the track needs some attitude, or punch (when it has got loud), the compressor kicks in and squashes it down, reducing the impact of the transient hits. Wouldn't it be great if we could flip this around but get the same result – so the quiet bits experience more compression, whereas the louder bits experience less?

The way to achieve this is by switching from downward compression to upward compression. This is where we compress from the bottom up as opposed to the top down. The way to do this is by setting up your compressor in a parallel fashion.

I would now like to introduce you to parallel compression which is sometimes known as 'New York compression'.

This is an extremely useful technique, I probably use it about 80% of the time. Its application is all the same situations as you would use a normal single-band compressor (bringing a mix to a whole for instance, giving it a layer of gel and gaining overall loudness) only this way of compressing is more friendly to the transient impact of the louder passages because it contributes most to the signal during the quieter passages instead.

Here's how it works. In terms of set up, it's really simple. You have your signal split in two, one goes through your compressor and one goes around it to join back with the compressed signal on the other side.

One signal is wet (compressed), one is dry. Here's a block diagram of how it may look:

Depending on whether your DAW has plugin latency compensation, you may need to insert a sample delay plugin into the dry signal to ensure both the wet and dry signals fuse back together in line with each other on the other side. This is to compensate for the delay caused by the processing time of the compressor plugin. If this is the case, to set the correct delay time, feed a signal through the parallel compressor network and set the compressor's ratio to 1:1 with the output at unity gain (0dB), then reverse the phase of either the wet or dry signal. In doing this, you will get silence at the output providing the wet and dry signals are perfectly in line. If not, adjust the delay plugin by samples at a time until you get total silence at the output (two identical summed signals give silence when one's phase is reversed). The setup may look like this:

Some compressors include a dry and wet parameter. By allowing an amount of dry signal through, you are effectively parallel compressing.

Typically, you'll be driving your compressor harder for this than you would for downward compression; in fact it's a good idea to set up your compressor so it's in constant gain reduction – bring down the threshold until it is.

By applying compression in this way, the louder passages give you more dry signal compared to wet (compressed) signal at the compressor's output than when the music is quiet, and it's the dry signal that has all the punch because the transients have not been squashed by a compressor. Try to visualise how the amount of compression changes with the level of the signal – as the signal level into the parallel compressor increases, the dry signal increases more than the compressed signal at the output.

To help you visualise how the signal levels change in respect to each other, I have set up a manual parallel compression network inside my DAW by duplicating the signal using an aux send (similar to setting up mid and side to be used on separate channel strips), then took a snap shot – see below. The first channel strip – Audio 1 (or 'dry' as I've named at the bottom) – is the dry signal. The second channel strip – Aux 1 (or 'wet' as I've named it at the bottom) – is the wet (compressed) signal. The third channel strip is the summed output of the other two channels; this is where we experience the parallel compression effect. I've named it 'ParaComp'. 

Notice the level of the dry and compressed wet signal. During the louder passages there is significantly more dry signal. The output signal (ParaComp) is mostly made up of uncompressed audio during the louder passages.

During quieter passages, the dry signal's level comes back down significantly, whereas the compressed signal's level is only marginally different as it was restricted from becoming louder by the compressor in the first place – see below:

The difference between wet and dry is much less during the quieter passages. The signal at the output (ParaComp) is now made up of more compressed signal than before.

You are still compressing the audio and reaping all the benefits of doing so, only now it's the quieter passages that are more compressed than the louder ones. So we have compressed from the bottom up instead of from the top down.

Another way to help you visualise this effect is to see the signals as waveforms. I have bounced the dry signal, the wet, and the combined output (ParaComp), then loaded them into an empty project and took a snapshot. Here it is:

Notice how the peaks actually look more intact than they would when using a similar amount of compression in a downward fashion. You get all the benefits of the increased RMS from the heavily compressed signal, along with the transient qualities of the uncompressed signal.

To adjust the amount of compression, adjust the output gain of either the wet or dry signal. The more compressed signal you have along side the dry, the more compression. In the snapshot below, I have offset the level of the dry against the wet by around -4dB, resulting in a more compressed signal at the output:

This way of compressing is fantastically useful when you need some attitude in your results, or even if you need transparency; parallel compression does both very well. Here's some settings to dial in for achieving attitude and punch, just when you need it – when the mix has got loud.

Parallel compression for attitude

Threshold: Bring down the threshold so the compressor is almost always in a state of compression, with only the very quietest passages passing by without triggering the compressor. You'll be expecting to get something like 5 to 7dB gain reduction.

Attack: Start with something similar to what we have used in previous examples – around 20 to 50ms. You don't want it too short, it's important we set the compressor to allow for the transients hits to pass before it responds.

Release: Setting the release should be approached in the same way as discussed in Part 1. Zone in on the rhythm, get the compressor moving in sync, responding to the heavy RMS energy of the lows and contributing to the fatness.

Ratio: Set to taste but don't be shy, get that compressor moving, as if it's under the kick and bass-line's spell! Perhaps start with something like 4:1, or may be 5:1.

With downward compression, once you've got your compressor moving with the rhythm, you would pull back on the threshold to achieve the desired amount of compression. For parallel compression, you keep that threshold right down, heavily compressing the mix, then simply introduce the dry signal into the chain which has just bypassed the compressor. This is how you determine how much compression you want. Now you can leave your compressor's settings and alter the amount of compression by simply adjusting the difference between the wet and dry signal.

Wet and dry levels: Between 3 and 5dB gain reduction will require something like a 50/50 blend (both wet and dry faders at the same level). With larger gain reductions, you should expect to need more dry signal to wet.

Remember that you're adding two signals; be sure the combined result isn't going into the red and clipping.

Compressing in this way is much more rewarding for all the hard work of getting your compressor moving with the rhythm. This is because you don't have to pull back on the threshold to decide on the amount of compression; the compressor is pretty much always in a state of compression, imprinting its fattening character across the mix, all of the time.

I can't tell you how incredibly useful this is when you have a very nice compressor at your disposal – an analogue compressor for instance, or a software version of a nice analogue compressor. You can drive your compressor really hard to reveal its wonderful character (curvature of attack and release), whilst retaining a suitable amount of compression and transient impact at the same time. This is when your skills of getting your compressor to move with the rhythm really come into play.

Here's some good settings to dial in when you require transparency (compression that sounds uncompressed); very useful for delicate recordings – classical guitar for example.

Parallel compression for transparency

Threshold: As before, you need to pull the threshold right down only this time you want the compressor to be in a constant state of compression – the very quietest passages must still be triggering the compressor. Expect quite a lot of gain reduction, more than the previous technique.

Attack: As fast as it will go. For this technique, you want the compressor to bite down hard on the transient hits (don't worry, the dry signal replenishes the lost transients).

Release: This should be approached a little differently to previous techniques. We still want the release to compliment the natural rhythm of the track but be careful not to have it set too fast as this might take away the transparency. Typically, fairly slow release settings work best for this technique, 2s or more.

Ratio: Set to taste.

Wet and dry levels: Suitable wet and dry levels are likely to be similar to the previous technique.

The parallel compression technique can be used anywhere, not just in mastering; you can use it on vocals or guitars for instance. Compressing the drums in this way is very effective. It's probably the most useful way of using compression. Enjoy!

Chapter 16: Mastering Reverb

Reverb is essentially a means of adding the illusion of a walled-in physical space with its own size, shape and sonic reflections, such as those found in a church or a small room in the real world. When applied correctly, the sound source will appear as if it really is in the desired environment. The effects can be minimal or dramatic depending on your needs. Many 'all-in-one' mastering plugins include the option of applying reverb. So when would such an effect be required in mastering?

The most important thing to bear in mind is that, most of the time, the recording/mixing engineer or the producer is responsible for the reverb effects heard in a piece of music. Often, different elements of a mix will have different reverb effects applied which allows the separate elements to have their own characteristics and tonality. In the hands of a good mixing engineer, these distinct parts will gel together and create an overall atmosphere of depth and space.

The use of reverb in mastering is a decision that should not be taken lightly, especially when working with material that was recorded with microphones in the real world. A recording such as this (a good example being a live drum kit) will have already captured a natural reverb from the room. A mix engineer could then have chosen to emphasise these natural reflections with additional reverb effects during the mixing process. It's likely that the final mix will already contain a sense of its own space and character due to the combination of recorded and added reverb.

When dealing with a mix that contains a strong sense of ‘room’ character there is little need for additional reverb at the mastering stage. In fact there is the danger of creating a problematic situation known as a ‘room within a room’. This can happen when the reverb applied from the mixing stage, encounters the additional reverb applied by the mastering engineer. It can sound as if one space is somehow reverberating into another, and will sound unnatural to the ear since this can’t happen in the real world. When used inappropriately, mastering reverb can lead to a lack of definition in the perceived depth and stereo image already inherent in the mix.

But with this in mind, an especially ‘dry’ rock mix could be helped towards further weight and sonic unity with a subtle amount of reverb – if it is applied carefully. In any mastering reverb situation, the amount of reverb will be minimal, probably never above a 9 to 91% blend of reverb to dry mix.

Mastering reverb can be a useful tool when working with electronic styles. Music created with samples, synths, outboard equipment and software can sound interesting for the very fact that vastly different sounds and textures exist in the same recording. In such situations, an amount of reverb can improve the sense of togetherness.

Another use of reverb in mastering is to draw more attention to an area of the spectrum in a situation where EQ alone might not be giving the desired results. For example, if the material being mastered needs to sound a little brighter, yet using EQ produces a result that sounds too harsh, applying a slight amount of reverb – which has been filtered with EQ to only allow the higher frequencies to be heard – can reinforce the brightness of the mix.

The idea of using reverb to purposefully alter the tone of the mix leads on to another point: On the flip side, the decision to use reverb may alter the tone of the mix against your desire. This is why EQ'ing the reverb is a very important point to be made. Using EQ, you can shape the reverb to blend better with the mix. Apply smooth reductions to areas where the application has drawn too much attention to itself. You will almost certainly low cut the reverb in any situation as not to add unnecessary muddiness to the lows and lower mids – perhaps a cut up to somewhere between 200 to 400Hz.

One of the best reverbs to use (not just in mastering) is a convolution reverb. It’s a plugin that allows you to load in short samples called 'impulse responses' (IR's for short) that have been recorded in real-world spaces. These samples are simply small audio files that have captured the immediate attack of a real sound such as a click, and also the first few reflections from the walls of a real-world space. The plugin then uses a process called ‘convolution’ to produce an algorithm from the characteristics of the sample, which allows you to treat any audio with the characteristics of that same real-world space.

Other kinds are known as algorithmic reverbs, which already have their parameters pre-defined. They use complex programming to create their reverb effect. But the benefit of using a convolution reverb is that the reverb can be changed drastically through use of different impulse responses, giving you ever-expanding reverb options based on the number of IRs you collect.

Typically for mastering, your reverb amount will be minimal. You're not likely to want to give the impression of the sound being inside a large reverberant space. Reverb times will be fairly short and the reverb type itself will be something more like a medium sized room with only a small amount of surface reflection.

In conclusion, reverb in mastering is a powerful effect but not something you will find yourself using all that often. Like compression, a suitable amount of reverb can help bring a mix to whole. It does this by giving the sense of all the different sound sources coming from the same physical space. Take care because the addition of reverb when it's not essential can have quite a negative effect on the mix.

Here's a link to a free convolution reverb plugin called 'SIR' (windows only I'm afraid):

http://www.knufinke.de/sir/sir1.php

Chapter 17: Ultra Depth Continued

If you haven't already, it's important you read the 'Ultra Depth' chapter at the end of Part 1 before you read this chapter. Even if you have read it, you might find it useful to have another quick scan over it to familiarise yourself with how Ultra Depth is set up before you proceed through this chapter.

Ultra Depth is a simple, yet effective technique for adding extra perceived depth to the mix at the mastering stages. The one problem you may find is that it doesn't always work on everything first time. Outlined in this chapter are ways to tweak the settings to make it more versatile.

However, before I get to all that, I'm first going to explain why this technique gives more perceived depth and talk a little bit about the previous studies I have read that led me to discovering this simple technique.

Papers and patents about enhancing depth using psychoacoustic manipulation

An AES (Audio Engineering Society) paper written by Paul M. Boers for Philips Research Laboratories dating back to 1983 titled 'The Influence of Antiphase Crosstalk on the Localization Cues in Stereo Signals' describes how the stereo width of a dual speaker system is limited to the distance between the two speakers and that all sound sources can only be located somewhere within those boundaries. The paper explains how the addition of cross-fed, delayed and anti-phase versions of the original signal can extend the perceived stereo width to exist beyond the distance between the two speakers. Boers refers to this as 'acoustic interference at the position of the listener'. He also talks of how this technique emphasises early reflections and reverberation (the ambience); he states 'an accompanying effect is the increase of the stereo difference signal with respect to the sum signal, (think about mid/side) thus emphasising signals not equally present in both channels, such as the early reflections and reverberation'.

Here's a link to Boers' paper:

http://www.aes.org/e-lib/browse.cfm?elib=11793

A 1995 US patent (US Pat. 5,412,731) awarded to Stephen W. Desper, Tujunga, Calif, titled 'Automatic Stereophonic Manipulation System and Apparatus for Image Enhancement' describes a process for enhancing the perceived sound field to be contained within boundaries larger than those normally reproduced by stereo speakers. Similar to the Boers' study, this process makes use of cross-fed, delayed and anti-phase reproductions of the original signal referred to in the patent as 'phantom boundaries'. The most interesting part of this patent for me is the explanation of how this process creates a 'virtual image' behind the listener's head, however the explanation is probably a little long winded for the purpose of this book. In simple terms, the paper describes how the left signal reaching the left ear and then the right, and how the anti-phase right signal reaching the right ear and then the left, provides information which the brain calculates as being sound coming from the rear – very interesting.

The patent also describes in detail its use of the 'precedence effect' in which a delayed second signal within a certain time (about 30ms) fuses with the original sound source. The brain's ability to fuse these two sounds together is referred to in the patent as 'temporal fusion'. This is similar to what I described in the 'Stereo Thinking' chapter at the beginning of Part 2 as the Haas effect.

This patent is quite a complicated read. If you want to have a look, you can find it here:

http://www.archpatent.com/patents/5412731

These processes make clever use of the Haas effect. They both contain a system which creates multiple reproductions of the original sound source which are delayed, attenuated, anti-phased and cross-fed into the opposite channel of the stereo signal. Ultra Depth is a stripped down and simplified version of these techniques.

Through my own experimentation of these two ambience enhancement systems, I have discovered that the most significant reproduction of the original sound source is the first one. In other words, it's the first 'early reflection', or 'phantom boundary' which provides the majority of the perceived effect. The following reproductions are still important for revealing more and more of the inherent ambience, however you can achieve a suitable amount of extra width, depth and space by simply introducing one single stereo delayed reproduction of the original sound source back into the signal. Rather than creating multiple phantom boundaries, or early reflections, Ultra Depth simply creates one extra early stereo reflection. This simplicity is why it can be easily built using plugins you already have as part of your DAW. Here's the process again:

Creating extra depth using psychoacoustic manipulation

So what's going on here; why does cross-feeding attenuated, delayed and phase-inverted versions of left and right back into each other give the sense of more width and depth?

In an acoustic environment, what we perceive as being the ambience within the music is the sound waves that have travelled and been affected by the surroundings before they reach our ears. Think back to the 'Stereo Thinking' chapter – after the direct sound waves from the snare had reached our ears, they were immediately followed by the sound waves that had bounced off the walls. These are what we call early reflections and, depending on the size of the room, are typically somewhere between 12 and 30ms (the Haas delay time). It was these sound waves that enhanced the sound, particularly the ones coming in from the sides as they create the biggest time difference between our ears. After about 30ms, we begin to perceive the reflections as an echo. This is where the brain separates the two sounds instead of fusing them together. The Haas delay is typically shorter for percussive sounds, and longer for sustained notes like the strings of an orchestra.

Using a simple but clever technique, we can artificially boost the perceived level of just the early reflections in a recording. To do this you simply set up the Ultra Depth process as illustrated in the 'Ultra Depth' chapter at the end of Part 1. The attenuated and delayed versions of left and right being cross-fed into their opposite channels injects new early reflections into the mix. The brain fuses the new early reflections with the original sound source, improving the perceived stereo imaging and boosting the apparent level of the existing early reflections. Reversing the phase of either the left or right within one of the cross-fed channels is what gives the effect its width and depth as it creates the illusion of the reflections bouncing around out of phase. This helps towards the impression that sound is coming from all around. The phase reverse creates a bigger overall difference between the left and right channels creating more stereo information.

Some larger studios are fitted with 'Haas reflectors' which are angled panels either side of the listening space. The extra early reflections created by these panels can improve the stereo imaging of the listening space. Whereas absorption of bouncing sound waves helps toward a critical listening environment, many believe that early reflections are very important and should not be removed, merely controlled.

Mono compatibility

Think back to the mono compatibility issues associated with using mid/side to widen the stereo image. To jog your memory:

Raising the level of the side will typically brighten the mix, but in mono, the brightness disappears. You also risk swamping it out in reverb and FX.

EQ'ing the side and widening certain areas of the spectrum is great but it vanishes when summed to mono leaving an obvious tonal difference between mono and stereo.

Scooping out frequencies of the mid channel can widen the image, but in mono the cuts are exaggerated.

Ultra Depth doesn't sum to mono perfectly well either but I find the side effects to be more appealing than the mono compatibility issues associated with mid/side processing. In fact, you could argue that Ultra Depth improves the mono compatibility of a mix.

When summing a master containing the Ultra Depth effect to mono, you get a cancelling of all the mono information contained within the artificial early reflection leaving only the difference. In effect, the mono signal will contain an extra delayed 'side' channel within it. This is down to the phase reverse inserted into either the left or right channel of the early reflection.

This side effect can be considered appealing as the delayed 'side' channel within the mono mix can actually still add perceived depth to the mix – even though we have summed the mix to mono. It can give the illusion of extra ambience arriving a little later in the form of an early reflection, enlarging the perceived depth of the sound stage.

Be sure to experiment with a mono switch inserted at the end of your master chain to understand how your mix translates to a mono system when using the Ultra Depth effect.

When can Ultra Depth be used? 

Ultra Depth can be appropriate on almost any mix to some degree, but as previously said, it will not always work on everything first time. An example of where Ultra Depth needs tweaking is when applying it to minimal styles of dance music, like tech or deep house. The very nature of these styles is to be very minimal in their content. The application of Ultra Depth (when not setup correctly) sounds really rather strange on such styles and is more exaggerated on headphones. You might possibly hear clicking noises created by the delayed versions of the minimal percussive content from the left and right being cross-fed into their opposite channels.

Any part of any style of music can give this strange clicking effect when the music happens to break down to a more minimal state, or contains any passages where there may be lone percussive sounds happening. This effect works best when the mix is busy enough to blend with the artificial delayed versions of the original sound source. EQ'ing the effect can help with this.

EQ'ing the effect

The first thing I'll mention here is the low end. You may recall how I talked about applying a low cut to the side channel when using mid/side processing. The very bottom end of the spectrum will usually not benefit from the addition of stereo information, or the boosting of ambience. I'm referring to around 100Hz and below. It's practical to keep the very lows pretty centred and fairly dry in their sound. The lows are like the foundations of the mix, they need to be solid and reliable. Adding stereo information down there or boosting ambience could muddy things up and take away the solidity. Plus, when listening on headphones, having stereo presence in the extreme lows can sound strange as this is something the ears are not used to hearing in the real world.

The next thing you need to think about is the high mids and the highs. When there exists a lot of percussive instruments, or sounds with sharp attacks, the effect can become exposed as being artificial. By reducing some of the high mids, and rolling off the highs, you are concentrating most of the effect into the mids which is usually the busiest part of the mix and so the effect will not be so easily exposed as being artificial. Having done this, you'll be able to add more of the effect to minimal or percussive styles of music. In my experience, it's a good idea to always apply a low cut, roll off at least some of the top end and shelve out some high mid. Doing this makes the effect more natural sounding and therefore more useable.

EQ'ing the effect does affect the overall tone of the mix to some extent but not so much that it becomes problematic in mono situations – like when using mid/side processing. With this in mind, remember to go easy. Heavy handed EQ adjustments to the effect might cause there to be obvious tonal difference when switching between mono and stereo.

Other ways to improve versatility

EQ'ing can help blend the effect into the mix but it will not totally eliminate the strange side effects you can get when applying it to certain styles or content. But the beauty of using Ultra Depth is the ability to alter the delay times and levels to whatever value you want. This control is how you can tailor the sound of Ultra Depth making it more versatile.

A little tip I gave you in Part 1 when explaining how to build Ultra Depth was to use short delay times for dance music styles. Try 15ms or even less. The general rule is the shorter the delay, the more transparent when applying to minimal or percussive content. The longer the delay, the more depth you can achieve – providing the style of music is appropriate. Try to keep within the Haas delay time.

Always listen to what's happening on headphones. Sometimes this effect can blow you away on speakers but as soon as you switch to cans, you hear the strange knocking and clicking effects caused by the delayed cross-fed sounds bouncing around. More often than not, you just need to back off with the effect level a little to get a suitable level in your headphones.

Automation

If no matter what settings you dial in, you just can't get the effect to blend, automation can help. You can automate the aux send to the effect bus to alter in level throughout the track. Simply drop out some of the effect during areas where the track breaks down and exposes the effect. Then bring the effect back in during busier passages where the effect can blend.

Doing this can actually give another new effect. The sudden increase in the perceived depth right at the point where a busier part of the track kicks in can help provide the desired impact at that point. This is most suitable in dance electro styles where the rules of stereo imaging are a little more 'anything goes', but it's certainly a valid option in any style of music.

The beauty of simplicity

As mentioned, Ultra Depth only creates one single artificial early reflection unlike the processes described earlier. Strangely enough though, I find that sometimes this can make Ultra Depth more useful as a long tail of multiple bouncing early reflections can make the process difficult to use on everything. But that doesn't mean to say you can't try to improve on the process I've described – why not try doubling the reflections? You can easily set up a second matrix in the same fashion being fed by the original source, in order to create a second set of early reflections arriving at a later time (set the time delay to be twice that of the first set of reflections). Naturally these second reflections would be lower in level, as would be expected in the real world. I would also suggest having the phase reverse on the opposite channel for the second set of reflections.

Bringing the mix to whole

Similar to how reverb can help bring a mix to whole, so can Ultra Depth. It can improve the sense of togetherness by giving the impression of all the sounds existing more convincingly in the same physical space.

Converts mono to stereo

Here's an interesting thing, when feeding a mono signal into Ultra Depth, you will get a stereo signal at the output. You can turn a mono signal into stereo. Try it; feed a mono vocal track into the inputs and listen to the output. The output will sound like the vocal was recorded using a stereo mic.

Summary

The introduction of delayed second signals into signal paths from the opposite channels with an added phase reversal, actually enhances our perception of the natural ambience within the music. It's a natural sounding and mono friendly way to find extra space, width and depth. Obviously nothing can replace carefully crafted space, width and depth in the mix, but this technique is probably the next best thing before reverting to mid/side processing.

Chapter 18: Saturation in Mastering

Once upon a time, everyone recorded to tape.

Let me start by saying recording to analogue tape, and the idea of tape saturation is a massive and very technical subject with a fascinating history which could easily fill a book by itself. I'm going to keep it simple and get you thinking about using saturation plugins in mastering situations.

Saturation is a lovely and very useful effect. It's a fascinating phenomenon that is basically achieved by trying to squeeze too much on to magnetic tape or force too high a signal through tube circuitry. In this chapter I'm going to focus mainly on tape saturation.

It's quite likely that many of you will have never used magnetic tape for recording; in fact it may be something you've only read about. Perhaps some of you might have used it at college or university. Advances in digital technology has brought an end to the necessity of recording to magnetic tape. A typical DAW is so powerful that there's no need to record to anything else. But there's something that recording to tape can do that recording digitally can't. This is why recording to tape will never disappear completely, and why there are plugins designed to replicate its sound.

To really appreciate the beauty of tape saturation, it's good to first think about digital clipping. When recording digitally, there's an absolute limit that the signal can reach. When the signal level on your mixer channel reaches the top of the meter, a red light is triggered indicating that the signal has reached the limit. The reason why the light is red is because it's considered bad to allow the signal to hit the top. Although just touching the top can be fairly inaudible, when the signal is really pushed against this limit, the sound can be very nasty. This is the sound of digital clipping (we covered this briefly in the 'Audio Suitability for Mastering' chapter).

Here's a snapshot of a waveform I bounced out of my DAW. I purposefully raised the level of the signal up against the limit to cause digital clipping:

The peaks of the waves are literally clipped off.

Most DAW's now include a 'soft limit' which allows for a signal to hit the top and go over a little without sounding nasty, providing you don't go too far.

Just like recording digitally, recording to tape has an absolute limit as well – there's simply only so much you can magnetise the tape. But the difference here is what happens when you reach the limit. When recording to tape, there isn't that solid ceiling that cannot be passed as with digital, there's a soft cushion before it – kind of like 'soft clipping'. As you push the signal up against the limit of the tape, the peaks are squashed down rather than clipped off. The effect is a bit like compression only there's no attack and release time – the effect is instantaneous.

When using tube circuitry, the same can be achieved by squeezing enough signal through the tubes to reach the absolute limit. The soft clipping effect caused by the tubes gives a similarly pleasing compression like effect. This could be referred to as 'tube distortion' although, when you are hearing obvious distortion, you may have gone too far – and the same goes for tape.

Just like single-band compression, saturation can be used to gain loudness without raising the peaks and can help bring a mix to a whole. But there is a key difference between the two which is why they both have different uses.

When a signal is recorded to tape at high levels, an amount of compression can occur but only the individual peak of each wave that has reached the limit experiences the compression, not the rest of the waveform below.

When a signal has experienced normal compression, the entire signal is affected once the limit (threshold) has been reached as opposed to just the peaks. The compressor reacts by reducing the gain which affects everything.

With compression, you can determine the attack and release time which gives compression a kind of rhythmical effect. The compressor will remain in a state of compression after the signal drops below the threshold for a length of time determined by the release parameter. Saturation is instantaneous and cannot be programmed to follow and compliment the rhythm of the track like compression.

Another very important point to mention with saturation is how it colours the sound. What I'm referring to here is that popular analogue warmth. There are a whole array of factors to think about when actually recording to tape, and it's these that give tape its distinctive warm character – things like harmonic distortion, weakened high frequency response, the addition of background noise like hiss, and of course the saturation of the peaks of the waves when recording at high levels will colour the sound too. In fact, the imperfections that we get from using tape for recording are what we miss when we record digitally. Ironically, it's the imperfections of analogue tape that makes it sound so good.

Now we know how saturation and single-band compression differ in their mechanics and sound, let's now imagine a scenario:

Let's say we have an over-dynamic mix – lots of loud percussive hits – sounding like it's not very well glued together. We basically need to reign things in a little.

Using a single-band compressor when the mix is in such a loose and dynamic state can cause problems, simply down to the fact that when the threshold is reached, the compression affects everything. What can happen is that every percussive hit that triggers the compressor may actually punch a hole in the rest of the mix. This may result in an undesirable pumping effect as the compressor reacts by reducing the entire level. Perhaps a very slow attack might prevent the hole punching effect but then we're not really compressing the track as we intended. Multi-band compression would probably do a tidier job but have a think about saturation.

Using saturation in the same scenario might be more suitable as every percussive hit would not give this hole punching effect as can happen with compression. This is because only the peaks of the individual waves that reach the top are squashed and nothing beneath.

Occasionally, a mix engineer will use a single-band compressor across the master bus to help the mix gel. This is fine when the mix is relatively tidy in the first place and the compressor is not set too aggressively, but when we have the scenario as above, using a compressor at the start of the master chain (or across the masters during mix down) could be a mistake. Saturation might be more suitable. Single-band compression is most powerful when the mix is tidy enough that you can use the compressor for fattening and loudening as opposed to tidying.

So if your mix is sounding a little loose, perhaps a helping of saturation at the start of the master chain will reign things in a little, making the rest of the mastering process more manageable. And there are other places where saturation comes into play, for example you could use it instead of a limiter to gain more RMS.

Aside from its control and loudening capabilities, probably the main reason most turn to saturation is for its analogue sound. That fuzzy warmth you get when pushing up against the tape's limit can be very flattering when a mix is sounding too digital. Be aware of how it colours the sound though; some situations may not benefit from the characteristic imperfections of tape saturation. If the mix is a little lacking in top end, or is already quite noisy, then the replication of recording to tape may not be so desirable. However, the beauty of using plugins is that you have easy control over many of the variables; you can choose the amount of hiss for instance.

The key thing is to understand how saturation works and how its mechanics and sound differs from compression so you can make more informed decisions about when to use it.

Probably the most popular tape saturation plugin is Avid's 'Reel Tape Saturation'. Take a look:

http://www.avid.com/US/products/Reel-Tape-Saturation

For those of you who are using mac, here's a free one worth checking out called 'TapeHead Medium' by Steven M. Massey:

http://diamond.he.net/~smassey/au.html

Chapter 19: Exciters and Enhancers

Exciters (also known as harmonic exciters or enhancers) raise the perceived level of chosen areas of the spectrum. They can manipulate the harmonic content to give more presence and perceived level, usually to the mids and highs.

Originally built back in the 1970s using tube circuitry, the 'Aphex Aural Exciter' was one of the first and was only available to rent in the beginning, charging studios by the minute of recording time for its use. Exciters are now readily available in plugin form.

The truth is, you shouldn't be too quick to grab an exciter as it is really meant for restoration purposes such as brightening up an old tape recording. Years ago, when tape was the main recording medium, the amount of overdubs a mix might experience could lead to an undesirable loss in high end; an exciter could bring it back. The added higher harmonics achieved by using exciters can brighten or add sparkle to a mix. Loss of high end is not so much of a problem in the digital age. I mainly use EQ or multi-band compression/expansion for spectral processing.

Although rare, there are times when a mix can benefit from a subtle amount of harmonic excitement, even though we seldom record to tape these days. There are other reasons why a mix can end up sounding a bit worn out, and it seems that some of the 'all-in-one' mastering plugins include exciters so it's worth giving them a chance. But the unlikelihood of needing to use one in mastering is why I'll be keeping this chapter short.

They can be used in a similar way to an EQ only they are typically used for enhancing, or boosting frequencies, whereas your EQ will be used for reducing as well. The different exciters/enhancers out there all aim to achieve more or less the same result, to brighten up the sound. However, not all of them process the audio in precisely the same way. Generally though, they will all contain some of, or all of the following processes – phase manipulation, dynamic EQ and harmonic synthesis.

Phase manipulation: This process attempts to correct phase shifting that can occur during playback on an audio system's speakers, restoring clarity. The audio signal is split into three bands, then the application of short time delays will realign the phase of each band.

Dynamic EQ: This means varying EQ in response to level. The signal is split into two, one is compressed and EQ'd then added back to the original signal.

Harmonic synthesis: This means the addition of synthesized harmonics extending further up the spectrum than the original audio. The audio signal is split in two – one signal is filtered to allow only high frequencies to pass, it is then fed into a harmonic generating circuit. The new harmonics are filtered to remove lower frequencies before being introduced back into the original signal.

Hardware units

The Aphex uses harmonic synthesis. BBE are a company who offer one called the 'Sonic Maximizer' which uses phase manipulation and dynamic EQ. SPL's 'Vitalizer' uses both dynamic EQ and harmonic synthesis on quite a complex level. There are a whole array of different kinds of exciters/enhancers in software form – you will probably already have a basic one as part of your DAW.

Exciters have probably earned a better place in areas other than mastering. In mixing, they can be used to add presence to a lead vocal to help it stand out in the mix. They're also sometimes used in PA systems giving the perception of better sound quality by counteracting the negative effect of budget speakers. They are quite commonly used in media playing devices or software.

So if your mix is lacking in sparkle, EQ will probably work, but it's certainly worth exploring what an exciter can bring to the table.

Chapter 20: Independent Left and Right Settings

You are likely to come across the option of applying settings to left and right separately on your processor. With your new found awareness of stereo imaging, it's important to have a look at the effects of processing left and right independently.

Stereo EQ

Using an EQ with the option of processing left and right separately can be an advantage when the particular part of the mix that needs adjustment has been panned some of the way to either left or right. You could notch out a particular offending frequency more in the channel where the sound in question exists and less in the opposite channel. Be aware though, EQ'ing in this manner can affect the stereo image. Cutting out more of a band of frequencies on one side can move that particular element back in toward the centre of the image.

Stereo compression

Multi-band compression can sometimes be more effective when the option of processing left and right separately is provided, especially in the highs. It may be that the very thing needing a little control to tame it down might well exist more in the left or the right allowing you to target it more efficiently. The same goes for multi-band expansion. Again, be aware of how you are affecting the overall stereo image. Obvious compressor movement on one side of the mix can sound as if the targeted area is shifting in and out from the centre.

When using single-band compression, things are a little different. Because the single-band compressor is more of a global effect, it's unlikely you're going to need to apply different settings to left and right independently, unless there's drastically different things happening between the two channels. Using the compressor to help things like the kick and bass translate, or to generally fatten the mix up, means it's best to keep left and right doing the same thing.

One option you will most likely be presented with on a stereo single-band compressor is the option to link left and right. This is where the compression rate of left and right becomes fixed together. The compressor will sum the left and right to mono to determine the amount of compression, it then applies this rate equally to both channels. It's probably a good idea to keep them linked to prevent any RMS heavy elements causing one side of the compressor to shift the entire stereo image around. Keeping them linked generally gives better focus to the stereo image – it keeps the centre in the centre.

Chapter 21: The Mix

Who is really responsible for a fantastic sounding master?

Mixing is justifiably a bigger subject than mastering and would easily fill a book or two by itself. For that reason, rather than give a tutorial on mixing, I am simply going to outline some of the most common problems I have come across when receiving mixes from clients. I won't go into too much detail with things like, what frequencies to boost, or cut to improve the mix, or any other mixing tricks. I'm just going to try and get across an overall concept of why some mixes produce good masters and why others don't. In doing so, I hope to shed a new light on this massive subject.

In a perfect world, all the problematic mixes I receive will go back to the mix engineer with a report on the issues I have found, followed by a remix, and then a great sounding master. The truth is, sometimes I have no choice but to make the best of a bad situation and master, to the best of my ability, a problematic mix. Having mastered many problematic mixes I am able to give some advice on why certain mixes don't produce the best results, and why others do.

Mixing sounds together

There's a fundamental rule when it comes to playing different sounds together at the same time. The lower down the spectrum the sounds lie, the more difficult they are to blend together. I touched on this briefly in Part 1. I used a bass guitar playing two notes simultaneously as an example. If you play two different notes way down at the bottom of the bass guitar's range then the two sounds will not sound clear, or very musical being played at the same time. Play the same two notes a few octaves up and all of a sudden it's not so unpleasant to the ears hearing these two notes. It's unlikely in any song that the bass player will be playing more than one note at one time, but there are other factors that can cause there to be more than one thing fighting for space in the bottom end of the frequency spectrum.

Occasionally, I will receive a mix where the choice of kick drum is that of the tuned category. By that I mean, will have an obvious frequency or musical note contained within the kick's sound. These are usually synthesized kicks or sampled (a natural sounding, and correctly recorded acoustic kick should not have any obvious tuned frequency or musical note). When a mix contains a tuned kick, problems can occur as the note usually clashes with that of the bass-line. These conflicting frequencies at the very bottom end of the spectrum cause it to sound unclear and muddy. Tuned kicks have their place. In house styles of dance music, tuned kicks can sometimes be quite suitable so long as effort has been made to ensure the kick and bass-line work together.

Another rule is the more natural each sound is – as in how well recorded – the more easily they will mix. If when you record the sound of an acoustic guitar, the mic picks up some undesirable resonance of an untreated room causing certain frequencies to resonate, then this guitar may well be stubborn during the mixing stages.

When listening to a musical instrument in the real world, you would probably not notice room resonance unless you have a trained ear. To effectively record a musical instrument, you need to avoid placing the mic inside areas where the room is causing certain frequencies to jump out. Poor microphone placement will force the mixing process to be quite difficult.

A musical instrument itself may cause certain frequencies to jump out a little when the mic is placed close in to the body of the instrument. The resonating body of a musical instrument is how it produces sound in the first place, but there has to be some distance between a microphone and the instrument for it to sound correct. Only once the sound has travelled a certain distance away from the instrument will it form the natural sound of that particular instrument. Depending on what part of the instrument body you point the mic at, you will get a different sound. Some areas will be desirable helping you towards a nice mix, but some will not and will cause trouble.

A good mix starts here – microphone choice, microphone placement, suitable room, good musicians. If you get all this right then the mixing becomes much easier. You end up needing much less processing like EQ to persuade the mix to, well, mix.

When using the EQ technique described in Part 1 (sweeping a boosted EQ across the spectrum), these undesirable resonant frequencies picked up by poor microphone technique are one of the things you are looking for. A very good mix will reveal little, or no unpleasant resonant frequencies when applying this technique – this is the best result you can hope for.

Separation

One major asset that all good mixes have in common is good separation. This is where all the many different things happening are clear and easy to pick out for the listener. So why do all the different sounds become so unclear and hard to pick out in the first place?

Resonance is one culprit. The resonating frequencies mask or blur other sounds occupying the same area in the spectrum. Also, when we hear sounds in the real world, the fact that they are all coming from different locations is one of the ways that the brain separates the sounds. When all the sounds are coming from a single set of speakers, some extra effort has to be made to allow the listener to separate all the different things going on. This leads to an important point – depth. If effort has been made to give the mix depth then we are able to separate all the different sounds more easily. But before we get on to creating depth within a mix (not using Ultra Depth!), I'd first like to raise some awareness about the idea of separation and how it is so easily lost.

When two elements in the mix occupy very similar frequency ranges, it can become difficult for the listener to distinguish between the two. When a mix has many elements struggling to be defined, the mix may sound muddy, especially when it's lower down in the spectrum. Throw in some unwanted resonance picked up during recording and the mix will become really rather stubborn.

Two sounds might occupy the same frequency range but have different textures, this would help them separate. Or, they might sit in different places in the stereo field which will help with separation too. Using the pan pot to separate sounds is good but you need to always ensure your mix sounds good in mono too, this is why EQ is an important tool for good separation. Some advice I was once given was to start the mix in mono, get as good a mix as you can, then move to stereo.

It's important to understand that it's okay for instrument's frequency ranges to overlap but you still need to be able to tell them apart. The tones of sounds can help with separation. Two sounds occupying similar frequency ranges could be EQ'd to have different tones or texture, one could be given warmth, the other could be brightened.

Muddiness can be the result of poor separation and is most often found in the low mids and lows, with the main culprits being the guitars and bass. A bass cab can create unwanted resonance. The room the bass cab is in can create resonance which tends to be more obvious in the lower frequencies down to the distance between the walls. Electric guitars typically extend pretty low in the spectrum and on their own sound great, but all this low end causes trouble when adding the bass guitar to the mix as the bottom end of the electric guitar extends right down into the bass guitar's range. Low cutting the electric guitar to allow space for the bass guitar will give much clarity to the low end. If you're worried that doing this will make the electric guitar sound thin at times when it's on its own, you can automate the low cut to only activate when the bass-line is present.

As mentioned in the chapter on Expansion, even a little low cut to the bass to ensure the low end of the kick extends that little bit further down can help with low end clarity too. Or the application of a notch in the bass at a certain frequency can allow the kick to cut through. The very bottom of the spectrum doesn't need much to fill. The kick and bass are plenty, and even they need to be separated from one another to some extent.

Pretty much everything in the mix, if played from a musical instrument, will have harmonics extending way up to the very top of the audible spectrum, and beyond. It's important that you try to keep these intact as they actually help with separation. Whereas heavy handed cuts can help clarify the lows, gentle persuasion in the mids and highs should be all that is needed to achieve separation. High cuts can sound harsh and will affect important upper harmonics.

I hope I have managed to make my point... that separation is clearly of huge importance!

As just mentioned, creating depth in the mix helps with separation. Depth in a mix is when all the different instruments (vocals, guitars, drums etc) seem to have their own actual space, not just from a tonal point of view, but from a 3D point of view as well. All the sound might be coming from two speakers, but we can still give quite a good illusion that they are coming from different places in the room. Depth can be achieved in many ways – EQ, reverb, chorus, panning and delays are all tools used for moving things around the sound stage. As I said at the start of this chapter, I don't intend to go too far into mixing techniques, I just want to get across the concept of good separation and definition.

Channel and bus compression

Some problems I come across when mastering are not always to do with muddiness or tone, sometimes it's down to compression. Let's start with the drums.

Compression can really change the tone or feel of the sound you are compressing and this happens mostly to percussive sounds down to their sharp attacks. Because a percussive sound happens so quickly, with quite a large amount of changing tonal information hitting you in such a small amount of time, the effect of the compressor changing the envelope (shape over time) of the percussive sound can really alter the overall tone.

Using a bus compressor for all the drums is definitely a good idea, it brings them to a whole in very much the same way as using a single-band mastering compressor across the mix. But care must be taken not to push too far. Given the chance, a compressor can really sap the life out of a percussive sound.

In general, excessive amounts of compression in a mix will not translate into a nice master. If a mix has been over-cooked, there's very little a mastering engineer can do to put things right. Occasionally, expansion has helped in such situations but not very often.

At the opposite end of the scale, I have experienced situations where certain elements in the mix could have really done with some more compression. For instance, dynamic range is not always that favourable with the vocals; at times I find vocals to be under-compressed. A vocal's expression (loud or soft) is translated through the way the voice sounds and doesn't need to have huge variation in the actual level. An under-compressed vocal can suffer from certain words or syllables being lost in the mix affecting its intelligibility.

Another aspect that can sometimes be under-compressed is the bass-line. Ideally, the bass needs to be pretty levelled out, at least in its lower regions. The bass-line, along with the beat, make up the foundations of the whole mix and need to be solid. What can sometimes happen is one certain note of the bass-line will jump out, most likely due to some kind of resonance. Or there can be variation in the level of each note down to the way the musician is playing. Don't be shy when it comes to compressing the bass. Multi-band compression can be useful to focus in on the frequencies that are jumping out.

For most other things in a mix, compression should be used lightly as not to over-cook it. It may be the case that a master can be too dynamic making it unsuitable for playing in a noisy environment, but a good master should still have a healthy amount of dynamic range overall. A piece of music's expression comes largely from the dynamics of the mix. Squashing out all the dynamic range with excessive amounts of compression will take away the life of the mix. However, there is one situation where being bold with your mixing compression may be more favourable – when you want the final result to be loud. The next chapter covers this in more detail.

In summary

Rather than teach you how to mix your music, I have merely pointed out some rules that mixing different sounds together follow down to the way we perceive sound. Mixing is very subjective, heavily dependent on the artist's taste. In my opinion, a nice mix is well balanced, has good separation, width, depth and punchy dynamics, which is a joint effort of recording and mixing skills. Nice mixes become nice masters, simple as that.

Chapter 22: Loud Masters

This chapter is in two sections. I will firstly explain what it is about a particular mix that allows it to be driven hard at the limiting stages to get it loud without it sounding bad. Secondly, I will give tips on how to get it loud using mastering processors other than just the limiter.

Before I go on, I must reiterate something I mentioned in Part 1, 'the loudness war'. This is where competition is forcing music to become louder and louder. Here's what I said about it:

'Do not fall victim to the belief that if one song appears louder than another, for some reason it's better. If you turn up the quieter of the two, you will probably discover that the quieter one packs more punch as it hasn't been squashed. Has your hi-fi got a volume control? Has your friend's hi-fi? You can always turn it up. We want a suitable amount of loudness to give a sense of body and weight. We also need to ensure all the details are audible amongst the distraction and background noise of our consumer's environment. Anything after that and you should be questioning the need for anymore loudness'.

Let's say you've finished mastering and you've made it super loud, RMS level up well above -10dBFS (decibels relative to full scale).

At first this may sound big and powerful, but the problem is what happens over time. When you listen to music with such extreme RMS levels, over time you'll experience fatigue and the music becomes tiring to listen to. If a whole album is mastered to ultra high loudness levels then it's very difficult to actually listen to the whole thing start to finish. Please remember this when you're mastering.

Section 1. Why does one mix permit loudness and another doesn't? 

There is no black and white answer to this. Sometimes a mix will totally surprise you by allowing extreme limiting without sounding bad, but in general, the better the mix, the louder it will go. As previously mentioned, mastering compression and/or limiting will surface all the finer details so the better quality the mix, the more you can get away with.

Electronica is probably the easiest genre for getting it loud because it's more acceptable for the original mix to end up sounding different. This is down to the fact that electronic sounds don't particularly have a right and wrong way of sounding in the first place. When it comes to other genre like metal for instance, changing the sound of the guitars, or drums, is not quite as acceptable, so reaching high levels of loudness becomes more complicated.

Muddiness is a major factor. As I have said, the harder you push the limiter the more of the mix is exposed which includes imperfections like muddiness.

The main dictator is the condition of the lows and low mids. When these areas are clear and tight, you are less likely to get distortion when pushing hard against the limiter.

In the previous chapter, I expressed the importance of dynamics and how a mix with good dynamics produces a better master. When the aim is to achieve a super loud master, it can often help to start with the mix. Healthy amounts of compression raising the RMS levels of individual elements within the mix is one way to obtain a loud master. A particularly dynamic mix won't take too well to being slammed by a limiter, whereas an already loud mix won't need slamming by the limiter to achieve the desired loudness in the first place. That being said, an over-cooked mix won't sound too pleasant when mastered to high levels of RMS. As always, go easy.

So the best way to achieve loudness is either produce a clean and tight mix that will let you throw lashings of limiting at it, or produce a loud mix in the first place, eliminating the need for heavy compression/limiting at the mastering stages. But here's a few tips for getting that little bit more.

Section 2.  8 Tips for getting a little more RMS

Firstly, parallel compression is great for achieving loudness without squashing the hell out of it. Large amounts of compression can be applied to the wet signal giving plenty of RMS whilst the peaks are kept intact in the dry signal.

Balanced tone – an even balance of tone will help toward loudness. When a mix sounds thin, bottom heavy or too bright for instance, you will really struggle to achieve a 'loud sound'.

Clipping the analogue to digital/digital to analogue converter (ADDA). Up to now I haven't discussed the idea of using analogue processors (this book is meant purely for learning techniques that can be used with any processors, analogue or digital), but for those of you who are using an ADDA, a clever way of getting a little more is to clip your ADDA just after the limiter. Most ADDA's have a soft clip (kind of like another limiter). This can allow you to achieve a tiny bit more loudness before you produce audible distortion.

Use multiple stages of compression. At the start of the master chain you could have some light compression or saturation. After that, multi-band compression, then single-band compression, then finally the limiter. This means each processor doesn't need to be pushed too hard causing it to sound bad.

Automation of softer and louder passages. Heavy amounts of compression/limiting can lift the softer passages so high in level, that when the louder chorus kicks in, it has little impact. You could insert a gain plugin at the very start of the master chain and automate changes in gain to make the softer passages quieter, giving more impact to the louder passages.

As we know, achieving loudness is much to do with the condition of the lows. Referring back to the 'Limiters and RMS' chapter in Part 1, low cutting the subsonic frequencies can create space to gain more RMS elsewhere.

Using Ultra Depth can contribute to the perceived loudness, allowing you to get a little more before it all sounds bad.

Last of all, and this may sound unusual, but as well as multiple stages of compression, use multiple limiters too. I discovered this by accident. Having two, or even three limiters carrying out the duty of obtaining the desired loudness level can allow you to push it further before it sounds bad. They would all still be found last in the master chain. You can use different kinds of limiters in combination. For example, you could have a PSP Vintage Warmer (with brick wall setting engaged turning it into a kind of limiter) followed by the Waves L2 Ultra Maximiser.

Transient processors

One of the problems with making it loud is the very fact that it takes away all the dynamic range and it's the dynamics that contain all the punch. So when your listener actually turns the volume down because your track is so much louder than the one just played, your track will sound lifeless down to the lack of dynamics and transient hits. One way to overcome this is to insert a transient processor into the chain. The negative effect suffered by the transient hits down to compression and limiting can be reversed to some extent by exaggerating the transient hits using a transient processor.

A transient processor is a tool used for shaping the transients within a mix. The processor detects a transient and allows you to boost or reduce its attack. Here's a few examples of transient processors: Sonnox TransMod, Waves Transform, CrySonic Transilate. Placing it before the limiter is probably best, any earlier and the boosted transients might play havoc with things like your single-band or multi-band compressors.

There's one more thing I'd like to mention. I don't recommend this but you'd be surprised at how many commercial tracks contain a little audible distortion down to how loud they are. It's not detectable on all systems, you need to have access to some high definition monitors to hear it. It seems that some engineers will simply allow for an amount of distortion knowing it will go un-detected on most systems. Either that or they hadn't realised it was there in the first place.

So there you have it, a whole bunch of ways to tire your ears.

Chapter 23: The Master Chain

Up to this point, you have a whole load of techniques and methods to use in bringing your music to a commercial standard. A question you might now be wondering, or have been wondering the whole time, is what order do you arrange your processors? This chapter goes through all the different processes and techniques described so far, explaining how one affects another.

The first rule of audio processing in general is 'everything affects everything'.

If you've visited MasteringTuition.com, you may have found some block diagrams of different master chains intended to achieve certain results, but I had to wait until the end of the book to actually talk about this subject so you fully understand the nature, or mechanics of each process when explaining how one process directly affects another.

Probably the most important thing to understand is how different kinds of dynamics processors behave with each other, and how dynamics processors behave with other processors like EQ for example. Out of all the different types of processor, dynamics processors are the ones that will behave differently according to what you feed in at its input. This is why making decisions about where to place processors in the chain revolves largely around dynamics processors.

Imagine you have an EQ inserted before a compressor. Changing the settings on the EQ will affect what the compressor decides to do because the compressor has a trigger threshold. Adjusting the EQ affects the signal level which is what causes the compressor to trigger. For example, EQ'ing out some low end will reduce the signals level. This could have a knock-on affect on the amount of compression the compressor applies. But by inserting the EQ after the compressor instead, the behaviour of the two processors will no longer be affected by each other according to what you dial in – the EQ doesn't have a trigger threshold, its effect is static. The EQ cannot change what it does in response to what's being fed in at the input. Let's look at the processes in more detail. I'll start with the two main tools in the kit, EQ and single-band compression.

EQ and single-band compression

As described in Part 1, using compression can affect the tone of the mix. Typically the lows may sound lessened, or you may interoperate it as a brightening effect. Inserting an EQ after the compressor can solve this. A gentle lift in the affected areas can revive some of the lost tone. Occasionally the somewhat brightening effect of the compressor is just the sound being aimed for, which is great because it means less processing.

Using EQ after compression has the benefit of being able to get back to the original tone of the mix. However, only EQ'ing after the compressor and not before can be a risky business. As you know, the compressor is most sensitive to RMS heavy elements like the lows. There's always the possibility that there may be some really low, undesirable sub-sonic noise or rumbling going on which can affect the compressor's behaviour. It's practical to have at least tidied the mix to some extent before the signal is fed into a single-band compressor. Even nuisance frequencies, like resonance, might be better dealt with before compression. Or multi-band compression could be used before single-band to control any overly dynamic elements that might cause the single-band compressor to do unwanted things. Remember, the single-band compressor's movement is imprinted across the whole spectrum. Referring back to the article on saturation, this is why I suggested it as an alternative to using single-band compression at the start of a master chain. Saturation doesn't run the risk of sudden dips in level when an unexpected RMS heavy element passes. Obviously, saturation has a sound of its own so this must be taken into consideration.

Multi-band compression

Multi-band compression is quite a versatile tool, it can achieve many different things so for where it works best, there is no straight answer. One way to look at it would be, the more it is used for controlling elements that are out of control (over powering hi hats, crashes, under compressed vocals or vocal ess), the earlier in the chain it should go. Placing it before the single-band compressor might be preferable to stop over-dynamic elements causing the single-band to misbehave. In my experience 'tidying' is better done earlier on.

Like the single-band compressor, multi-band can do a nice job of bringing a tidy mix to a whole when mild settings are used. In this case, you might place it further down towards the end of the master chain.

Multi-band expansion

Expansion is a great tool for injecting life back into elements that either lacked it in the first place, or are lacking down to other processes like single-band compression. If being used to reverse the negative side effects of single-band compression, like the loss of punch in the drums for instance, then inserting it after the single-band might be best. Inserting it before will affect the way the single-band compressor behaves. When being used to correct mixing issues like a weedy kick for instance, using it earlier makes more sense, perhaps at the very start. The mix needs to sound at its best before it experiences the main portion of the master chain.

Limiting

You already know where to place the limiter – last. There's a dedicated chapter in Part 1 about why they always go last. Obviously there are a few exceptions as outlined in the previous chapter. But more often than not, you'll find the most effective place to use a limiter to be the last position in the master chain. To jog your memory, here's what I said in Part 1 about why this is:

'Let's say you applied an EQ after the limiter. If you boosted any frequency bands, you will surely hit the top and trigger the red lights indicating you are clipping the audio (clipping off the tips of the waveforms). This is because the limiter has already pushed everything up to just before hitting the top. Doing so will probably produce audible distortion at the output. And if you reduce a band of frequencies, you will loose some of the loudness you just obtained by using the limiter in the first place.

If the EQ was to go before the limiter, there would be no risk of clipping when boosting a frequency band. Reducing a band's gain, which would lower the RMS level, could be replenished by pushing the limiter a little harder. For best results, put the limiter last in the chain. Try to think of a limiter as a final seal that cannot be broken.'

Moving away from dynamics, let's now look specifically at EQ.

Mastering EQ

Just like multi-band compression, its position in the chain comes down to what you use it for. Tidying is best done earlier on, so for things like low cutting out any unwanted subsonic frequencies or dealing with nuisance resonances, earlier in the chain is best. For correcting the negative side effect of another process like the change in tone from using single-band compression, there are two ways you can look at it. You can simply insert it after the compressor, this will correct the tone and not affect what the compressor is doing. On the other hand, correcting the tone before the signal enters the compressor can have an interesting and perhaps more desirable effect, depending on what you're trying to achieve. Boosting the lows for example, before entering the single-band compressor, will force the compressor to react more favourably towards the lows. As described in Part 1, this can be very useful when seeking that fattened up effect for electro styles, like dubstep or drum 'n' bass.

Aside from correction purposes, or playing on a compressor's trigger threshold, EQ is more generally used for tonal sculpting – finding warmth, softening harshness or adding a little sparkle. I wouldn't class this kind of use as tidying myself so it's not crucial that it's achieved early in the chain. Providing that you fully understand how the changes you make to the tone affect any dynamic processors ahead, then there's no right or wrong answer.

Mastering reverb

There are two main schools of thought here. The first is that the reverb could be one of the earliest, if not the first process applied, since it should be as if the mix always had this larger reverberant quality. The second method is that reverb could be much later, if not the last process added before the final limiter. This is because by this point the signal is neater and trimmed to the point of near completion, which means that the reverb is only acting on the elements of the mix that you intend to keep as the final master.

The main thing you need to be aware of is how compression affects reverb. Essentially, the compression can emphasise the effect of reverb in the way that it lifts the tail ends of any reverberating sounds (compression raises the quieter details).

Ambience enhancement

For this you might consider some of the same points as for reverb. Although ambience enhancement is not as heavy going as adding reverb and can be used more often, the same can apply in terms of where to use it. The effects of different layers of compression can emphasise or change the sound of the ambience enhancement when placed before compression.

Saturation and EQ

Saturation is a kind of dynamics processor and has similar properties to compression. There is a trigger point where the desired compression occurs, which reduces the dynamic range and allows for an overall rise in the RMS level. The trigger point is reached by forcing the signal up against it, as opposed to bringing the trigger level down over the signal. But unlike compression, saturation doesn't react in the time domain by dipping the level of the whole mix (or chosen frequency band) and returning back to an idle point determined by an attack and release parameter. The effect is instantaneous. Placing EQ before saturation can give different results to placing it after, but there's no danger of causing undesirable pumping as with compression. Placing EQ before saturation will affect the amount of saturation, and the colour of the saturator's sound, as it affects the signal's level going in. Placing it after the saturation will not affect the point at which saturation occurs, so any adjustments to the EQ will not affect the amount, or sound of the saturation.

Harmonic excitement

What harmonic exciters do is quite complex. In terms of where to place one, I guess its effect falls somewhere between a boosted EQ and multi-band expansion, so refer to what has been previously written in making your decision. Essentially, they brighten the sound. Just be aware of how the change in tone will affect any other dynamic processing in the chain like multi-band compression or expansion.

Mid/side adjustments

For corrective purposes, the earlier in the chain the better. For sweetening the sound, like fattening the lower mids, place further in, perhaps after any single-band compression as not to cause it to over-behave (unless that's what you want to happen). For general widening purposes, again be aware of how this is affecting any dynamics processors further on. Raising the level of the side in a mid/side matrix to widen the stereo image will most likely brighten the sound at the same time due to the likelihood of there being elements higher up the spectrum within the stereo field.

Mid/side and Ultra Depth

One interesting thing to think about when using mid/side is how it affects Ultra Depth. When you place Ultra Depth in the chain, you will have effectively introduced more stereo information into the mix (created a larger difference between left and right). Using a mid/side matrix to widen the sound might emphasise the effect of Ultra Depth when placed after it. This might be desirable or not. Just be aware of this fact.

If you have chosen to narrow a mix using mid/side in the hope to pull the vocals or snare out of all the reverb let's say, Ultra Depth can be used to widen the mix back out afterwards. The new sound could be as wide as before but with the vocal or snare sitting higher just as you intended.

Summary

In summary, dynamics processors are mainly what dictates the decision of where to place each processor. Whenever you make adjustments to processors placed early in an already working and nearly finished master chain, pay close attention to the 'domino effect' this can cause along the master chain as each process affects the next.

When mastering in the digital domain, the least processes used to achieve the desired results, the better. Every process, a long with its enhancing capabilities, will most likely also introduce loss of overall quality to the sound. For example, if you can achieve the same sound with one multi-band compressor as opposed to using a string of different processors then you're on the right track. Too many processors in the master chain can also result in an 'over-cooked' sound. It pays to keep things simple. The only instance where I would advise using multiple processors for the desired effect is when the target sound is to be loud. Other than that, try to keep the master chain as short as you can.

Chapter 24: Mastering Multiple Tracks for an Album or EP

When a number of tracks are to be mastered together for an album or EP, there are a few things you need to think about before you make a start on the first one. This chapter describes how to approach the mastering of multiple tracks, and provides tips for arranging and finalising the tracks once mastered.

Consistency

First of all, you need to listen to all the mixes before you start mastering, as you may spot issues that need more attention at the mixing stages. If so, these need to be dealt with before you start mastering, as ideally you need to master all the tracks in one go. That doesn't mean all in one day, but it's not practical to spread it out across a long drawn out period of time. When mastering multiple tracks for an album or EP, it makes sense to be in a similar mind set for each track to maintain consistency throughout the whole process. Just like compression can help bring a mix to a whole, mastering will help bring the album to a whole. All the tracks need to sound like they belong together – consistency is very important. As long as you master all tracks within the same time frame, consistency shouldn't be a problem.

If you have no choice but to master the tracks sporadically across a long period of time, you will need to keep listening to all the other masters every time you start on a new one to familiarise yourself with the original sound you achieved (tone, compressors attack and release times, spacial enhancements like width and depth, RMS levels etc). In theory, you should always arrive at the same sound regardless of how long it has been since the last master you did, but in practice this is not always the case.

Mastering a full album in one go usually takes less time because you don't have to keep listening to previous masters again and again, trying to get yourself back to the mind-set you had when you first started the mastering project.

One album I mastered for someone took around three years. They wanted their album mastering as they recorded it. As the recording took three years, there was about three years between the first and last master I did. In that time, not only had my mastering skills and techniques progressed, so had my mastering tools. I was even operating in a different studio. Mastering each track the same way I mastered the first became increasingly more difficult as time went on. If you're mastering professionally, it makes much more financial sense to master all the tracks together in one go.

Loudness

If getting it loud is on the agenda for the mastering project, then loudness is one of the things you need to be thinking about when listening to all the mixes before you start mastering. How loud should the masters be? To answer this, you might consider using another commercial album as a reference – something close to the sound you are looking for perhaps. Incase you are, let me just say, it's not always a good idea to try and match the loudness of another album. How loud you can go is dependent on the mix. Matching the loudness of other commercial albums may be pushing it too far.

Should they all be the same loudness? If all the tracks are very similar in style then getting them all to have the same perceived loudness is probably the way to go. But the likelihood is that some tracks might want to sound bigger than others. A soft acoustic track with just guitar and vocal might sound a bit odd if it possesses the same RMS levels as a heavy rock number a few tracks later. Here's a little tip on how to decide on the loudness levels of each; keep the perceived vocal level consistent between each track and let other sounds be different. It doesn't have to be exact, but if you at least try to keep the vocal coming through at similar levels, then the changes in overall loudness levels between each track will sound more realistic.

Where to begin

Now you have a decision to make, which track do you start with? If loud is the theme then which ever track is going to need to reach the highest RMS level is the one you should begin with. Here's why: Let's say you started with a relatively softer track, reached a given RMS level, then moved on to a more aggressive number. Mastering in this order can lead to the problem of not being able to obtain an appropriate perceived level of loudness for the aggressive number compared to the softer track, simply because the mix won't permit it. Starting with the track which you intend to be the loudest eliminates the problem of struggling to match the RMS of another track. Even if you don't aim to achieve high RMS levels, it's still a good idea to forecast which track will prefer the highest RMS levels and start there, just to be safe.

For all the tracks that you intend to share a perceived loudness level, it's not mandatory that you get every single one exact when mastering each individual track, although it helps to at least get them close. In reality, some tracks may simply need more/less compression or limiting purely for achieving a desired sound and not loudness. Because of this, you might end up with a collection of tracks with differing levels of perceived loudness throughout. As long as they're not drastically different, this should not be a problem because you have the option of making final adjustments to global levels once all the tracks are mastered. This can be done inside the software used for finalising all the masters. This final stage is where you arrange the tracks, choose gap lengths and input any meta data like the track's names for instance.

Arranging the tracks

There's a whole array of different software that cater for this final stage. Apple's 'Wave Burner' and Steinberg's 'Wave Lab' are two common ones. They're very simple to use; you simply drag and drop each audio file into the software in the order of the album. Below is a snapshot of Apple's Wave Burner with a collection of masters dropped in:

Global levels

This is the point where you can make final decisions for levels. You usually cannot raise the levels of any tracks as this might cause clipping when you render but you can reduce levels. If you left some headroom above the highest peak when bouncing your master, you can use that headroom to raise the level of that particular track during this final stage. Having all the tracks laid out in this manner makes it easier to get an idea of how loud each one appears compared to another, as it's easy to quickly flick from one to another, listening to the perceived loudness of each. You can fine tweak global levels accordingly.

Gap lengths

Now you can decide upon the length of time between each track. This is done by sliding the tracks back and forth within the time line. See below:

I have a few tips that may help you decide on what the gap lengths should be. Just before I get to that, there's a couple of important points I need to mention.

Capture

Did you capture all the audio when you bounced down your masters?

When you bounce each master down to a stereo wav file or similar, it's important that you allow for an amount of silence at the start and end of each. This is to ensure you capture the whole piece of music. You may think this is obvious, but I know from experience that engineers can get this wrong. There have been occasions when I've requested a mix to be bounced again with the markers wider apart because the end or beginning had been cut off. If professional mixing engineers can make this mistake, so can anyone.

Fade the silence

As well as a good few seconds either side of each master, a fade in before any audio appears and a fade out after the audio has completely finished is also recommended. This can be done when bouncing down each master, or inside the software used for finalising. The reason for this is the likelihood that there will be some noise inherent in the mix. This could be the tiniest bit of hiss but with addition of compression and limiting, the level of the hiss could have been raised. Without the fades, when one track finishes and another one begins, you might hear an awkward sudden stop and start of the background noise. The fades make sure that any noise/hiss smoothly fades in and out at the start and end of each track.

Overlaps

Don't worry that the added silence at the start and end of each track will complicate matters when choosing gap lengths. When you drop the audio files into the finalising software, you can simply overlap them like so:

For the duration of the overlap, both audio files will play on the finished album or EP.

When referring to gap lengths, obviously we're talking about the distance between where the audio ends and begins, not the actual distance between the audio files.

Having the option to overlap tracks allows for an amount of creativity when originally planning the album. For example, if one track ends with the gradual fading of an ambient soundscape, you could begin the next track before it has completely faded.

Gap length decisions

Choosing gap lengths is not always at the discretion of the mastering engineer; sometimes you will have instructions for gap lengths that have come from the artist. But more often than not, it will be up to you as the mastering engineer to choose appropriate gap lengths ensuring the smoothest flow throughout the album. If you're not feeling confident about getting the gap lengths right, here's some guidelines I have developed from my own experience of arranging tracks for an album or EP.

Please note, these guidelines are simply the way I like to do it, you might not totally agree.

The speed and style of each track is mainly what dictates the gap length. I'm not going to tell you actual lengths in seconds, there would be no point. The lengths are relative to the feel and flow of the album – only you can be judge of that. But what I can tell you is how the individual gap lengths will differ from one another, in relation to the style of each track.

When going from a fast, upbeat track into another of similar style, short gap lengths work well. Sometimes being so short that there's hardly a gap at all can really help provide the intended impact for the next upbeat track. As one fast track suddenly ends ...bam! – you're hit with the next track straight away. As a general rule, really short gap lengths should only be used towards the beginning and probably not past half way through the album. Really short gap lengths towards the end of the album just don't seem to sound as appropriate.

When there's a slow track amongst faster tracks, it's good to have an amount of silence either side of the slow track. The silence at the start of the slow track works as a means to slow the overall feeling of a fast pace down from the fast track, preparing the listener for a slower track. The silence at the end of the slower tracks is probably not as crucial as at the beginning, but slow tracks tend to call for at least some amount of silence at the end to help portray the softer nature of the track.

When going from a slow track into another slow track, there should be a fairly long gap but not too long. The pace is already slowed so the gap should only be long enough to provide a suitable rest before the next slow track.

In a nutshell:

Fast to fast = short gap

Fast to slow = long gap

Slow to fast = long gap

Slow to slow = medium to long gap

And for me the general rule is that, as you proceed through to the end of the album, shorter gap lengths tend to become a little longer.

I know this theory won't work for everything, your gap lengths may differ from what I've said. These guidelines have come purely from a pattern arising out of the albums I have mastered.

With global levels tweaked and gap lengths chosen, the only other thing left to do before you render is to input any meta data like track names, album name or ISRC codes. You can choose to render the audio into various formats like burn to CD or export a DDP (Disk Description Protocol – digital format suitable for sending off for replication). Head over to the hidden articles section at MasteringTuition.com for information about digital formats and ISRC codes.

If you're reading this as an eBook, you can follow this link to go there now:

http://masteringtuition.com/index.php/furtherlearning

If you're reading the hard copy and have a smartphone or tablet, use the QR code above to navigate there. Otherwise you can simply type the url into your browser.

Chapter 25: Final Word

So now you've mastered all your tracks to a commercial standard, arranged them, chosen suitable gap lengths and created a final product ready for replication or internet download. Well done. One thing I'd like to say right now is a reminder of how mastering is a series of very slight adjustments which, when all added together, creates a breathtaking enrichment of the music's sonic qualities. Mastering is not for the heavy handed.

I'll end with with one more valuable tip, possibly the most powerful tip of them all:

Fresh ears.

Never spend more than an hour on one track. Granted at first you may have to, but certainly not when you're starting to get the hang of it. Tired ears will play tricks on you. Walk away and come back at least eight hours later to listen to what you've done with fresh ears. Trust me, out of all the techniques and tips I have shared, this is probably the most effective tip of them all.

Mastering is both an art and a science. Enhancing the sonic qualities of the music is an art. The flawless translation across different sound systems is the science. Good luck.

Chapter 26: About Me and Some Friendly Advice

My name is David Eley and my mastering business is called TGM Audio.

As a kid I used to make my own electronica music on a sequencer. I soon developed a fascination for finding out how to make my own creations sound as good as the music I owned on tape and CD. Without realising it, I had embarked on the journey of becoming a mastering engineer.

I studied electronics and computer technology at college before going on to do a HND in professional sound technology. I worked as a systems tech and live sound engineer at a venue in Manchester UK. Some more study, a teaching qualification and working for free eventually landed me a job teaching Audio Operations at a college in North Manchester UK. During this time I had been building a sophisticated home studio where I would record my own band and other bands too. Some of my friends have gone on to be quite successful with the recordings they made in my home studio.

It was roughly the same time as I started teaching that I actually started mastering other people's mixes for cash, starting with friends, then friends of friends. Seeing a possibility to earn an extra income, I embarked on another journey – learning web design and online advertising. Soon I had built a website and began marketing my new mastering business online. A couple of years later, my business became so busy that I had to give up teaching. I left my home studio and set up a mastering only studio in East Manchester UK.

I've never been able to shake off my passion for teaching. As hard work as it was, I still miss it. This passion led me to create MasteringTuition.com and write 'The Audio Mastering Blueprint'. I hadn't realised how popular MasteringTuition.com was going to be until I was contacted by MusicTech Focus Magazine who wanted to include it in their 'Mastering Tuition Round Up' (I hadn't even finished writing it at that point). Since then, MasteringTuition.com has been included in another MusicTech Focus Magazine.

Once MasteringTuition.com was built and happily plodding along helping thousands of people around the world learn how to do a good job of their own mastering projects, I moved on to an even bigger task – writing the advanced second part to the Audio Mastering Blueprint... 'The Secret Notebook of a Mastering Engineer'.

Some friendly advice to help you on your way

Firstly, get mastering. The more tracks you master, the better you become, simple as that. I have included a section at MasteringTuition.com where you can download un-mastered mixes to practice with. I will try to add more and more un-mastered mixes to this section. When you first start out, it's difficult to get your hands on un-mastered material to practice with. Back when I was just about to start my online mastering business, MySpace.com was just starting to become known. I used to trawl through Myspace looking for un-mastered music. I used software to download the un-mastered music and would then practice my techniques on it. However possible, get mastering as many tracks as you can.

Secondly, get listening to as much music as you can, in particular pop music. Download the latest 'Now That's What I Call Music'. This advice came to me from another UK Manchester based mastering engineer, Paul O'Brien (Pob Audio Mastering). As you practice your mastering and listen to more and more music, you will actually start to hear the music differently. You begin to hear how it has been mastered. You become more tuned in to the tone and texture; things like compressor attack and release times start to become obvious. It's quite weird actually, and once you get to that point, I'm afraid there's no going back. Soon you will only be able to hear commercial productions with an objective ear. But don't worry, you will still enjoy listening to music, only you will always be learning as you do.

Thirdly, don't just learn about mastering. The deeper your understanding of the whole production process, the better you will be at mastering.

And lastly, read as much as you can. I'm a little obsessed with reading and I have accumulated masses of useful information in doing so, not just about music technology, but in other areas too. I have gained more from reading books than all of the study I have ever done. In audio, a couple of books that I have enjoyed and found really useful are:

'Mastering Audio: The Art and the Science' by Bob Katz

This book is jam packed full of technical information and will not only provide you with invaluable information about mastering, but in many other areas of music technology too. I still go back to it from time to time.

Don't be put off by how sophisticated Bob Katz' studio is and the sections in the book about mastering equipment. I was at first. I almost convinced myself that in order to do mastering, I needed to be able to afford the gear he uses and have as much of it. Well, I simply couldn't afford it at first and learned how to do a good job without. Over time, as my business has grown, I have been able to afford some very nice outboard mastering equipment. But I stand by the fact that you can do a professional job without. I did for years and customers kept coming back.

'Mixing With Your Mind' by Michael Paul Stavrou

This book is brilliant. You will never see the production process in the same way after reading it. What a feeling of enlightenment it leaves you with.

Just as I have tried to do in the Audio Mastering Blueprint, this book get's across a concept. There's a real 'penny drop' feeling as you read.

And finally, don't forget to read the bonus articles in the hidden section at MasteringTuition.com. There you will find more fantastic tips, and I've even included a list of links to free software I have come across over the years.

Follow this link to go there now or scan the QR code:

http://masteringtuition.com/index.php/furtherlearning

But just before you head over there, there's just one more thing, it's on the very last page (after the global contents)...

Acknowledgements

Thank you to all who have contributed to the creation of MasteringTuition.com and 'Pro Audio Mastering Made Easy'. Lee Calpee (SSR Graduate and freelance sound engineer), you came to me looking for work experience at just the right time; thank you for all your help with creating articles to publish on the MasteringTuition.com website – some of which have been used in the writing of this book. Thank you Lee Maguire (friend and business partner) for the inspiration I needed to bring this book into reality. Jonathan Rowland (my boss and friend), thank you for your contribution by providing help with web design, and for your commitment to making sure I have all that I need to run my mastering services. A huge thank you to Andrew and Marie Pascoe for your editing services. Thank you WAVES and Voxengo for providing suitable imagery for use with the tutorials. Thank you Logic for being such a fantastic DAW and a brilliant platform for educational purposes. And finally, may I thank my wife Alexa for her valuable contribution by putting up with a work obsessed husband. The writing of this book has been made possible through the valuable support I have received from my friends and family.

Global Table of Contents

Part 1: The Audio Mastering Blueprint

Cover Page

Introduction

Part 1 Table of Contents 

1. Definition of Audio Mastering 

2. Guide to a Better Listening Environment 

3. Calculating Standing Waves 

4. Tonal/Spectral Processing

4.01 Introduction to Tonal/Spectral Processing using a Linear Phase EQ Plugin 

4.02 EQ, Tone & The Frequency Spectrum 

4.03 EQ Parameters

4.04 EQ Decisions

4.05 Problem Frequencies

4.06 Technique for smoothening out the mix using a linear phase EQ 

4.07 Problem Frequencies Continued 

4.08 Linear Phase EQ Demonstration 

4.09 Final Word

5. Dynamics Processing

5.01 Introduction to Dynamics Processing 

5.02 How Compressors Work 

5.03 Circuit Types

5.04 Setting Up Your Compressor 

5.05 Demonstration of Compressor Technique 

5.06 Fatness and RMS 

5.07 Limiters

5.08 Limiters & RMS 

5.09 After the Limiter 

5.10 Demonstration of a Limiter 

5.11 Multi-Band Compression

5.12 Multi-Band Compression Situations 

5.13 Demonstration of Multi-Band Compression 

5.14 Final Word

6. Audio Suitability for Mastering 

7. Live Mastering Session 

Part 1 – Listen

Part 2 – EQ

Part 3 – Multi-Band Compression

Part 4 – Single-Band Compression

Part 5 – Limiting

8. Advanced Techniques Taster: Ultra Depth 

Part 2: The Secret Notebook of a Mastering Engineer

Cover Page

Part 2 Table of Contents 

Introduction

9: Stereo Thinking

10: Mid/Side Encoding and Decoding 

11: Stereo Widening, Or Is It? 

Widening the mix, how far can you go?

Narrowing the mix, how far can you go?

12: Mid/Side EQ and Stereo to Mono Compatibility 

Mid/side EQ

The lows

Sweetening the ambience

Focusing the vocals

Stereo widening

Stereo to mono compatibility

Midrange meat

13: Expansion

14: Mid/Side Dynamics

Compressing the mid channel

Compressing the side channel

Mid/side expansion

Mid/side processing conclusion

15: Parallel Compression

Parallel compression for attitude

Parallel compression for transparency

16: Mastering Reverb

17: Ultra Depth Continued 

Papers and patents about enhancing depth using psychoacoustic manipulation

Creating extra depth using psychoacoustic manipulation

Mono compatibility

When can Ultra Depth be used?

EQ'ing the effect

Other ways to improve versatility

Summary

18: Saturation in Mastering 

19: Exciters and Enhancers 

20: Independent Left and Right Settings 

Stereo EQ

Stereo Compression

21: The Mix

Mixing sounds together

Separation

Channel and bus compression

Summary

22: Loud Masters

Section 1. Why does one mix permit loudness and another doesn't?

Section 2. Tips for getting a little more RMS

Transient processors

23: The Master Chain (ordering your processors)

EQ and single-band compression

Multi-band compression

Multi-band expansion

Limiting

Mastering EQ

Mastering reverb

Ambience enhancement

Saturation and EQ

Harmonic excitement

Mid/side adjustments

Mid/side and Ultra Depth

Summary

24: Mastering Multiple Tracks for an Album or EP 

Consistency

Loudness

Where to begin

Arranging the tracks

Global levels

Gap lengths

Fade the silence

Overlaps

Gap length decisions

25: Final Word

26: About The Author and Some Friendly Advice 

Acknowledgements