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MIXING

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MIXING 2013 ALL NEW TECHNIQUES FOR YOUR PROJECT STUDIO

ALL NEW TECHNIQUES FOR YOUR PROJECT STUDIO

2013 EVERYTHING YOU NEED FOR A PERFECT MIX

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MIXING

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ISBN 978-1-909590-20-5

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MusicTech Focus: Mixing 2013

www.musictechmag.co.uk

Pro advice for mixing drums, vocals and instruments Essential production tips for every major DAW The best monitors and mixers reviewed The professionals reveal their production secrets 25 tips for mixing, monitoring and programming

Mixing in... Logic | Cubase | Reason | Pro Tools | Live | And more…

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Welcome MTF

Welcome

Welcome to a special Music Tech Focus on mixing. Firstly, let’s not get into a fight over this, but I would argue that the mix is the most important part of the music production process. You could say that the initial moment of musical inspiration might well be up there on the importance ladder, but if you then bury it in a bad mix, you might just as well toss it to the back of your hard drive and never visit it again. A great mix will enhance your initial concept and help bring that lightbulb musical moment to the masses. A great mix can, then, change the world! But, and there is a but. Mixing in 2013 is all over the place: in the box, out of the box, and with a gazillion genres to cater to, so where do we start? Why, at the beginning of course! So we open this special issue by mixing a track from scratch and, more importantly, going back to basics with the main elements of a mix – the vocals, drums, and instruments – and outlining some of the main mixing rules. Then we go into the detail, with specific production workshops on every DAW scattered throughout this issue, and a long hard look at mixing in the box (on p54) followed by that next step of mixing: the remix and mashup (p102). And the most important piece of equipment when mixing? Again, let’s not argue, but I’d say it is the monitor – the direct connection between your mix and your ears. So we’ve ladelled in a load of monitoring tips on p110 and reveal some of the best monitors out there – and mixers, come to that – in our reviews section starting on p117. Everything you need for the perfect mix in 2013? I’d say so. Enjoy the issue!

I would argue that the mix is the most important part of the music production process

Paul Pettengale Editorial Director [email protected]

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MTF Contents

Issue 28

Mixing 2013

100% PURE PRO MIXING TECHNIQUE

Pro advice for mixing drums, vocals and instruments ● Essential production tips for every major DAW ● The best monitors and mixers reviewed ● The professionals reveal their production secrets ● 25 pro tips for mixing, monitoring and programming ● And much more…

Workshops

Walkthroughs

p68

p114

p96

p80

p44

p18 p38 p26 4 | Mixing 2013

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Contents MTF

MTF Issue 28 Full listings…

MTF Pro Tips

MTF Interview

006 | Masterclass Mixing a track from scratch 014 | Logic Pro 9 Workshop Exploring stereo

25 tips for Mixing

Goetz Botzenhardt on 5.1 surround

018 | Step-by-Step Buss compression 022 | Interview Tony Platt

Mix processing – p30 Drum sound design – p48 Bass production – p64 Monitoring – p110

p72

026 | Ableton Live 8 Workshop Mixing tips & tools 030 | 25 Pro Tips for… Mix processing 034 | Step-by-Step Composing with sampled instruments

Reviews

p118

YAMAHA O1V96i UA The Boulder | Mackie DL1608

038 | Cubase 6.5 Workshop Working with loops 042 | Ten Minute Master Surround sound monitoring 044 | Pro Tools 10 Workshop Understanding gain-staging

Adam A77X | sE Munro Egg 150 | More…

048 | 25 Pro Tips for… Drum sound design

p122

p125 p120

054 | Masterclass Digital mixing & production 064 | 25 Pro Tips for… Bass production 068 | Sonar X1 Workshop FX Chains

p124

072 | Interview Goetz Botzenhardt 076 | Reason 6.5 Workshop Extreme loop processing 080 | Step-by-Step Groove arrangement & programming

10 Minute Master

MTF Interview

Surround sound monitoring Dolby Stereo Leftt

Centre

Dolby Digital 5.1 Right

Leftt

Centre

SSub LFE R Right

084 | Ableton Live 9 Step-by-Step Mix & workflow tricks 088 | Masterclass Buss compression 096 | Step-by-Step Layering bass

Left rround Surround Rear

p42

Righ g Right Surro Surround

110 | 25 Pro Tips for… Monitors & monitoring

Studio Icons Yamaha NS-10s

102 | Masterclass Remixing & Mashups

p128

114 | Logic Pro 9 Workshop Song structuring 117 | Reviews

Tony Platt shares 40 years of experience

p22

128 | Studio Icons Yamaha NS-10s 129 | Next Issue Recording 2013 130 | What’s on your MTF DVD

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MTF Technique Mixing from scratch

MIXING A TRACK MTF Masterclass Studio Technique

from scratch

Mixing is both an art and a science, and getting it right can mean the difference between a good track and a great one. Hollin Jones guides you through the process.

M

ixing is just one of the stages of the music production process but it’s arguably one of the most important. All the great songwriting, skilled playing and clever arrangement in the world means nothing if your mix isn’t good enough. A good final mix also means the best of starts for the next stage in a track’s production: mastering. A bad mix, however, cannot easily be fixed in the mastering stage. There is as much art as science behind getting a good mix, and you’ll need a grasp of both. Above all, a good pair of ears and an understanding of the kind of music you are mixing will serve you well. You might be lucky enough to be working with an engineer who can do the technical stuff to realise your ambitions for a production, or you may be engineering yourself, in which case you’ll be the one doing the automation, tweaking the EQ, compression and the many other small elements that combine to create the perfect mix. We’re going to look at how to mix the different elements of a typical track though, of course, no two tracks are the same, and all will require a slightly different treatment. The drums in a dubstep track will need to be powerful and brash, for example, whereas in a jazz production they will require a much more gentle approach. Nonetheless, there are some good general rules for carrying out the mixing process, both in technical and stylistic terms. Above all, sympathy to the kind of material you’re working with is crucial. If you are an engineer, this usually means being familiar with a very wide range of musical styles. As such,

There are some good general rules for mixing process, both in technical and stylistic terms

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Mixing from scratch Technique MTF

MTF Pro Technique Moving material between studios You might work for weeks or months on tracks but then, when it comes to the mixdown, move to a professional studio to take advantage of better acoustics and monitoring. If you have a limited budget, it’s worth considering spending it on mixing as it’s such a crucial part of the production process. We have mentioned that you could theoretically take your computer along and wire it into the studio, but it’s less disruptive and simpler to take exported audio files or OMF files on a drive or DVD. Your DAW project will almost certainly consist of audio and MIDI tracks, and some plug-ins. All the major DAWs allow you to export a project in OMF format, which is a sort of universal, cross platform file format designed to let you transfer material between systems. It works fairly well but has some significant limitations. OMF files store information about audio files and tracks, their position in a project and the project tempo. You can choose to embed audio files or link to them. OMF also stores MIDI tracks, but that’s about it. They don’t store automation data or any information about plug-ins. This is because all DAWs deal with these differently so it’s impossible to account for them all.

To export stems, which is the most foolproof way to move material, set your left marker at zero and the right one a few bars after the end of the project. Then choose to batch export all tracks as audio, in the same full quality as the project is using. If you have the option, export both wet tracks with effects and dry versions without. In the mixing studio you can then import Exporting OMF files is one way to get your projects to different systems, though it all the files starting at zero has some limitations. and get a pristine session for mixing. You lose editability, but in an emergency you can always return to your original project and re-export one or more tracks.

it never hurts to do your research by listening to commercially produced albums in the same style, or even auditioning samples or loops of, say, rock guitars or electric pianos, to see how they sound in isolation and thus how a specific type of instrument is supposed to sound.

Before you start Before you get to the fader-prodding stage there’s quite a lot to consider and understand. The first big question is whether you are mixing in the box or on a dedicated desk in a studio, and there are several reasons why this is an important distinction. If you are working in the box, as many people do these days, you will almost certainly be working with raw project data inside your DAW, complete with plug-ins, automation and edits. You may well also be relying on your DAW’s mixer to handle everything, rather than a hardware mixer, with your software feeding a stereo output to your monitors. There are several upsides to working in the box. Your project remains editable so if, halfway through mixing, you want to remove a couple of crash cymbal sounds from a chorus, it’s easy to do. That’s not to say you should be endlessly tinkering with arrangements at the mix stage, quite the opposite. But it’s handy to be able to make minor changes. You don’t have to export loads of stems, and you get to use your own plug-in collection, which you will be familiar with. If you have the budget, or are lucky enough to be able to use a professional studio, you will probably have to export your project as audio

Grouping tracks makes it easier to deal with big, complex mixes. It also helps you cut down on the number of plug-ins you run.

stems – ie, groups of tracks of pure audio, so that they can be imported into whatever system the studio is running. Even if you are running the same DAW, like Pro Tools or Logic, the chances of your having identical plug-in sets are small, so simply copying a project folder between computers is generally a non-starter. You would be well advised to export multiple versions of all tracks, one set with effects (wet) and one without (dry) and maybe even export a set pre-fader, so their volumes remain independent of your DAW’s mixer. The idea is to give you as many options as possible in the mixing studio. If you export wet tracks only, you will be unable to edit or remove the effects on the tracks, which can be a real problem. If you export only dry tracks you will have to re-apply all the effects from scratch, which isn’t ideal either. So having wet tracks and the option to replace any given one with a dry version for re-effecting offers the best of both worlds. Many leading DAWs now offer one-click batch stem export, so this process is much less time-consuming than it used to be.

Why go pro? The advantages of going into a dedicated professional studio to mix are pretty clear. They will have an excellent monitoring setup and probably acoustic treatment too, as well as a combination of high-end plug-ins and outboard processing hardware. They may well also have great metering facilities and multiple sets of monitors for testing a mix, which is something most home studios don’t have. Plus, if focus Mixing 2013

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Tech Terms ● SOUNDSTAGE The overall sonic character of a mix. Think of it visually – it’s about placing sounds in a certain space and making the end result make sense to the listener. ● STEREO IMAGE The relation between the left and right sides of the sound, and the centre. Bass, drums and vocals usually live in the centre, while you can add interest by panning other elements a little, or a lot, off to either side.

you are working with a mix engineer, they can usually offer valuable insight into getting a great mix, with experience and a fresh set of ears – yours will probably have been listening to the track for months already. There is a third option that has become increasingly popular as laptops have become more powerful and widely used for serious music production, and that’s to take your computer and plumb it in to the studio’s hardware. You can do this with a desktop of course, but it’s much easier with a laptop. The idea is that you retain editability and get to use your own plug-in collection, but can also use the studio’s outboard gear and monitors for mixing. There is one caveat, which is that you will need an audio interface with as many outputs as you have tracks that need to be fed to the hardware mixer. So if you have 24 tracks, you’ll need 24 separate outputs in order to use 24 separate hardware mixer channels. This can be a limiting factor since it’s generally only the higher-end models that have lots of physical outputs. You can get around this by using an interface with an ADAT option, which carries multiple channels optically, and also by submixing certain tracks like drums or backing vocals inside your computer, meaning they only need to be fed to a single stereo output pair. This can be fiddly because it introduces even more volume control stages into the signal path, so it’s worth asking yourself if you

You should go into the studio prepared to mix, not to tinker with your arrangements

Cubase 7’s new MixConsole brings together a range of tools and features in a single window, making managing mixes easier than ever.

might not be better off simply exporting audio stems and working entirely on the studio’s equipment. Whether you end up mixing in a home or professional studio, there are some good general rules to observe, regardless of the type of music you are working with. The first is that you should go in prepared to mix, not to tinker with arrangements. Of course, there will be times when the odd tweak or minor overdub needs to be made and that’s fine, but don’t go into mixing expecting to have to move sections of songs around – your attention should be focused on the sound, not the structure. Your listening environment is crucial. If you are mixing in a less than perfect space, like a home studio, be aware of its flaws. People do produce great records in spare bedrooms but usually only because they know that the bass response or the stereo imaging is wrong and needs to be compensated for. Audio analysis tools like plug-ins from Blue Cat Audio can be invaluable in helping you to combat the acoustics of a bad room, by showing you what’s going on with your audio before it has even been passed to the speakers. If you are in a space like this, testing your mixes on other systems is absolutely vital, since it will quickly reveal if, for example, your monitors are under or over-emphasising the bass, or if the vocals get lost when played on a hi-fi.

Retain a sense of perspective There’s a very physical side to mixing too, related to the tolerance of your brain to repeated listening of the same track. Be prepared to listen to a track tens, if not hundreds, of times while mixing it. It’s all too

MTF Pro Technique Files and formats You will almost certainly be working on a Mac or PC to create your music, and the mixing studio where you do your work is likely to be doing the same, if they are not running a custom recording setup like Radar. If you are going to move your project to another studio to mix it you will need to find out what format they need the files in, since this can head off the problems of turning up with a bunch of files that won’t load onto their system. Macs tend to use AIFF files and PCs prefer WAV files. However, if in doubt you should export in WAV format, since this is pretty much guaranteed to be supported on any system, whereas AIFF is not. You should be working in at least CD quality and preferably higher, though this will depend on your needs. 44.1kHz or 48kHz is standard, and many systems now support up to 96kHz or even 192kHz. There’s no point in upsampling however, as it adds

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only file size, not quality. So if you’re working at one of those sample rates, export your stems using the same sample rate. 16-bit is the minimum bit rate acceptable and you should probably be at 24-bit or maybe even 32-bit. Again, upping the bit rate on export will make no practical difference to your audio quality. It’s important to name your tracks properly prior to export, as the resulting files will inherit the track names assigned in your project. When you load your stems onto the target system, you really don’t want a load of files called “track 1”, “audio track 12” and so on as you’ll only have to solo and label each one manually. Correct naming will avoid confusion and headaches at the mix stage. Export your stems using the same settings at which you have been working. This information can be found in your DAW.


MTF Step-by-Step Subgrouping drums for mixing

CREATING A GROUP TRACK Here is a multitracked drum kit which we are going to mix, but will be easier to handle if we submix it, then assign it to a group. Start by creating a group track in your DAW. Here in Cubase it is done using a right-click or the Add Track menu. From the resulting window, select a stereo track, as we want to group the drums to stereo.

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REASSIGNING OUTPUT ROUTING Next, name the group track and then go to the first drum track and reassign its output routing from the main stereo out to the drum group. Repeat this for all other drum tracks and you will find them all playing through the drum group channel. Alter the individual drum levels to balance the kit, and move the drum group fader to change the kit level in the main mix.

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easy to lose perspective and start to believe that a mix is terrible, or that those tweaks you made to the drum sound have had a negative impact. Of course, these things may be true, but after several hours of listening to the same track it gets very hard to tell. Take breaks regularly, and don’t commit to saying that a mix is finished until the next day if possible. It’s common to leave the studio at night with doubts about the day’s mixes, then come back in the morning with fresh ears and realise they actually sound great. Comparing your mixes with commercial tracks is another really important thing to do. You might think you mix is sounding good but then fire up a comparable track off a CD and realise you don’t have enough stereo width, or that your bass end needs tightening up. Bringing comparison and context to mixing sessions helps keep you focussed and assists you with making the right call on mix decisions. Remember also that the commercial track will have been mastered, and yours hasn’t yet. So you shouldn’t be pushing for absolute volume at the mix stage, but rather aiming for a good balance and blend of elements and a decent overall level. Limiting, widening and overall EQ will be added during mastering but that’s not something to worry about too much while mixing; just focus on the task at hand.

Tech Terms ● AUTOMATION Changing parameters like volume, pan or effects over time. Much easier usIng software than hardware. Best used sparingly. ● BUSS COMPRESSION Applying gentle compression to the whole output of a track with the aim of making it ‘glue’ together and sound balanced and coherent.

ADDING INSERT EFFECTS As well as being able to change the kit level using the group channel fader, you can apply processing to the whole kit at once by adding insert effects to it. Here, a compressor has been added to compress the entire kit. You can also shape the kit’s sound using EQ and other tools on the group channel, as well as retaining editability on each individual drum channel.

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Beats mixing You can get now down to the business of starting to mix your track. You will almost certainly have a good working mix going anyway, as most people tweak and mix as they go along, while they are editing and arranging. Mixdown is where you make the final decisions. If you are mixing in the box you may already be pretty close to where you need to be. If you have exported stems to mix in a studio, a little more re-balancing is often necessary. There’s no hard and fast rule about where you start, but it makes a lot of sense to start with the

You shouldn’t be pushing for absolute volume at the mix stage, but rather a good balance

Using master buss compression prior to mastering can help to cut down on the amount of processing that is necessary later, but it’s entirely optional.

drums, since they generally form the backbone of any track. If you have used drum loops or samples, they will most likely be fairly well balanced within themselves already, and your control over them will be limited to using EQ and compression to draw certain frequencies in or out of the signal. If you have beats being generated by a virtual instrument, like Battery, BFD, BPM, Maschine or something similar, it will have an onboard mixer, complete with faders, panners and effects that you can use to submix the drum sound. There is also usually an option to route any of the channels out from the instrument to separate audio channels so they can be processed independently. This would be a good way, for example, to take a snare part and run it through a hardware effect like a Space Echo. This is even possible when you are working in the box, as most higher end DAWs have the ability to incorporate external effects on a send channel with delay compensation.

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If your drums are real, they will probably occupy a number of channels across the mixer, with one mic per channel, and invariably some bleed between the various mics. Bleed is a natural part of drum recording and not really a problem unless the mics have been poorly set up when recording. Since the drum channels all contain a recording of the same kit, it’s not usually an issue if there’s a bit of the snare on the tom track, or a bit of the floor tom in the hi-hat track – drum mics tend to be very directional anyway. If it bothers you, or the bleed is genuinely interfering, you can use gating on certain tracks to ensure that the track only sounds when that drum is struck. This works best for sharp, percussive sounds like the snare or higher toms, and can be used to tighten up the sound of the kit. Use a gate effect to keep the channel silent except for when the noise level crosses a threshold. You will want to use a very fast attack so that the gate opens quickly when a drum is hit, and a slightly slower release to allow for the natural sustain of the drum sound. Find a threshold setting where the gate opens quickly and only re-closes after the sound has tailed off. To find this point you will need to solo up the drum in question in order to be able to get it just right. If you do end up gating a few drums, be sure to listen to the whole

Tech Terms ● SIDECHAINING Using the audio output of one track, often a kick drum, to trigger greater compression being applied to another, often the bassline. ● A/B’ING Comparing effect treatments, or comparing your mix in progress with commercially produced material to get a better idea of how your version is going.

MTF Pro Technique Pre-mastering As we have noted, mixing is about balancing all the elements of a track together and placing every sound source perfectly so that it is not too quiet or loud, and not fighting with other sounds for space in the soundstage. This is easy to say, but in practice requires judgement, patience and, above all, a good ear. It’s important to remember that your mixdown isn’t the final incarnation of your file; that only Some gentle buss compression applied comes after mastering. So while it may be on mixdown can help to ‘glue’ your sound together. tempting to strap a limiter across your stereo outs to add some power to the mix, it’s rarely a good idea. Separating the mix and mastering stages helps you to retain perspective and has other benefits too, like allowing you to send the mixdown to a number of different mastering facilities, for example. During mastering you will compress, EQ, limit and perhaps add stereo widening, but some people like to perform pre-mastering at the mix stage. This mainly involves adding some buss compression to the mix, placing a stereo buss compressor across the master stereo outs of a project in order to ‘glue’ the sound together and add coherence. It’s not about adding volume; that comes later with limiting. But some gentle buss compression, correctly set up, can magically make a track come together and mean that you have to apply less compression during mastering. Other processing like EQ and limiting should be reserved for the mastering process, and you should leave a little time between finishing mixdown and starting mastering.

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De-essing can be helpful on sibilant vocals, and splashy cymbal sounds. To deal with plosives, try multiband compression.

kit again, in case the cumulative effect of the various gates is to drain the life and ambience out of the overall sound.

Join the group Drums tend to be the most track-heavy single element of any production, often occupying between six and ten channels, in comparison to one for bass, and a couple each for vocals or guitars. As such, they are a prime candidate for grouping, meaning sending them all to a group that is controlled by a single fader or stereo pair. The idea of grouping is to simplify your mix, and it can be applied to any sound where multiple elements are meant to work as one, such as backing vocals or groups of orchestral instruments. The aim is to balance the group of sounds within itself, so as to get a great drum sound or choir sound, and then assign those channels to a group channel so that the whole group can be brought up or down in the main mix simply by using a couple of faders. If you’re having to alter eight drum faders just to make the whole kit quieter, you’re likely to mess up your drum sound fairly quickly. By grouping them, the risk is much less. Although software allows you to create as many group tracks as you like, there’s not much point in grouping just two channels. Increasingly, DAWs also have advanced track linking options so submixing is getting to be more of a hardware-centric technique. It’s more or less a given that drums need to be punchy, so you will need to compress the kick and snare at the very least, and probably apply some compression to every drum. Individual drums, like the snare and cymbals can benefit from a little reverb, though the kick should be kept solid, dry and central in the stereo field. If you do pan any drums, it may be the splashy top-end cymbals. Even then it’s rare to hard pan drums; you might instead choose to just pan them a little off to the left and right to create more of a sense of space. Use EQ to work on each individual drum, shaping its frequencies to better tune the overall sound of the kit. You might pull the top end off the kick, for example, to make it more dubby, or emphasise the upper mids to make it more rocky. A snare can be made to ‘crack’ more sharply by rolling off some bottom end and pushing the highs, or it can be made more meaty by bolstering the lower mids. Hi-hats need to cut through a mix and again this is often done by emphasising the most attacking frequencies. If your toms are too boomy, rein in the lower frequencies and use EQ to make ambient room mics more or less prominent. As well as processing each individual drum you can apply processing to the whole kit, either by placing effects across a group channel or, in the case


MTF Step-by-Step Multiband bass compression

SYNTH BASS COMPRESSION Bass invariably needs compression on any track, but synth bass can be problematic, especially when it contains a wide range of tones from low to high. Typically, a single band compressor may not deal well enough with the whole frequency range, either sapping too much energy or allowing too much boom through at different ends. Try a multiband compressor to deal with it.

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BOTTOM END COMPRESSION This is specific to your bass sound and pattern, but often you might want to dial in more compression at the bottom end to tame the boomy part of the sound, while leaving the upper range more flat or with just a small amount of compression. You can alter the boundaries between the frequency bands to better account for the particular sound you are working with.

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of a loop or virtual instrument, as an insert across its audio output channel. This can be a good way of quickly changing the whole drum sound, to make it appear to be distant, for example, or crunchy, filtered or delayed. Some more conventional plug-ins, like compressors or EQ modules, have drum presets, so you don’t have to only use special effects here; you can use it as a technique for more general mixing. It’s worth paying some attention to the individual drum sounds first, but you may find that processing the whole kit in stereo saves a lot of time.

Big bottom end The bass is the other part of a track’s foundation, so it is often the part you will turn to next. Solo it up and have a listen to how it’s working. Like every other element of the track, you will have to work on it both in isolation and then back in the context of the track. Something that sounds great when solo’ed up may not sound good with everything else playing at the same time. Mixing is about the track as a whole, so don’t worry if something sounds a bit odd on its own, as long as it sounds perfect in the context of the track.

By understanding bussing and routing you can take better control of a mix. Insert effects are vital for shaping and processing individual tracks.

SIDECHAIN COMPRESSION You can also use a sidechain compressor to achieve the pumping bass effect associated with dance music. Assign the input from a kick drum to trigger the compressor and set the frequency accordingly. Used with less extreme settings, sidechaining is a good way to make space for competing elements in a mix, like bass track and kick drum, without turning either one down manually.

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Compression and EQ are usually the two main things to watch with bass, and maybe sometimes incorporated within saturation or amp simulation to crunch it up a little. EQ should stay centred in the stereo field, since the bottom end of a track is where the weight and energy lives; splitting this off to one side seriously compromises the overall sound. Again, the type of bass sound will determine your treatment. A rock bass might be quite pick-heavy and so need a decent top end and upper mid, whereas a fat dance bass sound might be bigger and rounder. As far as compression goes, you can use it to prevent the bass from booming too much in its lower reaches, while ensuring that it doesn’t lose energy if it gets higher. Compression can iron out the differences between the louder and quieter parts of a sound – this is especially important for bass, since lower sounds carry more energy and, without compression, can overwhelm the rest of the mix. Electronic bass sounds can be especially problematic because waveforms can generate very large, powerful signals which cannot be sufficiently tamed by a simple, single band compressor. In these cases you can use a multiband compressor to apply different levels of processing to the lower, middle and higher parts of the bass sound, or use sidechaining, if available, to make the compressor respond to only specific frequency bands. You should watch for conflicts between the kick drum and bass track, since both occupy the bottom end of the sonic spectrum. Sidechaining is often used to achieve the pumping effect in a lot of dance music, whereby the kick drum triggers compression of the bass track. Used to a less extreme extent, it can help to blend the two together, in conjunction with clever and careful EQ’ing.

Guitar mixing Where you go after drums and bass depends on what’s in your track, but for many people this will be

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electric guitars. These typically occupy the middle and upper ranges of a track and as such it’s important not to let them interfere with the vocals, which do the same. Electric guitars will usually benefit from compression, since they are quite loud – where there are several parts layered together, you can use EQ to make them fit rather than fighting for the same space. People often get creative with guitars at the mix stage – they can be panned around and effected more freely than other sounds. For indie music and rock, your guitars will probably need to be big, so you can experiment with tricks like double tracking them: duplicating the same part and then panning each one slightly left and right to thicken up the texture. You may also add amp or speaker simulations to bring more depth to guitar parts, and perhaps gating to silence amp buzz or hiss when there’s no signal. If you have recorded your guitar amps in stereo, watch out for phasing issues that can harm the strength of the signal. A simple phase meter plug-in will reveal any problems.

Tech Terms ● DYNAMICS Effects that change the volume or intensity of sound, such as compression, limiting and gating. ● PATCHBAY A hardware interface found in studios for routing sound sources and effects into the main mixing desk.

Work the keys Keyboards are used in many very different kinds of music and, when the sounds are electronically generated, you will tend to start from a point of

Getting a good sound from a piano comes mainly from mic’ing it up properly in the first place

Bass and drum tracks, or indeed any track where single band compression isn’t doing the trick, can benefit from sidechaining.

having pretty clean, strong signal. Virtual instruments, synths or workstations never actually play their sounds into the real world: they just go straight to the recording. As such they don’t have any specific requirements, apart from the general rules we have already looked at. Some compression may be required to make them sit nicely in the mix, and a little EQ will stop them fighting for space with other keys, guitars or vocals. Keyboard sounds run the gamut from very low to very high, so that’s something you will have to bear in mind. You can pan them around and add delays, reverb and other effects if required. One thing that can be useful with virtually generated keyboard sounds is to add reverb or space simulation and perhaps also some tape saturation, or very slight distortion, to make them sound more ‘real’, if that’s something you’re after. Real sounds, as in actual pianos, are much more complex to work with and getting a good sound comes mainly from mic’ing the piano up properly in the first place, which is an art form in itself. You will end up either with two or three separate tracks for the bass, middle and top ends of the piano, or one track onto which all three signals have been recorded. A bad piano sound is extremely difficult to rescue in the mix so you will be much better off concentrating on getting a really good take in the first place. With that much achieved, you should only really have to compress a little, EQ a bit and perhaps add a little bit of ambience.

MTF Step-by-Step Vocal processing

SOLO’ING THE VOCAL You’ll want to start by solo’ing up the vocal, though it’s also important to periodically listen back in the context of the mix as you go along. This is a male vocal, so we immediately need to drop some of the bass end out of it to make it sit better in the mix. You will need to tweak the vocal EQ again in the context of the main mix later.

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DE-ESSING Now apply some de-essing. Here we are using the DAW’s built-in model, which does a good job of pulling down the sibilance based on a threshold setting. You can use an auto threshold, or set it manually and specify the amount of reduction you want. Be careful not to de-ess too much because it can suck the life out of a vocal and make it sound artificial.

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AUTO-TUNING There are other vocal tricks, like auto tuning, you can use during editing, and during mixing you might want to double track a vocal to add weight to it. To do this quickly, try adding a doubler or cloner plug-in, which adds a second version of the vocal and lets you add slight delay and detune to make it sound like there are two versions of the vocal.

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MTF Step-by-Step Electric guitar mixing tricks

AMP SIMULATORS Here’s a project with a single electric guitar track recorded. Let’s say at mixdown you decide it needs to sound bigger, but aren’t able to do any overdubs. You can begin by adding a little warmth using an amp simulator plug-in, compression or distortion to the original track. If your original recording has lots of crunch already you may want to skip this step.

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DUPLICATING GUITARS A simple trick to fatten up guitars is simply to duplicate them and then pan them slightly, or give each one a different effect treatment. Select the track and choose to duplicate it, at which point the effects will be duplicated as well. You could choose to change the effects connected to the duplicated rack, or change the settings on the existing duplicated effect.

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Vocal mixing Vocals are a vital part of many productions and they may well be the last thing you come to when you are mixing a track – many people find it’s better to get all the music nailed down first and the vocal last, as they are quite different to the other sounds. That’s not to say that bringing vocals in won’t necessitate some tweaking of other sounds, because it might. Vocals tend to occupy a similar frequency range to guitars, keyboards and some orchestral instruments, so you can find yourself going back to the guitars, for example, and backing off some of their mid range to make space for the vocals, which are often more ‘important’ to the listener. You can employ automation to achieve this, so that when there’s no vocal the guitars come back to the fore, then back off again when the singing starts. Automation is actually a really handy tool for making more dynamic mixes and dealing with frequency conflict issues, though it’s easier to work with inside a DAW than on many lower-end hardware setups. The key to mixing a vocal well is to make it audible and upfront, but also to make it sound like it sits in the mix properly. Once again, the trick here is

PANNING GUITARS Now, in the mixer, pan one of the guitar channels a little to the left and the other to the right. You may need to drop their faders a little, since although they are now panned off centre, there are two instead of one, so their overall volume may be too loud. If you want to get really creative you could automate the panning of the guitars or their effect settings so they change over time.

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to use compression and EQ. Compression applied at gentle to moderate settings should help to even out the levels of a vocal so that quieter and louder sections sound less wildly different, and if your vocalist has moved slightly towards or away from the mic during a performance, it should be less noticeable. Male vocals will often need a lot of the bottom end taken out in order to place the vocal within the soundstage; female vocals may need some high end taming. Both usually sit mainly in the lower or upper mid ranges, and benefit from a little reverb – though not too much as it can get washy. If reverb

Vocals may need to be gated, to mute bleed from headphones, and perhaps de-essed

Multi-output virtual instruments like drum machines usually allow you to route and process each of their many channels separately for greater flexibility.

isn’t working, try a little short delay – a trick used extensively in rap. Vocals may also need to be gated, to mute bleed from headphones, and perhaps de-essed. De-essing is a specific form of compression that only acts on the sibilant parts of the signal. Use it carefully. If your vocal takes contain plosives – ‘b’ or ‘p’ sounds where air has blasted onto the mic diaphragm – you may be able to lessen the effect of these by using a high pass filter or a multiband compressor to identify and pull down only the low frequencies that are resulting from the plosives. As we have remarked, every track will require a treatment specific to its style and to your goals for it. But with the general rules and considerations we have looked at, you stand a much better chance of avoiding mistakes and of getting a great mix. Invest a little time and patience in working on your tracks and, with a good ear, there’s no reason your mixes can’t sound great. MTF

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MTF Workshop Understanding stereo

Logic Pro 9 Workshop

Understanding & exploring stereo It’s easy to overlook the stereo qualities of a mix, and Logic has some useful tools for better defining your soundstage. Mark Cousins has the lowdown.

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ithout stereo, the world of music would be a considerably more onedimensional experience. However, despite stereo being such an important component of a successful mix, it’s amazing how many of us – myself included – take it for granted. Scratch beneath the surface, though, and you’ll soon find that Logic has a range of tools that enable you to evaluate, refine and control the stereo image to create a realistic and engaging left-to-right soundstage. Equally, it’s also worth taking a few moments to consider the phenomenon of ‘stereo’, and what precisely we’re trying to achieve by feeding sounds to two speakers rather than one.

PRO TIP If you’re intrigued by the possibilities of M/S processing, take a look at Brainworx’s bx_control V2 plug-in. The bx_control is an advanced M/S matrixing tool that can turn any of Logic’s plug-ins into M/S processors. Just place the bx_control at the beginning and end of the plug-in chain – everything between it works in M/S mode.

On the disc

MultiMeter plug-in. In short, the Goniometer is a form of two-channel oscilloscope that provides a visual guide to both the phase and stereo qualities of a mix. Unless you’re in a perfectly treated control room, the Goniometer is essential for understanding how stereo works, and, in the example of this Workshop, how the mixing decisions we make influence the final output. 1 The Goniometer is currently placed across the main stereo output, which is being fed by two instances of Ultrabeat routed to buss 1. Playing back the song reveals a completely monaural mix of the drum and snare, which is indicated by the strong central line in the middle of the Goniometer’s display. Add some reverb to the snare, using the instance of Space Designer patched across buss 2. Notice how the display ‘dances’ in response to the stereo information generated by the reverb. Even with the monitors turned off, therefore, we can gauge some important information about the mix, particularly in relation to the instrumentation’s positioning and the selective use of reverb. 2

Take to the stage

Down the pan

Rather than focusing on one plug-in in Logic, this Workshop will take a broad look at a number of tools that impact on the ‘stereo-ness’ of your mix. To best understand what’s going on, though, we first need to explore the Goniometer options as part of the

Turn off the reverb for now as we’re going to explore how the pan pot can control and influence the stereo image. Using the pan control on the first two channels, therefore, slowly reposition the kick and snare so that they’re hard-left and hard-right respectively. On the Goniometer, you should have seen the two instruments slowly move from the middle to either the left- or right-hand sides of the display, and you should now also be able to discern some of the sonic differences between the two sounds (this will help later on, visualising the impact of our mixing decisions). 3

Logic has a range of tools that enable you to evaluate, refine and control the stereo image 1

2 The Goniometer is currently placed across the main stereo output, which is being fed by two instances of Ultrabeat routed to buss 1.

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Understanding stereo Workshop MTF

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Using the pan control on the first two channels, slowly reposition the kick and snare so that they’re hard-left and hard-right respectively.

In effect, what we’ve created here is a form of stereo signal on buss 1, with the two instruments sitting at the two extremes of the soundstage. Of course, whenever you select a stereo audio track (or stereo virtual instrument) you’re effectively starting at the same point, although the precise makeup of the stereo image will vary (you might be dealing with just a touch of internal reverb, for example, or a collection of samples captured with a pair of stereo mics). What’s interesting to note, though, is how the pan pot, as simple as it is, influences the representation of this stereo signal. To hear the results, therefore, try moving the pan control on buss 1 while listening to the output and watching the movements on the Goniometer. What should be apparent is that the pan control doesn’t pan as such, but instead changes the relative levels of the left and right channels. Pull the pan to the left, therefore, and the snare gets gradually quieter and vice versa. At extremes, of course, the snare or kick (depending on which way you pan) will be completely removed, which may or may not be a good thing. 4

The real world Thinking this behaviour through, it’s easy to see both pros and cons to the pan pot’s influence on the stereo signal. In the real world, of course, this operation is much the same as what you’d hear in real life, with a greater emphasis on one instrument as you moved to either side of the soundstage (although, of course, you’d still hear the room ambience and a small proportion of the opposite side). However, if you’re panning a stereo piano sample that has a left-to-right positioning on the keys, the logic doesn’t carry through so well. In this case, the pan pot changes not only the instrument’s position on the stage, but also the relative balance of the left- and right-hand keys (the low and high notes, in other words), which could lead to some unforeseen musical deficiencies. So if we don’t like the behaviour of the pan pot, what other options do we have? Well, one simple but slightly drastic solution is to use the Gain plug-in, found under the Utilities and Tools folder. By mono’ing our signal before we pan it, we negate any relative level changes across the two channels. Of course, the cost of this action is that the stereo interplay between left and right is lost, but as we move to one side of the soundstage,

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the signal becomes increasingly mono anyway. In short, therefore, extreme left and right instruments may well benefit from being mono’ed first. 5 While we’ve got the Gain plug-in open, it’s probably worth noting some other important features relevant to stereo signals. Arguably the most useful one is the ability to swap the two channels of the stereo signal using the Swap L/R button. This is an absolute godsend if your Virtual Drummer plug-in is panned from the drummer’s perspective; going back to the piano example, if the keys are arranged as if you were sitting at the piano (for reference, music should generally be mixed from the listener’s perspective). There’s also the ability to switch the phase of the left- and right-hand channels, as well as the ability to adjust the L/R balance ahead of mono’ing the output. 6

Image makeover

PRO TIP Technically speaking, the Stereo Spread tool is best used as a means of creating stereo information from a monaural sound source rather than increasing the width of an existing stereo signal. The plug-in works by dividing the frequency spectrum and then placing the respective bands on either side of the stereo soundstage. It’s an interesting tool, but something that shouldn’t be overused.

Another way of controlling the stereo soundstage is, of course, to use Logic’s various forms of Imaging Processor, most notably the Direction Mixer. The Direction Mixer solves the previously noted problem about panning a sampled piano, enabling you to reposition the instrument but retain the balance between the upper and lower keys. You can see how the Direction Mixer works by instantiating it across buss 1 and using it as a replacement for the pan pot. As you move the Direction control to the left, for example, the snare moves with the kick drum, so that both signals are present in the left-hand speaker. At extremes, of

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MTF Workshop Understanding stereo

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The Gain plug-in enables you to switch the phase of the left- and right-hand channels, as well as adjust the L/R balance ahead of mono’ing the output.

course, the signal effectively becomes mono, but we haven’t compromised the musical information carried in either channel. 7 The prowess of the Direction Mixer doesn’t begin and end with the Direction control. The Spread control is just as useful, either restricting the stereo width or expanding it. The reduction of the width is easy to understand, especially when using the Goniometer. Imagine our stereo buss fader as two mono faders, panned hard left and right. In the case of a reduction of the Spread, the control works in much the same way as pulling-in the pan controls, with the signal effectively becoming monaural when the parameter is at zero. 8

Going wide

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While it might seem like a shame to restrict the width of stereo signal, the Spread control is a great way of rationing the use of the soundstage. In truth, overpopulating the extremes of the soundstage can create clutter. Indeed, it is said that there are three ‘golden positions’ in a mix – hard-left, centre, and hard-right – and that these three positions should be taken by a few select signals. As an example, I often find that the stereo delay sounds too wide (the taps are panned hard left and right by default), but a quick instance of the Direction Mixer will bring the output closer to around 11 and 1 o’clock respectively. 9 What’s trickier to understand, though, are the results of moving the Spread control outwards. In short, the Direction Mixer uses a form of M/S matrixing (for more

A better understanding of stereo will ensure your mixes have the width and dimension they deserve

Compiled by the Logic Pro experts from Music Tech

information see MTM 85’s Ten Minute Master on middle-side mastering) to break a L/R signal into middle (centre) and side (stereo extremities) components. The easiest way to understand a wide spread is by mono’ing buss 1, re-instantiating the snare reverb, then instantiating the Direction Mixer on the main stereo fader. Now move between a width setting of 0 and 2.00. With Spread at 0, the mix is middle-biased, with just the mono components of the mix. Move to 2.00, though, and we hear just the extremities of the soundstage; in this case, the ‘sides’ of the reverb. 10

Panoramic vision Although there’s plenty more to explore, this Workshop hopefully introduces some of the key issues in relation to stereo in Logic. Ultimately, the result of the Workshop could just be that you’re more informed about Logic’s pan control, but equally, by using the Goniometer and your ears you might start to be more inquisitive about the stereo information that a plug-in offers and how this is best carried through into the mix. Either way, a better understanding of stereo will ensure your mixes have the width and dimension they deserve. MTF

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The easiest way to understand a wide spread is by mono’ing buss 1, re-instantiating the snare reverb, then instantiating the Direction Mixer on the main stereo fader.

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MTF Walkthrough Buss processing

Production technique Step-by-Step

Getting started with

buss processing

If you’re working with multiple channels and need a more efficient way of manipulating them, you need to get bussing. Mike Hillier shows you how.

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uss processing is one of the most useful techniques at your disposal for making a mix sit together. It can help sections of instruments to gel and work together as a whole, as well as simplifying the mix by giving you direct control over entire groups from a single fader and providing you with a quick-and-easy way to carve notches out of whole groups of instruments to make room for another group. In a standard mix we often have several group busses running simultaneously – one for drums, one for guitars, one for vocals and another for FX. However, it is likely that within each of these groups we will also have created several busses. If, for example, we have multi-mic’ed a kick drum, we will buss these two (or three) mics to a single (usually mono) channel; the same for the snare top and bottom mics. The toms will go to a stereo channel; the two overhead mics will go to a stereo channel, as will room or ambience mics. Each of these in turn feed the stereo drums channel. If any of the

MTF Navigation Buss processes

channels have additional parallel or triggered sampled channels added to the feed these also need to be added to the respective buss groups – a parallel compressor on the snare will also need to be feeding the snare buss channel, while a triggered 808 kick will need to be feeding the kick buss channel.

Routing a drums mix Sticking with the drums example, this provides you with plenty of control over your mix. If the blend of Kick In to Kick Out isn’t right you can remedy this with the

Most mix engineers will employ at least a stereo buss compressor on the Drums Buss individual channels feeding the buss; if the kick is perfect but too loud in the mix you can bring it down on the buss channel without altering the balance of the two mics feeding it. You can also adjust the EQ of each mic, using the Kick In mic to get lots of top end from the beater while rolling off the lows, and the Kick Out mic to get the bottom end while rolling off the tops. You can

FROM BUSS TO STEM The two drums busses – the primary one and the parallel one – are fed to an audio channel, which we will use as an additional buss, enabling us to control the level of the total drums mix without adjusting any of the balances. Once we’ve finished our mix, we will record this channel down to act as a stem.

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d a SUB-MIX BUSSING Even within a sub-mix we will often buss individual tracks to single channels for greater control. Here you can see how we’ve bussed the two Kick channels into a single mono Kick buss, which in turn feeds the main Drums buss.

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PARALLEL BUSSES Parallel busses are great for smashing and then blending in with the primary clean buss. Here we’ve used a UAD Fairchild emulation to smash the drums buss, which we’ll blend in with the primary drums, using different amounts of the parallel channel in different sections of the track.

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BUSS PROCESSING Just because you are grouping channels into a buss doesn’t mean you can’t process individual channels. A combination of channel EQ and compression with buss processing can define a sound in far more creative ways than either one alone.

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Buss processing Walkthrough MTF

MTF Step-by-Step Sub-mixes & processing

Nesting busses within busses can get a little confusing, but if you keep to a sensible system it should make your sessions easier to navigate and enable you to process channels together. Here we have bussed two mics on a kick drum (an AKG D112 inside the kick and a Shure SM7 outside the kick) to a single channel. This enables us to EQ and compress the channels individually and process them together with tape saturation.

Any nested groups will in turn be routed to the main group buss, in this case the drums buss, but you could do the same with harmony instruments, routing acoustic guitars to a buss, electrics to another buss and then all of these to a Guitars buss. How you organise your session will depend not only on the instruments, but the role each of them plays – an acoustic guitar might be used more as percussion than for harmony, while an electric guitar part might be used more as a pad.

While Pro Tools 9 and 10 include automatic delay compensation, earlier versions do not, and even in Pro Tools 9 you have only up to 4,095 samples. In this mix we have already gone over that figure, so be careful to check the amount of delay you are building up and if you need to, add short delay plug-ins to compensate. Otherwise you may suffer from phasing issues, which will make your mixes sound thin.

With a quick static mix of the various levels of the parts that make up the buss done, add the buss compressor and adjust the settings so that it’s giving you just the sound you’re after. Here we’re using the UAD SSL G-Series Buss Compressor, which is a favourite of many mix engineers on drums.

With the buss compressor and any other dynamic processing in place on the buss, you should find that the overall balance of your mix changes slightly. Go back and readjust the individual levels of the tracks feeding the buss to get the balance exactly how you want it. Remember at this stage that every adjustment to any fader will adjust the whole tonal balance. If the snare comes up in the mix and triggers the compressor it will reduce the level of all the other instruments at the same time.

Once you’ve got your full mix working, you can use the sub-mixes you’ve created to build a quick stems mix of your track. Simply route the output of each group buss to a new audio channel and hit record. Stems can be useful for going back and making adjustments to a mix in future, as well as being invaluable for remixing and for the dubbing mixer if you are lucky enough to get your music featured in a TV or film production.

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even shape the two mics differently using gates, with a fast attack and release on the Kick In and slower attack and release settings on the Kick Out. Once bussed, you can further shape these sounds. A saturation effect, like a tape emulation, can glue the sounds and make the kick punch as a single sound rather than two discrete channels. Similarly, a little compression across the kick can glue the sounds together, and different attack and release settings on the compressor can shape the overall sound in very different ways than if you compressed each sound separately. For the overheads and room mics, the advantage of grouping two separate mono mics into a single stereo channel is that you can process each of the mics identically to maintain the stereo image. Stereo compressors will squash both sides equally whenever the signal goes above the threshold, whereas were you to compress each side individually, you may find your stereo image compromised as the signal in one channel goes over the threshold at a different

PRO TIP Riding levels into a buss compressor will change the way in which the compressor responds and can be a great way of creating movement in a track. To get your guitars sounding huge when the chorus kicks in without too much change in tone, try riding the level up 2 or 3dB into the buss compressor.

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rate. Even slight changes in time and level will cause the two sides to be compressed differently, creating a stereo image that wobbles with the compression. The overall Drums Buss will then be made up from feeds of various other busses, so ensure that your bussing hasn’t introduced latency issues, which would create phasing problems in the mix. In most DAWs the software will compensate for latency, but be aware of this problem and make sure that you have any automatic delay compensation (ADC) turned on. If you don’t have ADC in your DAW, look into manual ways of re-aligning all of the channels using very short delay plug-ins. The Drums Buss itself is another place where you can add additional processing. Most mix engineers will employ at least a stereo buss compressor on the Drums Buss, and by pushing the various levels into this processor you can shape your sound as a single unit. We find that a combination of subtle compression on each of the individual kit pieces and on the Drums Buss is better than pushing focus Mixing 2013

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MTF Walkthrough Buss processing

either one too heavily. However, we know plenty of other engineers who prefer to rely more heavily on the buss compressor across the drums for compression. Getting the master buss right is the final key to unlocking a great mix. However, many inexperienced engineers will often get this stage wrong, either by applying too much compression at this stage and ruining the mix, or by ignoring it completely and relying on the mastering engineer to compress the mix for them. If you really don’t feel comfortable compressing your master buss yourself, leaving it to the mastering engineer is an option – and it’s certainly better to do this than to over-compress and ruin it. However, compression can have a huge tonal impact on your overall mix, and leaving this decision up to someone else is putting a lot of faith in their decision. As well as compression, we might subtly colour the master buss with a tape-emulation plug-in such as Waves MPX or UAD ATR 102. You might also want to shape the overall tone with an EQ. However, don’t try to

PRO TIP If you are struggling to hear the difference between attack and release settings on your compressor, crank the compressor threshold (or input) so that your compressor is really pumping and adjust from there.

master your track from the master buss; leave this for another stage and export the mix here unmastered. You can, if you wish, add a limiter for reference, but bounce an unlimited version for mastering from.

Set up before mixing If you are going to be using buss processing, particularly compressors, it can be useful to think backwards when adding additional processors to the mix. We usually add a buss compressor to our master buss early in the mix process, as any changes to the overall balance will change the way in which the compressor responds and there’s little point carefully balancing all the channels in your mix only to then add a buss compressor and find that you have to go back and re-balance the channels. Similarly, when approaching a sub-mix we will usually decide early on if we need a compressor across the buss and add it before getting any internal balancing sorted, leaving any channel compression to last, once we’ve already heard the signal through both the sub-mix

MTF Step-by-Step Group and parallel processing

EQ isn’t a natural tool for busses, as in most circumstances any EQ changes that you may want to make will be better achieved by digging into the individual channels themselves. However, if you want to carve a hole in the spectrum for the vocal or other lead instrument it can often be quicker, if more clumsy, to carve it out of the busses. You can also use EQ before a compressor to get it to respond differently, then another EQ afterwards to correct for this change.

Buss compression is a great way of gluing a group of instruments together. Even a small amount of gain-reduction (less than 3dB) will get the instruments moving as one. Pick the right compressor for the elements in a group. We’re fans of VCA-style compressors such as the SSL G-Series and API 2500 for buss compression. FET compressors like the Urei 1178 have an aggressive sound, while vari-mu designs like the Fairchild 670 are sought-after for their smooth tone and strong low end.

Tape, tube and transformer saturation effects are great tools for subtly colouring busses. Be sure to add the colouration subtly, as it can quickly add up over the different busses and leave your mix sounding muddy and unprofessional. If you think you’re getting too much colour, either pull down the input on the plug-in or pull down the output of the preceding plug-in.

You don’t have to rely on a single channel to get the sound you want from a buss. Sometimes it can be useful to route the same buss to the input of two or more channels and process each one differently. Parallel compression is a common use of this technique, blending one fairly clean channel with another that has been heavily compressed. You can then use the relative levels of each of these two channels to get the sound you want.

There are plenty of tricks other than compression to use with parallel buss processing. Distortion is one of our favourites on all manner of sources including vocals, guitars, drums and synths. This lets you really crush the signal with your distortion effect to get a distinctive sound, but retain some clarity by balancing the distorted and clean channels. It’s usually useful to decide what range of the signal you want to distort as full-band distortion can sound muddy very quickly.

Parallel processing has the problem of splitting the signal onto multiple faders, so to regain control of the sound from a single fader route the output of all the parallel busses into a further buss on another channel. You don’t need to add any additional processing on this channel, but it means that, again, once you have balanced your parallel channels you can bring them both up or down from a single fader without having to adjust the relative balance.

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and master buss compressors. If you start by compressing the individual instruments you can get a great sound, which is then spoiled by the sub-mix and master-buss compressors. Working backwards you will only add compression to individual instruments that still need compression after passing through the two buss compressors and you will likely add less compression, creating a more dynamic and more interesting mix.

What goes where While it might seem sensible to buss all instruments by type – placing the drums into one buss, the guitars into another, the backing vocals into another and so on – it can be useful to mix up the routing a little to get different instruments to gel together. For instance, routing the bass and kick to the same buss and applying some compression can help the two instruments to punch together as one and has a similar effect to sidechain compression in that the bass will be ducked every time the kick hits. This trick can be expanded across the whole

mix, creating busses for each of three or four groups, sorted by frequency content: low-end instruments in one, midrange instruments in one or two busses of their own, and high-end instruments in a final buss of their own. Don’t high- and low-pass filter the various busses, leave them full-band, as a bass instrument will most likely still have some high-range information that you want to keep. Adding compressors to each of these four bands creates a rough emulation of a multiband compressor, but is much easier to control and can be pushed harder without sounding unnatural as each compressor is acting on the full-band information of those instruments. Buss processing can make mixing a much easier task, giving you control over groups of instruments from a single fader, but it also opens up a whole world of creative possibilities, helping you to make better mixes. Experiment with different EQs, compressors and saturation effects to see which of your tools works best on each buss – they’re not always the same ones you’d use on similar ungrouped instruments. MTF

MTF Step-by-Step Master buss processing

While mastering can be done from within a session on the master buss, we recommend that you bounce your mix. If you are mastering yourself, do it from a new session. However, it can be useful to shape your mix from the master buss before you get to the mastering stage. A quality outboard processor is great at this stage for adding colour across the whole mix. Plug-ins are available too, including the excellent (free) Softube Saturation effect.

We like to mix into a buss compressor as part of our master processing as this can have a huge effect on the tonal balance, colour and dynamics of a mix. We like to use a fairly slow attack so as not to cut off any of the transients and tend to rely on a compressor’s Auto release mode, which will follow the dynamics of the track rather than a set level. We can always fine-tune this later once we have more of a handle on the tempo of the track, but it’s a great starting point.

EQ is something we usually leave until the mastering stage, but as many modern mixes tend towards a bright top end, it can be useful to add a little ‘air’ to the master buss to save yourself from having to add it to every channel individually as you mix. There are some superb air-band-type EQs available, as well as some highfrequency enhancers if your source material lacks top end from which to boost.

Just like our sub-mix busses, you don’t have to have only one master buss. You can create multiple stereo aux busses to act as parallel master busses and route your audio to these before going out to the stereo outs. This enables you to do all the parallel compression and distortion tricks you would use on an instrument or sub-mix to your master buss.

Stereo widening is probably best left to mastering, especially if you are going to send your masters to a mastering house as they will likely have some dedicated tools. However, to get some sounds it can be useful to use stereo-widening tools followed by additional processing (EQ and compression) in parallel with your primary master buss. Here we’re using Flux Stereo Tool to add width to our master buss before rolling off the low end to create a greater sense of width in the top.

The final processor is usually a limiter. If you are bouncing a reference track for the artist it can be useful to process through a limiter so it roughly reaches the level of commercial tracks, but bounce another version of the mix without the limiter for your mastering engineer. They won’t be too happy if the mix they are sent is already crushed as not only does it limit what they can achieve in limiting, it also restricts what they can achieve with EQ and other mastering tools.

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MTF Interview Tony Platt

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Tony Platt Interview MTF

‘‘Being parachuted into the mixing stage in order to turn a mix around is a fairly flawed approach’’ The MTF Interview Tony Platt

As he rolls into his 40th year in the business, MTF asks this legendary recording and mix engineer about the key to achieving sonic success. Photography by Zen Inoya

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tarting his career as tea boy at Trident Studios, Tony Platt rapidly built a solid reputation as recording engineer on the successful Bob Marley & The Wailers album Catch A Fire in 1973. Before the decade was out he demonstrated his competence again, mixing the AC/DC release Highway To Hell. Since then he’s worked on timeless records for a variety of artists, including Iron Maiden, Buddy Guy, Cheap Trick and many more. Having performed recording, mixing and production duties for a variety of projects, Tony is keen to talk to us about his methods for achieving a great mix. The first thing we want to know is what the mixing process is for him: “Well, there’s a tendency for people to think of mixing as a separate part of the music-making process. For me it isn’t, it’s actually part of a continuing process. The mix starts at the point you start to talk about arrangements and pre-production. At this point you’ll have had a conversation with the artist about the album they want to make and how they want it to sound. This informs the recording process, where you’ll collect the necessary audio information to be carried forward to the mixing and mastering stages. The very beginning of the process allows you to establish a necessary dialogue with the artist, so you both know you are meaning the same thing when you say something like ‘I want this to be brighter’ and so on. Without that, the project can soon start to go awry. I’ve found that being parachuted into the mixing stage in order to turn a mix around is a fairly flawed approach. I’ve had to do this myself and the only time it works is if the material is already there with an existing framework already established. If you’ve got a less than good recording that’s not already pointing in the right direction you have to spend most of your time repairing and refocusing it, not spending your time on the creative side of things, which is probably why you’ve been asked to do it in the first place. It’s a waste of your time and their money.”

Mix and match Tony has an example of how pre-production can affect the recording stage, which in turn heavily affects the mix. He tells us of a band he’s currently working with who are based in the sticks of north-west Canada. They needed some guidance to get their songs and sound in shape before the rest of the process began: “They started with hard rock songs that sounded similar to their heroes and the lyrics didn’t ring focus Mixing 2013

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MTF Interview Tony Platt

true, as they were all about living in the city. So I tried to encourage them to think about the whole thing differently. Three of the group are from Inuit families and they have lots of stories passed down to them from their forefathers. I explained to them how this would be really interesting to people listening to hard rock. Up until I met them they’d been rehearsing with everything cranked up, so I asked if they’d try working on acoustic guitars in order to make sure they were developing strong musical ideas first, before going loud. It was difficult for them at first and they had to start thinking of different types of chords and how the bass needed to underpin everything as well. After a few days they wanted to have a quick blast-out again with everything cranked so I said yes. The songs then really started to sound good and after 20 minutes they’d seen the results and went back to the acoustics again. They really embraced the approach and started to make songs that really represented them, with a sound that was much more original than they previously had. “On my second visit for pre-production we were really just adding touches to the songs, like deciding on playing in unison or at octaves apart for song sections and so on. This means by the time they record, it will be a simple process as we won’t be struggling with how we want it to sound and this then translates to the mix."

MTF StudioEye Strongroom Studios, London PRISM SOUND ADA-8XR We did a lot of interface testing when switching to using Pro Tools and decided on this. It gives us an absolutely rock-solid way in and out of our workstation. We were also impressed by the Lynx Aurora but that definitely added a character to things.

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LYNX HILO This is part of a test setup to capture my mixes at 192kHz on a separate computer. It runs directly out of the Neve summing mixer into here. I’m moving the computer to the machine room but the unit will stay here as it has handy interface settings and a nice VU metering.

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Rags to riches Tony tells us about the humble setup for his first mixing gig on Catch A Fire by Bob Marley & The Wailers. He says he pretty much used just a few compressors and an echo plate. He then goes on to say, “The recordings were done in Jamaica and sounded absolutely fantastic, so you’d go with what the recordings had to offer you.”

‘‘Every time you start applying EQ and other bits and pieces you risk degrading the sound and shifting the phase”

recorded. This was just to brighten-up the cymbals on the drums as the ceiling above was a little lower than I would have liked. When it came to the mix, this is where that planning was useful as we’re on a jazz budget so had little time for mixing, so the mix is pretty much pre-formed in the recordings and I haven’t done much in terms of EQ and compression. In fact, if you look at the faders [on the project he has open on Pro Tools], they are pretty much lined up, barring a few because compression has brought up their level a bit. Every time you start applying EQ, compression and other bits and pieces you risk degrading the sound and shifting the phase. If you’re not careful you can be taking away from the sound you want it to be.”

We then ask Tony for an example on how you can get a good recording in the first place. He shares a few tips with a modern example: “For mixing, one big lesson I’ve learnt is that 80–90 per cent of its sound comes from the studio floor. If you’re recording the sound that you want with the right mic and preamps you have to do very little else to it in the mix. The album I’m working on at the moment is mainstream jazz with Anthony Strong, recorded at Air-Edel Studios. We chose there because the resident piano was particularly good and I knew it was also an acoustically pleasant room to work in. I then proceeded to use some very good microphones to capture it. We chose musicians who already have a sound they know how to get and I only used EQ on two microphones when we

Modern mixing

Tony’s best-of...

Defining career moments

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1971 | Luther Grosvenor Under Open Skies This was the very first album I did all of the engineering on and it sounds terrible. At that point in a career almost everything is a learning experience and I learned a lot.

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In Tony’s studio there are quite a few bits of gear and the current jazz mixing session onscreen shows only the odd plug-in or two. We ask what his core essentials are for mixing in his modern setup: “What excites me about the tools now is how many are available to us. In a way it was much easier in the 70s and 80s as you had a limited selection of tools to use, so you’d make the best of them and your skill base was focused in a different place. Now there are not only loads of plug-ins to know about, there’s also lots of hardware. I think there’s more hardware now than there’s ever been. Compressors, summing amps etc are all being released on a regular basis. The biggest challenge in one respect is to know

1973 | Bob Marley & The Wailers Catch A Fire My most memorable experience of this was when I was visiting my home town and someone I knew shouted my name across the street and said they’d bought a fantastic album.

Tony Platt Interview MTF

M&K CR2401 I heard a larger version of these a while back and was really impressed with the detail, clarity and lack of colouration. We used those a lot at the time for surround sound mixing and when these models came out I wanted a pair for doublechecking things. As we set up this studio they were just the right size for this size of room.

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TRIDENT AUDIO DEVELOPMENTS A-RANGE I recently tried this in comparison with the UAD and Softube emulations and the software is in the style of it but isn’t exactly the same. This just means I have the hardware which I like, but also two other flavours in software to choose from.

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e RUPERT NEVE 5059 SATELLITE After some blind tests I ran for summing mixes, the results were that they all just had a sound of their own, making them another tool. Ideally, I’d like a few more summing devices so I have a choice of further colouration to the sound to choose from.

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SPL MTC 2381 There are a few monitor controllers out there which didn’t match up to this. The MTC doesn’t add any character to what you’re listening to and it also has all sorts of monitoring options as well as a much-needed talkback facility for recording sessions.

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RUPERT NEVE PORTICO 5015 We do quite a bit of vocal, percussion, acoustic guitar and other instrument overdubs here. These pre’s are nice and offer lots of headroom. For vocals I use a little bit of compression so they can move in and out from the mic.

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about enough of this new equipment to use it creatively. In terms of the core decisions I’ve made to get to my current setup, I had to ask myself whether I’d persist to record things through analogue kit and then whether I’d mix through analogue. When the sonic quality and flexibility of digital workstations got to the point of having more advantages than disadvantages, that’s when I made the decision to switch across. This made it a creative decision as well as a technical one. Along with the microphone choice and preamps, the recording medium also informs the mix. I used to mix to 1/2-inch tape at 15IPS with Dolby SR. So I was getting a lot of warmth there, even when mixing in a digital workstation. But as budgets got shrunk, tape machines got harder to find in studios and the tape quality itself wasn’t always there. I then had to make the decision to work in-the-box. Other factors – like how you get your sound in and out of your workstation – are very important, along with using monitoring you know and trust. You need to know you’re hearing what you want. In a way the workstation you choose is really a matter of preference. I prefer Pro Tools because it’s more flexible for what I need to do and it’s set up for working with good practice, whereas other software can encourage some less than best practice.”

Analogue or digital? When we arrive at the studio we immediately talk about a few bits of kit. It soon becomes evident that Tony doesn’t go with the analogue-versus-digital debate in respect of seeing it as

1979 | AC/DC Highway To Hell I mixed this album as Mutt Lange needed someone with experience of mixing British rock. I fed instruments through speakers to get the ambience it needed.

an argument of one or the other. From hardware choice to processing in-the-box, Tony seems to judge his tools on their merit on a case-by-case basis. We ask him what he likes to use for software processing – he tells us about one of his favourites: “The Pultec EQs by UAD are great. Whenever you used a hardware version, you’d make the decision as to whether you’d use the programme EQ or the mid EQ. So you’d patch each one in and try it out, then re-patch it for the mid. More often than not you’d end up putting two units in a chain as you’d need a bit of each one. UAD have not only emulated each one, they’ve also done the Pro version, which is the two states combined. They also emulate the added character the hardware gave when used in bypass, too. “I tested the hardware against the software on these the other week at a seminar in Holland. They are different, but the essence is there. I think that the way plug-ins are going at the moment is really exciting. Although regardless of whether you’re using hardware or software, you need to be careful to not process things just for the sake of it. For instance, when I go to mastering sessions, there’s often a large array of wonderful hardware lying around, so we’ll go through a lot of compressors, which you obviously want to use when you’re surrounded by such lovely kit. The temptation, though, is to think it must sound good if it goes through that, but there are various times when we’ve ended up going through a plug-in instead, as it did the job needed for a particular track much better.” MTF

2011 | Antonio Forcione & Sabina Sciubba Meet Me In London I’d always had the idea of working at 192kHz in the back of mind, but this was the first chance to put it into practice as it was all transferred from tape at 192kHz. Lots of things had to be figured out and it was a good opportunity for experimentation.

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MTF Workshop Mixing tips & tools

Ableton Live 8 Workshop

Mixing tips & tools in Ableton Live 8 Live has a range of tools and routing options that can both improve and potentially destroy your mixes. Liam O’Mullane shows you around.

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bleton Live is a very capable DAW for both composition and audio processing. Even the most basic package has a good range of mixing devices to help you create a spot-on mix. Its drag-and-drop functionality makes experimenting with certain processing combinations easy tasks to perform, enabling you to create intricate processing chains and routing setups that would take a lengthy investment of your time in another DAW. In this Workshop we’ll explore a few routing options while also raising awareness of a few technical factors regarding how Live works. Without knowing about these potential issues you might end up with a bad-sounding mix – and no clue as to why. So, pull up a project that needs some mixing attention and we’ll show you how to get better results from Live.

PRO TIP Another easy-to-use tool for playing with stereo balance is the Utility device. Use the Width amount to rebalance the mid and sides of a stereo source, or keep it set to 0% and enable the device every now and then to check your mix for mono compatibility.

Improve your view The first port of call for information about your mix is via the mixer in Session View. An often-overlooked option here is the ability to grab the top of a channel and

Even the most basic package has a good range of mixing devices to help you create a spot-on mix 1

expand the view to reveal more metering information. Just left-click and hold the black horizontal line above the channel fader to expand it. 1 You will now be able to see two new elements located at the top-left of the channel fader: the current fader position and the highest signal level encountered in decibels (this can be reset with a mouse click). These are invaluable when you want your levels to be under 0dB on a channel or master track. You can make very minor adjustments to a channel fader by holding down [Alt]/[Cmd] (PC/Mac) while you left-click and drag the mouse to move the fader. Our favourite technique for making those incredibly minor adjustments, though, is to click the channel fader so that it’s highlighted, then use the arrow keys to move up or down in single-decibel increments. If you hold down [Shift] at the same time these movements are even finer (0.1dB increments). To us this just feels better than using a mouse as you experiment with fine level adjustments.

On the level Although the obvious control to use for automating volume fades and making real-time adjustments is the channel fader, this can actually cause problems if you want to make an overall level adjustment later (because Live doesn’t have any automation trim controls). Instead you have to highlight the whole automation lane and manually move it up or down, which makes accurate changes or just a quick tweak hard to perform. To avoid this potential nightmare further down the line, instead add a Utility device at the end of your processing chain for that channel. You can then

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Make the most of the mixer in Session View by expanding your mixer channels to see important signal level information. Also pay attention to which devices have Hi-Quality enabled to maintain the best sound quality…

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On the disc

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Mixing tips & tools Workshop MTF

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5 …Sub-grouping your Return tracks in the correct way gives you easy control over how silent your silent parts are in a mix while incurring no signal delays…

automate its Gain amount instead, leaving the channel fader free for overall level adjustments at any point during the mixing process. 2

Right-/[Ctrl]-click and select Insert Return Track after the existing tracks. Rename it FX Sub to keep things clear. Now set all other Return tracks to Sends Only from the Audio To menu. This prevents them from going to the master channel. 5 Finally, enable the send to the FX Sub from each Return track that needs to be rerouted from the right-/ [Ctrl]-click menu. 6 Turn up each send to feed them to the FX Sub track. 7

Behaviour awareness There are a few devices in Live which have a Hi-Quality mode that can be enabled/disabled by right-clicking the device’s title bar. EQ Eight, Saturator, Flanger and Dynamic Tube all have this option and selecting it will improve the top end for Flanger, the bottom end of EQ Eight, and reduce aliasing on both distortion devices. 3 Using this mode will incur a slightly higher CPU hit per device, but the main thing to watch out for is the fraction of delay added to the audio signal. This isn’t something you’ll notice straightaway as it is incredibly minor, but if you start to use Audio Racks for parallel processing, your signal will soon start to sound worse due to the comb filtering effect of the delayed and undelayed signal playing together. As an example, we’ve added an EQ Eight to the master buss of the DVD Workshop project. We then selected Group from the Edit menu to put it in an Audio Rack. Then we right-/[Ctrl]-clicked the space below the Chain strip in the Rack and created a new Chain. 4 This creates two parallel channels for the project to play through – one with the EQ and one without. To hear this in action you just have to activate the Audio Rack and you’ll hear the negative affect it has on the sound. So, if you want to use any of the devices mentioned in Hi-Quality mode for parallel processing, you’ll need to add another copy of the device to the parallel signal to act purely as a dummy processor so it will also add a delay and keep the two signals in-phase.

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PRO TIP Sidechaining isn’t just reserved for pumping house and busy D&B mixes. Try adding a compressor to an FX Sub track as explained in the main text and set its sidechain input to be the drum group. Now all of your effects will be slightly pushed back rhythmically by the drums, giving everything a little more impact and energy.

Until Live 8 arrived in early 2009, the only option for multi-band processing was to split a signal manually using EQs or Auto Filter to create each band. Although this technique worked, it was very hard to get the crossovers to be audibly transparent. Fortunately, this is now a lot easier thanks to the addition of the Multiband Dynamics device. Once this is set up, you could try adding some light saturation to the midrange of a sound, compress just the high end to reduce sibilance in a vocal, or use a sidechained gate from a kick drum to cut out only the sub frequencies from a full-range bass instrument. But to process a single band you still need to create parallel signals so they can have devices dragged to them alone. Create a Multiband Dynamics device and Group it via the Edit menu. Before we duplicate the Chain to make more instances of this device, right-/[Ctrl]-click the Split Frequency boxes for High and Low and assign them each to a unique Macro number. This will enable you to fine-tune the crossover frequency points of all instances at any point. 8

Another process that involves unwanted delays is when you re-route a Return track into an Audio track in the main mixer. The reason for wanting to do this would be to send all Return tracks to one sub-track so you can easily control their volume or process them with a single effect. Having a master level control for all Return effects is very handy if you want to be able to instantly cut all ambience effects in a song for moments of dramatic digital silence. But there is a way to perform this task without any delays, involving the use of a final Return track as the sub-track. focus Mixing 2013

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…The Multiband Dynamics device makes multi-band processing simple to apply in a mix with the help of Audio Racks and Chains.

Solo the Low band by clicking on the small S box to the right of the Split Frequency box. Now you will hear just a single band. Right-/[Ctrl]-click the Chain and select Rename. Call it Low. Click again and select Duplicate, then select the Solo for Mid on this second device and rename the Chain to Mid and repeat the process again so you have a third Chain called High that’s solo’ed for high frequencies. You will now have three bands spread over three chains. 9 You can solo the Chains to hear each band on its own and drag and drop devices to process each one differently. Just remember the delay and sound quality issues with devices set to Hi-Quality and again use dummy devices to keep the timing of each band aligned.

Stereo EQ Live’s EQ Eight is a very powerful tool as it not only lets you EQ a mono or stereo signal, but also enables you to easily EQ the left and right channels independently or the mid and the side elements of a stereo signal, all at the touch of a single button. Via the left/right mode you can create subtle or severe stereo effects while not adding any type of ambience, such as reverb or delay. Start by switching an EQ Eight to L/R from the Mode dropdown menu. 10 Use the Edit button to the right to change the EQ shape for the left and right channels. A good starting point is to find an area of mid–high frequencies that makes a noticeable change to the sound when cut or

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WANT MORE? Music Tech Focus: Ableton Live 8 Vol 2 is available now. Find out more at www.musictechmag. co.uk/mtm/focus

boosted, then use the image of the left EQ in the background and roughly match it with the right. Try moving the frequency area a little to start creating a stereo effect. Subtle changes of frequency or gain amount work best if you want a stable stereo image. If you already have a stereo source, try using the M/S mode and play around with the Side EQ shape from the Edit menu. One approach is to remove low end with a low cut or shelf, then boost the tops or scoop out the mids a little. We’ve applied this to our drum track after a reverb that we added to deliberately make them sound muddy. After EQ’ing the Side signal, the drums become much more defined in the centre of the mix. To get a more airy stereo spread we’ve added another EQ Eight afterwards for a slight high-end boost on the left and right channels, which are slightly offset frequency-wise. 11 This type of treatment isn’t just reserved for source sounds, though – a bit of EQ control on your Return tracks can mould your reverbs/delays to fit your mix. This can be applied per effect or to a FX Sub track.

An optimum future Although much of what we’ve looked at here appears distinctly unsexy, having knowledge about what causes issues with audio quality will help you to get much better results in the future. After checking the EQ Eight Hi-Quality Audio Rack on the master track of our DVD project, we’re sure you’ll agree that it’s worth using a dummy to mix with from time to time. MTF

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MTF 25 Pro Tips Mix processing

Pro tips for

mix processing

There is a multitude of mixing tools and features to be found in every DAW, but how and when should you use them? Liam O’Mullane takes you through the options. GAIN STRUCTURE When working with any plug-ins, keep an eye on your input and output levels to get the best performance out of each processor in a chain of effects. Although floating-point systems enable us to get away with running into the red at various points in your DAW’s mixer and plug-ins, it can greatly affect the performance of your plug-ins as you are hitting them above their optimum levels of operation. For example, you might run a signal incredibly high into a compressor without any audible problems, but if the levels are coming in way above its threshold range you won’t have full control over what the compressor is reacting to. You can avoid this by adjusting the output levels of plug-ins or using a gain plug-in to keep the level down.

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DRAG, DROP AND AUDITION Although there are various pitfalls to avoid when working in a software mixing environment (and these include the inability to make a basic decision because of the huge range of options open to you), a big plus is that you can drag and drop effects around to change the processing order. This is not only incredibly convenient for finding the best processing order for a sound, it’s also a great help when it comes to understanding how the order of effects changes the sound and creates certain tones and styles of production. A good example of this is to move a low-pass filter from post- to pre-distortion. Post-distortion gives a clinically clean filtering sound as frequencies above the cutoff will be sharply rejected after the distortion. If the filtering is placed beforehand, it will still reject a lot of frequencies but the distortion will add a coating of extra frequencies above the cutoff, making it sound smoother.

SHAVING OFF THE TOPS FOR DEPTH In the early 80s, engineers in the US were pushing the top end extremely hard to tape to get that hyped pop sound of the era, yet this would always be counterbalanced by the limited top end that analogue tape could take, resulting in a brighter but saturated top end. Today we’re in the opposite situation: when we work in the digital realm we tend to have too much high end captured, which can make a mix sound

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Mix processing 25 Pro Tips MTF

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everything still sounds in-time. You might also find delay in any part of a parallel processing chain. If one parallel channel is slightly off you will start to hear a comb-filtering effect from the sum of its channels. Comb-filtering will make sounds appear washed-out and they lose punch, so it’s always best avoided. FLAVOURS OF SIDECHAINING When sidechaining is mentioned, it’s probably fair to say that most people think of the obvious setup: pushing down the volume of an instrument from an on-the-beat kick drum pattern. However, there are many other techniques that can be used to get a better-sounding mix. Try subgrouping your kick and snare separately from another group that contains the rest of your drum sounds, then add a compressor to the other drum sound group and set its key-input (external sidechain) to be the kick and snare group. This will increase the perceived punch of the kick and snare and keep it snappy no matter how busy the other drum elements get. The same approach can be used to get the kick and snare to push down all other sounds in your mix as well so they never get lost. Even just 1–3dB of gain reduction will make a noticeable difference to how clear things sound.

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brittle, cluttered and lacking in depth. Re-create the classic 80s production tone by using a high-shelf EQ on each sound in the mix and roll off enough top end so that only certain sounds occupy the very high end (cymbals, vocals and so on).

Use a low-shelf or high-pass filter to leave a clear area for your kick and bass to sit in WHAT TO DO IF A SOUND WON’T SIT IN THE MIX Sometimes you will come across a sound that just doesn’t want to sit in-line with the rest of your mix. Although reverb, modulation effects and dynamics processing can all help to change the perceived positioning of a sound in your mix’s plane of depth, EQ is the best way to move sounds back in a mix without adding other characteristics at the same time. A quick positive-gain frequency sweep with a parametric EQ set to a medium width can help you to find the frequency area of a sound that makes it instantly leap out of the speakers. Once you’ve found this area, gently tweak the width so that you are pushing all of the frequencies in this area that move the sound forward, then switch the gain from positive to negative to push the sound back with everything else.

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REMOVE LOW END WHERE POSSIBLE Like Tip 3, another EQ treatment that’s worth applying to most sounds is a low-end cut. Use a low-shelf or high-pass filter and remove anything below 100Hz or above to leave a clear area for your kick and bass instrument to sit in. The bigger the bottom end needed, the more you’ll want to remove it from other sounds to avoid conflict of frequencies, which results in a loss of headroom and, therefore, overall track volume. This also applies to effects, so get into the habit of placing an EQ after each send effect so they don’t add any unwanted rumble to your mix. An EQ placed here is also handy for shaping your send effects to fit better in the mix.

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HAAS EFFECT This is a simple yet effective trick that you can use to widen a mono sound in your mix. All you need is a delay added as an insert effect with its feedback set to minimum so that only one delay is created. Set both sides to different times in MS between 1–30ms.

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CHECK FOR PROCESSING DELAYS Although your DAW’s delay compensation should keep your audio signals in time, you can sometimes listen to your work towards the end of your track and wonder why it sounds less tight and defined. Certain plug-ins and configurations can cause delays that either accumulate as your effects chains increase, or will be bad enough to put a whole channel slightly behind the rest of your mix. A good trick to check for this is to occasionally flick on your click track to see if

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Be careful not to go too high as they will start to ‘flam’ when the mix is folded to mono. This will trick the ear into hearing this sound in stereo, but a downside is that you will get a comb-filtering effect when the signal is in mono. One way to minimise this is to use a filtered delay along with the original sound in mono. With the delays on the left and right being filtered so that they contain just midrange information, there’s enough information to create a sense of width but less frequency content to cause comb-filtering problems when summed to mono. focus Mixing 2013

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MTF 25 Pro Tips Mix processing

sure to make a habit of turning it off after a brief listen or you’ll soon start to get used to the mastered sound and potentially dislike your own.

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USING REFERENCE MATERIAL Whenever you’re in the process of mixing, have a few reference tracks to hand to keep your ears grounded. As well as taking regular breaks every hour or so, having reference tracks inside your project that can be muted and then solo’ed gives you a quick reminder of factors such as the amount of low, mid and high end you want in your mix, or how heavy your compression is sounding against the reference material and so on. It’s easy to get used to how much you’ve pushed your EQ or squashed your tracks, so listening to some reference points will prevent you from making damaging decisions because you’ve lost your perspective during a session.

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REVERB WIDENING There are two ways to quickly enhance the stereo width of a reverb used on an auxiliary send, which are particularly useful if you don’t get the width you’re after from plug-in settings alone. Firstly, use two reverbs on aux sends and have one panned hard left and the other hard right, then tweak the reverbs separately to give you a much more varied and therefore wide-sounding ambience. You can always pull back the hard panning a little so that they spill to the opposite side a bit. The other approach is to add a mid-and-side plug-in after the reverb so you can lower the mid signal level to make the side signal more dominant and therefore wide-sounding.

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PRESERVE DYNAMICS LIKE A PRO Too much compression can flatline any energy there may be in a sound source. As an alternative to using it as an insert effect to process the entire signal, try to work in parallel, so that the dynamics of the original signal can be heard alongside your compressed version. This also enables you to apply heavier compression than normal to the parallel signal, giving extra energy and volume. If it’s an option, use your compressor’s dry/wet dial to blend between the original and the compressed version. Alternatively, use auxiliary sends to an effect channel, where the compressor will process a parallel copy of the original sound.

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EQ ALL BUT THE VOCALS Along with the sidechaining options mentioned in Tip 6, a good trick for achieving a clear vocal in a busy mix is to group everything but the lead vocal(s) and use subtractive EQ on this group to reduce frequency areas that compete with the vocal line. If it proves difficult to determine where to make your frequency reductions on the group, try boosting areas of the vocal first to discover what makes it stick out most (see Tip 4), then take a note of the EQ’s settings and copy them to an EQ on the group, reducing these areas so that the vocal cuts through the mix better.

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TRIAL MASTERING There is an argument for and against having anything on your mix buss, so be your own judge after trialling this Tip. However, having a basic mastering setup that you can punch in and out occasionally can be very useful for gauging how your mix levels will change after some overall compression and limiting is applied at the mastering stage. This is where exploring presets can help you to get something of a ball-park overview of a ‘once-mastered’ version of your mix, but be

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A TOUCH OF COMPRESSION FREQUENTLY Instead of applying compression to just your drum buss (or even just piling it on at the mastering stage), try to apply less compression but more often at various points along the signal path. A little compression can be applied as you record external sources; you can add some to each sound source’s channel in the mix and a little more to a sub-group, then possibly more still on the mix buss. Compression is a lot more transparent when used in this way, and using different settings and flavours of compression will bring more and more tone to the mix, giving a richer sound.

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FAVOURITE PROCESSING CHAINS Spare yourself the time it takes to set up your everyday mix processing requirements by creating a template. This can include favourite auxiliary effects such as two reverbs for short and long ambiences, two delays with different timings, and a few parallel compression and distortion send effects as well. If you use certain processing chains for insert duties, save these chains in a folder for recall later, too.

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JUDGE FROM VARIOUS OPTIONS It’s always best to make educated decisions when it comes to deciding how you should process certain sounds in your mix, and by using parallel processing (or features such as Audio Racks in Live) you can take a channel, apply a

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Mix processing 25 Pro Tips MTF

DON’T OVERLOOK THE BASICS If you’re struggling to comfortably sit all of the elements into a busy mix, don’t be afraid to either remove some of them or heavily thin them out using EQ to get the space you need. Given the unlimited track counts of DAWs, there can often be too many tracks to deal with, and as a mix engineer you should sometimes remove or lighten the presence of sounds that don’t need to be in the forefront of a mix (or there at all). Remember that mastering will pull the levels of your sounds closer together, so if things are sounding incredibly clogged at the mix stage it may be problematic for the mastering engineer, too.

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EXPLORE PRESETS Although you may instinctively start to tweak a plug-in’s parameters from their default loaded state, don’t overlook the power and speed of working with presets. A lot of companies have taken the time to provide a highly usable range of settings that can often do the job straightaway (or at least give you a good starting point that you can then refine). Mixing as you produce is particularly useful for workflow and can give your track a pro sound while avoiding the momentum-crushing power of over-tweaking.

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A large bathroom is always good for capturing resonantsounding reverb certain processing approach, then mute it and start afresh, processing the same sound but with different settings or processors on the next channel. Audition each attempt at processing afterwards in the context of the mix. DYNAMIC MIXING If you have a track with lots of different dynamics, be prepared to apply different processing settings or treatments to individual sections – for example, the compression amount and its timing settings of attack and decay for a verse may not sound right for a chorus, and vice versa. Don’t be afraid to move audio to different channels so that you can apply different processing treatments to different sections of a song.

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AUDIO EDITS OVER GATING Although gating is a useful quick-fix for removing sections of unwanted lower-level noise, given the ease with which audio editing can be carried out these days it’s hard to find an excuse for not just quickly editing out any long passages of unwanted noise. You can even use a DAW’s automatic audio-chopping tools to separate these signals from louder parts in the audio, then use fades to make each edit sound right for the particular source.

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INCREASING MID DENSITY If you need a sound to cut through the mix a lot more in the midrange, try using some distortion in parallel and limiting its frequency range using low- and high-shelf EQ so that only the mids pass through. Much like Tip 2, you’ll get a slightly different sound depending on whether you add the EQ pre- or post-distortion. This can also be useful to have set up as an effect send in general so that you can increase the midrange presence of an entire mix when required. Just send different amounts from different instruments to add more midrange, while at the same time preserving each sound’s original tone.

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GATE TRANSIENT SHAPING Gates can be great for re-building a drum sound when resources or time are in short supply. If you run a parallel copy of a drum sound through a gate, you can set it to let through only the initial transient and suddenly cut off any tails. Then you can balance this signal’s level with the original to get more snap. Explore the possibilities of adding EQ to this gated signal so that you focus on the frequency area that gives the most snap and single it out.

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AUDIO INTERFACE PROCESSING A simple yet often overlooked processing trick is to use your audio interface’s preamps to reprocess a signal being passed from its outputs. Send out a signal (such as your drum buss), wire the outputs to the inputs, and find the sweet spot of analogue clipping while recording it back in. This is also a good technique for creating a smashed, distorted parallel signal.

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COMPRESSION FOR DRUM SHAPING Another approach to re-shaping a drum sound to achieve more snap is to apply extreme compression. With a high ratio and reasonably low threshold, set the release to 5–10ms and the attack to its minimum, then slowly increase the attack until you hear the transient slip through the compression. When this happens you can adjust the release to make sure that it’s long enough to prevent any strange changes in volume to the drum’s tail, but that the release isn’t so long that the compressor isn’t ready to act the next time it plays. MTF

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REAL AMBIENCE Although reverb rooms and chambers used to be a necessity, real ambience is still very hard to emulate in the digital world, and you probably have a room somewhere in your house or studio that you can use for this. A large bathroom is always good for capturing resonant-sounding reverb; a furnished room can be useful for the type of ambience that makes a difference but is hard to detect until it’s removed. Place a full-range speaker in the room, send your sound to be processed out to it, and experiment with mic patterns and positioning for the best results.

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MTF Walkthrough Composing with sampled instruments


How to work with the

Download content www.musictech mag.co.uk/mtm/ workshops

sampled orchestra

There’s no longer any reason for your musical ambitions to be restricted by the size of your studio or available instrumentation. Keith Gemmell explains why.

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Know the score

e’ve come a long way since the days when working with sampled instruments meant coaxing convincing sounds from a General MIDI sound module. Many of the sequencing techniques involved still apply, of course, but the samples themselves sound so much better. Although there’s still a long way to go, you could say that sampled instruments have come of age. Most come with hundreds if not thousands of individual samples covering the commonly used performance articulations of a given instrument or ensemble. Even so, creating a convincing performance from such a wealth of raw material is a daunting prospect for many musicians, particularly if they are unfamiliar with the instruments. In this Workshop we take you through the process of composing a piece of music using sampled instruments. You can download the audio and project files from the MTM website. Lasting just over a minute, it’s typical background music and just three instrument groups are used: strings, brass and percussion. It’s the kind of music

Most musicians working in popular music contend very well with the difficulties associated with sampled instruments. They usually play guitar, keys or drums and are familiar with the sounds and playing techniques of each. For them, sequencing these instruments within a DAW presents few problems. They know the score. Problems arise, though, when, like all of us, they step into an unfamiliar musical genre. This might entail adding sampled strings or a horn section to an existing pop track. Maybe they have been asked to write music for an indie film and an authentic-sounding orchestra is required. The key to working successfully with sampled instruments is quite simple. Get to know the instruments well. Study the playing techniques. This will help enormously when you’re confronted with a bewildering set of articulation choices. Study instruments’ playing ranges and musical sweet spots – how their timbre changes as they change registers. Only then will you know where and when it’s best to use them.

MTF Navigation Key tools (Albion II) a KEYSWITCHES Any keyswitched articulations that you record live or enter in your DAW’s piano roll can usually be viewed here, in your sample player. Audition the various articulations on offer by mouse-pressing these keys.

that’s often composed in a hurry – as was this – and for that reason we took a broad-brush approach and initially sketched out the piece using Albion II, which is ideal for this type of work. A similar library is Symphobia, popular with media composers who work to tight deadlines.

MIC POSITIONING If your sample library was recorded with several different microphone positions, their controls will appear in its GUI. Experiment with the different types, combinations and volume levels to vary an instrument’s sound.

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ARTICULATIONS Avoid playing just sustained notes on your sampled instruments and experiment with the articulations. They are there to help you produce a convincing, realistic performance.

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DYNAMICS Dynamics are important when playing sampled instruments. Use your controller to subtly vary long notes for authenticity. Use them, too, for adding deliberate dynamic and emotional expression.

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Composing with sampled instruments Walkthrough MTF

MTF Step-by-Step The string section

Let’s start with a moody string ostinato. Normally, we would play this manually, but as we’re using Albion II we can automate the process with the Ostinatum tool. Five notes are assigned to low strings. After drawing them as a chord we define their note length (16ths), velocities and playing order in the Ostinatum. A short articulation is selected and the pattern is repeated and varied throughout the piece.

After selecting a pizzicato articulation, we now play a bass line. To contrast the rhythmically precise ostinato, avoid quantization. If your library provides mic positions, for clarity, try keeping the close mic high in the mix. For more definition, experiment with compression. Here we use the excellent VSL Compressor with the ‘Basses Pizz – distinctive attacks‘ preset as a starting point (Attack 10ms, Release 225ms, Ratio 1.80:1).

Just before bar 9, the high strings play a soaring, slow-moving line. Not wanting anything too thick, we chose a half-section (one player tacit at each desk) for a transparent sound. Play these types of melodies live if you can. To breathe life into them it’s vital to vary the dynamics, which in many string libraries is assigned to CC1 and controlled using your keyboard’s mod wheel.

In step 3 we divided the high strings. Now the second half-section plays an inner line, triggering a legato patch. Overlapped notes will assist with realism, particularly if your samples were not recorded with legato transitions. The dynamic fade-down on the last note of the phrase is another realistic touch. Note, too, the addition of an ambient mic position, which further spreads the sound.

Between bars 17 and 21 the high strings play a series of staccato notes. To create a ’rise and fall’ dynamic curve using velocity as you play isn’t easy. Instead we used Albion’s CC1 Mapped Velocity feature, which enables you to control note velocity with the mod wheel. Key velocity is overridden so you can play as hard or as soft as you like or simply mouse the notes in.

Our half-section violins have a beautiful sound quality, but when mixed with the rest of the instruments are a little too prominent. Lowering the volume is not the answer. EQ is. By lowering the gain in the 3–4k frequency range they sound less direct. For the final chord, keyswitches are used to trigger tremolando and long articulations for the high and low strings respectively.

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Listen to the best examples that you can find, both as solo instruments and as part of an ensemble. At one time this kind of information would cost you a fortune in books and recordings, these days it’s freely available online. Read books on orchestration, even if you don’t read music (Gary Garritan’s website has a free orchestration course). Once you’re thoroughly familiar with the instruments you intend emulating, working with them will be much easier. Don’t over-concern yourself now with EQ, compression and mixing. Knowing how to write for and play sampled instruments is more important. Get the compositional and orchestration balance right and any frequency range worries and mixing problems will mostly disappear.

PRO TIP Many sample libraries assign the mod wheel to dynamic expression, and if you’re playing monophonic parts manually you should usually use it (particularly for shaping slow-moving melodies). Without it, your strings, brass and woodwind parts will sound lifeless by comparison. Sometimes, though, the mod wheel is assigned to vibrato, in which case you could set up a foot pedal for dynamics.

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Broadly speaking, orchestral sample libraries come in two types. Type 1 is complete recordings of orchestral sections such as strings, brass, woodwinds and percussion. These are further broken down into smaller ensembles – violins 1 & 2, violas, cellos and basses, for example. Solo instruments are often included, too. Type 2 consists mainly of ensembles. For example, Albion’s ensembles are recorded in situ, in a greatsounding hall, so you don’t have to worry about stereo placement and messing with convolution reverbs. What it may lack in detail, though, is made up for with a rich out-of-the-box sound and articulations carefully selected for cinematic-style composition.

Which library?

String fellows

Musicians with ambitions to write media music frequently ask which libraries they should buy. Putting synths, pianos, guitars, electric basses and drum kits aside, we’ll stick to orchestral instruments here because our brief for this Workshop was to write solely for strings, brass and percussion.

String players are probably the busiest session players in the music industry. With strings used so often for film and media work it’s no surprise that they are always in demand. They’re superb readers and will play just about anything you give them, no matter how complex. Working with sampled strings, though, can be tricky. focus Mixing 2013

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MTF Walkthrough Composing with sampled instruments

Slow attack times – the initial sound is heard a little later than it should – are a common problem. The solution is to set a negative decay time in your DAW’s track inspector. Staccato patches can sound too long. Unlike with synthesized strings, altering their lengths in a DAW’s piano-roll editor doesn’t affect the raw sample. However, some libraries, like VSL, offer time-stretching facilities in their sample players to remedy this. Writing convincingly for strings also requires a sound knowledge of music theory. Arrangements can be sketched on paper or in a notation program like Sibelius and recorded afterwards in a DAW. Another method is to export a MIDI file from Sibelius and import it into your DAW. From there you can edit and humanise it further. However, for non-readers, libraries such as Albion and Symphobia provide a broad-brush approach to string writing. Albion II’s strings, for example, are divided into high and low sections. Using these, convincing string melodies can easily be realised. Some libraries contain useful sequencing tools, such as Albion’s Ostinatum. We

PRO TIP Keyswitches are notes outside an instrument’s playing range that are used to replace one articulation with another. For a smooth transition, place them a few ticks before the change (they also show up as notes in score editors; if you intend printing parts for real musicians, place them on a separate track).

used this to kick-start our composition. It’s a form of pattern sequencing and in most cases wouldn’t work with strings. With a staccato patch, though, it does. The robotic quality provided the necessary sense of urgency. This technique is used frequently behind film chase scenes. String sections will often divide into two parts. This is known as divisi. Until recently, sample libraries supplied only full sections and if you wanted divisi you had to use the same patch twice, effectively doubling the sound not dividing it. As composers demanded ever more realism, developers began including divisi sections with libraries. In our Workshop composition two high string divisi patches were used. The result is very authentic because when these samples were recorded, one player at each desk remained tacit.

Brass facts If strings are the mainstay of cinematic orchestral music, brass comes a close second. Low brass in particular is prominently featured when strength and power needs to

MTF Step-by-Step The brass section

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Trombones from VSL provide the brass stabs here. Each part was played separately. The four notes of each chord start and end in different places, which is how real players perform. Our strings are from Albion II and come with ‘baked in’ reverb, but the VSL trombone patches are supplied dry, so we treated them with the Vienna Instruments PRO built-in reverb.

Apart from the last chord, all the trombone notes use staccato patches. However, some notes were too short-sounding (this is sometimes a problem with short string articulations too). To remedy this we lengthened them by approximately 0.3 seconds using Vienna Instruments PRO’s time-stretching facility. You can also alter the speed of pre-recorded string runs this way to make them fit a particular tempo.

As mentioned in step 1, the trombone parts were played manually without quantization so humanisation is unnecessary, at least as far as timing goes. However, we loaded a preset tuning curve into VI PRO’s Humanization area. Each time a note is played, a slightly different tuning curve is used. ‘In tune fast’ is a good one for staccato notes because the curve affects only the very beginning of each note.

the last chord a sforzarto patch was 04 For needed, where the notes are played with

To get the sforzarto patch to sustain for longer we used the Vienna Instruments PRO crossfade feature. By dragging a trombone sustain patch into the slot next to the sfz slot (top-right), the first patch crossfaded into the second patch, giving us the required two bars of sustain. Balancing the two patches is done using a slider (bottom-left).

Because of their stirring quality, four horns or more in unison are often used in film and media soundtracks. At bar 19, four horns (an ensemble patch) take over from the trombones and play a short descending melody. Note the expression fade-downs at the end of phrases and, like the trombones, the drawn sforzarto on the last note.

a sudden, strong emphasis. We also wanted the notes to suddenly drop in volume after the initial attack and then rise again over the two bars. So we simply drew the expression (CC11) into the Cubase controller lane. But we met a problem: the sfz patch was too short and died away.

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Composing with sampled instruments Walkthrough MTF

be projected. In our piece, four trombones play a series of closely voiced stabs over a low string ostinato. Close voicing isn’t generally recommended this low down and both instrument sections occupy a similar frequency range. However, because the trombone notes are short and sharp, for effect, the instruments project well and cut through the busy, fast-moving background. Dynamics sometimes present problems with sampled brass. As brass instruments increase in volume their timbre changes. Cross-fading is one way to simulate this. Our Workshop piece uses this technique on the final chord. Another way of creating a realistic-sounding brass crescendo is to assign a controller to both a low-pass filter and CC 11(expression). As the volume changes so does the timbre, which gets brighter.

Percussive adventures For film soundtracks, thunderous cinematic percussion effects are the norm, but usually, orchestral percussion is most effective when used sparingly. One way to avoid

overuse is to write for just two players – one for the timpani and another for the rest of the percussion. It’s also in most cases a good idea to write the percussion parts last. It’s rarely used to drive the music. Instead it’s used to provide colour, drama and interest that complements the other instrumental parts. It’s also the easiest section to emulate with samples and presents fewer problems in construction. In our Workshop piece, to position our percussion instruments in the mix we first rendered them to audio tracks. We then placed a pre-fader insert on each track and bussed them to an auxiliary track containing a reverb plug-in set to full-on wet. After balancing, we then grouped them together as a group track. To balance the percussion along with the strings and brass, we zeroed the group track level and auditioned the reverb output only (the wet sound on the auxiliary track, in other words). We then raised the group track to an appropriate level. (Download the Workshop demo track from www.musictechmag.co.uk/mtm/workshops) MTF

MTF Step-by-Step Adding percussion

When working with samples – for orchestral compositions anyway – it’s often best to leave the percussion until last. This leaves you free to accentuate and highlight specific events on the other tracks. As a rule of thumb, use it sparingly. Here, for example the timpani shadow the low pizzicato strings at important points. They don’t, of course, play the 16th-note runs, which would sound rather ridiculous.

The timpani samples here have a long decay that complements the short pizzicato string notes that they highlight. Most of the notes, too, overlap slightly, making the most of that decay. To add more punch to the notes a compressor was employed with attack and release times of 5ms and 150ms respectively. With a low threshold and a ratio of 4:1 a fair amount of attenuation was achieved.

Initially, the music occupies the lower frequencies. Because the timpani are used sparingly, there is room between their notes for something higher – castanets, perhaps. They occupy a higher frequency, catch the listener’s attention and maintain interest. When the trombone stabs enter, though, a percussion instrument with a sustained sound is required, for contrast. A triangle is the answer and is struck once every two bars, until the horns come in.

Once the horn line appears, musically, there is a lot going on. The high strings, for example, are very busy with 16th notes. The piece needs rhythmic percussion at this point. However, drums might prove too heavyweight; the timpani have already paused. Something light is needed, akin to a conventional drum kit hi-hat. A tambourine is more in keeping with the orchestral flavour and proved the best choice.

The percussion for this piece is now complete. However, after playing the track through it was obvious that something percussive was needed on the last chord, to accompany the timpani roll. Gongs were tried but they were too low-pitched and drowned out the timpani. A cymbal crash provided the ideal solution. Loud, brash and with a long enough sustain to cover the two bars, although, of course, it gradually fades.

All of the percussion came from the Vienna Special Edition and, like all the VSL samples, was recorded without reverb. So to mix it with the Albion II patches, the percussion was routed pre-fader to an algorithmic reverb with a church hall preset. Why pre-fader? It helps provide the illusion of a more realistic space.

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MTF Workshop Working with loops

Cubase 6.5 Workshop

Working with loops

On the & online

www.musictech mag.co.uk/mtm/ workshops

Tips & tecnhiques

Loops might be most commonly associated with Ableton Live, but Cubase is perfectly capable of dealing with them too, as Tim Hallas explains.

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hen it came to writing this Workshop I found myself in something of a quandary, having spent a day in the very pleasant company of the Steinberg crew previewing version 7 of Cubase, which of course has now been launched. So, what can I cover that will still be relevant in the new version? My decision is to take a look at the loops and looping techniques available in Cubase. For those of you thinking, ‘Oh no, not loops! That’s not music!’ I have to remind you that loops are here to stay, so it’s surely better to know how and when to use them than to dismiss them out of hand. Loops have been around much longer than we remember: early electronic pioneers used tape loops to create sound worlds and to manipulate and mangle electronic sounds repeatedly. And we also need to remember that Stayin’ Alive by the Bee Gees utilised drum loops as far back as 1977. Following the tragic passing of the drummer mid-session, the band turned to drum machines but didn’t like them. They then listened to the

already recorded drum part for Night Fever, recorded two bars of it to tape and looped it. The result is one of the most famous grooves of all time.

Finding your feet

PRO TIP When working with loops in Cubase make sure that the Snap control is set correctly, as due to the default zoom level of the Arrange window it is very easy to end up with loops out-of-sync with the bar. To adjust the snap, switch it on in the top toolbar and set the precision level to Bar, Beat or to link with the Quantise grid for smaller intervals. As most loops will be multiples of bars, I tend to find that setting Snap to Bar is the best option.

Electronic pioneers used tape loops to manipulate and mangle electronic sounds repeatedly

The loop feature in Cubase is arguably not as welladvertised as it is in some other software; nonetheless, it is definitely there and probably the best place to start is with the Loop Browser, which can be found by choosing Media>Loop Browser. 1 When opened, this brings up a large dialog that looks very similar to the MediaBay window (they are essentially accessing the information in the same way). The Loop Browser is split into three main sections: the category selection section at the top; the list of available loops that fulfil those category requirements in the middle; and a preview panel at the bottom containing basic transport controls and a graphic of the waveform of the loop. 2 We’ll begin at the top and find the loops we want. The categories run roughly left-to-right, with instrument groups on the far left followed by subcategories for specific instruments, genre, character and, finally, key on the right. A feature in this that I particularly like – and one that isn’t readily found in other DAW software – is the counter next to each category that tells you how many loops have that tag in the database – for instance, there are 633 loops in the key of C but only 158 in C#/Db. This constantly updates as selections are made and provides you with a live figure of how many loops are left with that tag rather than a static number (you may suddenly find that you have too many tags selected and there are none left with all the categories you have selected).

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The Loop Browser is split into three main sections: the category selection section at the top; the list of available loops that fulfil those category requirements in the middle; and a preview panel at the bottom.

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Working with loops Workshop MTF

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Filter loops via the buttons at the top (3), which enable you to select only MIDI or audio loops and sort by key, tempo or name.

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Audio loops within Cubase can be treated with its time-stretching and pitch-quantising tools As it is probably the most common use for loops we will begin with a drum loop, perhaps to use for the beginning of a dance track or – how I most often use them – as a placeholder during arrangements until the ‘real’ drummer turns up for the recording session. I’ve selected Drum&Perc to narrow down the options to just percussion and then also selected Beats>Pop/Rock to get the potential candidates down to a manageable number of files to audition. 3 Once you have all your preferred categories selected you can audition them to find the right loop. To do this, move down to the main list in the middle of the window and click on individual loops. As you do this the loop will automatically play back and a preview of the waveform will appear in the section at the bottom of the screen. There is also the facility to filter loops via the buttons at the top, which enable you to select only MIDI or audio loops and sort by key, tempo or name. You can also filter by ratings, but as I haven’t given any of my loops a rating – a loop might merit five stars for one project but be totally useless for another – this feature seems somewhat superfluous. Once the file is in the preview window you are presented with basic transport controls for playing back the raw loop. The all-important Cycle control is also present, enabling you to hear how well the loop links back into itself. On the right-hand side of the preview live two little controls which are often overlooked: the Align Beats to Project button and the Wait for Project Play button. The former changes the playback speed of the previewed loop to that of the project; the latter will play only the preview and the project at the same time. However, if you want to hear the previewed loop at the same time as the project and in time with the project you need to press both buttons. This might sound rather long-winded, but, on reflection, if you have a tempo-free loop you might not want the speed changed, so having the controls separate works well in this respect. 4

PRO TIP If you want more physical interaction with your loops you can drag and drop them into Groove Agent One, Cubase’s drum synth, and use the pad feature for triggering the loops. If you use Cubase in the live arena or like to record your loop-mangling via drum pads this is a great way to get interaction in a different way.

Finally, once you have selected your loop you can simply drag-and-drop it into the Arrange window. Alternatively, right/[Ctrl]-click on the selected loop and you are presented with several options for inserting it into the project at various locations.

Chop and change After loading your loop into the Arrange window – either by manually dragging it in or by specific placing directly from the Loop Browser – you may want to do some additional editing to alter the way in which it works within the project. Obviously, the easiest and most commonly performed task is to loop the file, as that is the purpose of a loop! To do this, click on Edit> Functions>Repeat ([Ctrl]/[Cmd]+[K]) 5 and select the number of repeats you want. This is slightly different from those DAWs that have a simple drag system, but it works perfectly adequately nonetheless. The difference here to other loop copies stored by other DAWs is that each copy is just that, a copy, and can be edited totally separately from the others. Unlike most pre-programmed audio loops, those within Cubase can be treated with all of the software’s time-stretching and pitch-quantising tools. 6 To access these windows, double-click on the loop you want to edit and it will open in the Sample Editor.

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MTF Workshop Working with loops

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Edit your loop as necessary via the Key Editor and tidy up any timing imperfections via Quantisation (Q).

All of the usual tools available for audio editing will be available for you to use on your loop. However, as already mentioned, if you have made any alterations to your original loop and repeated it across your project you will need to re-do the changes, as they won’t be copied over to the repeat instances.

Beyond pre-programmed If you are bored with pre-programmed loops but want to store some of your own for accessing later, Cubase enables you to do this with MIDI very simply, though doing it with audio is not so easily achievable. To create a MIDI loop, start with a MIDI or Instrument track and record the part or pattern that you want to store. Once you have done this, edit it as necessary via the Key Editor 7 and tidy up any timing imperfections via Quantisation (Q). This is important when creating a loop for the obvious reason that it will be repeating over and over and the timing needs to be perfect in order to loop back into itself accurately every time – even a minor glitch will eventually become noticeable. Once the loop is in its finished state click on File>Export>MIDI Loop. 8 This opens up a dialogue box that enables you to name the new loop and associate relevant search tags with it. You can see from my example that I have used bass and electronic tags for finding it in the future. 9 Once everything has been named and labelled correctly simply press OK and the loop will be added to the loop library. To find it again, re-open the Loop

Browser and you’ll see that just above the main list of loops is a search panel: if you search for your new loop it should appear every time you need it in a project.

Mash it up A looping tool that Cubase advertises more prominently is LoopMash, an instrument for the live manipulation and deconstruction of multiple loops simultaneously that can be used to create totally new sounds. It has good integration with the Loop Browser and supports drag-and-drop between the different tracks of the LoopMash instrument. 10 For more on LoopMash, see the article in MTM 106, Jan 2012, and use the techniques described in combination with those here and create your own loops to use with LoopMash.

On the make Exploring the loop-based features of Cubase 6.5 is something I strongly encourage you to do, even if you don’t use loops in much of your everyday music-making. I find the ability to quickly load up a loop to play along to when tracking in the studio – even if it’s replaced later on – invaluable. For those of you who create music in electronic genres, listen to some of the more experimental sounds for inspiration and then adapt them for your own purposes. And since this article is about loops we need to get back to my first point: Cubase 6.5 has now been replaced by Cubase 7, so let’s see what wonderful music we can make in preparation for the new incarnation of the software. MTF

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LoopMash is an instrument for the live manipulation and deconstruction of multiple loops simultaneously that can be used to create totally new sounds.

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MTF 10MM Surround sound monitoring

Minute Master

Surround sound monitoring If you want to break into surround sound, you’ll need to understand where and why you should put your monitors. Russ Hepworth-Sawyer is in the sweet spot.

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elieve it or not, surround sound has been with us for a great deal longer than you might think. During World War 2, a cinematic version of surround sound – Fantasound – was being developed at a rate of knots by Disney, employing a rear stereo pair to the front stereo much in the way we’re used to with today’s surround sound setups. Shortly afterwards, Pierre Schaeffer, of Musique Concrète fame, was playing around with spatial sound in performances using his Potentiometer D’espace in the early 1950s, much in the same was as electro-acoustic composers exploit ‘diffusion’ today. Schaeffer’s technique included a speaker behind the audience and one above, in addition to two at the front. The sound could then be manually placed around this soundstage as part of a ‘performance’. Due to its sheer excitement – and I dare say permanent installations – cinema has found enduring reasons to sustain and develop surround sound, with technology such as Dolby Surround enabling fourchannel audio to be decoded from two tracks. Later developments included Dolby Digital. Meanwhile, the music world failed to find a sustaining surround sound solution despite many near-miss attempts based around quadraphonic (quad, for short) in the 1970s. Early attempts at surround sound music such as Quad required extremely specialist and

Dolby Stereo Leftt

Dolby Digital 5.1

Centre

Right

Leftt

Left rround Surround

Centre

SSub LFE R Right

Righ g Right Surro Surround

Whereas Dolby Stereo offered only four channels of surround, the current 5.1 Dolby Digital format uses a full six channels for the various speakers.

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expensive equipment and media, which hampered the widespread consumer adoption of the format. One version even employed a matrix technique to squeeze four channels of audio into two channels on vinyl, making it backwards-compatible. In the 1990s, some clever solutions were developed permitting those listeners placed correctly in the stereo field (or wearing headphones) to literally hear 3D sound through the use of specific algorithms. By modelling the delay characteristics of a signal from one ear to the other it is possible to position sound in a wider space than the two speakers available to you. The main one of note is QSound, and a few albums (such as Roger Waters’ Amused To Death) demonstrate how sounds could be placed in a 3D space. Another system was Roland Sound Space (RSS). Although an

It has become tradition to refer to multi-channel systems in terms of the number of speakers utilised

Rear

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A controller such as SPL’s 2489 (around £550) enables you to switch between different 5.1 sources (a DVD and your DAW, for example).

expensive and not widely adopted standalone system, the RSS research later became distilled into effects algorithms that found their way into some of Roland’s 1990s 19-inch rack processors. Despite a number of prominent albums being mixed in QSound to emulate three-dimensional space, this format was not universally adopted, and by the mid-90s it seems we were back concentrating on the plain old stereo CD. The next step was to face the ‘update’ question: should we look to high-definition formats or remain with the status quo? Developments for higher-resolution audio led to the notion of stereo 192kHz and 24-bit sound as a stereo benchmark, perhaps on Super Audio CD (SACD) or a lower-resolution surround sound signal. The DVD, already popular for video, led the way to the DVD-A (the ‘A’ standing for audio). Being literally a Digital Versatile Disc (DVD) meant that standards could be extremely flexible and permit all manner of formats to coincide on the same disc. One of these, of course, was six-channel surround sound.

Making the point Over the years it has become tradition to refer to multi-channel audio systems in terms of the number of speakers utilised. A standard stereo set of monitors

Surround sound monitoring 10MM MTF

would therefore be considered as a 2.0 setup, while a stereo satellite monitor system with a subwoofer would be considered a ‘.1’ (‘point one’). Quad would have been a 4.0 system, while the standard surround sound system today is a six-channel system known as 5.1 – Left, Centre, Right, Left Surround, Right Surround and Low Frequency Effect (LFE). Other, more elaborate, systems do exist, and 7.1 is a fairly common expansion (as does 7.2, whereby two subwoofers are utilised alongside two extra side speakers). Taking things even further is 10.2, developed by Tomlinson Holman, the man behind the THX standard (a quality mark for sound reproduction within cinemas). In 10.2, the standard 5.1 system is retained, except the two subwoofers are placed at 90˚ left and right of the listener. Additionally, two full-range speakers are placed further behind the standard left and right speakers to provide a sense of depth.

Specialist monitor controllers such as the Audient ASP510 help to manage unwieldy volumes when working in surround sound.

Setting standards In the case of traditional stereo, the correct placement of the listener in relation to the speakers is still quite misunderstood in the average living room, but there is less likelihood for error than with a surround system. Naturally, many homes were not designed with surround

Task master

In the case of stereo, placement of the listener in relation to the speakers is still misunderstood sound in mind: a door situated in the corner of a room, at the ideal point where the left surround speaker should be; a fireplace where ideally the centre speaker ought to be... Such obstacles cause issues with regard to the optimum surround sound listening position. The industry standard recommended by the International Telecommunication Union (ITU) suggests that each of the five main speakers ought to be full-range, and that the subwoofer should be employed only as an LFE channel. The ITU standard also

-30º

00º

+30º

Tech Terms

110º -110º

+110 +110º

The ITU standard specifies the placement angles of the five monitors in a surround sound setup.

recommends that the speakers be placed in a field approximately 1.2 metres from the ground and surrounding the listener at certain angles. At 0˚ at the front is the centre speaker. To either side of this (at -30˚ and +30˚ respectively) are the front left and front right speakers. The rear speakers are ideally placed at 110˚ to either side of the listener at the same distance away as the front speakers. The ‘point 1’ LFE subwoofer should be placed in the front plane, ideally.

● LOW FREQUENCY EFFECT The term used for the subwoofer in surround sound setups. An LFE is not usually a means of compensating for poor performing monitors, but for film directors to place some cinema-shocking rumble for impact. ● MATRIX A matrix permits for more than two channels to be encoded in a stereo channel by use of a high-frequency modulator signal.

Once you’ve set up your studio with a 5.1 solution, it is time to consider the wider implications. One significant challenge is how to control simple tasks such as the volume – but for six discrete channels! Products such as the Audient ASP510 Surround Sound Controller make this experience seamless, as it handles the six audio signals with the one volume controller. What’s more, these solutions provide you with the options you’ve come to expect from a stereo monitor controller, and specific features you’d require when working in surround. Should you be experimenting with 5.1 without a specialist monitor controller – by perhaps connecting your monitors directly to the outputs of your audio interface – be sure that the monitor levels are down and that the output configurations are set up as you might expect. Connecting the standard stereo left and right to an audio interface would most likely output using channels 1 and 2. However, output 2 in many 5.1 surround systems would be reserved for the centre speaker, with 1 being left and 3 being right. In the 1970s it was clear why widespread adoption of a surround sound format for music was likely to be curtailed by the technology and expense. Despite significant attempts to get music surround sound back on the agenda in recent decades, nothing has taken over from stereo. Some might say that’s the iPod culture and convenience, while others still believe in the quote from George Massenburg’s AES Keynote Speech in 1997: “Stereo. What a concept! Hey, two ears? Two microphones and two speakers! I think the expression that my kids use is, ‘Duh, dad’.” (www.aes.org/ events/103/). MTF FURTHER INFO ● For ITU recommendations for multi-channel sound, see: http://tinyurl.com/d4ld64d ● For more information on Audient’s monitor controller for 5.1: www.audient.com/products/asp510-surroundsound-controller focus Mixing 2013

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MTF Workshop Understanding gain-staging

Pro Tools 10 Workshop

Getting to grips with gain staging in PT10 Setting up your signal chain correctly before you start recording will make for a smoother, less frustrating mixing process. Mike Hillier explains the rules.

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roper gain-staging is a key concept within audio engineering, yet it’s one that time and again is misunderstood within the digital world. In the ‘good old days’ of analogue recording engineers would have to properly gain-stage their recording equipment to guarantee a good signal-to-noise ratio without distortion. Drop the signal too low and you risk raising the noise floor in your signal path; raise the signal too high and you run out of operating headroom and the signal will begin to distort. Once the signal has reached the digital domain the noise floor issues aren’t really important anymore. Even entry-level A-to-D converters have a signal-to-noise ratio of over 100dB and with 24-bit recording offering a dynamic range around 144dB (the new 32-bit floating point format in Pro Tools 10 brings this up to a staggering 1,680dB dynamic range) 1 the noise floor becomes almost forgettable. This wasn’t always the case: early digital equipment had a very high noise floor, so users had to run equipment as close to 0dBFS as possible without clipping, leaving very little headroom.

PRO TIP A brickwall limiter such as Maxim on the stereo output buss can be used to bring up the signal level while also ensuring that clipping doesn’t occur. Maxim employs automatic make-up gain, so as you reduce the threshold, gain is applied to bring the signal up to the ceiling. Leave a little room in the ceiling by setting this parameter to around -1dB.

However, despite the extra dynamic range of modern digital equipment, many users still try to record as close to 0dBFS as possible, leaving themselves very little headroom before harsh digital clipping occurs. The first rule to good gain-staging in Pro Tools (and, in fact, any DAW) is to record at 24-bit and give yourself sufficient headroom in your initial recording. 0VU on an analogue system equates to around -18dBFS on most A-D converters and there are many engineers who prefer to use this figure as a guide when recording in Pro Tools. The meters might not look like they’re doing much, but at -18dBFS you’re giving yourself a good amount of headroom and you’re still well above the noise floor. There’s no reason to stick exactly to this figure, though; we usually find ourselves aiming for around -12dBFS. 2 The important factor is to not push the recording too close to 0dBFS. This is especially important when recording singers and acoustic instruments without a compressor in the analogue signal path, as the signal can suddenly leap up – and if you’ve given yourself 12–18dB of headroom, the recording will most likely not clip if it does.

Gain in the channel strip

Many users still record as close to 0dBFS as possible, leaving themselves very little headroom

Once the signal is inside Pro Tools there is no excuse for clipping. Any clips that occur after this point can and should be fixed as you go along, unless you are deliberately degrading audio for sound-design reasons. Understanding where each gain change occurs is key to ensuring you maintain a solid gain structure throughout your signal path. In the case of an audio channel the gain is first determined by the clip (previously known as a region in Pro Tools 9 – a name change which is going to make discussing clipping quite

1

3 2 Recording with sufficient headroom in the first place is the best way of ensuring that you maintain good gain structure throughout your mix.

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Understanding gain-staging Workshop MTF

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The Trim plug-in and clip-based gain can be used to bring down the level of any overly hot signal before it hits any processing.

Leave plenty of headroom in the audio signal or risk pushing the audio into digital distortion complicated!) itself, which then goes to the inserts and, finally, to the fader. This means that any gain changes made by an insert effect cannot be fixed by simply pulling down the fader. You can test this very quickly by applying a Trim insert 3 to a clip and pushing up the level until it distorts. Once it’s distorting from the Trim plug-in, pull down the fader and the signal will still be distorting. In Pro Tools 9 and earlier versions the Trim plug-in was an important tool for gain-staging. Placed in the first Insert slot, it can be used to bring down the level of a signal before additional processing, ensuring that even a hot recording was given some headroom. In Pro Tools 10 the Trim plug-in is still an option, but the new Clip Gain feature lets you dial down the level of the clip directly on the clip itself. This gain occurs before the inserts, and not only does it therefore perform a similar function to the Trim plug-in without using up one of your Insert slots, it also provides visual feedback as the waveform shrinks and grows relative to the clip gain level. What’s more, it can be used to quickly alter sections of clips individually, or multiple clips across multiple channels simultaneously. Simply highlight the clips (or parts of a clip) you want alter and press [Shift]+[Win]+[Up] (PC) or [Shift]+[Ctrl]+[Up] (Mac) to raise the gain; [Shift]+[Win]+[Down] (PC) or [Shift]+[Ctrl]+[Down] (Mac) to reduce the gain. 4 This is especially useful for vocals, when you would normally ride the fader to smooth out a performance. Now the performance can be smoothed before any processing, requiring less compression and making it easier to set the compressor threshold. To view the clip gain line ensure View>Clip>Clip Gain Line is enabled. 5

PRO TIP The TL MasterMeter lets you view not only any clipping on your audio, but also whether the audio will produce what is known as inter-sample clipping, which occurs when the re-created analogue signal goes above 0dBFS even if none of the samples themselves actually does. This is just one of the reasons it is advised to leave a little headroom, even on your final masters.

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can push a hot signal into clipping. This may appear counter-intuitive since filtering should reduce the overall signal level, but in reality an EQ will actually boost slightly around the edge of a cut, which in some circumstances can be enough to push a signal into clipping (watch out for this with high-pass filters 6 , which are commonly used to clean out noise below the fundamental but can also sometimes boost the fundamental frequency slightly). If you’ve recorded with sufficient headroom or used the Trim plug-in or Clip Gain feature to give the signal headroom, you will be able to boost or cut with an EQ without worrying about introducing any digital clipping distortion. Of course, now that your signal is lower you will notice that any level-dependent plug-ins – such as dynamics processors or saturation effects – will be behaving differently. However, any level-dependent plug-in is also likely to have its own level controls, so it should be possible to control the level going into and from the plug-in directly from the plug-in itself. If you feel that you aren’t getting sufficient level to get the necessary effect, simply boost the level going

Keeping headroom between inserts If you don’t leave yourself plenty of headroom in the audio signal before it reaches the Inserts there is a risk that you could push the audio into digital distortion through processing. Even something as simple as applying a high-pass filter or an EQ notch to a signal focus Mixing 2013

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MTF Workshop Understanding gain-staging

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Sending a signal with sufficient headroom to any processing will give the process more room to work its magic on the signal without creating digital clipping.

into the plug-in, but counterbalance this by reducing the level coming out of the plug-in so that your overall level remains in the -12dB to -18dB range. The Bomb Factory BF-76 compressor 7 , for example, doesn’t have a threshold control to set the level at which the compression kicks in, but it does have an input knob that can be dialled up to achieve the necessary compression characteristics, as well as an output knob to control the level coming out of the BF76 and into the next plug-in in the chain. Leaving this kind of headroom will ensure that you find it much easier to work, especially when EQ’ing, as you should have more than sufficient room to make large boosts without running the risk of pushing the signal too high. Additionally, when working on large projects with lots of busses you won’t risk peaking the signal when summing two or more channels together.

The one place where you want to see the meters starting to move is the final stereo output buss

Headroom and the summing buss Aux and instrument channels have a similar gain structure to audio channels, although an aux channel’s basic level is determined by the level of signal coming out of the buss, while an instrument channel’s basic level is determined by the plug-in instrument in the first Insert slot. Master channels, however, have an important difference: the fader on a master channel comes before any Inserts and, importantly, it also comes before any summing, so you can use a master channel to prevent a buss from clipping (not that we would recommend this – it’s better practice to turn

PRO TIP Many MP3 encoders now work best when delivered a 24-bit file. This means that you don’t always have to bounce down a dithered 16-bit version, except when burning to a CD. Some online tools, such as SoundCloud – which is built-in to Pro Tools 10 – even enable you to upload audio in 24-bit WAV format.

down the individual channels going into the buss). As well as the master summing buss, this is useful for any sub-groups or send effects you might have created, as they can be turned down before they reach the aux channel. What’s more, it is useful when using outboard equipment as it enables you to add the outboard as an Insert on the master channel and control the level going to the outboard equipment using the fader, ensuring you work closer to your gear’s optimal input level and you’re not constantly reaching for pads. 8 The one place where you do want to see the meters really starting to move is the final stereo output buss. You still don’t want to clip the signal at all, but you can afford to have the peaks much closer to 0dBFS than you would want on any of the individual channels or sub-groups. If you’re producing the final master you want to get the signal as loud as possible, although leaving a little headroom – around 1dB – can improve the quality of any MP3 encoding that might happen to the signal further down the line. If you’re just producing a mix, which will be mastered later on, leaving a little more room is fine. The mastering engineer will also want some headroom to apply EQ and gain changes. MTF

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The Bomb Factory BF76 doesn’t have a threshold control to set the level at which the compression kicks in, but it does have an input knob that can be dialled up to achieve the necessary compression characteristics.

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MTF 25 Pro Tips Drum sound design

Pro tips for

drum sound design

From big beats to a minimalist drum section, there are numerous techniques you can draw on when it comes to creating your own drum sound. Liam O’Mullane shares his favourites. LAYERING CONSIDERATIONS One of the best ways to create a unique drum sound is through layering – but this needs an organised approach (avoid just piling up similar sounds and thinking it sounds better because the output is louder). A more productive approach is to designate layers for specific roles. For instance, you might have a sub layer for the low rumble of a kick drum, but then add a knock or splat-like sound to take care of the kick’s transients. A snare drum can consist of a lower-weight layer for the 180–240Hz area of the sound. This is the physical part of it, but you’ll also need something in the midrange for the bite, and a sharper element for the initial transient. A third layer is then usually needed for air and ambience. This can be a live-sounding hit or a hit with reverb applied, but the main treatment needed for this layer is a high-pass filter to remove all but the top end. When combined, these layers alone can be the basis of your drum sound.

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AUDIO ON THE TRACK VS TRIGGERING VIA MIDI If you plan on sequencing drum sounds (ie, not record a drummer playing a kit live) you’ll need to decide on whether to work with MIDI or audio samples on an audio track. Both have their merits, but you don’t have to choose strictly one or the other. Triggering drum samples from a drum machine or sampler via MIDI has obvious advantages as this approach is ideal for live performance and facilitates interaction with a controller. You can then render the MIDI parts as audio later on when you want to commit parts and move on. Working with audio has a disadvantage in that it’s not as easy to simply swap out already sequenced sounds like it is with a MIDI sampler. However, working with audio on a track has a certain

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tangible feel to it. The advantage of being able to see the duration of each sound over time is also very handy when programming and producing tight drum sounds. REMOVE EXCESS FREQUENCIES Nearly all untreated drums will need a certain amount of frequency trimming, but the amount will depend on the size of drum sound you’re after. This is a decision that directly

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Drum sound design 25 Pro Tips MTF

time. As different parts of the sample move in and out of phase you can determine which positions work best for the sound you’re creating.

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relates to the space you’re giving them in the context of your overall mix. A live drum kit will need a lot of low end removing from most microphones to eliminate rumble and lowerfrequency mud. In the case of overhead or cymbal mics, the amount of low–upper-mids to remove can be quite significant. All sounds need their own space to live in, and depending on how much the sounds overlap each other, you may have to remove more frequency content to prevent the overall sound from becoming too dense.

Strategically removing sounds will help to retain a certain level of sonic definition

SOUND MASKING If you’ve got a lot of drum parts playing various rhythms in a programmed pattern, try muting one sound or another when they play at the same time. This can help to reduce sound masking, which happens when too many sounds play at the same time and cover similar frequencies. Your pattern can still sound busy if that’s what you want, but strategically removing sounds when too many double up will help to retain a certain level of sonic definition.

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ADSR AND ENVELOPES If you’re using a sampler you’ll be able to shape the amp Attack, Decay, Sustain and Release of each sample. The attack stage is often overlooked, but it can turn an overly bright or brittle drum hit into something a lot more rounded and fat-sounding. Just slowly increase the attack time until the tip of the drum transient is softened. If you need to create a tighter, snappier sound, use the decay time while the sustain value is set most or fully at its minimum. This lets the decay control how long a much louder transient is heard for. The release time is useful for a natural drum sound, giving a natural decay at longer values. However, a low release time will let you control the playing duration per drum sound by the length of a MIDI note triggering it, enabling you to program shorter stutter-like sounds one minute, then longer, more natural-sounding hits the next.

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EXAGGERATING PEAKS Any decent-quality drum sound sample will have one or more resonant frequency peaks. These can be easiest to spot using a spectral analyser. Certain styles, such as the low-end 180Hz thud on modern heavy electro snares, require you to exaggerate these peaks. A general low-end boost with an EQ will make the lower frequency louder but also potentially make the snare too dominating in the bottom end of the mix. Instead use a parametric EQ set to a narrow width (Q) and then boost the specific frequency of the resonant peak.

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MORE LAYERING CONSIDERATIONS In contrast to the strategic layering illustrated in Tip 1, simply putting two or more random samples together will in most cases yield some less-than-pleasant results. But by applying a few phase-based tricks you can make the most out of them. First pay attention to the direction of each layer’s phase. Phase can be flipped via your DAW’s mixer channels, by a plug-in, or by choosing ‘flip phase’ from the audio-editing options. Assess which phase direction sounds better (or worse!) as you introduce each new layer. Next explore slightly offsetting each layer by tiny movements in

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GATE OR EDIT If you’re working with a multitrack drum project it’s easiest to set a gate to mute any unwanted bleed in a recording. This is the quickest approach but it’s hard when the dynamics and speed that the drums are being played at varies throughout a track. Editing the hits individually is time-consuming, but it gives you a lot more control over where the mutes occur and you can use envelopes to fade things in and out smoothly.

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IMPROVING TRANSIENTS WITH COMPRESSION A compressor has a variety of uses when it comes to drum sound-shaping. This Tip involves applying extreme compression with a high ratio and medium threshold settings. Without any make-up gain enabled, this will make the drums very quiet. Raise the attack time to deliberately let a slither of full-volume drums pass through uncompressed. This will add a new transient to the drums; adjusting the attack time will let you shape this transient differently.

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TRIALLING DIFFERENT SOUNDS It should go without saying that it’s a good idea to trial different sounds together when first creating your drum sound. Sample choice or kit choice (for a real drummer) can have a big impact on how suitable a drum sound is for a given track. With sample-based drums, don’t be afraid to swap out sounds at the mixdown stage. As your mix gets busier

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MTF 25 Pro Tips Drum sound design

the decay time with a low sustain and attack settings. Experiment to find the right amount of modulation and the most suitable decay time. SYNTHESISING RIDE CYMBALS A synthesised ride cymbal is very different in texture from a real cymbal, but its tone has become iconic over the years (with a big helping hand from the Roland TR-909). All you need is an FM-based synth to get the required metallic tones. Start with a square wave oscillator pitched up from middle C by 24 semitones, then set the FM modulating oscillator to 44 semitones and start applying the modulation amount to the first oscillator until you start to hear a bell-like sound. Next bring in two more square waves and set their pitches to sound harmonically musical. Detune them to get their tones to pulse against one another, then shape the transient of the ride using a low sustain, long release, short attack and short decay time on the amplitude envelope.

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UTILISING SAMPLE TRIGGERING If you’re working with recorded drum tracks but need a more rigid level of intensity from certain sounds like the kick and snare, drum replacement is worth exploring to either augment or replace these sounds. Logic, Pro Tools and Cubase all offer drum replacement tools, and third-party options are available as plug-ins. This is the simplest way to get consistent power from your drums with minimal effort in terms of sound production and editing.

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through its evolution, you may find that changing your kick and snare choices at the end will transform your final mix for the better. GATE SHAPING Gates can be used for drum-shaping in a way that always sounds different from the other methods discussed. Set the threshold high enough so that the gate starts to open with each hit of the drum. Then you can explore using a quick attack and no hold time for a sharp start to a drum, with a ramped-down fade that is controlled via the release or decay parameter. The other option is to set a short release time and use the hold time to create a classic ‘gated’ effect with an on-and-off volume change.

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WORKING WITH GROOVES Although a drum groove won’t have a direct impact on the production sound for drums, the correct use of velocity when programming is certainly a large contributing factor to the finished production sound. If you use accenting to make different drum elements sound louder when playing together, you can further imply the desired rhythm and feel, making those particular parts in the beat sound much more powerful. The timing and feel of a groove is also worth exploring as a well-produced drum track might still sound like it’s lacking something if the timing is sterile. Without paying attention to the groove, you might unnecessarily start to mess with your drum sound when it’s really not the cause of any problems.

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Drum replacement is worth exploring to augment or replace kick or snare sounds

REVERSE SOUNDS FOR MORE VIBE A great way to get a push-and-push feel from your drums is to add some reverse sounds before they are played forwards. A tried-and-tested way of doing this is to add a reversed snare before it plays forward. But less is more, so try adding one before the fourth snare when played on the second and fourth beats per bar (the second snare in the second bar). This will give a nice push to the end of the drum pattern. You can add to this

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SYNTHESISING KICKS AND TOMS The kick and the toms are the simplest of drum sounds to synthesize. Any synthesizer that lets you modulate the oscillator’s pitch with an envelope can be used for this job. This might be hardwired as a pitch envelope, or you may have to manually route this modulation. After choosing a subtle tone – like a sine or triangle wave for the oscillator – create a very fast attack shape on the pitch envelope. With the sustain down, use the decay to shape an almost instant pitch drop. Now explore how strong this modulation needs to be to get the kick sound you’re after. The pitch at which you play the oscillator is the main difference between this sound being a kick or a tom (also explore setting longer decay times for higher tom sounds).

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WORKING WITH FOUND SOUNDS If you want to create a truly unique-sounding collection of drums, grab a portable recorder or run a microphone from your audio interface and get into the kitchen. There are so many potential sounds with the character of plastic, wood and metal. These can be hit with a wooden spoon, a metal spatula and so on to generate more tones. Shakers and rattles can be created by putting dried foods such as rice and pasta into bowls and pans as well.

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PITCH MODULATION TO SAMPLES A short but sweet trick to really get snap from your drum samples is to use a pitch envelope on samples. In the same way as synthesising kicks, create a rapid pitch descent using

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Drum sound design 25 Pro Tips MTF

CHOOSING THE RIGHT REVERB Your choice of reverb can vary widely. If you want a super-tight and upfront sound from a live recording, you might want to add a tiny amount of a small room setting. But the ambience in the recording (even in the driest of rooms) might already be enough. Larger reverbs have their place on drums if there’s space between each hit so the reverb can be heard. The stereo width of a reverb is also important. If you want a wide drum sound, a wide-sounding reverb will enhance your overall sound, but in many cases the drums might need to be in the centre of the mix so that other instruments can have space at the sides. In this instance you’ll either want to go for a mono reverb, or use a width tool to control how wide the reverb is.

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WORKING INTO DISTORTION Distortion has its uses when crafting a drum sound. You can apply a tube-like saturation to a drum group for added volume, glue and intensity as newer harmonics are added and the dynamics are reduced. Another approach is to apply distortion to a group in parallel using a more extreme, hard-clipping distortion. Much like parallel compression, this will added a constant level of background information, increasing the perceived energy in your drums.

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feature by minimising other sounds while the reversed sound plays. This adds further focus to the moment and makes the returning forward snare all the more dramatic. GROUP PROCESSING When you pass all of your drum elements through a group channel you get much more than just control over their volume as a whole. You can add compression to get a much more gelled drum sound, or apply a transparent EQ to bring out important information as your mix starts to get busy. Pick a musical EQ, something like a Pulteq can give a very professional finish to a drum sound, adding elements of saturation for extra harmonic content.

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GROUP PROCESSING 20 PARALLEL Running a parallel channel from a group can work wonders, increasing the energy and density of a drum sound. A classic trick is to smash the parallel channel with a very hard compressor like a Urei 1176. With the attack and release at low settings, nothing slips through this compressor, meaning that you can really slam a signal through it. The output will be a flat and very noisy sound that you mix in below the original for a constant level of energy. SYNTHESISING SNARES The process of making a snare drum from scratch starts in much the same way as it does for a kick (Tip 12). With a subtle amount of pitch modulation and a higher oscillator pitch in general, you can create the sound of the top skin of the snare. The snare rattle needs emulating, which you can do with a noise oscillator. You’ll need to give it a dedicated amp envelope as it needs to sound for a short period of time for the attack portion, then stop while the top skin resonates. A high-pass or band-pass on the noise oscillator can help to add bite to the attack stage; passing the whole sound through a distortion will give more of an aggressive sound.

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DELAYS AND PROGRAMMING As delays have specific repeats after a sound they aren’t a great effect for adding ambience to drums. But since a delay is really a constant rhythm, they can be interesting to

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A classic trick is to smash the parallel channel with a very hard compressor like a Urei 1176 interact with. Try adding a delay to a percussion part and programming a few hits, then experiment with delay times until something really jumps out at you. For extra funk, set the delay time to MS instead of quantized values and go for around the same timing as before, but add a loose feel by not setting it precisely in time. TREATING YOUR CREATION AS NEW LOOPS Many of you may have played with drum loops in the past and also sliced them up to re-edit their order and create unnatural silences between hits and so on. If you’ve created a drum beat that sounds great in terms of production but isn’t doing anything for you in your current composition, try treating it like any other drum loop. Render it to a mono or stereo audio file, then slice it up and program something new out of it. Turning your drums into a new format to interact with can really help you to move forward with ideas when you get stuck. MTF

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MTF Technique Digital mixing & production

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Digital mixing & production Technique MTF

MTF Masterclass Studio Technique

DIGITAL MIXING and production Producing a track from start to finish with only the minimum of outboard gear is now well within the capabilities of anyone with a fairly recent computer. Hollin Jones illustrates the back-to-basics approach.

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ver the past decade or so, the music technology landscape has changed almost beyond recognition. As in so many other areas of life, the computer has become an inescapable part of the way people work every day. This is all the more remarkable in the world of music, where hardware had ruled since it first became possible to record and mix sound. And of course, hardware still exists and is still used, but we have started to see a new phenomenon, whereby entire productions are created ‘in-the-box’, meaning that from recording through to the finished product, sounds do not leave the domain of the computer. So how has this happened, and how did we get to a point where not only was it possible to produce in-the-box, it was also a preferred option for some? In the past, the sound of records was determined not only by the instruments used to perform them, but also the equipment used to record them. This is still true of course, but there used to be a much larger possible combination of components involved in making a finished recording than you necessarily have to use today. Imagine 30 years ago going into a studio to make a track. The sound you ended up with would be affected to a great extent by the model of mixing desk used, the particular outboard compressors, EQs and reverb units, the cabling and preamps and, of course, the recording medium. Different sizes of tape had different characteristics, as did tapes running at different speeds. The ‘sound’ of a particular studio could be one of its greatest assets and to an extent still is – just look at the virtual instruments that are still created using sessions recorded at Abbey Road or other famous locations.

In with the new All of this began to change with the advent of digital technology, which has none of the inherent ‘flaws’ or quirks that lend analogue gear its character. If you record a CD to a cassette tape, you lose quality. If you duplicate an AIFF file on your computer, it’s a perfect copy. The first stages of digital’s takeover of recording came with digital tape, then, later on, hard disk-based recording, which would capture what you fed it without adding anything at all in terms of colour. It had other positive attributes, too, like far higher recording and track

Special thanks to Digital Village, Bristol, for loan of the Euphonix MC Control and the Focusrite Liquid Channel. www.dv247.com

We have started to see a new phenomenon: entire productions created ‘in-the-box’

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count capacity, ease of backup and duplication. Eventually, of course, this all filtered through to the more affordable end of the music market in the form of powerful and capacious home computers that had the ability to multitrack digital audio and, eventually, run software instruments and effect plug-ins. An important thing to remember is that although for years people spoke of the possibility of producing entirely in-the-box, stock plug-ins and affordable audio interfaces were no real match for professional outboard gear. Even computer-centric dance acts used loads of outboard simply because it sounded better. Now, though, things are rather different. It is genuinely possible, with the right approach and a little knowhow, to make radio-ready tracks using just your computer for recording, processing and mixing. In fact, it’s much more common than you might think, and listening to radio shows that play new music reveals what excellent results some people are getting without ever going into a commercial studio. There are some really compelling reasons to go ‘in-the-box’, not least of which are the money that it saves you on paying for studio time, and the space you save by not having racks of outboard cluttering up what is in all probability a spare bedroom. You can be portable if you’re using a laptop, and work at your own pace. Plug-ins are excellent these days and some carefully chosen ones can do the job of costly outboard processors; best of all, you can load up as many instances of a synth or a reverb as you like, which isn’t possible with physical kit. There are rules to this as well, of course. Simply deciding to do everything in-house won’t make your recordings sound great, you have to do a little homework. Understanding the part that each component in your setup plays is vital, so there’s no

Tech Terms ● IN-THE-BOX A broad description used to describe producing music entirely inside a computer using digital audio recording, MIDI sequencing and virtual effect plug-ins to process sound once it has been recorded. Mixing and mastering typically also take place inside-the-box. ● OUTBOARD A term used to describe audio processing and mixing hardware, usually rackmounted effects units of one sort or another. It can also refer to virtually anything non-virtual, such as mixing desks, drum machines or mastering processors.

The quality and character of your original sounds are hugely important things to get right

Use analogue emulation plug-ins like Steinberg’s Portico EQ and Compressor to get the sound of analogue gear inside your DAW.

point, for example, in having Logic Pro running top-notch Waves plug-ins, but recording through your computer’s audio-in jack. Or recording your guitars too low then having to boost their volume in your DAW, which can also amplify the background noise that will almost certainly exist somewhere in the signal. Gain structure, bit-rates and an optimum project setup are vital, and that’s before you even get to mixing. The good news, however, is that it isn’t as hard as it sounds.

Start at the source As we have mentioned, digital systems record exactly what you feed them, so the quality and character of your original sounds are hugely important to get right. Before you even consider recording, assess what you are going to be recording. If you are lucky enough to have a live drum kit, this should be tuned and the mics placed optimally for the kind of sound you want. Guitars and basses should be in tune and their pickups clean, and any cables in good order. Vocals should always be recorded with a pop shield and preferably some sort of portable isolation unit to create a manageable space for the sound and avoid reflections from walls. Whenever you mic anything up, try to use an appropriate model of mic and experiment with its placement in relation to amps or instruments in

MTF Step-by-Step Processing a vocal

With your vocal recorded, try applying a little compression. Use gentle settings to start with and adjust the threshold, knee and ratio so that any peaks are being caught but the vocal doesn’t start to sound squashed. Starting with a preset based on the vocal type can be helpful.

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Using either the channel strip or a dedicated EQ plug-in, adjust the EQ curve for the vocal. Vocals always need some EQ, whether it’s to cut some of the bass off a male vocal to make it sit in the mix better, or enhance the upper mids of a female vocal to add shine.

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Vocals can benefit from some careful gating, so that when the noise level drops below a defined threshold, the sound cuts out. This gets rid of background noise and bleed, but be careful not to have the gating come in or drop out too violently.

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Artist Q&A

Grammy Award– winning engineer S Husky Höskulds.

Husky

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ith 15 years’ experience in both big LA studios and his home setup recording such artists as Sheryl Crow, Norah Jones, Elvis Costello and Mike Patton, Husky now chooses to work fully in-the-box. He tells us why. What led you to an ‘in-the-box’ production setup? Was it a gradual process or did you make a conscious decision to move away from hardware? I suppose it was somewhat gradual, even if it was conscious. I’d been mixing analogue for years – API console, 24-track 2-inch, 2-track 1/2-inch – but wanted to expand a bit into a wider range of colours, so to speak. I’d always used rooms, PAs, guitar pedals and the like while mixing, and in a way as an extension of that I started integrating plug-ins into the workflow.

‘non-linear this and that’ than even looking at their monitoring setup. If you haven’t created a level playing field before you start mixing in-the-box you will be chasing your tail forever. I’ve spent hours testing D–A converters, digital cables, analogue interconnects, speaker cables, volume pots, fine-tuning my system... and it’s hugely important. People forget that in established studios this is the foundation of it all, and it’s all been done way ahead of time by professionals.

Was there one bit of kit that marked a turning point for you in your move towards working in-the-box? One of the game-changers was Altiverb. I never liked the digital reverbs back in the day, and Audio Ease brought a whole new palette to the table. I would also

The most important thing about this ‘mixing in-the-box’ business is your monitoring

Can you offer any EQ and compression tips that are specific to working in-the-box? Well, the thing about being in-the-box, the beauty of it, is that you have all your tools available to you at all times. No more ‘I used the good compressor on the vocal, so I can’t use it on the mix buss’. That leads to some great flexibility.

use Logic to drive MIDI effects in my rack as well as for automating plug-ins and even tracks when I’d run out of faders on the API but needed some panning effect or similar. Having the plug-ins as part of the workflow was like having an extra effects rack, and that was a great inspiration. It certainly got me thinking. What would you say are the most important things you gain by going in-the-box? More flexibility? Lower costs? Portability? Well, yes, those are all the obvious ones of course. For example, I have a studio in LA and another in Iceland – exact copies – and I take the computer with me from one to the other. That’s not an option in the analogue world. What tips would you offer to anyone considering this approach? The most important thing about this ‘mixing in-the-box’ business is your monitoring. Hands down. I’d say 95% of people working in-the-box spend more time and money worrying about ‘tape emulation plug-ins’ and

How much time do you spend shaping your sounds at source as opposed to using plug-ins and processing post-recording? If I’m recording the stuff myself I try to get it right at the source, of course. This holds true no matter what the destination format is. Of course, sometimes you are forced to work fast, and you tend to let things go for the sake of speed, perhaps knowing that in the digital world you can fix it, but for the most part I try to get it right on the day of recording.

Getting acquainted with Apple’s Logic Pro was a key factor behind Husky’s decision to start working in-the-box.

What do you think is the next big thing for mixing in-the-box? What do you want to see coming soon? Well, I would like to see the computer form factor start to change a bit. With solid-state drives now there’s the potential for a lot smaller and much quieter machines – and that would be great to see. Having a Mac Pro or PC tower seems awfully outdated these days by comparison to what’s going on in the iOS world. I’d love to have something the size of a Mac Mini – no fan, no moving parts, fiber optic (Thunderbolt, I suppose) to HD monitors and some super-SSD RAID system. You can check out Husky’s artist collective at www.groundlift.org focus Mixing 2013

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MTF Pro Technique Getting depth One of the biggest challenges when you work in-the-box is to create a sense of space and depth using recordings that generally have either been recorded through a mic in a bedroom or generated using software instruments and thus have never actually existed in a real space. The most obvious answer is to use reverb, but this carries its own issues. For a start, a lot of simple reverb plug-ins that might come with basic DAW versions are less than stellar. They might be serviceable, but apply more than a touch of the effect and they will start to sound washed out. Another problem is that as you add reverb to multiple tracks, the overall effect can be one of making the whole track sound too spacey. A good solution to this is convolution reverbs, which offer control not only over the size and shape of the reverb, but also let you EQ the reflections and the damping plus much more. The impulses tend also to be much more believable. Using short amounts of delay rather than reverb (especially on vocals) is another trick that can add depth without adding space at the top end of your tracks. If you want to get really involved, Audio Ease makes Speakerphone, a remarkable plug-in that can emulate just about any kind of physical environment, including placing sounds inside containers far away from the listener. Another way to add depth, as we have noted, is to use samples that already have some ambience on them. A sampled drum loop or guitar riff with some great room ambience, for example, can underpin your sound, add character and mean that making your other sources a little more dry won’t be a problem. Finally, warming effects like the vintage tape or amp simulators we have looked at can add a cohesive depth to your overall sound by Convolution reverbs are the best option ‘glueing’ everything together. for extra depth and space in the digital domain.

A well set-up room will help you to achieve professionalsounding mixes order to get the best sound you can. You will find much more in-depth guides to recording specific instruments in the MTM archives (see http:// musictechmag.co.uk/mtm/features/microphonescatching-waves as an example). The reason this is important is that getting the best possible sound at source will save time later as there will be less tweaking to do. Software is great, but try to make less work for yourself by capturing a tone that’s as close to perfect as you can manage. Plug-ins and EQ will then only have part of the work to do later. A rotten vocal sound or a weedy bass might not be recoverable, even after the use of plug-ins, so pay attention to the sound before you even consider hitting record. The quality of your audio interface also plays a big part in the sound you get, as do the mics you use. Skimping on any particular part of your recording setup can negate the benefits brought by quality components elsewhere, so get a decent mic that’s suitable for what you’re doing, and an interface with good preamps and as many inputs as you think you’re going to need. More advanced interfaces have

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Tech Terms ● SAMPLE-BASED INSTRUMENTS Virtual instruments that generate sound using sampled audio. These are usually replications of ‘real’ instruments like pianos, drums, orchestral sounds or other things that are much easier to create by recording short samples of real sound than through synthesis alone. ● VIRTUAL MICS Increasingly, advanced virtual instruments can give you the option to move virtual microphones around your virtual sound source, providing much more control over the character of the sound. This is often more control than you would have even in a real physical studio.

Software-based mastering systems like iZotope’s Ozone can give that all-important radio-ready finish to your productions.

features that can really make working in-the-box easier, since you may well not be using an external mixing desk. Look for features such as direct monitoring, zero-latency performance, onboard DSP-powered effects (reverb, compression and EQ) and flexible, software-based audio routing. All of these will give you more recording flexibility without needing a desk, and can be found variously on models from Focusrite, MOTU and on Yamaha’s n-series. Your listening environment is also crucial to get right, and a well set-up room will help you to achieve professional-sounding mixes.

Sampled sounds It may be, of course, that you are only doing a little real-world recording using microphones or DI’ed sources, or working entirely using sampled audio and MIDI-triggered sounds. In this case, your focus might be less on the sound that’s being fired into your computer and more on the kinds of sounds you are loading up inside your DAW. Let’s start by looking at samples and sample-based instruments. The great thing about samples is that if they are from a commercial source like Loopmasters, they will almost certainly have been recorded using precisely the kinds of outboard gear that you have chosen not to use. As such, you can incorporate the sound of specific kinds of hardware into your tracks without ever having to actually use that hardware. Some samples producers make a point of explaining exactly how their loops were recorded and what gear was used in the process, so this is an excellent way to ‘cheat’, as it were, without really cheating. This isn’t a new idea of course: a lot of early hip hop and jungle music gained its character via sampled drum breaks lifted from old funk records.

Use stereo panning and widening with plug-ins to add a sense of physical space and interest to your tracks.


Virtual instruments have also come on leaps and bounds in recent years, and even if your average soft synth doesn’t expend much energy on trying to emulate physical spaces, there are plenty of other instruments that do. These tend to be sample-based, and so are often drum, piano, percussion or orchestral in nature, with Native Instruments’ Kontakt-based instruments and Toontrack’s drums being good examples. Some virtual instruments now give you the option to ‘place’ sounds in virtual physical spaces, moving microphones relative to the source and switching mic models. These are usually stunningly effective, and in many cases enable you to control the amount of bleed between drum mics around a virtual kit – something you can’t do in a real studio. It’s also simple to flip between amplifier models, or control the ‘age’ of the sound digitally. Virtual effects are also far more capable than they were, a notable example being convolution reverbs, which are able to accurately mimic real physical spaces such as aircraft hangars, churches or phone booths. Altiverb is a good example, and most leading DAWs now come with a decent model of their own to help you to re-create a believable sense of space. Other plug-ins increasingly either model analoguestyle operation or are switchable between digital and analogue modes. Typically, digital operation has a cleaner sound, while analogue is warmer and perhaps slightly more saturated. Digital modes might make a point of not colouring the sound; rather, they simply perform their stated task of compression, EQ or similar. Analogue modes might have built-in ‘imperfections’ or softer characteristics that replicate the slightly more imprecise way in which

Tech Terms ● DIRECT MONITORING Better-quality audio interfaces have a direct monitoring feature that enables you to listen to the signal you are recording before it has been passed to the software, thus removing any latency from the equation. This can be especially important to have if you are working in-thebox and without any kind of external mixing desk. ● LISTENING ENVIRONMENT The space in which you work. The way your speakers are positioned and the shape of the room you are in as well as things like the distance of speakers from corners or walls all have an impact on the way you perceive your mix. People who mix in-the-box need to pay particular attention to this, as visiting higher-end studios is often not part of the production process.

MTF Buyer’s Guide The best plug-ins for working in-the-box VIRTUAL CONSOLE

Manufacturer Slate Digital Price $199 Getting a great analogue sound without any analogue gear is now possible thanks to plug-ins like Slate’s Virtual Console. Bringing the sound of five legendary analogue consoles to your DAW, it models both the characteristics of the audio channels and analogue audio summing and gives you the ability to mix and match consoles to perfectly tailor your sound. Web www.slatedigital.com

MIXBUS

Manufacturer Harrison Price $149 The Harrison Mixbus is actually a DAW in its own right, albeit one designed specifically with pro mixing in mind. Taking its cues from the much-loved Harrison series of consoles, it provides professionallevel features to import or record an unlimited number of audio tracks to your computer, edit them and mix them together. Web www.harrisonconsoles.com

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iZotope’s Nectar is an advanced vocal processing tool that can model microphones and physical spaces.

real analogue gear works. In digital mode, pushing a signal hard can result in clipping in some cases, whereas in analogue mode, doing the same might cause some overdrive or warm distortion but not necessarily clipping. When you choose to use either mode is down to personal choice, though sometimes it’s good to use analogue modes on digital sources like soft synths that might otherwise have a rather clinical sound, or digital mode on sounds that already have character, such as a well-recorded vocal or acoustic guitar part. Ultimately, though, you’ll get a feel for what works best on your particular setup.

Taking effect When you record or mix in-the-box, gentle compression will usually required on some tracks, or even harder compression if you’re making heavy or electronic music. If you have some experience with your setup, it’s a good idea to use track presets in your DAW, which recall a load of settings at once. So,

TUBE-TECH CLASSIC CHANNEL

Manufacturer Softube Price £207 Softube’s Tube-Tech Classic Channel offers a surgical equalizer as well as a fast buss compressor and is perfect for adding character and warmth to digitally recorded tracks. It has two Pultec-style EQs and a CL 1B compressor; the modules can be used separately or together as a channel strip. Web www.softube.com

AMPEX ATR-102 Manufacturer Universal Audio Price $349 The two-channel Ampex ATR-102 Mastering Tape Recorder has a cohesive sound and punch, plus the ability to provide subtle-todeep tape saturation and colour. A fixture in major recording and mastering studios, it is considered by many engineers to be the best-sounding tape machine for final mixdown. This emulation can provide a rich and full sound to finish off digitally recorded tracks perfectly. Web www.uaudio.com


MTF Step-by-Step Drum buss processing

Here we have recorded a live drum kit and routed the drums to a Group channel, meaning that we can process the output of the kit as a whole by applying plug-ins to the drum group channel. To leave out any particular drum element, re-route it back to the main mixer.

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Now we have applied a special tape plug-in across the drum group channel as an insert, which will process every channel that’s routed to it. We can play around with the tape speed, feedback and other controls to really warm up the sound. This effect also works well across a project’s master outputs.

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if you’ve spent hours getting a perfect vocal sound, save a preset and load it up instantly in future. Reverb, EQ and delay are other effects that can be used to sweeten sounds and add depth, even if nothing in a project has been recorded using a mic. Another trick is to use stereo panning to create the impression that sounds exist in a physical space around the listener. In fact, this is easier to achieve in-the-box than in real life, since stereo mic’ing brings its own problems. In your DAW you can duplicate and pan tracks, use stereo panning delays and stereo widening effects to add to the soundstage.

A good audio interface with flexible routing capabilities will ensure that you capture the best possible takes.

master outputs of your track or a group of tracks (for example, the drums) to colour their sound. The advantage of applying an effect to the whole mix – whether at the mixdown or mastering stages (preferably at mixdown) – is that it can be used to ‘glue’ the mix together and take digitally recorded tracks and add punch and sparkle to them collectively, not individually.

Master processing

The sounds you use and the way you process them will be key to getting a good end product The bottom line when you are considering producing and mixing in-the-box is that you probably don’t want it to sound like it’s been made purely on a computer. That doesn’t mean you have to add record hiss artificially or anything, but if you’re not careful, a track made on a computer can sound cold and lifeless compared to something that’s been made with outboard gear, at least to trained ears. As we’ve mentioned, much of this is in the planning and the kit you choose. Use good instruments, proper mic placement and quality interfaces with good preamps. Set up your DAW up for recording at 24-bit and 48 (or even 96kHz if you are comfortable with it). The sounds you use and the way you process them will be key to getting a good end product as opposed to something that sounds sterile. Analogue emulation plug-ins can help to add warmth and character to digitally recorded sounds. If you’re looking to apply subtle warmth across an entire mix, consider strapping a specialised tape plug-in across the

We have added a convolution reverb to the drum buss and dialled in a hall preset, then dropped the wet/dry mix down low so only a tiny amount of reverb is applied. You can separate off the lower end of the kit – say, the kick drum – and exclude it from the reverb if you find the results too swampy.

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Tech Terms ● CONVOLUTION REVERB Advanced and sometimes CPU-intensive reverb effects that don’t just add a standard shimmer to a sound but actually calculate reverb based on impulse responses and so are able to accurately place sources inside virtual physical spaces. Generally they are much more believable than standard reverb units. ● TAPE EFFECTS Effects that simulate the effect of playing from or recording to analogue tape. Much of the character of recordings in the pre-digital era came from their being put down onto vinyl or tape, and in the digital world it can be desirable to try to recapture and re-create this sound.

Master buss processors used to be restricted to some less than convincing tape saturation plug-ins, but now, things are different. As more people moved to ‘in-the-box’ working, they needed better ways to soften the sound of digital recordings and add character to tracks. Developers were quick to respond: plug-ins such as UAD’s Studer 800 Multichannel Tape, Slate Digital’s Virtual Console Collection, and Waves’ Kramer Master Tape among others are extremely adept at replicating the sound of classic tape machines. They can be used on individual channels and groups as well as the master buss if you want to add analogue sound to some tracks but not others. Producing music in-the-box is now more of a realistic proposition than ever. By taking care with your sources if you are recording, or choosing samples that have been recorded through quality gear in the first place, you have a good head start in terms of avoiding flat, sterile-sounding mixes. Choose plug-ins carefully and maybe employ some simulated ‘real-world’ instruments and effects that simulate recordings made in physical spaces; use analogue modes where they add feeling, and consider a little master buss compression or tape emulation at mixdown. Follow this advice and there’s no reason why your digitally produced tracks shouldn’t sound indistinguishable from studio recordings. MTF

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MTF 25 Pro Tips Bass production

Pro tips for

bass production

The bass forms the foundations of your music, and as such needs to be solid, well-formed and often expertly executed. Liam O’Mullane gets busy in the bottom end. INSTRUMENT PREPARATION If you plan to record a bass guitar, you need to check a few basic elements to get the best sound possible. New strings can make a huge difference to the tone and bite of your sound, but they’ll need time to settle in, so try to have them changed a day before you record or make sure the performer gives them a really good play if they’re changed on the day. An alternative option for the budget-conscious is to boil the strings in water to remove excess dirt and grease. Five minutes should do it, and the grease at the top of the water is a good indicator for when they’re ready. If a softer tone is required for something dubby in nature, dirty strings may work. Intonation also needs to be considered so make sure the instrument has accurate tuning up and down the neck (something that’s usually best set up by a seasoned professional). Finally, make sure all pots and switches are crackle-free, batteries are full of power (for active instruments) and that string and pickup heights are suitable for the player’s style. This will avoid unwanted clicks, bumps and thumps during a recording.

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EVENING-OUT LEVELS Although compression is designed to even-out levels, even the best device will struggle in an audible way if you try to use it to even-out drastic changes in volume. These can be caused by poor playing technique or general inconsistencies in lower-quality instruments, so try to do any major level work by hand instead so the compressor has less work to do. You can use an auto-detect function or manually split the audio into each note played, then use the volume controls for each clip to even it all out.

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PLAYING STYLES Make sure you choose a suitable playing technique for the part in question. You’ll struggle to achieve a sharp and rattly bass sound when someone is playing with their fingers; likewise, a plectrum player will sound too aggressive if you want to achieve a softer, more rounded tone. The instrument can to some extent also influence the natural playing style, so take that factor into consideration, too.

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Bass production 25 Pro Tips MTF

PERFORMANCE REPAIR Although it’s always best to get things right at source, sometimes you just have to make the best of what you’ve recorded. If you’ve captured popping noises from an overloaded DI, strings accidently smacking pickups, excessive hand and neck noise or perhaps crackles from a faulty lead, there are ways to make a rough fix. Any crackles, clicks or pops that are only momentary can usually be cleaned up with a Pencil tool in your waveform editor. You’ll have to zoom right in so you can see individual samples, then draw out the offending items so you are visually smoothing out the waveform. Excessive hand noise can be reduced by finding its main frequency area with an EQ to then reduce.

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ADDING WARMTH If you’re wanting to get a smooth and subby sound from a bass guitar recording you can try rolling off the top end with a low-pass EQ. The lower you go to eliminate the higher note information and hand noises, the more into dub territory you’ll get. Try adding compression to make the sub more consistent. Another trick is to drive the instrument after EQ through a valve-like saturation effect for some lower-order harmonics to thicken the bottom end a little more.

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SIDECHAINING External sidechained compression is often associated with using a kick drum to push back the volume of a synth pad as it plays four-to-the-floor (an obvious example being Eric Prydz’s hit Call On Me). However, it can be used to make a bass track or just the sub element be pushed down in volume when a kick plays for a more consistent bottom end. Just pass the bass through a compressor and wire the output of the kick to the external sidechain input. This will also help to achieve a louder mix as you’ll eliminate any rise in volume when the bass and kick play together.

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A bass drum microphone such as the AKG D112 is good for a boomy bottom end EVERYTHING BUT BASS One of the most powerful techniques to apply to your mixing is the removal of bass from other sounds that don’t actually require it. Although elements such as guitars or vocal lines might sound great in isolation with some thumping bottom end, this will most likely conflict with your bass parts, creating a somewhat muddy mix while also increasing the overall output of your mix in a (usually) negative way. Try to apply high-pass filters, therefore, to the majority of other sounds to leave sufficient room for your bass to live and breathe undisturbed. You can also apply this technique to groups of sounds and even create an ‘all but the kick and bass track’ group. A touch of low roll-off placed here when the mix is nearly finished can be enough to fine-tune your bass balance to perfection.

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CAB RECORDING There are two approaches to recording bass guitar. One is through a DI box for a direct sound with minimal added colour. This gives you an upfront sound and tight bottom end. If you want more ambience and tone in general, record through a cabinet. The amp along with the guitar will shape the sound quite a bit, but so will the cab position and microphone choice. The style of bass part will affect which mic you’ll use. If you want mainly lows, a bass drum mic such as the AKG D112 is good for a boomy bottom end and will handle the loudest of amps. If there’s a lot more top end like you’d capture – such as in slap bass – you need something that will capture higher information but also withstand aggressive transients. A dynamic mic like a Shure SM57 or Sennheiser MD421 are good starting points. You could try a condenser mic, but it needs to be able to handle high SPLs.

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MELODIC EQ WORK If you want to bring out the fundamental note frequencies in a bass or raise the presence of upper harmonics you’ll need different EQ positions for each note

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played so they change and track changes in the part. This can be done by automating the position of the EQ’s frequency so it boosts the correct frequency for each note. More recently, new tools like Sound Radix’s SurferEQ have removed the need for this time-intensive approach. It works much like the pitch-tracking in auto-tuning plug-ins except it moves EQ positions. Once you’ve set up the harmonic you want to boost for one note, it will track and follow the changes through the bass part. It also has multiple EQ bands so you can boost or reduce various frequencies in a musical way. focus Mixing 2013 |

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LAYERING SOUNDS As well as layering synth sounds for texture and depth, a simple trick of layering a pure sine wave or filtered triangle wave can be used to add a huge amount of solid weight to your bass parts. This is usually played an octave below higher bass parts so it sits well below the 80–100Hz region, depending on the musical key you’re working in.

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BASSLESS BASS Not everyone has huge sound systems and subwoofer setups at home or work, so it’s important to cater for your potential audience. A good tip is to start producing on small speakers such as a laptop or active PC speakers. These will help you to focus on creating a bass where you can hear its notes in the mix through a combination of compression, EQ and distortion. After, when you’re mixing on full-range speakers again, you can adjust the amount of bottom end to ensure good playback on both types of sound systems. This is essential if you want your tracks to be radio-friendly.

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DI BOXES The choice is yours whether you record a bass guitar through an amp and microphone setup or directly into your audio interface. For the latter, make sure it’s a highimpedance instrument input. If you don’t have one you can use a DI box (Direct Inject) which will match the impedance of the bass guitar and convert it into a balanced lowerimpedance signal. This can then connect to a regular XLR connection. DI boxes also have the advantage of earth-lift, which can be useful when you’re wiring your bass signal to various destinations (two amps, for instance). Phase-reversal switches are also common and these are very handy when you’re positioning microphones to be blended to a DI recording later on.

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PARALLEL RECORDING If you intend on recording both DI signals and a mic’ed cabinet, you need to be aware of phase accuracy. Although abusing the phase relationship of one signal when blended with another can be a creative tool for tonal shaping, with bass, you’ll want the best phase alignment, so neither signal has a negative affect on the other when combined. The best way to do this is to reverse the phase of one signal while listening to them combined in the control room or on headphones, then move the distance of the microphone until you get the most frequency cancellation possible. Basically, you’re trying to get the worst, thinnest sound you can. Flip the phase back again and you’ll now have the most powerful sound possible as the majority of frequencies are now reinforcing each other.

REGGAE RATTLE If you want your bass to sound like it’s hurting your speakers, try mic’ing up a bass part through a speaker while placing a piece of paper over a bass port or against the cone itself. This

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EVENING-OUT A MIDI PERFORMANCE If you’ve played in a bass track via MIDI and the sound is velocity-sensitive, you may have too much dynamic range in your bass parts (much like the approach of levelling out audio segments in Tip 2). You can do this with MIDI by making single adjustments with the Pencil tool to the velocity values; alternatively, some DAWs will have velocity data compression which, like an audio compressor, will bring the range of values closer together, giving a more consistent sound.

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FINDING YOUR BOTTOM END Those without subwoofers or near-perfect room acoustics will struggle to hear all of the notes in the sub-bass region. If you do hear them, some will be emphasised and others reduced in volume due to standing waves in your room and many potential performance issues with your speakers. The best way to minimise these factors is to invest in some studio-quality headphones, which should be able to deliver the very low notes more accurately. Other tricks are to look at the vibration of the speaker cones themselves as an indication that you’ve got the right amount of bottom happening in your mix. Finally, another aid is to look at the sub content in an analyser as this tool doesn’t suffer from interference from the real world.

creates a flapping noise as though the cone is broken, a noise which is picked up by the microphone. You can then high-pass this recording to single out the rattle and blend it with the original bass track. MONO YOUR BASS It’s important to keep the lower-end of your bass focused within your mix by having it in mono. This is so the sub information is identical in both speakers and is therefore delivered with efficiency and power. If you’ve used any stereo effects while processing your bass track, try splitting the signal into low and high and add a stereo folding plug-in to the lower track so it’s mono. Plug-ins like iZotope’s Ozone can make this a lot easier to implement.

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Bass production 25 Pro Tips MTF

will respond just to average levels and not jump to the fast attack level changes. Classic compressors such as the 1176, LA-2A and DBX160 behave in a similar way and are good for adding tone at the same time.

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BIGGER BASS SOUNDS If you want your bass to fill up more of a mix, parallel processing can be used to add distortion to a second copy of the sound without losing the punch and weight of the original. Amp simulation plug-ins are good for this. Try piling on distortion with something like a Marshall JCM800 for a high-gain sound. This will automatically lose lots of low end so it won’t cause any major conflict with the original. You can then add other effects, such as chorus, reverb and delays.

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BASS GUITAR TONE POSITIONS To get the right bass tone for the job you’ll have to contend with plenty of variables, some of which you may or may not have control over. The model of guitar is important and so is the choice of amplification and DI boxes, but the player can also vary the tone through their playing position. If you ask them to play closer to the bridge you’ll get more midrange information and a sharper tone. This helps the notes to cut through the mix. Playing closer to the neck will yield a fuller, bassier and softer tone. The notes will also ring out for longer, whereas you’ll get less sustain when you play very close to the bridge.

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SUB DESIGN As well as layering a sine wave under other bass instruments, you can create quite thicker sub sounds to stand on their own by stacking two sine waves, either an octave apart or with the higher sine set to be at a harmonic interval. A 7th above is popular for this task. You can then thicken it up even more with some slight saturation.

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PEDAL POWER There’s a huge number of guitar and bass pedals out there if you want unique and interesting sounds, but the majority are designed for guitars and not bass. This means

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To get the right bass tone for the job you’ll have to contend with plenty of variables that they can rob your sound of the lower frequencies as they haven’t been designed for bass instruments. A way around this is to split your bass signal beforehand and run a clean version to be recorded alone with all its low-end glory. You get the best of both worlds at mixdown. KICK AND BASS KEYS Although we can sidechain bass parts out of the way of a kick for a more consistent low end, a little thought at the composition stage can help your kick and bass play along in a nicer way than when they conflict. Try looking at the lower resonant peak of your kick in an analyser. None of your sub-bass should play a note in this area. Try one or all of the following: change the key of the bass part, play a harmonic or octave equivalent for the note that shares the same space as the kick or, finally, change the kick to fit the bass and melody better.

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BASS COMPRESSION Compression can be used in a variety of ways to help mould your bass tracks. To accent the attack stage of bass notes, set the compressor to peak mode, pull down the threshold and ratio so you’re hearing the most drastic compression for setting up, then set the attack stage so the attack of the bass slips through at normal volume. Time the release so you don’t get a pumping sound before the next note is played and raise the threshold and ratio to taste. For general levelling of your bass RMS settings or the various classic compression hardware or emulations are useful. RMS

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RECORDING DOUBLE BASS Double bass can be recorded in two main ways. A classical music context requires getting an overall picture of the instrument tied in with its ambience. In this situation you’ll want to go for some distant and closer microphone positioning which you can blend at mixdown. If you want a more contemporary sound, however, for styles like pop, jazz and rock, for instance, go for a closer mic position of one to three feet away to capture a tighter and more defined picture of the sound. Use the microphone like a camera: point towards the plucked area for more attack; you’ll get more bottom end and air movement if you point towards the F hole.

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REVERSE GATING We’ve already mentioned the option of using a kick to lower the volume of a bass for a more consistent bottom end in your mix using compression. But to improve a good performance (or a poor one) you can gate a bass and reverse the process by setting it to momentarily raise the bass volume in time with the kick, making the two sound more in time with each other. Just set up the kick to be the external sidechain input for the gate on the bass and raise the range level so the gate is just a fraction quieter when closed than when it’s open. Your bass will become punchier when the kick plays. MTF

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MTF Workshop FX Chains

Sonar X1 Workshop

Boost your productivity

with FX Chains

FX Chains are a powerful means of getting more out of your audio plug-ins, and in version 2.0 they’re even more flexible. Hollin Jones examines how they work.

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ne of the most useful advantages that software offers over a hardware-based music production setup is that everything is much more flexible. And nowhere is this more apparent than in the case of audio effects, which in the real world are much more limited than they are in the software domain. If you have five outboard effects units, each can be used to process one track at once – or several tracks when used as a send – but with the same preset. Alternatively, they can be chained together using a patchbay or simply connected in series using regular cabling. What you can’t do, however, is use more than one setting at a time within any single effects unit. Software totally does away with this limitation, enabling you to load as many plug-ins as your hardware can cope with. That’s a given, but in recent years, developers have started giving us yet more innovative ways of getting more utility out of the effects at our disposal. FX Chains 2.0 in Sonar X1 are an excellent example of this. They are essentially a way of chaining

PRO TIP FX Chains can have their interfaces customised by right-clicking and choosing Customize UI. You can use custom images for backgrounds and button/knob icons, as well as text colours. Use this either to make things look cool or simply to improve legibility if you have poor eyesight or need to be able to see the interface more clearly on a dark stage.

Software enables you to load as many plug-ins as your hardware can cope with

plug-ins together into presets that can be loaded, modified, controlled and saved. The idea is that if you have spent many precious hours getting a vocal processing chain just right, you save it as a Chain preset and then recall it instantly for use in another project with just a couple of clicks. What’s even better is that they all have their own interface, which helps to keep screen clutter to a minimum and enables you to assign quick controls to any parameters within any effect in the Chain – and, of course, there’s support for Active Controller Technology, too. So, let’s have a look at how FX Chains can help you to save time and achieve a better sound.

Getting started Sonar X1 comes with 100 FX Chain presets to help get you started, and these are a good way of understanding exactly how the system works. In a project, reveal an audio track’s FX bin or go to its channel inspector and right-click in the FX area. From the menu, choose Load FX Chain Preset. 1 You should now be taken to the preset folder that installed with Sonar (inside the Cakewalk Content folder). Pick a preset and open it. 2 The resulting window is the standard generic FX Chain container, but along the bottom you will see a list of the plug-ins that are contained within it. It’s a simple routing system that has an input on the left and passes audio from left to right in series, plus there are input and output level meters and volume sliders present as

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FX Chains are a great way to combine and control multiple audio effect plug-ins. Sonar X1 comes with 100 presets to get you started, grouped by category.

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FX Chains Workshop MTF

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Add your own custom controls to chains and assign them to multiple parameters on any effects within the chain for ultimate control.

If you have spent hours getting a processing chain right, save it and recall it in another project well. The Chain can be switched on or off using its power button, and double-clicking in the name field at the top enables you to rename it. 3 Plug-ins can be rearranged in the Chain by simply picking them up with the mouse and changing their order. This is important because the order of the effects can change the sound at the eventual output. If a sound passes through an EQ unit before a compressor, for example, the compressor will behave in a certain way based on the EQ’ed sound it receives. Place them the other way around, however, and the compressor will work differently, with the EQ acting on the already compressed sound. It’s not crucial in every case to arrange things in a specific way, but it’s worth understanding that this can be an issue, so experiment to hear the interaction for yourself.

Exploring the details

PRO TIP Sonar lets you manage your plug-ins using the Plug-In Manager window. To the left can be found plug-in categories, and you can also set up specific lists of plug-ins and manage these, which is handy for separating mixing, tracking and special effect plugs, for example. You can also create exclusion lists, which are good for disabling buggy or troublesome plug-ins without having to permanently delete them from your system.

For argument’s sake, try adding a knob and you will be taken to a Control Properties window, from where you can set up your quick control. You may want to begin by assigning the parameter before naming it, and if you click on the first Destination menu you will see every plug-in in the Chain, and within each one, every available parameter. 7 Try assigning the first destination to something – here we have chosen a flanger delay, and Sonar has helpfully placed the word Delay into the name field, though we are still free to change this.

User defined Since this is a knob we are also able to set the start and end values. These default to zero and 100 per cent respectively, but can be set anywhere you like between these two values. There are many reasons to do this – for example, if you wanted the knob to set a delay level of between only 15 and 75 per cent to avoid accidentally turning it all the way down or all the way up. You’ll have noticed also that there are four Destination menus available, and that’s because a single knob can be made to control up to four

To add a new effect to the Chain, right-click in the area at the base of the window and go to the Audio FX contextual menu, from where you can add a new effect. To remove an effect from the chain, right-click on it in the Chain and select Delete. 4 Double-clicking on the name of any effect will open its native GUI, giving you all the controls you are used to seeing. Here’s another interesting trick: if you right-click on a plug-in’s name in the Chain you can choose to read- and write-enable any of its automatable parameters without having to open the GUI first. 5 This is handy when you have a MIDI controller hooked up, as it enables you to more easily control what is automated or suspended from automation without having to open lots of windows. You might have noticed that the area in the centre of this particular plug-in Chain is largely empty, and that’s because it doesn’t contain any user-assigned controls, at least not yet. If you right-click in this grey expanse you will see a menu that lets you add knobs and buttons, among other tasks. 6 focus Mixing 2013

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Set ranges for controls so that you can precisely tailor how each one behaves. You can set different ranges for different destinations.

parameters on any of the effects inside a Chain at the same time, and with different Start and End values. So the knob might turn a delay all the way up or down, but also a reverb between 50 and 95 per cent, for example. The Position menu lets you arrange your quick controls within the Chain’s interface. 8 Once a parameter has been added, you can right-click on it to go back and edit any of these settings from the contextual menu, including the ranges of all parameters assigned to it. 9 The same technique applies to adding a button, though since buttons have only on and off states they are better for assigning to something that needs to be at one of two values. Since you can set the start and end values for buttons, too, it’s still possible to toggle a reverb depth between, say, 17 and 36 per cent, but not modulate between those two values. For that you would use a knob. To assign a physical MIDI control to a quick control within an FX Chain, right-click on it and choose Learn. 10 If your controller is set up correctly, moving one of its controls should now cause Sonar to link the two so that they are mapped together.

Multi functional If you have existing tracks with audio effects loaded onto them you can combine them into an FX Chain easily so that they can be shared with other users and used in other projects. Go to your FX pane and right-

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FX Chains are easier to work with than you might think thanks to some clever interface design WANT MORE? Music Tech Focus: Sonar 8 is still available online. Find out more at www.musictechmag.co.uk/ mtm/focus

click in it, choosing Save FX Chain Preset and then assigning it a name in the resulting window. Alternatively, choose Convert Bin to FX Chain from the same window to have the plug-ins automatically grouped together into a Chain. 11 You can then double-click on the Chain to edit it or right-click to rename it. Yet another option open to you is to right-click on a chain and choose Extract FX Chain Plug-ins, which will unbundle them back out into separate modules. 12 FX Chains in Sonar X1 are incredibly powerful and easier to work with than you might think thanks to some clever interface design. Effects can be chained together, rearranged, automated, controlled and even unspun back to individual plug-ins with just a few clicks of the mouse. For many everyday production and mixing tasks it’s useful to be able to call up a readymade, preset chain of audio effects for everything from vocals and drum processing to getting a great guitar sound. Explore FX Chains in Sonar X1 and see how they can speed up your workflow and help you to get even more out of this excellent DAW. MTF

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12 Existing tracks can have their effects combined into a new Chain, or a Chain exploded back out to separate effects.

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MTF Interview Goetz Botzenhardt

“Speed of mixing was hardest to learn. I now mix a movie in the same time as I’d normally mix a pop single’’ The MTF Interview Goetz Botzenhardt

A good soundtrack is vital to any film’s success, but its making is far from a straightforward process. MTF finds out what 5.1 soundtrack mixing entails... Photography by Zen Inoya

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oetz Botzenhardt is a successful writer, engineer and producer in the pop music world. He recorded and mixed the first Sugababes album and worked on various tracks with the Pet Shop Boys, Björk, Faithless, Blondie and many more A-list pop artists. We catch up with him at a point in his career at which he has switched from two speakers to six. Although he is still active in stereo work, today we are sitting in his commercial studio – Soho Sound Kitchen. Goetz is also one of Air Lyndhurst Studio’s engineers, but SSK is home to himself and fellow composer Christian Hendon. We arrive with the sole purpose of finding out exactly what working on soundtrack mixing for film truly entails. We previously interviewed Youki Yamamoto in MTM March 2011 about composing for film score, but today we want to know about the engineer’s side of the mixing process. Goetz is happy to share his techniques and has two film cues set up to show us. The first is from The Imposter; the second is Storage 24. Both of these were released last year. Film cues are the sections of music that apply to single scenes in a movie – and Goetz reveals that there can be 50 cues or more to mix for a single film. In terms of time frame, he may have two, three or four days at the most to deliver these, although, as we continue to talk, it sounds like three or four is considered a luxury and doesn’t happen often...

Cueing up Goetz opens up The Imposter and tells us how a project might arrive: “The composer may have some programmed stuff in a project which was used in the writing stage. They will record this as audio and some of it gets replaced with recordings of real performances. This is dictated by the film budget so I can get a varied combination of real and sample-based parts

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On the disc

from one project to the next. In this particular cue, strings, flutes, clarinets and woodwinds are real. Synths, piano and harps are all programmed parts.” We’re then told how Goetz works the files into Pro Tools: “I’ll drag the files into a template session that has the routing I use already set up. Live parts are routed through the SSL desk for mixing, but synthetic and programmed sounds will be routed directly to a buss inside Pro Tools called ROM. This stands for Rest Of Music, so it’s easy to understand where my

‘‘You’d be surprised how different your perception of the music is when looking at picture” real and synthetic parts are at any time. Everything will end up at the 5.1 stem buss in Pro Tools, so here [in The Imposter project] the orchestral, percussion and ROM buses will route to this 5.1 stem buss; a following stereo buss is used for creating a stereo mix as well. When I’m ready to commit, I just hit record once on these buses to quickly capture the cue. It’s all about being economical with time and this is a quick way to finalise work to audio.” Once he’s set up his live mics through the SSL, they will usually remain consistent throughout the movie. But ROM sounds can very greatly from one cue to the next, so these are balanced within-the-box for quick recall: “I’ll work by getting a

Goetz Botzenhardt Interview MTF

MTF Navigation Anatomy of a movie cue

SAMPLED STRINGS These are here to back up the live strings and add size to the real instrument recordings. The Albion strings you can see here have been recorded at Air Studio’s Lyndhurst Hall. They sound big, ambient and beautiful.

LIVE STRINGS Here you can see the types of mic configurations used for real recordings. From top to bottom: spots, Decca Tree (left, centre & right), outriggers and surrounds.

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STEMS To create the stereo mix I will re-balance the output of the individual surround channels. The standard for doing this is centre and LFE channels set to -3dB, -6dB for surrounds, and the stereo channels remain at 0dB.

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COLOUR-CODING As well as colour-coding the audio files into groups, these buses are also given colours to make their grouping easier to see. They are here to sub-mix existing audio from above the video track, FX returns, and channels that have passed through the SSL desk. They are then routed into the 5.1 surround stems.

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MTF Interview Goetz Botzenhardt

few cues sounding good, then do smaller level adjustments from one cue to the next within Pro Tools so it can be recalled without changes on the desk. The real strings are pretty consistent throughout a film, but if there’s a radical change, we’d deal with that on a different session. So that’s why the programmed material stays in-the-box as that can vary a lot.” We ask Goetz about how he approaches all these cues: “I’ll usually divide them up into groups, such as action cues, character themes, emotional themes and so on. As the cues will be related you can mix one then jump to another, and they’ll be quite similar for the mix.”

MTF StudioEye Soho Sound Kitchen, London ADAM S3A The top end from the folded ribbons is amazing. My only issue at the moment is that my new control room is bigger and I need to go for some bigger speakers soon. The sub woofer is to the right of the SSL desk, letting me monitor my occasional use of the LFE channel.

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TC ELECTRONIC SYSTEM 6000 This is mainly for reverb on film mixing, but it’s also good for mastering stereo work. I wouldn’t do this on a 5.1 mix as that’s not my job, but might do on a pop record. The Large Warm Hall is my preset of choice for strings. I use more of it when the live room isn’t enough, or less if it’s recorded in quite a nice live space.

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Creating the desired effect We ask Goetz if there’s a prescription given to him as to how his mix should sound: “Often, the composer will sit down before I start and if I’m lucky they’ll have a selection of reference songs or movies for me to listen to – a Tom Waits record, for example, if someone was trying to get something that sounds quite old and crusty. We’ll also have a conversation about how we want the music to sound. I listen to the demos from the composers as they can be pretty high-quality these days if a good sample library has been used correctly. The piece will have already been signed off by the director and the film is also edited to the sound of that demo, so my mix doesn’t want to be radically different to it. But The Imposter was a film in which we had quite a free rein on how we thought it should sound.” For live instrument mixing and placement, Goetz consults the recording engineer. This helps him to create the correct imagery with the assortment of mic recordings he’ll have.

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Into surround The next step is to actually start mixing. Goetz starts with live instruments and creates a balance between spot mics, these are referred to as ‘desks’ as they are recordings of two instruments at a desk or music stand at a time. The first violins in a piece might comprise four desks, for instance: eight violinists performing and four spot-mic setups. As well as spots, Goetz utilises the left, centre and right mics from a Decca Tree, room mics and outriggers located at the back of the room, up high. He often puts the outriggers into the rear left/right channels, giving the listener a real idea of the room’s shape and making the sound surround you. Room mics are directed more towards the front speakers. We ask Goetz about his use of the .1 – Low Frequency Effects channel (LFE) – in soundtrack mixing: “The dub engineers don’t like us using it, at least not too much, so it’s best to mix the music without it. I might use it for a big drum hit or for sine waves as an occasional accenting tool, but it can’t be used to make a mix sound bigger in general. If I used it in that way, they’d just take it out of the mix at the dubbing stage and my mix would sound like something is lacking.” We assume that the central channel is kept free for the dialogue, but Goetz reveals that there is a little room for music: “It is for dialogue but it’s used a little; the central mic on Decca Tree recordings ends up on it, for instance. But placement-wise it’s still very stereo-based for the general mix. Instruments like strings and flutes would be a particular problem if you placed them in the centre channel. They’d directly conflict with the main frequencies in speech.”

Mixing in context While listening to other cues we realise we’re closing our eyes to focus on the rich orchestral work. We then remember to look up and watch the movie scene playing back on the

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projector screen, but we’re hearing only music, so we ask Goetz whether he works a mix around the dialogue during the mixing process or not. He reveals: “I’ll listen to the cue once with dialogue, then take it out until I think I’m ready to sign off the mix, then I’ll bring it back in to check in the mix again.” He explains how dialogue can affect his mixing decisions: “If there are gaps in dialogue you have much more room to feature elements of the mix, but it’s obvious that you don’t want a big flute, for instance, being high in the mix when someone is trying to speak, or a significant part being drowned with a drum bang and so on. It’s important to get the balance right in this respect otherwise the dubbing mixers will just have to duck all of the music from my stems to sort out any conflicts with dialogue.” Although Goetz removes dialogue for most of his mixing, he leaves the film playing on his projector while he works. His reasons? “You’d be surprised how different your perception of the music is when looking at picture, so I look at it quite a lot. When you look at picture, music becomes secondary and works on your emotions, whereas if you just listen to music, you become focused on the details. I’ve started mixing a few times without picture, then when I’ve put it on it’s made me hear the mix in a completely different way.” Moving from The Imposter to a cue in Storage 24, Goetz talks us through the setup for this specific cue: “Real strings are again routed through the SSL; there are Spitfire string samples in there as well, which can add a lot more size for a really big sound. I started mixing with these first, then I brought in the percussion.” The percussion parts are programmed, and although they sound quite authentic, Goetz will use some compression and EQ to get the most out of them. He turns to his GML EQ hardware to help make certain parts stand out if needed.

Goetz Botzenhardt Interview MTF

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AVID ARTIST CONTROL This just makes automation a lot quicker . I prefer the real-time, hands-on approach to changing levels over using a mouse to program changes. Plus, my 5.1 monitor controller is on the other side of the studio, so this is also used as my master volume control through Pro Tools.

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GML 8200 This gives the expensive and professionalsounding top end with a little edge that we all recognise. It has that radio-ready quality to it. I use it more for pop work than soundtrack music unless I really need to make something in-your-face.

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c DBX 120A SUBHARMONIC PROCESSOR There are three of these so each stem can be processed, like strings, percussion and then ROM. They add bottom end, so if I put in 200Hz it will generate 100Hz. It’s the opposite effect to an exciter really, which would add higher frequencies.

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MANLEY MASSIVE PASSIVE This adds that nice and expensive, warm sound. The top end is bright without sounding harsh. This is the main treatment I apply for strings. I also like to add a touch of bottom end with this too, as well as lifting the presence in my string sound.

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EMPIRICAL LABS DISTRESSOR This is an amazing compressor that’s great on vocals and other solo instruments. It is great for slamming duties in nuke mode but is versatile and can also provide less hard-edged dynamic processing.

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In terms of software, Goetz lists some of the plug-ins he regularly uses in Pro Tools: “For compression I like both the Bomb Factory BF76 and Fairchild emulations. The Waves SSL plug-in is my standard channel strip for EQ; everything else I use my real SSL channels for. It’s a workhorse plug-in. EQ work on soundtrack music is different from that on a pop record. It’s more a case of bringing out higher frequencies to help certain sounds cut through the mix than aggressive cuts to make space for other big sounds. It’s just about minor

‘‘If there are gaps in dialogue you have much more room to feature elements of the mix” tweaks here and there. I use a lot of compression throughout the mix, though, as it makes sounds more interesting and helps them sit better. Sometimes it might be applied quite heavily, such as when I’m needing to control the bass end. The double bass can also need this, as well as the synthetic bass and drum sections, too.”

Parts of the process Before moving to the final part of approving a mix, we ask if there are any more exotic types of processing involved from time to time. Goetz then shares a few tips with us: “Sometimes, for intense scenes in action movies, I might

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process the bass parts with a distortion effect to help them cut through the mix and sound more aggressive. The SoundToys Decapitator is great for adding some dirt and grit into the mix. Also, if a sound is a bit tired and needs a lift, I like to use the Chris Lord-Alge plug-ins. It’s got lots of ways to shape a sound and some great presets, too.” Once all processing and balancing is done, it’s time for Goetz to invite the composer to sit in for playback and he’ll make adjustments based on their feedback before committing the work to 5.1 stems. Finally, it’s printed to audio files, which are sent off to the dubbing mixers. For Goetz it’s very important to have all of his cues approved as his setup on the desk doesn’t allow for recall. Once he’s committed and the desk is reset, the mix is lost forever. However, this doesn’t seem to be a problem as he says he’s usually working on the cues the day after recording and a few days before it all goes off to the dubbing mixer, so strict decisions need to be made anyway and his desk can remain set up as he’ll work only on that project until it’s completed. Before we wind up the interview, we ask Goetz about the transition from stereo work to soundtrack surround: “Speed of mixing was the hardest thing to learn. I now mix a movie in the same time as I’d normally mix a pop single – that’s quite a change to adapt to.’’ MTF To see how a typical orchestral recording is captured by Geoff Foster at Air Lyndhurst Studios, visit our YouTube channel: www.youtube.com/user/musictechmag More information on Goetz’s studio can be found at: www.sohosoundkitchen.com The Imposter: www.imposterfilm.com Storage 24: www.storage24movie.co.uk focus Mixing 2013

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MTF Workshop Extreme loop processing

Reason 6.5 Workshop

Extreme loop processing tricks

On the & online


Loops can sometimes seem lifeless, but Reason’s tools let you mangle, process and mess them up to your heart’s content. Hollin Jones explains.

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Happily, Reason 6.5 has all the tools you need to do this. Here, we’re going to take a look at some of its built-in tools, as well as some of the new Rack Extensions that can help you to achieve this goal.

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eason has featured support for audio tracks since version 6, which means that as well as being able to import audio samples and loops to sample-playback modules, you are able to use drum loops in a more conventional way on audio tracks in a sequencer. This is a big step forward, since you no longer have to fiddle about with loop points and synchronise samplers (or, indeed, turn your loops into REX loops) to get everything working nicely. The advantages to being able to work more flexibly with loops are numerous. You may already have a big loop collection, and it’s easy to record and edit down your own. Beats and samples make great music beds for your tracks, but there is a potential problem: if you use commercial loops (or those from Reason’s Sound Bank) there’s a pretty good chance that others will have used them too, so sometimes what you need to do is mangle the loops, change and reinvent them in ways that make them more interesting or, indeed, unrecognisable from their original state.

PRO TIP Propellerheads’ online Rack Extensions store is where you can preview, try out or buy any of the currently available Extensions. Once installed, these are available for use inside any project and have full Rack-access privileges just like any of Reason’s own modules. There’s an ever-growing collection comprising both instruments and effects.

Use Reason’s built-in tools and techniques to make your loops more interesting

To begin with you can try using some of Reason’s built-in tools and techniques to make your loops more interesting. Start by importing a loop into an audio track. It will help greatly if your loop is already cut to the right length, as any commercially produced sample will be. Reason won’t stretch the loop to fit your project’s tempo on import but you can fix this by simply making sure that snapping is switched on then dragging the loop to the nearest bar marker while holding down the [Alt] key (making it longer will slow it down; making it shorter speeds it up). 1 Play back and your loop should be in time. Here’s the first thing you can try to change your beats: some simple tempo changes. By default, Reason stretches your audio sympathetically and won’t introduce any distortion. Sometimes, however, you may actively want to create a weird, slowed-down effect, so you can use the same tool to stretch out a loop much further. Try stretching the same loop over 50 bars, for example. 2 Reason will do this happily and the result is somewhat strange and robotic. Rhythmic loops cease sounding like beats and start sounding like textures and soundscapes instead. Of course, you can do the opposite and speed up a loop, making it very fast and perhaps repeating it to fill time. Another option is to copy and paste the same loop multiple times and stretch each one to a different duration to create yet more variation.

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2 Load up an audio clip or loop into an audio track in Reason and you have access to a number of time-stretching and slicing tools that can quickly make them sound completely different.

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Extreme loop processing Workshop MTF

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Clips can be reversed with a simple command, and edits that you have made can be formalised by bouncing clips out to new samples or new copies.

Use Reason’s audio comping tools in ways you may not have thought of before Another thing you can do easily is reverse a loop by right-clicking on it and choosing the Reverse option. 3 To make this more interesting, why not try reversing some parts of a loop but not others? This is easy to achieve by using the Razor tool to cut pieces of a loop (leave snapping on if you want the sections to be evenly timed) and then using the Reverse command only on specific sections. In fact, you can time-stretch certain parts of a loop, too, by combining the stretching technique with the cutting one, resulting in some pretty extreme sounds. If you want to render the edits you have made to loops down to the timeline (or, indeed, out to a new file), right-click and choose to bounce the clips to new samples, recordings or to disk. 4

Comp winner Another clever way to mess with audio loops is to use the audio comping tools in ways you may not have thought of before. If you double-click on an audio clip you will open it in Edit mode and see the Comp lanes, even if to begin with there is only one present because the clip has not been recorded in multiple takes. Locate the lane’s solid area to the left – it will probably be blue – and hold down the [Alt] key while dragging the mouse downwards. This creates a duplicate lane with an identical copy of the loop in it. 5 Place the track into Comp mode by clicking on the Comp Mode button. Now, using the Razor tool, make a couple of edits to add Comp markers to the first lane. With the Pointer tool, double-click the area in the second lane that falls between the two Comp markers so that it lights up in blue and is thus audible. 6 What you will now find is that you can move the instance of the clip in the second lane by dragging it left and right, and as the loop as a whole plays back, you

PRO TIP Long before Reason supported audio recording, Propellerhead created ReCycle, its audio-slicing tool that is able to take regular audio files and turn them into REX loops. These can be loaded into various DAWs (and, of course, into Reason’s Dr Octo REX) and triggered in interesting and unusual ways. Although Reason now supports audio recording, ReCycle is still a great way to work with loops as it boasts some unique tools and tricks.

have created a variation by telling Reason to play a different bit of the loop when the playhead passes between the Comp markers. You have effectively substituted one part of the loop for another and done it non-destructively. You can repeat this as many times as you like per clip and get really detailed with the edits by using very precise snap values (or even turn snapping off altogether). The loop will continue to be editable once you leave Edit mode, or you can repeat the Bounce technique from the right-click contextual menu to create a new, ‘glued-down’ version of the clip with all edits as part of the waveform.

Taking effect You can see from the above that loops are as flexible as any audio in Reason. You may also want to use the software’s many processing tools to get creative with loops, either in conjunction with or in place of the editing and comping tricks we have mentioned

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MTF Workshop Extreme loop processing

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The bundled effects are great for loop-mangling and The Echo module in particular brings some much-needed movement to any audio loop or recording.

so far. To make significant changes to the sound you will want to explore some of Reason’s more extreme effects, starting perhaps with The Echo. This module is capable of some great tape-based echo effects but also of some more heavy processing. Load up an instance and explore the presets. Try the one called Kraftwerk Drums, for example, which introduces some spring-style modulation as well as echo and sounds really dynamic on drums. 7 The Feedback dial controls how much of the spring is put into the signal. Other presets, such as Jaw Harp Drum Delay, introduce a synth-like quality to a beat, and you can use the Delay and Feedback section controls to add or remove the synthetic quality of the sound. 8 Another module that’s great at messing up loops while staying in tempo sync is the Pulverizer, and this is also well worth exploring. Presets range from moderate squeezing to extreme modulation, which you will hear if you fire up a preset such as Slap The Drums, which turns a beat into a sort of rhythmic synth sound. 9 Playing with the Filter section and its Frequency and Type controls enables you to emulate everything from an old-style radio sound to a buzzy synth. In terms of distortion and grit, you get the Scream 4 module and also the Line 6 guitar and bass modules, which are excellent for dirtying up loops. Add anything from gentle warmth to extreme crunch, and choose different amp and speaker types to customise the effect. 10 The most heavyweight effects can be achieved by combining these kinds of modules in a chain, taking advantage of all their talents.

Reason’s flexibility in regard to audio editing and processing means you need never run out of ideas Reason 6.5 also has the new Rack Extensions, which can be purchased from the Props’ online store. They fall into various categories, but for loop mangling it’s worth exploring modules like Buffre, Slice Repeater and Pulsar (which is free). It’s a dual pitch-shifter module, and you can select multiple parameters to create involving effects that add dynamism and interest to the simplest of loops. Try, for example, a preset called The Grainmaker on a drum loop – it really spices things up. 11 By using some or all of the techniques explored here you can change existing loops either by just a little or in more extreme ways by mangling, editing and processing them. Whether you’re working with Reason’s own library of loops, commercially produced samples or clips you have recorded yourself, Reason’s flexibility in regard to audio editing and processing means you need never run out of ideas. MTF

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Rack Extensions like Buffre are great for slicing up loops, but you can also experiment with free models such as the Props’ own Polar pitch-shifter.

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MTF Walkthrough Groove arrangement & programming


Groove arrangement

& programming

Drums patterns vary greatly from one genre to another, but many of the tools used to create them are universal. Liam O’Mullane goes from ideas to full arrangements.

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his walkthrough focuses on the programming aspect of creating drum tracks, but the production side of things will inevitably come into play either at the start or end of your drum programming activities. The production sound of drums isn’t just achieved with mix processing: it’s also a by-product of the choice of drum sounds you have to hand. But with so many samples packs covering a wide range of snare, kick hits and so on, it can be all too easy to get sidetracked when trying to find hits that will match each other and give a good overall drum sound. We strongly suggest working with drum kit patches instead – at least initially – so you can guarantee a well-balanced drum sound. You can always swap out sounds after programming. Doing it this way prevents the potentially creativity-crushing task of searching through folders of samples when you really want to be writing beats. The relationship between the drums and the key of your song is also very important from a production point of view. If you’re going to be writing all the main elements

On the disc

of your song as you go, experiment with the key while creating just your main melody and bass parts. The range of the notes you use in your melodies will dictate the available keys to a greater or lesser extent, so choose one that will ensure that the instruments don’t fight with the drums and vice versa. Similarly, make sure that the key enables the bass lines to remain in the audible range and that melodic parts hit the right midrange spot for their timbre. If you’re

It can be all too easy to get sidetracked when trying to find hits that will match each other going to record vocals, the vocalist’s range is an ultimate deciding factor in terms of key (unless you intend to find a singer to match the track when it’s completed).

The first pattern Programming a drum track typically starts by placing the core elements of kick, snare and cymbals. A typical rock beat, for example, will often have very open and

MTF Navigation Essential programming tools

ENVELOPE PROGRAMMING As well as note length, the amplitude shaping of drums can also be changed over time for more variation. This can be easily done with a sampler or audio clips.

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VOLUME AND VELOCITY Don’t leave all drums sounds at full volume – even minor changes in volume on certain drum sounds can help them to sound more lively.

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NOTE LENGTH CONTROL If you’re using a drum machine or sampler you can set the amp release low so MIDI note length controls how long the drum sounds for. An excellent accenting tool with an unnatural, synthetic sound.

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d CHOKE GROUPS Choke groups tie different drum sounds together, so the next note played cancels the last. Primarily used for programming open and closed hi-hats, this is also useful for maintaining constant levels when applied to kick and snares.

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Groove arrangement & programming Walkthrough MTF

washed-out-sounding hi-hats that play on the beats of the bar, while the kick and snare fall on the beats as well. This rocky pattern also features on various commercialsounding electronic tracks these days. However, by simply shifting the hi-hat to the off-beat, a more push-and-pull feel is created as the emphasis of the rhythm is spread over eighths rather than fourths. See the walkthrough box Genre-specific Patterns for other potential starting points. If you’re intending to create a drum pattern with various layers of information – hi-hats, shakers, percussion, kick, snare, break loops/slices and so on – decide on the core sounds first, then add subsequent layers once these are established. This approach gives

you the ability to control the drum’s overall energy level by simply adding and stripping layers away at the arrangement stage. When it comes to drum layering, do be careful about what types of sounds you choose to layer. If they have a lot of mid and low-end content they will increase the overall drum volume quite a bit when they’re playing at the same time. Cymbals and higher-pitched percussion are therefore good starting points.

Creating a groove On the topic of drum placement, groove plays an important part in giving your drums some feeling and movement. Groove can be used as a general term to

MTF Step-by-Step Step-sequencer programming in Reason

The ReDrum’s step sequencer-based sequencer makes it fun and easy to use. The first decision to make when you start creating a pattern is what resolution the pattern will run at and how many steps you want it to run for before it cycles back to the beginning. The most straightforward patterns will require 16 steps and a resolution of 1/16 (this is the default setting).

At 1/16 resolution one page of the pattern will equal one bar when Reason is running in a 4/4 time signature. If you want to work in triplets you can set the Resolution dial to 1/16T, which will split this bar over one and a half times as many pattern steps. You then need to set the Steps amount to 24. Now, instead of the on beats being 1, 5, 9 and 13 they will be 1, 7, 13 and 2 on the next page of steps.

When you’re building your step pattern for each drum, make good use of the dynamic accenting options of hard, medium and soft. All three can be utilised on 16-step hi-hat or shaker patterns to differentiate the parts playing on the beat (making them louder) from the parts between them (fade these out a little). Doing this on just a kick and clap can bring a lot of dynamics to the production sound of your drum pattern.

Try reversing the emphasis. Right-click and choose Copy Pattern, then select Pattern 2 and Paste Pattern from the right-click menu. This time put the emphasis on the off-beat, so steps 3, 7, 11 and 15 are set to Hard. If you flick between the two you’ll hear a dramatic change in energy.

You can use this simple approach of copying and pasting patterns to create various versions of the same drum pattern by adding or removing parts to differentiate them from each other. Drum fills are also important, and are best added to the end of an existing pattern so they are easier to trigger as you require them.

Record the pattern changes into Reason by hitting the transport’s Record button or by dumping the patterns from ReDrum onto Reason’s sequencer. Set the left and right locators to frame the area where you want the pattern to be, then right-click on the ReDrum and select Copy Pattern To Track. Dump all your drum patterns into Reason’s sequencer, then click Enable Pattern Section to disable patterns still onboard the ReDrum.

It’s also possible to pair the power of ReDrum’s step sequencer with the sonic options of Kong. Create a Kong, then right-click and select Combine. This will make it easier to create a Combinator patch that can be loaded at any point in the future.

Creating the Kong first will also make sure its audio outputs are routed to the Combinator and not the ReDrum. Add a ReDrum and flick the Rack around by pressing [Tab]. Now you can drag and drop CV wires from each module’s Gate output on the ReDrum to the desired Gate in on the Kong.

Click the disk icon at the top of the Combinator and save this setup for use later. If you want to use all of Kong’s pads just simply add another ReDrum and wire up the remaining six pads around the back.

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MTF Walkthrough Groove arrangement & programming

describe a drum pattern to other musicians, but when used in reference to programming it describes a combination of drum volumes and timing factors within a drum pattern. The timing aspect of a groove can involve anything from large, obvious off-grid placement to tiny micro-movements that give a more human feel. Although rhythms such as a four-to-the-floor kick drum will tend to be mathematically perfect, shifting other repetitive elements – a clap, cymbals or percussive parts, for example – can have a huge impact on how the feel of your pattern is perceived. Try experimenting and draw in drum parts with the snap grid turned off, or move them around ever so slightly while listening in a loop to hear the changes as you make them. When it comes to volume manipulation, accenting decorative sounds such as cymbals on the main beats of the bar by applying a higher velocity value will result in a solid, stomping groove, especially when combined with kicks and snares, which helps to reinforce the feel. If you switch the volume accenting to fall between the beats of the bar you again get that jumpy off-beat feel, but in a much less obvious way than just playing a single hi-hat on the off-beat.

PRO TIP

of this approach is that you won’t fall into the trap of copy-and-paste, which alone doesn’t really cut it in the modern world of production. The disadvantage is that it can be hard to keep the essence of the groove established in the first place as each new bar might be too different from the last. Another option is to copy, paste, modify and extend. This works particularly well when you have a few layers of drum sounds to play with. You can copy the existing part and focus on the new copy, leaving some parts as they are while slightly modifying others. Then you can copy the whole part so far and focus on the new section using the same approach. This method maintains consistency in energy as you make only slight changes, but the variation is enough to give the listener a more dynamic experience, even if they can’t hear the subtleties themselves.

A quick way of getting feel in your drum beats is to make use of your DAW’s groove tools. Some will just have different types of quantise grids, but others, such as Ableton Live, will offer much finer control. If you create your own grooves you can make a template from it so that other instruments can lock into it as well.

Alternate sections and repeats When you’ve created a full song section of drums, explore various alternate patterns to decide where you want your song to go. Doubling or halving the timing of the first drum section is very effective. A less extreme trick – but one that nonetheless creates almost the same type of energy change – is to halve or double the timing of just one or two layers in your drum pattern. So hi-hats that are playing an eighth pattern could change to a 16ths-based pattern; alternatively, the kick and snare could be changed while all surrounding parts continue as normal. Don’t be afraid to choose different sounds to stamp a clear and unique sonic identity on each song section. In the case of acoustic drums, the easiest way to do this is to move from playing the hi-hat to the ride or crash cymbal while the kick and snare provide a sonic constant. If you want to create a rhythmic hook that can continue throughout the track, again, a change of the sound that plays this pattern will easily create more sonic variation to use at the arrangement stage.

Extending a pattern Once you’ve got to the point of a short drum pattern with a few layers of energy, you’re in the perfect position to take those one or two bars and extend them into a whole song section. One approach is to program from left to right, creating each new bar one after the other by entering completely new rhythmic parts. The advantage

Shifting repetitive elements can have a huge impact on how the feel of your pattern is perceived MTF Technique Drum programming tips

Another method for getting less rigid drum timing is to record parts in live from a MIDI controller. Try speeding up or slowing down the project while recording to get a different feel when played back at normal tempo.

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When you’re staggering layers of drum patterns for less repetitive drum parts, glue/bounce them into their individual loop lengths to make it easier to spot what’s repeating and what isn’t.

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Automating effects can also add variation to your drum sounds. Try using delays to bring additional rhythmic elements on occasion.

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Along with half- and double-time switches, try changing elements such as hi-hats from regular to triplet timing to change the pace of one song section in relation to the next.

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Groove arrangement & programming Walkthrough MTF

When it comes to drum fills, they’re most commonly associated with rock drummers, who use big and loud tom tom runs to take you from one section of a song to the next. But drum fills don’t need to be based around just tom toms, and they also needn’t be placed purely at the end of song sections. An unexpected drum re-edit in the middle of a song part can serve to surprise the listener. This edit can be done with the existing sounds or with the introduction of something new. When it comes to arranging your patterns, it’s a good idea to create various versions with different levels of layering to make it clear how many options you have to

play with. You will probably have too many, but you can get rid of the weaker options and set about laying them out in the right order for arrangement dynamics.

Keep it simple A good rule of thumb is to keep things stripped-back for the verse/pre-maindrop sections, then give the listener the full intensity of the drums along with the other instruments for the main sections of a track. However, don’t be afraid to experiment, as alternating from high-energy drums with low-energy instrumentation (and the opposite) can often make a refreshing change. MTF

MTF Step-by-Step Genre-specific patterns

HOUSE A four-to-the-floor beat is simple to build and the kick is always bang on the grid for each beat in electronic music 99% of the time. Nudging the clap/snare and the hi-hats slightly forward of their grid positions will give the drums a sense of rushing ahead; when nudged back slightly it results in a lazy, loose feeling.

FUNKY TECHNO Although the core of a funky four-to-thefloor beat is always the kick, a sense of funk can be achieved by off-setting different drum elements by different amounts and in different directions of the rigid grid line. Here we have a funky techno beat with the clap slightly ahead of time and a double hi-hat pattern slightly behind and working with a triplet feel.

THE JIGGY DRUM BEAT The ‘jiggy’ beat is common in hip hop and dubstep. It has a double-time bounce to it, but regular, normal-time kick and snare parts. Creating one is a case of less is more – the feeling of the beat is implied and not dictated by hi-hat parts. Put the kick and snare on the beat, with occasional hi-hats placed loosely in between.

BASHMENT AND DANCEHALL BEATS The typical bashment beat is a powerful tool if you want a party vibe in your music. The kick is on the beat, but so much snare action is placed around it that it has a very different feel from most kick-led patterns. Sync’ing closed hats with the kick and an open hat with the snare will also accent what the snare is doing.

JUNGLE SNARE-LED BEATS The snare-led beats of jungle music range from around 140BPM upwards and give a very full, dominating and high-energy feel. It involves various cross-rhythms, with a shuffle 16th pattern driving it along and the snare blasting out as an irregular-sounding pulse. Constant cymbal crashes will give an Amen break-like sound, but keep them low in the mix.

DUBSTEP TO BREAKBEAT SWITCH This is a common technique used to go from the half-time feel of dubstep at 140BPM to the normal-time pace of breakbeat. It has a similar effect to the time changes commonly heard in hardcore metal and is easy to create.

ELECTRO BEATS From the days of early 80s electro, the funky placement of the kick drum is key to the high-paced feel of this genre. Fast 16th-speed hi-hats can be used and extra funk will come from removing the occasional hit. Try to accent certain hi-hats with a clap as well as a snare that plays on the second and fourth beats of the bar.

DRUM ‘N’ BASS When it comes to commercial drum and bass (acts like Pendulum and Nero), there isn’t that much going on with the cymbals as far as rhythm is concerned. Use washed-outsounding cymbals; a single closed hi-hat or shaker can be used to provide the slow feel of these rocky-sounding beats.

SALSA FLAVOURS Salsa music takes onboard a lot of the irregular placing of various drum elements, much like the bashment and jungle rhythms already discussed. The main thing to remember is that the kick and snare are rarely played dead on the beat, so don’t try to imply a regularly spaced rhythm with them as you would in house music.

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MTF Walkthrough Mix & workflow tricks

Ableton Live 9 Step-by-Step

Mix and workflow

On the & online


tricks for Live 9

Don’t waste valuable music-making time finding your way around the latest version of Ableton’s venerable DAW. Liam O’Mullane gets you introduced.

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s every Live user will know, version 9 is now just around the corner. Many years in the making, the exciting new features list has already done the rounds in our news section and on the web, and when we got our hands on the beta version and started making music with it, it didn’t take us long to notice some great new mixing and workflow refinements. Some of these changes are major – such as the new incarnation of the EQ Eight and Live’s dynamic devices – while others are much more discrete. So discrete, in fact, that you appreciate them only when used in practice. We therefore did the legwork for you and had a good look around, bringing you some highlights here. To begin with, let’s take a look at the very root of all workflow: Live’s new file management system.

It didn’t take us long with Live 9 to notice some great new mixing and workflow refinements setup. Although bookmarks could be set up for your preferred locations, they had to be accessed via a dropdown menu, so it wasn’t a lightning-fast procedure. In Live 9 the entire browser has been re-designed, so along with the expected tabs for Instrument, Audio, MIDI devices and third-party plug-ins there’s also Sounds, which you can search to find what you want rather than

Getting started Ableton’s approach to screen layout has always been to keep it simple – a single fixed master screen with non-moveable windows that surround the main

MTF Navigation New features for better mixing

sequencing area. The Device View Selector window is a fixed size and the Clip Overview window at the bottom of the screen can be re-sized only vertically, but this is perfectly fine to work in. Something that has always been a little restricting in previous Live incarnations, however, is the Browser. You used to be limited to using dedicated File Browser tabs for quickly accessing specific parts of your hard drive. The first of these would normally be left in its default location within Live’s own library, so in reality it was actually two not three tabs available for your personal

AUDIO & MIDI TRACK DEFAULTS You can now save your favourite default devices onto audio and MIDI tracks. Use this feature to store your favourite processing chains, such as a mixer channel strip.

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EQ SOLO The headphones icon enables you to hear just the EQ filter you’re editing. This is very useful for hearing exactly the frequency area you want to hone in on.

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BROWSER The Add Folder option enables you to add as many destinations as you need. The new folder system also automatically closes one folder when you open another.

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MAX FOR LIVE DEVICES Max For Live offers many things, but the most important mixing tool is the new Convolution Reverb. An IR-capture device is also included for creating your own impulse response files.

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Mix & workflow tricks Walkthrough MTF

MTF Step-by-Step Slamming drums with Glue

Both Compressor and Glue now have Dry/Wet dials for quick parallel compression. Here we’re playing a drum beat that doesn’t have quite enough energy in the high end so we’re going to create a slammed sound to then re-balance with the original. To start we’ll enable the Soft Clip button to catch any peaks at the output.

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For an over-the-top sound you can simply start cranking up the Makeup gain dial to raise the signal into the ceiling of the Soft Clip section. This goes from a general lift in energy between 5–10dBs to an increase in distortion as you keep raising the level. Now set the mix to Dry and slowly bring in your slammed parallel sound.

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having to go via the Live devices they were created with. Use this when you know you’ll need specific sonic elements, such as a pad, for instance. It also helps to minimise the time spent learning Live’s library content. If you’re very specific as to the type of source you’re after, use the tabs labelled Clips and Samples, which are useful for choosing pre-prepared audio or audio on your drive. Be aware, however, that Samples will display all samples contained in any location added to Live’s Browser, but the search box at the top helps to narrow down the options available. The shortcut is [Ctrl]/ [Cmd]+[F] (PC/Mac). In terms of adding your own preferred locations, the new Add Folder option enables you to add them to the same fixed panel in the Browser as for all other elements already mentioned. Folder management is now a lot easier to deal with too, as other folders close when you open a new one. This and the separate Browser labels for Instruments, MIDI and Audio Devices makes navigating your tools much easier as you create or mix. It also

PRO TIP You can edit automation to perfection using the new MIDI Stretch Marker function. Highlight the area of automation you want to work on and drag the markers to condense/expand your edit to perfection. A Pseudo MIDI Stretch Marker can be placed by clicking in the Marker area between the two existing markers. This is great for experimenting with prominentsounding parameters.

Another approach is to set the ratio at its highest as this is when the knee of the compressor is at its steepest. Set the Attack time to a super-fast setting of 0.1ms, then lower the Threshold until it starts to flatten the transients. To reduce pumping, try different release settings then add Makeup gain as required.

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negates the need to scroll up and down when the list gets too long due to multiple folders being open. This behaviour can be overridden if you prefer by holding down [Ctrl]/[Cmd] as you open a folder (useful for keeping go-to presets to hand when mixing and so on). Another way to keep this type of workflow available is to make use of the new defaults option for Audio and MIDI Tracks. This enables you to store preferred devices or plug-ins on a track.

Two into one The relationship between Live’s two primary views has improved quite a bit in Live 9 – certainly, Session and Arrangement views are now much better integrated than before. You’ll notice the benefits during those moments when you want to take a group of musical ideas or song sections in Session view and turn them into an arrangement in Arrangement view, or take an already laid-out song arrangement and move it into Session view for a live performance. For both tasks, automation for

MTF Step-by-Step Visual feedback with Compressor

Compressor now has three views: Collapsed View, Transfer Curve and Activity Display. The Activity Display is especially useful for understanding how the compressor is applying gain control in relation to the input signal. Enable this option by clicking the third switch to the left of the Knee setting display.

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As you lower the Threshold setting you will start to see a line that represents the gain reduction being applied. With a metronome-like sound it’s easy to see how the two Release curves behave. The first portion of this line is a linear release curve and the second a logarithmic curve. The Lin/Log button is to the left of the Dry/Wet dial.

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If you switch from GR mode (Gain Reduction) to Output you’ll see a representation of the compressor’s output on top of the input in grey – good for seeing how the attack stage and amount of compression is shaping your source material. You can visually separate the two readouts by disabling the Make Up button and using the Out amount to lower the Output line.

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Session Clips is a welcome feature, one that’s been very high on the Live user’s wish list for many years. It’s a slightly different process from recording automation in Arrangement view as you need to enable a new Automation Arm button for session capture, then a Session Record Button when you’re ready to go. Sadly, neither of these has dedicated shortcuts like Live’s transport controls, so use Key Map mode ([Ctrl]/ [Cmd]+[K]) and assign them to your preferred keys. When you move these Session clips to Arrangement view they behave just like a clip would if it originated from Arrangement view. It’s worth pointing out here that this capture can take place in one of two ways, set within the Record Warp Launch panel in Preferences. One captures parameter changes wherever they happen; this is called All Tracks. The other is much more like recording MIDI control of a specific instrument; this is called Armed Tracks. As the name suggests, you have to manually arm the track you want to record Session automation to.

PRO TIP EQ Eight now features an adaptive Q mode that changes the Q amount along with changes in gain. This is to imitate the behaviour of more classic EQs and goes from a wide Q at low gain amounts to a narrow Q at extreme gain settings. Use this mode for a much more musical sound as you explore what EQ treatment you want to apply.

Another improvement to Arrangement-to-Session workflow is the Consolidate Scene To Time function. All you need to do is select the area of an arrangement you’d like to move into a scene and right/[Ctrl]-click to select the function. If you’ve tried this before you’ll remember that it used to involve highlighting the area and selecting Split to create a clean section of clips, which you then had to drag to Session view. In itself this was quite a simple process, but you would also need to Consolidate sequenced pieces of audio on one track so they became a single clip that looped in Session view. The new approach saves quite a lot of time and all new clips are looped automatically.

Fix in the mix The last new element we’ll explore is the Audio To MIDI functionality, using it as a basic interface for creating rhythmic or musical parts. Start by recording yourself beat-boxing or singing and begin creating a track without any note programming or MIDI controller input.

MTF Step-by-Step Spectrum analysis and EQ grouping in EQ Eight

EQ Eight’s new onboard spectral analyser is a handy tool for several reasons, the most obvious being for seeing what content is in your sound. This can act as a visual reminder to keep your audio in good shape. Bottom end usually needs some sort of removal on most sources and Live’s new 48dB low-cut filter does the trick nicely.

When in Mid-Side mode, the analyser will also display the Mid or Side option you are currently listening to. Having two instances on your master channel with one set to Mid and the other set to Side can be very useful. With all EQ filters de-activated you can see where you may have some Side information to remove or bring out.

It’s easy to focus on single musical peaks in audio using the analyser. If you have multiple peaks, the new Group function enables you to grab an existing EQ treatment and move all highlighted EQ numbers up or down the frequency spectrum while their musical relationship is kept intact – handy for when you might have to change the key of a part and want to change the key of the EQ treatment as well.

Another use for grouped EQ filters is to highlight all EQs you want to alter and experiment with their Q amount. This can help to change the perceived size of a sound and is a handy feature to use when you’re getting towards the end of a and need to fine-tune one element to sit better in the mix. Gain can also be changed here, but if you simply want to do this for the whole EQ you can still use the Scale function instead as in Live 8.

For quick reference you can use the mouse pointer to find the frequency and musical note associated with a specific frequency area. Double-click the display to open the extended display. The EQ will show you controls for all EQ filters at once in the device and the bottom-left shows you the gain, frequency and musical note position of your mouse pointer.

When the analyser is extended, the left-hand side of the device reveals three options for changing how the spectral display works in both Condensed and Extended modes. The block size can be increased to add more frequency detail, but its response time will suffer. The Refresh rate can control this to some degree, while the Avg control moves between peak response and a more averaged readout over time.

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Its potential as a remixing tool should also not be overlooked, enabling Live to transcribe parts from a track to use with your own chosen drum or melodic sounds. Both are remarkable features, but we tried it for sound replacement and layering while mixing. For example, if you’re mixing a track for someone else, you may have the parts only as audio, and if an element is letting down the mix you can use these tools to add the necessary sound(s) to improve them as needed. This facility makes mixing work with audio a bit more flexible. Extract Drums To New MIDI Track works very well on anything with clear transients, but we find that a little additional processing prior to conversion helps Live to pick out more subtle drum hits in certain breakbeat sources that might not be as defined as you’d like. First we add a Gate device, setting the Attack, Hold and Release to very short values while the Threshold is set to only tickle the gate open. It doesn’t need to sound good at this point, you just want the gate to open for all the drum hits you think are in the audio. Try using the EQ

WANT MORE? Music Tech Focus: Ableton Live 8 Volume 2 is available now. Find out more at www. musictechmag.co.uk/mtm/ focus

for the Sidechain section on the Gate to hone in on the most dynamic part of the frequency spectrum if you don’t initially get the results you’re after, but be sure to Extract Drums after flattening your newly gated sound using Freeze and then Flatten from the right/[Ctrl]-click menu. You have to do this because the conversion process listens to the audio clip rather than any subsequent processing. Extract Harmony To MIDI is for working with polyphonic material, and if it’s melody-dense there are a couple of tricks you can try to optimise the results. We found that adding a simple high-pass filter to remove the lower, body frequencies of the sound helps Live to detect smaller, flourish-like notes. If problems still persist you can use additive EQ to raise the volume of the notes in question to help Live out a little bit more. Of course, we’re only scratching the surface of what’s new in Live 9 and needless to say we’ll be bringing you more tutorials and features in upcoming issues, so stay tuned and stay ahead of the game! MTF

MTF Step-by-Step Expanding audio

Live 9’s new version of Compressor now includes expansion, which was previously reserved for Multiband Dynamics and Gate devices. To use Compressor for expansion, click the Expand option on the right-hand side and the Ratio dial will update with reversed ratio values of 1:1 to 1:2.

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While watching the input signal on the Activity Display, decrease the threshold until it sits above the sound you don’t want to increase, but below the peaks of sounds you do. This is a guitar recording and we want to emphasise the plucking. We’ve placed the threshold above the string resonances between each peak caused by plucking.

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Raise the Ratio setting to start expanding the level of anything above the Threshold setting. We’ve gone for an automatic Release setting as the rhythm changes quite a bit throughout the piece. The Attack dial can be used with a low setting for sharp transients, or you can round off the plucking a little with a higher setting.

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MTF Step-by-Step Reducing chatter with Live 9’s new Gate device

We’re trying to gate-out the bleed from a vocal recorded with a loud backing track in speaker monitors. We can set the Threshold low enough to have the level of the vocal keep the gate open, but the gate is rapidly opening and shutting (chatter) and we can’t go any further down in level as this is the level of the bleed.

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This is were the Return value is useful as it tells the gate when to close after being opened. So the Threshold can stay towards the top levels of the vocals while the Return can be set at a lower value. Because this is only for closing the gate, the Return can go much lower than the Threshold could on its own.

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Now you can set the Attack, Hold and Release settings to taste so the gate is open for long enough to make sense rhythmically in your piece. You may want to go for a higher Floor setting so the gate doesn’t fully close to silence, especially when there’s a lot of bleed as the stark open and closed sound may be too much.

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MTF Feature Buss compression

BUSS COMPRESSION MTF Masterclass Studio Technique

Buss compression has become the secret weapon of many a leading mix engineer, helping to drive a mix and achieve punch. Mark Cousins gets dynamic.

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Buss compression Technique MTF

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hen you’ve struggled for hours to get a mix to sit together, you can see why the cohesion that a buss compressor can deliver is so desirable. Applied correctly, buss compression can ‘glue’ the disparate elements of a mix together, gelling instruments into a cohesive and musically effective whole, as well as adding punch and energy to the overall performance. However, while a liberal dose of buss compression has the potential to lift a track, it can also ruin an otherwise good performance if applied in a sloppy and incoherent way, or without access to the correct type of hardware or software compressors. In this feature we’re going to explore the powerful sonic effects of buss compression, understanding what is meant by terms like ‘glue’ but also exploring the practical ways in which you can implement buss compression as part of your studio setup. More than just being a gimmick, though, we’ll see how buss compression can have a profound effect on your mixing process, transforming your approach to balancing and processing a track as well as giving your mixes that elusive professional sheen.

On the busses For the uninitiated, buss compression refers the process of inserting a compressor across your main stereo outputs or across a grouped section of instruments routed to a buss fader. Arguably, the technique of buss compression was conceived when mix engineers started to explore the unique effects that a compressor could deliver when it was patched across the outputs of the console rather than inserted across individual channels in the mix. As the process became more popular, companies such as SSL even started embedding a dedicated buss compressor into the console’s signal path, with SSL’s G-Series compressor arguably becoming the de facto standard buss compressor. What’s interesting to note about buss compression is the variety of opinion regarding its validity, and, more importantly, the number of ways in Tech Terms ● OPTICAL COMPRESSOR An optical compressor uses a small LED as its gain control element – the stronger the light glows, the more gain reduction is applied. ● VARIABLE-MU COMPRESSOR A variable-mu compressor uses re-biased vacuum tubes as its gain control element. The tubes have an interesting non-linearity that lends them a unique sonic colour. ● FET COMPRESSOR FET compressors were a technical evolution of earlier Optical and Variable-MU designs, using field-effect transistors as a means of quickening the attack and release times.

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MTF Technique Buss compression

which it can be applied. As we’ve already hinted, the application of buss compression tends to fall into two distinct camps: either using a single compressor patched across the main stereo outputs, or a series of two-channel compressors applied across groups of instruments (a drum submix, for example, or a collection of backing vocals). Each method brings its own distinct set of pros and cons, as well as transforming the methodology and workflow of the mixing process. Although each engineer will express a different opinion, I’d argue that master buss compression should be approached with a large degree of caution, while buss compression applied across groups of instruments should be positively embraced. The issue with master buss compression is that it somewhat duplicates the use of a two-channel broadband compressor as part of the mastering process, although by having been applied at the mix, it limits what can be achieved in mastering. Put simply, once the buss compression has been overcooked during the mix (easy to do as your ears soon become fatigued) it’s next to impossible to restore the original dynamics later on in the production process. By compressing groups of instrument, though, we can assimilate many of the positive attributes of buss compression without killing the overall dynamics of the track. Importantly, we gain control of where and how buss compression is applied – instruments that need to be ‘glued’ together, for example, can be compressed accordingly, while other instruments might pass unaffected through to the main stereo buss. We can also be more creative about

Tech Terms ● VCA COMPRESSOR VCA compressors use a Voltage Controlled Amplifier for gain control - offering finessed and detailed control over attack and release times, as well as the knee of the compressor. ● KNEE The knee defines the transition between a compressor’s linear and non-linear response. The wider the knee the smoother the transition between the two.

The application of compression is largely driven by the need to shape the dynamics of a track

As with all buss compression parameters you need to pay close attention to the attack and release settings so that the compressor’s movements are empathetic to the music.

the types of compression used for each instrument group, maybe contrasting a fast-acting FET compressor on a drum kit, for example, with gentle variable-mu compression across a strings submix.

Sticky stuff Before diving into the specifics of buss compression it’s worth taking time to consider what we’re trying to achieve by applying compression in this way. First and foremost, the application of compression is largely driven by the need to shape the dynamics of a track, arguably understanding that even though individual instruments might be controlled, their combined input might vary at points throughout the song. As we’ll see later, it’s often about discerning and controlling the differences between your mix on a micro level – with individual instruments – and on a macro level, where instrument groups combine to create a complete arrangement. Beyond dynamic control, another major role of buss compression is this elusive term of ‘sonic glue’. A mix that lacks glue is something that all of us has encountered at some point, whereby a mix sounds like a combination of disparate elements rather than an entity that works as a whole. Given that most songs are recorded as a series of separate overdubs and that we are initially driven by the process of separation in a mix (usually by EQ’ing instruments separately to avoid overlapping frequencies) you can see why the end result might lack cohesion. Contrast that approach with the ‘Wall of Sound’ recordings

MTF Pro Technique French-style compression Although it’s not technically buss compression, so-called Frenchstyle compression is an interesting variation on the buss compression principles we’ve looked at in this feature. Daft Punk first exploited the technique back in the early 00s, although artists such as David Guetta have arguably taken the effect to its logical conclusion with hits such as Titanium, in which the bass drum appears to pump the mix in an extreme way. The trick behind French-style compression is the use of a compressor’s sidechain input. In a conventional application, the gain-reduction circuitry is driven by a version of what’s present at the compressor’s input. By feeding an alternative signal into the compressor’s sidechain input, though, we can manipulate the gain reduction using a completely different source – in this case a kick drum. To re-create the effect in your DAW you need to ensure that your host and any third-party plug-ins supports sidechain routing. Start by routing the required channels through to a spare buss fader –routing everything from keyboard lines

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through to hi-hats and reverb returns. Depending on the routing in your DAW, you might also need to route the kick to a separate buss so that it can work as the sidechain input to your compressor. Instantiate a compressor across the buss fader with the grouped instruments, setting the kick buss as the sidechain input to the compressor (sometimes referred to as the key input). The kick should now control the compression, so that gain reduction is applied whenever the kick is playing. The finesse lies in how you fine-tune the effect. Aim for a high ratio (8:1 or more) and a moderate threshold so that you achieve plenty of muscle in the gain reduction, applying as much as 10dB of compression. The attack should match the movements of the kick, but you really need to work with the release to find the setting that best suits the tempo of the track. In theory, the compressor should ‘return to zero’ by the end of each beat, so you’ll need to carefully adjust the release time based on the tempo of the track. In French-style compression, a kick (fed to the compressor’s sidechain input) drives gain reduction.


Phil Spector made in the early 60s – which often comprised large ensembles of players all recording at once – and you can see why a modern mix might struggle to be ‘greater than the sum of its parts’. In effect, therefore, the glue that we associate with buss compression is arguably formed by the instruments sharing the same amount of gain reduction – in short, the more gain reduction you apply, the more glue you add to the mix. Having created the separation, the application of buss compression sticks the mix back together, but (unlike the Phil Spector approach) in a controlled way. In short, we can decide which instruments are stuck together, which instruments float free, and how much adhesive we want to apply. Beyond the ability to control dynamics or glue instruments together, the final piece of the buss compression puzzle is its ability to add a degree of mojo into your mix. Rather than just being a vague term, mojo arguably encapsulate all the positive and interesting sonic attributes compression can deliver. Applied in the right way, compression can add energy and punch to a piece of music, and in some cases even appear to make the track ‘breathe’ in a musically effective way (French-style compression is a good example of this).

Group mentality Having understood some of the reasoning behind the application of buss compression, let’s look at practical ways in which we can apply it. First, we need to consider at what point in the mixing process it’s best to apply buss compression – should it be running from the start of the mix, for example, or is it the final ‘icing on the cake’ once the mix is reaching perfection? In truth, the answer lies somewhat between these two points. Starting clear of buss compression will enable you to assess the dynamics of your instrumentation and how the instruments naturally sit together. Once you’ve achieved this initial understanding, though, it’s worth adding the buss compression into the equation so that you’re effectively mixing ‘through’ the compression. Given that we’re opting for groups of instruments rather than the master buss, the next stage is to start to think about how you want to organise your instruments and the various compressors at your disposal. The solution, of course, can be as complicated or as simple as you see fit. You could, for example, just use the buss compression on a single instrument group (usually the drums) to add a selective amount of glue. A more involved solution, however, might use multiple groups, each with a different choice of compressor and differing amounts of gain reduction. Equally, I’ve achieved some great results using two distinct buss compressors – bussing one half of the mix to one compressor and the other half to another compressor, seeing how the two interplay with one another. Given the choice of compressors now available, it’s also worth making some distinction between the type of compressor used for the respective groups of instruments. Optical or variable-mu designs, for

MTF Buyer’s Guide Software buss compressors VSC-2

Company Brainworx Price $299 Available in both software and the original hardware version (£3,642, from Vertigo Sound), Vertigo’s VSC-2 is a great example of the power of VCA compression for buss processing. Uniquely, the original hardware compressor has four VCAs, giving it an enviable combination of power and transparency. In particular, the fast-acting attack and release settings really help the VSC-2 in difficult peak-limiting tasks, where it always delivers a musical performance. There’s also a useful sidechain filter, freeing the gain reduction from a heavy-handed low end. Web www.brainworx-music.de www.vertigosound.com

SSL 4000 COLLECTION

Company Waves Price TDM $1300. Native £285 The SSL G-Series Bus Compressor has to be one of the most well-respected buss compressors ever made, with countless hits having passed through its unique VCA circuit path. It’s no surprise, therefore, that there are several developers – including Waves – producing their own versions of this unique buss compressor. As with the original unit, the Waves plug-in (as part of the SSL 4000 Collection) is pleasantly transparent – especially in the unique ‘wide knee’ 2:1 mode – but it also seems to lend its output a slightly ‘larger-than-life’ quality. Web www.waves.com

1176 CLASSIC LIMITER PLUG-IN COLLECTION

Company Universal Audio Price $299 Not the most transparent tool in the box, but the recent 1176 Classic Limiter Plug-in Collection delivers plenty of character compression that’s great for buss processing activities. The series actually features four variants on the classic 1176 design, from the original Rev A Bluestripe model through to the subsequent Rev E Blackface designs and the newer 40th Anniversary Edition. The 1176 works well both as a subtle gain-reduction tool – where the colour of the output is as important as the amount of gain reduction – but it can also deliver some great results when pushed slightly harder. Web www.uaudio.com

Try combining different compression types, aiming to match the type of gain control and its tonal characteristics with the instrument group you’re processing.

example, favour a more gentle application of gain reduction, especially in situations when you want to preserve transient details such as guitar picking. A FET compressor like the ubiquitous Urei 1176 tends to work best when you want plenty of character and movement to the compression, especially if you’re working with drums. A VCA compressor, on the other hand, offers detailed control and transparency, which is ideal in situations where you want the compressor’s fingerprint to be discrete. With so much choice available in hardware and software, it’s a shame not to explore the veritable Aladdin’s cave of compression styles on offer. Be wary, though, of compressors that might be adequate or even characterful on individual instruments but don’t quite cut the mustard when it comes to groups of instruments. The integral compressors included with most DAWs are good examples of this, often lacking the subtlety and musicality that the more prestigious compressors can deliver. While it isn’t essential to use a dedicated buss compressor (like the aforementioned SSL G-Series Compressor) it’s

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MTF Buyer’s Guide Hardware buss compressors XLOGIC X-RACK STEREO BUS COMPRESSOR MODULE

Company Solid State Logic Price £1,381 If you’re of the opinion that the original is always the best and can’t fit a full-size G-Series console in your bedroom, the X-Rack Stereo Bus Compressor Module should be top of your wish list. You’ll need to factor in the X-Rack or Mynx unit to house the compressor, with the X-Rack arguably being the best option as you’ve still got six slots left to fill (with another three Bus Compressors?). As you’d expect of a VCA compressor, it’s a versatile tool, catering for everything from a soft 2:1 ratio to a harder 10:1 mode. Web www.solidstatelogic.com

AM670 STEREO LIMITER

GAIN REDUCTION

Instrument In Ins nstru ns trumen trumen m t2 ment

Instrument 3

BUSS MASTER

Time

The ‘glue’ that engineers associated with buss compression is largely brought about by groups of instruments sharing the same gain-reduction movements.

worth opting for a compressor that is designed to work with complicated and varied programme material, some of which we’ve highlighted in the Buyer’s Guide boxes.

that many of the US-based mix engineers using this approach often have their compressors barely tickling their gain reduction meters, often using the compressor more like a colouring device. For the softer style of compression, look at using a combination of both a low threshold and a low ratio – no more than about 2:1. In this respect it’s interesting to look at the SSL Buss Compressor and its unique behaviour in the 2:1 ratio. In this lowest ratio setting, you’ll find small amounts of gain reduction being applied even at relatively high threshold settings. The reason for this behaviour is that the 2:1 ratio has an extremely wide knee, so that small amounts of gain reduction (using 1.25:1 ratio, for example) are applied beneath the threshold, with the full 2:1 ratio achieved only as the signal rises above the threshold. In short, the compressor spends plenty of time ‘compressing’, but only ever applying small amounts of gain reduction. When it comes to defining the attack and release of the compressor, the approach should be equally moderate so that the buss is massaged into place rather than being aggressively squashed. Keeping the attack around 10ms preserves transient detail, while an auto release setting adapts the compressor’s release characteristics so that it’s empathetic to the complex dynamics inherent in a collection of instruments. Put simply, it keeps the compression transparent, allowing you to glue the instruments without the compression becoming too noticeable.

Soft drinks

Peak slicing

Having grouped your instruments and chosen a particular compressor, we now need to start thinking about the settings we use. To best illustrate the methodology behind buss compression, we’re going to start by exploring two contrasting techniques – one using a soft, ‘over-easy’ style of compression, and another style of gain reduction that aggressively clamps down on transient energy. Each style has its place, of course, and between the two approaches there are plenty of other solutions to explore. With either approach, though, the key component of any form of buss compression is subtlety: small amounts of gain reduction can go a long way. Whatever ratio or threshold setting you use, therefore, aim for small amounts of gain reduction – from around 0 to -3dB – rather than any hard ‘gain pumping’. As a point of reference, it’s worth noting

To illustrate a completely different approach, we could look towards the buss compression controlling some of the transient energy routed to a given buss. As you can see, we immediately make some distinction between the different qualities present at each subgroup – a percussion buss, for example, will have lots of transient energy, while a strings buss will have much more sustain and body. Matching the compression to the type of instrumentation enables you to be more discerning and refined in respect to how you control dynamics, rather than the ‘one-sizefits-all’ solution of a single stereo buss compressor. For the fast-acting transient compression, we’d pick a high ratio and threshold so that the compression occurs more sporadically than in the low ratio/low threshold model previously described.

Company AnaMod Price £3,295 The Fairchild 670 is arguably one of the most sought-after compressors ever built, with original units now worth well over £20,000. The Fairchild 670’s variable-mu design doesn’t make it the fastest compressor in the world, but it delivers surprisingly musical gain reduction. AnaMod’s AM670 isn’t entirely faithful to the original tube-based design, using a form of analogue modelling to replicate the classic sound of a Fairchild 670. At over £3,000, the AM670 is still expensive, but it’s a great way of acquiring 670-type sound without a second mortgage! Web www.anamodaudio.com

2-1176 TWIN VINTAGE LIMITING AMPLIFIER

Company Universal Audio Price £2,230 If you’re ‘breaking out of the box’, why not make the trip worth your while and buy a compressor with plenty of character? Two original 1176s are a considerable expense and difficult to run in true stereo operation, so the 2-1176 Twin Vintage Limiting Amplifier is perfect if you want a two-channel version of the 1176’s classic FETbased compression. All of the original features of the 1176 are present and correct, including the all-important ‘All-buttons-in’ ratio, which sounds like gain reduction on steroids and is perfect for more extreme forms of parallel compression. Web www.uaudio.com

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MTF Pro Technique Compressing reverb

Mixing ‘through’ compression changes the way in Xxxxxxx which you look at mix balancing. Automating channel faders, for example, will raise a channel’s level and determine how hard the compressor is driven.

The attack and release settings should also be suitably fast so that the compressor works only with the transients rather than modifying the body of the sound. In this respect, a VCA or FET compressor should be the preferred option given that both of these types are known for their fast response times. Between these two extremes there’s plenty of ground worth exploring – particularly if you want the compression to leave more of a sonic footprint (or, to put it another way, mojo). Compression ratios of 3:1–8:1 can inject punch into the compression, but in this area you’ll find that the fine-tuning of the attack and release times becomes far more important, especially if you start to work the compressor harder. Watch out for the attack being too fast, reducing too much of a buss’ transient energy. As for the release, move away from the relative transparency of the auto setting and start to experiment with the release times (from around 100–400ms) so that the compressor audibly ‘breathes’ or ‘pumps’ in time with the music.

One of the more interesting and unusual facets of buss compression is how you deal with the application of reverb. The decision is principally two-fold: do you send the reverb directly to the master buss – and therefore distinct and separate from the buss compression – or do you embed it into one of the existing buss compression paths and have it squashed alongside the other instruments in that particular group? Routing the reverb through to a buss is an interesting option if you want the reverb to form part of the ‘glue’ of the group you’re trying to process. Of course, it makes sense that only the instruments in the group use a single shared instance of reverb, which is also good for defining the group’s own acoustic space. Obviously, the buss compression helps reverb to gel with the instruments in question, and can itself be enhanced by the pleasing pumping effect and added sonic intensity of some strategic buss compression. Running the reverb separately, however, can help it to remain distinct from the rest of the mix – almost floating the space away from the instruments routed to the groups. It’s also interesting to consider whether you feed the reverb pre-buss compression – using aux sends from the individual channels – or post-buss compression, using aux send faders on the buss faders. OK, you lose some instrument-byinstrument control from the busses, but it’s an effective way of processing groups of instruments complete with a touch of flattering compression. Try to consider the implication of buss compression and reverb. Routing the reverb through to the buss can yield some interesting results, but it might not always be appropriate.

Mixing through compression Beyond the settings of each compressor, what’s really interesting is the effect of mixing ‘through’ the buss compression. In effect, any dynamic change

For softer styles of compression, look at using a combination of both a low threshold and a low ratio MTF Pro Technique Buss compression and equalization Although you don’t want to get too carried away EQ’ing your buss faders, it’s still worth noting the relationship between buss compression and EQ. Indeed, it’s an interesting opportunity to contour the timbre of the mix, as well as affecting how the compressor behaves and how it colours the sounds it’s processing. If you want to hear the EQ modifications, place the equalizer post buss compression. Although its input is subtle, compression can alter the timbre. Timbral alterations are brought about either through tonal colourations or distortions within the compressor (a slight sharpening in the midrange from an 1176 is a good example of this) or because compression can marginally reduce the amount of high-frequency energy. Given this propensity to dull the sound, it can be a good idea

that you apply ahead of the busses has a knock-on effect on its accompanying buss compressor. For example, push a lead guitar routed to a guitar buss up by 6dB and you raise both its level and, of course, the amount of compression. The final level that you achieve, therefore, becomes more about the interaction between the individual channel fader and the compression strapped across the buss rather than the level on the fader itself. As you can imagine, it’s a complex interwoven relationship, but it’s also one with huge musical and sonic potential. With a little bit of practice, you’ll soon start looking at your mix in a different way, understanding the difference between a signal’s absolute amplitude (leaving the buss, in other words) and the given

Compression can reduce the amount of high-frequency energy, so consider adding an air boost post-compression.

to place a small ‘air’ lift around 12kHz on the output of the compressor, especially if the group needs to cut to the front of the mix. If you feel the need to EQ the group in respect to a key deficiency in the sounds, try placing it before the buss compression. By placing EQ pre-compression you ensure that the compressor is presented with a version of the signal you feel is most appropriate. If it’s a problem with the low end, for example, this is especially important as a compressor’s movements are often heavily influenced by LF sounds.

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MTF Pro Technique M/S buss compression Applying buss compression in the M/S domain is a powerful way of controlling the stereo presentation and power of a drum submix and is well worth experimenting with as an alternative to conventional L/R compression techniques. If you want to compress in the M/S domain you can either purchase a dedicated M/S compressor (the Fairchild 670’s Lateral/ Vertical mode being a good example) or pair a stereo compressor plug-in with an M/S encoding/decoding matrix such as Brainworx’s bx_digital V2 or any number of freeware versions that are also available. You’ll need to ensure that the compressor can address the left and right channels individually (in other words, it has two discrete channels) rather than working with a single set of controls across both channels. Once moved into the M/S domain you have complete control over the mid and side components of a stereo image. As the kick and snare largely reside in the mid channel, you can be more heavy-

intensity of sound (defined by how hard the buss is driven). Familiar tools such as automation take on a new light, so that channel faders can be used to drive a compressor harder, for example, while automation on the busses can yield some interesting results by raising or lowering the absolute signal level. In short, amplitude becomes a far more multifaceted beast, with significantly different results based on where the gain change is applied. Another important change that buss compression delivers is that you’ll start to think far more about the micro/macro structure of your mix. Groups of instruments now gain far more importance, with the

M/S compression is an interesting option for buss processing, especially on a drum kit or anything else with a defined stereo image.

handed with your compression, adding punch and focus to the sound of the kick without having the cymbals (found in the side channels) aggressively pumped. Alternatively, add some subtle compression to the side channel to enhance the detail, width and sense of ambience across the drum submix. Even something as simple as your final output levels can also affect the presentation of the kit, moving the bias between a strong, centralised mono image from the mid channel and the exaggerated width formed from a bias towards the side channel.

Buss compression is one of the most powerful tools a mix engineer has at their disposal

MTF Pro Technique Parallel compression Using a palette of compression techniques – including channel compression, buss compression and parallel compression – seems to be an essential part of modern mixing. Parallel compression uses the effect of compression almost like reverb, whereby a number of signals are sent to a compressor via an auxiliary send, compressed and then bled back into the mix. The important point to note is that the eventual output contains a proportion of both compressed and uncompressed signal, allowing you to retain the dynamics and transient details of the original signal but also enhance the sound with the added body and sustain that compression delivers. Given that parallel compression is only ever used in discrete amounts, you can afford to be much more heavy-handed with your settings (and indeed, the effect almost works best when it breaks the accepted wisdoms of ‘transparent’ compression). Look at using a tough ratio – such as 8:1 or above – so that there’s a sharp transition into compression. Also, reduce the attack time so that the compressor clamps down on the transients as, of course, these will still be evident in the dry version of the sound. The final linchpin is the release time setting. With parallel compression, the effect seems to work best when the compressor really begins to ‘suck’ the sound, with a notable, graduated release. The actual release time depends on the dynamic structure and tempo of your track, but start with a relatively long setting (around 600ms or so) and work backwards until you achieve just the right amount of ‘pump’. Parallel compression is usually applied using aux sends, creating a unique effect that has both compressed and uncompressed elements.

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final mix often being as much about how the groups balance against each other (and the amount of compression applied, of course) as well as the level on an individual fader. As such, you arguably start to work in a way that’s closer to how the listener perceives your music (in other words, the relative balance between rhythm, melody and chords) rather than the channel-by-channel balance you’re painfully aware of as a mix engineer. As you reach the end of your mix, it’s interesting to bypass the buss compression to see exactly what it is delivering to the overall sound and feel of a track. Not surprisingly, many mixes will fall apart once the compression is bypassed – not because they’re fundamentally deficient in any way, but because the buss compression, though subtle, has become an integral part of the musical balance. It also explains why buss compression tacked on at the end of mixing also doesn’t work, because it has to be an integral part of the balancing process.

Musical control Buss compression is arguably one of the most powerful tools a mix engineer has at their disposal – and it’s not a practice that’s restricted to only high-end studios with expensive outboard. However, despite the abundance of tools now readily available, it’s still an easy process to get wrong. The professionals that make such great use of buss compression all seem to use it in a subtle, musically intuitive way, responding to the emotional drive of the music rather than feeling the need to squash every last decibel from the performance. Ultimately, therefore, buss compression isn’t a ‘one-stop solution’ to a great-sounding mix, but an important part of a palette of tools that an engineer uses to bring out the best in a piece of music. MTF

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MTF Walkthrough Layering bass

Production Technique Step-by-Step

On the & online

Techniques for bass layering & processing www.musictech mag.co.uk/mtm/ workshops

Getting your bass to translate across a wide range of speakers can be tricky, but there are solutions to your low-end ills. Alex Holmes gets to the bottom of it.

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he bass line is often the most important element in most genres of modern club music, acting as the key melody or hook of a track, and combining with the drum groove to get people moving on the dance floor. But getting a big sound that will translate on both a PA system and over headphones or laptop speakers can be a daunting task, as the tiniest of changes to the low end can cause imbalances elsewhere in the mix. One way to get more control over your frequencies is to layer bass sounds, separating out the frequency bands to create a richer, more balanced final tone (especially in genres that favour bright, full-range synth bass parts such as drum ’n’ bass and electro). Here we’ll look at layering subs and synth bass sounds to create a well-rounded bass track that will sound good through both small speakers and a big PA.

How low can you go? When thinking about bass the most important factor is going to be your sub part, which generally ranges from around 40–100Hz. This is where you will make or break

MTF Navigation Bass-making tools

your track, and picking the right pitch and key is essential: too low and the speakers won’t be able to re-create the sound (plus it will be difficult for the ears to determine the pitch); too high and you run the risk of the bass line sounding painful and resonant (plus you’ll lose weight and power). Although most club PA systems will pump out sounds below 40Hz, we wouldn’t recommend going that low unless it fits the genre you’re working in. The optimum range for a weighty sub is around 44Hz (F) up to around 73Hz (D) give or take, going as high as 80–90Hz if you want something more resonant.

One way to get more control over your frequencies is to layer bass sounds Ideally, you would also consider how your kick drum is going to fit around your sub, either placing it above or below to create space for each sound, or tuning its fundamental to avoid clashing frequencies. The sub bass sound itself is often a simple sine wave, as the pureness of the wave allows precise placement at a single frequency. However, without any extra harmonics, a deep-down sine wave can be rather quiet as the human

TOP SYNTH LAYERS Our top layers are much brighter synth sounds with an array of processing and stereo widening effects. You’ll want to EQ each layer so that it occupies its own frequency range.

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DISTORTED SUB LAYER Once your main hits are in place, look for gaps to place your percussion hits. These will make for a more interesting groove and also ensure that frequencies don’t overlap.

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SUB BASS The clean sub bass layer carries the main low-end weight of the track. Try to keep this in mono and processing to a minimum to retain a consistent sound.

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a BUSS b BASS All basses are fed through a buss, where a little additional EQ and compression are used to gel the sounds together. This is also a good place to automate changes to the whole bass sound.

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Layering bass Walkthrough MTF

MTF Step-by-Step Digital and analogue subs

Although we’ve talked elsewhere about how to thicken up your sub, there are other ways to make sure you start off with a beefier sound. One of these is to use a wave taken from a hardware analogue synth, the incredibly subtle imperfections of which add character and warmth. Let’s try to highlight the differences by taking a simple sine wave in your DAW of of choice and drawing in some bass notes. We’re using the EXS24 sampler in Logic, without any samples loaded.

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Draw in bar-long notes at F2, C2, G1 and D1 and listen to how each one sounds through both your speakers and headphones, taking note of how your speakers handle the low D. Now load up the Clean Analogue Sub audio file from the coverdisc and listen to the difference. There is a certain fatness that shows itself best in the lower notes.

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ear is less sensitive in this area. An interesting alternative to using a simple clean sine wave, therfore, is to take a richer wave – such as a saw – and carefully filter out the top end, down to somewhere around 100Hz. Taking this approach ensures that the additional frequencies add a thickness to the sound while simultaneously retaining the original low pitch. You can also experiment with adding some subtle distortion to the sine wave, which gives a similar result by adding upper harmonics to the fundamental pitch. Unfortunately, however, this is where proceedings get rather more complicated, as too much thickness will detract from the impact of the clean sub sound. Extremely fine tweaking is often necessitated to reach a compromise between low-end impact and the overall thickness of the sub, and finding this ‘sweet spot’ will

Finally, load up the Processed Analogue Sub audio file and note the difference. This version has been heavily processed with outboard kit. Although we lose some low-end weight, the extra harmonics make the sound louder and fatter, making it a good sound to layer with the clean sub. It’s not essential to use analogue subs, but they can be a good way to fill out the low end, and there are plenty of sample packs available with sub basses taken from classic synths.

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very much depend on where you place your bass notes and the interference of the kick drum, as well as the overall balance of the rest of the mix.

Weighing it up So how do we strike the perfect balance between weight and thickness in our sub bass? One solution is to keep the original sine wave clean, and add another layer with additional harmonics on a separate track, enabling us to blend in the thickness to taste. The advantage of this approach is that you can retain the power of your original sub sound while carefully controlling how much bulk to add in the crucial low-mid region. This is similar to how the Waves MaxxBass processor works, whereby the plug-in examines the original bass signal and generates additional harmonics that can be blended in.

MTF Step-by-Step Matching the kick and bass

If you attempt to combine a deep subby kick drum with deep sub bass, the frequencies will pile up, meaning you’ll hit the 0dB ceiling much quicker – and ultimately struggle to get a loud-sounding mix. To get around this we want to carefully pick a kick with fewer low frequencies. Start by putting down a simple sub bass line with most of the emphasis in the 45–60Hz range.

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Now load up a bank of your favourite kick drums and put in your kick pattern. Cycle through the different kicks until you find something that has more of a knock sound than a low whoomph. If we pull up a spectrum analyser on the kick and sub we should hopefully see them peaking in slightly different places. If needs be, gently massage a little low end out of your kick with a low-cut filter to create more space.

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If you want the weight of your track to come from a heavy kick drum instead, we can easily switch this idea around. Start with a sub-heavy kick sound and transpose up the bass line to sit on top (match the two harmonically if possible, a distance of a 5th or a 7th can work well). If you’re unsure of the pitch of your kick, try putting it up an octave and adding some distortion. This should make it much easier to detect the pitch, then you can drop it back down again.

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MTF Walkthrough Layering bass

Alternatively, a simple solution can be to duplicate the sub an octave up and gently blend this in, although it results in a different, slightly thinner kind of sound. It’s through the careful control of this low-mid region that you can create bass lines that translate well to smaller speakers, as any information below around 100Hz can be lost on low-quality in-ear headphones and small laptop speakers. Of course, one could argue that a club track doesn’t need to translate to small speaker setups, and if you genuinely aren’t worried about the track sounding thin outside of the club, then you may not need to worry about this. However, if you listen to the majority of successful club hits, both underground and more commercial, they manage to achieve a balance that sounds heavy no matter what system they’re played on.

PRO TIP To achieve optimum balance between kick and sub, put a high-cut EQ on your master out and remove information above 100Hz. This way you’re concentrating only on the low frequencies of the track. With everything playing, bring up the sub until it starts to swamp the impact of the kick. Pull back a bit and you should have a good balance.

your room more loudly than others. Combine this with the fact that most studio monitors can’t pump out the deepest of subs and it’s no wonder that many an amateur mix is plagued by too much – or not enough –sub bass. The best solution is to use a combination of metering with spectrum analysers and monitoring on headphones, so the effects of the room become null and void. Analyse your favourite tracks to get a good idea of the amount of bass in relation to the rest of the mix, and remember that a little sub can go a long way. You want weight at the bottom end but you don’t want it to swamp the mix, as too much sub bass will cause the limiters to overload at the mastering stage, resulting in a quieter mix. If you’re writing bass-heavy music, maybe invest in a subwoofer, but be sure to calibrate it properly with your monitors.

Meter maid

Upper crust

As very few people writing electronic music will have the perfect room in which to work, it’s highly likely that there are certain frequencies that will resonate around

With the sub part of the track out the way, you can start thinking about adding in some upper layers. It’s here that your bass line can take on a personality, from

MTF Step-by-Step Layering basses

One way to get a truly thick and detailed bass part is to layer up sounds. Although this can be done by stacking oscillators in most synths, or layering up samples in a sampler, we prefer to use separate instrument tracks for maximum flexibility. Start by laying down a bass pattern with a fairly bright synth patch, then roll off everything below around 200Hz and take away some tops with a low-pass filter to leave a midrange sound.

Now copy this MIDI pattern to a new track with your sub bass instrument. You might want to think about transposing the whole pattern up or down a little to get the heaviest impact and to fit with the kick drum and the rest of the track. If you have a broad range of notes, take the highest and drop them down an octave. This gives your sub line a more consistent weight, with the main melody still coming from your top basses.

To make sure your bass translates on smaller speakers, send a copy of your sub to a buss and apply a little distortion to add harmonics. Trim with some EQ to make sure the sound is not too bright, and to take out any overlapping deep sub frequencies, then blend this channel in with your main sub sound. Alternatively, you could use a plug-in like Waves MaxxBass to add the additional harmonics in the 100–300Hz range.

You should now have a fairly thick and warm sound. If we want something a little more aggressive, we could create a higher top layer to add some sizzle. Copy over your mid synth part and make some tweaks to the sound, then cut lows below 2kHz. To add interest and movement, try applying some subtle phaser or chorus effects. Of course, you don’t have to use the same synth for this layer – why not try a completely different synth or sample instrument?

A great way to bring top synth patches to life is to use something called the Haas effect, whereby one side of the signal is slightly delayed to give a wider sound. In Logic we can use the Sample Delay plug-in, but there are solutions in most DAWs. The only slight issue is that the sound can lose definition when the track is played in mono due to phasing problems. To combat this, try sending a copy to a buss and add the stereo effects there, that way the phase of the original is left intact.

Similarly, a small amount of short ambient reverb can work wonders to add excitement and width, and also combat the ‘in-the-box’ dryness of many soft-synths. Finally, you’ll want to feed all of your bass tracks through a buss so they can be controlled by a single fader. Subtle processing here can help to glue the sounds together, and just to get things working with the kick, we’ve also applied a sidechain compressor.

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Layering bass Walkthrough MTF

resonant synth waveshapes to distorted guitar chords or plucked cello samples – indeed, the options are limited only by your imagination! And what’s more, the advantage to having this solid low end already in place is that now it doesn’t really matter if the sound you choose doesn’t have any bass in it, because, of course, this aspect has already been dealt with. The only essential thing at this point is to remember to roll off any low frequencies to make sure that the upper layer doesn’t overlap too much with the sub. If it’s a seriously full-range sound you’re looking for, you can take things further still and create additional layers on top of the foundations you’ve already built, giving each its own unique identity and applying different processing treatments to individual layers to better distinguish them. One example would be a sizzling top layer to which you could apply a bitcrushing effect or some bright distortion with the low-end cut fairly high. And as we have these various layers placed on different channels, we also maintain tight control over the stereo

PRO TIP If you’re using a sub patch with a lot of release, you might end up with notes overlapping, which can cause an odd wobbling effect as the sine waves clash. Although this can be an interesting effect, for a cleaner sound make sure that you put your instrument into monophonic mode. That way the release of any note will be cut off by the next, which is especially important for busy bass patterns.

field, enabling us to keep the subs in mono to ensure maximum impact and apply stereo spread effects to the top parts to generate interest.

Bringing it all together Once you’re happy with your layers, feed everything through a buss, where additional processing can gel the sounds together. Keep the layers on separate tracks so you have options for automating channels across the length of the track or bringing sounds in and out in different sections (especially useful for complextro-style bass lines, where you can experiment with a variety of top synth sounds while retaining a solid low end). Or, bounce the bass sound and import it into a sampler, which opens up yet more processing and envelope controls. We’ve looked at bass here, but these techniques can add thickness to other instruments, too. And remember, there’s more than one way of creating big bass with lots of control, but if you can create the sound you’re after with a single synth or sampler, so much the better! MTF

MTF Step-by-Step Bass line automation tricks

Now we have our stacked bass sound, let’s look at how we can add detail across the length of a track. As we have our layers on different tracks, we can create movement by automating parameters on specific instruments. First up, try automating the cutoff of the top layer to open out and close as it moves from one section to the next.

A great way to increase the impact when your track kicks in is to duck a little low end out of your bass just before the drop. Automate a low-cut sweep using a filter or EQ plug-in on the bass buss, taking care to make sure you get the full range back when everything kicks in. For more subtle sweeps use a shallow slope around 12dB/oct. This could also coincide with a similar low-end fade on your drum track or other instruments.

Sticking with the bass buss, you can create snappy edits in your bass line by automating the bypass on an extreme effect placed across the buss channel. One example would be to use a crunchy bitcrush plug-in with a fair amount of downsampling, and draw in nodes so that the plug-in comes out of bypass only for the final bar of a 16-bar section.

One of the bonuses of having a separate sub bass channel is that you can chop and change your top bass sounds without affecting the solidity of the track. Try creating a new 16-bar section that uses the same sub but a different sound for the top bass. This could be a different synth or sample instrument with slower attack, tighter decay/ release, or even an extreme FX chain to give it more character.

Now we’ve got our alternative 16-bar section, let’s use a reverb wash on the bass buss to help move back to our main bass sound. Add a reverb plug-in on the buss channel with a long setting of 4–5 seconds. Automate the wet and dry parameters just before the next section so that the sound slowly recedes into the distance before slamming back in. This is even more effective when combined with the low-cut EQ sweep described previously.

Finally, an issue when coming to the end of a bass section is that the bass track can cut off a little abruptly. To help with this, try sending your whole bass buss to a delay effect on another buss channel, automating the send level so that you’re catching only the final note or notes of the phrase. This can work really well when moving from a full-sounding section to a drop-down or outro.

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MTF Technique Remixing & mashups

REMIXING & mashups MTF Masterclass Production Technique

Although the tasks of remixing a track or creating a mashup have a different starting points and end results, the process of creating them requires a lot of similar skills. Liam O’Mullane talks to nine professionals in the industry and finds out how they go about it.

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here’s a multitude of reasons for wanting to create a remix or mashup track. For many people, it’s a stepping stone into the world of production, either as their very first move into working with music or to add to their existing skills in DJ’ing. A mashup is the combination of two or more songs edited to sound as one, and the creation process will be familiar to many DJs if they’ve found an acapella that mixes well over another instrumental track. A remix is a reworking or re-interpretation of an existing track, and is created using elements from the original production. These can be singular stems or, if you are doing it for your own entertainment, they could be parts of the final stereo file from the release (provided it has the parts you want in a reasonably isolated form). Successful remix work has boosted the careers of many DJs and producers over the years and it involves a lot more in terms of production work than most mashups. But as you’ll find out, the task of how you remix a track differs from one artist’s interpretation to the next, as their outlook on what a remix is and should be can vary a lot. In short, the earliest definition of a remix has its roots in the dub reggae culture, when various artists explored the ability to rework the arrangement of a track. Using the individual stem parts to drop in and out at will using volume faders, mute switches and effects, this technique evolved along with digital technology to become the art of creating the extended 12-inch mix. These were aimed solely at the dance floor and DJ arenas, making a song more suitable for club playback – something that Carl Cox still sees as his role when it comes to

Successful remix work has boosted the careers of many DJs and producers over the years

THE PROFESSIONALS SASHA This dance music veteran needs no introduction. Sasha has remixed hundreds of tracks since he started in the early 90s.

BT His first remix was Not Over Yet by Grace. He’s since carefully selected other jobs as for BT, the process is more involved than original writing.

CUT UP BOYS Ministry Of Sound’s mashup superstars. This duo have released ten mashup mixes so far. Mash Up Mix Bass 2012 came out last April.

www.djsasha.com

www.btmusic.com

www.cutupboys.com

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DOCTOR P A remixer who always takes things in new directions. His remix of Ed Sheeran’s Drunk is a great example of this. www.myspace.com/ doctorpdubstep

CARL COX Another dance legend from the early era of the scene. Carl puts his remixes into a club style that makes him want to play them in his own sets. www.carlcox.com

Remixing & mashups Technique MTF

THE PROFESSIONALS MICHAEL WOODS His first remix was of the seminal Ibiza track Cafe Del Mar by Energy 52. He’s since remixed Lady Ga Ga, La Roux, Deadmau5 and more.

FLUX PAVILION Often seen alongside Doctor P, FP takes a very musical approach to his remix work. His remix of the Freestyler’s track Cracks is a prime example.

MARK BREEZE Until recently, Breeze was the master of hardcore. Now, as he moves into his take on bass music, he shares his remixing tips from many years of experience.

SAM TOWNEND Sam is one side of the Tidy DJs alongside Deano, flying the flag for hard dance. Sam shares his experiences in creating mashup tracks with us.

www.michaelwoods.co.uk

www.myspace.com/fluxpavilion

www.markbreeze.co.uk

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Tech Terms ● PHASE In the context of this feature, phase is the movement of an audio waveform above (positive cycle) and below (negative cycle) the zerocrossing line. If two copies of the same audio file are played together but one has its phase reversed, the result will be silence as the two cancel each other out. ● ZERO CROSSING The middle line of an audio waveform is the zero-crossing point. This is when a speaker is in its central position, then the positive cycle pushes the signal forward and the negative pulls it behind the resting zero point. Editing without snapping to this point will result in audio clicks.

remixing. He tells us: “I’ll use the elements of the track that really make the original what it is in the first place, then I’ll turn the track into something I would play, or something I feel will enhance someone else’s music. Every remix I do, if the original has a really good hook, you’re going to hear that hook in my remix; if it has a good drum pattern, bass line or strings, you’re gonna hear that in my remix as well.” Sasha has a similar viewpoint and shares his intentions at the start of a remix: “For me, there has to be a melodic hook that I feel I can take in a different direction from the original track. It could be an instrumental hook if it’s really strong or a vocal. For me, that’s the whole point of doing a remix. A lot of the time I’ll hear a track and think it’s great but it doesn’t have the club impact, and that’s what I’ll go for. The first thing I add are my drums. That’s the first way to stamp your own mark on the record. I mean, a club record is 90% groove, so the drums are a big part. But it changes from one track to the next. There’s some remixes where all I’ve done is changed the drums to bring it to the dance floor; the other sounds and the arrangement have stayed the same. But other times I’ve written a new track underneath a melody to take it to a different place.” BT explains: “It’s about framing a remix from a compositional vantage point. I’m usually looking for the interesting elements of the song that aren’t at the forefront, the interesting ear-worms. Often, if there is a vocal, I like to start with that.”

Getting started Mark Breeze knows that if he’s been asked to do a remix, it’s because they want his sound. He tells us a bit about how he achieves this from one track to the

As well as musical considerations, the tempo of a donor track is an important consideration

Unless you’re already producing tracks professionally, it’s unlikely that you’ll be able to get hold of stem files from a label. However, remix competitions are a great place to get them from. Try: www.remixcomps.com www. findremix.com www.indabamusic.com

next: “When I do a hardcore remix it’ll be my drum set and my style of bass and so on, so the key is also important as I can only remix a track that will fit the keys I use for my style. The biggest thing I’ve learned is to turn down remixes when the key just won’t work for hardcore. It’s different now as I’m leaving the older sound and moving forward with the bass-driven stuff. But much like drum ‘n’ bass, there are certain keys that work best.” As well as musical key considerations, the tempo of a donor track is an important consideration. Michael Woods explains his usual starting point: “If the vocal is the main hook in the track then I tend not to go near anything less than 110BPM or more than 140BPM as this would require severe timestretching or compression to the vocal to get it to my usual 127BPM. If the track is more music-led then it’s fine as I tend to replay all musical elements with my own sounds anyway.” How far you decide to push the tempo or pitch of the original track is definitely an artistic decision. If you want everything to be incredibly clean in terms of sound quality then you really want to work within 10BPM or less of the original tempo. Like Michael

MTF Step-by-Step Creating a sampler instrument from stems

The quickest way to turn a stem into an instrument is to first find one of the longest notes and cut it out of the stem file. This may involve you having to export this smaller piece out of your DAW as a new audio file if you want to load it into a third-party sampler. But if the sample is native to your DAW, you can probably just drag and drop this new slice directly into the sampler.

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Once inside the sampler you can use its auto-pitch detection (if available) so the notes you hear are in the right place on the keyboard, or set a middle-C MIDI note playing while you use a tuner or analyser to set the sampler’s pitch control to the right place. If the sample has multiple pitches within it try playing it along with a synth at the same time and change the tuning until it sounds close.

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You can now create any melody line with the sampler instrument, but here’s another tip to make it more playable. This sound has a long portion after the initial attack spike in volume that is reasonably similar in level so it’s quite easy to loop. Most samplers can do this and it helps if you can use a fade option to smooth the sound of the loop as it repeats itself.

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said, the other option is to re-create the parts either from your own transcription or by requesting MIDI parts. Then you can feed your own instruments with the same melodies and craft it to work for your tempo and style. However, this doesn’t always work. Mark Breeze shares his experiences of this when remixing Azora’s Tell You A Secret: “It was around 140BPM originally and I had to get it to 170BPM. I tried various synths with the MIDI parts I’d been given and couldn’t get it to sound right. The producers weren’t keen on sending me their original synth patches so instead I requested a dry bounce of the main triplet-based riff. Then I could get it to the right tempo with minimal artefacts.”

All in the mix The main piece of software used by our professionals for both remixing and mashups is just as varied as original composition work. But as a working aid, Mixed In Key was name-dropped quite a few times. Sam Townend tells us about how he uses MIK: “It’s just a godsend when it comes to mashups. I used to just trawl through banks and banks of acapellas, tracks and samples trying things together; however, now, with Mixed In Key, I can organise all my sounds into keys so I have a pretty good idea if a few things are going to fit.” Sasha also uses MIK: “I use it to organise my samples so everything on my computer can be viewed by key, so if I want to write a remix in E minor I have a folder for it. A lot of the time I’ll drop a load of sounds into Maschine [his studio hub alongside Ableton Live] and re-sequence them. Because everything’s in key, it’s good for happy accidents, and within ten minutes or so I’ll have three or four sounds that work together really well.”

An idea from the outset Even though the artists Doctor P and Flux Pavilion usually put the tracks they remix into completely new contexts, they both told us that they will have a

If you can manually beat-match, using DJ software is the quickest way to audition tracks for a potential mashup.

Tech Terms ● STEMS The single or grouped parts from within an original track that are exported as new audio files. Usually they will be different instrument groups like bass, lead, vocals, backing vocals, drums and so on. ● MUSICAL KEYS The musical key of a song, broadly speaking, represents the choice of notes used within a piece of music. By matching songs that are in the same or musically compatible keys you can find tracks that will work well together before even listening to them.

good idea of what they want to do before the parts even arrive. Doctor P says: “I find it’s good to work towards something rather than just randomly doing things until it sounds good. On my remix of Plan B’s Love Goes Down, I knew that I wanted to do a drumstep/metal/hardcore-style track, and for my recent Scroobius Pip remix I knew I wanted to do a ‘dubby’ groove that was also loud and aggressive.” Flux Pavilion is of the same mindset before he starts: “A remix must always start with an idea. I don’t remix anything unless I have a good idea of exactly what I want to do with it. When I first did some remixes it was because I had a really good idea, not because I was approached to do one. I’ve tried to keep that mentality when I choose which ones to go for.”

Starting techniques for a remix The starting point of a remix can be the act of just re-creating what’s in your head, or it may be a point of experimentation to see where you can take an element of the track. BT tells us about one of his

MTF Step-by-Step Reworking stems as audio on the track

Whether it’s a synth line, drum part or lead vocal, a common way to rework parts in a remix is to chop and manipulate the original stems. You can use an automatic transient detection function in your DAW to do this or split them manually, which is easy enough if there’s just a few sounds to separate. Turn off the grid but have zero crossing turned on and slice up each part.

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If you split the parts using an automatic function you can also quantize the audio. But be careful – this can have an effect on sound quality, especially if you’re making the part play slower. If you manually sliced the parts, snap them to the grid if you want to remove the original groove and use volume envelopes to smooth out the ends. This will keep the quality intact as you slow the tempo.

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The next option is to play with re-ordering the parts or to repeat a few pieces for stutter effects. A classic trick is to play the first two beats of a track’s main hook as normal and then repeat the second beat until the end of the second bar. Pitch-manipulation and filters are always useful ways of adding more of your own style to the piece.

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MTF Pro Advice Legal Stories When we ask our professionals about any legal issues they’ve experienced in the past, two of them have some pretty solid advice to pass on. First, Sam Townend: “The number one rule is don’t release anything until you have full permission from all parties you have sampled in the mashup. If you’re ever releasing a compilation album containing mashups you need to state on any license agreement that the tracks requested will be used in a medley/mashup, and often the original writers of the work as well as the label will need to give permission for the tracks to be used in this way. People can be understandably protective about their biggest-selling record being mixed with a different vocal. Having said that, some big dance records are a result of ‘mashups’ and can sometimes give an old record a new lease of life by adding a new element to it. So if you’ve done something that you really think works, don’t be shy to contact the label that owns the rights to see what they think!” The Cut Up Boys have to consider the legal side of things way before they start working past the basic ideas stage. This became obvious when we asked them about their approach to creating a themed mashup as they do for Ministry Of Sound. They tell us: “When you’re doing stuff for radio, you can kind of use any tunes you want. But when it’s going on sale and licensing is involved, we can’t just use anything we want as certain labels won’t just license everything you want to use. When we start, we make a list of things we’d really like to use. We then find out if it’s even possible before moving on.” Mashup mixes like this release from the Cut Up Boys require legal clearance very early on in the creation stage.

If you’re doing a mashup for your own use you are limited only by the tracks you have access to favourite tricks: “One of the things that I enjoy doing is a non-linear cut-up of a vocal and experimenting with the non-linearity of what was once a straight line. I can come up with really interesting vocal hooks that way.” Cutting up the main vocal hook or instrument melody (depending on the type of track) is a great way to see how you can re-order what can be a musical sequence that your listeners already know in its original form. This is therefore an easy way to create excitement. Sasha has some interesting approaches to share when it comes to getting from the remix stems to a working idea: “I like to make composite bounces of stems like the melody and the bass and other bits in other groups. Then I can process them like samples. First I’ll explore what can be bounced together. For instance, if you feed a single string part through a guitar pedal it’ll sound very different from feeding a bounce of multiple sounds. The exciting thing is that you’re not going to know what pops out when feeding grouped sounds through a processing chain and I’ll spend a good couple of days processing sounds.”

So, if you have your remix files in front of you, try grouping certain parts and processing them in different ways to inspire you. You can always bounce the results to a new audio file and treat them as new starting points.

Starting a mashup If you’re deciding to do a mashup for your own use you are limited only by the tracks you own and have access to. But as mentioned in the Legal Stories box to the left, if you’re planning to have the mashup released officially or have been approached to do it, you need to find out what you’ll get clearance for before you go too far with the project. If you start by sorting your tracks by key it’s also worth viewing them by BPM as well to figure out which two tracks are likely to work together musically without being pushed too far in terms of tempo to be in-sync. The Cut Up Boys tell us a little about their general approach: “Knowing the keys of acapellas will give us an idea of what might work straightaway as you can get away with re-pitching them by a semitone or two. Sometimes, though, vocals sound really good pitched up, like the helium vocal effect. It just depends on the style of the project. It’s really nice with the specialty albums that we do for Ministry of Sound, like the 90s and old-skool themes, for instance, as it gives you a remit to work to. Then you can really get into it. When things are a little too open you can get a bit lost.” If you don’t have an imposed restriction like a theme or the legal side of things you might find your music library is just too vast to work with. Sam Townend shares his approach: “You have to be patient as it can be tedious searching through samples and tracks trying to figure out which parts will fit together. Sometimes it just works from an idea you’ve had or a DJ mix you’ve done that you want to turn into a mashup. But other times you’ll be banging your head and nothing seems to fit. Trial and error is as important as having an actual idea sometimes! EQ is essential to get the tracks you’re using to sit together by carving out some space. I also find that you need to drop everything down by around -9dB as you’re working on sounds that have already been mastered and processed, so allow for this to avoid clipping the final mix.”

Tech Terms ● VARI-SPEED Like a record turntable, varispeed changes the pitch and tempo of audio by slowing down or speeding up the audio. ● TIME-STRETCH AND TIMECOMPRESSION A process that adds information to fill the gaps created when stretching audio over a longer length of time than the original playback. Time compression is the opposite and parts of the original sound will instead be lost and squashed.

It’s possible to extract the vocals from a track if you have the original and an instrumental. Flip the phase on one so when played back you’re left with the difference, which is the vocals.

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The Cut Up Boys have some good advice on picking tracks that will most likely work from tempo information long before you get round to putting them into your DAW: “An example would be say a hip hop acapella at 90BPM. That isn’t going to speed up to 140BPM for a dubstep mash very well. But if it’s, say, 80BPM you could possibly get it down to 70BPM to work instead.”

Tempo and pitch

Tech Terms ● ALGORITHM Different software and plug-ins use calculations or commands for performing time-stretch/compression. These commands are known as an algorithm. ● BOUNCE To re-record or export audio from a collection of sounds that you want to become one single sound. The master file of a song is technically a bounce of all the tracks in the song.

While we’re discussing the relationship of a song’s BPM, the key it’s in and how far you can potentially stretch or condense it to work for your project, we get a few insights into how remixers consider and cope with these potential issues. Doctor P tells us: “Tempo and pitch can be a nightmare sometimes. Time-stretching usually works pretty well, although sometimes it can just sound wrong. I try to approach everything creatively, so every track is done in a different way. Cubase 6 has a nice feature whereby you can input the original tempo of a sample and it time-stretches everything automatically to the new project tempo. It also has a feature called VariAudio Pitch And Warp, which is similar to Melodyne. I use that a lot on vocals – it’s how I did the choppy vocal effect in the Ed Sheeran Drunk remix. Time-stretch algorithms are so good these days that you can get away with a lot of changes and it will still sound acceptable. Plus, ultimately, everything goes towards giving a track its character, so a bit of wrongness isn’t always bad!” For mashups, the Cut Up Boys share some more of their wisdom: “There’s certain tricks for maximising on sound quality. For instance, if I

MTF Pro Technique Favourite remixes We ask our professionals about some of their favourite remixed tracks. BT states: “One of the best remixes I’ve ever heard is the Professional Widow remix that Armand Van Helden did of Tori Amos. I was so impressed by how reverent and evocative the remix was and how it really captured the essence of the original song while elevating it to new heights. One of the best recent remixes I’ve heard in years is Cinema by Skrillex. Although we’ve heard it plenty of times now, I’m always blown away when I hear it. What a fantastic, breathtaking piece of work that is.” Sasha also recalls the Professional Widow remix as being: “Such a big track back in the day. The way he turned what that record was originally into a club monster was really quite stunning.” Doctor P gives us some more examples after first telling us what it is about these remixes that does it for him: “I like remixes that realise the full potential of the original parts; sometimes tracks have some good ideas in them, but they are produced in a slightly uninteresting way. Remixes can often make a track come to life, and the Crookers remix of Day And Night, Fat Boy Slim’s remix of Cornershop’s Brimful of Asha and the Cinema remix by Skrillex are some examples that I think do this well. It’s also worth checking out the originals first to see how much they’ve been transformed.” Flux Pavilion states that although he’s never been hugely into dance music, he did like how Chicane remixed Hoppípolla by Sigur Rós and took it to a different place musically. The Armand’s Star Trunk Funkin remix of Professional Widow by Tori Amos is a favourite with our professionals due to how transformed the track became for the club format.

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Using hands-on effects processing can make it quick and intuitive to find cool ways to process your remix stems or mashup tracks.

wanted to run a tune with an acapella from 110 to 125BPM, it would be best to pitch-shift the acapella up a semitone with vari-speed and then time condense the rest. With a whole tune I wouldn’t want to move it more than a couple of semitones up as you’ll start to lose the bass end and the tonality of it will change as well”. Whenever you use any form of time-stretch or time-compression you should always explore the different algorithms on offer. Different types will perform better than others on a case-by-case basis. Ableton Live has a quite a few different options, with Beats mode being focused on maintaining the timing accuracy of the material. But this wouldn’t be best for a melodic part. The Tones Warp mode might be more suitable in this case and Complex mode is geared towards warping full or complexsounding musical material so it’s the first option for mashups alongside Re-Pitch for vari-speed changes in tempo.

An idea to an arrangement There are as many options for laying out the arrangement of a remix as there are for original compositional work. But, if you’re aiming to create something for a specific audience such as a track for the dance floor, your best option is to study the typical arrangements used for the particular genre you’re going for. Club music does have some common forms, though, regardless of genre. A 16-to-32-bar intro is common, then a significant change will happen at bar 33. This could be a breakdown to bring the dynamics of the track down for 8–16 bars, or 32 bars if you really want to create a lot of anticipation for the main hook section that will follow. Another option is for the track to simply roll into the main hook and perhaps hold off both the bass line and the energy in the drums until this point to help announce its arrival without the need to break down all the track elements. After this, the rest of the arrangement really is a case of artistic creativity or something to be based on the more specific nature of the genre you’re remixing for. A common trick is to make sure that you have a signature sound in the intro that will make it easy for the audience to remember the track and know it’s


being mixed in each time they hear it being played out. The obvious choice for this signature sound is an element from the original track to let the audience know from the start that it’s a remix of a track they may already know. When it comes to making mashups, Sam Townend tells us about how he tackles arrangements: “I stick to the rule of mixing verse vocal parts with verse melody parts and chorus vocal parts with chorus melody parts. This will make sure you get a proper balance and all your sounds have the correct space in the mix. Trying to ‘shoe-horn’ sounds together tends to not work!” So, this may mean that you need to re-edit parts so they work together better, and from our experience it’s best to get the best match for this when choosing the tracks rather than relying on heavy editing, which can mess with the song structure too much. If you’re unsure about how to tackle this, it’s worth creating a few different versions of your project and trying out different edits early on in the process.

Finding the right tempo for an acapella can be difficult. If you struggle, grab the original and get that in sync using the drums and then apply the same changes to the vocal track.

Advice for the future When it comes to mashups, you shouldn’t need much more than the right selection of tracks, some initial editing to get things lined up in terms of arrangement, and then the application of EQ to create space between the different layers you’re using in the track. You might also need to apply some reverb or delay on an acapella to help it glue to the backing tracks being used. Then you can create smaller edits to add more excitement.

If you keep things really simple you’re going to end up with the biggest sound

Unless you plan to simply re-edit stems for a remix you will need to explore the production side of things to help establish a sound over time so you can consistently put your own sonic stamp on things. Carl Cox had some good advice for anyone starting out, which we’ll leave you with: “Be simplistic, try not to overcomplicate things too much. If it’s a bass line or drum pattern, don’t make it too intricate. With drums, go for something that’s just straight up and more like how a drummer would play a kit rather than how a computer can play them. Of course, you can go as far with content as you want to, but it can end up confusing the listener. On some older remixes I’d have 80-odd tracks going, but some of the best remixes I’ve done have been more like 12 tracks or so. As long as they’re big and phat-sounding parts they will come across well. If you keep things really simple, you’re going to end up with the biggest sound.“ MTF

MTF Buyers Guide Hand picked tools for the job

ABLETON LIVE INTRO

Price £64 Ableton Live is well-known for its flexibility when it comes to beat-matching audio and pitch shifting. You may want to go for the full version for remixing, but Intro has enough features for creating mashups quickly and easily.

VENGEANCE PHILTA XL Price £40

A current favourite of the Cut Up Boys. A good workhorse filter is always useful for creating builds and movement in your work and this filter sounds great and is well-priced. Web www.vengeance-sound.de

Web www.ableton.com

MELODYNE EDITOR 2 MIXED IN KEY

Price $58 A musical blessing for DJs but also a great way to either help match tracks for mashups or make musically matching your samples easier. Web www.mixedinkey.com

Price £289 This Grammy-winner is great for adding harmonies to vocals or re-pitching and editing them in the style of artists such as Skrillex. The DNA technology can be used for re-working musical content for mashups as well, but this is always at the cost of sound quality. Web www.celemony.com

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MTF 25 Pro Tips Monitors & monitoring

Pro tips for

monitoring

Monitors are one of the most crucial components of any studio, so it’s vital that an engineer understands every aspect of them. John Pickford shares the knowledge. FULL-RANGE MONITORS Sometimes referred to as main monitors, all professional recording studios, mixing suites and mastering facilities are equipped with at least one pair of full-range monitors. These are usually very large and contain several drive units, more often than not with separate, dedicated drivers designed to handle bass, midrange and highfrequency reproduction. Full-range monitors are able to cover the entire frequency range, from the deepest bass (which can be felt as well as heard) right up to super-high frequencies that are inaudible to human ears. Capable of performing at very high SPLs, this type of monitor is often situated several metres from the listening position, so it’s vital that they integrate well with the room in which they are used. Some studios build theirs into the far wall of the control room.

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NEARFIELD MONITORS Often used as alternative speakers to the main full-range monitors in large studios, nearfields are also the first choice of main monitor in smaller project and home studios. Usually no bigger than an average domestic hi-fi speaker, modern nearfield monitors can give a much more faithful account of how your recording will sound over home stereo systems. Because these monitors are normally situated quite close to the engineer’s ears, the influence that the room has on the overall sound is reduced by comparison to full-range monitors. Many engineers will choose to mix through nearfields these days, saving the full-range monitors for occasional loud playbacks and more accurate assessment of deep bass response.

HEADPHONES Musicians wear headphones in the studio to monitor their performances, and they are essential in order to hear previously recorded tracks while overdubbing. Engineers also find them useful to check stereo balance, although it’s unusual – and not really advisable – to mix down through headphones alone. They are also useful for pin-pointing faults such as hiss or hum without subjecting others present

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Monitors & monitoring 25 Pro Tips MTF

front of the monitors. To get it spot-on, it’s a good idea to experiment with the angle of your monitors. Some are designed to fire straight down the room, while others perform better when the speakers are toed-in to point towards your ears.

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at the session to the process. Most studio headphones are large, closed-back designs rather than the in-ear types favoured by the iPod generation. DOMESTIC AND CAR SPEAKERS Studio monitors are designed to be accurate across the audio spectrum, whereas hi-fi and in-car systems are more inclined to flatter a recording. Accuracy is important to achieve a good sound-balance and tone, especially when recording acoustic-based (as opposed to electronic) music. However, many producers of pop, rock and dance music like to use monitors that are tuned to sound more like domestic speakers. It’s common practice nowadays to run off a CD-R of a final mix to play in a car, giving an idea of what it might sound like over the radio.

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LOW-GRADE MONITORS Although not as common these days, many studios still make use of an alternative low-grade monitor, traditionally used to hear how a recording might sound on a cheap transistor radio. This type of monitor has a very limited bandwidth, with no top or bottom end and a generally tinny sound. Overly loud or quiet lead vocals can be easily detected by using a low-grade monitor. Auratones (often referred to as horror-tones!) were popular for this purpose and regularly used to monitor sound in television studios. A good alternative today would be to use cheap computer speakers.

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SWEET SPOT The listening position, in relation to monitor placement, is crucial in order to hear the best possible sound, with the sweet spot being the place where optimum sound quality is achieved. As a starting point, think of the stereo speakers and your head forming an equilateral triangle, so that if your monitors are positioned, say, two metres apart, the sweet spot will usually be found centrally placed, two metres in

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OFF-AXIS Not everyone listens to music glued in the sweet spot. Sometimes it can be useful to listen to your mix off-axis, elsewhere in the room. Even moving to an adjacent room can reveal basic sound-balance problems that may have gone unnoticed, particularly when critically listening to the finer details within a mix in your usual recording and mixing position. Top producer Trevor Horn likes to check his mixes standing at the back of the mixing desk, with the monitors firing into the room in front of him.

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ACTIVE MONITORS Most ordinary loudspeakers are passive devices that require power from an external amplifier. However, it has become increasingly popular to use active monitors in the studio. These units have bespoke power amplifiers built-in to the cabinet enclosure, often with a dedicated amplifier powering each individual driver. The main benefit of this type of monitoring system (apart from not having to buy a separate amp) is that the amplifiers will have been tailored to suit the speakers, both in terms of power handling and tonal balance.

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AMPLIFIERS If you choose to use passive monitors, a power amplifier will be required to drive them. Generally speaking, small speakers are less efficient than larger ones and require more power. A common mistake when putting together a home recording setup is to blow the budget on expensive monitors, then attempt to drive them from a cheap, lowpowered domestic amp. Using an underpowered amplifier will result in poor reproduction and, at high SPLs, will clip and produce a distorted sound. It’s far better to use a powerful amplifier that won’t become stressed when high monitoring levels are required.

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STANDS AND SUPPORTS In order to get the best possible performance from your monitors it’s important to place them on suitable stands. Simply plonking them on the meter-bridge of a mixing desk or on a rickety shelf will not produce stellar sonic results. Monitors are sensitive to vibrations and will not reveal their full potential unless they are placed on sturdy supports. Speaker cones move back and forth in operation and do not benefit from the

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MTF 25 Pro Tips Monitors & monitoring

cabinet dancing along. Some engineers place spiked cones and other vibration-busting devices underneath their monitors to further isolate them. However, a good pair of stands should provide adequate isolation, with just a small amount of Blu-tack in each corner of the stands’ top-plate to secure the monitors. CABLING Passive monitors receive audio signals from a power amplifier by means of speaker cable. Most cables are made from strands of copper wire (solid-core cables are less common) and range from very thin and inexpensive types suitable for doorbells but not much else right up to thick multi-stranded types costing hundreds of pounds per metre. It’s important that both runs of cable are the same length, regardless of the location of the amplifier in relation to each monitor. The bare wire can be attached directly to the amplifier and speaker terminals or banana plugs can be used for easy connectivity. Active monitors usually accept XLR cables from a preamplifier, with some also featuring phono inputs. Active monitors also require mains power via IEC (kettle) leads.

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WOOFERS In large, full-range monitor systems the woofer’s job is to reproduce bass frequencies, with separate drivers employed to handle mid and high frequencies. Nearfield monitors are usually two-way designs, with the woofer responsible for both bass and mid frequency reproduction. Woofers feature a diaphragm (speaker cone), traditionally made from paper although nowadays synthetic materials such as woven Kevlar are often used. The cone is attached to a voice-coil and magnet that drives the speaker cone as current from the driving amplifier flows through.

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usually made of fabric or very thin but stiff metal. Ribbon tweeters are becoming increasingly popular in modern monitor design, with some very accomplished designs being produced by Elac, among others. Because they are so light, ribbons possess an extremely fast transient response characteristic along with superior high-frequency extension. Some monitors utilise super-tweeters that deal with harmonic content that gives a sense of air and space. SUBWOOFERS Subwoofers are relatively new to the world of monitoring systems, first appearing on the scene in the late 1960s. Designed to augment existing setups and capable of delivering the very lowest frequencies, the first known studio use of subs was during the mixing of Steely Dan’s Pretzel Logic album in 1973. Most full-range monitors do not benefit from the addition of a subwoofer; however, nearfield monitor users often find them indispensable, particularly those producing bass-heavy electronica or similar genres. Professional subwoofers tend to kick in at around 100Hz– 200Hz and reach down to 20Hz.

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Small speakers are less efficient than larger ones and require more power BASS-REFLEX PORTS Many monitors, including subwoofers, utilise a bass-reflex port to aid bass extension. Described in the most simplistic terms, the port is a physical hole in the cabinet that is tuned to allow low bass frequencies to escape. Depending on the monitor’s design, the port can be situated at the rear of the cabinet or on the front baffle. Some designs employ two ports, while those that do not make use of bass-reflex port loading are known as sealed-box or infinite-baffle monitors. Although good bass-reflex designs can usefully extend the bottom-end output, some engineers complain of ‘chuffing’ noises heard from the ports of poorly designed monitors.

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COAXIAL SPEAKERS In the early days of studio recording, the majority of professional studio monitors were coaxial designs. This type of speaker features a tweeter situated in the centre of the main driver. The idea is to create a point-source, with the sound emanating from a single spot rather than having the tweeter placed above the woofer. Tannoy is arguably the most famous monitor manufacturer to use coaxial drivers, with its Dual Concentric designs gracing dozens of models

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MIDRANGE SPEAKERS Once referred to as ‘squawkers’, the majority of dedicated midrange speakers are constructed in the same way as woofers, albeit on a smaller scale. Some manufacturers, such as ATC, use dome-type speakers to reproduce frequencies in the 300Hz to 5kHz region. Horn speakers are also employed for midrange duties, though they are more commonly found in PA systems rather than studio environments. Midrange speakers operate in the audio spectrum where most of the sonic action takes place.

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Monitors & monitoring 25 Pro Tips MTF

over the years. (See the Studio Icons feature in issue 110 of MTM for the full story on Tannoy Dual Concentrics.) SURROUND SOUND The vast majority of studio recording and mixing is performed in stereo; however, with the advent of home theatre systems, recordings that contain front/rear audio information as well as left/right have become increasingly popular. Obviously, extra monitors are needed for these productions and studios equipped to mix in surround sound are much less common than standard stereo studios. Surround sound isn’t a totally new concept, though. Quadraphonic sound was a 1970s phenomenon that failed to take off and now resides in the same graveyard as the eight-track cartridge and the Ford Capri.

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MONOPHONIC It’s quite common during the tracking stage of recording to monitor in mono, with stereo panning saved for the mixing stage. Although stereo recording techniques using two or more microphones is not exactly an uncommon practice, many stereo mixes of pop music are produced from a number of mono signals that are subsequently carefully panned to give a sense of separation and space. It can be useful to collapse your stereo mix to mono in order to check the basic balance and also highlight any phasing issues that may have arisen within the recording.

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FAMILIARITY Regardless of the size, quality or cost of your monitoring system, it’s crucial that you are intimately familiar with its sonic signature. If, for example, your monitors are overly bright, a mix that sounds great coming through them may sound rather dull when replayed on systems with a flatter response. One of the reasons why so many commercial studios possess Yamaha NS10 nearfield monitors is that most jobbing engineers – who very often work in many different studios – know what they sound like, so even if the studio’s main monitors are unfamiliar models, the famous NS10s provide a frame of reference.

output from a specific driver. Bass-reflex-ported monitors sometimes have a switch that reduces the output from the port. This can be useful when rear-ported monitors are sited close to a wall (this increases the perceived bass response). Don’t engage tone controls to flatter your mix: if the sound isn’t right, adjusting the tone of your monitors will make no

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METERS Whether using a mixing desk or DAW, engineers rely on meters – VU, LED or otherwise – to visually monitor the output of individual channels, sub-groups and the main stereo mix buss. As useful as these are, it’s worth remembering that many great sounds have been created with the meters well into the red, particularly when recording in the analogue domain. Take note of what your meters tell you, but it’s better to mix with your ears rather than your eyes.

There are no hard-and-fast rules when it comes to the SPL at which you monitor

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PFL AND SOLO Most mixing desks feature a Pre-Fade Listen (PFL) and/ or Solo switch on each channel. The PFL switch allows an individual audio channel to be monitored before the signal reaches the fader. This is useful when adjusting the input gain. When this switch is engaged all other channels are muted. The Solo switch, sometimes called AFL (After-fade Listen) allows the individual channel to be heard at the level set for the mix. These features are useful for identifying and subsequently dealing with audio imperfections.

difference to the true sound of your mix when replayed elsewhere. MONITORING LEVEL There are no hard-and-fast rules when it comes to the SPL at which you monitor. Obviously, if you’re recording and mixing a mellow singer/songwriter there’s little point in monitoring at ear-blistering levels. Similarly, if you’re making tracks designed to be played in clubs or you’re recording a heavy metal band, you won’t get a feel for the final result unless you occasionally increase the volume. However, no matter what you are working on, consistently high SPLs can become fatiguing and cloud your judgement.

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TONE CONTROLS Some monitors feature tone controls that allow you to tailor their sound to suit the room in which they’re being used. Usually these controls will boost or cut the

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RECORDING MONITORS Some sounds can be improved by routing the signal through an external amplifier/speaker and recording the result. This is known as re-amping, and monitors can be used for this technique. The re-amped sound can be used to replace or augment the original. Close-mic’ing the monitors can add punch and drive to previously recorded tracks, while positioning the mic(s) at a distance from the monitors can add some room ambience. Echo chambers operate in much a similar way, albeit in a more reverberant space. MTF

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113

MTF Workshop Song structuring

Logic Pro 9 Workshop

Song structuring tools & techniques

On the disc

Being able to quickly adapt and refine your song’s structure is an essential part of producing music in Logic. Mark Cousins gets close to the edit.

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he cornerstone of any effective piece of music production has to be the structure of a track – defining how a song evolves and develops its musical ideas over time. It’s essential, therefore, that your DAW has a range of song structuring tools, ideally so that structural ideas (whether an extended chorus, a new middle eight or a shortened intro) can be executed in an intuitive and efficient way. As you’d expect, Logic is equipped with a good number of these song structuring tools, but it’s easy to overlook the subtlety in their operation – and, indeed, how the editing buttons can actively be used to enhance the creative process. In this Workshop, therefore, we’re going to explore Logic’s principal song structuring tools – features that enable you to change the ‘macro’ proprieties of a piece of music, rather than the smaller details such as individual notes. As we’ll see, there are plenty of useful tools and tips to be found, but also some important caveats in their operation that, once understood, will allow you to execute song restructuring without the

PRO TIP Use Logic’s Skip Cycle feature as a means of auditioning a cut before you initiate it. You can create a Skip in much the same way as you place locators to cycle, only this time you’ll need to drag backwards rather than forwards over the timeline for any bars that you want to skip.

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To place unassigned editing buttons, drag them from the palette of options into the toolbar.

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Tooling up Before we get started, it’s worth taking time to ensure that the toolbar is set up correctly, and, in particular, whether we’ve got a full set of editing buttons assigned. When it comes to structuring a song, the editing buttons found at the top of the Arrange window are invaluable tools. However, depending on what version of Logic you’re running and how you’ve configured your setup, you can find some small variations in the default toolbar setup. To ensure we’ve got our full assignment of editing buttons, therefore, [Ctrl]+click anywhere on the toolbar and select Customize Toolbar. 1 For this exercise we need access to the following tools: Cut Section, Insert Section, Repeat Section, Insert Silence and Split by Locators. To place any unassigned editing buttons, drag them from the palette of options that should have now opened into the toolbar. It’s worth noting that this arrangement will form your default toolbar every time you open Logic, so spend some time creating a spread of buttons that matches your workflow. 2

Director’s cut

When it comes to structuring a song, Logic Pro’s editing buttons are invaluable tools 1

fear of important details – such as automation moves, tempo changes and even midi notes – getting lost in the process.

To understand Logic’s structuring tools we’re going to explore a number of different musical tasks to see how we can best accomplish them. In the first example, we’re going to remove a passage from our song (in this case, bars 13–17), shortening the overall track length but also keeping any subsequent tempo changes and automation moves in place. Before we make the edit,

Song structuring Workshop MTF

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5 As an alternative to using the editing buttons at the top of the Arrange window, you can use a menu function to achieve the same editing task.

therefore, check that you have the Global Track open and that Automation view is enabled (View>Track Automation), so that we can see how the entirety of the song’s data is being affected. To make the cut, set the locators, then click on the Cut Section editing button found in the top of the Arrange window. 3 If you’ve carried out the Cut Section correctly, Logic should have removed the corresponding passage of music and moved the accompanying automation moves and tempo changes accordingly (indeed, Logic will ask you if you want to erase the old automation data as part of the move). The point to note here is that the changes are applied globally in response to using the editing buttons at the top of the Arrange area – something that will become more relevant as we move through the subsequent tasks. In short, all data is moved, irrespective of what’s been selected. 4 As an alternative to using the editing buttons at the top of the Arrange window, it’s also worth noting that you can use a menu function to achieve the same editing task (Region>Cut/Insert Time>Snip: Cut Selection Between Locators). The point to note here is that the cut is dependent on the currently selected regions – in other words, only the selected regions between the two locators will be cut, with subsequent data shuffled along as before. If you want the menu function to behave like the editing button, therefore, you need to invoke the Select All command first – although, of course, it might be that you want to apply only a selective cut in the first place! 5

PRO TIP Be wary of automation data if you use any menu-based song structuring command. The problem relates to tracks without regions that might have automation on (like an aux channel, for example). To solve the problem, simply insert an empty region over the automation data before you make the edit.

chorus, for example, or a double-length verse. To create a simple repeat, therefore, use the locators to define the area you want to duplicate and select the Repeat Selection editing button at the top of the Arrange window. Once initiated, the rest of the song will be shifted by the desired number of bars, and the area between the locators doubled in length. 7 If you want to relocate a verse/chorus to a new part of the song (or duplicate part of the song in any number of locations), you’ll want to make use of a combination of the Cut Section and Insert Section editing buttons. Firstly, use the locators and the Cut Section editing button as a means of defining and removing the section of the song you want to move. To relocate it, use the playhead to mark the point of insertion and click on the Insert Section editing button. Once pressed, the cut portion of the song is spliced into the song, moving any subsequent material further on in the arrangement. 8

Structural caveats Although the various structural editing tools do a fine job, it’s worth pointing out a caveat in relation to MIDI information, particularly on regions

Silence is golden Another useful option is the menu feature Region>Cut/ Insert Time>Insert Silence Between Locators (replicated by the Insert Silence editing button). This is a great tool for quickly inserting a portion of ‘blank canvas’ into your composition, adding the desired number of empty bars defined by the locators and shifting all subsequent material to the right of this empty space. Again, you might want to differentiate between the onscreen editing button (which applies the process to the entirety of the song) and the use of the menu command (which inserts silence only between the selected regions). 6 As well as cutting and inserting silence, it might be that you want to repeat a given section within the composition – maybe adding a repeat of the last focus Mixing 2013

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MTF Workshop Song structuring

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If you want to relocate a verse/ chorus to a new part of the song, use a combination of the Cut Section and Insert Section editing buttons.

with long sustained notes. To illustrate the issue in action, try inserting two bars of silence (using the Insert Silence editing button) to occur over bars three and four. Once you’ve created the silence, open the Synth Pad region in the Piano Roll editor. Note that even though the region has been resized, the note still plays for its original duration. The issue, therefore, specifically relates to MIDI notes held over the section that you want to edit – something that may or may not be a problem given the track you’re working on. 9 In situations where you think overlapping notes might be an issue, it’s worth using the Split By Locators editing button before you initiate any Silence, Cut or Repeat editing functions. The advantages of splitting the region first is that you get the option to spot any overlapping notes, and, if so, determine what you’d like Logic to do to remedy the situation. Where overlapping notes are found, you’ll be given the option to Shorten, Split or Keep the notes. Of course, your choice will largely be dictated by what you want to achieve. Shorten, for example, would be the logical choice in situations where you want to use the Insert Silence editing button, whereas Split might be useful if you want to carry the segment (with all of the notes intact) over to a new position in the song. 10

The ability to fluidly restructure a composition is an essential part of creative workflow workflow whereby parts of the song are manually spliced and re-edited to form a new structure. Given the way in which the editing tools intelligently grab the entirety of the song’s data (automation, tempo changes and so on) they’re well worth using, enabling you to quickly and confidently restructure a track. Ultimately, unless you’re completely sure of your arrangement and structure before you begin recording, the ability to restructure your composition in such a fluid way has to be an essential part of your creative workflow. Beyond the essentials we’ve explored here, there are also several other creative ways in which you can further extend your interaction with the structure of a song. Options such as Folders – which we partially explored in Issue 101’s Logic Workshop – can be called into service as an alternative means of arranging a song; going even further, you can use features such as the Environment’s Touch Tracks to create ‘live’ arrangements triggered directly from a MIDI keyboard. Whichever method you choose, though, you’ll appreciate the more fluid approach to shaping a song’s structure – not to mention the creative opportunities that it brings. MTF

Free form Despite their prominent position in the Arrange window, it’s easy to overlook the powerful editing buttons and instead turn to a more labour-intensive mouse-driven

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If overlapping notes are an issue, use the Split By Locators editing button before you initiate any other editing functions.

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Toontrack EZmix 2 Reviews MTF

Toontrack

EZmix 2

For PC & Mac

If you need a guiding hand or simply a good mix fast, EZmix 2 could be just what you need. Alex Holmes assesses its talents. EZmix 2 Manufacturer Toontrack Price £115. Upgrade from EZmix 1 £31 Contact Time+Space 01837 55200 Web www.timespace.com Minimum system requirements PC Windows XP (SP3), Pentium 4/ Athlon processor Mac G5 processor, Mac OSX 10.5

Key Features ● All-in-one mixing plug-in ● More than 350 effects chains ● 2 main knobs, input and output trim controls ● New amp, cab and mastering effect modules ● Flexible search engine

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earning the craft of mixing audio can be a daunting task, especially when you factor in not just the number of different types of processing available, but also the sheer amount of choice when it comes to picking a certain manufacturer’s product. We’ve seen a number of plug-ins over the last couple of years that aim to simplify this task by offering a stripped-back interface and limited but carefully selected controls, forcing you into using the most important tool of all – your ears.

Pick’n’Mix EZmix 2, from drum specialists Toontrack, is an all-in-one mixing tool that includes a range of specialist effects chains created by industry pros. There are presets for vocals, drums, keyboards, guitars and bass, plus mastering and buss processing settings, each of which uses a combination of different effects. One of the main improvements over EZmix 1 is the much more advanced and flexible search engine, which now enables you to filter different criteria or search for a specific instrument, effect or genre. Once your preset is selected, all of the complex controls are

Measuring Up Along similar lines are several simplified Artist Series plug-ins from Waves, such as the Tony Maserati and Eddie Kramer Collections ($375), along with the One Knob range ($229), where each effect has a single control. There are also iZotope’s Alloy and Nectar channel strip plug-ins (£169) for mixing and vocal processing, which are more expensive but more flexible.

simplified down to two knobs, with text to either side informing you about what each one does. For example, one knob may increase compression, while the other brings up the level of some parallel distortion. In another scenario,

older DAW sessions. There is also a fully functional demo version, with three projects in different styles, downloadable from the Toontrack website. These songs feature unmixed stems with an instance of EZmix 2 used on each instrument to create a full, pro-sounding mix. All of the effects modules have been provided by plug-in-makers Overloud and the sound quality is superb, especially in the crunchy new guitar amps and cabs. We also found the mastering settings to be a surprisingly quick and effective solution to bulking up the overall mix with minimal artefacts. We welcome the more flexible search options, which become essential when you start to add in the additional presets packs, and the new section showing each module on the right hand-side of the GUI is a nice touch. The main issue we have is with the massive price jump from the previous version, which was available for around £50. To be fair, the upgrade price isn’t that bad, but to more than double the

There are many aspects of EZmix 2 that have been very well thought-out one may add some thick chorus and doubling to a vocal, while the second blends in some reverb and adds brightness to the EQ. The 205 presets from the original have also been added to, with 148 new signal chains that make use of the added cabinet and amp modelling, and the new multiband compressor, mastering limiter, stereo enhancer and aural exciter modules. To help illustrate which processors are present in each setup, the GUI has been extended to show darkened images of amps, effects and pedals that illuminate when in use. There is also a decent number of genre-specific preset packs available for £21.95 each.

The perfect blend There are many aspects of EZmix 2 that have been very well thought-out. Firstly, Toontrack has streamlined its registration process – no longer do you have to open your browser to enter your serial, and more importantly, if you’re upgrading from EZmix 1, the plug-in simply replaces the previous version, meaning backwards-compatibility with

cost for newcomers seems a bit steep, despite the new presets and improved search options.

Speedy Gonzales There’s no denying that EZmix 2 can get good results quickly, especially if you’re mixing band-based music. Pros might find the lack of controls a bit frustrating, and beginners may struggle and end up over-processing due to a lack of visual feedback. However, if you’ve got good ears and can make sensible judgements about what sounds good for each instrument, you’re likely to get on well. MTF

MTF Verdict WHY BUY + Great sound, high-quality effects + Broad range of uses WALK ON BY - Pricey for newcomers - Lack of visual feedback A pricey but great-sounding and easy-to-use mixing plug-in.

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MTF Reviews Yamaha O1V96i

Yamaha

01V96i

The latest incarnation of Yamaha’s new digital console continues the evolution of the 01V. Mike Hillier takes to the stage. 01V96i

Key Features

Manufacturer Yamaha Price £2,620 Contact Yamaha 01908 366700 Web www.yamahaproaudio.com

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amaha’s digital consoles are a common sight in small-tomedium-size venues. They’re both reliable and easy to use, and although not having a single knob for each function might be a little confusing at first, the advantage of being able to save your settings after a soundcheck and simply recall them when the band comes on is actually a much more useful function. Yamaha’s 01V96i is the latest offering in a range that began with the 01V and has been steadily improved and developed, Phantom power switches are located at the rear (top-left) of the O1V96i, which could prove awkward.

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● 12 mic preamps ● Motorised faders ● USB 2.0 ● 96kHz multitrack recording ● VCM effects and REV-X reverbs

bringing new and useful additional features as well as improving existing ones with every iteration.

What’s on the board? Anyone coming from one of the older models will be pleased to see that the overall layout has remained consistent.

channels in layers, with the first controlling the 16 analogue inputs, the second layer controlling the additional 16 inputs, a third ‘Master layer’ for controlling the eight aux sends and eight output busses, and a fourth ‘Remote layer’ that enables you to use the 01V96i as a MIDI controller to take charge of a DAW. The 16 analogue inputs include 12 mic/line XLR inputs (each with an insert jack and 20dB pad) and four additional line-level inputs, which can be paired to form two stereo inputs. Phantom power for the mic inputs is applied in banks of four via recessed switches at the rear. This is somewhat annoying, as you have to use a screwdriver or similar device and reach around the back of the mixer to switch on phantom power. Given the lack of space in many venues, we foresee some house engineers simply choosing to leave phantom power on at all times, which surely works against Yamaha’s intention of making it hard to accidentally switch on. Each channel has its own digital EQ and dynamics processor with three rotary encoders permanently assigned to the EQ frequency, Q and gain, plus buttons for switching between the four EQ bands. This makes it very simple to make EQ changes by quickly selecting whichever channel you want to tweak and grabbing the EQ knobs – although it is easy to accidentally end up EQ’ing the wrong channel the first few times you use it. The dynamics processor is a little less easy to use as it doesn’t have dedicated buttons or knobs. With the channel selected you have to navigate to the gate, compander or compressor window using the Dynamics button and

The 01V96i is the latest offering in a range that has been steadily developed and improved You get 16 motorised channel faders, with dedicated Select, Solo and On/Off buttons. These are assigned to

use the four direction buttons to navigate to the parameter you want to tweak, then dial in the value with the Parameter wheel. Dedicated knobs for the dynamics section, such as those on the EQ section, would be a nice addition, but given the limited space we can see why EQ has been singled out in this respect. EQ tweaks often involve altering frequency and Q simultaneously, which would be impossible with the single Parameter wheel encoder.

Yamaha O1V96i Reviews MTF

In addition to the onboard EQ and compressor, the 01V96i has the full suite of Yamaha VCM effects and REV-X reverbs built-in, with up to four DSP effects engines at 44.1 or 48kHz (only two engines are available at 88.2 or 96kHz sample rates). These effects can be placed as inserts on individual channels, used as send effects, or even placed on the master buss. Considering you have EQ and dynamics on each channel it’s somewhat surprising to see additional EQ and dynamics processors available for the effects engine, but it goes some way to showing what the 01V96i is capable of. The built-in channel dynamics module is all well and good, but with the additional DSP effects you get to choose different compression behaviours by choosing one of the analogue modelled compressors, such as the Comp260 (which has a VCA compression characteristic clearly modelled on a dbx 160) or the Comp276, which is presumably

route any of the direct outputs or the eight stereo buss channels to your DAW for recording. For capturing live performances this feature is a godsend. If you’re using 16 or fewer channels you can capture each one directly in whatever DAW you want to use; you could even capture your sends if you have enough spare channels. Alternatively, if you are using more than 16 channels you can buss them however you want, giving you control over which instruments you feel need their own channels, and which can be bounced down with others. DAW communication works in both ways, so not only do you get a built-in 16-channel recorder with the 01V96i, you can also route 16 channels out of your DAW back into the mixer, making the 01V96i not only a fantastic mixer for use onstage, but also a great project studio mixer. Without any additional A–D or D–A conversion you could feed channels into the 01V96i to be processed by the built-in effects, or

Measuring Up The Yamaha 01V96i manages to squeeze a lot of features into such a compact frame, and it’s no surprise that we see more Yamaha digital consoles in venues than other brands. Roland’s V-Mixing system provides similar flexibility and has the advantage of greater expandability thanks to Roland’s REAC technology. The M-300 has a similar feature set to the 01V96i, with 32 mix channels, eight aux busses, four matrices and EQ and dynamics on each channel with additional optional multi-effects.

The 01V96i has the full suite of Yamaha VCM effects and REV-X reverbs built-in modelled on the UREI 1176 FET compressor, having a very musical breakup when pushed hard, especially on drum busses. We’re not sure quite how faithfully Yamaha has tried to emulate the original hardware, however, because while the hardly subtle naming should give enough of a clue as to which compressor is being modelled, the parameters don’t quite match the originals – often with additional options not available on the hardware, such as a 2:1 ratio on the Comp276. What’s more, to our ears the compressors don’t behave quite like the originals. We don’t see this as a problem, though, as both the Comp260 and Comp276 sound fantastic. Were exact matches to exist in hardware form, they’d almost certainly be just as sought-after as the UREI 1176 or dbx 160.

Beat connection So far, then, everything is pretty much as it was on the older 01V96 VCM. However, Yamaha hasn’t skimped on updates, the most useful being the new 16x16 USB 2.0 audio interface built-in to the 01V96i. This feature lets you

even feed 16-channel stem mixes to the 01V96i for mixing. There are even some great effects that we probably wouldn’t want to use on a live mix, but would love to have access to for our studio mixes, such as the OpenDeck processor, which models a selection of analogue tape machines (including the Studer A800 and Ampex ATR100). Yamaha has even bundled a copy of Cubase AI with the 01V96i to get you recording straight away. And of course, you can now also use the DAW as an additional effects processor for gigs, sending audio from the mixer to your DAW and back, again without any of the additional A–D and D–A conversions and the latency overheads this entails.

Soft solutions Whether you’re using this mixer in a live environment or in the studio, the bundled 01V96i Editor software is a useful tool for saving your mix setups, and even for controlling parameters on the desk from your computer. You get complete control over the console from the software, with improved visual monitoring for each channel while it is connected via USB. When not

connected you can still work on your patching and scene data, making it possible to work on your setup from a laptop while on the way to a gig. The software also makes it possible to switch between different 01V96i units at different venues while keeping the same patch and scene data, effectively meaning that touring bands can maintain consistent patch setups between venues. MTF

Despite the additional features of the O1V96i, the general layout remains similar to the previous model.

MTF Verdict WHY BUY + EQ/dynamics on every channel + Great built-in multi-effects + 16x16 USB audio interfacing WALK ON BY - Few dedicated controls - Easy to accidentally work on the wrong channel - Navigating the various screens is tricky at first A fantastically powerful mixer with some great recording features that make it ideas for both project studios as well as small/mediumsize venues.

★★★★★★★★★★

Method Spot The flexible routing system onboard the 01V96i makes it possible to use studio tricks in a live environment. For example, you could route all your drums to a single stereo buss and apply group compression to this channel, even using another send channel to apply parallel New York-style compression.

focus Mixing 2013

| 119

MTF Reviews sE Munro Egg 150

should be able to see the light fade as you move forwards and backwards or from side to side.

Hard or Soft boiled?

sE Munro

Egg 150 sE has partnered with Andy Munro to hatch a new brand of monitor. Are they all they’re cracked up to be? Mike Hillier finds out. Egg 150 Manufacturer sE Munro Price £1,649 Contact Sonic Distribution 0845 500 2500 Web www.seelectronics.com

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he Egg 150 is sE’s first foray into the world of loudspeakers. The company is better known for its microphone range, which spans entry-level USB models right up to the Rupert Neve collaborations (RNR1 and RN17) that are causing a stir in some of the biggest studios around the world. Jumping from microphone manufacture to loudspeakers might not be an obvious move, but sE had been discussing a range of acoustic treatment products with acoustics expert Andy Munro (who co-founded Dynaudio and designed, among others, Air and Sphere Studios) when the idea of designing a loudspeaker system incorporating ‘extreme ideas that have been bandied about in the acoustics world’ was brought up by Andy.

Shaping up The most striking of these ‘extreme ideas’ Andy had was the egg shape itself. According to Andy, the ideal shape for a loudspeaker is a sphere, but this isn’t practical because of the shape of the drive units. However, the egg shape, or ovoid, has many of the same properties and is therefore a

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Key Features ● 6.5-inch polypropylene LF driver ● 1-inch silk-dome HF driver ● 50W RMS LF ● 50W RMS HF ● Balanced XLR main inputs ● Unbalanced RCA aux inputs

viable alternative. The science behind this explains that the sphere and ovoid shapes eliminate diffraction and resonance, enabling a loudspeaker within them to produce sound that’s free from these distorting effects, giving stunningly clear reproduction. As well as the drive units, the Eggs themselves house a Speakon connector and an ingenious blue LED to aid in positioning. They sit on a simple stand, which can be tilted down to angle the monitors directly towards your ears. Everything else is built-in to the accompanying amp unit – unlike conventional powered monitoring solutions, where the power amplification and controls are housed in the cabinet itself. The absence of these components from the speaker cabinet ensures absolutely no compromises in the cabinet design, furthering its ability to provide perfect acoustic performance.

Included with the Egg Monitoring System is a 2U power amplifier that also incorporates all of the controls for the Eggs. The amp has a gorgeous brushed-metal finish and two big chunky volume pots, one each for the balanced main inputs and the unbalanced aux inputs. There are also two chunky rotary switches, one for switching between the two sources and another that controls the mid frequency response. This switch has three settings: Soft, O and Hard. In the central ‘O’ position the amp retains its perfectly flat mid band; in the Soft position the mids are cut slightly around 2kHz, which sE Munro claims emulates the response of a hi-fi loudspeaker. In the Hard position that 2kHz region is boosted slightly, which sE Munro claims brings the midfrequency response closer to that of the Yamaha NS-10. Comparing the three settings reveals only the slightest of changes. Using the O position as a reference, the Soft setting seemed to flatter a simple jazz track we were using to test the speakers, while on a harder rock track the Soft setting brought the bass slightly more into focus while cutting the punch from the guitars. The difference is subtle, and we had to flick between the two several times to really get a grip on what was happening, so don’t expect to throw out your hi-fi speakers just yet (we have ours plugged into the B output of the desk, enabling us to determine just how much different speakers can affect the sound of a mix).

In position Method Spot The aux input enables you to have a CD or mp3 player permanently attached to your rig for quickly referencing other tracks. It has its own volume control so you can check your references at the same volume without having to adjust levels.

The positioning LED is an excellent aid when it comes to setting up these speakers. It’s one of those simple, elegant solutions that makes you wonder why it’s taken so long for someone to come up with it. Simply get comfortable in the listening position and have a friend angle the Egg until you can see the blue dot at its brightest, then repeat for the other side. This process ensures that your listening position is right in the sweet spot between the two speakers, and you

A blue LED sitting just above the tweeter ensures you get the Eggs optimally positioned.

Egg 150 / Mixing Audio 2nd Ed Reviews MTF

Switching to the Hard setting pushed the vocals and guitars right into focus on our rock reference track, and gave the jazz track a little extra brightness around the otherwise deep vocal. We were actually shocked by how such a tiny change in the EQ response across the mids reminded us of the NS-10s. The bright, harsh top end wasn’t there, but the mids certainly

both across the left and right field, and also in terms of soundstage depth, with some instruments standing back beyond the speakers and others jumping out in front of you. Bass reproduction is outstanding, defying the size of the 6.5-inch woofer and without any hint of breakup. Deep drum’n’bass synth parts were incredibly smooth and the fast transient response

The pure honesty in the sonic reproduction had us going back through many of our old mixes carried the character of the famous studio standards. For mixing we think we’d keep the Eggs in their O reference position most of the time, but could easily imagine switching to the Hard position when focusing on a tricky vocal and guitar mix.

Eggsquisite We were incredibly impressed with the Eggs during our listening tests. It was almost like sitting in on a live recording, with each instrument clearly positioned,

meant details in a mix that we’d not heard through other monitors suddenly became audible. Equally, in the high end the Eggs don’t exhibit any harshness, which might betray a fatiguing quality (something we find quite common in metal-dome tweeters, which tend to exhibit brighter characteristics). The pure honesty in the sonic reproduction had us going back through many of our old mixes, listening and hearing details we’d missed at the time. This brutal honesty isn’t going to flatter

Mixing Audio 2nd Ed Manufacturer Focal Press Price $54.95 Contact Focal 01865 844640 Web www.focalpress.com

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he art of mixing has been covered in many books and we’ve seen plenty over the years. However, there’s no harm in hearing someone different explain theory and techniques, as you’ll inevitably end up picking up some tips you didn’t know before and reinforcing certain points you were unsure about. Mixing Audio 2nd Edition, from Focal Press, is a huge 600-page tome packed with mixing theory, diagrams and explanations, and is now updated with the latest plug-in/software information. Author Roey Izhaki divides the book into three main sections: Concepts and Practices, Tools, and Simple Mixes. Although some may be tempted to skip the opening section, which discusses background theory and why we mix music, it’s actually well worth a read, as it makes us question the purpose of our

Key Features ● 600 pages ● In-depth descriptions and audio examples ● 5 sample mixes plus audio stems ● Detailed illustrations ● Included audio content available from website

your mixes in any sense, but it reveals all the flaws in your recordings and mixes, giving you the opportunity to fix them long before they make it to the final master. And once you’ve got your mixes sounding good on these speakers, you can be pretty sure they’ll translate well onto anything else. Because the Eggs are a fully integrated system, they aren’t the easiest things to pick up and lug from one studio to another (something we know some engineers like to do when working outside their own room). However, they’re far easier to set up once they’re installed, which makes up for the extra carrying. MTF

MTF Verdict WHY BUY + Tight, focused bass + Crystal-clear imaging + Deep soundstage WALK ON BY - No digital input

Measuring Up The Eggs are among the best nearfields we’ve used, easily positioning themselves alongside other favourites of ours such as Unity Audio’s The Rock (£1,899), Genelec’s 8050APM (£2,374) and PMC’s DB1S-AII (£1,825). More speakers in the Egg range are expected, including both smaller and a larger sets, which we look forward to hearing.

If pure honesty is what you want from your monitors, the Eggs are serious contenders.

★★★★★★★★★★

mixes and is very well-written. The main event is the Tools section, which focuses on each element in the engineers toolkit, from EQs and compressors to expanders, distortion and automation. Each topic is covered in an almost obsessive amount of depth, but ultimately offers excellent explanations for each item, with a large number of informative diagrams helping to explain each point. In fact, we’d even go so far as to say the 60-plus pages on compression is one

of the most comprehensive explanations we’ve seen, and anyone struggling with this complex topic will undoubtedly learn a great deal. For the final section, Izhaki takes us through five sample mixes in a variety of genres (rock’n’roll, hip hop/grime, techno, DnB and rock) and explains how he worked on each element of the mix to create the final result. While the book by itself is a worthwhile investment, what seals the deal are the accompanying website and downloadable audio files. You’ll find a DVD’s worth of audio with hundreds of examples helping to explain each and every section of the book. There are also the original mix stems for each track, plus rough mix, final mix and mastered versions. This is an incredibly detailed book that should be essential reading for anyone looking to take their mixes to the next level. MTF

MTF Verdict Incredibly detailed explanations of each tool in the engineer’s arsenal, plus superb audio examples and sample mixes.

★★★★★★★★★★ focus Mixing 2013

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MTF Reviews Mackie DL1608

wirelessly. It’s an intriguing concept, and many touring engineers do have iPads for those long plane journeys and hours spent on tour buses. The hardware itself is relatively minimal, though very sturdy and nicely put together. Around the back you get 16 inputs powered by Onyx mic preamps and with high-end Cirrus Logic converters. These are 12 regular XLR plus four combo inputs and there are, of course, stereo XLR master outs, six aux send jack outputs and a headphone out with its own gain control. Sixteen physical knobs are provided for manual level control, though presumably these are mainly as fallback on occasions when there’s an issue with the wireless or the iPad.

Get connected

Mackie

DL1608

If you want to break free from the FOH desk and happen to own an iPad, Mackie has a solution for you. Hollin Jones tries it out. DL1608

Key Features

Manufacturer Mackie Price £959 Contact Loud Technologies 01494 557398 Web www.mackie.com

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he line between mobile technology and professional music equipment continues to blur, as iOS devices become increasingly used not just as sketchpads, but serious professional tools. Granted, in this context they are most often used as controllers rather than production environments, but nonetheless, the pace of change in the mobile arena is nothing short of staggering. Mackie is a big-hitter in the live sound world so it’s perhaps not surprising that it is among the first to take this fusion of technologies to the next level with the

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● 16 Onyx preamps ● Cirrus Logic converters ● Ultra-low noise, high-headroom design ● 6 aux sends ● DSP-powered plug-ins ● Touchscreen operation ● Use up to 10 iPads

DL1608, an iPad-controlled digital mixer. And when the company says iPad controlled, it means it: there are no physical faders. So it’s a bold step forward, but how does it perform? First: a look at the idea behind the DL1608. Live sound engineers do spend a lot of time running back and forth from the desk, changing levels and checking settings. This mixer aims to free the engineer from standing behind the desk – at least for some of the time. It does this by placing pretty much all the controls inside an iPad app (free to download) and enabling this to work either when docked with the mixer or

An Ethernet port is provided which must be used to connect the mixer to a wireless router if you intend to use the iPad as a wireless as well as a docked controller. Setup is really very easy and begins with downloading the Mackie Master Fader app from the App Store (any iPad model) and connecting the mixer to the router. You then dock the iPad with the mixer and it may perform a firmware update; it did this automatically on our model and the update was seamlessly performed in-app. The iPad also charges while docked. Once that’s completed and the iPad and desk are connected we had full control of the mixer. The desk can be secured with a Kensington lock and the iPad can be secured in the desk if you choose, both of which are good for busy gig environments. The app itself is well-designed and uncluttered and the faders responded very accurately and smoothly on our second-generation iPad. Metering is fast and there’s mute and solo on each channel as well as a master fader that can be flipped to control the six aux outs as well as the reverb/delay channels. The app works wirelessly as well as when connected and there’s a brief sync of data that occurs when you disconnect the iPad from the desk. Overall, though, the transition is seamless.

App, app and away The app can store presets and there are some supplied that can be loaded into any channel; you can also store The 16 inputs are powered by Onyx microphone preamps and equipped with high-end Cirrus Logic converters (12 XLR, four combo inputs, stereo XLR master outs, six aux sends and phones out).

Mackie DL1608 Reviews MTF

Measuring Up Yamaha’s StageMix provides free iPad-based remote control of the company’s M7CL desk, though not of all parameters since the desk is large and complex. You do get fader control, EQ editing with multiple EQ types, gain, tap tempo and more. Yamaha makes some other remote apps for its mixers, too, including the LS9, MGP and CL, all with varying levels of control depending on the complexity of the hardware. None has the small footprint of the Mackie, which is perhaps more suitable for smaller venues.

Undock the iPad from the mixer base and enjoy the freedom of wireless sound control.

your own. Channels can be named and have an image associated with them – this extends to taking pictures with the iPad camera. That might sound gimmicky but it could be incredibly useful. Imagine being able to snap a picture of each band member and then place that picture on their mixer channel. It would make quick reference much easier, especially during a show. You can also take mixer snapshots and there’s a Recall Safe feature that will exempt specific channels from having settings overwritten. Presets can be named, so you can flip between snapshots easily, which is handy for recalling setups for different bands or songs.

The EQ factor Tap on a channel’s EQ window and you open the EQ section. Each channel has its own four-band EQ with draggable points and bell or shelf types, plus adjustable Q values. You also get a high-pass filter and a phase-invert switch, and scrolling down from the EQ window reveals further effects on each channel. There’s a gate – again with draggable points and sliders for easy setup – and a compressor with soft or hard knee plus all the controls you’d expect. These are simple to use and since all processing is carried out using the mixer’s internal DSP, they place no strain on your iPad. Scroll down further again and you will find reverb and delay, assignable on a per-channel basis. The reverb section has multiple types as well as Predelay, Damping, Decay and Rolloff controls along with a master level fader. The delay has multiple types including ping-pong and tape echo, and left/right buttons for manually setting the repeat rate. There are also Feedback and Damping controls. Again, these effects are processed in the mixer itself and the iPad acts as a control surface, so no processing is done on the portable device.

The master fader also features a 31-band graphic EQ with a Draw function, enabling you to enter smooth EQ curves by hand.

Last but not least Other clever features in the app that are worthy of mention include a dedicated iPad channel through which audio can be streamed from any other app that supports background audio, so you could use the iPad as a jukebox. It’s also possible to record a stereo WAV

By leveraging the best aspects of the iPad – touchscreen, menus and wireless – the system simplifies the process of live sound mixing and setup to the point that it’s easier to understand than when looking at a load of physical controls. This applies equally when using the iPad docked (which means the whole unit has a smaller footprint) and when using it wirelessly. Wandering about turning channels up and down, altering EQ and reverb could be hugely liberating, enabling engineers

By leveraging the best aspects of the iPad, this system simplifies the process of live sound mixing and setup file of everything running through the desk, which is a nice touch as it makes recording a show easy. Both of these features work only when the iPad is docked and not when it’s in wireless mode. It’s also possible to use up to ten iPads with a single system so that more than one engineer can work on the system remotely, or musicians can set up their own monitor mixes from the stage. This last concept is a great one – providing that the engineer trusts the band member with remote access to the desk. Getting monitor mixes right by continually shouting to the engineer is tiresome and this could be an interesting solution. In practical terms, the DL1608 is surprisingly easy to set up and use, and won’t prove daunting even to beginners. The app is logically laid-out and provides almost all the tools you might need, though we’d like to see a way to link faders. At present it’s possible to move multiple faders with several fingers, but not link them so that one will move the others by the same amount, which is useful for grouped sources such as drum kits. Since the app can be easily updated by Mackie, it seems likely that something like this could be added in the future.

to account for the idiosyncrasies of specific rooms and meaning that they won’t have to keep running back and forth between stage and desk. Snapshots, camera integration and audio recording are great features that will be genuinely useful, and overall the package feels like a real step forward in the live sound world. If you didn’t use the iPad wirelessly the case would be harder to make for this versus a standard mixer, though the sound quality is excellent. We’re willing to bet, though, that once people get a taste of it, they won’t want to go back. MTF

MTF Verdict WHY BUY + Beautifully designed + Wireless setup is easy + Gentle learning curve + Powerful DSP-based plug-ins + Great preamps; clear sound + Excellent iPad integration + Software is easy to customise WALK ON BY - Not possible to link faders

Method Spot The desk plugs into a wireless router that doesn’t need an internet connection for general use, but creates a local network which the iPad can join to gain access to the mixer functions remotely. When docked, the iPad charges and operates the mixer locally. When undocked it continues to control the mixer functions as long as it remains within range of the router.

A well-designed and beautifully implemented solution for controlling a live setup remotely.

★★★★★★★★★★ focus Mixing 2013

| 123

MTF Reviews Adam A77X

strikes easy to discern and quick transients in the attack frequencies around 2kHz.

Opening up

Adam

A77X

The A77X is the first new monitor to be released by German firm Adam in two years. Huw Price puts it through its paces. A77X

Key Features

Manufacturer Adam Price £1,180 pair Contact Adam Audio 01992 525670 Web www.adam-audio.com

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ircumstances sometimes dictate that monitors need to be flipped horizontally, but sonics generally suffer unless the monitors were designed to be used that way. Hitherto, the only Adam monitors designed for horizontal use were confined to the company’s flagship SX range. The A77X features Adam’s exclusive X-ART (Accelerating Ribbon Technology) tweeter, which has been developed to extend to 50kHz with higher efficiency and higher maximum sound pressure levels. The two 7-inch woofers are identical to those in the A7X, but they don’t cover identical frequency bands. Only one handles the midrange; the other joins in below 400Hz. Power specs are impressive, with a 50-watt A/B amp for the tweeter and 100-watt PWM amps for each woofer. Adam sees the A77X as having studio/ home audio crossover appeal, so unbalanced RCA audio inputs are provided in addition to balanced XLRs. The rear panel also accommodates a gain control for the high frequencies (±4dB) and two shelf filters (±6dB) for high frequencies (5kHz) and low frequencies (300Hz).

Side issues In our excitement to set the A77Xs up we omitted reading the manual. Don’t make the same mistake, because the

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● 2 4-inch carbon/ rohacell woofers ● 1 X-ART tweeter ● Front-mounted input sensitivity control ● Front-mounted power switch ● High- and lowshelf EQ ● Tweeter gain ● Frequency response: 38Hz–50kHz

Measuring Up The A77X’s most obvious competitor would seem to be the Focal Twin6 (£1,210 each) where one of the two 6.5-inch woofers works in large band (midrange–bass) and the other reproduces from 40–150Hz. The Twin6 Be can also be installed vertically or horizontally to respond to the space requirements of a studio. There are many other superb monitors in this price range including the PMC DB1S (£1,825 pair), but you’ll need to consult the manufacturers to find out if they can be used horizontally.

labels ‘Speaker A’ and ‘Speaker B’ on the back panels indicate left and right respectively. If you get them the wrong way around the stereo imaging suffers – just as it would when two-way monitors are placed horizontally with the woofers on the outside.

Similarly, bass lines were solid and even throughout the frequency range without the irritation of vanishing notes or boominess caused by hyped low-mids. At the treble end we found the stock settings a little bit subdued in a test room with a moderate amount of acoustic treatment, but things opened up considerably when we boosted the tweeter by a couple of notches. The midrange may be the area in which the A77Xs will divide opinion. By comparison to our Focal CMS-40s the Adams sounded a little on the thin side. Old-school vocal recordings didn’t come over with quite the sense of body or rich valviness we are used to hearing. Consequently, centrally panned instruments never seemed to burst free of the speakers themselves to sit solidly in the air between the drivers. This is an observation rather than a criticism because what we like

The A77Xs delivered a wide and spacious soundstage with solid left/right imaging Previous experiences with Adam monitors led us to direct our attentions to the low frequencies. Adam favours front-ported cabinets and many users have noticed wind noise at the openings at certain low frequencies. This is often called ‘port chuffing’, but we’re pleased to report that the A77Xs were chuff-free. Apparently, Adam addressed this issue a while back by tweaking the design of the front and back of the ports – and it has done the trick. However, sweeping through test tones indicated a slight kick up in the response around 50Hz and a fairly rapid drop-off by 40Hz, despite the two bass drivers and the generous proportions of the cabinets. Unsurprisingly, the A77Xs delivered a wide and spacious soundstage with solid left/right imaging. However, we didn’t experience the same sense of depth or roominess when we were listening to acoustic recordings. Switching over to some electronica revealed a bottom end with tremendous energy and a nice degree of slam. Kick drums sounded extremely well defined, with differences in the intensity of pedal

to hear and what we need to hear aren’t necessarily the same when it comes to monitoring. All monitors have their own character to an extent and it’s incumbent on the user to learn how they respond in any given space. Not everyone will find the A77X’s slightly lean and two-dimensional presentation an enjoyable experience. However, they are fun and easy to work with, covering all the essential functions required of high-end monitors with aplomb. MTF

MTF Verdict WHY BUY + Power/volume at front + Five-year warranty + Solid, well-defined bass + Wide soundstage + Plenty of power WALK ON BY - Unsuitable for LFE work A high-quality active monitor with no obvious issues, but you may still require a sub for LFE work.

★★★★★★★★★★

Unity Audio The Boulder Reviews MTF

Choice Unity Audio

The Boulder

Can Unity Audio’s second monitor offering better the success of its first? John Pickford is bowled over. The Boulder Manufacturer Unity Audio Price £5,160 Contact Unity Audio 01440 785843 Web www.unityaudio.co.uk

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ack in MTM 102 we reviewed The Rock, Unity’s first foray into the world of professional studio monitor design and manufacture. We were extremely impressed by this small, two-way active design and looked forward to hearing what the company would come up with next. Well, The Boulder is here, and we’re pleased to report that it’s a very fine monitor indeed. Like its smaller brother, The Boulder is a sealed-box (infinite-baffle) active design, with a cabinet made from high-quality Baltic birch plywood and a front baffle made from corian. However, the larger three-way Boulder – measuring 549x256x368mm (HWD) – is constructed from much thicker examples of these materials, which no doubt contributes to its 23kg weight. Two 180mm Elac woofers handle bass duties, with middle and high frequencies reproduced by a new dual-coaxial ribbon design, also manufactured by Elac. The woofer is the same as that used in The Rock, featuring a 0.2mm aluminium foil bonded to a pulp-fibre cone, which allows +/-15mm of cone movement and gives extremely tight and clean bass that extends down to 39Hz. A unique new unit has been designed to take care of both midrange and high frequencies. This is a very interesting design that allows the

Key Features ● Dual coaxial midrange/ tweeter unit ● 2 dedicated bass drivers ● 4 bespoke 100watt amplifiers ● Adjustable midrange and treble controls ● Balanced XLR connectors

Measuring Up Adam Audio’s S4X-V (£4,699) is an active bass-reflex design incorporating the company’s X-ART (Extended Accelerating Ribbon Technology) tweeter along with a HexaCone midrange dome and woofer. Comprehensive controls are provided for altering sensitivity and frequency response.

midrange (and tops) to be reproduced without splitting the task between a mid/bass driver and a tweeter, creating an acoustic point-source. This unit consists of a flat aluminium honeycomb ring radiator coupled with a folded ribbon that extends to 50kHz.

box-speaker we’ve heard. In fact, they sound more like electrostatic panel speakers (which are natural acoustic point-sources) such are their powers of audio resolution. Electrostatics are renowned in hi-fi circles for their pure, accurate sound, but are rarely found in recording studios. Comparing them to ordinary speakers can be initially underwhelming, with seemingly curtailed frequency extremes and a stark yet dry sound. This is because, like The Boulders, they don’t do the things that most conventional speakers get wrong. There’s no boom or bass overhang, no midrange glare and no fizzy treble. What The Boulders offer is natural and faithful reproduction. This can be disconcerting at first because we are used to the hyped-up sound that most monitoring systems deliver. Listening to familiar recordings highlights minute dynamic and tonal shifts that most speakers gloss over. That subtle bit of fader-riding becomes transparently obvious through these incredibly detailed monitors. Our only reservation is how The Boulders might

Astonishingly accurate audio reproduction that can be a little bit frightening The Boulder’s amplifier modules are the same as those in The Rock, designed by Tim de Paravicini of Esoteric Audio Research. Tim is a highly respected designer who’s known for creating some of the best valve amps available today, but the four 100-watt amps powering The Boulders are bespoke solid-state devices. At the back of the monitor, below the large heatsink, are sockets for a balanced XLR lead and an IEC mains lead, as well as a gain pot and switches that allow 2dB of boost/cut at high frequencies around 10kHz, and 2.5dB of boost/cut over the broad midrange.

Rock on Following our review of The Rock – which we praised for its incredibly fast transient response and superior detail retrieval – we spoke with Unity’s Kevin Walker, who informed us that The Boulder would not only give more bass than the smaller monitor, but also sound even more detailed and accurate. He wasn’t wrong. It’s an ear-opening experience hearing The Boulders for the first time as they sound like no other

integrate within an existing monitoring setup. Flipping between them and, say, Yamaha NS10s is a complete waste of time. The sound is chalk-and-cheese. Used in isolation, or with The Rocks as alternative nearfields, The Boulders provide astonishingly accurate audio reproduction that can be a little bit frightening. Get used to them and they will make you a better engineer. MTF

MTF Verdict WHY BUY + Incredible resolution of detail + Super-fast transient response + Uncoloured midrange + Tight, clean and deep bass WALK ON BY - This level of honesty might be too much for some The Boulder has electrostatic-like levels of clarity and resolution not available from any other boxspeaker we’ve heard. What you hear is the truth, the whole truth and nothing but the truth...

★★★★★★★★★★ focus Mixing 2013

| 125

MTF Reviews Focal SM9

genuine vintage Neumann on hand for comparison. Listening through our regular monitors, the two mics sounded almost indistinguishable, but switching over to the SM9s revealed subtle differences in the high-frequency response. The acoustic image was so solid that the mono signals seemed to be coming from a centre speaker.

Stand corrected

Focal

SM9

The SM9 and its innovative two-in-one approach tops Focal’s range of active studio monitors. Huw Price finds focus. SM9

Key Features

Manufacturer Focal Price £4,290 pair Contact SCV London 020 8418 1470 Web www.focalprofessional.com

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he SM9 is billed as two independent monitor systems housed in a single enclosure. On the one side there’s a two-way monitor with a 6.5-inch bass/ midrange driver and a 1-inch pure beryllium tweeter. On the other side there’s an 8-inch bass woofer with an 11-inch passive radiator. The fun part is that you can switch between the two-way system and the full system, where all four drivers are in play. The SM9 has numerous frequency response adjustment controls, which we’ll describe a bit later on. There’s even an EQ in/flat response switch, conveniently located next to the half-/ full-system switch and the standby switch located on the side of the cabinet. As one SM9 owner observed: it’s almost like having four monitor systems in one. And at 35kg each they weigh about as much as four monitors, and measuring 320 x 490 x 390mm they take up almost as much space. The quoted frequency response of the SM9 is impressive, ranging from 30Hz to 40kHz with all drivers engaged. We can’t comment about the upper frequencies because we’re merely human, but test tones were clearly audible below 30Hz. With 400 watts driving the woofer and 200 watts evenly distributed between the bass/midrange driver and the tweeter, you’ll soon identify any loose fittings in your room.

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● Frequency response: 30Hz–40kHz ● Input: XLR with -10dB/+4dB sensitivity switch ● Total onboard power: 600 watts ● Onboard EQ (bypassable) ● Standby switch ● Two-way driver switching

Measuring Up Active nearfield monitors with a built-in subwoofer are rare indeed, but Barefoot Sound has made waves with the Micro Main 27 (£7,200 pair). Billed as a three-way active monitor with dual integral (active) subs it can be mounted vertically or horizontally and onboard power is rated at 810 watts. Frequency response reaches down to 30Hz. Alternatively, check out the more compact Micro Main 35 (£5,040 pair), which has a 5-inch mid/bass driver and a 1-inch tweeter on the front and reaches down to 35Hz with the help of dual 7-inch subwoofers mounted on the sides of the cabinet.

An XLR socket is provided for balanced audio input along with a -10dB/+4dB sensitivity switch. First up is the High Pass Filter, with the Full Range position disabling the HPF. The filter slope for the three other positions (45Hz–60Hz–90Hz) is 12dB/octave.

Shelf life The LF Shelving potentiometer adjusts frequency levels between 30Hz and 250Hz; LF EQ adjusts the bass level

Next we decided to try some mixing. Recalling a recently completed mix in Logic, we embarked on some finetuning. Through the SM9s it seemed that the bass could come up a tad, along with the drum overheads. The lead vocal also exhibited occasional sibilance. The adjusted mix was burned to CD along with the original to compare on a variety of systems. Bearing in mind that we had been happy with the original mix, it has to be said that the minor alterations the SM9s had prompted us to make improved it. The SM9 mix simply translated better. While low frequencies are said to be non-directional, hearing ultra-low frequencies from the SM9s at ear level rather than feeling them in your chest takes some getting used to. The sensation is odder still when you’re standing up. Besides that, we have

Full-frequency monitors with a unique feature set and extraordinary sound quality (with a central frequency of 50Hz); and LMF EQ sets the level of the mid bass frequencies around 160Hz. The central frequency for MF EQ is 1kHz and HF Shelving affects frequency levels between 4.5kHz and 40kHz. The Q varies from band to band, but all of the control pots are indented, allowing +/-3dB adjustments in 0.5dB steps. Our expectations were high and the SM9s didn’t disappoint. Essentially they sound like utterly superb nearfield monitors – but without the usual missing frequencies. If your studio already has a pair of soffit-mounted big monitors, the SM9s could make them all but redundant. Basically they eliminate the guesswork. Besides listening tests involving our usual test tracks, we used the SM9s during a couple of microphone reviews. The mic in question was a repro of a classic Neumann, and we had a

nothing negative to report about these speakers. The imaging is surgically precise, the effortless transparency is world-class and they’re a pleasure to work with. MTF

MTF Verdict WHY BUY + Stunning clarity/imaging + Comprehensive EQ controls + Side-mounted standby switch + EQ bypass switch + Clever two-way option switch + No sub required + Fast and articulate bass WALK ON BY - Fairly large (and heavy) - Quite expensive for some budgets Full-frequency nearfield monitors with a unique feature set and extraordinary sound quality.

★★★★★★★★★★

Fostex PM8.4.1 Reviews MTF

Fostex

PM8.4.1

The latest product from Fostex marks the company’s first foray into three-way territory. Huw Price lends them an ear. PM8.4.1 Manufacturer Fostex Price £398 a pair Contact SCV London 020 8418 1470 Web www.fostex.com

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lthough the ‘PM8.4.1’ designation may appear to be describing the latest in surround sound formats, it’s actually the name of Fostex’s debut in the three-way active nearfield monitor market. Drivers comprise an 8-inch woofer, 4-inch midrange driver and 3/4-inch tweeter, powered by independent 60-, 18- and 18-watt amplifiers respectively. An integral channel divider provides overlapped frequency crossover points for smooth transitions between the drive units. With the midrange and highfrequency drivers arranged symmetrically, the PM8.4.1s must be configured for left and right operation, with the tweeters sitting on the outside in upright mode. If your particular working environment dictates sideways placement, however, all you need to do is swap the speakers over so that the tweeters are placed above the midrange driver. Interestingly, the inside of the speaker enclosure is divided into different sections for the woofer and the midrange driver. This has been done to minimise interference between the drivers. The woofer and midrange diaphragms are both constructed from the same highly rigid and elasticpapered Kevlar that was designed by Fostex’s engineers to ‘reproduce any type of input source’. The tweeter is a

Key Features ● LF driver: 8-inch Kevlar cone ● Midrange driver: 4-inch Kevlar cone ● HF driver: 3/4inch soft dome ● Frequency response: 50Hz–25kHz

Measuring Up If you’re keen to find three-way monitors you’ll probably struggle to find others in this price range. Twoway alternatives could include the lower-powered KRK VXT 4 (£479/ pair), which has integrated mount support, ground lift, auto on/off, switchable limiter, 4-inch woven Kevlar woofer, 1-inch silkdome neodymium tweeter, and one XLR/1/4-inch combo input. The Yamaha MSP5 Studio (£399/pair) is a bass-reflex monitor with 67 watts of power running a 5-inch woofer and a 1-inch tweeter, with controls for level, low trim and high trim. The Adam A5X (£524/pair) is a bass-reflex speaker with a 5.5-inch carbon/rohacell/ glass woofer (50 watts) and X-ART tweeter (50W), plus tweeter and woofer shelving filters.

soft-dome type with a diaphragm constructed from UFLC (Urethane Film Laminated Cloth).

Port side The enclosures themselves are fairly sizeable, measuring 270 (W) x 432 (H) x 290mm(D). You’ll need sturdy speaker

the low end certainly doesn’t sound out of proportion. In fact, there’s a fairly abrupt cut-off below 40Hz but the low mids and bass did sound wellcontrolled and even in our listening room. The PM8.4.1s do exhibit a touch of port-chuffing around the 60Hz region, but it’s barely noticeable in use (to put this in perspective, we have heard far worse from active monitors costing several hundred pounds more than the PM8.4.1s). Moving up to the midrange, the PM8.4.1s provide a clear and transparent picture that presents vocals, guitars and snare drums nicely. By comparison to some small-box nearfields the midrange isn’t quite as upfront and dominant, but we didn’t find it hard to adapt. The treble is clear and defined, but we found the tone a little strident with the tweeters beaming directly into our ears from a distance of just over a metre. However, notching down the treble improved things. Overall we’re very taken with the Fostex PM8.4.1s and they sound as solid as they look. What’s more, they’re

We’re very taken with the PM8.4.1s and they sound as solid as they look stands, too, because the PM8.4.1s weigh in at 13.6kg each. The cabinets are front-ported with rounded edges at the opening of the tubes. All the control action is located on the rear panel, where you’ll find the IEC socket with accompanying power switch and three step-level adjustments for high frequencies (around 10kHz) at +1/0/-1dB and low frequencies at +3/0/-3dB (around 60Hz). There are two input sockets for XLR and TRS jack connectors. The XLR socket operates at an input level of +4dB and the TRS jack operates at -10dB. Both have input impedance rated at 20k ohms or more.

Loud and clear The PM8.4.1s are big speakers so you’d obviously hope for a big sound. In that regard they don’t disappoint. Given the ported design and the size of the cabinets we were slightly concerned that the bass would have been overcooked, but the Fostex engineers have shown admirable restraint and

fun and easy to work with and they can get pretty loud whenever the need arises. They arguably fall behind some more expensive monitors because acoustic instruments miss a little of their natural timbre and there’s a slight lack of depth in the overall image. However, left/right imaging is commendably crisp and, all things considered, they’re certainly a lot of speaker for the money. MTF

MTF Verdict WHY BUY + Solidly made + Ample power + Clear sound quality + Tweakable frequency response WALK ON BY - Power switch at rear - Volume control at rear - Slightly strident treble Three-way active monitors offering ample power, impressive sound quality and good value for money.

★★★★★★★★★★ focus Mixing 2013

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MTF Studio Icons Yamaha NS-10

Studio Icons

Yamaha NS-10

Yamaha’s enduring NS-10M has become the reference by which all other small nearfield monitors are judged.

We close this Mixing Focus by looking back at a monitor that has mixed more songs than any other. John Pickford explains why.

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ith its iconic white woofer, the Yamaha NS-10 loudspeaker is one of the more instantly recognisable pieces of audio kit, and undoubtedly the most well known studio monitor. Launched in 1978 as a bookshelf speaker designed for domestic hi-fi use, it quickly became popular among freelance studio engineers who wanted a portable reference speaker to transport from studio to studio. The original NS-10 was designed by Akira Nakamura, who also designed Yamaha’s flagship NS-1000 monitor, a sophisticated design that made use of beryllium in its midrange dome and tweeter. The large three-way NS-1000 is still highly regarded as a full-range hi-hi speaker, however, the NS-10 proved to be unpopular in the world of home audio and could well have been quickly discontinued had it not been embraced by professional studio engineers. It’s unknown who first used the speaker as a professional monitoring tool, but credit is usually given to studio legend Bob Clearmountain for kick-starting the NS-10 movement. The speaker itself is a two-way, closed-box (infinitebaffle) design that does not make use of a bass-reflex port to aid the low-end response, which is flat down to around 60Hz. Early NS-10s suffered from poor integration between the 3.5cm tweeter and 18cm mid-bass driver, which caused unwelcome phasing issues. What’s more, many users found the treble to be harsh and over-exposed, making long-term listening somewhat fatiguing. Some technical engineers actually modified the speaker’s crossover to tame the top-end response, while others resorted to covering the offending tweeter with tissue paper. Again, it’s Bob Clearmountain who is thought to be the first to employ this trick, and as this unorthodox modification became well known by word of mouth, engineers experimented to find the perfect tissue. Audio expert Bob Hodas even carried out extensive tests in the 1980s to determine the effect that different types of tissue had on the NS-10’s frequency response,

The NS10M features a unique sheet-formed white cone 18cm woofer and a 3.5 cm soft dome tweeter.

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going as far as publishing graphs to demonstrate his findings. Yamaha responded to this tissue-paper phenomenon by redesigning the crossover and tweeter to give a smoother treble response. At the same time the company altered the front-baffle logo so that it was legible when the speaker was placed horizontally, the generally preferred method of use. This tweaked design, known as the NS-10M (M for monitor), was marketed as a professional studio monitor and remained in production until 2001. So what does it sound like? The most often-quoted phrase is ‘if it sounds good on NS-10s, it’ll sound good anywhere’. This is largely due to the speaker’s midrange performance, which is ruthlessly revealing, making poor mixing judgements stick out like a sore thumb. The bottom end is not particularly extended, but the bass that is present is taut and fairly dry, lending the speaker a sprightly nature.

NS-10s remain something of a Marmite monitor – you either love them or hate them The treble is well-lit, providing bags of detail, although the extreme top end does not possess much in the way of air or sparkle. When driven hard at ear-splitting levels they have a tendency to shout and can sound compressed, but at moderate levels they are fast, revealing and capable of dealing with small dynamic shifts remarkably well. Despite their ubiquity, NS-10s remain something of a Marmite monitor – you either love them or hate them. Some engineers find their prominent midrange fatiguing to listen to for long periods, preferring the less brutal presentation of more easy-going monitors. Others, especially those freelancers who work in many different studios, are aware that familiarity is often the key to accurate monitoring. Whether used as nearfield alternative monitors to complement full-range speakers in professional studios or as the main monitors in project studios worldwide, Yamaha NS-10s have been used to mix many thousands of recordings, ranging from amateur bedroom productions to multi-platinumselling hits. No other studio monitor comes close to the NS-10 in terms of popularity, and its familiar tone – like it or not – will ensure that it continues to be the reference by which all other small nearfield monitors are judged. MTF

Next NextIssue Issue MTF MTF

100% PURE RECORDING MIXING

OnOn sale sale 7 February 5 April 2013, 2013, £8.99 £8.99 with with free free DVD. DVD. Available at WHSmith (UK), Barnes & Noble (USA) and all good bookstores in Australia, Canada, and throughout Europe. Or order online at www.musictechmag.co.uk/mtm/focus

MTF On Your DVD

The modern computer can be a powerhouse of high-end mixing tools, from analogue modelled EQs and compressors, to intelligent, transparent digital limiters, reverbs and more. Yet all your powerful plug-ins are no use if you don’t know how to use them – and choosing the right ones for you can be a minefield too. Your free Mixing Focus DVD has more than three hours of pro mixing tips and tricks, the latest software demos and freeware tools, and 1GB of promo videos. You’ll also find highly polished royalty-free samples, plus all the files you need to follow the tutorials. MTF What’s on the disc More than three and a quarter hours of pro video tuition

COMPRESSION & REVERB EXPLAINED The mixing experts at Groove 3 have provided a host of videos, including five chapters on compressor controls and buss compression taken from Compression Explained and three chapters on dialing in reverb from the Reverb Explained tutorial.

PRO TOOLS 10 AND SONAR X1 Groove 3 takes an in-depth look at mixing the piano, keyboard and bass parts of a track using Pro Tools, plus mixing a rock vocal in Sonar X1 using the pro channel EQ and compression, along with the Softube Saturation knob.

LOGIC PRO macProVideo has supplied tutorials covering a range of topics including track doubling and dynamic EQ tricks in Logic Pro. Please note, these videos require the NED player, which is only compatible with OSX, and 32-bit Windows systems.

CUBASE, LIVE, REASON & PRO TOOLS Other videos from macProVideo take a look at distortion effects in Ableton Live, automation, mixer, and channel controls in Cubase 6, the EQ section and Filter Faders in Reason 6, as well as routing techniques in Pro Tools 10.

MTM PRO-TECHNIQUE The MTM team take a visit to two very different producers studios to pick up some top tips. Far Too Loud shows us some heavy moombahcore bass and drum production tricks, and Goetz Botzenhardt discusses his workflow when mixing down 5.1 soundtracks.

LOOPTV SOFTWARE OVERVIEWS The Plugin Boutique and producer Dom Kane take a look at Kilohearts’ Faturator saturation plug-in, the Sonitex STX1260 multi-fx distortion unit, and a whole heap of plug-ins that are included in the Waves Gold Bundle.

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On Your DVD MTF

MTF Your Disc

For PC & Mac

MTF DVD 28 Mixing 2013

MTF DVD 28 Mixing 2013 PROMOTIONAL VIDEOS

We’ve got over 1GB of videos showcasing the latest mixing software and hardware. There’s high-end processing tools and multi effects from iZotope, Eventide, FabFilter, and Pro Audio DSP, vintage tape emulation from Slate Digital, and custom artist-built guitar presets from Toontrack. Plus, Mackie’s iPad mixing dock, mixing and music apps from Allen & Heath, and Image Line, hardware controllers from Focusrite, and more.

COPYRIGHT ANTHEM PUBLISHING © 2013 FAULTY DISC? Check www.musictechmag.co.uk for known issues Return to: Anthem Publishing (disc returns), Suite 6, Piccadilly House, London Road, Bath BA1 6PL, UK

SOFTWARE DEMOS

From cutting edge effects processors to meticulously modelled vintage gear, we’ve rounded up a range of demo and freeware plug-ins for you to try out. You’ll find plenty of EQs, compressors, limiters, stereo spread tools, audio analysers, reverbs, and saturation plug-ins.

For PC & Mac

USING OUR WORKSHOPS

Whether you’re mixing in Cubase, Pro Tools, Logic, Live, Reason, or Sonar, we’ve got you covered with a host of mixing workshops. Where appropriate you’ll find hi-res images, project files and audio on the disc so you can follow along at home. Do please be sure to copy all the files to your computer before opening a project.

SAMPLES

One of the best ways to fine tune your skills is to listen to highly polished audio crafted by the pros. We’ve included over 400MB of royalty-free samples from Sample Magic, Zero-G, Sounds of Revolution, and Nine Volt Audio covering a range of styles and instruments for you to use.

MTF DVD 28 Mixing 2013 COPYRIGHT ANTHEM PUBLISHING © 2013 FAULTY DISC? Check www.musictechmag.co.uk for known issues Return to: Anthem Publishing (disc returns), Suite 6, Piccadilly House, London Road, Bath BA1 6PL, UK

On the disc

YOUR DVD CONTENT FILES

ZIP FILES To maximise the amount of content we can bring you on each DVD, the video, Workshop and samples files are supplied compressed (‘zipped’). Mac users should be able to decompress zip files simply by double-clicking on them; PC users may need to download a utility such as WinZip (www.winzip.com). WORKSHOP FILES The software Workshops that feature in each issue of MTF are almost always accompanied by files and audio so you can work through them on your system. These files are zipped to reduce the space they occupy on the DVD.

Download them to your hard drive and unzip them to access the individual files (remembering to eject the DVD to prevent your computer from slowing down).

WHAT IS ROYALTY-FREE?

Any MTF DVD content marked ‘royalty-free’ can be used in your own original compositions (even commercial ones). You may not, however, resell these samples in any other form.

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endeavour to supply you with a replacement disc immediately. Please note that we’re unable to provide technical support for the software on the MTF DVD – please check our website at www.musictechmag.co.uk for any known problems.

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If your disc is missing, contact us at [email protected] with your full postal address and the issue number.

In the unlikely event that your disc is defective, please return it to: Disc Returns, Anthem Publishing, Suite 6, Piccadilly House, London, Bath BA1 6PL. We will focus Mixing 2013

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