Compression Made Easy

Demystifying Compressor Controls & Parameters

Technique : Effects / Processing

Getting precisely the results you want from compressors can often be the key to a tight, modern-sounding mix. We explain what the controls are designed to achieve and how they relate to what you hear. Audio Examples On-line — We’ve placed a number of audio files to accompany this article on the SOS web site at www.soundonsound.com/sos/sep09/articles/compressioneasyaudio.htm

Mike Senior

Few things confuse home-studio owners more when mixing than compression. It’s easy to find out what compressor parameters do in the abstract, but answers to more critical (and less tangible) questions are thinner on the ground. How do you know whether or when to compress? How much compression is enough, or too much? What are the right attack and release times? This article should lend a helping hand with such questions. Instead of beginning by explaining about compressor design, as most tutorials seem to do, I’ll start with some common mixing difficulties and show how the main compressor parameters provide tools to overcome them.

Dynamic Range

Possibly the single greatest challenge facing the mix engineer is finding the right balance. On the face of it, the task should be simple: you adjust the channel faders until you hear everything in the right proportion. In most styles of music, though, the chances of setting up a reliable ‘static’ balance like this are slimmer than a gerbil’s toothpick.

Let’s take the example of a lead vocal where some words are mumbled. If you set your fader so that the majority of the vocal is nicely audible in your mix, the lower-level mumbled words will start playing ‘hide and seek’. If you push the level up so that the mumbled syllables come though, the rest of the vocal will eat Manhattan. No single fader-setting gives a good balance because the difference between the highest and lowest signal levels (the ‘dynamic range’) is too large.

Compressors remedy this by reducing a sound’s dynamic range: compression will reduce the level differences between the mumbled and unmumbled words, making it easier to find a static fader setting that works. The compressor does this by turning down (or ‘compressing’) the louder signals so that they match the quieter signals more closely — and all it needs from you is an indication of which signals you think are too loud. Every compressor has a control for this, but it can be implemented in different ways...

Compression, Peak Reduction, Threshold & Input Gain

These plug-in compressors illustrate the three most common control layouts for compressors (whether hardware or software). The Antress Painkiller takes the approach of the famous Teletronix LA2A compressor: you turn up the peak-reduction knob to increase the amount of compression. Waves’ Renaissance Compressor takes the approach of SSL’s popular bus compressor, where you get more compression as you bring the threshold down. Cubase’s Vintage Compressor uses another approach, as found in the UREI 1176, whereby an input gain control pushes the signal up against a fixed compression threshold to increase the amount of compression. Each approach also incorporates a gain control to compensate for the unavoidable volume changes caused by the compression process.

The simplest approach is to have a single control that makes the compressor react to more of the signal the more you twist the dial: at the minimum setting, the signal remains uncompressed; as you turn the control up, only the signal peaks are reduced in level; and as it reaches the maximum setting, all but the softest signals are sat on. This knob is sometimes called ‘Compression’ (on some JoeMeek and Focusrite Platinum units, for example, as well as plug-ins such as Digital Fishphones’ Blockfish), but on the iconic Teletronix LA2A it was called ‘Peak Reduction’, a term that can be found on other hardware and software (for example Tin Brooke Tales’ TLS 3127 LEA). Both terms make sense, because you get more compression (more reduction of peak levels) as you crank the knob.

A more common approach is for compressors to have a ‘Threshold’ control, which works the opposite way around: you get more compression as you turn the knob down, because the threshold is the level above which the compressor considers the signal too loud. With the threshold set to maximum, very little is considered too loud, so precious little compression occurs; but set it to minimum and most things will be too loud, and the level of all but the very softest bits will be reduced.

A final control layout you may encounter is the one used on the Urei 1176LN and which now appears on many of the plug-ins it has inspired, such as Cubase’s Vintage Compressor. In this design, there’s a fixed signal level, above which the compressor will turn the volume down. The only way to specify the amount of compression is to adjust the input level with an input-gain control. The more you turn up this control, the more the signal exceeds the threshold, and the more compression you get.

If the action of the Input Gain control sounds rather like that of the Compression and Peak Reduction controls above, you’re not wrong. They all increase the amount of compression as you turn them up. The crucial difference is that as you compress with a Compression or Peak Reduction control (or a Threshold control), the overall processed sound tends to reduce in level, while with the Input Gain approach the overall signal level gets louder. For this reason, I tend to steer newcomers to compression away from 1176LN-style processors, because the overall level increase that you get as you turn up the Input Gain control always tends to give the impression that your processing is improving the sound — even if the amount of compression is inappropriate. That aside, it pays to get comfortable with all three common control setups, so that you have the widest available choice of different compressors while mixing.

Gain-compensation Controls

Irrespective of which compressor you choose, you’ll almost always find that squishing a signal’s dynamic range to taste will change its apparent overall level. You could use the channel fader to compensate for this, but because of the large level changes that compressors can bring about, this is rarely a good solution in the real world. Almost all compressors include a simple output gain control, usually called Output Gain, or Make-up Gain (or simply Gain, or Make-up), but whatever it’s called, all it does is allow you to reinstate the compressed signal to roughly its former level in the mix.

Having said that, there are a few ‘one-knob’ compressor designs with only a single Compression control. What you’ll typically find if you use one of these is that the designers have implemented some kind of automatic Make-up Gain function behind the scenes, keeping the subjective level of the audio consistent no matter how much compression you’ve dialled in. This does make the compressor simpler to control, but these designs almost always make compressed signals feel louder than uncompressed ones which, once again, can encourage inexperienced users to dial in more compression than is necessary.

That’s a lot of explanation for only two controls, but I make no apologies for that, because they can actually deal with a lot of compression tasks on their own. Furthermore, if you find compression confusing, these controls make it possible to make useful headway with any compression presets in your recording software. So before we discuss any other compression parameters or controls, let’s look at how you can make the most out of what we’ve already covered.

Balancing, Multing & Compressor Choice

Here you can see a single lead vocal multed across three tracks to allow for different vocal processing and levels for different sections of a song. In some cases, multing tracks can allow you to avoid compressing at all, but even when it doesn’t, it can still make it easier to improve your compression results, because you can better adapt each multed track’s compression to the context of its section of the song.

First, let me repeat myself: concentrate on the balance of the tracks in your mix. If the tracks balance fine as they are, no-one will arrest you for leaving them alone! The trick is to wait until you spot a fader that you can’t really find a suitable level for (the sound may disappear in some places, or have sections that feel too loud): that’s where you may need to compress. In the first instance, though, see if you can solve the level problem by splitting the audio onto two different tracks and balancing them separately. This is a common technique often referred to as ‘multing’. It’s easily done in most DAWs, and can head off a lot of rookie compression mistakes. Again, you may find that you don’t need any compression at all to find a balance that works.

Multing can solve a lot of problems on its own, but quickly gets very fiddly if you try to use it to deal with lots of short-term balance problems (lots of single notes or words that are too loud or quiet), and this is where the automatic processing offered by a compressor can begin to complement multing. For example, you might mult out a guitar solo from the main guitar track to give it a higher fader level, but still compress that solo so that a few over-zealous notes don’t pop out too far. So try multing to solve balance problems at first, but don’t be afraid to let compression take over when it suits the job better.

Which compressor should you choose? At the risk of uttering studio heresy, there are more important things to worry about when starting out than the model of compressor you use. You might as well use whichever one comes to hand, albeit bearing in mind the advice I offered above regarding 1176LN-style or one-knob designs. It’s also worth finding one that has a gain-reduction meter of some type, because this helps you to see when and how hard the compressor is turning down the level of louder signals. Gain-reduction displays are typically in the form of VU-style moving-coil meters or LED bargraphs, and sometimes the compressor’s normal level-meter can be switched to show gain-reduction instead. If you’re still in a quandary, and have access to a VST host, you could try TBT’s freeware TLS 3127 LEA as a starting point, because it has a pretty simple interface, where the Peak Reduction and Gain controls are right in your face.

Starting To Mix With Compression

These three compressors (Digital Fishphones’ Blockfish and Tin Brooke Tales’ TLS 2095 LA and TLS 3127 LEA) all sound quite different even for similar amounts of gain reduction — and you don’t need to know why this is to take advantage of it. 

Now insert your chosen compressor into the channel in question, and if there are presets available, select something likely-looking — again, there’s no need to give it too much thought for now, just go with your gut. To start with, pile on a fair dollop of compression using the Threshold control, so that the gain-reduction meter (usually calibrated in decibels) shows at least 6dB of compression occurring on signal peaks. Once this is set up, adjust the Make-up Gain control to compensate roughly for any overall change in signal level caused by the compression. (For the sake of discussion I’ll refer to Threshold and Make-up Gain controls, but the same principles apply with a different control set.)

Now’s the time to return to the main question: can you now find a level for the channel fader that delivers a better mix balance? There are a lot of possible answers to this question, so let’s look at how you deal with each in practice. Clearly, if your compression solves your balance problem, the job is done, but even if you think that this is the case, it makes sense to try turning the threshold back up a little and seeing how little compression you can get away with. Pushing your channel compressors too hard is a common mistake that can slowly suck the life out of a mix if it’s duplicated across all your tracks, so it pays in the long run to be a little wary.

If the balance problems can’t be solved, try rolling the threshold down further, to see if that makes it easier to find a decent fader level. Feel free to completely max out the control if you like, even it if makes the result sound rather unnatural for the moment: the important thing is to keep concentrating on the balance, and whether the compression can deliver the static fader-level you’re after. Again, if you can get the balance you’re happy with, and you find any side-effects of the compression appealing (as they often can be), then consider yourself a hero, and turn your attention to the rest of the instruments.

On the other hand, although you may find an appropriate balance through heavy compression, you could find that the processing isn’t doing nice things to the instrument’s sound. Perhaps it’s making the performance feel lumpy and unmusical, or altering the tonality in some unpleasant way. The remedy? Just switch to a new compressor or preset and have another crack. Different compressors and presets can respond very differently for similar settings of our two main compressor controls — and you don’t really need to understand why this is to reap the benefits. Try out a few different ones and choose the one that does the job best. With a little experience, you’ll soon build a shortlist of personal favourites for different instruments. In fact, if you’re working with a computer sequencer, you may well be able to set up several different compressors in the signal path for comparison purposes, bypassing them as necessary.

When Compression Is Not The Answer

Compression is usually not enough to deliver the kind of up-front lead-vocal sound that many modern styles demand, so rather than trying to push your vocal compressor too hard, finesse any final balance tweaks using careful level automation. Here you can see some fairly typical level automation for the lead-vocal phrase featured in the audio files that accompany this article on the SOS site.

There are almost always cases where no matter which compressor you use, or how you set the threshold, you can’t find a good fader setting for the track in the mix, even if you’ve already done some sensible multing. This is the point at which a lot of inexperienced engineers throw in the towel and simply settle for a compromise between dodgy balance and unmusical processing side-effects. What you need to realise, though, is that your mix is probably trying to tell you that simple compression is not what you’re looking for. Furthermore, if you’ve already tried a few different compressors and/or presets, then it’s pretty unlikely that any of the other compression controls are actually going to salvage the situation. So the best thing is to step away from the compressor with your hands in the air, and look for alternative processing options instead.

An article about compression isn’t the best place to go into all the other processes you might use at mixdown, but here’s one example to demonstrate what I’m talking about. If you have a bass guitar recording with loads of very low frequencies, it’ll be difficult in most mixes to find a fader level where the bass is audible enough in the mid-range without absolutely swamping everything else at the low end at the same time. No matter how much you compress that sound, you’re unlikely to solve the problem because you won’t be fundamentally changing the balance of the instrument’s frequency content. It’s much better to address this problem with EQ first. You’ll be able to tell when you’re doing the right things with the EQ when it starts getting easier to find a suitable fader level for the bass, and you might, again, discover that you don’t need any compression at all.

Another very common occasion where compression can’t provide a complete solution to mix balance issues is when dealing with very critical tracks, such as (typically) lead vocals. Commercial expectations for the audibility of lyrics are very high, and compression, no matter how expertly set up, is simply not an intelligent enough tool to keep a lead vocal exactly where you want it throughout most mixes. If you try to keep your vox parts up-front and audible in a mix entirely with compression, they’ll usually sound over-processed, and it’s a better tactic to keep the compression within musical-sounding limits before dealing with fine, moment-to-moment level tweaks manually, by moving the vocal fader during the mix. All the main sequencers now have good fader automation systems, allowing you to edit and refine fader moves until they sound exactly right, so if you’re after the best vocal intelligibility possible, you should make a point of learning how these facilities work in your own software.

Before I move on, let’s quickly recap what we’ve covered so far.

First, concentrate on the balance. Can you hear everything you need to?

When you can’t find a static position for a fader, mult and/or compress to try to remedy that.

When compression does solve your balance problem, ask yourself a second question: do I like the subjective ‘sound’ of my compression? If not, try a few different compressors or presets.

If you still can’t find a static fader position that works, you probably need to do some other processing or automation work to reach that magical balance.

All Those Other Controls

So why do we need all the other controls? If you’ve already taken the opportunity to try out a variety of compression presets on the same sound, you’ll have noticed that some work more effectively in evening out the levels of the instrument in question than others, and that’s because the deeper compression parameters in each preset tweak a variety of more subtle aspects of the compressor’s gain-reduction action. If you can learn to adjust these parameters for yourself, you can match the compressor’s action more closely to the specific dynamic-range characteristics of the input signal — and more effectively achieve the static fader level you’re looking for.

Although the technical raison d’être of compression is gain-reduction, compressors also change the tone of processed signals quite a lot, even when compressing comparatively little. So if you like the general tone of a compressor, but you can’t find a suitable preset for the instrument you’re processing, it’s useful to be able to tweak the gain-reduction action manually to suit. And once you get some practice working with all the extra controls, it actually ends up being quicker and easier to set them up from scratch anyway.

To see how the compression ratio control can work, let’s take the example of a slap bass part, the waveform envelope of which might appear as the top one in the diagram below. The big spike is where a slap note has created a large level surge in an otherwise comparatively even line. Setting a compression threshold above the majority of the note peaks allows you to compress just the rogue slap note, but if you used a normal moderate compression ratio (as in the middle waveform envelope) you wouldn’t be able to contain it as well as you might like. Increase the ratio higher, though, and the gain-reduction will stamp down much more firmly on the offending level spike, preventing it from leaping out unduly within the mix.

So let’s introduce some of the more advanced controls and look at how each can be used to adapt the compressor to specific tasks. As a first example, let’s consider a slap-bass part. Now, as everybody knows, the best processing for slap bass is that button labelled ‘Mute’, but let’s assume for the moment that this option has been ruled out This particular slap-bass part is nice and dynamic and balances fine with the rest of the track, except that the odd slap note really takes off and leaps out of the track. You only want to turn down the sporadic signal peaks — but you want to turn them down pretty firmly in order to match the levels of the rest of the bass part.

What compressors do is reduce the amount by which a signal level exceeds the compressor’s threshold level, so in this case you want your compressor to put up a proper fight and all but stop the input signal from exceeding the threshold. That way you can set the threshold just above the level of the majority of the bass part, and it will then kick in at full force only when the over-zealous slap notes hit.

In contrast to the slap-bass example, lower ratios tend to be better for instruments which have good musical dynamics, but simply have too wide a dynamic range. Let’s assume that the waveform envelope at the top of the diagram below represents such a part, then compressing with a low ratio can be used to gently squeeze the dynamic range such that it will maintain its position in the mix balance. However, if the ratio is set too high, as in the bottom waveform envelope, the compression will iron out the part’s internal performance dynamics and render it unmusical.

By contrast, imagine an electric guitar part where there are no dramatic level spikes, but where the overall dynamic range is still militating against finding a static fader level. You want your compressor to act more gently on signals overshooting the threshold level, so that you can set the threshold just above the level of the softest notes and then subtly squeeze the whole dynamic range down to a more manageable size.

Ratio

It’s a compressor’s Ratio control (sometimes labelled Slope) that allows it to tackle these two contrasting problems, effectively setting how firmly the compressor reins in signals that overshoot the threshold level. At low ratio settings (something like 1.5:1) the overshoots are nudged politely back towards the threshold, whereas at higher settings (12:1, for instance), overshoots are beaten back by club-wielding thugs. At the highest settings — and some compressors offer infinity (or ):1 — overshoots are effectively stopped in their tracks, unable to cross the threshold at all. So for our slap bass, we’ll be looking for high ratios, and for routine dynamic-range reduction tasks (like the electric guitar example) the lower ratios (up to about 3:1) will fix balance problems in a more natural-sounding way.

When I’m talking about a ratio of 3:1, for example, you might wonder what that figure actually means. Put simply, for every 3dB by which the input signal exceeds the thresold, only 1dB will be allowed to pass by the compressor. I could give you some lovely graphs, but I don’t think it’d be a lot of practical help, because some compressors don’t label their Ratio controls and different compressors can react quite differently for the same Ratio setting. A much more practical and intuitive approach is simply to use a compressor with a gain-reduction meter so that you can see when and how much the compressor is working as you juggle the Threshold and Ratio controls.

In the case of the slap bass, you’d set the ratio up fairly high to start with, and then find a threshold setting that caused the gain reduction to kick in only on the slap peaks. Once you’d done this, you’d listen to ascertain whether you’d solved the balance problem, and then adjust the Ratio control accordingly. Still too much slap? Increase the ratio to stamp on the peaks more firmly.

With the electric guitar example, you might start off with a fairly low ratio (maybe 2:1) and then set the threshold so that gain-reduction happens for all but the quietest notes. With the threshold in roughly the right place, you could then turn back to the ratio control and tweak it one way or the other to achieve your static fader level. If some quieter notes are still too indistinct, increase the ratio to reduce the dynamic range further. Why not just max out the Ratio control? The danger is that if you turn it up too high, you’ll iron out the important performance dynamics that make the part sound musical, leaving it a bit flat and lifeless — so try to turn up the Ratio control only as much as is required to get the balancing job done.

At this point you might very well ask: what would I do if that slap-bass part needed not only high-ratio control of the slapped notes, but also more general low-ratio dynamic-range reduction? The answer is that you could deal with the problem by chaining more than one compressor in series. This is quite common in commercial practice, and lets you dedicate each specific compressor to a different task. If you’re wondering what order to put the different processors in, though, the answer isn’t quite as clear. The best solution is to try both ways and choose the one that best resolves the balance.

Why Attack & Release Matter

Up to this point, we’ve been dealing with the compression controls that are the easiest to get a handle on, but when it comes to a compressor’s Attack Time and Release Time parameters, a lot of newcomers quickly become confused. So let’s once again examine some examples of real-world balance problems to illustrate the practical purpose of these controls.

Let’s say that we’re mixing a song where a strummed acoustic guitar has a nice, natural sustain that works really well when it’s at the right level in the mix, but you find that you have to turn the fader down whenever the player digs in more during your song’s choruses. “Sounds like a job for Captain Compressor!”, you cry, but when you actually start dialling in the processing you find that, rather than just reducing the level differences between song sections, the compressor seems intent on evening out the much shorter-term level differences between the attack-transient and sustain portions of each strum. Although you can sort out your overall balance problem, you’re having to pay an unacceptable price: the impact of each strum is softened, or the instrument’s sustain is over-emphasised.

The Attack and Release controls provide a remedy to this ailment, because they determine how quickly the compressor’s gain-reduction reacts to changes in the input signal level: the former specifies how fast the compressor reacts in reducing gain, while the latter specifies how fast the gain reduction resets. The reason why the compressor in our example isn’t doing the job it’s being asked to do is that it’s reacting too fast to changes in the signal level. In other words, its attack and release times are too short. Increase these and the compressor will react more slowly, which means that it’s likely to deal with this particular balance problem more effectively, because it’ll track longer-term level variations (such as those between our verse and chorus) rather than short-term ones (such as those between the individual strum transients and the ringing of the guitar strings between them).

If you look at these controls’ legending, you’ll notice that the times are usually expressed in milliseconds, although you do occasionally find microseconds and whole seconds. However, as with the Ratio control, I wouldn’t recommend getting too hung up on exact numbers, because they’re only ever a rough guide to how a specific compressor responds in practice. A much better tactic is to focus on finding the best balance with the fewest unmusical side-effects, adjusting the Attack and Release controls by ear. A compressor’s gain-reduction meter can be a very good visual guide here, as it’ll show you not only how much compression is being applied, but also how fast it’s changing. Because the effects of gentle compression can be subtle, the visual feedback from the meter can be a great aid in setting up the controls.

Snare Compression: Three Different Settings

A compressor’s attack and release times can have very different effects on the waveform envelope of a snare-drum hit. Beneath the unprocessed hit (top), you can see the transient-suppressing effect of very short attack and release times. The third waveform shows how combining a fast attack with a slower release gives you pretty much just an overall level change, with little change in the nature of the snare sound. In the last waveform, increasing the attack time a little in has boosted the level of the initial percussive transient in relation to the sustain.

The ability to adjust Attack and Release controls independently significantly increases the range of balance problems than can usefully be tackled, so let’s look at another common example: a snare-drum backbeat. Set the attack time too fast and the compressor will respond quickly to the fleeting initial drum transient, reducing the gain swiftly. If you then set the release time very fast, the gain reduction will also reset very rapidly — well before the drum sound has finished, such that the lower-level tail of the drum hit won’t be compressed as much. The drum transient will be de-emphasised relative to the overall snare sound.

On the other hand, if you partner your fast attack with a slower release, the gain-reduction will reset very little during the drum hit itself, instead resetting itself mostly between the hits, so the balance between the transient and sustain phases of the drum will remain pretty much unchanged. The compressor in this case is simply making the level of each drum hit appear more consistent. However, if you then increase the attack time, you’ll find that some of the drum transient begins to sneak past the compressor before its gain reduction clamps down, effectively increasing the level difference between the transient and the rest of the snare sound.

So the attack and release controls have made possible three different balance results — less transient level; more consistent hit level; and more transient level — all from the same compressor. This ability to achieve very different effects is partly what confuses some newcomers to compression, and it’s also one of the reasons why promisingly-named compressor presets often don’t do the trick: if your ‘Snare’ preset has been set up to reduce the drum transient, it won’t help if you actually need more transient in your mix!

Side-effects

Although thinking in terms of balance answers most questions about attack and release times, in certain circumstances you may find that some settings produce unwanted side-effects. The first problem occurs when you set a fast enough attack and release that the compressor begins to react to individual waveform cycles, rather than the overall signal-level contour. The gain reduction then effectively changes the waveform shape, producing distortion — the nature of which will depend on the sound being processed and the compressor you’re abusing. Bass sounds, with their slow-moving waveforms, are particularly prone to this, but delicate acoustic instruments can also present difficulties because they’ll ruthlessly expose the smallest of distortion artifacts.

Another common problem is with percussive bass sounds, such as kick drums, which can appear to lose bass content if you compress them with attack times under about 50ms. This is because the compressor begins clamping down during the first couple of waveform excursions, which seems to affect lower frequencies more than higher ones, shifting the tonal balance of the sound. Once you know this is a danger, it’s not that tricky to avoid, but if you’re not listening for it, it’s easy to miss while you’re concentrating on balance issues.

One final thing to say is that changing the attack and release times will affect the amount of gain reduction that’s occurring for a given combination of threshold and ratio settings. For example, a side-stick sound (which comprises a short transient and very little sustain) might completely bypass a compressor that has a long attack, even if its level shoots way over the compressor’s threshold. So it’s not uncommon to keep adjusting Threshold and Ratio controls alongside your attack and release, to take account of these kinds of changes.

Summing Up

There’s more to using compressors than just solving mix-balance problems, but until you’re confident with their fundamental gain-reduction properties, their more advanced and creative applications will be a bit baffling. I hope this article has clarified these basic functions in such a way that you can start putting them to use sensibly straight away, while avoiding the most common processing mistakes. Once you’re confident of which controls you need to reach for in any given case, you’ll find that the more subtle differences between compressors begin to become more relevant, and that the purpose of more advanced multi-band and parallel-processing techniques becomes more logical. All of which can be exciting and interesting stuff — but that’s a very different article!

Former SOS Reviews Editor and regular contributor Mike Senior has worked professionally with artists such as The Charlatans, Nigel Kennedy, Therapy, and Wet Wet Wet. He now runs Cambridge Music Technology, delivering modular training courses based on the studio techniques of the world’s most famous producers.

www.cambridge-mt.com

Published in SOS September 2009

Sound Devices Mix Pre-D - NAB 2011

Classic Compressors

Choosing The Right Compressor For The Job

Technique : Effects / Processing

Now that we have digital models and dynamic convolution technology, we’re spoilt for choice when it comes to classic compressor flavours. But whether hardware or software, which is right for which job?

Mike Senior

Whether they truly recreate the real thing or just give you a hint of their flavour, there’s now a bewildering number of classic compressor emulations in software.

At its most basic level, compression can be seen merely as a labour-saving solution to mix-balance problems: if you control the levels of an overly dynamic instrument, it becomes easier to find a consistent fader setting for it. While the practicalities of setting up compressors can take a little time to get your head around to start with, there is still a certain logic behind the process, as I’ve tried to demonstrate in preceding pages of this issue of SOS.

As you venture beyond general-purpose balance discussions, though, the subject of compression becomes shrouded in mystique, particularly with respect to the qualities and quirks of the most famous hardware compressors, many of which are prized as much for their tonal qualities and audio side-effects as for their ability to control dynamic range. For years, such debate remained the preserve of those few lucky souls in professional studios who had the opportunity to compare a range of such rare and expensive hardware, but recent advances in the DSP emulation of those same units has brought the topic slap-bang into the mainstream.

That’s great news in some ways, but it also presents many recording musicians with a significant problem. Seasoned studio jockeys can bring a wealth of past experience to bear on the task of choosing the right model of compressor for any task, but those without the benefit of such an apprenticeship can now find themselves adrift in a sea of complex possibilities, with little but guesswork and experimentation to assist them.

In this article, then, I’ll try to shed light on which classic compressors tend to be used in which processing situations, based on my ongoing research into the techniques of ‘A-list’ engineers and producers (see the box at the end of the article for their backgrounds). I’ll also look at a few of the design features and side-effects that adapt the different processors to these tasks, as well as talking about some of the ways they’re actually combined in real-world sessions.

Classic Vocal Compressors

Let’s start where the money’s at in most productions: those all-important vocal parts. There’s no shortage of conflicting opinions from those in the know, but certain compressors do seem to crop up a great deal more than others.

Where transparent level control is required, the Dbx marque is a popular choice, with engineers such as Steve Churchyard, Humberto Gatica, Eddie Kramer, Ed Cherny, and Mike Clink all specifically name-checking the Dbx 160-series processors. The clean, predictable response of the low-distortion, Blackmer VCA gain-reduction element is clearly part of the appeal: “A Dbx 160 works really well if you just want to protect the tape,” comments Ed Cherny, for example. The majority of SOS readers working with high-headroom, 24-bit A-D conversion will be less concerned than analogue engineers about fitting their recordings into the dynamic range of the recording medium, but cleaner-sounding compressors can be just as useful while mixing, especially on instruments with complex harmonics (12-string guitars, pianos and harpsichords, for instance), which can respond badly to distortion.

However, the compressor’s speed, achieved through a feed-forward circuit design, also seems to be a key feature for many engineers, with Mike Clink referring to his ‘fast and quiet’ Dbx 160s and Steve Churchyard commenting that “typically the Dbx 160 seems a little faster than 1176s or Fairchilds, so if I need something with a faster attack time, I’ll use that instead.” In fact, so fast can this compressor react that it can easily generate distortion as it tracks individual waveform cycles, especially on later models with variable attack and release controls, so you do need to listen carefully for this. Whereas the original Dbx 160 was a hard-knee device, the later 165 retained the same basic design while adding in an ‘Over Easy’ soft-knee compression curve, and this markedly improves its capabilities with regards to unobtrusive control of extreme level fluctuations. In the words of Mark Nevers: “You can squash the shit out of it, but you don’t hear it.”

Despite their reputation for cleannness, later Dbx models do still offer subtle colour and flattery to the vocal tone, by virtue of their transformer-coupled inputs and outputs. High-spec transformers can contribute a kind of euphonic, low-level, frequency-dependent distortion that regularly elicits descriptions such as ‘warm’ and ‘fat’, and which is often called upon to make vocals seem somehow ‘larger than life’. The combination of VCA gain-control and audio transformers can also be found in Focusrite’s Red 3 and Red 7 units, which also exhibit this kind of subtle warmth and are used for vocals by producers such as Steve Power and Stephen Hague.

The ability to heap gain-reduction onto a vocal without audible side-effects is associated with another well-known unit, the Teletronix LA2A (now re-issued by Universal Audio), which achieves its low-distortion soft-knee characteristics, as well as a programme-dependent release time, using a unique electro-optical gain-reduction element. This unit has been in regular use since the ’50s, and could lay a strong claim to being the most popular vocal compressor of all. Where the Dbx’s feed-forward VCA and solid-state circuitry are fast, controlled, and clean, the LA2A’s electro-optical feedback design works more slowly and in a non-linear fashion, which nonetheless produces very musical results, preserving the impression of performance dynamics despite quite extreme level management. While it, too features audio transformers, it also rounds out the tone via its internal valve amplification stages. “I added the LA2A for warmth,” comments Serge Tsai, for example, echoing similar ‘gentler and warmer’ epithets from producer Glen Ballard.

The hugely popular Teletronix LA2A (now built by Universal Audio, and one of the first units modelled for their UAD platform) is great at transparent gain reduction, but can also yield the kind of vocal overdrive favoured by artists like Jack White of the White Stripes.

However, a few high-profile engineers have also learned to take advantage of the extra attitude on offer from this king of smoothness as you push it outside its comfort zone. “Sometimes the LA2A works really well [on vocals],” says Joe Barresi. “If you’re looking for more dirt you can turn a tube compressor up a little more.” Joe Chiccarelli recalls a similar tactic when producing the White Stripes: “Jack [White] always wanted more distortion on the vocals... When it came to the mix, Jack wanted still more edge, so I overloaded an LA2A compressor... This meant that I was getting the distortion from the last tube stage of the compressor, which creates a really beautiful distortion.”

The LA2A’s winning combination of electro-optical compression, transformers and valve amplification can also be found in a couple of other units well-respected for vocal use, Tube Tech’s CL1B and Avalon’s VT737SP, so if you like what an LA2A can do for your vocals you may find that these models provide useful alternative colours. For faster, cleaner, more VCA-style control, and a slightly crisper transformer-free sound, Summit’s TLA100A and DCL200 are also popular with audiophile engineers such as Al Schmitt, Jon Gass and Tony Maserati. “I found I could use it on vocals that I needed to squish a fair amount but didn’t want to lose all the nice top I got off the microphone,” explains Joe Chicarelli in an interview on www.summitaudio.com.

If you like the character of the LA2A’s non-linear optical gain element, but the valve distortion artifacts aren’t appropriate, then the slightly faster, valveless version of the same basic design, the Teletronix LA3A, provides another option favoured by two of the most respected mix engineers on the planet: Bob Clearmountain and Tom Lord-Alge. “To make a vocal command attention,” says Lord-Alge, “I’ll put it through an LA3A and maybe pummel it with 20dB of compression, so the meter is pinned down. If the beginnings of the words then have too much attack, I’ll put the vocals through an SSL compressor with a really fast attack, to take off or smooth out the extra attack that the LA3A adds.”

Further Vocal Coloration & Enhancement

Beloved of Beatles engineer Geoff Emerick is the rare and mind-bogglingly expensive 

Many engineers use the Fairchild 670 for the subtle sonic enhancement of its circuitry as much as they do for gain-reduction. Although the original is now prohibitively expensive for all but the most maniacal of collectors, there are some excellent software emulations, such as this one from Universal Audio.

Fairchild 670 stereo valve limiter (and its pretty much identical 660 mono sibling), and his penchant is shared by other high-profile names such as Roy Thomas Baker, Steve Power and Tom Elmhirst. Where the LA2A will give you fairly slow attack times, the Fairchild has a much faster, sub-millisecond, onset, as well as a different type of extremely soft-knee compression curve, courtesy of its valve-based ‘vari-mu’ gain-control circuit — the compression curve is unusual in that the ratio just continues increasing as the input signal level gets higher.

While these characteristics mean that the 670 can control levels very effectively, probably the main reason that top engineers call on it is actually for the sonic enhancement of its circuitry, which incorporates a total of 20 valves and 14 transformers! Geoff Emerick always put the Beatles’ vocals through it for precisely this reason: “Just the sound of the amplifier, even if you didn’t do any limiting, just added a certain presence.” Steve Churchyard also talks about using the 670 “more for the sound than for the compression,” a sentiment echoed by Tom Elmhirst in SOS August 2007.

The Fairchild 670’s extreme rarity, combined with its ability to make notes, in the words of Jack Joseph Puig, “long and lush and beautiful” has spawned not only fastidious modern hardware replicas (such as Anthony DeMaria’s ADL670 and the Analogue Tube AT101) and heavily-inspired variations (EAR’s 660, Fairman’s TMC, Manley’s Vari-mu, and Pendulum Audio’s ES8, for example), but also a slew of software emulations — so you don’t need to look too far these days to find something akin to the Fairchild sound. If you like what you hear, you might also want to hunt down emulations of some other vintage valve-based vari-mu designs, such as the Altec 436, Collins 26U1 or 356E1, Federal AM864U, Gates Sta-Level or Level Devil, RCA BA6A, and Universal Audio 175 or 176. Fairchildren they aren’t, but each is nonetheless packed full of its own unique vintage valve character, and is frequently referred to reverentially by mix engineers. Michael Brauer, for example, used a Gates Sta-Level and a Federal AM864U alongside the EAR 660 for his mix of Coldplay’s recent hit ‘Violet Hill’.

The Urei 1176 is famous for adding a sense of edge and excitement to vocals.

The final classic vocal-processing contender is Urei’s 1176 (and the same marque’s near-identical stereo 1178). Whereas the 670’s valves are smooth and warm, the 1176’s equally fast FET-based design brightens the tone and adds extra edge and excitement in a way that has made it a fixture on vocals for many top engineers for a half-century. People differ in terms of how exactly they use it, particularly with regards to the choice of ratio setting, so it pays to experiment with this aspect of any emulation — and don’t forget to try the unusual ‘all buttons’ mode if it’s on offer, as it is, for example, in Universal Audio’s software 1176LN for the UAD platform. The 1176’s enormous popularity has given rise to legions of impersonators, including hardware units such as Universal Audio’s re-issued 1176 and Purple Audio’s MC76, as well as myriad software emulations — so this should be one of the easiest compressor sounds to get hold of, even if you’re working on a budget.

That pretty much wraps up the vintage units, but we can’t leave the topic of vocal compression without touching on one very popular modern processor too: the Empirical Labs EL8 Distressor. This phenomenally versatile, digitally-controlled, analogue compressor, introduced in 1995, was designed partly to emulate the characteristics of different classic compressor actions, including those of all four principal units we’ve discussed so far: Dbx 160, Teletronix LA2A, Fairchild 670 and Urei 1176. As such, it’s not easy to say exactly what kind of compression you’ll get if you dial up a Distressor emulation, although a little knowledge of the way the aforementioned classic compressors work may provide you with some useful pointers. What is worth a special mention, though, is that the Distressor has a number of built-in, side-chain EQ options, one of which emphasises the upper-mid frequencies in the detector circuit and thereby reduces sibilance problems that can often accompany heavy compression.

Smash & Grab: Compressors For Drums

It’s pretty clear, looking at the practice of real engineers, that fast-acting compressors and limiters are the most popular ones for drum processing. No prizes, then, for guessing that, of the models we’ve already encountered, the LA2A doesn’t really get a look-in. Units from Dbx receive mentions in this role from the likes of Tom Lord-Alge, Jack Joseph Puig, and Butch Vig, while Glen Ballard singles out the 160X variant for special praise: “I think the 160X is the best compressor for drums that I’ve found. You just give it a little bit on the kick and snare while you’re recording and it brings them into focus. I don’t think you can find anything better for kick and snare — they’re amazing! You can really slam them too! I love the fact that they have a really quick recovery, and you can exaggerate it if you need to.” The extremely fast attack and release times available on many Dbx units are particularly worthy of investigation if you find yourself in a situation where you really need to sculpt the front end of your drums with precision.

The Fairchild units have been responsible for much drum-mangling ever since the 670’s prominent use by Emerick on Ringo Starr’s kit, and since then engineers including Steve Churchyard, Tom Elmhirst, Manny Marroquin and Elliott Scheiner have all used the Fairchild brand for drum processing, on account of its colourful sonics and speed. “The 660 has a nice smooth sound,” comments Fabian Marasciullo, “and it really controls the peaks.” However, the 670 also has an unusual trick up its sleeve in that it has M/S matrixing built in, allowing you independently to compress the Middle and Sides components of a stereo signal. By plugging the 670 into a drum bus and then compressing the Sides signal more heavily, you can emphasise and/or pump the room sound without compromising the transient snap of your main kick and snare sounds.

It’s no surprise that Urei’s 1176 and 1178 processors also crop up frequently, with mixing heavyweights such as Chris Lord-Alge, Michael Brauer and Rich Costey all partial. Despite the reputation these units have for fast attack, though, Lord-Alge was actually using an 1178 more for its sonic flavour with slower attack settings when we interviewed him back in SOS May 2007. Rich Costey’s preference for a parallel processing setup with this compressor also suggests that the tonality of the Urei, especially when driven hard, was of more importance than its ability to catch transients. (See the ‘Parallel Compression Explained’ box for more information on parallel techniques.)

It’s to be expected that Empirical Labs’ Distressor also turns up frequently when drum processing is discussed, but that’s not just because of its emulative power: it has also earned a reputation of its own, particularly for its ‘Nuke’ brickwall-limiting mode, which has a logarithmic release time originally optimised for pulverising room-mic recordings.

Empirical Labs’ Distressor is one of a few ‘modern classic’ hardware compressor designs, and is incredibly versatile. As well as mimicking some longer-established designs, it brings some of its own special tricks to the table.

The side-chain high-pass filtering is handy, too, where you want to compress a drums bus heavily without the kick drum triggering excessive gain pumping, and the ability to switch different harmonic distortion flavours in and out of circuit provides a great deal of tonal flexibility. Perhaps because of the compressor’s enormous power to mould the envelope and timbre, Michael Brauer, Joe Chiccarelli, Fabian Marasciullo and Robert Orton all use parallel processing setups to give extra control over the extent of its contribution to the mix. Take Marasciullo, for instance: “On drums [the Distressor] gives me an attack that no other compressor can give me. I will have one drum just raw, exactly as it was given to me, with just a bit of EQ on it. I will then mult it into a Distressor and that will give it this extreme attack, and then I blend it in the way I like it.”

Moving on from the vocal workhorses, another compressor renowned for drums is the TG12413, a limiter originally built into each channel of the Neve-designed custom consoles that EMI studios used for famous records such as Abbey Road and Dark Side Of The Moon. Although original units are as rare as hens’ teeth, the widespread availability of Chandler’s TG1 re-issue (and its more flexible TG12413 Zener sibling) and EMI’s more recent plug-in emulation has ensured this sound a regular role in mainstream commercial mixing.

“I like the EMI compressor for its fast attack,” remarks Joe Barresi, and this is a recurrent theme in other producers’ descriptions, but the tone of the TG12413’s unique diode-based gain-reduction circuit is also an important factor. Although Joe Chiccarelli characterises it as “more of a round, warm, vintage sound,” pushing harder transforms it into something more aggressive. Nick Launay: “I don’t like a lot of compressors because I find they’re not fast enough or they’re a bit too subtle or they make things sound too nice. I like the stuff that’s got a bit of grit, like the old EMI compressors, if you can find them.” Michael Brauer concurs, speaking on the Chandler web site: “When used across the drum room [the TG12413] brings out an aggressive quality that I’ve not yet heard from any other compressor. When used as a send, it brings out the punch and clarity of instruments without affecting the overall level of the stereo output.”

The RCA BA6A. Like the Fairchild 670, this compressor is often used as much for it’s sound as it’s ability to achieve the desired gain reduction.

If you read Will Haas’s article about mix-bus compression back in SOS May 2008, you’ll already be aware of many of the industry-standard mix-bus compression choices, but it tuns out that two of these regularly appear in the role of drum submix processor too: the API 2500 and the SSL G-series console master compressor. Both of these feature fairly clean, VCA-based gain reduction and are good at increasing average levels without trampling on the punch of transients or robbing too much low end. However, the SSL also has a reputation for adding a dose of desirable mid-range crunch when more heavily provoked, and Smart Research’s C2 emulation emphasises this aspect of the unit further with its additional ‘Crush’ mode.

“[The C2’s Crush mode] adds field-effect transistor distortion,” explains Jason Goldstein. “I’m a big fan of using distortion, in small amounts. Tubes distort more harmonically, whereas transistors and Class-A stuff are more aggressive, which is why I think many guys still like to mix on the SSL 4000 — those consoles are always just shy of distorting. It adds to the overall aggression of the mix.” But that’s not all that Crush does. According to Tom Elmhirst, it “completely annihilates all the transients and brings up the air.” Small wonder, then, that Goldstein, Elmhirst and Rich Costey all take a parallel processing approach with the C2, for greater control.

Max Bass

Bob Clearmountain.

That the Urei 1176 convincingly topped the polls for bass processing in my survey of producers might raise an eyebrow, given what I’ve said about it thus far. After all, fast compression can introduce serious distortion on slow-moving bass waveforms. However, as long as the time constants are kept away from the fastest settings (in other words, somewhat counter-intuitively, away from the higher-numbered, clockwise end of the control settings on the unit), the circuitry’s tonal brightening and subtle FET distortion really help the bass to cut through the mix in a way that other designs can’t match. The lowest ratio (4:1) is the most common choice, as evidenced by independent recommendations from Steve Churchyard, Chris Lord-Alge and Ralph Sutton, although Tom Elmhirst likes the result you get when you push in the 4:1 and 8:1 buttons together.

Not far behind the 1176 in popularity is the Teletronix LA2A, numbering people like Mike Clink, Mark Endert, Tony Maserati and Jim Scott amongst its friends: the slower-moving, electro-optical design is clearly better suited to clean level management of low-frequency sounds. The valve sound, though, is clearly a crucial part of the equation too, lending thickness and warmth to the tone, as evidenced by the wide range of other valve units from Altec, Fairchild, Gates and Summit that turned up in the interviews from time to time. That said, mix engineer Chuck Ainlay raves about the original Joemeek SC2 solid-state optical compressor for bass, so that might be worth a look if FET or valve additives don’t suit your taste.

What About Other Instruments?

Joe Barresi.

I’ve spent a good deal of time looking at compressor choices for vocals, drums, and bass, and in each of these cases I’ve been able to distil some degree of consensus from the available evidence. What I’ve discovered in the process of my research, though, is that compressor choices for other common instruments, such as guitars and keyboards, are much less commonly discussed in interview and, furthermore, that there also seems to be very little agreement between different producers as to what processors to use on them.

For example, with electric guitars, aforementioned units from Dbx, Fairchild, Gates and Urei all turn up sporadically, as well as Neve’s 2254 and 33609

Joe Chicarelli

models (both of which use similar bridged-diode gain-reduction elements and lots of transformers, to deliver a silky-smooth tone that often seems to be called on to remedy thinness or abrasiveness in a recording). However, little justification is given for these processing decisions, so any attempt at generalisation remains firmly in the realms of surmise.

Perhaps the lack of information on processing other instruments simply reflects that vocals, drums and bass are more important than other elements in modern productions — so by the time an analogue engineer gets to mixing the brass parts, for instance, the most desirable compressors will probably already be in use, and he or she will, in practice, be more concerned with the compressor settings than the specific choice of unit. Or maybe it’s that vocals, drums and bass just tend to need the most dynamic-range control in a typical mix, so the choice of compressor (and indeed the use of compression at all) becomes less crucial with instruments holding a more consistent level.

Well known for bus compression, amongst other things, the Neve 33609, which offers silky-smooth gain reduction, is equally at home on electric guitar.

Another argument for the lack of consensus with regards to compressor choice for things like electric guitars and synths could simply be that parts like these vary so much in character from mix to mix. A clean and choppy funk guitar is clearly a very different beast to a heads-down Emo wall of noise in terms of attributes such as transient definition, dynamic range and frequency balance, so it makes sense that the same compressor would not necessarily suit both.

Whatever the reasons, though, hopefully this article has provided enough of an insight into the basic characteristics of different classic compressor models that you can begin to extrapolate towards more educated guesses when it comes to processing other instruments. The aforementioned funk part might benefit from the speed of a drum compressor to catch its transients, for instance, while the Emo landslide might need some of the 1176’s tonal definition to make it bite. Once you get used to the way in which compression can react on vocals, drums, and bass, you’ll find that there’s a lot less guesswork involved in processing pianos, guitars, strings, brass, zither, nose flute, one-handed clapping...

Published in SOS September 2009

SSL Duende Native - Musikmesse 2011

Audio Technica AT2022 - NAB 2011

Studio SOS: Dominik Johnson

EQ, Compression & Reverb

Technique : Effects / Processing

This month, we tame a strange space, record a sitar and dispense a little wisdom on EQ, compression and reverb, as we sort out Dominik Johnson’s home studio.

Paul White & Hugh Robjohns

Dominik Johnson plays his sitar, as Paul White searches for a good mic position.

Dominik Johnson studies world music at Leeds University, and plays all manner of plucked stringed instruments to a very high standard. His compact Manchester apartment is crammed with classical guitars, banjos, bouzoukis, mandolins, a renaissance lute, a tenor mandola, and even a sitar. However, his attempts to record these instruments at home had been compromised by the rather odd acoustics of his living space, and by the layout of his budget recording setup, which comprises Logic Pro running on a small Macbook laptop computer, an Edirol UA25EX interface and a pair of tiny Alesis M1 active USB desktop speakers. He’d made a good choice of microphones (a pair of Rode NT55s, to allow for stereo recording), but at the time of our visit the lack of acoustic treatment was working against him.

Starting Setup

Dominik’s flat is a nice, large space with high ceilings, but it was completely untreated when the team arrived.

We arrived to find that Dominik had enlisted the help of his friend Joel Davis, who is at the University of Huddersfield studying Music Technology. This turned out to be useful, as Joel is very tall and Dominik didn’t have anything rigid to climb on when we were fixing things up!

The studio apartment is a rectangular room, around four metres wide and 10 metres long, with an enclosed ‘island’ in the centre serving as kitchen, bathroom and storage. This stops around 80cm short of the main room’s unusually high ceiling (3.2 metres), leaving some storage space on top, which is accessible by means of a ladder but also allows sound to pass over the top. Essentially, this island divides the apartment into two roughly square spaces, separated by two short corridors on either side (flanking the island) — which unfortunately means that any sound created in one space sets off resonances in the other.

Dominik had his system set up on a small computer desk backed right up against a side wall. Playing some known test material revealed that he had a huge room-mode resonance set off by the bottom ‘A’ of his guitar.

Speakers Corner

An unusual open-plan layout with an ‘island’ in the middle caused problem resonances.

Because this was a rented apartment, we couldn’t do anything non-reversible, and there was no budget for bulky bass traps (although there was space on top of the island that could be used), so our first step was to try moving the desk, to find a place where the bass end would be more even to start with. Dominik’s little Alesis monitors have rear ports, and his initial setup placed them very close to the wall. That being far from ideal, we experimented by moving the desk away from the wall, and then over to the corner of the room nearest the glass windows and doors leading to the balcony. By arranging the desk at an angle, we got the speakers further from the walls, and helped reduce side-wall reflections reaching the listening position. We also placed Auralex MoPad foam pads under the speakers (Dominik had already bought some but wasn’t actually using them!). Because the speakers were so small, only one half-width pad was used under each speaker, set up ‘reversed’ — so that the speaker was angled up slightly, towards the listening position, and the foam that extended in front of it would help a little to reduce reflections from the desk surface. Using both foam pads would increase the area of foam and reduce reflections even more, but the desk was quite small and the best working compromise, leaving enough space for the laptop, was to use a single pad.

Packing duvets into the storage area above the central ‘island’ room tackled the unwanted resonances, as well as providing some useful bass trapping.

Further listening with test recordings and a ‘staircase’ of sine-wave semitones (from Logic’s EXS24) showed a noticeable improvement, although the room mode was still there. Examining the speakers showed that Dominik had engaged the bass-boost switch: flipping this back to the normal setting cleaned up the low end further, as did pulling the speakers forward on the desk so that they were further from the walls, and closer to the listening position.

As an experiment, we improvised some foam plugs to put in the bass ports, which further tightened the low end, both with and without the bass-boost switch engaged. My preference was to leave the bass-boost off and insert the foam plugs halfway, which felt like a good compromise between a clean bass and having enough low end: tiny monitors like these are never going to be great at deep bass frequencies, but as Dominik was recording non-bass instruments, the arrangement was workable. Even so, we did recommend he buy some good-quality headphones to use as a secondary reference.

All Of A Flutter

With plastered concrete walls, the apartment exhibited very obvious and distinct reverb, and produced some of the most dramatic flutter echoes we’d heard to date, so our priority was to dry this up as much as possible. We’d brought half a dozen sheets of two-inch Auralex foam with us, but this wasn’t going to stop sound leaking around and over the centre island, and exciting the bedroom space at the other end.

After considering the problem over coffee and chocolate biscuits, we set out for the nearest DIY store to buy self-adhesive plastic hooks, nylon cord, impact adhesive and two king-size polyester duvets. The plan was to glue wooden strips to the top rear edge of the foam panels using contact adhesive, and then hang these, picture-style, on a self-adhesive plastic hook stuck directly to the wall. We’d use the duvets to screen off the space over the centre island by hanging them over a nylon line fixed to plastic hooks and going all the way between the side-walls, via a convenient centre support where we could put more hooks.

In most respects this worked well, although the hot weather made all the plastic hooks gradually peel off the walls under the very modest weight of the foam panels, so we had to provide a helping hand using some of our contact adhesive. We ended up with one foam panel each side of the listening position (one on the wall and the other fixed to the metal door/window frame), three on the wall behind the sofa, which was roughly behind the mixing position, and one more set horizontally, as high as we could place it to damp down some of the flutter echo that developed high up in the ceiling space.

With the panels and duvets in place, the liveness of the space was significantly reduced but was still enough to flatter acoustic instruments, especially if recorded with the floor rugs lifted to reveal the nice parquet floor. Test recordings sounded cleaner, with better imaging, and although the ‘bottom A’ bass hump was still present to some degree, it was far more benign now than it had been at the outset.

Recording System

When we arrived, Dominik had his small monitor speakers placed directly on the surface of the desk...

Having addressed the acoustics issues as best we could, we turned our attention to the recording system. Dominik had been plagued by high latency, and couldn’t get the direct monitoring on his Edirol interface to work. We were initially concerned that as the monitors and interface were both connected via USB, the setup might require an aggregate device driver to allow both to work at once (this approach can cause increased latency), so we wired the speakers into the analogue outs of the interface, using a couple of jack cables, so that the interface could use its own drivers.

A quick look at Logic’s Audio Preferences showed that Dominik had it set to a 1024-sample buffer size, and we reduced this to 128 to cut down the latency. Most players can live with this tiny delay even if they don’t have zero-latency hardware source monitoring, although we were determined to find out what had been going awry with the Edirol. As it happened, nothing was wrong, but Dominik hadn’t realised that you need to turn down the track monitor level in Logic as you record. If you don’t do this, you hear both the direct monitor signal and a slightly delayed version coming back from the computer — and his initial 1024-sample buffer size would have created a very distinct slapback echo.

We went back to the Audio Preferences, ticked the Independent Monitor Levels box, and showed Dominik that he could now turn down the fader level while in record mode, and it would automatically spring back to its playback level when he left record. Keeping the channel fader down while recording prevents the current performance from being monitored in the DAW output while tracking, so that only the direct source and existing DAW tracks are heard. We made some test recordings in this way after selecting source monitoring on the interface, and confirmed that all was now well, with no unwanted echo.

Mic Technique

...so we isolated the speakers from the desk surface, before putting foam in the ports to get the most appropriate sound for Dominik’s music and room.

For recording vocals, we took along a Reflexion Filter kindly provided by Sonic Distribution. This would also be useful for placing behind an omnidirectional mic when recording Dominik’s instruments. As usual, we rejigged the mounting hardware to make it sit closer to the centre of gravity of a typical mic stand. We also stressed that for the best recorded vocal result, some absorbing material should be placed behind the singer to intercept reflections that would otherwise bounce back over their shoulders, into the microphone, from hard surfaces behind them. Here, Dominik could use one of the foam panels or an additional duvet. He asked what was the best place in the room to stand for vocal recording, and we explained that as long as he avoided the exact centre of the room, or positions too close to the walls, the results should be OK more or less anywhere.

Dominik was also keen to see how we’d mic the various instruments that he plays, so we did some test recordings to see what worked best, using the Rode NT55 with its omni-pattern capsule fitted. Acoustic instruments always seem to sound more natural when recorded with an omni mic and, as explained above, if the room is intruding too much you can screen off the sides and rear of the mic using a Reflexion Filter or improvised screens. Because omni mics have a much better-balanced off-axis response than cardioids, positioning also seems less critical.

Above right: To the rear of the mix position, we decided to hang three Auralex acoustic foam panels to cut down on unwanted reflections. Right: we also placed foam panels at the mirror points to create a tighter stereo image from the monitoring setup.

We explained a simple ‘universal’ miking rule that often yields good results when you’re recording an instrument for the first time: you measure the longest part of the instrument that creates the bulk of the sound (such as a banjo body diameter), and then place the mic no closer than this distance, so that it has a chance to pick up sound from all parts of the instrument rather than from just one spot. Then you need to find the placement that gives the most natural sound, which is best done by wearing headphones as you move the microphone (or move the instrument relative to the mic). Most recording textbooks give you typical mic placements, but every situation (instrument, player and room) is different, and it always pays to search for that sweet spot.

When recording the sitar, I tried a mic position very close to the floor, to make use of the boundary effect and boost the low end slightly. The Sitar is an unusual instrument in which some sound is also produced by the (hollow) neck and, where attached, by a secondary resonator gourd at the back of the neck. We found that placing the mic close to the floor and aimed at the bridge from around 18 inches away produced a well-balanced sound with the required degree of warmth. The banjo was altogether simpler, as it seemed to produce a consistent tonal balance over quite a wide angle, so the distance between the mic and instrument was actually more important than the mic position. We also experimented by having the mic look over the player’s shoulder, working on the principle that if it sounds OK to the performer, it will also sound good to the mic. This can yield excellent results, but as we found here, the mic picks up more of the player’s breathing (or humming, in Dominik’s case).

EQ, Compression & Reverb

As Dominik was a relative newcomer to Logic, Paul created a ‘starting’ song template that included screensets for the Arrange and Mixer pages, as well as a reverb set up on a post-fade send. This lets you get up and running without having to set up a project from scratch every time. He also gave a crash course on compression, the most important lesson being that if you choose a preset compressor plug-in setting, you still need to adjust the threshold controls so that the gain-reduction meter is showing activity on the loudest peaks: if the gain-reduction meter isn’t showing anything, then you’re getting no compression. For gentle instrument compression, a peak gain-reduction of 4-6dB is probably fine, although more aggressive sound sources may need larger amounts of gain reduction. Ultimately, of course, your ear has to be the judge.

This led to the question of the best order in which to place insert plug-ins. On one very transient-rich, percussive part that required very assertive level control, Dominik had placed a reverb plug-in before the compressor, creating a splashy, unpleasant sound that got in the way of the instrument. Normally, you’d use reverbs via post-fade sends, but on small projects there’s no problem using them as inserts. In this case, we reversed the order, so that the compressor controlled the peaks before the sound hit the reverb, and the result was less ‘splashy’.

EQ, on the other hand, can go before or after compression, depending on the effect you’re after: for natural-sounding tonal tweaking, I tend to favour filters first (to remove unwanted out-of-band noises such as rumbles), then the compressor, then creative, tonal-shaping EQ if it’s required. Placed the other way around, the compressor will react more strongly to any frequencies boosted by the EQ, and tend not to control frequencies you’ve cut. We also demonstrated the way in which quite severe narrow EQ cuts can deal with resonances, but narrow EQ boosts rarely sound natural. For natural results, use as little EQ as you can, and boost with fairly wide (low) ‘Q’ settings.

Our final venture was to explore some Logic Space Designer reverbs to see what was best for acoustic instruments. While the temptation is always to go for a concert hall — or something equally familiar sounding — you can get great results from less obvious spaces, such as tiled rooms, parking garages, or even woodland and other outdoor spaces. Impulse responses with complex early reflections and low-level tails often sound interesting, but don’t over-use them: a 20 percent mix level is usually a good starting point. About 60-90ms of pre-delay can add to the sense of space, while rolling off some low end below 200Hz can clean up an otherwise messy reverb.

As rush hour was approaching, we decided to make our getaway before any more of those B&Q hooks peeled off the walls — but we left Dominik with the contact adhesive just in case! 

Published in SOS February 2010

Alesis 3632 - Musikmesse 2011