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FEATURING Essential advice for preparing your track In-depth tutorials for mastering at home Discover the pro techniques that deliver today’s toughest tracks TUTORIALS / ADVICE / OPINION / EVERYTHING YOU NEED TO POWER UP YOUR MUSIC TODAY
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WorldMags.net SPECIAL 64 2013 Future Publishing Ltd. 30 Monmouth Street, Bath BA1 2BW Tel: 01225 442244 Fax: 01225 732275 Email:
[email protected] Web: www.musicradar.com/computermusic EDITORIAL Editor-in-Chief: Daniel Griffiths Art Editor: Stuart Ratcliffe Sub-editor: Jane Glover Tutorials and features: Jono Buchanan, Marco Migliari, Greg Scarth, Tim Cant Editor, Computer Music: Lee Du-Caine Cover illustration: Luke ONeill
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welcome Is a track ever really finished? It’s a question that unfortunately plagues every stage of the creative process. From writing, through recording and mixing, the temptation to dabble, rework, experiment, scrap and start all over again is often too much for some music makers to battle against. And with technology increasingly delivering the power we need to minutely adjust every part of the track, this temptation to never actually finish anything is on the rise. It’s a good job, therefore, that the rise of mastering and its adoption by even the most casual of tune-makers has given us a final ‘full stop’ to our music making. Mastering is the definitive underscoring of your work. After mastering, there is nothing more that can be done and all your niggling doubts have to be discarded as your newly minted tune begins its life as released music. If you’re taking your first steps at truly finalising your work and are keen to explore the various pro and DIY options available, then this Computer Music Special is for you. We’ve all the theory you need to prepare your track and ensure that it’s laden with the ammo the mastering process needs. Then we’ve expert guides to the process itself, revealing the pros’ secrets and giving you the knowledge to take the job on yourself. If you’re not feeling confident right now, then we guarantee you will be by the end of this Special. Good luck! ENJOY THE ISSUE…
© Future Publishing Limited 2014. All rights reserved. No part of this magazine may be used or reproduced without the written permission of the publisher. Future Publishing Limited (company number 2008885) is registered in England and Wales. The registered office of Future Publishing Limited is at Beauford Court, 30 Monmouth Street, Bath BA1 2BW. All information contained in this magazine is for information only and is, as far as we are aware, correct at the time of going to press. Future cannot accept any responsibility for errors or inaccuracies in such information. Readers are advised to contact manufacturers and retailers directly with regard to the price of products/ services referred to in this magazine. If you submit unsolicited material to us, you automatically grant Future a licence to publish your submission in whole or in part in all editions of the magazine, including licensed editions worldwide and in any physical or digital format throughout the world. Any material you submit is sent at your risk and, although every care is taken, neither Future nor its employees, agents or subcontractors shall be liable for loss or damage.
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Make great music on your PC or Mac!
Also available digitally on App le Newsstand, Zin io, Google Play and more!
Computer Music is the magazine for PC and Mac musicians. It’s packed with tutorials, videos, samples and exclusive software to help you make great music now!
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WorldMags.net SPECIAL ISSUE 64
contents 06 WHAT IS MASTERING?
54 MASTERING FOR WIDTH AND POLISH
Welcome to the dark art that makes ‘finished’ tracks even more finished! But what is mastering, and what can it do for you?
12 GET READY FOR MASTERING
Mastering isn’t just about keeping levels in check and removing nasties. Breath new life and space into your track with our tips
60 THE PERFECT MASTER CHANNEL
Top dance music producer Sharooz dishes the tips for preparing your track for the perfect mastering session
Start up your computer and build yourself a great start-point master chain in your choice of DAW
14 THE PRO PRODUCER’S GUIDE TO COMPRESSION
64 UNDERSTANDING DIGITAL Your final mastered track is just a bunch of ones and noughts. Understanding what’s going on will help you get the best possible sound
If you want your music to sound right before, during and after mastering, you need to know compression
69 LOUDNESS AND PSYCHOACOUSTICS What tricks can the ear play on you and is the never-ending pursuit of loudness just an illusion? It’s time to get scientific…
22 UNDERSTANDING LIMITING Meet the compressor’s brother in arms, the limiter. We get you up to speed and expert, fast
27 UNDERSTANDING METERING AND VISUAL MASTERING TOOLS
78 HEADROOM AND LEVELS EXPLAINED Navigate the danger zone of mastering with our guide to signal-to-noise, gain staging, headroom and more
Are you dazed by decibels? Struggling with spectrograms? The world of mastering is strewn with complex meters and visual indicators. We take the hard work out of it for you…
86 THE PRO’S GUIDE TO MASTERING Bigger, brighter, better: learn the myths and methods behind a professionally mastered sound
34 HOW TO GET LOUD (AND WIN THE LOUDNESS WAR) Achieving loudness to compete with professional tracks while maintaining dynamic range is an art. We show you how…
94 MASTERING ESSENTIALS Grab your final scientific fact-pack of mastering know-how
99 CM DOWNLOAD 44 MASTER LIKE ABBEY ROAD What happens when the great Abbey Road gets to grips with your track and how can you get that sound at home? We ensure your next track is sparkling with mastering magic
Get the download that’s free with this digital edition. It’s your one-stop shop for all the tutorial files and audio that accompany this Special
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What is
MASTERING? Welcome to the dark art that makes ‘finished’ tracks even more finished! But what is mastering, and what can it do for you? Let’s find out…
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Welcome to possibly the most mysterious aspect of the music making process: mastering. Thanks to a combination of misinformation and good old-fashioned misunderstanding, a number of myths surround the mastering process. The good news is that we’re here to dispel them as well as talking you through the equipment, processes and knowledge that mastering engineers use to add the finishing touch to your tracks. So, what exactly is mastering? Simply put, it is the process of taking a complete mix and preparing it for distribution. Historically, the mastering engineer’s job related very closely to the vinyl record manufacturing process. Vinyl discs had certain limitations which meant that mastering engineers had to be careful to get the balance right in order to create discs that would be loud and have enough bass for the job without causing the needle to jump or grooves to wear prematurely. The same general principles apply to modern distribution methods, albeit with slight differences. A good master nowadays will sound good on any playback system, will be loud while retaining dynamic range, will have a good balance between all parts of the frequency range and will sound, for want of a better word, finished. Let’s take a look at how it’s done.
The mastering environment If you thought that recording studios paid close attention to acoustics and high-quality equipment, mastering studios take it to the next level. Given that the mastering process acts as a final check on the quality of tracks already mixed in recording studios, it stands to reason
that the playback equipment and acoustic treatment of the mastering suite should offer an even more neutral, revealing sound to ensure that the finished product is of the highest sound quality. Studio acoustics are of vital importance to a mastering suite and no expense is spared to ensure that the space offers a flat, neutral frequency response and is free of excessive reverberation, room modes, standing waves and other acoustic nasties. It should go without saying that the monitors themselves must be of the highest quality. Debate rages as to whether subwoofers are preferable to full-range systems, but whichever approach is chosen the playback system should be capable of delivering a neutral, uncoloured reproduction of the full frequency range. In these digital days, D/A conversion is also crucially important so that the monitors are fed with a high quality signal, as is A/D conversion to print the finished master back to digital. Some mastering engineers still choose to print the finished master to a top-quality two-track tape machine, although tape’s declining popularity in the tracking process is mirrored in mastering. The most important, and perhaps most underrated, pieces of the mastering puzzle are something we all possess but not all of us use to their full capabilities: our ears! In order to carry out the mastering process it’s essential that the mastering engineer has a well-developed ability to listen to music and analyse it closely. Luckily, that hearing can be trained – although it’s not
really in the scope of this short article to explain how to develop your critical listening and analytical hearing skills. Don’t be put off if you feel that you don’t have so-called ‘golden ears’. With hard work and careful training using a wide variety of recordings, just about anyone can learn how to use their ears to a high enough standard. As with so many aspects of music making and production, practice makes perfect.
The mastering process So, what’s involved in the process itself? Firstly, the mastering engineer listens to the track and assesses its characteristics and requirements, including overall EQ balance, dynamics and
“Debate rages as to whether subwoofers are preferable to fullrange systems for monitoring” stereo width and balance. At this point, consider the fact that the mastering engineer might decide that the track needs absolutely nothing doing to it. Although this is often not the case, it’s a good reminder of the fact that there are no rules as to what equipment should be used in the mastering process, what processing should be applied or what settings should be used. Anyone who tries to peddle sweeping generalisations about processing and settings should be viewed with a extremely healthy dose of scepticism! Once the mastering engineer has a good idea of what’s required, the sound can be processed
Understanding the track Don’t forget: the final piece of the mastering jigsaw puzzle is the track itself. At this stage let’s think again about the aim of the mastering process: adding the finishing touches to a good mix. Your aim when mixing is still to make the track sound as good as you possibly can and as such it’s a good idea to separate the mixing process from the mastering process. Needless to say, trying to mix and master all at the same time or carrying out a half-hearted mixdown with the intention of fixing it at the mastering stage are best avoided. If you find it hard to avoid the temptation to start tweaking your mix during mastering, practise with a bounced stereo
version rather than simply loading the project up in your DAW. Taking a break between mixdown and mastering is also a good way of creating a psychological distinction between the two processes. On a related subject, we would strongly suggest that you avoid adding any kind of mastering effects to your track until you’re sure that your mix is as good as you can possibly get it. An EQ or a compressor on the mix buss is not a problem as long as it suits your overall sound, but mixing or even writing tracks with a stack of multiband compressors, limiters and exciters on your stereo output is a very bad habit to get into.
Beware of mixing and mastering at the same time. Take a break between the two processes to aid separation
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WorldMags.net Processing explained Regardless of what software you have it’s wise to consider the importance of effect order. A typical basic mastering chain would consist of an EQ followed by a compressor or multiband compressor, then a limiter. The majority of extra effects such as exciters or stereo widening tools would typically slot in after the compressor and before the limiter. It’s also a good idea to put a metering tool such as a spectrum analyser, scope or level meter at the end of your chain in order to see how your changes are affecting the output level, frequency balance and waveform shape (this can warn you as to when clipping will occur or dynamic range is being harmed). Most DAWs have built-in tools for this task and we’d also recommend Bram’s free s(M) exoscope plugin (www.bram. smartelectronix.com). The next thing to consider is how the separate stages of your mastering effects chain interact with each other as you make changes. This phenomenon occurs just about any time you chain effects together. To simplify matters, let’s think about effects in a production
The Bram smartelectronix s(M)exoscope waveform visualisation plug comes highly recommended
context or a moment. Consider a situation where a sound source (say, a drum machine) is being run through an EQ and then a compressor. If we boost the bass on the EQ, the drum machine will hit the compressor harder each time the kick is triggered, resulting in a more dramatic compression of the entire drum channel. As such, the EQ can alter the way the compressor works almost as much as adjusting a setting on the compressor itself. The same process occurs when running a full track through
in order to achieve the desired results. Taking a look at the typical outboard equipment used in mastering studios will give us a good idea of the kind of adjustments that might be made. The tools of the mastering trade are very closely related to equipment commonly found in recording studios and used for tracking, producing and mixing. The most commonly used gear like EQs, compressors, expanders and limiters should need no introduction, although the versions used here tend to be masteringgrade equivalents, designed with the intention of offering features demanded by mastering engineers and also providing the highest possible sound quality. Digital or analogue versions of each tool may be used.
mastering effects, although the fact that the waveforms are more complex and contain sounds from across the entire frequency range means that mastering effects interact with each other to an even greater extent. Sticking to established processing chains and keeping an ear on the bigger picture when making adjustments should help you to avoid unexpected results. As you work, think about how each change might affect other settings in your mastering chain. For example, if you’ve boosted the
in this area. Compressors and expanders are typically used to control dynamic range but may also be used to add a subtle colouration to tracks. Units including the Manley Vari-Mu, Chandler LTD-2, Pendulum OCL-2 and classic Fairchild 670 offer a variety of different flavours, including valve and solid state operation plus VCA, FET and optical circuitry. For the more aggressive forms of ‘loudness maximiser’-style limiting, Waves’ L2 digital hardware limiter is the top dog.
“An unfortunate side effect of software mastering is that it can encourage the use of presets”
Precision EQ Accurate and precise EQ is crucial to the mastering process and various options are available, with many offering mastering-friendly features such as linear phase response, linked stereo channels and discrete switched settings rather than continuously variable rotary controls. The classic Pultec EQ and Manley Massive Passive are de facto industry standards 8 / COMPUTER MUSIC SPECIAL
highs to add clarity maybe you need to adjust your multiband compressor to take the change into account. Good quality mastering is largely based on trusting your ears. If you’re really struggling, visual aids may help, so try running a commercial track through a spectrum analyser or looking at its waveform to get an idea of its frequency balance or overall level. You should soon train yourself to hear subtle differences and adjust accordingly without looking at the meters.
bads boys offer every type of mastering tool in a single unit. Software equivalents of just about all mastering tools are now available thanks to the likes of iZotope’s Ozone suite, IK Multimedia’s T-RackS, TC Electronic’s MD3 and Brainworx’s suite of M/S tools. Many will argue that sound quality in some cases may not match the hardware equivalents, but the lower price and ease of use of the software approach has seen it rise in popularity over recent years. Software also has the unfortunate side effect of encouraging the use of presets, an idea that would be met with derision by any mastering engineer worth their salt. When working with software approaches, we recommend you don’t use presets and determine appropriate settings on a case-by-case basis.
Alternative methods Together, these tools carry out the most important parts of most mastering jobs. Other tasks may include fixing problems with elements of the mix, for example by applying noise reduction or widening the stereo image. In some cases, mastering engineers may also choose to employ multiband compression in order to control the dynamics of separate frequency ranges individually. All-in-one digital mastering processors are also produced by various companies. These
Although the mastering process usually runs along the lines we’ve discussed here, there are a few alternative techniques that some engineers might use when they feel it’s necessary and appropriate. The most common technique in this case is for the artist or producer to supply the mastering engineer with separate stem files for different sections of the track. This could be as simple as providing a stem of the vocals and a stem of the instrumental, or may involve splitting the mixed version into
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Preparing a track for mastering three or four component parts, all with the levels matched so that they simply need summing together to create the desired mix balance. The aim is not for the mastering engineer to carry out the mixdown for you, but to allow more freedom and flexibility during mastering. Processing the fully mixed stereo file of a track may be satisfactory in the majority of cases, but the ability to adjust the balance of elements or process separate parts individually may be preferable in some circumstances. Bear in mind that there comes a point where a higher number of stems is less convenient and mastering engineers will certainly not thank you for handing over a disc full of unmixed stems!
Distribution-specific mastering A less common technique still employed by some is the idea of creating different masters for distribution by different methods. This is a considerable bone of contention among the mastering fraternity, with most mastering engineers arguing that it shouldn’t be necessary to produce versions with different characteristics and that a good master is a good master regardless of how you intend to distribute your track. If you really feel that your track needs special treatment, discuss it with your mastering engineer and they’ll be able to offer you the best advice. Good luck!
Even when you’re entrusting your tracks to a professional mastering engineer there are a number of things you can do to help make the process run more smoothly. We had a chat with Joe Caithness (above) of Subsequent Mastering in Nottingham (www. subsequentmastering.com) who gave us some great advice that should help you get the most out of the mastering process. “The single most important thing is to ask questions if you’re not sure about something” says Joe. It’s essential that you and your mastering engineer understand each other’s aims, so don’t be afraid to discuss what you want. “It works both ways,” Joe tells us. “Sometimes we need to ask questions, too. A quick email can save hours of messing about.” Joe advises against applying stereo mix buss effects like limiters, heavy compression and mastering plugins before sending your tracks to him. “Although there’s nothing wrong with applying processing to your main output, most of the time it’s going to confuse issues as you’re essentially doing bits of our job for us. If you use any mix buss effects, please tell us what
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they are, how they’re set, and why they’re set that way. If possible, you should also send a version without any processing applied, too.” Don’t be tempted to push your mix too hot to try and get it louder, either. After all, that’s part of the reason you’re sending it for mastering in the first place. “Like all tradesmen, we need some space to work. It’s better to leave a little bit too much headroom than not enough. Even more importantly, do not let the mix clip!” It’s another aspect of the mixdown that causes the biggest headaches for mastering engineers such as Joe… “Probably the biggest tonal problem in tracks I receive is the excessive use of effects such as exciters, saturation and overdrive. What I get is a lot of ‘solo mixing’, as I call it. It’s obvious when someone’s soloed each instrument in turn and made everything hotter and hotter. When you add it all back together, you get a mush. It’s kind of like preparing a casserole and adding a bit of salt every time you chop a vegetable. Each piece might taste OK on its own, but, man, your casserole will be salty and it won’t work as a whole. Not good!” COMPUTER MUSIC SPECIAL / 9
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Mastering has its origins in expensive studio hardware. You don’t get presets with these…
Bin the presets Warning: as tempting as they might be, presets are best avoided for mastering purposes. The idea that you can load a generic dance mastering preset in your mastering package and hope to get something appropriate for the particular track you’re working on is simply nonsensical. A dance preset EQ might boost the low frequencies to try and emphasise the kick drum and bass notes, but its suitability will depend entirely on the level of bass already in your track. If you’ve made a bass heavy mix, picking a mastering preset that boosts the bass again will have an entirely detrimental effect. If you want to learn about mastering and get the best results, by all means start with a preset or some of the suggested chains you’ll find in this magazine, but begin adjusting and tailoring the settings to suit your own track as soon as possible. Just like mixing, there’s no one setting to suit every type of music! 10 / COMPUTER MUSIC SPECIAL
iZotope’s excellent Ozone is a popular mastering solution for home studios
Mastering at home Mastering engineers work long and hard for years to hone their craft and build a reputation for quality. However, if you put a bit of time into training your ears and learning the tools of the trade, you can do a damn good job at home – and for free. Here we’ll focus mainly on software since it makes sense that most DIY mastering beginners will choose the cheaper software-based approach. Armed with the tips on offer here you should soon be able to get started with mastering your own tracks. The quality of mastering relies heavily on the quality of your monitoring environment and equipment. In order to understand their importance, you should probably consider the consequences of getting it wrong. Poorquality monitors are likely to suffer from an uneven frequency response and if your room is badly treated (or worse, untreated) then this problem could be exacerbated to the point where there are significant peaks and troughs in the frequency response of your monitoring setup. Attempting to master under these conditions, you are likely to compensate for the problems as you go along, leaving you with a mastered track riddled with problems. The fact that you’re looking to master at home rather than employ an expert probably means that money is tight, but it’s important not to cut too many corners. Mastering is your final opportunity to spot problems with your tracks, so it makes sense to give yourself a fair chance. At a very minimum, do consider building some cheap DIY bass traps, diffusers or absorbers to get your mastering space up to scratch. As with mixing and producing, there’s
also no harm in having a second set of monitors, a hi-fi or some multimedia speakers on hand for cross-referencing. In fact, it’s a wise idea to double-check that your mastered tracks translate well to different systems just as you’d check that a mix translates. In the long run you’ll probably train your ear to the point where this stops being necessary, but it’s a good training tool as you learn the art of the mastering engineer. In terms of processing, there are a number of options these days when it comes to software mastering packages. iZotope Ozone, IK Multimedia T-RackS, TC Electronic MD3, the various WAVES packages and the Brainworx bx_
“Poor-quality monitors and a badly treated room could mean a mastered track full of problems” bundle are the main commercial options, but a number of freeware and shareware packages are also available. In addition to this, you’ll probably find a reasonable set of mastering plugins bundled in with your DAW. Logic 9, for example, offers handy mastering tools in the form of plugins such as the Multipressor, Adaptive Limiter, Linear Phase EQ and MultiMeter. The last decision to make is whether you want to use a mastering plugin in your DAW or master in a dedicated wave editor such as Steinberg WaveLab, Sony Sound Forge or the excellent and free Audacity. There are several advantages to working in a wave editor, but some may feel that the familiar DAW environment aids workflow. Once all the tools are in place, you’re ready to start.
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Get ready for
MASTERING Top dance music producer Sharooz dishes the tips on preparing your track for the perfect mastering session
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PROCESS THE MIX BUSS SPARINGLY “There’s a great temptation these days to reach for the multiband compressor or limiter to give your track an instant gloss. However, it’s important to ensure your actual mix is solid in the first place. If buss processes are dramatically altering the sound of the mix, then I’m afraid you’re doing something wrong. They should serve simply to glue things together, not change the entire fabric of the track.”
GIVE EACH INSTRUMENT ITS OWN PLACE “Scrutinise each part, in particular, paying attention to the midrange as this is where things can easily get cluttered. Using subtractive EQ should guide you well. Unnecessarily long reverbs can also cloud your track. Add effects sparingly – ask yourself if the part really needs it and if so, how much does it really need to get the desired tone?”
REDUCE BASS FOR MAXIMUM LOUDNESS PAY ATTENTION TO THE ARRANGEMENT “If your mix is too bassy, you’re compromising “Break your tune down into its sparsest form and volume. It may be difficult to fully hear what’s layer on parts to check if every element is going on in the 20-80Hz range if you’re not using doing its job. Remove parts that clash a sub. Check your mixes on a system with a good, rhythmically. With multiband compression now accurate bass reproduction and make sure your kick tails commonplace, try to evaluate the impact this will have on aren’t overly long or boomy as these often go unnoticed the sum of parts.” and get in the way of the bassline.” RETAIN DYNAMICS “The biggest club tunes always retain a keen GET ANOTHER PAIR OF EARS TO LISTEN TO sense of dynamics. Try it for yourself in your YOUR MIXDOWN arrangement – going from a quiet or silent “Whether it’s something as simple as turning the section into an all-guns-blazing rhythmical vocals down a notch, or fine-tuning EQ, it’s often assault usually does the trick and always works wonders easy to overlook critical elements if you’re the on the dancefloor!” only person working on your creations. Take a few days away from the mix then pull the original track back up to check if it still sounds as powerful as it did when you first USE A REFERENCE began. Get a friend to listen, too.” “When you think you’re done, open up your favourite tracks and carefully scrutinise their mix on your system, then use that to see what CHECK YOUR MIX ON VARIOUS SYSTEMS you can learn about your own mix. Try and “Every monitor has a different characteristic steer clear of over-using spectrum analysers to evaluate a and unique detail can be picked up from even mixdown, but focus instead on using your ears to check the poorest quality speakers. If you make club what’s what!” music, checking mixes on a big rig or PA is a must. For vocal tracks and general balancing, a flat, no-frills monitor such as the classic Yamaha NS-10M is a WHEN IN DOUBT, GET THE PROS IN valuable tool.” “It is never a bad idea to use a properly trained engineer to master your track. Strip any excess limiting or multiband LESS IS MORE compression from the mix and allow the “Generally speaking, the fewer elements in the mastering engineer to rebuild the process using their own mix, the more punch and volume they yield. tools. Convinced that you’ve got the best possible sound With the mass of plugins and soft synths currently available there’s never been a greater already? Even if this is the case, you’ll always find that a little added pro sparkle never goes amiss. Besides, pro temptation to whack on multiple parts. Trimming the fat mastering has never been cheaper or more accessible will help you focus on creating more unique and than it is now, with some of the finest studios offering memorable elements, as well as making it easier on the great rates.” final mixdown.”
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The Pro Producer’s Guide To
COMPRESSION Above all other effects, compression will have the most effect on the quality of your mastered track. If you want your music to sound great after mastering, get to know compression!
Try to imagine riding the first ever bicycle, the legendary ‘boneshaker’. Without suspension, every lump, bump, ripple and pothole in the road was connected directly to one of the most sensitive parts of your body. Without compression, audio is the same – every click, thump and subtle nuance of the music is connected directly to another of the most sensitive parts of your body: your ears. Sometimes, that’s exactly what you want. But in most modern genres, compression is an integral part of the sound of almost all the music we listen to. Just as good suspension smooths 14 / COMPUTER MUSIC SPECIAL
“Without compression, every nuance of your music is connected directly to your ears” out the ride on your bike and car, compressors even out audio signals. This can make them gentler on the ear, or make them easier to
balance in a mix, or add a unique ‘warming’ character to the sound. Alternatively, like the suspension on a great racing car, they can be used for control and power, rather than comfort. Compressors can add punch and impact and enable you to push your music to its absolute limits. This feature will tell you how to achieve all of these goals – or at least, get you started on the road to using compression effectively in your mixes. We’ll start with the basics, talk about the creative uses of compression later and give you a perfect track ready for the next step in the mastering chain.
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the pro producer’s guide to compression / make music now <
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The basics: manual compression Sometimes the simplest approach is best Electronic compressors as we know them only became widely used in the 60s. That doesn’t mean people weren’t using compression before then, though, they were just doing it ‘by hand’… In the broadest possible sense, compression is about managing dynamic range. Dynamic range is the difference between the loudest and softest sounds in a song or performance. Voice, bass guitar and percussion all have a naturally wide dynamic range, whereas solo flute or sustained string chords tend to be much less ‘peaky’, that is, more consistent and with less dynamic range. In any mix there will be an ideal dynamic range for each instrument – enough variety to keep interest and emotion, but controlled enough to remain audible, without getting lost or overpowering other elements.
Microphone technique The very simplest and most direct way to manage dynamic range is at the source. A great example of this for singers is to use what is broadly described as ‘microphone technique’. You’ll often notice a great live singer pulling the mic away for the loudest notes and bringing it close in for quiet, intimate moments. You can achieve the same thing by asking a singer to move in and out from the mic in the studio. This strategy has pros and cons. On the plus side it’s immediate, effective and cheap. On the other hand it takes skill and practice to get right. Many singers overdo it and removing the effect can end up being more time-consuming than using our second manual compression option…
Riding the fader In a nutshell, you can turn anything that is too loud down and anything that is too quiet up. Just like microphone technique, there are pros and cons to this approach. As with microphone technique it’s free, and it sounds very natural. The mixer chooses the ideal level for every phrase, and so long as the level doesn’t change, the part sounds entirely natural. On the downside, this is a very time-consuming method. Even with modern computer-based automation technology, it can take many runs through a track to get everything just right. Also, like mic technique, it doesn’t have that distinctive compressed sound that a lot of genres require. More on that later. It’s worth experimenting with though. We’ve done a few mixes using only minimal compression with everything else being done by hand; the results can be superb. In particular, riding a fader into a compressor can get great results – you can get even more control and the compressor doesn’t have to work as hard. Often there are times when manual compression isn’t enough, though, or is simply too time-consuming. For example, if a vocal can be balanced line by line, that’s great, but the human voice is one of the most dynamic sound sources you can record and often we need much finer control. In such cases it’s time to break out a compressor.
There’s more than one way to manage your dynamic range. You can use a compressor, of course, but you can also work the mic or ride the faders for an alternative take on it
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COMPUTER MUSIC SPECIAL / 15
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The knowledge: compression techniques Compressor controls explained Like manual methods, a compressor also manages dynamics by reducing level, but it does it automatically, depending on the settings you choose. To help understand how this works, let’s keep the car suspension analogy in mind. The suspension is a compressor, of sorts, controlling the ‘dynamics’ of the road’s surface – the bumps, peaks and troughs.
Smooth it over The overall smoothness of the ride is determined by the stiffness of the springs in the suspension. If you lean on the car, how easily does it sink down to absorb the weight? In a compressor, this overall ‘softness’ is determined by the ratio. A lower ratio gives you a smoother ride; most family cars have their suspension set up this way, but there’s more movement in the car and less control for aggressive driving. Higher ratios are analogous to a sports suspension in a car, where the smoothness of the ride is less important than control. Next up is the speed with which the suspension reacts to bumps in the road. In a compressor, the attack and release times determine how quickly a compressor reacts to the input signal (attack) and how quickly it relaxes afterwards (release). The attack time needs to suit the material. Imagine a car hitting a
gain as standard – so the effect you hear as you dial in more compression is simply that the quieter signals get louder, with the peaks staying roughly the same. This can make using them quicker and easier, but it can also lead to them being over-used. Our ears tend to assume that anything louder sounds better. With automatic make-up gain, we keep piling on more compression, thinking it sounds better, and being distracted from the negative aspects: reduced dynamic range, pumping and even unnecessary distortion.
Gain reduction One way to avoid this is to keep an eye on the compressor’s metering: the overall gain reduction. How much gain reduction is needed is dependent on what you’re recording; vocals and bass may need heavy compression while keyboard sounds and strings hardly need any. As a rule of thumb, though, pay close attention if you start to see more than 8-10dB
> Step by step
gain reduction. If you wind the threshold control down further and don’t hear any real changes, you should probably ease it off and try some different settings.
Knee The knee of a compressor refers to when and how the ratio starts to change when the compressor starts to take effect. A ‘hard knee’ means the compression becomes immediately active as soon as the input signal hits the threshold, whereas a ‘soft knee’ means the compression becomes audible more gradually. A soft knee also means gentle compression starts happening further below the threshold. If a car’s suspension bushes start to wear out, you’ll feel something like a ‘hard knee’ effect. Normally they smooth out the smaller bumps that aren’t big enough to involve the main springs. As they become worn, however, they don’t do this as effectively and the action of the suspension becomes more abrupt.
Classic compression
“How much gain reduction is needed depends on what you’re recording” speed bump too fast – the suspension can’t react fast enough to smooth out the thump. The same thing happens with audio if the attack time of a compressor is too slow. The final crucial setting is the threshold. The threshold determines when a compressor starts working, depending on the input signal.
1
For a natural sound, use slower attacks (longer than 75ms), gentler ratios (less than 2:1) and always allow the compressor to ‘relax’ back to zero several times a bar.
3
For a thicker, denser sound, use faster attacks, medium ratios and lower thresholds. Be prepared to see much more gain reduction, though.
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For a harder, punchier sound, use higher ratios and thresholds, but make sure you keep an ear out for distortion.
4
For an overstated, ‘pumping’ effect, use fast attacks, high ratios and a longer release time. Use it on a stereo buss to affect only part of the mix (eg bass and drums) and make sure the effect works in time with the music.
Make-up gain One setting on a compressor that doesn’t fit the car analogy is make-up gain. Compressors control the dynamics of the input signal and usually reduce its dynamic range. Loosely speaking, they turn the louder stuff down. This means that when you first patch an analogue compressor in, it will probably make the overall signal sound quieter. To avoid having to keep pushing the fader up to compensate, most compressors allow you to add make-up gain. This is just a way of lifting the signal back up to balance the reduction in the dynamic range caused by the compressor. You may not have seen this control if you use a compressor plugin, though – many digital compressors now include automatic make-up 16 / COMPUTER MUSIC SPECIAL
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Pro tips: complex compression Because every setting affects the way the others sound Compression is a complicated system. The controls interact with each other and give different results depending on the source signal. They need to be carefully balanced to get the sound right, making learning compression tricky. Let’s look at some examples. A family car needs a soft suspension. The ratio needs to be high enough to absorb energy and smooth out the ride, but if it’s too high the suspension will feel stiff and you’ll be jolted around. In music, the ratio needs to be high enough to control the signal level in the way you want, but not so heavy that the life and energy of the music is squashed out. A car’s suspension needs to react fast to bumps. If it has a slow ‘attack time’, you’ll feel the bump before the springs have had time to absorb the shock. It needs a fast release time, too, to allow the wheels to follow the surface of the road, otherwise you’ll get a noticeable ‘bounce’ afterwards. Similarly, the attack time of a compressor needs to be fast enough to catch the changes in level, while if the release time is too long you’ll hear it ‘bounce’ afterwards. The threshold needs to be set so that a car suspension works when it’s needed. If the car is heavily loaded, it may be sitting low on the suspension with the springs already working. If you hit a bump there may not be enough play left to absorb the impact. This would be equivalent to setting the threshold level too low on a compressor. If the threshold is too high, the compressor hardly does anything. This would be like a car that wasn’t heavy enough to bring the springs into play.
If you need a sense of excitement on your track, a Urei 1176 could be the compressor for you
Exploring the subtleties Not all compressor settings are car suspensionfriendly… Imagine you want that classic ‘pump and suck’ sound that Daft Punk used on One More Time. That is achieved using a compressor with a high ratio, fast attack and a long release. In car suspension terms, that’s the equivalent of the car dropping down suddenly on hitting a bump, then gradually smoothing out. So, the rules of thumb we’ve discussed so far are only the beginning of the story. Like all processing in audio recording and mixing, it’s just as valid to ignore the rules as it is to do things by the book. If it sounds right, it is right! Compare a handful of compressors and they will sound different when used on the same signal, even with the same settings. In this way, some engineers use compression to add ‘character’ to the sound, rather than to control it. Character can mean the effect that dynamic control has on the sound (a well-compressed vocal can sound full, warm and intimate, when the original sounded thin, hard and distant). It can also mean the sonic characteristics of a hardware compressor – the valves of a Fairchild, or the lightning-fast attack time of a Urei 1176. Remember, the rules of thumb suggested here apply to ‘mathematical’ software compressors. Don’t expect a 4:1 ratio on Logic’s compressor to sound the same on a classic hardware unit, or even a modern software emulation.
Compression dos and don’ts
DO:
DON’T:
DON’T:
DO:
DO:
DON’T:
avoid extreme settings to begin with if you are just trying to control dynamics.
add compression to every channel as a matter of course. Start off with minimal compression and carefully choose where to add compressors. experiment with different types of compressors, both hardware and software. There can be a significant difference in the sound you get from the various different types.
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forget to bypass the compression occasionally just to check you’re getting a good result. remember to balance the output gain so that the level doesn’t change when you hit Bypass. That way, you get a fair comparison. be afraid to experiment. Some of the greatest sounds in the history of recorded music came from misused and abused compressors!
COMPUTER MUSIC SPECIAL / 17
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Software versus hardware Need to make your sounds gel? Check out the SSL G-Series Master Compressor
5 hardware compression superstars The hardware versions of all of these units are highly desirable, and their cost reflects that – although a dbx 160 can be had for a very reasonable price on eBay these days. Let’s take a look…
Urei 1176 A superb hardware compressor with a very fast attack. Sounds great on drums, or vocals and bass with a slower attack and release.
Teletronix LA-2A
While software tools are very capable, there are certain jobs only components can conquer The compressors most of us use are software plugins, often the ones that came with our DAW, and the truth of the matter is, they do a great job. The flexibility to be able to put as many compressors as we like, anywhere in the audio chain, at any time, is the sort of luxury the engineers who mixed most of our favourite classic tunes could only dream of. But there’s a catch. Simple software compressors bear no relation whatsoever to the hardware compressors that those same engineers still use in their mixing right up to the present day. These days, it’s pretty easy to mathematically define a compressor in terms of ratio, attack and release times, and so on, and build a plugin that slavishly follows that definition. Back in the golden age of audio, though, that wasn’t the way things were done… Compressors were built with electronic components – transistors, resistors, amplifiers
and valves – and these components simply couldn’t offer the infinite control and flexibility that a computer can, so designing compressors was a fine art. It was a constant process of balancing the technical requirements of the hardware against factors like noise, headroom and cost. And this is where things start to get interesting… If you listen to a digital compressor with a ratio of 1:1, the chances are its output will sound exactly the same as what you put in, even at quite a high level. A hardware compressor such as the classic Fairchild 670 Limiter, however, uses no less then 20 valves in its design. This means that even when it isn’t compressing things, its output may well cause all kinds of lovely colouration and sonic changes that have nothing to do with compression. It’s ‘extras’ such as these that are one of the reasons big-name engineers love the sound of their expensive racks of analogue hardware.
A classic model now re-issued by Universal Audio, featuring a choice of transformer or optical compression, plus valve amplification for warmth. Great on vocals and bass.
Fairchild 670 Valve Limiter Contains over 20 valves! Pioneered by Geoff Emerick on the Beatles’ recordings. Great on vocals, drums, or anything else, if you can afford it, that is…
dbx 160 The dbx 160 is a classic vocal compressor. It’s fast and clean, but take care it doesn’t distort the signal.
SSL G-Series Master Compressor This archetypal two-buss compressor is great for gluing things together. Mix through it but don’t whack it on afterwards, and be prepared to spend time learning it.
The Fairchild 670 Valve Limiter: more valves than you can shake a stick at
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It’s the ‘extra’ sounds you get from hardware compressors that set them apart from their digital cousins
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WorldMags.net Pro technique: parallel compression Run compressed and clean signals side by side Parallel processing is an excellent way to introduce subtle changes in a mix and nothing suits it better than parallel compression. Parallel compression operates from the ‘bottom up’, so it doesn’t crush in the way that ‘top down’ compression does. It’s great for filling out a part, enhancing quieter details and making everything sound fuller and warmer without removing any of the punch. And luckily, with modern computer-based recording, it’s a piece of cake. Parallel compression can work especially well on drums, to bring up room ambience and subtle details without crushing, or across a complete mix. You basically run a heavily compressed version side-by-side with a clean version of the same signal – they are running ‘in parallel’, hence the name. When setting it up, mute and unmute the parallel compressed channel to
> Step by step
“Parallel compression is great for making everything fuller and warmer without removing any punch” hear the results clearly. The advantage of this technique is that you can control exactly how much of the heavily compressed character sound you want in the mix, while keeping all the impact of the original. See the step-by-step walkthrough below on how to set it up for yourself.
Pro technique: parallel compression
It isn’t the cheapest, but many would argue that Waves Renaissance is one of the best
5 top software compressors There are some great emulations of hardware compressors available. UAD-2 plugins specialise in this, but you do need one of their hardware units to run the software.
What you already have The truth is, the built-in compressors of most modern DAWs such as Logic and Pro Tools LE will do a great job of controlling dynamics. Make sure you’ve mastered using these before splashing the cash on expensive alternatives, but don’t expect hardware character from them. Best of all, they’re free!
Massey CT4
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Route the signal you want to process to a new buss, so there are two channels sending the same thing to the output, and solo the new buss. Now compress the hell out it!
2
Use very fast attack times, high ratios, and a nice low threshold. You’ll end up with something that sounds very flat, very squashed and unnatural – but has lots of compressed character. Look out as vintage emulations may add distortion – or maybe this will sound good!
Many people swear by the Massey plugins, and ‘lite’ versions are available to try out for free, but even the full version is a steal at only $69. Try it out for yourself.
Waves Renaissance Compressor Somewhat of an ‘industry standard’ in the world of plugins. This includes an optical emulation mode. It’s more expensive, but watch out for promotional offers on the Waves website.
Stillwell Rocket Compressor
3
Pull the fader right down to 0 and then un-solo it, so you’re listening to the original, uncompressed channel again.
4
Now gradually lift the fader for the heavily over-compressed signal and blend as much as you like of the heavily compressed version back into the mix. Make sure you have plugin delay compensation switched on to avoid any problems with latency.
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Stillwell’s Rocket Compressor makes the bold claim of “offering analogue character in a digital plugin”. All Stillwell plugins are available as trials and are very affordable if you like what you hear.
McDSP CompressorBank The CompressorBank offers a huge range of different emulations in a single plugin, a unique ‘Bite’ control, and has an army of loyal fans. COMPUTER MUSIC SPECIAL / 19
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Advanced techniques Take it to the next level, just like the pros By now you’ve got to grips with the basics of compression. Here are a few advanced techniques to add to your repertoire…
Chaining compressors It’s often better to use several compressors in a row, each one only working quite gently, than to push one compressor very hard. This technique is known as ‘chaining’. A Urei 1176 followed by an Teletronix LA-2A is a real power-house of a combination, for example. This technique was standard in the days when using analogue tape was a much more common practice. It was fairly normal to compress gently when going to tape, to get the best signal-to-noise performance. With tape, it was possible to record quite ‘hot’, allowing a little saturation to compress the sound further. After this, more compression would be used in the mix-down.
Limiting A limiter is a compressor with a very high ratio – typically more than 20:1 – and very fast attack and release times. Limiters are often used in
“Too much multiband compression can remove internal dynamics, leaving you with a mush” mastering because they offer a great way to control unwanted peaks and spikes in the sound. This enables you to use slower attack times elsewhere.
Multiband compression
There’s no need to rack up a bunch of expensive hardware. These days it’s much easier to do the same in software
Sometimes used on an entire mix during mastering, a multiband compressor splits the signal into several different frequency bands and applies compression to each one separately. Care needs to be taken if you’re using multiband compression, though. Just the right amount can pull a mix together nicely, but too much can remove internal dynamics leaving you with nothing but a confused, overcompressed mush.
Sidechain compression Some compressors can be set up to react to the input from another channel – the sidechain – so that this external effect triggers the compressor. Use fast attacks, high ratios and a longer release time – and make sure the effect works in time with the music. Heavily EQed tracks can also be routed to a sidechain and used as a crude form of multiband compression; one example of this would be as a de-esser on a vocal. 20 / COMPUTER MUSIC SPECIAL
Back in the days when analogue was king, it was common practice to chain compressors for the best result on tape
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the pro producer’s guide to compression / make music now <
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Finishing touches: 5 top tips Our guide to compression is nearly done. But before you master, try these techniques on your track
1
Decide what you want to achieve. Are you looking to control a dynamic signal or add punch and impact? Or do you want to change the sound and create an unusual effect? Keep listening with your final goal always in mind. Choose a neutral starting point. We suggest a ratio of around 2:1, 75ms attack time and around 100ms release.
2
Overdo it to begin with. Wind down the compressor’s threshold until it starts working – it can be useful to start with an exaggerated version to get the settings right. You can hear how the attack and release is working at the more extreme settings. If you find yourself having to turn
the threshold a long way down, try boosting the input level a little instead.
3
Listen! Fine-tune the settings, remembering the goal you have in mind. Once you’re getting close, you can adjust the threshold to get just the right amount of compression to achieve the effect you’re aiming for. For a smoother sound, try a faster attack time and higher ratio, but be sure to keep enough energy in the sound. To reduce ‘bounce’ (where you can hear the level duck as the compressor cuts in and then spring back up when it releases) try a shorter release time and easing off the threshold, or try using a lower ratio.
To add punch, experiment with higher ratios, slightly longer attack and shorter release times, but watch out for ‘pumping’ (where the end of the note is louder than the start) and any distortion that’s introduced.
4
Listen again. How does it sound? Balance the different settings against each other. For example, higher ratios usually need higher thresholds if you want to avoid a heavily over-compressed sound. But maybe you don’t…
5
Experiment, imitate and listen again. Some of the greatest sounds in popular music owe their existence to the use of compressors – and some of
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the most unpleasant, too! Using too much compression when it’s not needed is almost always worse than too little. At the end of the day, though, there’s no right or wrong when it comes to compression settings – only what sounds right to you. Here’s our three-point summary: Ratio: controls the degree of ‘squash’ you have Threshold: determines how much squash happens Attack/Release and Knee: dictates how fast and how abruptly it squashes Every source and every mix is different, so keep these points in mind, fire up whatever compressors you have to hand, listen, experiment and enjoy.
COMPUTER MUSIC SPECIAL / 21
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UNDERSTANDING
LIMITING By now you’re an expert in compressors, but what about its mastering brother in arms, the limiter? Let us guide you through its subtleties
22 / COMPUTER MUSIC SPECIAL
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One subject that seems to cause more confusion than any other among beginners and experienced engineers alike is the real difference between compression (in its many forms and plugins) and limiting. Here, we’ll build on your knowledge of compression to explain what these subtly different tools are, how they work and how they can help you.
Limiters, levelling amps and expanders The difference between a compressor and a limiter (sometimes known as a peak limiter) is essentially very small. Both operate in a similar way, reducing the volume of a signal when its level exceeds a given threshold. The usual rule of thumb is that if the ratio is 10:1 or higher, the unit is considered a limiter. However, there are a huge number of exceptions to this rule and you’ll find many compressors that offer higher ratios and limiters with lower ratios. A ‘brickwall’ limiter uses a ratio of ∞:1 in order to ensure that the signal never exceeds the threshold. Occasionally you may also come across a levelling amplifier. This is really just an oldfashioned term for a compressor. It was most famously used on the legendary Teletronix LA-2A of the 1960s and crops up occasionally on new equipment. Thanks to the vintage link, it’s more closely associated with retro-sounding compressors that colour the sound of any signal they compress. However, this is only a
“The rule of thumb is that if the ratio is 10:1 or higher, then the unit is considered a limiter”
very vague rule of thumb. In reality, the differences between compressors, limiters and levelling amplifiers are so small that they should really be considered different variations on the same basic idea. The good news is that this means the controls tend to work in pretty much the same way. If you know how the threshold, attack and release settings work on a compressor you’ll have no trouble with a limiter or levelling amp. Expanders, which seem to have fallen out of fashion in recent years, are effectively the opposite of compressors. Rather than reducing the level of a signal above the threshold, expanders reduce the level of any signal below the threshold. The result is to allow louder parts of the signal to pass through untouched while reducing the volume of quieter parts. You could think of it as a more flexible version of a noise gate – while a noise gate completely mutes any signal below the threshold, an expander simply makes it quieter and allows you to adjust how dramatic the gain reduction is.
Maximising Maximising is the term most often used to describe the process of increasing the volume (or perceived volume) of a signal. The key to the process is reducing the dynamic range. We can amplify any signal until its loudest peak just hits 0dB, but depending on the headroom between this peak and the rest of the signal, the track still might not seem loud. Maximising with a limiter or compressor effectively increases the average level of a signal by reducing the level of the peaks, allowing us to push the gain up and make the whole track seem louder. The subtle distinction between general compression and using a limiter to maximise a signal is that maximising specifically aims to push the overall level of the input signal as high as possible. This can be achieved in two main ways. The first is to route your signal into the limiter, pull the threshold down until the desired level of gain reduction is achieved and then add makeup gain to push the output up as high as
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“Understanding what you’re doing to a signal’s dynamic range is key to getting good results” possible (again, until the loudest peak is at 0dB). The other method is to set the limiter threshold at, or close to, 0dB and amplify the signal before it hits the limiter in order to achieve the desired level of gain reduction on the peaks. The downside to this process is you lose some of your dynamic range, which is essential to keep a track musical.
Limiting and dynamic range The effects of compression and limiting on dynamic range always seem to cause confusion so it’s worth thinking about them again. Understanding what you’re doing to a signal’s dynamic range is one of the keys to getting good results from dynamics processors. Compression and limiting effectively reduce the difference in level between the quieter parts of your signal and the louder parts, so let’s take a look at three different ways in which this can occur and look at the implications for whole track limiting, drum buss compression and kick drum compression. Firstly, consider the track as a whole. For argument’s sake, let’s say we have a verse that is quieter than the chorus. If we push the whole track through a limiter in order to make it seem louder, we’ll eventually pull the threshold down to a point where the loud parts (the chorus) are being limited and the quieter parts (the verse) are being amplified. As a result, the dynamic balance between the two sections is being affected. If the overall level of the track can’t be increased without detrimentally sacrificing COMPUTER MUSIC SPECIAL / 23
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Dynamics processors such as Flux BitterSweet II keep your transients in check
Alternatives to limiting In mastering, compressors and limiters can be incredibly powerful when it comes to shaping the transient envelopes of percussive sounds, but there is an alternative. Rather than requiring you to set ratios, thresholds and attack and release times in order to squeeze the sounds into place, transient designers offer you control of the amount of attack and sustain. It can be a much easier way of achieving similar results, allowing the effect to do the hard work. The SPL Transient Designer is the most wellknown transient shaping hardware unit, but a plugin version is also available and a number of other companies make similar effects. It’s most commonly used on drum tracks, to manipulate the amplitude characteristics of hits, but transient shapers can be just as useful on other sounds with a percussive element. Synths, guitars and even vocals can be tamed or twisted. There are plenty of options out there for you to get started with, but the free Flux BitterSweet II plugin comes highly recommended (www.fluxhome.com). 24 / COMPUTER MUSIC SPECIAL
Limiting hardware is a studio essential but in-the-box plugins will do the job at home
some of the track’s intended dynamic variation, then it suggests there’s a problem that should be addressed in the mix rather than with broad, sweeping application of a limiter.
Transient shapers Secondly, on a smaller scale, let’s consider the situation when we compress a bussed drum mix. When a drummer plays, some hits will be louder than others due to the drummer’s emphasis on certain beats and potentially due to some degree of inaccuracy (accidentally hitting the snare harder than intended on certain beats or fluffing a kick drum hit). If we use a compressor to tame the dynamic range of the hits, we can smooth out the variation in level between each one. Louder hits can be squashed, making the level more consistent, and this allows us to push the level of the quieter hits up to match. But as the drums are compressed harder and harder, the dynamic range between hits is reduced. This may be desirable in some cases, but care should be taken to avoid squashing the dynamics to the point where the drummer’s expression is lost. On an even smaller scale again, consider a single kick drum hit. At shorter attack and release times, a compressor on a drum hit can dramatically affect its overall amplitude envelope. For most of us, the envelope-shaping effect of a compressor on a drum track is an invaluable tool – one of the characteristic sounds of electronic music is the effect of a
drum machine being pushed through heavy compression – but if you don’t think and listen carefully to the effects of your compressor, it’s easy to end up with a drum hit that doesn’t sound anything like you want it to. The desired amount of dynamic range may vary according to genre. Jazz tracks or more gentle acoustic music would typically benefit from the highest levels of dynamic range in
“Jazz tracks or more gentle acoustic music would typically benefit from the greatest dynamic range” order to retain a realistic, lifelike sound, whereas the (slightly controversial) trend in recent years has been for pop, rock and dance music to be more squashed. This would mean a lower dynamic range might be acceptable if it meant the track would have the loud, squashed sound which we’ve become used to. The temptation with limiters and compressors is often to go overboard, but care should be taken to avoid the squashed, lifeless sound which often occurs.
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Future Music is the mag for the latest gear and how today’s cutting-edge music makers use it. We’ve been making the future since 1992. Make sure that you’re part of it. WorldMags.net
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Understanding metering and visual mastering tools
Are you dazed by decibels? Struggling with spectrograms? The world of mastering is strewn with complex meters and visual indicators that all aim to show you what you’re hearing…
Modern audio gear and software comes loaded with meters to try and give visual indications of signal levels and a host of other information. The problem is that a lot of us don’t really understand exactly what our meters mean and what they’re trying to show us. There are dozens of ways that understanding how to interpret meter readings can help you. To this end, we’re going to take a closer look at decibels, then examine some of the various meters, dials, gauges and indicators in more detail. This should help you understand how they all work and how they might be useful to you in your music making.
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What’s a decibel? The hardest part about metering is getting to grips with exactly what a decibel is. In fact, many experienced engineers and producers struggle with the scientific definitions. A decibel is most accurately described as a logarithmic method for measuring the magnitude of a signal relative to a given reference level. Only when we know what the reference point is does the decibel measurement take on any real meaning. As such, there are a number of different signals that can be measured using decibels, although for audio purposes we’re mainly interested in acoustic sound signals and (analogue) electronic sound signals.
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WorldMags.net Sound pressure levels explained One of the most common uses of decibels is to measure sound pressure levels. The measurement dB(SPL) is the one most people use to refer to how loud a sound is. To confuse things, SPL measurements are sometimes made using weightings designed to replicate the response of the human ear. A-frequency weighting is the most common system, based on the equal loudness contours, while C-weighting is used to measure peak sound pressure. SPL meters often have a speed setting for adjusting how quickly the meter responds to level changes. The S (slow) setting usually uses a 1s integration time, F (fast) uses 125ms and I (impulse) uses 35ms. In the studio, the most important use for an SPL meter is to measure your monitoring level. Prolonged exposure to high sound pressure levels not only leads to fatigue, it can also cause long-term damage to your hearing. As a result, it’s advisable to determine your preferred monitoring level and stick to it, avoiding the temptation to push the master level control up. Typically your monitoring level should be calibrated with a pink noise reference signal (at 0dBFS) and the SPL meter in your listening position set to C-weighting, slow response. We’d recommend picking a level somewhere between 80 and 90 dB(SPL), although some prefer even lower levels.
Our pic shows critical notch control data. A 25.57dB drop at 1.245kHz with a bandwidth (Q) of 37.369. The result? One nasty resonant spike tamed!
Notching nasties Almost all sounds, whether produced acoustically or electronically, contain resonance. This is where certain frequencies are amplified as a result of the ‘construction’ of the sound and the way it is subsequently produced. Occasionally, these can become exaggerated to the point where they become unpleasant to listen to. However, even before such extremes, it’s a common producer and mix engineer’s task to target unpleasant, prominent frequencies and employ EQ tools to lessen volumes at specific points where needed. It’s important that such ‘nasties’ are removed prior to mastering as the broader mastering process could boost this unpleasantness further.
Part of the reason why so many people find this confusing is that most of the time, we foolishly insist on referring to each of these different measurements simply as ‘a decibel’ and abbreviating them all to ‘dB’. What we should be doing is specifying what the reference point is. The result of this lazy naming is that we often have to infer the reference point from the context. There are three main areas where you’ll find decibels used for audio measurements, and the context should give you a clue about which type is being used. Probably the most common everyday use of the decibel unit is as a measurement of the level of intensity of an acoustic sound. This is more accurately termed dB(SPL) and is referenced to sound pressure level in air, where 0dB(SPL) equates to 20 micropascals (uPa). This measurement of micropascal is the lowest 28 / COMPUTER MUSIC SPECIAL
pressure (and therefore quietest sound) that can be perceived by the human ear. For electrical audio signals, the most common scales used to measure level are dBu (the professional standard) and dBV (more common in consumer equipment). In the case of the dBu, it is referenced to a signal with a root mean square magnitude of 0.775 Volts (RMS is
“Only when we know what its reference point is does the decibel measurement take on any meaning”
The process of removing specific problem spots in the frequency range is often referred to as ‘notching’ because, rather than using EQ to soften, brighten, energise or dampen a whole range of frequencies, you are reducing the bandwidth to a sharp, narrow point; identifying the offending frequency and then hugely reducing its volume, leaving the rest of the sound ‘intact’. There are three key parameters to adjust when it comes to a notch EQ: frequency, which is necessary to target the problem spot; bandwidth, to set a narrow range around this particular point; and gain reduction, to remove lots of volume. Visual tools work to make all three parameters easy to identify and adjust.
essentially a method of calculating the average magnitude of a wave with positive and negative elements), while dBV is referenced to 1V RMS. If you’re working exclusively with professional equipment then you can just about forget that dBV exist and concentrate on dBu. The only caveat is that you should pay attention to capital letters – dBu was originally known as dBv and is occasionally still referred to in this way by Americans. The dBv unit is exactly the same as dBu. We did warn you that decibels can be confusing! Decibel meters on hardware and in software are usually used to display dBFS or dBVU, measurements of the level of a signal relative to a defined maximum (in digital equipment this is the point just before clipping occurs, whereas in analogue equipment it refers to the highest level before the signal starts to degrade). In digital equipment, 0dBFS is the highest possible signal
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WorldMags.net Know your GUI While the walkthroughs elsewhere focus primarily on tone shaping treatments, once you begin searching for them, plugins dedicated to Pan, EQ, Reverb, Distortion and countless other parameters are awash with GUIs aimed at providing visual support for the sounds they produce. This becomes particularly important as, when used in isolation with preset settings, the parameters within plugin effects can often sound rather too extreme. For instance, if you manufactured a Tremolo plugin designed to produce creative panning effects, you would ensure that most of your presets push sounds from the far left to the far right so that they showcase the plugin’s potential fully. Equally, distortion plugins often produce deliberately enormous treatments that fill bass, mid-range and treble with saturated, overwhelming sounds requiring an EQ after them to tame their wilder frequencies. While plugin manufacturers can’t be blamed for attracting us with their effects this way, if every element of your mix uses treatments with such extreme settings, it will prove hard to balance as every element will be ‘inflated’. Aside from the originality from which your mixes will benefit, if you construct your own effects treatments, this is one of the main reasons we always recommend you avoid presets wherever possible. Or at least edit the preset starting points you like to bring them to a more appropriate treatment for your sounds. GUIs will assist here, particularly in the form of helping you identify the parameters that may lead to too extreme an effect solution. Use them to check stereo width, the results of filter treatments, length of reverb tails or the amount of gain reduction within a compression setting. Compare what you hear to what you see to find the very bestsounding results.
level. Attempting to push a signal beyond this level will instantly result in the waveform becoming squared off at the peaks. On professional audio equipment, 0dBFS is usually set at +4dBu. Commercial equipment tends to use -10dBV as the 0dBFS reference point. Analogue equipment tends to use dBVU (referenced in Volume Units) in a similar way to dBFS, where 0dBVU is the nominal maximum signal level. The Volume Unit is a hangover from the days of analogue radio broadcasting and tape recorders, where 0VU was the highest allowable signal level. Whether you’re working with digital or analogue, any signal above 0dBFS or 0dBVU is usually bad news in terms of sound quality.
Captain’s log One of the most important things to remember about all of these measurements is that the
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Volume reference The relative volume difference between the compressed and uncompressed signals is easy to see at a glance.
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Frequency boost Notice from the bottom of this line the centre frequency of this loud area in our treated sound. Now identified, it can be modified.
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Pan indicator The display shows just how wide and extreme this pan treatment is. Reign things in with the Width dial.
“Whether you’re working with digital or analogue, any signal above 0dBFS or 0dBVU is bad news” decibel scale is logarithmic. As such, a 100dB signal doesn’t have twice the power or amplitude of a 50dB signal and isn’t perceived to be twice as loud (power, amplitude and perceived volume are related, but not directly proportional). The base-10 logarithmic scale used by decibel measurements means that a
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3dB increase in signal level equates to doubling the signal’s power, while a 6dB increase equates to doubling the signal’s amplitude. However, the human ear’s response is also logarithmic, so it takes a level increase of approximately 10dB for the signal’s volume to be perceived as twice as high. Bear in mind also that this is true no matter whether you start at 10dB or 100dB, so 20dB(SPL) is perceived to be twice as loud as 10dB, while 110dB is twice as loud as 100dB. Small numbers on the decibel scale can relate to significant perceived level changes, so always be suspicious of anyone who uses phrases like, “It’s only a few dBs”!
VU and PPM A very specific type of level meter known as a VU meter was commonly found on hardware in the analogue era. VU is sometimes erroneously used to refer to any mechanical signal level COMPUTER MUSIC SPECIAL / 29
WorldMags.net > Step by step
Placing a vocal using only visual tools
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We’re starting with an extract from a song, constructed from separate beats, piano, bass and string parts, plus an unprocessed vocal on the final track. At this time, we’ve merely set an ‘overall’ level for it with no in-channel or auxiliary effects. As a result, it sounds flat and dull.
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The first of Alloy’s Dynamics modules is activated to provide visual feedback on settings, with the Threshold pane on the left showing how much gain reduction is applied, the Knee curve graph showing fluctuating levels, and the top pane showing gain reduction of each vocal phrase in real time.
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Delay and two reverbs are added to the vocal before attention moves to the output channel strip where iZotope’s Ozone 5 is on. The Harmonic Exciter module uses multiple Bands to apply modelled Tube saturation. The top display shows Bands and Amounts to help us make appropriate tweaks.
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Now iZotope’s Alloy is placed in the channel strip for the vocal so that we can control each stage of the processing with dedicated modules providing relevant feedback. Start with EQ, shaping a vocal sound to enhance breath and intimacy, while reducing some of the hollow, lower frequencies.
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With Sonnox’s SuprEsser set up on the second insert slot, we use a GUI to target sibilance within the vocal. Placing the frequency markers on either side of the area we want to reduce and then the Dynamics slider (with its own Attenuation LED ladder), helps us get a good level of gain reduction.
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Finally, the Maximizer module adds strength to the mix. The GUI helps, with the Threshold/Margin graph showing us the extra volume added. The top pane illustrates the amount of reduction/ make-up gain applied overall. Using our eyes as well as our ears, the vocal now sits well in the mix.
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Advanced analysis tools
Basic tools are included in most DAWs, but Blue Cat Audio’s analysis tools are much more comprehensive
Spectrum analysis Spectrum analysers are the easiest analysis tools to use and probably the most commonplace after level meters – so much so that they were once a regular feature on home hi-fi equipment. This is mainly because they looked pretty as they bounced around in time with your music. Oldfashioned spectrum analysers break the signal down into a series of frequency bands and display the levels (typically average and peak) of each band independently. More advanced modern software versions display a continuous curve, representing all audible frequencies. Most DAWs have a built-in spectrum analyser either in the EQ or as a separate plugin. Alternatively, check out commercial options such as NuGen Audio’s Visualizer suite (www.nugenaudio.com), which includes many of the tools discussed here. They can be put to a range of uses during production. Primarily, they can be a visual aid for EQ and mixing challenges. The
meter, but the VU metering system is just as clearly defined as the various decibel scales. VU meters display the average level of the signal, with the meter taking 300 milliseconds to rise from zero to its full level when a signal of 0VU amplitude is applied. Since the VU meter is so slow to respond, it isn’t useful for monitoring peak levels but should be used to monitor overall averages or average perceived loudness. Many VU meters feature an integrated peak level LED indicator to aid peak detection. It is worth noting that some level meters in hardware and software exhibit pseudo-VU style damped readings without truly following the VU specification. Care should be taken whenever you’re unsure whether a true 300ms integration time is being used. Peak programme meters (PPMs) are similar mechanical meters but with a much shorter 5ms integration time in order to display
frequency range of a sound can be quickly identified to determine where it sits in comparison to other elements. Unwanted low- or high-frequency content can be identified and the response of the entire mix can be analysed and compared to other tracks to see if it’s hitting the same frequencies as a good commercial mix. A further variation on the spectrum analyser is the waterfall diagram, which introduces time on a third axis, creating a three-dimensional graph of how the level of each frequency band changes over time. Spectrographs (often referred to as spectrograms after the name of the diagram they generate) are tools that produce a twodimensional representation of the same data shown in waterfall diagrams. Rather than using a third axis, spectrograms typically represent time along the x-axis, frequency up the y-axis and level using colours. Many people find them easier to read than 3D waterfall diagrams.
audible peaks in signal level. Unlike VUs, which fall as quickly as they rise, PPMs typically take between 1.5 and 3 seconds to fall back 20 to 26dB after they hit a peak (the fallback time varies by country). This means peaks can be held by the meter, making it easier for you to read their level.
“The decibel scale is logarithmic. A 100dB signal isn’t perceived to be twice as loud as a 50dB signal” WorldMags.net
So what are analysis tools? A good clue for the question above comes from another term sometimes used to describe them: sound visualisation tools. Though it has been appropriated by the iTunes-style visualisers, which create pretty patterns in time with the beats and melodies, it refers here to tools that provide a visual indication of the key sonic characteristics of signals you feed them. This is very useful for a simple reason: we can’t always hear the fine details of an audio signal as well as we’d like. Aside from the limitations of human hearing, there are other factors which come into play here. No monitoring setup is 100% accurate, transparent and revealing. In addition, some frequencies are beyond the range audible by humans. The lowest sub-bass frequencies can’t be heard by anyone who listens to your music or reproduced by even the heaviest club sound system – but they still take up valuable bandwidth in your mix. Spotting the presence of that low-end signal with a spectrum analyser would give you a clue that rolling off the bottom end with a high-pass filter could free up some bandwidth and allow you to make the rest of your track louder. Even if it only helps you to bump the rest of your mix up by a fraction of a decibel, that’s still a valuable improvement.
Peak indicators Let’s take a closer look at peak indicators, those tiny red LEDs that tell us a signal is too hot for our gear. On analogue equipment, it’s not the end of the world to see the peak LED indicator flicker momentarily every now and again (pushing into the red might even result in pleasant analogue distortion) but on digital equipment it should be avoided at all costs, especially when recording. In the days of analogue tape, it was desirable to use up every last bit of headroom in order to maximise the signal-to-noise ratio and dynamic range of the recording. However, this practice is no longer necessary with digital equipment, due to higher dynamic range and improved SNR. Furthermore, exceeding 0dB results in the painful and highly undesirable sound of digital clipping. At best, that means another take to fix COMPUTER MUSIC SPECIAL / 31
WorldMags.net Low frequency clear-out The low frequencies are always a problem area even for the most careful and clean mix. Uncluttering your bottom end prior to mastering is essential for final clarity and impact no matter how careful you’ve been. EQing kicks and basses so that they sound great together isn’t easy. Providing you have followed our advice and tuned your kick to the mix, you should find its fundamental frequency may well be the same as your bass. That means, if kick and bass notes play together the mix becomes soupy as your speakers struggle to reproduce overlapping frequencies accurately. It can be hard to hear what’s happening during a ‘flabby’ bottom end and mastering it will only make the matter worse. However, using visual tools, you can see what’s going on and use them to punch different ‘holes’ so the two work together.
> Step by step
If your kick drum and bassline combine they can dominate a track’s level and throw off your mastering
Using tools to see where bass and kick lurk
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Here we have the beginnings of a track with just a four-to-the-floor kick and a syncopated bass pattern. When a kick and bass note play at the same time, the frequency curves of the two sounds – which are incredibly similar – ultimately clash, muddying the mix. Fret not, for our EQ’s Analyzer immediately identifies the issue.
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The bass is rolled off below 80Hz using a 3.5dB low shelf with a 24dB/Octave high-pass filter employed below 34Hz. At 118Hz, a 4.5dB narrow boost is applied with a gentler, less ‘targeted’ 2.5dB boost centred 200Hz above this. The effect is a nice warming of the low mid-range. Again, the before and after treatments can be heard in our audio examples.
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To rectify the problem, we have to separate the sounds, prioritising the kick in the lowest frequency range and the bass sitting above. We EQ the kick with a +1dB shelf at 80Hz, plus a 3dB ‘Bell’ boost at the same frequency. Then, 4.5dB is cut at 118Hz to create a hole into which our bass fits. Listen to the pre and post EQ treatments in our examples.
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Ta-da! Here is the combined result. As with the single sounds, you will be able to hear EQ Out followed by EQ In via the clips found in our examples to compare mixes. Remember, we could swap this approach around with the bass carrying the lowest frequencies and the kick carrying higher, low-mid range ones.
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WorldMags.net Oscilloscopes Oscilloscopes are one of the most important tools for electronic testing and signal analysis, allowing waveforms to be visualised in real time. Originally developed as hardware units, oscilloscopes were primarily designed for engineers to test the functionality of electronic circuits on a variety of gear (not just for audio and not just operating in the audible frequency range). They are often used with function generators – similar to synth oscillators, but with a much higher frequency range – which feed a test waveform into the circuit. For audio purposes we already have a waveform to test – the audio signal itself – but not all signals are suitable for oscilloscopes. Oscilloscopes tell us most about signals of relatively stable frequency, amplitude and harmonic content. With these signals we can synchronise the oscilloscope’s
scan period with the frequency of the wave, making the waveform appear stationary on the display (most software scopes do this automatically). Want to check the inaccuracies that make one synth’s raw square wave sound fatter than another’s? Want to see how a lowpass filter rounds off the shape of that square wave until it’s a pure sine? Want to see how a distortion effect turns that sine slowly back into a square as you crank up the gain? You need an oscilloscope. It’s the best way to visualise that kind of simple process. Oscilloscope’s aren’t the kind of equipment you’d use every day but for visualising simple signals in real time they’re unbeatable and a great tool for learning about synthesis. Try Blue Cat’s Oscilloscope Multi (www. bluecataudio.com), which is available for just about every DAW priced at £33.
STEREO CORRELATION METERING
the problem. At worst, you’ve ruined the recording and won’t get another chance. Although many recording engineers will have their own slightly higher or lower preferences, we’d advise you to try and hit a peak level of around -15 to -12dBFS when recording to a 24-bit digital format. Leaving this much headroom should minimise the chances of clipping while giving you dynamic range and SNR far in excess of analogue tape. Even when mixing in your DAW, remember to keep your levels down. It’s a common mistake to push all the faders up and then use a peak limiter plugin on the master buss to bring the output level back down to 0dBFS without realising you’re harming your signal. Aim to leave headroom in every step of the signal chain (including between effects plugins) and you’ll ensure that no damage is being done.
If you want to get technical and really see what’s going on at the heart of your track, invest in a specialist oscilloscope
Things get a little more advanced when you begin using tools that make comparisons between two channels of audio (spectrum analysers and spectrographs typically simply average the two stereo channels when used in stereo mode). There are a number of highly specialised tools that use comparisons to provide us with information on variables such as phase and the stereo relationship between left and right channels. These tools take various names – balance meters, vectorscopes, stereoscopes, correlation meters and goniometers are probably the most common – and all represent the results in slightly different ways (typically a correlation bar running
from -1 to +1 or L to R, a polar stereo map, or a ‘Lissajous figure’ – a form of metering that is a little more tricky to read but can give very detailed information on phase, frequency and stereo relationships). All these tools essentially do a similar job of comparing two channels of audio. That usually means the left and right side of a stereo pair, but there are occasions when you’d want to compare two mono signals. For instance, you might want to check for phase cancellation when double-tracking, recording with multiple microphones or using parallel processing techniques.
“On digital equipment you should avoid making your peak LED flicker, especially when recording” Advanced metering Every DAW will include basic metering functions on channel strips, in the mixer section or as a utility plugin, but if you want to take your metering to the next level, there are a number of more advanced ways to visualise your sound
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signals, most of which involve measuring multiple variables. Metering starts to get a little more interesting when we measure another variable alongside the amplitude of the signal and the most common way of doing this is to measure frequency. Here are some examles… A spectrum analyser displays the individual levels of a number of frequency bands in order to show their relative amplitudes. Oscilloscopes measure amplitude and time in order to give a visual representation of the waveform you route into them. Spectrograms add a time element to the spectrum analyser principle, plotting three variables simultaneously in order to show how the spectral composition of a signal develops. These meters are incredibly powerful tools and a thorough understanding of how to use them will pay dividends in the long run.
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HOW TO GET LOUD (and win the loudness war) Achieving loudness to compete with professional tracks while maintaining dynamic range is an art in itself. How can you maintain dynamic integrity and get loud? We show you how… 34 / COMPUTER MUSIC SPECIAL
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When your track is played out on a big club system the venue’s limiters and compressors will squeeze the track to its maximum potential. Make sure your mastered track hits their equipment with maximum clarity and potency to ensure it fits in the DJ’s set and sounds as good as the tracks around it
Over the last ten years, a war has been raging that has remained invisible to all but the most observant of audiophiles and producers. Interestingly, though, despite its apparent invisibility to the masses, this war is audible to anyone with a radio, who buys a CD or who purchases a download. This war, known as the ‘Loudness War’, is waged in all forms of broadcast, recording and production of music. Even the most casual of music listener is now ‘enjoying’ music that has had the dynamic life strangled out of it by this war. So what are we to do?
Range wars The first option is to wring our hands, moan about it and demand that everyone simultaneously stop using hard mastering limiting. After all, if good old-fashioned dynamic recording was good enough for Steely Dan, then it’s certainly good enough for the so-called ‘music’ that today’s youngsters like to bounce around to… The second option it to get real. Sure, the artificial mangling of acoustic and dynamicsrich recordings is clearly a no-no, but if you’re making that ‘music that youngsters bounce around to’ you need to make your tracks as loud, punchy and in-your-face as the competition. And that’s what this feature is all about: making the most of your dynamics while still winning the loudness war.
So, leaving aside the technicalities for a moment – and wearing the hat of a music consumer, rather than producer – in what guise does this war exhibit itself? If you peruse any online music store and audition a range of tracks from different albums, artists and eras, you will notice that some will sound louder than others. Although there may be differences between genres, you will notice that modern-day pop productions jump out of the speakers more readily and in order to maintain a similar listening level for each, plenty of volume control action will be required (if you are using iTunes for this, ensure that Sound Check is disabled, otherwise you will get false results). This trend for ever-greater perceived volume is how the loudness war manifests itself, but as we shall discover it is quite possibly a war that can never truly be won and winning over ‘hearts and minds’ is a fruitless pursuit.
“Human hearing determines loudness not by its highest peak, but by a shortterm average” WorldMags.net
What is loudness? The key point in all this is that we are talking about perceived loudness, not peak level. Take a look at the Peak and RMS Explained box (page 39), for more detail on this. In a nutshell, though, we need to be aware of the fact that the human hearing mechanism, including the processing required by our brains, determines loudness not by its highest peak, but by a short-term average. If we take our selection of listening tracks and pass them through a level meter, we will see that the peak levels often remain reasonably consistent between tracks. Those that sound ‘louder’, however, spend more time in the top range of the scale. By manipulating an audio signal in a number of ways we can fool the listener into thinking that it is louder, although this can be to the detriment of the original’s sonic character if pushed too far.
Manic compression So what is happening in our louder modern tracks? Critics describe the worst examples as being ‘squashed’ and this refers to the restriction of the dynamic range (to boost perceived loudness) by using some form of compression. This compression usually takes the form of a limiter (generally, a compressor with a ratio higher than 10:1) which works by reigning in spurious peaks over a given level and then allowing the overall signal to be COMPUTER MUSIC SPECIAL / 35
> make music now / how to get loud
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You’d expect a Metallica track to be at the louder end of the sound spectrum, but not at the expense of the sound quality…
Going too far… Metallica’s Death Magnetic, or how not to master an album Released in 2008, the album Death Magnetic – and more specifically a track called My Apocalypse – caused controversy in some circles due to the extreme peak limiting exhibited on its CD release version. Mastering engineer Ted Jensen is alleged to attribute it to extreme processing at the mixing stage,
by production legend Rick Rubin, no less. The ITU (International Telecommunication Union) that has a standardised system for loudness measurement confirms that the album is one of the loudest ever produced. You’d think that, for a band like Metallica, this was exactly what
they had in mind. However, the levels peaked too high (as our shot of the waveform, below, shows) resulting in ‘clipping’, that is, the digital medium couldn’t handle such high levels and rounded them down, resulting in clearly audible distortion. The result? The first casualty of war as the technology used to
If your track’s waveform looks anything like this one belonging to Death Magnetic, it’s time to rethink your limiting
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produce the track proved detrimental to its quality. Interestingly, the Guitar Hero versions of the album are from a less compressed, some argue better, master, which only served to heat up the debate. However Death Magnetic remains an important lesson in pushing things too far.
WorldMags.net boosted (using a simple gain increase) right up to the maximum available peak (see the Headroom box at the bottom of page 39).
Audio distortion Clearly, the audio is being reshaped (distorted) in some way – peaks are rounded or chopped during the limiting process – and while very short distortions are difficult for the ear to perceive, the further this process is pushed into the realms of completely flattened peaks, the more unsatisfactory the result. This brickwalling or ‘crushed-to-death’ clipping leads to tracks that are tiring to listen to, especially when heard on the radio where further processing is used. While punchiness is a good thing if handled well and can make a track really stand out, whether in the car, club or at home, pushing it too far can result in a track lacking authority and sounding bland.
The blame game So who has been driving this pushing of the envelope? Clearly, mastering engineers have a part to play in this as they are the ones sitting at the controls for the last part in the music production chain. But to lay the blame firmly with mastering engineers would be unfair. They generally have to do what their clients (record companies) request in terms of dynamic tweaking and might conceivably have a hard time convincing an A&R person that the next Girls Aloud album should have a greater dynamic range to allow the tracks to breathe.
“Hotter mixes passed through the average radio processing chain merely sound distorted, not louder” There are a number of arguments as to why music should be squeezed to maximise loudness, mostly based on the idea that if a track sounds quieter next to the competition, sales will suffer. The fact that we increasingly live in a single song (as opposed to album collection) culture is often cited, as is the need to ‘sound big on the radio’. Neither of these is particularly convincing, especially given that radio stations have so many tricks of their own to max out the loudness. In fact, research conducted by two radio processing gurus and published by mastering legend Bob Katz has proven that ‘hotter’ mixes passed through the average radio processing chain merely sound more distorted (and tiring) rather than noticeably louder.
Making Waves Ultimately, the Loudness War appears more like a nuclear arms race than a search for sonic perfection in that it’s never going to end well. There is, however, one additional part of the
equation we’ve yet to mention that helped foster the ‘make it really loud’ mentality, and that is digital processing. In fact, we can narrow it down to one plugin in particular that started it all: the Waves L1 Ultramaximizer. One of the great advantages of digital technology is that there are easy ways to utilise so-called lookahead processing. If a plugin knows what is about to happen in terms or levels, then pre-emptive action can be taken. When applied to limiting, this leads to greater possible degrees of squashing without noticeable distortion. Now anyone at home can mix a track and run it through a plugin to make it sound louder and ‘better’. This is not a criticism of plugins, rather the way in which they are applied. So how much should you squash your own mixes? Well that depends very much on genre and you should compare your track to similar commercial tracks – there is no right answer here. Just remember: the idea is to present your tracks to best effect, not just to make them sound loud.
TUTORIAL
FILES Broadly speaking, the range of audible frequencies can be divided into seven sound characteristics. Need more or less of these? Cut or boost at the relevant frequency
Using EQ to achieve more punch and power Assuming you have done your best to mix a track with plenty of space and a sympathetic spread of frequencies, there might still be some room for enhancement before moving to overall dynamic control. This is where a master buss EQ might be useful, but before embarking on a dose of EQ tweaking, you should be aware of a few basic rules… Boosting frequencies is more noticeable than cutting them and is usually best applied in a broad manner rather than selecting a narrow band on which to work. Boosting a frequency adds noise and data
from your DAW or plugin. It might sound great, but in reality you’re adding signal that isn’t part of the original. Cutting, on the other hand, removes original signal and doesn’t add anything ‘fake’. So, if your mix lacks bass, instead of boosting the low end, how about cutting the high end? That way you’re not introducing the colour of that particular EQ’s gain stage, which may be undesirable. In terms of specific frequencies to boost or cut we need to turn to the world of subjective timbre. If your mix sounds thin then you’ll
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need to address the frequencies from about 70-550Hz. Take care, though, as too much boost between 300-800Hz can sound boxy. Added warmth can be achieved by dipping between 3-7kHz, but this can remove brightness if overdone. Probably the best place to start is to remove extreme low end below about 40Hz and perhaps apply a scoop at around 250Hz. Edginess can be added in by boosting at around 4-5kHz. Be sure to check out our nine audio examples of the EQ treatments discussed, on varied sources.
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> make music now / how to get loud
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Space is the place Silence in your track is one of the most important aspects in maintaining an high perceived volume level Given the prominence of selfproclaimed ‘miracle’ mastering plugin suites and loudness maximisers, it’s easy to forget that the source material plays the most important part in all of this. No amount of multiband übercompression or equalisation can turn a drab track, mixed with no flair, into a hit. This means that you should start with the basics when addressing any shortcomings in your track. Even before you start mixing a track, the placement and choice of instruments and sounds will have a massive impact on the final result. Space – the stuff that isn’t there – is almost as important as the music itself. In fact, the gaps between notes determine how effective any overall compression will be and how much you can get away with. Moments of silence will cause compressors and limiters to retrigger once the signal returns, assuming your attack and release times are shorter than the gaps in the sound. This provides punch and perceived loudness more effectively than constantly driving your limiter. This also extends to the way in which elements are spread throughout the frequency spectrum. Choose the right sounds that sit in their own sonic space and don’t tread on other areas of the mix. Many a mix can be tightened up by tweaking the existing sounds, something that should be relatively easy in the plugin realm. Take the time to revisit basslines and synths and explore how the amplitude and filter envelopes might be adjusted for more punch. Sometimes just shortening the release stage will do the trick. This is also the time to check that the timing of notes isn’t sloppy. A classic and highly versatile production trick is to use a compressor in Sidechain Input mode to momentarily push heavier elements out of the way and increase impact. Let’s take a look at how to achieve that effect. And don’t miss our audio examples, where you’ll hear lots of sidechain compression examples, including sidechained reverb and triggering sustained synth notes.
channels through the effect easily. The compressor should now be configured to trigger its gain reduction using a separate input. The classic treatment for dance tracks is to use the kick drum as an external trigger and to route bass, synth or pad elements to the compressor. With the kick being so important, this means that each time the kick plays it will push down the level of other elements routed through the effect. Remember, though, that you
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genres of music, not just dance, and a kick drum isn’t the only instrument you can use to trigger the dynamic squashing, either. You could try using the snare channel to trigger a level change in a full-on percussion loop (as seen in the screenshot, below). This is something that works particularly well in electroinfluenced genres where a full and fat snare is an essential component. And what’s to stop you routing both the kick and the snare to the sidechain? Absolutely nothing at all, that’s what.
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Let’s start with the basics
1 and apply a compressor to a mix buss or group channel. This allows us to route different 38 / COMPUTER MUSIC SPECIAL
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Something that works on a
3 more subliminal level is the use of sidechain compression to achieve ducking delay and reverb effects. In the example below, the channel features a synth riff that needs to sound punchy and dry when playing, but requires some ambience to fit into the mix as it ends. By routing the channel (via an effect send) to a long-decaying reverb treatment and then applying a compressor after the reverb (to push the level down when the synth is playing), it is possible to create more space in your mix.
WorldMags.net Apple Logic’s MultiMeter charts decibel response across the frequency scale
UAD’s Precision Limiter offers super-accurate metering and tracking of peak levels
Peak and RMS explained Measuring loudness is all about perception, really, and it’s a tricky beast to nail down. How loud we perceive a sound or mix to be depends on how long we have been listening to it for and its overall frequency content, among other things. The peak level meters in most DAWs generally do a good job of showing audio activity and flag up potential clipping/distortion problems, but they don’t indicate actual loudness. To attempt to quantify loudness, we need to take a look at
average level – the peak level averaged over a given time frame. The problem here is that there are numerous ways to calculate that average. Thankfully it is generally agreed that the RMS (Root Mean Square) calculation method provides the best results. Running two different tracks with very different peak levels through an RMS-capable meter allows you to make informed choices about their relative loudness, regardless of musical style.
Another useful concept here is that of the Crest Factor – the peak-to-RMS ratio. The lower the Crest Factor, the more likely an audio source has been heavily compressed. To get a sense of how far to push the dynamic processing of any mix buss or master section, you should pay attention to the RMS level of one of your tracks and compare it to that of a similarly styled commercial counterpart that you like the dynamics of.
What is headroom? Headroom is defined as the level permitted for peaks above the nominal level. The nominal level is the average or RMS level (see the Peak and RMS explained box, above) at which an audio device is specified to operate. The idea of choosing a nominal level comes from the days of analogue signal chains. A wide range of kit could work together with a guarantee that the working range of any internal amplifiers was not maxed out. It is worth bearing in mind, however, that digital meters are not always entirely accurate and this might cause problems with transient detail. This also assumes that no further processing might occur – as soon as tweaking takes place on a ‘hot’ digital signal, peaks once again begin to clip. Noise
TUTORIAL
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is not generally an issue with digital systems, but it is worth getting a healthy level when recording or bouncing a final mix to a DAW. 24-bit offers clear advantages over 16-bit here as it’s possible to leave a
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wide amount of headroom (26dB or more) with negligible impact. Don’t take our word for it – listen to our audio clip to hear what increasingly less headroom sounds like.
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WorldMags.net Quick steps to great mastering Let’s kick proceedings off with a disclaimer: what we’re describing here is not intended to replace the ears, equipment or expertise of a decent mastering engineer. However, there are some key tools and techniques that will allow you to maximise your own sound and that’s what we’re going to discuss… In an ideal world, you should aim to achieve a well-balanced and dynamic mix. Get this right first and then treat master processing as the icing on the cake. However, if you are on a tight schedule, mixing with a full master-style plugin chain can deliver reasonable results. It also often allows for less accurate channel levelling, although this is possibly at the expense of flexibility further down the line. That said, sometimes moderate master buss compression will help tie everything together
> Step by step
and there are engineers who favour mixing with it in place from the start. As a general rule, the basic digital mastering chain tends toward the following routing: EQ->Compression->Limiting->Dither-> We’ve discussed EQ elsewhere, but it is worth remembering that this comes in many flavours – DAW channel strip, linear phase, vintage emulation, etc – and each has its own ‘vibe’. Compressors also come in different styles, and each plugin will have its own range of controls. A standard broadband compressor works across all frequencies equally. This can work well in many cases, but if you want to experiment with greater levels of compression with fewer side effects, then you may wish to explore a multiband variant. Beware though: it is
very easy to ruin a perfectly good mix with the haphazard application of multiband processing. Limiting should be applied after all other dynamic processing and its aim is to maximise the final level (as opposed to compression, which is also used for dynamic shaping). If you are producing a mix for CD or commercial download then you will probably be looking at exporting to a 16-bit audio file. If this is the case then you should consider applying dither as the last stage in the chain. Specialist tools, such as those for spacial management, are sometimes used at various stages of the chain, as are sonic maximisers, but these all have their own particular way of interacting with the audio. Use your ears to determine if the result sounds better or worse and try to avoid the temptation to use only factory presets.
Essential first steps for preparing your track
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Adjust EQ: try adjusting the bandwidth of EQ frequencies, but keep boosts broad (have a large Q setting so that peaks appear wide and flat – that way you’ll be making broad musical changes rather than boosting a particular element of the sound). Only use EQ for sweetening the track or correcting problems when mastering.
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EQ types: explore the options by switching between Peaking, Shelving and Filter settings. These three basic types of EQ will give you the ammo you need to isolate and enhance (or remove) anything you need, but – as we said earlier – use very broad strokes when mastering. Mixing is the place to truly blend and complete a track’s frequencies.
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Compressor ratio: compression doesn’t have to be heavy-handed and it’s important to know the basics of your chosen compressor before you go too crazy at the controls. The ratio determines how much compression occurs when the threshold is reached.
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Limiter auto release: if enabled, this adapts the release time to the program material and tends to deliver more transparent results. Turn on the Auto Release first and odds are you’ll be able to get a sound you like. If it’s not cutting it, turn it off and tweak the Release yourself until you’re happy.
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Compressor threshold: pushing the threshold down introduces compression on lower level areas of the mix (so a lower compression threshold ‘exposes’ more of the track to the compression effect). Always start with a high threshold and then reduce it until you hear it start to take effect. Stop when you like the sound.
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Compressor attack and aelease: use these to shape the way compression is applied. Slower attacks allow transients to breath. That is to say, you won’t get any unnatural, choppy sounds where levels are artificially altered. A slow attack gives a subtle, natural sound. Faster attacks let you hear the compressor work – a cool sound when mixing, but not something you want to inflict at the mastering stage.
WorldMags.net How to create a slamming master chain Don’t want to fork out for a professional mastering session? Here’s how to achieve solid results with your built-in plugins… Like any professional mastering engineer, getting to know your kit is essential for good results, as is comparing your track with others. So far we have talked about compressors and EQs almost as kinds of generic processes, but as in the analogue world, different designs result in widely varying results so choice is everything. Here’s a typical master chain: first up is a Pultec Pro EQ that provides the coloured sound of the Pultec MEQ-5 and EQP-1A equalisers. Next, let’s go with the
> Step by step
UAD 4K Buss Compressor that emulates the console dynamics from the famous SSL 4000 G+ mixing desk, although this is sometimes replaced with the FATSO Sr Analog Tape Simulator & Compressor for a wider range of ‘warmifying’ options. These are followed by the UAD Precision Equaliser and Precision Limiter for uncoloured tonal tweaking and dynamic control. These are all premium priced plugins, but there are plenty of more modest alternatives – and don’t discount your DAW’s builtins. Don’t forget to check out the included audio examples to hear the before and after!
Modern pop – such as that from The Black Eyed Peas – is prone to slamming mastering
Build your own master chain
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EQ: a good starting point here is to ensure that there are no extreme low frequencies present. Use a high-pass filter to get rid of anything below about 40Hz. This might sound counter-intuitive but it clears the way for greater perceived loudness with compression and limiting. Sweet spots vary, but try exploring 2-5kHz and 250-600Hz and see how tweaking these change the overall sound.
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Spatial enhancement: here we’re using a stereo enhancer to increase the width of the mix. This technique mixes elements from one half of the stereo field to the other in anti-phase (and vice-versa). A stereo processor such as this one from Sonalksis also provides a way of removing low frequency stereo information that cannot be detected, but may defocus the mix and mess with low-end punch.
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Compression: start with a broadband compressor and set it for a 2:1 ratio with a medium/fast attack and moderate release. Bring the threshold down until you get 4-6dB of compression. Tweak the release. Get the compressor to apply gain reduction with a rhythmic flow that follows the groove. This is a great way to add punch, especially if the attack is adjusted to maximise transient detail.
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Limiting: this is the process that has more to do with loudness than any other. Most limiters have a small range of controls. Push the Input Level control up to achieve more limiting and use your ears (and a decent RMS-equipped meter) to keep things in check. The Release control is often used in Auto mode to track the dynamics of a mix. Pushing limiting too far leads to peak chopping, which is not good.
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Transient designer: for more extreme manipulation of the transients, consider a transient designer plugin. This was a concept originally put into practise by SPL, but is now available in a number of plugins, including Logic’s Enveloper and the Cubase Envelope Shaper. Unlike a compressor, the Attack and Sustain/ Release controls directly enhance or inhibit their respective components.
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Dither: the least glamorous part of any mastering chain, but of particular importance if your music features a wide dynamic range. Dithering is the process of adding a tiny amount of noise back into a digital system in order to randomise quantisation errors. In basic terms, this improves low-level signals such as reverb and piano decay tails when converting from a higher bit depth.
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Need a track mastering? Why not go to the world’s most famous recording studios and see what they can do?
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Making a track is hard enough. There’s the business of writing, recording and bringing all the many track elements together. There’s blending them through the mix stage where sounds are treated with effects such as EQ, compression, reverb and delay. Then, before your track goes out to the public, there’s one final stage: mastering. But what is it and is it really necessary? Before you begin thinking about any moves to improve the track, it’s worth remembering that mastering tackles administrative criteria as well as sonic ones. It’s frequently the case that pop albums feature tracks mixed by a number of different producers and the first job a mastering engineer faces is to get the running order of an album right and ensure a smooth flow. He is also there to make sure that no track is ‘clipped’ at the front and that each finishes smoothly at the end, whether this means a long fade over an outro chorus, for example, or a shorter one to curb the length of a long reverberant tail.
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He’ll also tag tracks with the necessary ISRC codes to make sure that royalties you deserve arising from radio or TV play of a track are assigned to the right artist.
Tools of the trade The fun bit is making sure that the levels and dynamics of tracks are consistent and this is where the creative part comes to the fore. A mastering engineer has control of tone and dynamics. This is the final stage in which the stereo (or surround) mix is treated one last time, to wrest the best possible result. This could be adjusting bass or treble balances, addressing problem areas relating to dynamics or specific frequency bands and ensuring that the overall volume of the track is competitive – that is, that the volume is roughly the same as other music of similar genres and that the resulting track won’t sound limp when played out in a set. The tools used during mastering are broadly the same as those used at the mix stage as,
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Master like Abbey Road What is mastering? Do you need it? What happens when the great Abbey Road gets to grips with your track and how can you get that sound at home? We’ve tips, tests and more to ensure home mastering magic
effectively, a similar job is being done again, albeit gelling together all the various mix elements with a single cohesive treatment. So, rather than mastering representing an opportunity to right any wrongs, it’s an opportunity to add gloss, sheen or weight to a track that is already complete, either to upload it to an MP3 store, write it to Red Book CD format or prepare it to be cut to vinyl, where mastering is a must.
Misconceptions Perhaps the biggest misconception about mastering is that it represents an opportunity simply to slam the output level of a mix as hard as possible. As you’ll know, digital recording and mixing provides us with a maximum volume level of 0dB, so it’s easy to find this confusing. After all, if your mix peaks at 0dB for one brief moment, it’s as loud as it can be – right? Well, no, as it happens, as that’s ignoring the way that we actually hear. Humans subconsciously respond
to average volume, so if the rest of a track is made louder leading up to and after this peak we’ll hear it as ‘being louder’, even though the peak moment will be at the same volume as before. Taken to extremes, this means that you could bring all program material up to the same volume as that peak, resulting in a track that is within touching distance of 0dB from the moment it starts until it ends, the net result being a ‘loud’ mix. This drive towards maximum volume results in records which flirt dangerously with distortion. Fortunately, the current mood in mastering circles favours a more natural result, with less obviously squashed dynamics. This is both preferable and increasingly common, though loudness maximising remains a crucial part of the mastering process. The trick is to play the game, to master loud, but to do it smartly. It’s still true that a record mastered without any loudness maximising stands little chance of being received as warmly as those
with the treatment. Much of the natural energy and weight of a track is derived from that perception of a high average volume.
Mix before mastering As well as basic volume the other crucial weapons at a mastering engineer’s disposal are EQ, to address frequency balance issues, compression and multiband compression, to provide different volume solutions within different frequency bands, and effects such as stereo spread, to widen and lift a mix. While these elements can all be brought into play to ‘give punch to drums’ or ‘bring out the groove’ of a track, it’s vital that mastering is never seen as a cure for a bad mix. The levels of access and potential for recovery in mastering are markedly less than when you’re mixing. So make sure you’ve done your best before you commit to stereo or there’ll be some timeconsuming, frustrating and potentially costly work ahead.
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WorldMags.net Before you begin… 5 steps to getting your track master ready
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Firstly, you should ask yourself if you are the right person for the job. Are you too close to your track? Are you sure that your track wouldn’t benefit from a fresh pair of ears? Even if you’re bent on not paying a penny for a pro, it can be an interesting idea to at least let your mate have a go on your behalf. Try it. You might be pleasantly surprised.
2 Where will your track be played? Decide before you start the mastering process
Decide what the mastered track is principally for. Is it heading to radio or TV, or is it being manufactured for CD? Is it heading straight to digital download or will it be mostly played at home? Or is this the next club smash intended to be played out large? Your track needs to be mastered according to its purpose so be sure to tailor your output channel solution accordingly.
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Beware: mixing and mastering are not the same thing
Don’t confuse mixing and mastering. Increasingly, would-be producers and programmers put an output channel solution into their autoload song starts or, at the very least, have one on the output channel long before a track is fully mixed. While there’s mixed opinion on this approach, you’re more likely to have difficulty isolating problems in your mix if you’re sticking to a set preset or chain and not adapting the master to a finished mix or fully understanding what’s happening in your signal chain. If you are determined to start with a master chain, it’s good practice to regularly A/B your mix with it in place, then bypassed.
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It’s important to master in a room that is providing accurate playback of the track you’re working on – even more so than when mixing. Complete acoustic treatment solutions can be expensive, but if you’re determined to carry out all of your own mastering then this would definitely be a worthwhile investment.
5 Make sure the room you master in gives accurate playback of your track
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Give your mix some room. Remember that whether you’re sending your mix off for mastering elsewhere or tackling it yourself at home you should start with a mix that isn’t already bumping into the maximum. Mastering will take care of this, so give it room to do its thing.
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Abbey Road VS
Pro Studio VS
Your Studio Which mastering process is for you? We take a reader’s track and master it three different ways…
THE TEST TRACK Lumeria, John Andio
We wanted a track that had potential but was let down at the final hurdle. We chose Lumeria by John Andio, aka Cyberoptikon – https://myspace.com/cyberoptikon. It’s an ambient dance track with an unruly sound. What does its maker reckon the track needs? “I have noticed that it doesn’t have enough impact and may not be loud enough,” says John. “Over the past year I have become a fan of Above and Beyondstyle mixing. I wanted a thick bottom end to work with the high choral arrangement.”
So far we’ve discussed many of the common misconceptions of the mastering process and hopefully spelt out the options available to anyone who’s serious about making their tracks sound as good as they possibly can. So, where to go and what to do? The first option, of course, is to have your track mastered by a professional mastering house. Many large, well-respected recording studios have mastering rooms attached and staff who have spent decades learning the dark art. Alternatively there are many mastering studios that solely concentrate on the mastering process rather than any recording. If you are cutting bass-heavy vinyl dub plates you might want to find someone that’s a true specialist in this area. The next option is to trust your local friendly recording studio (or benevolent skilled engineer) to master your track using their gear, time and ears. Any engineer worth their salt will advise you and serve up a bunch of suggestions as a fix. If nothing else, having your track played loud on a studio’s system by someone who will give you a second opinion on your mix is always going to be useful. Ask at your local studio.
They’ll doubtless do you a competitive rate for mastering or, if they think there’s the smallest chance of you recording there, they might give you their opinion (if not a full-on correctly formatted 16-bit 44.1kHz WAV) for free. Finally, there’s the good old-fashioned DIY option. With so many mastering effects plugins available and the ever-increasing power of your DAW, you really do already have all you need to do the track yourself. The only question is, are you brave enough – and objective enough – to handle the process?
“You have all you need to do the track yourself. The question is, are you objective enough?”
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In order to put all three options to the test, we employed the services of three experts and charged them with the task of making our test track sound better. So is there a right way to do it? Will all of our engineers approach the challenge in the same way? And just what will the results sound like? Read on to find out whether one of the three methods comes out on top and exactly how our three mastering experts achieved their results. COMPUTER MUSIC SPECIAL / 47
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The Pro Studio Mastering with Stuart Bruce
Stuart Bruce Stuart began his career at Sarm Studios, London, in 1981 where he worked with Duran Duran, Frankie Goes To Hollywood, Stevie Wonder and more. He immersed himself in the emerging technology of the period and began a career as a programmer and remixer. Today, he splits his busy schedule between the UK and France.
With a wall of analogue gear as big as a library bookcase and snakes of cabling winding every which way, Stuart Bruce’s studio is an Aladdin’s cave of classic kit. This veteran of production employs any means possible to get the sound he’s after and if that means experimenting into the small hours with options that ‘just might work’, then he’s your man. His dedication to creativity has taken him all over the world in his 30-year career. Faced with our test track, Stuart intended to use the best of analogue and digital to deconstruct then reconstruct the track for the better. And while some of the techniques he used owed a little to his engineering and production skills, he couldn’t wait to get the track out of its Logic home (from where he played the 24-bit 44.1kHz track original) and out into his wall of hardware goodness. “It’s very dense, isn’t it?,” Stuart says on first play through. We agree. His Sonodyne monitors put in an able job of presenting the wall of sound that the track packs, but it’s clear that there’s work to be done. Meanwhile the smaller, powered Yamaha MS101 MkII near-fields only serve to reinforce the lack of range in the track. Stuart spins back, sits back and goes again. By the second and third playback he’s made an important discovery. “Look,” he says, sparking up BX Digital’s bx_control. “A huge amount of the track is in mono. All of the drums, in fact,” and to demonstrate he flicks a virtual switch to mute the mono signal and – bingo – all of the drums, save for a shimmering ghost of the snare and hats, disappear. “The only part of the drums that’s in stereo is a little reverb,” he observes. “Everything else is straight down the middle and that’s why it sounds so dense.”
The heart of the matter It’s time for some bolder moves. Stuart sets about routing the audio out of Logic and back in again and it’s clear very early on that this monoism is the heart of the track’s problems and one Stuart has an idea to fix. Using bx_control, Stuart separates the track into Mid/Side format. This recording technique splits the track into a central mono element and a combined left and right element. In this case, it allows us to get at and treat the problematic central component independently of its soupy middle. The central element heads first to the SPL Transient Designer for a little sharpening. Attack set at five and Sustain at three just boosts the presence of the beats. Next, on to a GML EQ where it’s boosted “a little” at 50Hz, 200Hz and 1.5kHz. And again “a little” at 450Hz, just to make sure. “This setup has the effect of increasing the air and clarity in the drums. “Meanwhile the surround element goes to the Tubetech CL 1B compressor where I’m giving it a 6:1 ratio with medium Attack and Release. This is 48 / COMPUTER MUSIC SPECIAL
just for a little warmth and colour,” says Stuart. “It’s a very digital sounding track, very grainy, so this helps cure that. “From there, the stereo signal gets sent to a second GML EQ, cut at 50Hz and 120Hz, effectively taking out some of the weight of the pads, and allowing room for the drums. Again there is a little top-end boost at 1kHz and at 16kHz that has the effect of opening up the track so that it appears more stereo. This is to offset the increasing level in the centre – the drums – as that has the effect of making the track seem less stereo.” Time to recombine back in the box. “It all comes back into bx_control which converts the track from M/S back to stereo,” explains Stuart. “I’ve dialled in a little of the stereo width parameter here. It’s at 128% – again, just to get some space in there.” Is that it? “No, there’s more!” says Stuart, reaching for a final couple of plugs. “There’s a Waves JJP
“The result is remarkable – the detail on the octave bassline is audibly more pronounced” Compressor here – a PuigChild 670, which is their Fairchild emulation. It’s in lateral and vertical mode and is compressing the mid of the track lightly. It’s just tickling those drums, no thrash at all.” Sounds good so far. “And then we’re into the UAD Precision Limiter to just knock off the peaks. It’s not giving the track any more level. It’s at +1dB on input and -0.1dB on output, just to make sure there’s no clipping come CD authoring time.” The final result is remarkable. The detail on the rapid octave bassline is audibly more pronounced as we toggle between the original track, leaving Logic and the returned, recombined version of the track arriving back. Similarly there’s an increased ‘slap’ to the snare drum and more of a sense of hi-frequency excitement. The track definitely feels more exciting and buzzing than before and while it’s actually a little quieter than the original, the increased sense of space and stereo width is a very welcome addition. Plus it’s ever-so-subtly wider, giving that wall of pads and reverbs room to breathe and make their individual presence known. It’s anything but conventional, but the result is a bettersounding track.
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A few playbacks and Stuart has identified an issue: much of the track is in mono
What was done: step by step
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Abbey Road Mastering with Geoff Pesche
Geoff Pesche Geoff has worked at Sanctuary Townhouse Studios and Abbey Road Studios. New Order, Mike Oldfield, Beta Band, Aphex Twin, Therapy, Lily Allen, William Orbit, Adele, Arctic Monkeys, Snoop Dogg, Coldplay, Kylie and Gorillaz are just some of the artists he has worked with.
Call tech support “A good example of the technical expertise here at Abbey Road is this fader. It broke the other day and went open circuit. It’s noisy – it’s a big deal. “This isn’t new kit but they were able to take it out and replace it with an identical tested replacement. Look at the sticker: tested 24/08/98… ready to go. It was taken out of its box and slotted in here. Fantastic.”
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If you’re going to have your track mastered professionally, why not have it done by the very best? We spared no blushes and went to the top of the tree, Abbey Road Studios, aka the world’s most famous recording studio for their take on our track. You’d be forgiven for thinking that our nerve must match our wallet, but today’s Abbey Road has its iron curtain folded neatly away and it is possible for mere mortals to get the same treatment afforded to the Fab Four. It costs just £90 to have your track mastered in the selfsame building that produced Sgt Pepper. The pricing is all thanks to Abbey Road’s new online mastering service. You visit their website (www.abbeyroadonlinemastering.com) choose what you want from simple drop-down options, upload your track, key in your card details and within five working days one of the experts at Abbey Road – the same experts that master smash hits for artists from Kylie to Coldplay – will give your track the treatment. There are three mastering rooms in Abbey Road Studios, each staffed by a pair of engineers who work alternately. Dishing out the jobs is Lucy Launder, who heads the Mastering department and uses her own keen ear to match the right engineer to the right piece of music. Remarkably, tracks submitted via the online mastering facility aren’t treated differently in any way to the studio’s other work. Lucy administers a central spreadsheet and your chosen engineer really will be working on your track in between mastering future hits for worldfamous names.
vinyl showing no signs of disappearing, Geoff can just as easily cut your track to 12" or 7" vinyl masters in addition to the rather more boring but absolutely vital WAV file and CD. The first part of any mastering session is the listen through. Geoff assumes the position (ensuring that all journalists, PR, management and photographers are well behind the magic listening point) and cranks out the track. “Perfect!” shouts Geoff above the mix. “I know what this track needs,” and with that he begins to quickly tweak the desk. “I can reach a verdict on a track quite quickly because I know the room so well,” he says. It’s a curious process – at no stage do the changes employed seem wild or sudden – but Geoff is perpetually working the desk. Another playback. And another. “I don’t use plugins, I use boxes. It’s all hardware,” explains Geoff. “People don’t have this kind of stuff in their homes because you can’t buy it. These are EMI modules from 1972 and I use this on passive stuff that doesn’t need a
Vintage magic
lot of work. This track needs a little more so the Prism EQ is a little harder and more precise,” says Geoff. “Good mastering rooms will have two sets of EQ and a good stereo compressor, good faders and good converters. If you see a mastering room with racks and racks of gear, the guy’s in a quandary – he’s got too much stuff! What we’ve got here is proven and really works.” So what does Geoff think of the track so far? “It’s loud, but it’s not brick-walled. People think it’s got to be loud before it gets to me, but that’s not the case. I make it loud – with a four-grand limiter! Headroom is important, so they need to retain the dynamics and let me do that.” Geoff continues his work. “20 years ago we were putting the kitchen sink in, but people can spend their time perfecting what they’re doing now because they’re in their bedroom. They’re not paying for studio time. A lot of it sounds great – mixes are better than they’ve ever been – and with mastering, if it ain’t broke, you don’t fix it. It’s not always necessary to do something and in those cases what you’re paying for is me saying, “It’s right”. You’re never going to hear your track as good as in here because this is a controlled listening environment, so if it sounds good in here, bingo.”
Our designated engineer is Geoff Pesche. With 20 years of experience he has a CV that reads like an encyclopaedia of music’s great and good. Two PC-based Sadie systems – one for playback, one for recording – take care of the audio as we listen. Monitoring is taken care of by a pair of fridge-sized B&W Nautilus 801 monitors, each with a Bryston amp, while industry standard Yamaha NS10s provide the grim-faced boxy nearfield. The real magic, however, comes in the form of Abbey Road’s bespoke EMI TG12410 mastering desk, with each of the three rooms having its own identical console. Essentially this is a rare analogue EQ and filter with notched dials for settings. There’s no total recall here. Geoff logs all the settings of every session in a notebook so that he can return to a project should a tweak be required. After the vintage EMI magic comes a Prism Sound MEA-2 Mastering EQ for EQ and compression and then on to a Jünger Accent 2 Digital Dynamics Processor for limiting duties. Surprisingly, Geoff loves and trusts this box so much that he rarely has cause to adjust it beyond a simple preset. A new addition to the room is an amazing vinyl cutting lathe. Yes, with
“People think it’s got to be loud before it gets to me, but I make it loud – with a fourgrand limiter!”
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WorldMags.net All in the detail Geoff’s moves and the changes in the sound are becoming less noticeable now. We sit taught, silent and… “I think that’s it!” booms Geoff and suddenly the music stops. We all breathe again and gather at the desk for a closer look. “There, press that,” instructs Geoff. He motions to a small, illuminated square button at the west-most extremity of the desk. “It defeats the EQ so you can hear the before and after.” The track starts again. From here, up at the desk the sound is full, dense, swamped even and the track is cranked enough to reveal detail we’d not heard before. This track needs work. We tentatively stab the button and… boom. Instantly the bottom end drops away revealing a much more discernible middle and delicate, brighter, less brittle top. We hit the button again, undoing Geoff’s work and once again we’re hit with a hidden bass note so intrusive as to act like an annoying drone. How had we not noticed this earlier? “It’s night and day, but subtle,” observes Geoff. “It’s evened out, sonically more detailed, even at low volume. It’s more controlled and… easier.” So what exactly are we listening to? “Well there are two set frequencies that I’m cutting,” says Geoff. “There’s 150Hz, which is high low end, and 90Hz which is above the kick drum, in the low low end. I’m subtracting at both and adding a little bit of top end to sweeten it. “The strangest bit of the original is at the end,” Geoff comments. “When the final synth part comes in it just joins the track. It’s a huge sound and there’s no room for it. I’ve programmed a gain change so when the synth kicks in, the track comes down 2.5dB. It’s a fade over 10 seconds so you don’t notice it.” He’s right: we don’t. The track seems at once louder, yet dynamic, with a renewed clarity to the middle and a new shiny top end. And no more drone. All in all, some sterling work and arguably the most professional, calculated and precise time we’ve ever spent in a studio.
You won’t find Abbey Road Studios’ bespoke EMI TG12410 mixing desks in a home studio
What was done: step by step
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Your Studio Mastering with Jono Buchanan
Jono Buchanan Jono Buchanan is a producer, composer and songwriter. He has produced remixes for Whitney Houston, LeAnn Rimes and Shania Twain and has written for TV and film around the world. He’s recently completed albums for Professor Penguin and Annakin and lectures at London’s Guildhall School of Music & Drama.
Jono is the Logic master. If belts were issued for Logic skills, Jono’s would be a battle-worn black. Always up for the challenge, Jono jumped at the chance to fabricate a finished master entirely within the box. Although he has recently moved his studio out of the family home, Jono’s setup is still your typical home studio. There’s a Mac, a pair of medium Genelec monitors, a Mackie Big Knob and little else – an ideal scenario for our ‘in the box’ challenge. Time to get busy, and Jono loads the track into Logic. “In terms of the mix it’s very full,” says Jono as he listens. “Without the drums, it’s an ambient track. It’s quite ‘drifty’ but has a four-tothe-floor beat on top. It’s quite woolly in the middle – there are lots of pads going on!” Jono listens closer, spins back and starts again. “There’s no definition at the bottom end and no sparkle at the top. If this is a dance record then I need to make it sound fuller and less reliant on the pads in the middle.” It’s time to leave the master to his tools with the brief that he is only to use Logic’s built-in plugs. We slip away in search of tea. When we return a half hour later things are almost done. It’s sounding bigger and better. “The first stage was to EQ the track, which I did with Logic’s Channel EQ,” explains Jono. “My main goal here was to bring out the top end. The original is bright but there’s no space. After that, I wanted to find the kick frequency and boost that to make it punchier. Finally there’s that dense mid range. I used Channel EQ to scoop that out. You can use the EQ’s Analyser feature to tell you where the track is busiest.”
So with a new, cleaner, less cluttered signal, what came next? “A Multiband Compressor,” says Jono. “I used a 2:1 compression ratio at the bottom end to make it tighter. I dropped the low mid and upper mid in level and boosted the top a little. The use of the Multiband really helped me get to the parts that needed tweaking. “After that, I put the whole thing through a regular Stereo Compressor to pinch it all together. I used a really low ratio to get the whole track to match its peak, then a long attack and release. “Next came a Stereo Spreader. This is a very dense, middley track and I wanted to make it wider. I limited the range of the Stereo Spreader – just the top end however – from 700Hz to 7kHz. This made the mix that bit shinier and put a bit of life into it,” Jono observes.
Minimum headroom Jono, too, comments on the track’s overpowering synth sound near its conclusion. “It has got a resonant low mid element and it’s all on the left. It’s very peculiar but the settings I’ve made limit it and pull it back into range. That said, I could have done with a bit more headroom to work with here. “Finally, there’s the Logic Limiter set to zero gain in and out. This isn’t a loudness maximiser as such, it’s just there to keep things in check, and to double-check that, I’ve put a level meter in there too – to be sure there’s no clipping.” And the result? Much bigger and more up-front. There’s still a lot of bottom end in there and the top boost and spread makes the track more impressive without being louder.
What was done: step by step
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Conclusion: which do you prefer? Perhaps the most important thing to learn from this exercise is that the process of giving your music to a mastering engineer – or anyone else for a second opinion – always gives you a new perspective. To our ears, all three mastered tracks sound fuller, more defined, more exciting and, interestingly, not necessarily any louder. We were surprised by just how subtle and sympathetic all three of our engineers were. Our conclusion would be that while you can master a track yourself, getting a fresh pair of ears with a fresh perspective is always a good way to go. Are you confident enough to go the whole way, from conception and production to mixing and mastering? Do you really know best? It’s perhaps this element of doubt that’s the strongest reason to get the experts in, rather than any shortcoming of the software and hardware involved. Remember: if nothing else, you’re paying for someone who’s new to your music to tell you exactly where your mix is going right or wrong. And if that person is an engineer at Abbey Road with 20 years of experience, then that makes the process of handing over control that little bit easier.
subjective, after all. The three masters and the original are downloadable and unzippable here: http://bit.ly/apCM0G (that’s a zero by the way). We’ve called the three tracks Lumeria Mastered
1, 2 and 3. All three are 16-bit WAV files – that is, exactly as you would get from each engineer, ready for burning to a CD. Along with the WAVs is a text document that tells you which engineer each track is by, but please, humour us and decide which you like best before you open the text file. You might be surprised…
The artist’s verdict We gave all three masters to the maker. Here are his thoughts… John Andio, aka Cyberoptikon: “I must say that all three versions are highly impressive and much better than the original. Master 1 sounds really nice and thick with a clear high end. Master 2 is definitely cleaner and flatter too. When I raise the bass on playback there’s absolutely no distortion. This sounds good in the latter half of the track where the bass is more important. I like this version as it seems less risky to send to a DJ without fear of rejection! Master 3 really stands out though. The drums are like an earthquake!”
Three of a kind That’s all well and good, but which is best, you ask. Well, we’re going to let you decide – it is
MASTERING AT HOME
7 essential tips for DIY mastering
1
adjustments there before attempting mastering again. Often, for well-mixed tracks, some gentle compression can do a more effective job.
Start with an EQ to prepare frequency changes that will help compensate for the output limiter and then use either a stereo or multiband compressor to address levels. Finally, there’s our loudness maximising. This needs to come last in the chain as this plugin sets the volume ceiling, which you don’t want to risk exceeding. If the mix is solid, these three plugins should be enough.
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It’s too easy to select Master EQ, Mix Compressor and Brickwall Limiter from the respective preset lists of your bulging plugins and assume the mastering stage is finished. Nothing could be further from the truth… You must tailor the plugins you use to the unique qualities of the track you’re mastering.
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Beware multiband compression. Nothing has the potential to mess up a well-crafted mix like this plugin if used incorrectly. Be subtle with your settings and if you are finding you need to be too extreme, go back to your mix project and make
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The louder the volume the better the quality – right? Not necessarily. People have an unfortunate tendency to associate slammed volume with a well-puttogether tune. For dance music, in particular, your record will need to be competitive in terms of loudness maximising if it’s not to be less ‘full of impact’ than other records played in the same set. Again, work your way to loudness carefully by choosing good EQ and compression settings and a maximising treatment that doesn’t just sound loud for the sake of it.
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We all have working habits and know the tricks that work well for us throughout the production process. As you become better at mastering, you’ll learn which plugins and settings to use to get close to a good result for your own music. If you’re asked to master someone else’s work though,
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the slate needs to be clean. This will be a record mixed differently, using different instruments and sounds, so those same tricks won’t necessarily yield the best results. Open your ears and respond to what you can hear in the moment, not what has worked in a different context before.
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Headphones are great for picking out mix and mastering details in particular, but they provide a false sense of acoustics. Many don’t monitor bass well and while it’s a pleasant pat on the back to have your headphones confirm that your mastered track is sound, to trust in them completely isn’t a smart move.
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Finished? Maybe not. Test your mastered track on as many systems as you can lay hands on and compare it to other music in the same style. Invite someone whose ears you trust to hear your new master. If collaboration no longer exists it is easier for poor decisions to survive. Remember: some constructive criticism is always useful. Good luck! COMPUTER MUSIC SPECIAL / 53
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MASTERING FOR WIDTH AND POLISH Mastering isn’t just about keeping levels in check and removing nasties. Breathe life and space into your track with the following tips Regardless of the music you make, the same adjectives can be applied to describe the final result; on the one hand, positive descriptions such as ‘energetic’, ‘rich’ and ‘detailed’ or conversely, the less welcome associations of ‘bloated’, ‘lacking definition’ and ‘dull’. To discover how your tracks can avoid negative connotations and instead bask in the glories of success, read on as we highlight essential advice, not only for correcting your track, but improving it too. With many of us now carrying out our own mastering at home it’s a good idea to have an output channel strip setting ready at hand that you can call upon to act as a starting point. The following walkthrough will introduce us to the 54 / COMPUTER MUSIC SPECIAL
building blocks that will set us on our way to putting one together. It’s all too easy to pay attention to the volume of individual tracks in your mix without keeping a watchful eye on the output channel volume. This can lead to a situation where none of the individual tracks of your mix are overloading and yet their combined volume is forcing the output channel strip into the red. If you then start adding plugins to the output channel strip, each of these will be receiving a distorted signal, so address any overloading issues your mix has before you start thinking about how to add polish via the output channel. The best way to rectify internal distortion is to drag the volume faders of each channel of
your mix down. However, if you’ve used volume automation in your mix, this won’t work as when you press Play the ‘drawn’ volume of the automation will supersede the mix slider position. There are two solutions to this… You can add a Gain or Volume plugin at the end of the plugin chain on each channel of your mix with a negative offset to ensure the output fader doesn’t peak. Or, you can add the same plugin, once, at the top of the output channel to create a global volume offset. As the plugins you’re going to add for polishing are likely to bring up the overall mix volume, look to create a volume starting point that peaks at around -3dB. Then, you can get creative with ways to enhance your final track as it heads into mastering.
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WorldMags.net > Step by step
Start here and build a mastering chain on your output channel
TUTORIAL
FILES
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You work on a track, filling it with sounds, effects and automation, only to find that the combined parts overload the output. Here, no single sound is peaking, but the output channel has been pushed into the red by their combined volumes. This needs fixing.
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After the Gain plugin, we’ve inserted DMG’s Equilibrium EQ into the output channel. To add weight and sheen, boost a little around the fundamental frequency of your kick drum, as well as using a high shelf to add sparkle at the top end. Also, roll out a little sub-bass for some extra clarity.
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Next, we’re looking to get overall dynamics – the range of levels produced – a little. We’re using the FabFilter Pro-C with a ratio of just 1:69:1 and a Threshold setting which picks out the loudest 4dB or so. With no output gain added, this will reduce the overall volume but will control the dynamic of the output nicely.
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Bring the volume of all faders down by 6dB or so, to provide a better mix gain structure. If your track features volume automation, this won’t be possible. In which case, use a Gain or Volume plugin in the output channel to create an offset that leaves your mix at around -3dB.
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It can help to widen the stereo image of your mix, too, as this will add richness. We’re using the UAD Precision K-Stereo to widen ambience and bring up the volume of the mix’s sides. Stereo Spread plugins will help you achieve something similar.
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Finally, we add an output Limiter to allow us to push the overall mix volume up. We’re using the Sonnox Limiter, adding 6dB of Gain to ensure a loud mix. After the limiter has been added, it’s often a good idea to return to the EQ setting and make adjustments.
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WorldMags.net Building width Only got two speakers to play with? So be it, but with careful use of reverb, pan and your imagination, you can create some stunning widescreen mix effects. We know by now that the key to successful mixing is contrast. To hear loud things, you need to have also heard quiet things. The relationships between far and near and dull and bright are also crucial, as is that between left and right. It’s all about light and shade… This isn’t only true in a ‘music mix’ context, however. In fact some sound design sets out to achieve such extreme versions of these relationships that mixes start messing with your head.
> Step by step
We’ve all heard moments where we’ve been tricked into thinking that a sound is behind is, or further left or right than our speakers are placed. It’s great fun to try to extend the stereo field so wide that you’re pushing at the perceived limits of that width. In the following walkthrough, we’ve created a music and sound design scene where a battle is happening behind a saloon bar. An armed guy is hiding in the bar and someone, similarly dangerous, comes looking for him. Having listened to it, import the final WAV into your DAW and complete it – you can decide who kills who in the inevitable shoot-out!
A professional mastering house employs gear designed to widen your mixes
Mix tricks to give your track extra space and depth
TUTORIAL
FILES
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The first step is to provide a musical foundation upon which our sound effects will rest. We use Omnisphere and a program that combines a shimmering nylon guitar sound with a low pad. To this we add a 3.6s reverb with a 67/33% dry/ wet balance. Volume is kept low.
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As he arrives at the saloon door, he knocks hard. Against the distant, more muted sounds we’ve heard, a hard, loud, dry door knock, panned +40 (right), comes as a shock. After hearing nothing after his first knock, he knocks louder. Automate volume to make the second knock more purposeful.
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We import a ‘battleground’ fight scene, which is naturally bright. This is an ‘outside’ texture with low volume and the same reverb as the musical background, but with a lower high cut level. We use an EQ to roll off frequencies above 1.38kHz using a 12dB/octave slope. Volume is -22.5dB.
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The response from the guy hiding inside is to load his gun, panned left to -50 to create a wide panorama. The knocking effect also causes a fly to buzz, and we use the GRM Doppler effect on this sound to inject some side-to-side and front-to-back movement in the centre of the mix.
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In our scene, either a hero or villain (you choose!) breaks from the fight to pursue a guy who’s hiding in a saloon bar. We use footsteps on dirt to signal his approach ‘around the outside’ using volume, pan and a filter cutoff rise to signal the distance and direction of our pursuant’s approach.
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We add a dry barn door open effect after the second knock and, to create a heightened sense of excitement, we add some fast breathing. This works well kept very dry and close to the listening position, but needs to be panned as in Step 5, at -50 (left) to attribute it to the correct character.
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Increasing stereo spread when mastering
TUTORIAL
FILES
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To showcase the highs and lows of stereo spreading plugins at the mastering stage, we’ve prepared a bass-heavy mix of drums, piano and twin bass parts. The mix is run through a mastering chain of EQ, stereo compressor and loudness maximizer.
Bass frequencies don’t enjoy being spread like this, so we’re moving the Lower Frequency Threshold to 340Hz so that bass and low mids are ignored. We apply a spread amount of 25% at this Lower Threshold so that spread to frequencies above 340Hz is introduced incrementally.
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In the gap between the compressor and the maximizer, we add Logic’s Stereo Spread plugin. By default, this stretches from 20Hz to 20kHz, distributing bands of frequencies hard left and right. The overall impression is wider, but the mix is completely compromised and the bass end suffers.
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The top end isn’t suffering as much as the low end, but it is too dispersed now that the bass is controlled. We drop the upper Threshold to 8.1kHz and the Amount to 45%. This produces a wider mix than we had back in Step 1, but without over-distribution of frequency content.
Auto width In synthesis terms, tremolo relates to the process of patching an LFO into the amplifier section of a synthesizer, to provide fluctuations in volume. Guitar tremolo pedals adopt the same approach to create ‘undulating’ phrases. Tremolo plugins within DAWs provide similar functionality, but rather than creating volume rises and falls, their movement tends to be from right to left, creating auto-panning effects. These will nearly always allow you to clock the ‘speed’ of their movement to the tempo of your session and provide control over the rate at which they’ll send sounds from one side to the other. Their ‘depth’ controls decide how wide the left-to-right movement will seem. The ‘shape’
A simple panning or tremolo plug can automate the process to give life and breadth to your track
of movement can usually be determined, too, with ramps moving from left to right through the middle, while ‘square waves’, jumping from side to side, avoid the middle.
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So if you track is lacking a certain spatial something, try injecting it with a bit of stereo width courtesy of a tremolo effect prior to mastering - enjoy the results!
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Essential dos and don’ts DON’T
Danger zone: highs Sheen is one thing, but a saccharine bag of air is something else. The best time to judge whether you’ve overcooked the top end is first thing in the morning. That first playback of the day will help you analyse your whole mix and shine a light on your treble levels. If you burnt the midnight oil, don’t be surprised to discover that the mix sounds too hyped at the top end – this will be a natural product of craving more energy through the process of working up your mix. So long as you’ve restricted your major treble moves to an EQ in the output channel, it shouldn’t be too hard to bring the levels back down to earth.
Become seduced by too much treble. As the day wears on and your ears grow tired, they’ll crave the fresh excitement of extra fizz and sparkle at the top end. It’s easy to just keep on nudging up the upper frequencies as you search for the perfect mastered track brimming with excitement, but be aware the finished result might not sound so great!
DO
Try out your mix and your master on different systems. Be as familiar as possible (and as happy as possible) with your mix on every size of system available to you before moving over to the mastering stage. Then, once mastered, be sure to try that out on a few systems to make sure you’re really happy before unleashing it into the wider world.
DON’T Danger zone: lows Many mixes are ruined by an over-saturation of unnecessary sub-bass. Unless your mixes are heading for the sound systems of your local night spot, you won’t need any of the frequency content in your mix below 40-50Hz. Your speakers have to respond to the frequencies you send them; so, if they’re busy trying to get their cones around the massive low frequency information your mix is producing, they’ll use this as a foundation for more audible frequencies higher up, offsetting the mix horribly in the process. Scoop out unneeded bass from individual sounds and, with an EQ in the output channel, clarity will quickly return.
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Don’t start work on the definitive master of your track at 2am after an intensive 14-hour work day. Go away, sleep, wake up, freshen up, then listen to it again. The objectivity of a fresh listen the following morning is often telling and always invaluable.
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> make music now /
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Master Channel in Reason
Mastering EQ
Master Compressor
1 The Mastering EQ features a low
2 Record users, on the other hand,
cut that we often use. When engaged, it can tidy up a lot of the very low subs (below 30Hz), allowing the bassline some space to breathe. Try this for size: close your eyes and flick the Mastering EQ on, then off a few times. If your track sounds punchier and clearer (subs often overdrive limiters such as Reason’s Maximizer device) and your bassline cuts through the mix better with it on, it’s probably worth keeping it that way!
have arguably one of the finest mix channel tools you’re ever likely to come across: a faithfully recreated SSL 9000k master buss compressor. Countless engineers the world over swear by its ability to pull together a mix. Our settings of choice are Attack at 10ms, Release on Auto, Ratio on 4 and Make-Up on 0. We then start playback with the Threshold at 12 o’clock and dial it clockwise until we’re getting peaks of about 4dB on the compressor’s meter.
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The perfect master channel Build yourself a great start-point master chain in your choice of DAW 60 / COMPUTER MUSIC SPECIAL
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/ make music now <
WorldMags.net Master Channel in Logic 1
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Channel EQ
1 Firstly, you’ve got the Channel EQ. Use this to tweak settings subtly. If you’ve got radical cuts and boosts here, something’s wrong in your mix. Compressor
2 This will enable you to add some extremely subtle stereo compression – just pull in the peaks with high threshold and low ratio settings. Multipressor
3 This is another tool that has subtle
settings, perfect for warming up the bottom end and making the top end just that little bit shinier. Spreader
4 The Spreader plugin is used to subtly enhance the stereo image of your mix. It’s a great effect, but be careful not to push it too far. Adaptive Limiter
5 Lastly, the Adaptive Limiter is used to prevent mix overload and to push the mix a little harder.
By now, you’re probably wise to the fact that there’s no real ‘right’ way of performing a perfect master. All you really need is a few basic pointers to make a difference and get your brain and ears working in the right direction. Here we show you the ins and outs of some classic mastering chains of effects for the popular
DAWs. We’re using nothing but built-in plugs here to make things super-easy to copy. Simply place the selections shown (in the DAWs shown) on the final output channel of the DAW and hear the effect for yourself. You’re then just a few tweaks away from your first basic mastered track!
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WorldMags.net Master Channel in Cubase Studio EQ
1 Let’s start with general tonal shaping. The most important thing here is to make sure that there are no super-low frequencies present. Use a high-pass filter to remove anything below about 30Hz. Too much above 10kHz can be fatiguing. Compressor
2 Set the compressor to a ratio of 2:1 with
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medium/fast Attack, moderate Release and 4dB compression. That should give some subtle dynamic shaping. VintageCompressor can shape the sound in a more appealing way, although if you want the ultimate control, MultiBand Compressor is probably the best choice. Envelope Shaper
3 If the track needs more extreme transient shaping you could explore EnvelopeShaper. Limiter
4 These settings can then be rounded off with the Limiter to increase loudness (or Maximizer with Soft Clip engaged). UV22HR
5 Finally, use UV22HR (in a post-fader slot) and Multiscope to get more detailed visual feedback.
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Master Channel in Ableton Live EQ Three
1 Try an EQ with the mid band switched off as an effect. This quickly gives you control and is handy for removing problem bottom-end rumble. Or how about sweetening the top end with a little gain on the high band?
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Auto Filter
2 Slap this DJ-style filter on your master for quick and easy filtering effects. Master Slicer
3 If you want to get creative, why not put a Beat Repeat into the line so you can create stutter effects on the fly? Limiter
4 As is standard practice, use a limiter as the last insert in the chain to avoid any nasty digital clipping and distortion.
Master Channel in Pro Tools Seven-band EQ
1 Adjust the overall tone of your mix using a basic
1
equaliser with five fully parametric bands and filters. Make sure you keep the low end under control.
AIR Enhancer
2 Add harmonics in the upper frequencies to add sparkle, or beef up the low end with some subharmonic content.
AIR Stereo Width
3 Control the stereo spread of your mix, but be careful – your brain gets used to phase-incoherent signals very easily.
Maximise
4 Maximise the level of your mix by setting the threshold to affect all the peaks. Avoid destroying the dynamics, though.
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Phase Scope
5 Check for phase coherence of your mix and run Peak and RMS metering on the levels.
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UNDERSTANDING
DIGITAL Your final mastered track is, at its core, just a bunch of ones and noughts. Understanding just what’s going on in a final master recording will help you get the best possible sound
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WorldMags.net To use digital music equipment, we need to convert an analogue signal (such as that produced by a microphone, synth or electric guitar) into a digital signal. More specifically, right now we’re interested in turning the multitude of live instruments, voices, synths and plugins into a single, cohesive stereo track. It’s probably fair to say that most equipment in modern studios operates in the digital domain to some extent, whether it be hard-disk based recording setups, computer-based DAWs and all but the purest analogue synths or outboard effects. As such, the messy process of analogueto-digital conversion is necessary for most of us. Although it would be possible to build a fantastic studio consisting solely of analogue equipment without using any digital signals, the practical disadvantages of doing so would seriously outweigh any theoretical advantages. This process, whereby an analogue signal is represented digitally by a regular series of quantised samples, is known as pulse-code modulation (PCM) and dates back to the late 1930s and experiments by the English inventor Alec Reeves, which led to the creation of digital telephony systems. Rather than record an exact value for the amplitude at each sampling point, the PCM process quantises the exact (continuous) value to the nearest whole (discrete) value. We’ll go into more detail about this process when we discuss quantisation and sampling resolution in depth later.
Sampling rate When we refer to sampling in this context we don’t mean what you do with a sampler, such as sampling a loop. The data sampling process refers to the way in which measurements of the amplitude of an analogue signal are taken at regular (evenly spaced) intervals. These measurements are referred to as samples. The
Sampling points
This image shows how the amplitude of the continuous analogue signal (grey) is sampled at each of the regularly spaced sampling points (represented by
frequency of the samples is known as the sampling rate and measured in Hertz (times per second). To achieve an acceptable level of accuracy in the sampled signal, this sampling rate needs to be in the order of thousands of samples per second. The higher the sampling rate, the more accurately the digital signal represents the analogue source. The 44.1kHz standard sampling rate used by audio CDs equates to 44,100 samples per
Amplitude quantisation
In order to create a digital PCM signal, the exact amplitudes must be quantised to the nearest discrete value, as above. The horizontal dotted lines represent a
simplified example (this would be a very low-resolution signal!). The digital signal must follow the dotted lines and so is less accurate following quantisation.
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the vertical dotted lines). The resulting digital representation (red) is what will become the digital audio signal after the quantisation process.
second, or, one reading being taken roughly every 0.00002268 seconds or 22.68 microseconds. For the record, the 44.1kHz rate itself was adopted from PCM adaptors – largely forgotten pieces of equipment that allowed PCM digital audio to be recorded and stored on video cassettes. This process pre-dated the DAT format by a decade. Much of the theoretical basis for our understanding of sampling was developed by two men. Harry Nyquist, an electrical engineer and physicist, worked at the US telecommunications giants AT&T and Bell from 1917 to 1954 and laid the groundwork for later research. Claude Shannon, a mathematician and electrical engineer, worked primarily in the late 1930s and 1940s and was instrumental in developing the field of information theory. Together, their work defines and explains much of what we now understand about digital signals and, perhaps of most interest to musicians and producers, the importance of sampling rate. For audio purposes, the two most important aspects of Nyquist and Shannon’s work are probably the concepts of Nyquist rate and Nyquist frequency. The Nyquist frequency is defined as the upper frequency limit that may be accurately represented by a set of data, and is equal to half the sampling rate. Conversely, the Nyquist rate is the lowest sampling rate which may be used to accurately represent data of a given bandwidth. When applied to audio, these theories have helped to determine the standard sampling rates now in use. The Red Book audio CD standard’s 44.1kHz sampling rate, for example, allows the digital audio to reproduce accurately frequencies up to 22.05kHz, exceeding the widely accepted 20kHz upper frequency limit of the human ear. Listening to the audio examples included, you’ll hear clearly that lower sampling rates severely affect high-frequency response. COMPUTER MUSIC SPECIAL / 65
WorldMags.net Clocking explained Clocking is the central timing that binds our digital recording world together. It’s either not a concern at all as your system and gear are simple and reliable enough to co-operate perfectly, or hugely important (for instance when mastering) as a multitude of digital gear and signals all try to combine together in a single whole with occasionally bizarre and detrimental results ranging from glitchy, imprecise audio to flanging effects and dropouts. The sound quality of something as simple as a regular consumer CD player or as complex as a huge digital mixing desk relies entirely on its clock signal. Even the tiniest inaccuracy in the clock signal will have some influence on the resulting sound. It may only be a subtle effect, but it’s an unnecessary and avoidable inaccuracy being introduced into the signal chain. Of course, if you have multiple devices running, the timing becomes even more crucial.
Resolution, dynamic range and SNR The resolution (also known as bit depth) of a digital audio signal is determined by the accuracy of the sampling method itself. The information stored in a PCM stream cannot be perfectly accurate for practical reasons. If a continuous signal is sampled with perfect accuracy, the amplitude at each sampling point could be any value between the minimum and maximum. To record the exact value at each of these sampling points would require a relatively large amount of data to be generated for each sample. To avoid this problem, each sample is quantised, reducing the amount of information. Quantisation in this sense (which differs to the kind of quantisation you’ll be familiar with from your DAW) refers to the way in which the exact value of the amplitude measurement is rounded off to the nearest of a set of pre-defined values. The amount of data to be stored is reduced at the expense of some accuracy. The bit depth defines the number of bits of information, which refer to each sample’s amplitude. If the audio is recorded at 8-bit resolution (ie using all binary numbers with eight digits) then the total number of values is 28 or 256. This means that the maximum possible scale of the analogue wave (from the lowest possible magnitude to the highest) is split into 256 discrete values and each sample measurement is quantised to the nearest value. The resolution of the sampled audio is the main influence on the difference in dynamic range and signal-to-noise ratio (SNR) between the analogue audio source and the resulting digital audio. that is to say, how much the dynamic range and SNR are damaged by the analogue to digital conversion process. Assuming the existence of a perfectly efficient ADC (analogue-to-digital converter) ie one that does not introduce any outside noise to the signal, the dynamic range of a digital audio signal, ie the difference between the loudest and 66 / COMPUTER MUSIC SPECIAL
By connecting all your digital equipment to just one clock signal, you can minimise the problems arising from out-of-sync clocks. The benefits of a good-quality master clock are obvious: with a perfectly synchronised, extremely accurate clock signal being sent to all your equipment, all the potential problems of inaccurate, unsynchronised clocks will be avoided. So is there a downside to an external master clock? Just one: the price. A decent master clock will set you back a pretty major chunk of cash when compared to your other gear. For even a very basic clock like the Black Lion Audio MicroClock you’ll need to spend a few hundred pounds. Once you get to topend products like, Apogee’s Big Ben, you’re looking at well over a thousand. This is a substantial investment for the home user and it’s not something you’re likely to consider until you’ve got the rest of your studio set up to a high standard.
quietest sounds) is determined by its bit depth according to the formula SNR (in dB) = 1.761 + (6.0206 x Q) where Q is the bit depth. In the case of a 16-bit digital signal, this equates to SNR = 1.761 + (6.0206 x 16) or a maximum signal-tonoise ratio of just over 98dB. The same formula gives theoretical dynamic ranges for 24-bit and 8-bit signals of 146dB and 50dB respectively. Take a listen to the examples included to hear how different resolutions affect the sound.
The weaknesses of digital The strengths of digital signals are numerous. At least in theory, digital facilitates copying, storage transmission without any loss of fidelity or degradation over time. However, digital does have certain weaknesses which we should be aware of… Although there’s a widely held misconception that digital audio is fully efficient in capturing the detail of any recording, this is clearly far from the case when we observe the way in which the analogue-to-digital conversion process discards information, reducing it to a less accurate digital waveform. As a result, the process leads to an inevitable reduction in fidelity, which may or may not exhibit itself as a perceivable reduction in sound quality. In the case of music recording and production, the aim is to minimise the degradation of the signal to the point where it becomes negligible. In terms of sampling variables, this simply means that we need to keep the sampling rate and resolution high enough to avoid sound quality deterioration. Digital audio is prone to a phenomenon known as ‘aliasing’, which occurs when, as Nyquist observed, digital formats fail to represent frequencies higher than half their sampling rate accurately. The effect is that erroneous elements may be introduced to the signal, usually in the form of distortion or rogue sounds known as artifacts. The solution to this problem is to process the analogue audio signal through a low-pass filter prior to sampling.
A Big Ben master clock will sort out your clocking issues, if you’ve got a grand or so…
Digital to analogue Although we’ve focused on sampling and the analogue-to-digital conversion process, the opposite process, digital-to-analogue conversion, is equally as important and the majority of issues surrounding sampling and ADCs are equally related to DACs (digital-toanalogue converters). Fortunately for us, the majority of equipment carries out the ADC/DAC process apparently seamlessly and without any fuss. However, very few digital processes are as perfect as they should be and anything from a volume change to the errors introduced by a less-than-perfect clock signal can cause deterioration of the sound.
Digital studio equipment If the de facto standard set by the Red Book audio CD format is 44.1kHz sampling rate and 16-bit resolution, why does most professional audio equipment exceed this? The old adage that you can always remove more information but never replace what you’ve lost applies here. The superior resolution, dynamic range and SNR of a 24- or even 32-bit signal is clearly advantageous to maintaining higher overall sound quality, regardless of the intended resolution of the end product. Sampling at a higher rate than strictly necessary is known as oversampling and helps to avoid aliasing problems by raising the Nyquist frequency. For this reason, digital synths and outboard equipment often operate at high sampling rates during internal digital processing stages, although it’s also worth noting that some software is limited in its ability to work with higher sampling rates and resolutions. Although this is only a brief introduction to the complex world of digital audio, a basic understanding of the processes involved in A/D conversion should be an essential tool in any modern studio and can only serve to help you make your music better.
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&Loudness
psychoacoustics Can psychological factors affect the way we interpret sounds? Is the pursuit of loudness just an illusion? It’s time to get scientific…
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COMPUTER MUSIC SPECIAL / 69
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Mastering sub-bass Sub-bass presents its own mastering considerations. A lot of the techniques you’d normally use on a bass part become redundant on most sub-bass parts. A sine wave doesn’t respond to EQ and filters in the same way as a harmonically rich signal because there are no harmonics other than the fundamental frequency. EQing a sine wave can only act as a frequencydependent gain control. Likewise, applying a lowpass filter to a sine subbass will simply progressively reduce the volume of higher notes. A lot of techniques commonly used to try and ‘thicken up’ sub-bass are also counterproductive. If we add distortion or induce harmonics using saturation and warming effects, the mix can easily become messy and hard to balance as harmonics clash. We’d then need to EQ out a space in the frequency range, removing harmonics and taking us back in the direction we came from. Rather than trying to thicken up the sound or use dramatic harmonically rich effects, it’s generally best to add interest to sub-bass parts by modulating volume, glide or oscillator detuning. Mix rescue techniques such as bass maximisers, sub enhancers and even multiband compression can be troublesome. Many add harmonics to the bottom end in
Classic machines such as Roland’s TR-808 drum machine are famous for the huge amount of sub-bass they kick out
order to make it cut through, while multiband effects are often used to paper over cracks of weak mixes.
How low can you go? Amid all this obsession with deep bass, it’s important to remember that there is a point where subbass becomes too low to be effective. The lower limit of the human hearing range is typically around 20Hz, but sound systems don’t necessarily go this low. Most tend to struggle with bass
Although you may not be aware of it, the way we perceive sounds is strongly affected by a series of psychological effects known as psychoacoustics. It is vital that we have an understanding of exactly what we’re hearing and the effects of the changes we make before we push things too far, particularly at the mastering stage. Our hearing is not as straightforward as you might imagine and a number of psychoacoustic phenomena can dramatically affect the way we respond to the sounds we hear. Let’s take a look at three very important characteristics of audio – frequency, volume and timing – in order to examine some of the ways the human ear responds to sound and how our perception can be affected by subtle variables.
Equal loudness In the world of mastering, loudness is critical, 70 / COMPUTER MUSIC SPECIAL
below 30-40Hz. As such, adding low frequency content around this range might not be a good idea. The best way to figure out the ideal frequency range for your sub is to test it on a variety of systems, but this isn’t always possible. To save you the time and effort, a good rule of thumb is that 40Hz and upwards tends to be the sweet spot. Thinking in musical terms rather than frequencies, that means your lowest sub-bass notes are probably going to be hitting
but the way we perceive the loudness of sounds can vary wildly. Our ear’s frequency response is not linear, meaning that we perceive some sounds to be louder than others despite the fact that they may measure at the same sound pressure level. The audible frequency range is usually approximated to 20Hz – 20kHz, although this varies on an individual basis, but the ear is most sensitive to sounds around 2-4kHz. Check out the diagram (page 75) of an equal loudness curve for an example of how it takes significant increases in sound pressure level to make low and high frequencies appear as loud as these mid-range frequencies. Note also that 2-4kHz is the typical frequency range of the human speaking voice. It’s also the typical frequency of a baby’s cry – the ear is naturally biased towards the sounds that we need to hear.
somewhere around E1 (41.2Hz) or possibly Eb1 (38.9Hz). It’s also advisable to keep lower frequencies in mono. Likewise, applying delay or reverb effects on sub-bass can create a real mess. If you do decide to use stereo effects on a sub-bass part, check mono compatibility carefully. As an alternative, mid-side processing will ensure your track’s bottom end doesn’t collapse when it gets summed to mono. Keep it simple and you won’t go far wrong.
One important point to make here is that your perception of the loudness of different frequencies changes significantly at different levels. At an lower listening volume, low and high frequencies appear to be much quieter relative to the midrange. Anyone with an older hi-fi will be familiar with the now unfashionable ‘loudness’ button, which boosted the low and high frequencies at lower volumes in an attempt to compensate for this discrepancy. This phenomenon helps to explain why your mixes can sound significantly different when you turn the volume up and down. One of the most common mistakes made when mixing is to monitor at a high level and push all the midrange elements up; this leaves them sounding unbalanced at lower levels. The classic solution to this is to mix at a low volume, especially when checking the relative levels of vocals and other crucial midrange elements.
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WorldMags.net Louder is better, right? Scientific research has shown that listeners consistently perceive louder versions of a recording to sound subjectively ‘better’ than quieter versions of exactly the same waveform. The implications of this research to musicians are significant. In a music production and mixing context, the temptation to push the volume up in order to make the track sound ‘better’ is powerful, but the effects are highly undesirable, massively affecting our subjective response to sounds. Our sense of pitch is seriously affected by volume. The effects are more extreme at the limits of our hearing range, making low notes sound lower when played louder and high notes sound higher. This can be problematic for instrument tuning and vocal pitching. As a result, making critical judgements at higher volumes is best avoided. However, the most significant consequence of loudness for most of us will come before the final mixing process, and relates to production and sound design. When synthesising sounds, choosing samples and applying effects processors to a track, we often make comparisons between two versions of a track at different levels, and this is where the subjective ‘louder is better’ effect can seriously warp our judgement. A number of effects – from compressors with incorrectly adjusted make-up gain to overdrive plugins – impart a volume change to your signal and it’s all too easy to trick yourself into believing an effect is improving the sound simply because it’s boosting the volume. When adding effects, careful balancing of gain settings and use of the bypass button are highly recommended to avoid tricking your ears into thinking a louder version is better.
Careful use of today’s visual tools and notch EQs can reveal then eliminate problem frequencies
Auditory masking and missing fundamentals Two more interesting psychoacoustic principles are useful to understand when engaging in critical mixing and mastering. The first is known as auditory masking and refers to the way in which one sound can affect our perception of another. In pronounced cases, one sound can be completely hidden by the other when the two are played simultaneously (or near simultaneously). The MP3 format takes advantage of this by discarding masked elements of a recording in order to reduce bandwidth and file size. In a studio context, some degree of masking can occur whenever we have two instruments playing in similar parts of the frequency range. For this reason, it’s wise to give each instrument its own space in the frequency range, either through careful EQing or by addressing the problem at its source and having each instrument play notes in different ranges. This is often as simple as shifting a troublesome part up or down an octave. A contrasting phenomenon known as missing fundamentals is also common, with the ears and brain filling in the gaps where components of a sound are expected but not present. Take for example a sawtooth wave, which contains all the integer harmonics of the fundamental frequency. Even when the fundamental frequency is not present, our ears still perceive the pitch of the note to be at that fundamental, filling in the gaps where they believe the missing fundamental (or phantom fundamental) should be. This is useful for giving the impression of lower bass notes without overlapping with your kick drum and using up vital mix headroom. Harmonically rich bass sounds such as sawtooth and square waves allow us to roll off the bass frequencies and let our ears do the rest of the work for us.
However, there’s a strong argument in favour of checking final mix EQ or master at a higher level if you’re making music that will primarily be listened to at high volume in clubs. Either way, picking an approach and sticking to it is probably the best bet for consistent mixes.
Out of range It’s worth considering the effect of frequencies outside the audible frequency range here, too. We may not be able to hear subsonic or hypersonic frequencies, but they can’t be disregarded. In particular, the weight imparted by subsonic frequencies plays a major part in how we feel music. To get an idea of how this works, try taking a heavy track with lots of sub-bass that you know well. Listen to it with the whole track running through a steep high-pass filter at around 80Hz, then gradually reduce the filter frequency. You’ll
A hardware spectrum analyser reveals every element – even those you can’t hear
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Compressing sub-bass
In the classic stereo field a sound can be placed exactly. With sub-bass, however, it’s much harder to position
Compression can often have unexpected results on sub bass. Rather than simply controlling the dynamics of your sub, it’s common for a compressor to colour the signal and add harmonic content. The wave-shaping effects of compressors are often overlooked, but at low frequencies they become even more important. Feed a low-frequency sine wave into a fast compressor with short Attack and Release times and you’ll hear a dramatic change in the tone of the signal. Check out our audio clips and the graph above to see how dramatically a 40Hz sine wave can be affected by compression. This particular effect occurs when a compressor is set fast enough to reduce the peak of a wave before releasing in time for the next peak, chopping off the top of the waveform as it goes. Even with the threshold and ratio set so that the compressor only hits a couple of decibels of gain reduction, the effect on the sound is noticeable. The sub itself might end up sounding thicker and warmer or cutting through the mix more, but the extra harmonics that result from squaring off the wave could clash with the kick drum and bass. It’s easy to get caught up in a cycle of adding harmonics to the sub, then EQing them out and trying to cut it back to avoid clashes.
notice as you reduce the frequency that there comes a point where all the fundamental frequencies are present, but the track lacks weight and doesn’t feel like it physically hits you as much as before. Although most of us can’t hear sound above 20kHz (and many of us struggle from as low as 14-16kHz), studies have suggested that humans perceive music differently when hypersonic frequencies are present. This has also been suggested as one of the weaknesses of the Red Book audio CD format (which can only reproduce frequencies up to the Nyquist frequency of 22.05 kHz). Just because they’re not obviously present, it’s important not to forget these inaudible frequencies.
It’s all about timing The timing of signals massively affects our perception of their position in a 3D space. Since 72 / COMPUTER MUSIC SPECIAL
“We may not be able to hear subsonic frequencies, but they play a major part in how we feel music” sound waves take time to travel from their source to our ears, pushing back the timing of one element (relative to others) gives the illusion that its source is moving further away from us. It’s quite easy to experiment with timing delays in order to get and understanding of this phenomenon. Since sound travels quickly, it
Why do low frequencies sound omnidirectional? Our brain derives positional information – the likely direction, distance and elevation of a sound source – with a complex array of measurements. These are frequency dependent. As sound travels to each of our ears, there are two main variables that our auditory system is able to measure: time differences and sound differences. Even the smallest delay between the ears is enough to triangulate a lateral position. Sound differences are caused by our body mass when the sound travels through the body to reach the eardrums and also by the filtering effect of the pinnae (the outer ears). Such differences vary depending on the frequency content and the position of the sound source and are responsible for elevation information. Frequencies lower than 60Hz can defeat this complex system . Delays between waveforms become less measurable when the wavelength is several metres and elevation data less relevant as the filtering effect of the pinnae and body decrease. When the system is unable to work out a position, our brain will interpret it as a paradox. Both eardrums are stimulated at the same time and from within, and that is our perception of nondirectional lower frequencies.
only takes very small timing differences to hear the effect. The speed at which sound waves travel through air depends on the barometric pressure, temperature and humidity of the environment, but for our purposes here it’s easily approximated to one foot per millisecond. Try adding a delay plugin to one element of a mix, set to 100% wet signal, and gradually increasing the delay time in the order of milliseconds. A 10ms delay should be enough to hear the sound get pushed back behind other elements of your mix. Of course, there’s more to 3D space than timing, most notably left-right sound location. For more on this take a look at our feature, Mastering for width and polish, on page 54. The frequencies of sounds are affected by their environment. The sound of a dog barking half a mile away is not just a delayed version of
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WorldMags.net Phase coherence Keeping your sounds in phase is vital for mastering impact. Follow our quick guide to phasing fixes… During mastering, it’s often desirable to send the signal (be it the whole track or part of it split by frequency) through external hardware for treatment. This can then be mixed back with the original signal to imbue a certain something and beef up the sound. However, when a single source is split into two or more channels and recombined, any change in the relative phase of their waveforms at any particular frequency will reduce their combined amplitude at that frequency. They will be
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phase incoherent. Though this can be an interesting and sometimes pleasant shift at low frequency. It can also be devastating to the power of a mix. Incoherency can come from numerous processes: poorly set up stereo mics, overcooked stereo width adjustments, parallel effects processes, layered sample slices, faulty turntables, the list goes on… There are many manifestations of phase shift. Identifying them and rendering them coherent again involves just as many processes. The walkthrough below includes six ways of tackling phase shift for you to get to grips with
Classic hardware EQ units let you split a finished track into separate frequency bands
Staying in phase
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Learning to identify by ear: it’s almost impossible to describe what phase incoherency sounds like, but at low frequencies it is at its most obvious and destructive so acquaint yourself with what this sounds like. Take a stereo source (eg a pad), phase flip (180°) one side and listen to how it changes. Delaying one side in sample or single millisecond increments will show sub-180° shifts.
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Complex incoherency: sometimes processes in mixing can cause subtle and cumulative phase shifts that are not immediately obvious, such as EQ and filtering. Though linear phase EQ is possible in the digital domain, most introduce a degree of phase shift that varies with frequency, so when two phase-related channels are treated differently they can become irresolvably incoherent.
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Phase analysers: most DAWs and some soundcard utilities feature a phase analyser or vector scope. These are an X-Y oscilloscope-style display of two channels that plot their relative phase positions. They are useful for single sources and full mixes alike. More simple is the phase correlation meter, which is a single bar ranging from +1 through 0 to -1 (0°, 90° and 180° shift, respectively).
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Time (re)alignment solution: often if a multi-channel (stereo or multitracked) source is phase incoherent and cannot be resolved with a simple phase flip (180°) then one or more channels may have to be realigned in time, either by hand or the use of a short delay plugin. Nudging the channels on the DAW Arrange page to align their low frequency waves is usually quickest.
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Looking at the waveform: looking at phase-related channels’ waveforms can often show up phase incoherency, especially at low frequencies where the waves are often larger. If a single source featured in all channels (eg a kick drum) has its waveform starting or crossing zero (the infinity centre line) at different times then it is not phase coherent.
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Phase alignment tools: there are a few tools that will allow a shift of anywhere between 0° and 180°, and thus allow recombining with a phase-related source. Little Labs IBP does just this and has proved very successful, inspiring some less comprehensive freeware versions. These make light work of the problem and are a useful aid for training the brain to identify phase (in)coherency.
WorldMags.net Mono vs stereo For maximum mastering impact, lower frequencies are best reproduced in mono. Vinyl records are cut in mono below 80-90Hz – and sometimes higher than that – due to the restrictions imposed by their technology. If you pack too much bass energy into the groove it’ll simply cause the needle to jump out. Most 5.1 surround sound systems, home theatre and professional alike, are equipped with a bass management system that applies a crossover to each discrete audio channel. The audio is usually filtered around 80-100Hz and the low frequency content from each of the five channels is summed and added to the subwoofer feed. This process has the double advantage of reducing the amount of power required to drive each speaker – satellites can be smaller and cheaper – and providing a single point source for low-end playback. While 80Hz is perhaps too conservative, there is little doubt that 60Hz is the threshold below which sound reproduction ceases to be directional – if our ears cannot tell where the sound is coming from, there’s little or no point in attempting any stereo effect beyond that. On the other hand, the band 100-200Hz can easily lend itself to all sorts of stereo trickery and sometimes throwing in a little mid/side processing can make room in the middle. Two good examples would be a stereo bass sound with a mono kick in the middle, or a wide stereo synth sound with a mono bass. Stereo is just another way to make room for elements that wouldn’t otherwise get on.
Why do low frequencies excite? Our brain is wired to react to danger noises, but while anything loud will attract our attention, low frequencies have a specific ancestral association with natural phenomena such as earthquakes, rockfalls or landslides. Such events have affected humanity since the dawn of time and our brains are primed to instinctively react to them. There are countless medical studies on the effects of low frequencies on the body and it is a known fact that they can stimulate the production of adrenaline.
Subsonic frequencies (below 20Hz) also affect the body in a number of very odd and more physical ways: as the resonance frequency of our internal organs is approached, it’s possible to experience a range of sensations ranging from discomfort, blurred vision, confusion, nausea and even pain. The excitement induced by loud low frequency reproduction is put to good use in club systems, where the overhyped LF content is aimed at stimulating the crowd.
The equal loudness contour shows how much louder low and highs need be in order to sound as loud as mids
the same waveform we’d hear if we were standing next to the dog. For one, it’s quieter, but it also has a different tonal characteristic as a result of the diffusion of the sound over space. Disregarding echoes for a moment, the sound appears to lose some of its high frequencies as it travels through the environment to us. Since high frequencies are diffused, gently rolling off the top end of the delayed track with a low-pass filter can also help to recreate the effect of a sound coming from further away. Echoes are also a crucial factor in the way our ears perceive the sounds we hear. Most natural environments will impart echoes and reverberations on a sound and our ears are surprisingly adept at decoding the acoustic environment from sounds we hear. On an obvious level, a sound recorded in a concert hall will have a significantly different sonic signature
“Rolling off the top end of the delayed track with a low-pass filter can make it sound further away” to one recorded in a small club. The decay time of the reverb and the time delay between the original sound and early reflections are two of the most important characteristics of reverberations, and our hearing uses this information to calculate the size and shape of the 3D space of sounds we hear.
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The right way We’ve only covered a select few psychoacoustic phenomena, but you may still be wondering how on earth you can account for all of them as you mix and master. With so many interacting variables, surely it’s impossible? The good news is that you’re probably already compensating. For example, if you have ears (and you probably do!) you’re already ‘using’ equal loudness contours as you make music. Likewise, most producers almost instinctively compensate for timing discrepancies, subtle shifts in frequency balance and 3D imaging as they work. The biggest practical lesson we can take from this is that consistency is essential. If you’re wondering why your mix sounded great at full volume but the vocals popped out when you turned it down, or why your track lacked 3D depth, psychoacoustics could be the answer. COMPUTER MUSIC SPECIAL / 75
WorldMags.net Maximising your listening environment There are no cheap tricks to bottom end reproduction. Professional rooms are heavily treated to improve the performance of the monitoring system and most acoustic designers will couple the main speaker system’s characteristics to the specific design of the room. Low end is deeply affected by the dimensions of the listening environment. A typical untreated rectangular room will boost the bottom end if you’re listening close to the walls and will reduce the bottom end if you’re standing in the middle. If you’re using a subwoofer, it is usually better to use a crossover to split frequencies between satellites and the sub. Most active subwoofers feature a built-in one, but they might not have a filtered thru-feed for the nearfields they’re coupled to. It is also worth experimenting with sub placement and its phase – it is not uncommon for subs to perform better facing the wall and positioned off-centre. In order to improve your monitoring environment, below are a few tricks to try:
A recording studio is the perfect listening environment, but it’s easy to replicate the same quality at home
Simple tricks from the pros
Corner bass traps and tuned, heavy acoustic panels can effectively help to control and reduce exuberant wave modes caused by the room’s dimensions. They needn’t be pro – simple wood frames will do.
Avoiding parallel walls and ceilings and you should be able to avoid wave mode build-ups.
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Avoid placing speakers against the wall or in the middle of any of the dimensions of the room, as they will excite specific low frequencies dictated by the room size.
Have one or more alternative pairs of speakers set up sideways to the listening position so you have a range of listening options.
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78 / COMPUTER MUSIC SPECIAL
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Headroom and levels explained Navigate the danger zone of mastering with our guide to signal to noise, gain staging, headroom and more If the world of mastering is about making things louder, pushing the envelope and getting everything ‘on the limit’, then it’s clear that in the wrong hands your track could be heading into the danger zone. The world of digital levels is an unforgiving place that, unlike the warm analogue love afforded by the gear from yesteryear, will shred your track and turn it into nonsense if you overstep a single dB mark. So here’s the trick: give your track lots of digital space (known as headroom) to manoeuvre in, build signal chains (known as gain staging), that allow you to add sounds together (known as summing) in a pleasing fashion while keeping noise low. Digital headroom is the breathing space we leave in a track, as opposed to trying to push all our levels as high as possible. But do peak indicators matter? Is it OK to clip channels? How do you set the master output level? We’ll be answering these questions and more. There are two major aspects of digital headroom we need to look at: recording and
“If using summing stages and effects during production, consider their effect on headroom” mixing. Recording is straightforward: you get an analogue signal into the DAW as a digital recording, so for example, you sing into a microphone and it’s recorded in your computer. Next comes mixing, which in this case doesn’t just refer to the final mixdown process after composing and producing your track. Most electronic music producers mix as they go to some extent, so if we’re using summing stages and effects as part of our production technique we also need to consider the effect they’re
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having on headroom and the overall sound. Let’s start off with getting the sound into your DAW in the first place and see exactly how headroom affects recording.
Recording Headroom is important from the moment you start recording signals. To understand digital headroom, let’s first look back to the analogue days. Most of us are now recording digitally, but we can still learn a lot from the analogue recording process. When tracking to tape it’s crucial to set the gain of each channel to maximise the signal-tonoise ratio. If a track is recorded too quietly, it’ll be necessary to increase the gain at a later stage in order to bring it up to a suitable level. Doing so would also increase the volume of the tape’s inherent noise. On a multitrack recording, the cumulative effect of boosting this inherent noise can be severely detrimental to the overall signalto-noise ratio. So, how does this relate to digital recording? Essentially, the key factor to consider when COMPUTER MUSIC SPECIAL / 79
WorldMags.net Signal-tonoise ratio explained All because tape had to be hit as hard as possible to avoid noise interference It’s important to maintain the highest possible signal-to-noise ratio. In simple terms, this means keeping the signal as loud as possible while keeping background noise as quiet as possible. The perfect example is tape hiss. When music was recorded to analogue tape, a gentle hiss in the background of a recording was an inevitable side effect. Recording engineers learned that a certain technique could make it less prominent in the final mix. If we record our tracks to tape at a relatively low level, the first thing we’re likely to do when mixing the recording down is to amplify those recorded signals. But when we amplify playback from the tape, we’re not just boosting the signals we were trying to capture, we’re also boosting the tape hiss. That could add up to a noticeable background noise. The solution is to record those signals at a higher level, meaning they don’t have to be amplified as much. We can then achieve the same overall mix at the same level with a lower level of hiss. The same approach applies no matter what type of audio signal we’re dealing with. Most of us now
Back in the days of tape, hitting it as hard as you could kept signal up and noise low. In the digital world things are rather different
record digitally, so tape hiss probably isn’t an issue, but there are still unavoidable elements of noise that exist in a nominally ‘digital’ recording setup. Ultimately, you might be recording to a digital medium (your DAW and computer’s hard drive) but there are analogue signals along the way – every piece of equipment could introduce noise. The general approach of most recording engineers in the analogue tape era was to hit the tape as hard as possible without overloading it, guaranteeing the signal-to-noise ratio was as high
recording digitally is similar: higher signal levels result in better signal-to-noise ratios, lower quantisation error and more accurate representation of the analogue wave. However, whereas tape compresses, saturates and gradually overloads when the input level exceeds a certain nominal maximum level, digital is much less forgiving. If the input level of the signal exceeds the maximum level of the analogue-to-digital converter, you will find you’ll get clipping. Clipping is the condition that occurs when a digital signal attempts to exceed the maximum level. Since the maximum possible amplitude of a digital waveform is fixed at 1, any signal which attempts to go above this level is truncated (limited) to 1. The waveform is distorted and depending on how badly clipping occurs you’ll hear anything from a short click or pop to a harsh, undesirable digital distortion. Back when 24-bit recording took over from the previous 16-bit standard, many users failed to understand the point of 24-bit recording if 80 / COMPUTER MUSIC SPECIAL
as possible. One principle to consider is starting with a relatively hot signal, making it quieter where needed rather than starting with a quiet signal and amplifying it along with its noise. Whenever we amplify a signal, we’re amplifying every part of it. If we’re amplifying to compensate for a weak signal that could have been boosted earlier in the chain – before the introduction of unwanted background noise – we’re not getting the best signalto-noise ratio. With so much equipment on the market it’s impossible to give a 100% foolproof formula for
“By recording in 24-bit we can achieve the dynamic range and accuracy of 16-bit, and avoid clipping” their tracks were destined for the 16-bit 44.1 kHz format defined by the Red Book audio CD standard. One significant part of the answer is that although 24-bit recording produces larger files and is unlikely to result in major increases in sound quality by itself, it solves the headroom conundrum when recording. By recording in 24-bit, we can achieve the dynamic range and accuracy of 16-bit while keeping the input level low enough to avoid any chance of clipping.
setting input and output levels. Gain staging is a balancing act – it’s about juggling input and output level settings on each piece of equipment until you find what gives the best results. Thankfully, it’s a relatively easy process, but one you should run through before you start recording. Hook up all your gear – whether it’s a synth into an interface or a complex chain of preamps, compressors and effects – start with a loud signal from your source, determine the overall combination of settings that give the cleanest sound and then start the recording process.
Generally speaking, an average level of around -18dBFS with peaks hitting no higher than -6dBFS is a good rule of thumb. Of course, in some cases you’ll be limited by your preamps or the dynamic range of your sound source, but trial and error should help you to determine the best settings.
Mixing There’s one question that we get asked more than any other about headroom when mixing, and that is: do the peak indicators in my DAW matter? It might seem like a simple question, but as we’ll see, the answer isn’t entirely straightforward. In fact, it’s a multi-part answer that depends on exactly which peak indicators we’re talking about. First, we need to break the DAW’s signal path down into chunks. Let’s assume that we have an audio recording or virtual instrument as our sound source. Most DAWs run the audio through a similar signal chain. Firstly, the signal runs through the effect chain on each channel strip
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WorldMags.net Plugins such as PSP Vintage Warmer give your signal a ‘vintage’ gain boost that is pleasing to the ear
“The outputs of your audio interface have a fixed limit: 0dBFS. Go above this and you’ll be clipping the signal” (auxiliary sends carry out essentially the same function but with slightly more complex routing). Secondly, the outputs of each channel get summed to a single (mono, stereo or multichannel) output signal. Finally, this signal is sent through the master fader and to the output, which is either a digital-to-analogue converter or a digital audio file, depending on whether you’re playing back your mix or bouncing it. Let’s take those three stages and consider how each one affects the signal level and the sound.
Mastering It’s best to tackle the three stages in reverse order, mainly because the master channel gives us the easiest answer: do not, under any circumstances, let the master channel peak. Let’s say you’re bouncing to a WAV file. Regardless of bit depth, your output level cannot exceed 0dBFS. Any signal over this level will be clipped and induce harsh digital distortion. Likewise, the outputs of your audio interface have a fixed limit: 0dBFS. Anything above this and you’ll be clipping the signal. It might not be obvious at first, but as you push harder it’s likely to become very clear that something’s not right. Some producers choose to set a brickwall limiter to 0dB at the end of the master channel strip in order to ensure the signal can’t go any higher and avoid the output clipping problem altogether. A much better idea is to leave a few decibels of headroom and then save the limiting for the mastering stage. This ensures that the output doesn’t clip but also doesn’t colour the sound or squash the dynamic range of your track. One notable exception is when playing live – a clean, neutral limiter on the output can be a useful safety net in this case.
To sum up The summing stage of your DAW is where things get slightly more complex and it’s all due to the fact that most DAWs now use 32-bit floating point (as opposed to integer or fixed point) processing for internal calculations. We haven’t got enough space to go into detail on floating point data representation here, but essentially it means that the potential accuracy and resolution of a digital signal is much higher than with the older 16- or 24-bit integer standards. 32-bit floating point effectively means that you can’t exceed the maximum signal level. Back in the days when everyone used analogue
Optimum signal levels Consider every element in your signal chain for the perfect path The temptation is to crank up the volume to get a loud, hot signal, but if you follow that approach you’ll run into trouble. All audio hardware or software is designed with a specific operating range in mind. Let’s take an analogue compressor as an example. If we hit it with a quiet signal, immediately we have a problem. The meters and range of the threshold setting are designed for a signal within a certain range. If the signal is too quiet, it will be hard to monitor its level on the meters. In extreme cases, it may even be impossible to set a low enough threshold to achieve the level of gain reduction we need. Some compressors have an input gain stage, but trying to fix our mistake by boosting the signal at that time also boosts the noise that came with it. The correct approach is to ensure the signal arrives at the compressor at the optimal level in the first place. So what happens when we feed an overly loud signal into a piece of equipment? Every signal processor has a limit beyond which it can no longer operate in the way the designer intended. Above 0dBFS in a digital system we get clipping (a harsh form of distortion). Analogue devices tend to break up more gradually as the signal exceeds the optimal level, but distortion is still introduced. Generally speaking they’re undesirable unless you are deliberately choosing to overdrive the equipment. Once inside the DAW, the rules change. The majority of DAWs and plugins tolerate overly hot signals in most cases. It’s also much less likely that processing will introduce noise. However, we can still run into similar problems. Plugins designed to emulate the operation of analogue gear almost always need signals within a certain range. Effects like tape saturation or overdrive are often designed to break up
more noticeably when hit harder, emulating the way modelled analogue hardware would respond. Running signals outside their intended optimum range means they won’t work the way the developers intended. Likewise, if we run overly loud signals we will eventually hit the master channel. Master output signals above 0dB will clip when they hit the DA converter of our interface. Trying to bounce the mix will lead to similar problems. So why not just pull down the master fader? We could, but that’s a work-around rather than a proper solution. There are still compelling arguments for paying attention to signal levels and gain structure in the box. Even something as simple as a level meter relies upon an optimum signal level, whether that’s on a channel strip or inside a plugin. Most level meters are designed to be at their most accurate between around -20dB and 0dB. Run outside that sweet spot and they become virtually redundant. Let’s go back to the subject of rules changing in DAWs for a moment… Most DAWs and plugins now use floating point processing, making them virtually impossible to clip no matter how loud your signal is, but that shouldn’t be taken to mean all plugins work flawlessly. Even if you are confident that your DAW processes everything cleanly no matter how hard you slam a signal through it, are you 100% certain the same can be said for every thirdparty plugin you use? Probably not, and it will either harm your sound or take time to resolve when you do run into a problem. Whether working in the box or with hardware, maintain a logical approach to gain structure and signal levels. A little attention to these will save you headaches further down the line.
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Both real world and virtual mixing consoles work in the same way – adding effects or boosting EQ will push up the overall level, potentially into dangerous extremes
Digital recording Optimum headroom and the ideal level for a recording signal In the past, recording engineers would try to hit the tape as hard as possible in order to maximise the signal-to-noise ratio and take advantage of the full dynamic range of the tape. Gain staging remains vitally important when recording digitally but, whether we realise it or not, a lot of us are still clinging on to rather outdated ideas of optimum recording levels that have lingered around from the days of analogue tape recording. The same principles still apply in terms of getting a hot enough signal to optimise the signal-to-noise ratio, but if we exceed the limit of an analogue recording device (0dBFS at the analogue-to-digital converter on the way into your DAW), we don’t get the pleasant saturation and compression we would likely hear with tape. Instead, ‘overs’ (peaks above 0dB) are clipped (squared off and distorted). The introduction of 24-bit digital recording means we have more room to manoeuvre. Back when 16-bit recording was the industry 82 / COMPUTER MUSIC SPECIAL
“The ideal level for a recorded signal? As near to 0dB as possible, right? Erm, wrong…” standard, we had a much smaller target to aim at if we wanted to get a signal with a high dynamic range while avoiding the possibility of overs. The additional dynamic range of 24-bit over 16-bit enables us to record at a lower level and to achieve the same signal-tonoise ratio. So, the obvious question would be, what’s the ideal level for a recorded signal? As close
to 0dB as possible for maximum dynamic range, right? Erm, wrong… Not only does that run the risk of overs if you suddenly get even a small peak in the signal, but unless you’re making a song that only contains a couple of audio tracks it will also inevitably mean your master output peaks over 0dB and you will have to turn all your channel faders down anyway to compensate. The dynamic range of the now-ubiquitous 24-bit recording methods means that this is completely unnecessary (in fact, even if you are recording in 16-bit it’s still a bad idea). Recording engineers continue to debate what can be considered the optimum headroom for 24-bit recording, but as a general rule of thumb they would say somewhere in the region of -16dB is a pretty good target to aim at as a starting point. That said, a couple of decibels in each direction won’t make a big difference so don’t get hung up on trying to hit precisely -16dB.
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WorldMags.net The perfect level Getting a good signal into the DAW is the first step for anyone using hardware synths, samplers or drum machines. When recording digitally, it’s important to set input levels in order to achieve optimum dynamic range and avoid the possibility of clipping. Below you will find examples of recording a synth part multiple times at various settings. In many ways, this is the perfect example of gain staging in practice, in analogue and digital contexts. First off, there’s the concept of balancing the synth’s output gain with the input gain of the preamp. Next, it’s the effect of hitting the AD converter at different levels (represented by the input level meter on the DAW channel). Follow these rules to find a suitable input level:
> Step by step
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You don’t want your input levels going over 0dB or you’ll get clipping
Setting levels using a DAW’s metering system
Too high. The most common mistake made when recording digitally is to try hitting a peak level just below 0dB. If peaks exceed 0dB, clipping will occur. Sustained periods of signal above 0dB will sound seriously distorted and harsh.
mixers to sum their sound sources, it was obvious that a flashing peak indicator on any channel meant it was being pushed too hard and the sound was likely to start breaking up soon. Likewise in a 24-bit integer system, there isn’t anywhere higher than maximum signal level so attempting to push a signal higher will lead to clipping. Floating point audio processing eliminates this problem altogether. There is no clipping because there is no maximum – the digital value can represent a signal level from minus infinity up to infinity. As an aside, floating point also has its benefits for low-scale signals since accuracy around zero is much higher than with fixedpoint systems. Dynamic range is practically infinite, so signals can be attenuated until they’re inaudible, then amplified back up to their original level without any signal loss. To all intents and purposes, pulling down the master fader has exactly the same effect as pulling down all of your channel faders (assuming there are no plugins on the master channel strip, of course). So, headroom isn’t really an issue at the summing stage. Does this mean you can just run your signals as hot as you like so long as you pull the master fader down to
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Too low. For obvious reasons, going too far in the opposite direction is just as bad an idea. The signal will have to be amplified after recording in order to bring it up to a suitable level, boosting any noise along with the signal.
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Just right. As a rule of thumb, start by aiming for your peaks to hit somewhere in the region of -16dBFS. This gives plenty of dynamic range and provides ample headroom, so avoiding the risk of clipping.
You can’t go above 0dB with a 24-bit integer system without clipping occurring
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If you want to make some nasty noises, try Logic’s Bitcrusher
Bit crushing and digital clipping: deliberately breaking the rules Now that you’ve learned the principles of clean mixes and digital headroom, you are probably wondering what happens if you break the rules. Just because you know how to make nice, pleasantsounding mixes doesn’t mean you have to, and overloading an input slightly or abusing a digital effect can lead to some seriously aggressive sounds. In general, digital distortion is so harsh and abrasive that it’s best in small doses. Used as an effect on a breakdown or for special FX it can be seriously effective. Of course, you might want to go the whole hog and clip the master output, but don’t blame us if you send your audience running for the exits. We’ve mainly looked at the software world in this article, but a lot of the same principles apply to digital hardware. Most will operate with fixed point internal processing so it’s possible to clip various stages. Some of the filthiest sounds in industrial and techno come from abusing digital gear by overloading inputs and outputs. For a more controllable take on digital distortion, take a look at plugins like Logic’s built-in Bitcrusher. This underrated little plugin offers an excellent digital clipping option entirely independent of the bitcrushing and sample rate reduction features. Three modes allow the waveform to be folded, cut or displaced and the adjustable clip level allows the intensity of the effect to be finetuned or even automated. 84 / COMPUTER MUSIC SPECIAL
Most DAWs use 32- or 48-bit, giving you masses of headroom above 0dB
24-bit recording: a thing of the past Why 24-bit is out-dated and why paying attention to your DAW’s gain and peak levels is always best practice Now that 24-bit recording is industry standard, especially at the mastering stage, we have huge accuracy and flexibility when recording signals into our DAW. The 24-bit standard provides a dynamic range of 144dB, meaning it can accurately reproduce levels ranging from below the threshold of hearing right up to beyond the point where the signal becomes unbearably loud. However, 24-bit is now generally considered insufficient for signal processing. The audio engines found in DAWs tend to use higher resolution fixed point processing or an alternative called floating point processing. In practice, this means that if your DAW uses 32-bit or 48-bit fixed point processing you will have plenty of excess headroom beyond the nominal 0dBFS limit (the total dynamic range of 32-bit fixed is 192dB, allowing up to 44dB headroom above 0dBFS). Floating point processing doesn’t quite deliver infinite headroom as some might suggest. That said, it’s possible to exceed 0dBFS by over 1000dB before you run into trouble, so it’s as good as infinite unless you are doing something very unusual. So why even worry about gain and peak levels? Because it’s best practice. If we’re not using any plugins, the audio engine of our DAW will handle a lot of abuse (those flashing red clipping indicators don’t mean anything in terms of the sound, they’re just visual warnings that a signal exceeds 0dBFS). But given there’s nothing to be gained by running overly hot signals, and the small possibility that one of our plugins might not behave nicely, there’s no point taking chances.
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Modern home studio DAWbased recording allows you to make recordings of extremely high quality
compensate for it afterwards? Well, the answer is: yes, and no… The one potential problem with running hot signals during the mixing process comes from the third stage we identified: plugins. The issue here comes from the fact that not all plugins are created equal. While one effect plugin might happily take a +20dBFS signal and process it in just the same way as it would handle a -10dBFS signal, another might not be quite so predictable. In practice, it’s much easier just to avoid the situation altogether by keeping the signal low.
For the record Recording audio is where the issue of headroom really requires some attention. Too low a signal and you’ll lose your dynamic range. Too high and you’ll ruin the take with pops, clicks and clipping. Following good recording practice and aiming to keep your signal somewhere around -18dBFS with peaks no higher than -6dBFS should ensure that you don’t have to worry about either problem.
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“A signal of around -18dBFS with peaks no higher than -6dBFS should take care of your audio recording” Generally speaking, as long as you make sure not to clip inputs or outputs you won’t run into too many problems. Of course, there are a few exceptions to these rules but the lessons we’ve learned here can be applied to all situations. In some rare cases a plugin won’t use floating point processing and you’ll have to keep the levels down. Likewise, inputs and outputs for effects sends and returns should be treated just like the main stereo outs and mic inputs and be kept below 0dBFS at all times. COMPUTER MUSIC SPECIAL / 85
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THE PRO’S GUIDE TO
MASTERING Louder, harder, bigger, brighter, better: the myths and methods behind a professionally mastered sound
Back in the bad old days of analogue, a finished studio mix would be played through the mixing desk and recorded to two-track (ie, stereo) tape. This master tape would be the last that the mix engineer would see of their hard work before sending it off to be turned into vinyl. A mastering engineer would then transfer the tape to a master disc, from which all the records would be pressed. During this transfer, the engineer may have chosen to apply some further processing to improve the sound of the mix. Perhaps the mix was created using speakers that lacked bass, so leading to a bottom-heavy sound – the 86 / COMPUTER MUSIC SPECIAL
mastering engineer could correct this by applying EQ to the stereo mix. As time went on, these sort of post-mixing corrections and improvements became more commonplace and keen-eared mastering engineers would perform their critical work using extremely accurate speakers, in an acoustically treated room that typically far exceeded the listening conditions of the average mix room. Today’s mastering engineers work in much the same way, though the emphasis these days is on the processing more than the transfer to a physical medium. Typical processes include
applying EQ to balance the track’s frequencies and compression and limiting to ‘glue’ the track and increase its volume. They either apply these effects plugins on the master buss in the DAW, or they render out a track as a WAV file (at least 32-bit) and then load it into a fresh project, placing the plugins on the appropriate track. Given that mastering is your last chance to make your track blast the competition out of the water, is this stage something you should attempt, or is it best left to the pros? We discuss state-of-the-art mastering with three pros for their expert opinion and an insight into what goes on behind the mastering studio’s doors…
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WorldMags.net EXPERT #1
Conor Dalton mastering engineer, Glowcast The in-house expert at Glowcast spells out the basics
What is mastering? “In my experience there seems to be a lot of confusion surrounding the world of mastering, its capabilities and limitations. I’d describe it as the last link in the music production chain, somewhere between a final mix and a completed project, where audio can be optimised through subtle processing, errors checked and all involved made to feel good. It’s the process of creatively enhancing audio before preparing and transferring recorded audio to a device such as a CD or vinyl. It’s not a chance to fix an otherwise bad recording.” What does mastering involve? “It’s your final chance to sonically change your music through techniques such as EQ and compression and the last chance you get to bring out the best of your sound. As a mastering engineer, it’s my job to make all the individual elements of the song gel together, as well as balancing the song tonally with EQ and removing minor flaws such as pops and clicks.
“Mastering is the last link in chain where audio can be optimised through subtle processing” “If your mix has problems, you’re going to struggle to get an great-sounding mastered record. Mastering isn’t the place for fixes, rather it’s the art of balance: audio feng shui. That means that it can never make a bad mix good, but it can certainly help make a good mix great. “When mastering for CD, tracks are also spaced with appropriate distance between them so the whole album flows well artistically and ISRC and UPC/EAN codes can be embedded if necessary. Then the ‘master’ is created, for example a digital WAV or AIFF file, CD or vinyl.” What is a master track used for? “The master is the copy from which all further copies are created, so the goal is for the finished songs to sound great on any sound system, whether that’s a stadium live rig, or a home stereo. You want mastering to be mainly constructive instead of corrective: you want to be enhancing the audio and not spending all your time fixing mistakes that occurred during the mix. “Therefore, it’s essential to get your mix sounding just the way you like it to the best of your ability before mastering. Your job when applying creative processing techniques in the mastering stage is to recognise the point at
which you have successfully pushed the audio to its optimum position before the sound begins to deteriorate through over-processing. It can also include adjusting stereo width, surgical EQ to remove unwanted resonances and limiting, for example. Is it possible to make a track ‘loud’ at the mastering stage? “Yes, but something you must understand is that loudness comes at the expense of dynamic range. Mastering is not just simply making something loud. I get asked a lot if I can make someone’s song as loud as another artist. One of the most common problems I encounter on tracks I get sent as a mastering engineer is overcompression on the mix to achieve that ‘loudness for the sake of loudness’.
“Mastering can never make a bad mix sound good, but it can certainly make a good mix sound great” WorldMags.net
“Loudness is really an illusion, as we all have volume control on our hi-fi or stereo. While it’s often desirable to aim for competitive loudness, it’s not wise to completely kill the dynamic range of your song in order to achieve that. When you reduce the distance between the loudest peak and the quietest sound through compression to achieve loudness, you can achieve a ‘fuller’ sound. However, if you go too far and over-do it, then you’re sacrificing some of the essential and powerful peaks for the benefit of the quieter sounds. Basically, you compromise some sonic integrity to achieve that loudness. Understanding your options and limitations is the essence of mastering.” Can someone master their own music? “Of course they can. However, one of the major benefits of having your music mastered professionally by someone else is that you get a fresh set of ears to provide a new perspective on your music. We all know that feeling you get when you’ve been listening to your track for weeks and you just can’t tell if creative processing changes are required anymore. This is where a fresh perspective can come in very handy: a first impression can reveal a great deal about your audio.” Go to www.glowcast.co.uk for more information on the services Glowcast can provide. COMPUTER MUSIC SPECIAL / 87
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8 mastering myths debunked with Conor Dalton
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engineer will use whatever tools they see fit for the job. That could be digital, analogue or a combination of both. All that matters is that the engineer uses their experience and knowledge, implementing the tools wisely to achieve the desired result and that the engineer knows the characteristics of their tools well enough to choose what process is right for a particular task.”
“Don’t worry about the mix, the mastering guy can fix it.” “No, no, no, no, NO! If you can fix something in the mix, make sure you do. Remember that mastering is the art of compromise. While the mastering engineer can improve sound, balance and correct minor mistakes, if there is an opportunity to fix something early on in the mix, then do so. Remember, the secret to a great-sounding master is a great-sounding mix. A mastering engineer cannot turn a terrible mix into an great-sounding master.”
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“All the mastering guy will do is make your tracks loud.” “Loudness is not the ultimate goal of mastering. Loudness will often be achieved through mastering, but there is much more to mastering than that. Mastering provides an element of quality control where the engineer uses their experience and opinion to decide if a project is ready to be released on the world, or if it could do with subtle processing to enhance the listening experience. Mastering can also include editing fades, surgical EQ, tonal EQ, compression, reducing hum and hiss, adjusting volumes of tracks to match so the whole project feels like a smooth listening experience, ISRC codes and CD text, track spacing and sequencing.”
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“Mastering is a rip off!” “There are some high-end engineers who charge a fortune and quite rightly so as they’re the best in the business and have studios with the best equipment available. There are also a wide range of different mastering studios with varying ranges of experience, equipment, quality of previous work, ability and price. Find one that matches your budget and one you’re happy with.”
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“I can do just as good a job with my plugins.” Don’t look at the plugin to solve your problems. The plugin is just a tool. What you’re paying for when you go to a mastering engineer is their experience and
Of course it helps to have A-grade monitoring when mastering. These industry leading PMC stacks at Optimum Mastering in Bristol show how serious things get
judgement on when and how to use the tools available to them. Good mastering comes down to good decisions and not just having a particular piece of equipment or software. Knowing how and when to use your tools is the essence of mastering. Amazing results are achievable with simple tools if you know what you’re doing. Likewise, you could have the best tools available, but if you don’t know how to use them you won’t produce good results.
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“The louder the better!” “Not true. Loudness is not desirable if you have sacrificed sonic integrity to achieve it. If you smash your songs with compression to make sure they are louder than other tracks, you could be left with music that can feel
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squashed and tiring to the ears. Considering the listener will have a volume control on their stereo, loudness is not the main goal of a mastered piece of music. Maintaining sonic integrity is much more important. Competitive loudness for the genre is what mastering engineers usually aim for, but it’s not true to simply say that a louder song is more desirable than a dynamic song.”
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“You can only achieve a good master by using analogue equipment.” “I, and many others, would strongly disagree. There are a wealth of incredible digital tools available today, but more importantly, it really comes down to who is using the tools rather than the tools themselves. A good mastering
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“You need ‘golden ears’ to be good at mastering.” “This myth refers to the mystery of mastering and the fact that people view it as a black art that can only be conducted by people with incredible, sensitive hearing. While there is truth in the fact that you need to be fully conscious of what your ears are trying to tell your brain, mastering is something you can learn. No one is born with natural abilities to EQ. You learn these techniques through hours and hours of practice. When you become comfortable and confident in your room and in what your ears are trying to tell you, then you can make decisions very quickly and you can immediately tell if something is ‘too bright’ or bass-heavy. Getting to that stage takes time, but you can learn how to become golden-eared with enough practice.”
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“Everything has to be mastered.” “Not strictly true. This depends on whether you’re happy with the sound or not in the first place. Sometimes mixes come in here at Glowcast Audio Mastering that sound near to perfect, and if that’s the case then creatively adjusting the audio with EQ and compression is not helpful. It depends on the client and their aims for the project. If they are delighted with the mix and see no reason or way to improve it, then they might decide to skip mastering and use their version as the final master. It is worth remembering that a fresh perspective from an objective and experienced set of ears should not be underestimated, however.”
WorldMags.net EXPERT #2
Mazen Murad mastering engineer, Metropolis Studio What Mazen Murad doesn’t know about mastering isn’t worth knowing
How did you get started? “I started out recording and mixing,” explains Mazen. “Then, when CD came in, they got me working on a lot of CD remastering work. There was so much work that I ended up getting into mastering full-time.”
Lukas Lyrestam / Vanishing Point Soundcloud.com/Lukaslyrestam We gave one lucky Computer Music Twitter follower the opportunity to have their track mastered by Metropolis’s mastering expert Mazen Murad. We asked followers to submit a track and we chose a track that we thought demonstrated some good examples of common mixing mistakes as well as offered good headroom and potential for mastering. Lukas Lyrestam’s Vanishing Point was the chosen track and you can hear the before and after via this download link:
How do you start the process? “I start by listening to the whole track. You’re balancing all the frequencies. You listen completely differently when mastering. It’s got to sound good wherever you play the track. If the band doesn’t come in, I’ll look at their work and see what they’re about. You can get it all technically right, but some things are a matter of taste. So, the chain I use varies according to what the track needs. Sometimes it’s EQ, compression and a limiter; or if it doesn’t need compression, it would comprise EQ and then limiters. Or, if the mix needs compression before mastering, I would put the compressor first and then drop in the EQ and limiters. And I sometimes use parallel compression or
http://bit.ly/1bUxl50 But what does Lukas reckon to the finished result? “I was very impressed with the mastered version of the track as it retained the mix balance I wanted while having that bit of extra sheen. The bass became warmer, rounder and gelled much better with the kick drum, while the vocals were given body and presence. I was particularly impressed with how well the mix translated onto a club sound system.”
“The chain I use varies according to what the track needs. Sometimes it’s just a matter of taste” multiband to get the job done. It just depends, really. The chain is very important, but you certainly don’t always need a compressor to make tracks loud.” Do clients expect you to work wonders with their tracks every time? “Sometimes you might cut at 20Hz on the master buss if there’s too much information. But that is best done at the mixing stage. If you have a high string with low rumble, cut the rumble on the track. You often get situations where the kick and bass aren’t working together because somebody’s boosted a shelf that’s covering everything. So, if the kick is around 100-120Hz, take a bit of that out of the bass. You don’t need any kick at 300-500Hz, so maybe take a little bit out of that out, too. “Maybe the guitars don’t need boosting at 3kHz – perhaps boost them higher to make space in the 1-4kHz range for the vocal.” Do you get tracks that are already ‘mastered’ reaching you? “It’s important not to slam things at your mixing stage when you’re comparing it to tracks that have been mastered. Doing that will limit what
the mastering stage can add. So, keep your limiting low if you use it. I find people overdoing the limiting happens a lot, especially with dance remixes – they’ll send it turned up full. All I can do is turn it down 5dB or whatever, do whatever processing it needs and then turn it back up. Of course, that’s not great. But then if you mix it with the limiter on and then take it off, your balance is going to change, so maybe send a version with and without.” What’s your top tip? “Trust your ears. Just because you can,
“We can EQ for CD, for vinyl… We know how to trick limiters for radio. We even did tests with the BBC” WorldMags.net
really doesn’t mean you should. Sometimes less is more. I was recording a band for months and I was getting too close to it, so I suggested they bring somebody else in to mix it. Likewise a mastering engineer is in one room and he knows it inside out. We can EQ for CD, for vinyl… We know how to trick limiters for radio. We even did tests with the BBC. That’s one of the best reasons to use a place such as Metropolis for mastering. Even the wiring is the best of the best! We have different clocks for all our digital equipment – some work better on different styles and even a tiny shift can change the whole groove. Nobody else knows how to clock their system – it scares them!” And a pro master is more affordable than ever, right? “People have this impression that it’s superexpensive to get your mastering done by a pro, but our pricing is tiered and caters for all budgets, big and small.” Mastering with Metropolis starts at £60 for one track. Further info can be found at www. metropolis-group.co.uk. COMPUTER MUSIC SPECIAL / 89
WorldMags.net EXPERT #3
Robert Babicz artist & producer The producer and engineer talks mastering tech
What does mastering mean for you? “Mastering for me is like being a makeup artist; a nice-looking woman comes to you and you turn her into the most beautiful princess in the universe. It’s a philosophy of how you work on a track and how you colour it with your own view.” It seems like it’s been confused with ‘loudness’. Do you agree? “If you just want to be loud, no need for a mastering. Just put a limiter on the end and push it to the limit. Real mastering is more – so much more.” Is there an argument that some mastering plugins have created confusion in what mastering is for? “Yes, sure. Recently I had a customer who was so into a particular preset on one of his plugins that it was his only reference point. Only when he was able to listen to his music on my speakers and in my room could he hear how much his music had been fucked by this thing.
“For my own music, the mastering is done in a few minutes because I really take care during the mix” “But in the end it doesn’t matter because if you love the fucked-up sound then that’s the right way. I personally love sound too much to be trusting a preset.” What’s the most important thing for you when mastering a track? ”To listen to the track. I mean really listen and understand what the track wants from you. Then you know where to go and what limits are better not pushed too far.” How separate do you think mastering and mixing need to be in order for them to be really effective? “For me, the mixing process is much more important than the mastering stage because that’s the point where all important decisions are made. In mastering, you are mostly able to repair and make the nice things nicer.” Do you master your own work or would you prefer a fresh set of ears and perspective? “I totally know what I’m doing for my own music and the mastering is done in just a few minutes because I really take care during the mix. There is really not much left to do when I master my own music.” 90 / COMPUTER MUSIC SPECIAL
What tips would you give someone who didn’t have the budget to have their track mastered professionally and wanted to give it a go at home? “The most important thing in mastering, or even doing music in general, is having a place were you are really able to hear what is going on in your music. That is when you can hear what is wrong or right. “So that means that your tiny laptop speakers are not the ones you want. The room you do your music in is also very important. If you can hear big-sounding bass waves in your room and just a tiny kick in the track, then that’s a sign that something is not right.” How important is knowing your room and your speakers?
“The most important thing in mastering is having a place where you can really hear what is going on”
“This is your main equipment, I would say. It took a little while for me to understand this for myself when I started. You need to be able to hear what you do. It’s the same as driving a car and having just a tiny hole for a window – you need to be able to see where you are going!” If you were to give some advice on achieving loudness while maintaining a dynamic RMS level, what would it be? “It’s almost like you are playing a game – you have to try to find the right balance between how far you can push the sound you have and how much you can let your music get distorted. So, you have some parameters to play with. “On the other side of things, listen to what happens to the music when it’s played loud; does something hurt you? In that case, you need to bring down this frequency part a bit. You really don’t want to be causing anyone any pain, do you?” What’s the best level to listen at when mastering? Does cranking up the sound reveal more, or should you master at a more realistic listening level? “I listen in three volumes usually – normal, loud and then very low. Usually, the very lowest volume is the one that shows me where the loudest part in the music is.”
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WorldMags.net Parallel compression Parallel processing is a powerful weapon used by many mastering engineers. This process involves duplicating the signal, ie having two versions of the same song (let’s call them ‘A’ and ‘B’), and then very heavily compressing signal B while leaving signal A untouched by compression. As a result, you turn the negative implications of compression around into upward expansion. Signal A’s dynamic range remains entirely untouched, which is great, and it becomes complimented by the heavily squashed signal B, which can really beef up the overall impact of the song. It’s kind of like having an ugly wingman to make you look good. Signal B is only there to compliment signal A and so not much of signal B is needed to have a nice effect. Let’s see how it’s done...
> Step by step
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Just some of the top-grade analogue-to-digital converters you’ll find in a pro mastering studio
Double your track’s power with parallel compression
In your choosen DAW, add a compressor to your track. Make sure the audio is passing through the compressor completely unaffected, like here: Ratio – 1:1 and Threshold – 0.
Blend to taste with the volume fader. A good tip is to start with B fully turned down, close your eyes and slowly turn it up until it feels right. Remember, you won’t need much of B to achieve the desired effect – it’s only intended to add support and strength to A.
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Duplicate the track, add a group to the project and assign the output of the two tracks to the new group. This combines the two signals.
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Remember to volume match. The way we have set up our signal flow here will mean we need to create another clean version of the song as a reference track. Here, we’ve turned down the group ‘combined signal’ by 0.7dB. We can see the peak values are each hitting -5.3dB.
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Now it’s time to squash the life out of track B. Consider what effect you’d like to achieve. Would you like the drums to become a bit punchier? If so, a slower Attack time may help allow some of the peaks through.
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You can achieve very different sounds by changing Attack and Release times. If the track feels a little weak, try a fast Attack on B’s compressor and fast Release, removing all transients on B. When this is added to track A, it can provide a bit of reinforcement.
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WorldMags.net Dynamic EQ Dynamic EQ really is a versatile and useful tool and for many producers it can be a fantastic and muchneeded problem solver. In short, it is a virtual processor that can be used for eliminating the shortcomings of static EQ by providing a dynamically applied equalisation to your work. It combines multiband compression and EQ and can be a powerful tool in many ways. For example, removing problematic frequencies, boosting a weak kick drum, de-essing, or toning down a troublesome element that is causing you problems when it’s playing in the song.
> Step by step
Dynamic EQ works just like a regular static EQ except for one important thing – you can set a compression threshold, so only when a specified frequency range passes over that threshold is it reduced or boosted accordingly by the desired amount. For example, if you cut a frequency in a dynamic EQ, it won’t apply that cut until the frequency crosses a particular threshold. This can be exceptionally useful in toning down a hi-hat that’s too loud, for instance, or for reducing the volume on a very resonant tom drum that only occurs every so often in your track.
Keeping bass in check via EQ is a vital part of mastering for vinyl
Let’s take a look at the possibilities of the dynamic EQ a little more...
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We’re using a Mid/Side dynamic EQ from Brainworx called bx_ dynEQ. By sweeping about with an extreme boost and a narrow Q (and threshold to the right), we’ve found a problematic frequency at 349Hz that occurs once every four bars. We’re going to remove some of it, but we only want the EQ to cut when the sound plays.
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In this next image, we’ve done the opposite and used the dynamic EQ as an expander and boosted the sub in the kick drum every time it plays, while also sucking out some of the bass in the sides each time the kick hits.
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Switch the EQ to Cut and reset the Threshold to Off. Slowly move the threshold to the right until you can see the narrow red cut take action once the sound is playing. This tells us we’re targeting the right frequency as it only seems to spring into action once the sound is playing. We’ve successfully targeted the problem frequency.
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This only springs into action when the kick crosses the threshold. It’s not the same as using a standard static EQ to simply boost the low end as it is specific to parts of the song where the kick is in action.
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WorldMags.net Mid/side EQ Using a Mid/Side EQ can be an extremely useful tool in the mastering process. Basically, it splits a stereo signal into mid and side channels. The mid channel contains all the mono information – that is, everything that is identical in the left and right speaker – and the side channel is the opposite. It contains any information that is different between the left and the right speakers. You will find that bass and drums often reside in the mid channel, whereas wider elements, such as stereo reverb and panned elements, can be found occupying the sides.
> Step by step
It’s sometimes useful to have a listen to this split signal on a stereo song as it gives you an opportunity to get a glimpse ‘under the hood’. Maybe there is a strange resonance that is problematic only in the mid channel that you can get rid of with surgical EQ with a narrow Q cut, leaving the nice sides alone. Maybe you want to roll off the bass in the side channel to tighten up the low end and focus that kick drum straight down the middle. This gives the mastering engineer a few options they wouldn’t have with only a standard EQ and is an invaluable tool when used correctly. Let’s check it out…
Routing a stereo track through an expensive outboard to split it is common practice
Split your stereo track to deliver bespoke EQ treatments
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Pro-Q is a great EQ from FabFilter which has mid/side capabilities. Simply switch to Mid/Side in the Channel Mode section, then you can EQ the mid and side separately.
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You could give the sides a boost in the high end to bring out the intricacies lurking in the reverbs and effects. When used subtly, this can bring out a bit of shine in the top end and help open up the track. It’s useful to be able to test whether this high-end boost sounds better on the sides alone, the mid alone, or the mid and side combined. It gives you more flexibility as a mastering engineer to have these extra options available to you.
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In this instance we have rolled off the low end in the sides, with a smooth 6dB per octave curve starting quite high up, while the mid is EQed slightly differently: rolling off with a steeper curve and a little boost on the subs before the cut.
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Here is the resonant frequency adjustment option we discussed earlier in action. We are removing a little bit of resonance from a problematic frequency, but only in the mid channel, while leaving the sides untouched. This adds another level of control to your bag of tricks as a mastering engineer and gives you much more precise and flexible EQ options when dealing with problematic frequencies than just a standard EQ.
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MASTERING ESSENTIALS We wrap up our mastering guide with your one-stop-shop glossary of mastering tools and techniques and a brace of top tips
TOOLS OF THE TRADE To master your own tracks, there are a few pieces of kit you need to do a decent job. Let’s have a look at the essential ingredients of a mastering meisterwerk.
Monitoring The point of mastering is to create a mix that sounds great anywhere you hear it, and such a feat requires a decent pair of monitors. There’s no point troubleshooting a frequency only to find out that it was your monitors at fault, or to completely miss a problem sound because your speakers didn’t reproduce it properly. In general, the more you spend on speakers, the better they will be, so this is an area that will demand your cash. And just as important is the room itself, as even the best speakers won’t sound any good in a poor acoustic environment, which can cause poor stereo imaging, a bumpy frequency response and a general lack of detail. This is a whole topic in itself, but do take it seriously because without a decent-sounding room you will struggle to create good mixes and masters. The things to look into are speaker placement and the use of acoustic treatment products such as bass traps and absorption/ diffusion panels. Assuming that you’ve got all that sorted as best you can, listen to as much different material as possible until you learn the limitations and idiosyncrasies of your environment. When mastering, you should also listen to your tracks 94 / COMPUTER MUSIC SPECIAL
on many different systems. Even top engineers who use a well-calibrated system and have used the same room for years check their work on midfield speakers and iPhone headphones as a matter of course. If your master sounds awesome in your room but unbearable on earbuds, then it’s back to the drawing board. If you’re really stuck, there are plugins that claim to compensate for flawed monitoring and poor acoustics, such as IK Multimedia’s ARC. While these work to an extent, they’re not a magic bullet and you’ll get the best results using them in combination with a well-treated room and great speakers.
EQ This is one of the most critical processes in mastering and is also important for setting the general tone of the finished master. For mastering you’ll need a fully parametric EQ with shelving filters for the top and bottom end, as well as a high-pass filter. This allows precise work on specific frequency ranges as well as broad top- and bottom-end adjustments.
Compression We all know what compression does in general, and it’s no different in mastering. A compressor’s main use is controlling transients and groove. So, the process can be used to accentuate or smooth the punch of your bass and kick, or to add – or even remove – pumping.
It’s important to choose the right compressor, as some are better suited for sounds with fast attacks, while others will shine with smoother grooves. While valve and analogue emulation is all the rage, these models are not always ideal for mastering, where you often need something accurate yet not overly clinical.
Multiband dynamics These allow you to compress and limit (and sometimes expand) different parts of the frequency spectrum individually. When it comes to troubleshooting bad mixes, this is an incredible boon, as it allows a huge amount of control over each part of the mix. This means that these tools can be used to tame bass, bring out vocals, reduce harsh top end and fix a multitude of other issues. It can also be used more subtly to pull together parts of a mix and add some sonic glue. With more bands come more problems, however. Multiband dynamics is not mandatory, and some mastering engineers rarely use it, or perhaps only a band or two. If you’re relying on multiband dynamics to get a good sound, you need to work on your mixing skills.
Limiting Limiters have become synonymous with the often overly loud style of modern mastering. This processor prevents peaks in the signal exceeding a certain threshold, and so the signal
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WorldMags.net 112dB’s Redline Equalizer has an Auto Gain mode that’s massively useful in evaluating your tweaks
Analogue emulations such as this one from Nomad Factory can provide a touch of warmth and smoothness
FabFilter’s Pro-Q can operate in Linear Phase mode, meaning the sound won’t be subject to phase distortion
can then be turned up louder without those peaks causing digital clipping.
Stereo tools Stereo processors come in a number of flavours, and these have varied uses in mastering. They are typically used to create a greater sense of space in the stereo mix, but this can have a knock-on effect on the clarity and focus of certain parts. You may also reduce the stereo width to create a more central punch. The most important things to be aware of when using stereo tools are that bass should usually be mono and vocals should normally take centre stage. It’s also important to realise that stereo widening often increases the perceived level of high frequencies, so that is something to consider when EQing your mix.
Signal analysers It’s true that you should trust your ears above all else. However, even pro mastering engineers like having a visual meter on hand to guide and confirm their impression. When you’re mastering your own stuff in your own studio, using a meter is even more important because your monitors may have problems reproducing certain frequencies. This way, you can keep an eye out for such ear-evading problems. Stereo correlation meters can also help you to keep an eye on phasing issues (which are sometimes introduced by stereo processors).
So where should you place your metering? As you’d imagine, stereo analysers should be placed after any stereo field processors in the signal chain (any decent stereo mastering plugin should include one). Frequency analysers should be applied at least after your EQ, though ideally you want to be able to check what’s going on before and after the equaliser to see the effects, and so we recommend you place two into the chain. Many modern EQ plugins feature built-in frequency analysis anyway. Dynamics processing can have an impact on the frequency content, too, so we like to have an analyser right at the end of our chain, to check the final product. Try different analysis modes, too. Some analysers feature an RMS/Hold mode that gives an average of the signal’s frequency content, rather than a constant update. If you aren’t sure what a good mix looks like, this is where playing back professionally mastered material of the same genre comes in handy. Play your favourite mastered tracks through the analyser and note how the meters behave, then compare it to your own stuff. If there are obvious differences between your tracks and theirs, you’ll have an idea of what needs attention on your tracks.
Processor order As we’ve already explained, you might use some or all of these tools on any given track.
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However, it’s no use applying the appropriate processors if you’re not using them in the right order. It is vital to have an understanding of the signal path and how this will impact on your finished master. There are two considerations to think about here: processor order and gain structure.Let’s have a think about processor order first. Some tools will affect the quality of results further down the line. EQ, for example, affects the level of frequencies within the signal. This means that it will impact on the results of any subsequent dynamics processors (since these react to signal level), particularly on heavy bass frequencies. For this reason, it is often a good idea to place your EQ before the compressor to ensure that it’s working on a well-balanced mix. The limiter should always come last in the chain. This is because it doesn’t just create loudness but is also a functional device for preventing clipping and overloaded signals. Gain structure is the other consideration. Each processor will affect the level a little, especially compressors and limiters, so be careful that none of them cause any overloading or add unwanted distortion. Any decent mastering processor will feature an input and output meter so you can be sure it’s getting enough signal and not overloading internally. Most modern plugins are tolerant of overly hot levels, but many still do work best with a sensible signal level. COMPUTER MUSIC SPECIAL / 95
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Tape emulation Quick tips Read and heed our quick list of wrap-up mastering advice MOVE THE BASS If your bass is muddy, try using Mid/Side EQ. By reducing the clutter from the stereo field, the centre bass can poke through, and in addition, it’s not getting muddled up in the busy side signal. COMPRESSION AFFECTS WIDTH A narrow mix can be widened using Mid/Side processing, as compression and gain on the outer frequencies will accentuate them. Just be careful not to obscure important parts in the centre such as bass and vocals. An overly wide mix can be reigned in by compressing the centre signal, then raising its level back up.
Tape emulation plugins recreate the warmth of analogue kit and are best applied at the mixing stage
Many modern producers are convinced that sounds used to be ‘warmer’ back in the days of analogue gear. They cite the cosiness of old recordings and seek out tape emulation and valve saturation plugins to try to recreate the sound for themselves. But is there any truth in their holdings? The reality of the matter is complicated. Old recordings were indeed made courtesy of analogue units, but the warmth of the sound provided was due in part to the hugely expensive, high-end gear, jockeyed by an army of highly specialised and gifted professionals. Such records were often also completed in very laborious ways, such as endless bouncing to accommodate a full band using just a four-track recorder. Many factors contributed to the sound of old tracks and, indeed, most of us never listen to them on the sort of equipment they would originally have been played on. So, is there any place for tape emulations in mastering? In our opinion, warming is best applied at the mixing stage, rather than at the mastering stage. Instead of whacking a tape emulator on your final mix and setting it to one of the ubiquitous mastering presets then basking in the subtle distortion and compression achieved, try applying the effects to individual channels and use a compressor on your final mix. Most tape emulation plugins are based on multitrack tape recorders and although they can add character to individual channels, they can also take away from the clarity and dynamics of a full mix. Tape emulation can take some of the edge off a sharp digital signal, then, but so can EQ, and without distorting things. Of course, there are some situations (such as a soundalike or a song intro) where you might want to properly mimic that warm and fuzzy style. And in that case we say: go for it!
TRICKSY KICKS If your kicks are cutting through too much yet turning them down takes out too much, try compression with a fast attack and/or slow release. Conversely, if your kicks don’t come through enough, try a slow attack to let the crack of the kick through, plus a fast release. Multiband compression can zero in on kicks. DROP THE OUTPUT When creating MP3s you’ll often get better results by converting from a WAV that has been mastered to a slightly lower output level than 0dB. We recommend -0.5dB. Hard to believe, but try it for yourself!
“Old recordings were made with hugely expensive, high-end gear, jockeyed by an army of gifted pros”
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USE MULTIPLE LIMITERS Don’t just use one limiter. Try two or three, each doing a little bit. This allows you to gradually scale up the limiting. Two limiters with 1.5dB of gain reduction could sound louder and punchier than the same limiter with 3dB reduction. TOO LOUD TO BE PROUD It’s a misconception that mastering alone is responsible for loudness as any track’s potential for ‘going loud’ is really down to how well it’s mixed. If your mixes fall apart when pushed as loud as commercial ones, it’s likely it is your mixing that’s at fault, not the mastering.
If your track can’t stand up to being played loud, it’s your mixing you need to address, not your mastering
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iZotope Ozone is wildly popular for good reason, but feel free to use other plugins alongside the suite
All-in-one solutions
Need help getting your vocal line to cut through? Try boosting the mids or use a Mid/Side processor to make the centre channel louder
CUT BEFORE YOU BOOST If your mix is too dull, don’t add top end – try taking out some bottom end first. There’s only so much space in a mix and too much of one thing will obscure others. KNOW THE LIMITS For the radio, sometimes louder isn’t better. Most people think that to get your tracks really loud on radio, you need to limit them. However, if you limit a track too much at mastering, it will actually wind up quieter on radio once it has gone through their system. DO DITHER Dithering is a complex subject, but all you really need to know is that you should apply it at the very end of your mastering chain, after all other processors (many master limiters include dither for this very purpose). Dither makes a very subtle difference that you may only hear on the quietest parts of your track, but it’s dead easy to apply and there’s no reason not to do it, especially when rendering out to 16-bit WAV (eg, for CD). So just do it!
Choosing individual plugins can be a dizzying prospect so the appeal of all-in-one solutions such as IK Multimedia’s T-RackS 3 or iZotope’s Ozone 5 is obvious. But are these one-stop shops a good idea? iZotope Ozone is one of our favourite such packages. This mastering suite has much to recommend it, offering compression, EQ, frequency analysis, loudness maximising, demonstrative presets, stereo imaging tools and even (the slightly contentious) mastering reverb. It sounds fantastic, too. So, what’s the catch? If one thing is clear from these pages it is that mastering tools and settings vary as much as music itself and the taste of the people who make it. So, most all-inone solutions will come with certain limitations. You may find that a certain tool within the package just won’t provide you with the results you desire. We aren’t suggesting you dismiss all-in-one mastering plugins or bundles altogether. Indeed, Ozone 5 and T-RackS 3 both offer individual effects, so you can use just the bits you like. Bear in mind, though, that you may get better results by involving other plugs too.
Exciters
HAVE SOME RESTRAINT Just because you have all these great tools, it doesn’t mean you have to use them on everything. If the music is wellbalanced, maybe you don’t need a compressor. It’s all about gain structure. You want to turn your track up, but you don’t want to ruin the transients.
One of the simplest tools for adding top-end sparkle to a sound is an exciter. This process can be carried out in a variety of ways, including generating higher harmonics based on the information in the signal, applying more traditional EQ boosting and subjecting the signal to other more unusual methods. BBE’s Sonic Maximizer range, for instance, claims to introduce tiny signal delays at certain frequencies to compensate for the delays introduced within electrical circuits. However they work, we’re wary of using exciters on the master buss. Firstly, if your mix is OK you shouldn’t need to add much to the top. Secondly, by adding to the harmonic mix, you’re increasing the amount of sonic clutter. While top end is important for a clear-sounding mix, it’s equally important that you don’t overload the frequency spectrum or expose unpleasant harshness. We’d rather use an EQ to add top-end presence and brightness, leaving exciters for individual channels. Maximisers and bass boosters affect the whole track so they’re better used on just the parts that need them.
BACK TO THE DRAWING BOARD If you find yourself using effects to excess, chances are you aren’t dealing with a good-enough mix. While a pro engineer might have to make the best of things in this situation, we DIY types have the luxury of being able to revisit the project and make mix fixes at any time. Check out the audio examples included to hear the effect of mastering at work.
Think twice about adding exciters at the mastering stage – you may be better off using them in the mix
NOTCH THE KICK Don’t use shelving EQ to boost your kick as when you do this, you are boosting all the harmonics too. Always use notching to just control the thud of the kick. Make sure you actually need to boost the kick, rather than making space for it in your bass using EQing and sidechain ducking. RAMP UP THE VOCALS If a vocal isn’t cutting through, go to the centre and add some mids. Boost around the 2, 2.7, 3kHz range , depending on the vocal tone. You can also use a Mid/Side processor to make the centre channel louder.
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WorldMags.net
WorldMags.net
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Audio examples for our tutorials and tips How to get loud Hear our headroom, limiting, master chain, space and using EQ tutorial audio Metering and visual mastering tools Hear for yourself the difference that tweaks via visual tools can make Pro producer’s guide to compression Audio examples of sounds under compression
Understanding digital A set of samples at a range of digital bit rates Width and polish Hear our track take shape through walkthroughs for width and stereo spread Mastering essentials A selection of mastered and unmastered tracks let you hear the magic at work
WorldMags.net
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WorldMags.net
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