Effects: All You Need To Know... And A Little Bit More

Sound Workshop

Technique : Effects / Processing

Whether you're a beginner or a seasoned pro, there's always something to learn about adding colour to your mix.

Paul White

The world of audio effects is one that can be confusing even for experienced engineers. Especially in modern computer-based recording systems, there's a bewildering array of options, and to add to the confusion, some effects are widely referred to by more than one name. In this article, I'll take you through the most common effects, explaining how they work and where you might want to use them in your music. Meanwhile, SOS's team of writers has contributed a wealth of expert tips and tricks, which you can find in the boxes scattered through this article.

Effect Or Processor?

Let's begin by getting clear on what we mean by 'effect': an effect is a device that treats the audio in some way, then adds it back to a dry or untreated version of the sound. Echo and reverb are obvious cases, and you can use pitch-shift and pitch modulation in a similar way. 'Processors', by contrast, generally are those devices that change the entire signal and don't add in any of the dry signal. Things like compressors and equalisers fall into this category: as you'll see from the tips and tricks, processors can often be used as effects in their own right, or as part of an effect chain, but until you know exactly what you're doing and what the consequences are likely to be, it is a good idea to stick to these guideline definitions, as they dictate how you can connect the effects into your system.

If a device has a Mix control on it that goes from 100 percent wet (effect only) to 100 percent dry (clean only), then you can be pretty sure it is an effect. If it doesn't have a Mix control, and doesn't rely on electronic delays to create its results, then it is probably safe to assume it is a processor.

Effects can be connected via insert points, or the effect send and return loop that is included in most consoles and DAWs (Digital Audio Workstations). When effects are used in the send/return loop, their Mix control should be set to 100 percent wet, so you add back only effected sound to the dry sound, which comes directly through the mixer channel.

Processors, on the other hand, comprise an entirely different water heating appliance filled with piscean vertebrates, as they tend not to need any of the dry sound, other than in a few specialist applications. As a rule, processors such as EQ and compression are connected only via track, bus or master insert points — at least until you have the necessary experience to understand why you might want to break the rules once in a while. Having got that off my chest, let's look at some specific effects (we'll look more closely at processors another time).

Tips & Tricks: Filters Used as a send effect, a resonant filter can be used much more creatively than as an insert. You get very different results from your filtering depending on where you put the filter in the signal chain. To introduce some real movement into delay lines, for example, place sweeping low-pass filters before the delay. You then get the movement of the dry sound contrasting with the movement in the delay line. If both the filter sweep and the delay lines are tempo-sync'd, you can create interesting effects where the filter appears to be moving up and down at the same time. Filters are also great for use on drum loops. One trick I like is to send the drums to a modulated resonant filter set up as a send effect, with a narrow band-pass EQ beforehand. This creates a rather bizarre metallic melody that accompanies your drums. It can get fatiguing if over-used, but brought in at a low level in some sections of a song, it can create plenty of interest, particularly if followed by a modulated delay. Matt Houghton

Echo & Delay

Echo and delay are created by copying the original signal in some way, then replaying it a short time later. 

There's no exact natural counterpart, though the strong reflections sometimes heard in valleys or tunnels appear as reasonably distinct echoes. Early echo units were based on tape loops, before analogue charge-coupled devices eliminated the need for moving parts. Today, most delay units are digital, but they often include controls to help them emulate the characteristics of the early tape units, including distortion and low-pass filtering in the delay path and pitch modulation to emulate the wow and flutter of a well-used tape transport.

While pure digital delay produces perfect echoes, an analogue emulation can be more musically useful, as each successive echo becomes less distinct, creating a sense of distance and perspective. Hi-fi echoes tend to confuse the original sound, while the human hearing system seems better able to separate lo-fi echoes from the original clean sound.

The feedback control regulates the number of echoes by feeding some of the output back to the input. If you apply too much feedback the delay unit will self-oscillate — an effect often used in dub music. Delay normally relates to a setting with no feedback whereas echo uses feedback to produce a series of diminishing repeats.

You don't have to use long, distinct delays: short delays up to 120ms can be used to create vocal doubling effects, normally set with little or no feedback. Nor do you have to dedicate a delay to a single sound: you can configure it via an aux send so that several tracks can be treated with different amounts of the same delay or echo treatment, which not only saves on processing power (or buying separate units!), but can help to make elements of your mix work better together.

You can often use a tap-tempo or tempo sync facility to get your echoes exactly in time with the song if that's the effect you need, but many echo/delay plug-ins can be locked to your sequencer's master tempo, enabling you to create precise, rhythmic delay effects.

Tips & Tricks: Panning One special effect I used quite a lot in analogue studios, but which is surprisingly tricky to implement in a lot of software sequencers, is where you feed the left and right outputs of an auto-pan effect to two different effects processors. With this setup, the outputs of the two effects can then be mixed together to create a variety of different modulation-style treatments. This patch always worked well in a send-return loop with a pair of phasers, especially if you also EQ'd the two returns wildly differently. The same setup used as an insert could do great things with distortion and ring-modulation processors, and if you were feeling really adventurous, you could fiddle with the panning rate in real time while mixing down. Mike Senior

Reverb

I think it's fair to say that we all have a pretty good idea of what reverb is, though there are several ways of emulating it in the studio. Early reverb chambers, plates and springs have now given way to digital solutions, which fall into two main camps: synthetic and convolution. Synthetic reverbs take an algorithmic approach, setting up multiple delays, filters and feedback paths to create a dense reverberation effect similar to what you might hear in a large room. Though these often sound a bit 'larger than life', they've been used on so many hit records that we now tend to accept their sound as being the 'correct' one for pop music production. Most can approximate the sound of rooms, halls, plates and chambers, but in comparison with a real reverberant environment, the early reflections often seem to be too pronounced. The advantage of a synthetic reverb is that the designer can give the user plenty of controls for altering the apparent room size, brightness, decay time and so on.

In recent years, convolution reverbs have become both affordable and commonplace. These differ from synthetic reverbs insomuch as they work from impulse responses (or IRs), recorded in real spaces to faithfully recreate the ambience at the microphone's position when the IR was made. Sometimes these are referred to as sampling reverbs but there's no sampling involved as such, even though the process seems akin to sampling the sonic signature of a room, hall or other space.

Because IRs can be recorded in virtually any space, convolution reverbs generally come with a library of IRs ranging from small live rooms to famous venues, top studio rooms, forests, canyons, railway stations and just about anything else you can think of. They sound very convincing, and there's plenty of variety to be had, but once the IR is loaded, there's only a limited amount of editing you can do without spoiling the natural sound. Usually you can apply EQ and also change the envelope of the reverb decay to make it shorter, and adding pre-delay is not a problem, but after that you pretty much have to take what you get. Some companies, such as Waves, have managed to create additional controls but, as a rule, the further you move from the original IR, the less natural the end result.

Ironically, the sound of certain synthetic reverbs is now such an established part of music history that most convolution reverbs come with some IRs taken from existing hardware reverb units or from old mechanical reverb plates. Also, if you have a convolution reverb, it is worth checking the manufacturer's site, as additional IRs are frequently available for download.

All serious reverb units have a stereo output to emulate the way sound behaves in a real space and, in the case of convolution models, the IRs are often recorded in stereo, using two microphones. Some surround reverbs are also available.

Reverb creates a sense of space, but it also increases the perception of distance. If you need something to appear at the front of a mix, a short, bright reverb may be more appropriate than a long, warm reverb, which will have the effect of pushing the sound into the background. If you need to make the reverb sound 'bigger', a pre-delay (a gap between the dry and wet signals) of up to 120ms can help to do this without pushing the sound too far back, or obscuring it.

Though reverb increases the sense of stereo width, it dilutes the sense of stereo position. If you want to pinpoint the placement of something in a mix, you should consider using a mono rather than a stereo reverb, and panning this to the same place as the dry sound.

Most synthetic reverbs allow you to balance the level of the early reflections and the later, more dense reverb tail. If you want to keep the sense of space but without the reverb tail taking up too much space in your mix, you can increase the early reflection level and reduce the tail level.

As a rule, you don't add much, if any, reverb to low-frequency sounds, such as bass guitar or kick drums. Where you need to add reverb to these sources, short ambient space emulations usually work better than big washy reverbs, which tend to make things sound muddy. Taking this a step further, you can also make a mix sound less congested by EQ'ing some low end out of your reverbs.

Tips & Tricks: Amp Modelling One of the most useful features of guitar-amp simulation plug-ins is that they can help mask some quite serious problems with whatever you're putting through them, without necessarily changing it beyond all recognition. I've found that even relatively clean settings can disguise such horrors as clipping on transients to a surprising extent. If you're ever faced with a badly recorded guitar part (even one that's played on an acoustic guitar, or through an amp), try putting it through an amp modeller. Processing audio before it passes through an amp simulator is a creative alternative to adding effects to its output. As described elsewhere in this article, pitch-shifting can work well in conjunction with amp simulation, but other ways of editing and processing the raw guitar file before it goes through the amp modeller also yield interesting results. Reverse reverb, resonation, vocoding and Auto-Tune can all produce distinctive effects. Try chopping small sections of guitar out, for an interesting stuttering effect that's nothing like tremolo. A piece of guitar that's been reversed before being fed through an amp modeller sounds quite different to what you get by reversing a guitar part that's already been through an amp, and this technique can be very effective. Likewise, recording three or four separate tracks of single guitar notes and routing them simultaneously through the same guitar amp simulator sounds very different from playing chords. Sam Inglis Re-amping a DI'd keyboard or bass can really liven up a sound, but if you don't have access to a nice amp or amp modeller, you can simulate the effect by sending the audio to a bus with a delay plug-in set to a short delay time and with the wet signal set to 100 percent and dry to 0 percent. Then send the bus's output to another bus with a distortion (or better still, a guitar amplifier emulator) plug-in inserted. This simulates the delay you get from miking up a speaker, and if you blend this in with the DI'd sound, it can give the recording a live feel — especially if you use a convolution reverb to add some 'room' ambience. You may also want to roll off the very low and high frequencies to help get rid of that DI'd vibe. Nicholas Rowland

Modulated Delays

Effects such as chorus, phasing, flanging and pitch vibrato are created using pitch modulation and, except in the case of vibrato, the modulated sound is added back to the original to create the effect. The pitch modulation is generated by delaying the signal by just a few milliseconds, and then modulating the delay time, using a low frequency oscillator (or LFO). For vibrato, this is all that needs to be done and, because the delay time is actually very short, the effect is perceived as happening in real time. The other effects, however, generally rely on an equal balance of the dry and modulated signals to achieve the strongest effect, so it is easier when working with plug-ins to adjust the wet/dry balance using the plug-in controls, rather than adding the wet only signal via a send/return loop. As a rule, these effects aren't very processor-intensive so, if you're working with plug-ins, you can probably afford to insert as many as you need into track or bus insert points as required. Stereo versions of these plug-ins may generate different modulated delays for the left and right channels to create a more dramatic spatial effect.

Chorus and flanging are created in fairly similar ways, the main difference being that chorus doesn't use feedback from the input to the output and generally employs slightly longer delay times. Phasing is similar to both chorus and flanging, but uses much shorter delay times. Feedback may be added to strengthen the swept filter effect it creates. Phasing is far more subtle than flanging and is often used on guitar parts. With chorus, phasing and flanging, the delay time, modulation speed and modulation depth affect the character of the effect very significantly. A generic modulated delay plug-in allows you to create all these effects by simply altering the delay time, feedback, modulation rate and modulation depth parameters. Most of the time, low modulation depths tend to work well for faster LFO speeds (often also referred to as the rate), while deeper modulation works better at slower modulation rates.

Chorus is useful for 'softening' rhythm guitar or synth pad sounds, but it does tend to push sounds further back into the mix, so it should be used with care. Adding more brightness to the sound can help compensate for this effect. Chorus also works well on fretless bass, but tends to sound quite unnatural on vocals. Phasing can be used in a similar way to chorus but, whereas chorus creates the impression of two slightly detuned instruments playing the same part, phasing sounds more like a single sound source being filtered, where the frequencies being 'notched out' vary as the LFO sweeps through its cycle.

Flanging is the strongest of the standard modulation effects. The feedback control increases the depth of the 'comb filtering' produced when a delayed signal is added back to itself. Because it is such a distinctive effect, it is best used sparingly, though it can also be used to process a reverb send to add a more subtle complexity to the reverbed sound.

Though modulated delays are essentially effects, the need to balance the dry and delayed sounds as a means of regulating the effect strength means that using these devices via insert points makes them much more controllable than trying to use them in an effects send/return loop. If you do use them as a send effect, you can achieve this balance by automating the send level.

Tips & Tricks: Dynamic Effect Modulation (MIDI Control) One of my favourite hardware effect units is the Electrix Mo-FX (sadly no longer in production). It is superbly constructed for hands-on performance and it offers full MIDI control over the panel's knobs and buttons. I use this in conjunction with the Sequentix P3 (a hardware step sequencer). Not only can the P3 generate patterns of controllers suitable for varying multiple Mo-FX parameters, but it can generate evolving or shifting patterns, courtesy of its 'accumulators'. In a nutshell, accumulators are designed to prevent your sequences becoming annoyingly repetitive: controller values (actually values directed at any internal sequencer parameter) can be added or subtracted on each pass of the pattern, with rules and limits directing the behaviour as the accumulation progresses. Digging through the Mo-FX manual quickly reveals all the MIDI Continuous Controllers you need. Usefully, you can also trigger the tap-tempo function via MIDI, and this offers a rather wonderful way of generating clock intervals. As you can decide exactly where to place your tap-tempo trigger events, and the P3 sequencer can shift or vary these events according to rules you devise, you can find clock sync intervals unseen on any other device. Paul Nagle

Pitch-shift

Pitch-shifters work by slicing the incoming audio into extremely short sections (typically a few tens of milliseconds long) and then lengthening each section where the pitch is to be decreased, or shortening each section where the pitch is to be increased. Though cross-fading algorithms and other techniques are used to hide the splice points, most pitch-shifters tend to sound grainy or warbly when used to create large amounts of shift (a couple of semitones or more), though they can sound very natural when used to create subtle detuning effects, using shifts of a few cents. A refinement of the system, designed for use with monophonic sources, attempts to synchronise the splicing process with whole numbers of cycles of the input signal, which makes the whole thing sound a lot smoother but, as soon as you present these devices with chords or other complex sounds, the splices again become audible.

Though some sophisticated processors combine pitch detection with pitch-shifting, to generate musically correct harmonies in user-defined keys, simple pitch-shifters always change the pitch by the same number of cents or semitones. In musical terms, that means that only the octaves, parallel fourths and fifths are very useful. Other intervals tend to sound discordant, as they don't follow the intervals dictated by typical musical scales.

When using subtle detuning to thicken a sound, I suggest trying values of between five and 10 cents and, where possible, adding both positive and negative shifts, to keep the pitch centre correct. Then combine with the dry sound and adjust the level to control the subjective depth of the effect. This is very effective for fattening up guitar solos or backing vocals.

By putting a pitch-shifter before a delay, then feeding some of the output of the delay back to the input of the pitch shifter, you can create delays that keep climbing or falling in pitch as they recirculate. Though not always very useful in a musical context, this effect is often used in TV and film dream sequences.

Because large pitch-shifts can sound grainy, it is common to combine the effect with the dry signal, rather than using only the 100 percent effected signal, though ultimately this is an artistic rather than technical decision.

Though pitch-shifting is an effect, it is easier to control when used via an insert point. However, if you need to use the effect on several tracks in varying amounts, you can use it via a send/return loop, providing the shifter is set to 100 percent wet. That way, you can adjust the effects depth for individual mix channels by using the send control feeding the pitch-shifter.

Tips & Tricks: Circuit Bending There are an awful lot of boutique guitar effects manufacturers out there who make pedals designed to create all kinds of twisted and bizarre sounds. Sadly, their products are often very expensive, often prohibitively so — so what about the more budget-conscious would-be sonic terrorist? Well, one option is to 'circuit bend' more conventional (read 'cheaper') guitar effects. The basic idea behind circuit bending is that you experiment with short-circuiting the pedal until it makes a noise that you like, and then solder in a connection, with a switch or potentiometer in place if you think you may want to turn the noise off again at some point. Guitar pedals are perfect for this. Firstly, and perhaps most importantly, because you should never try to circuit bend anything mains-powered, they can run off 9V batteries. Secondly, their internal circuitry is usually very simple, and they already have audio I/O. Thirdly, you can get them for almost no money from eBay, and the other tools required — soldering iron, wire, switches, and so on — are also very cheap. There's an almost infinite number of sonic possibilities to be explored here, from finding new ways to process a signal (of course you don't just have to use them with guitars) to creating a machine that goes 'Eeeeeeooooowsquelch blipipipip' in a different way every time you turn it on — and who would say no to that for less than a tenner? For more information and some examples of unintentional sonic mayhem, check out www.circuitbenders.co.uk and www.anti-theory.com/soundart/circuitbend. Have fun. Rob Kemp

Tuning Correction

By contrast, tuning (or pitch) correction processors and plug-ins are normally considered processors rather than effects, but they do have creative uses. The idea behind these devices is to monitor the pitch of the incoming signal, then compare it to a user-defined scale, which can be a simple chromatic scale or any combination of notes. Pitch-shifting techniques are then used to nudge the audio to the nearest semitone in the user's scale but, because the amount of pitch-shift required is usually quite small, the result doesn't sound grainy or lumpy, as often happens when large amounts of pitch-shift are generated. Because pitch tracking is used to identify the original pitch, only monophonic signals can be treated.

When used with the human voice, it is important that the pitch correction doesn't happen too quickly, otherwise all the natural slurs and vibrato will be stripped out leaving you with a very unnatural and robotic vocal sound. If only a few notes need fixing, consider automating the pitch-corrector's correction speed parameter so that it is normally too slow to have any significant effect, then increase the speed just for the problem sections. This prevents perfectly good audio from being processed unnecessarily.

If you stick to a simple chromatic scale (all the semitones), you also run the risk of the pitch correction moving the audio to the wrong note if the singer is more than half a semitone off pitch. A user scale, containing only the desired notes, generally works much better. Some systems also allow you to dictate the correct notes via MIDI. If the song contains sections in different keys or that use different scales, it is often simplest to split the vocal part across several tracks and then use a different pitch-corrector on every track, each one set to the appropriate scale for the section being processed.

If your audio track suffers from a lot of spill, or includes chords, the pitch correction may not work correctly. Where spill is loud enough to be audible, you'll hear this being modulated in pitch alongside the wanted part of the audio as it is corrected. As a rule, chords are ignored, so guitar solos, bowed stringed instruments and bass parts (including fretless) can be processed, and only single notes will be corrected.

The main creative application for pitch correction is the so-called 'Cher effect', which is achieved by setting the tracking speed as fast as possible to deliberately generate a robotic-sounding result. It's a matter of taste, but for me, this is one effect that has already been done to death!

Tips & Tricks: Delay Tempo Delay: Most plug-in and hardware delays now allow you to automatically sync delay times to MIDI clock and then specify the interval of the repeats in terms of note values rather than milliseconds. A trick here is to use two simultaneous tempo-based delays with, say, a triplet delay setting, panned hard left, and a straight-note delay panned hard right. Things can get more interesting still if you apply this technique using ping-pong delays, so that alternate repeats bounce from one side of the stereo spectrum to the other. To create a true 3D effect, play around with the amount of original signal left in the middle. Depending on the intervals between your repeats, you can turn simple guitar and synth lines into complex, arpeggiator-like patterns or totally spaced out ambient pieces. Stephen Bennett Ostentatious Delays: If you're making very rhythmic music of any kind, it makes sense to use tempo-sync'd delays, to avoid undermining the main pulse. However, simple tempo-sync'ed delays tend to be masked by the main rhythmic stresses, so they sink into the background of the mix unless mixed very high in level, which makes it difficult to create ostentatious delay effects in rhythmic music without swamping your mix. One solution to this problem, very common in trance music, is to set a delay to a three-16th-note duration, which means that although the delay repeats never step outside the 16th-note grid, they'll often miss the main beats and therefore remain clearly audible. Mike Senior Keep It Reel: Perhaps because a humble tape echo was the first effect I ever owned, delay has always been my primary effect. Whether to liven up repetitive loops or add apparent complexity to simple solos, it's worth getting to grips with delay the old-fashioned way. This means daring to switch off MIDI sync and manually setting delay time, driving feedback to the brink of madness, or routing the pure delay output through equalisers, filters and so on. Many of today's digital delays allow you to darken the delay iterations, but there's no reason not to find your own method to achieve this: adding alternative colours and discovering your own favourite processes. I find precise, perfect digital delays can be rather generic and characterless — so the more I delve into additional treatments, the more interesting and organic the results are. Paul Nagle Softer Delays: I'll usually have at least a couple of delays as auxiliary effects in a rock or pop mix, but I often find that bringing the general level of the delay as high as I want it makes any transients stand out too much. When I'm sending single notes on a clean electric guitar to a delay line, say, I tend to want to hear a wash of sound, not the rhythmic 'CHA-Cha-cha-cha-cha' of a repeated note attack. For this reason, I'll often put a gate or expander before a delay, with an attack time set to 10ms or so. This is enough to 'chop off' any abrupt transients, and makes the delay sound much smoother. Sam Inglis Non-sync'd Delay: We are so used to perfectly sync'd delays that it's easy to forget that manual sync and a pair of ears has a charm all of its own. Even delay times that bear no obvious relationship to the tempo can add dynamic movement and feel to a track: check out some early King Tubby if you need reminding of this. Paul Nagle Subtlety: You don't always have to make longer echo or delay effects obvious in the mix for them to be effective. Once you've set up the delay times and panned them to suit your song, try dropping the delay levels until you scarcely notice them during most of the mix (listening on headphones often helps set the most suitable level). This generally results in intriguing little ripples of repeats that you notice at the end of verses or during pauses, that add interest and low-level detail to the mix. Martin Walker

Distortion

Technically, distortion is defined as being any change to the original signal other than in level. However, we tend not to think of processes such as EQ and compression as distortion, and the term is more commonly used to describe processes that change the waveform in some radical and often level-dependent way. These include guitar overdrive, fuzz, and simply overdriving analogue circuitry or tape to achieve 'warmth'. In the analogue domain, heavy overdrive distortion is usually created by adding a lot of gain to the signal to provoke deliberate overloading in a specific part of the circuit. Such high levels of gain invariably bring up the level of hum and background noise, so it may be helpful to gate the source. Though overdriving analogue circuitry is the traditional way of creating intentional distortion, we now have many digital simulations, as well as some new and entirely digital sound-mangling algorithms.

The most musically satisfying types of distortion tend to be progressive, where the audio waveform becomes more 'squashed' as the level increases. Hard clipping, by contrast, tends to sound harsh. All these types of distortion introduce additional harmonics into the signal, but it is the level and proportion of the added harmonics that creates the character of the sound. Harmonically related distortion can be added at much higher levels than non-harmonically related distortion before the human hearing system recognises it as such, so there is no way to define a percentage of distortion below which audio is acceptable or above which it is unacceptable. The reason that digital distortion has its own character, which most people find less musically pleasant, is because it is not usually harmonically related to the input signal. For example, quantisation distortion, which results from sampling at too low a bit depth, sounds quite ugly, though many dance and industrial music producers have found a use for it, and some plug-ins deliberately introduce it.

The use of overdrive distortion as a musical effect probably originated with electric guitar amplifiers, where the less pleasant upper harmonics created by overdriving the amp are filtered out by the limited frequency response of the speaker. If you use a distortion plug-in without following it up with low-pass filtering (or a speaker simulator) in this way, you may hear a lot of raspy high-end that isn't musically useful. This is why electric guitar DI'd via a fuzz box or distortion pedal sounds thin and buzzy unless further processed to remove these high frequencies.

The warmth associated with tube equipment and analogue tape is quite subtle when compared with deliberate overdrive effects. As a rule, if you're trying simply to warm up a sound and you can hear the distortion, you should back off it a little, as there's probably too much of it.

Adding a little distortion to sounds such as drums, electronic organs, and even vocals, can help them stand out in a mix, and give substance to a sound that's too thin or uneven. Software guitar amp models often sound more convincing if you use external guitar pedals to create overdrive prior to the audio interface.

A Word Of Warning

Effects are fun, and can make mixing a more creative process, but it's worth bearing in mind that they won't help in situations where the basic principles of recording have been ignored! Used with care, effects can help turn a good mix into a great one, but they are seldom successful in covering up other problems. It is also very easy to over-use them — sometimes their most valuable control is the bypass button, and it is certainly worth learning to use the basic effects well before throwing lots of complicated tricks at your sound. As long as you let your ears decide what is right, you should be OK, and a little critical listening to your favourite records will give you a feel for what works and what doesn't. 

Published in SOS May 2007

Focusrite Control 2802 - Musikmesse 2011

Mechanical FX

A DIY Guide To Unusual Audio Effects

Technique : Effects / Processing

Some of the best signature sounds have been developed through experimentation with mechanical devices. We offer some inspiration...

Paul White

When we think of creating effects in the studio, we invariably think first of electronic processing using hardware or plug-ins, but many of the great sound effects and sound processes heard both on film soundtracks and records were created by mechanical rather than electronic means.

Classic Mechanical FX

A prime example of this in the film world is the laser gun sound used in Star Wars, which started life as someone hitting the steel support wires for a radio mast or similar structure. The resulting twang has rather more attitude than most synthesized sounds.

It's the same in music, where one of the most established effects, the rotary speaker cabinet used with electric organs, works via a system of rotating baffles and sound guides rather than any form of electronic manipulation. Essentially the audio is fed through a crossover to split it into high and low frequencies, and each band is sent to its own amplifier. The low end feeds a conventional cone driver, facing upwards or downwards into a rotating drum that has an angled reflector inside and an aperture in the wall of the drum to allow the sound to escape. As the drum rotates, the sound is beamed out, rather like the light from a lighthouse. This results in an audibly complex modulation, involving level and frequency content, with a hint of Doppler shift for good measure.

One of the most famous and instantly recognisable mechanical effects: the Leslie rotating speaker. You can use similar principles to create DIY mechanical effects.

The high frequencies go into a fixed tweeter, which feeds into a rotating horn system, usually counterbalanced by a dummy horn. Again, the sound is sprayed out in all directions as the horn rotates and, as each rotor has its own motor, the speeds are never exactly the same, which adds to the complexity of the effect. The performer changes the rotation speed or applies brakes during the performance, and a novel characteristic of the system is the way in which, when the speed is changed, the rate of rotation ramps up and down due to mechanical inertia. Because of how the moving sound source interacts with surfaces in the room, the result is actually very complex and quite difficult to synthesize precisely — though there are some very plausible electronic emulations.

There's also the spring reverb, which uses the mechanical properties of a loosely tensioned helical spring (or, more often, two or three springs: see the picture at the top of the page) to transmit audio from a small transducer to a pickup at the other end of the spring. The complex way in which the sound travels along the spring, and is reflected back and forth along its length, produces the characteristic sound. By combining two or more slightly different springs, the sound can be made more complex.

The familiar plate reverb is also mechanical but, instead of a spring, it relies on a fairly large, thin, suspended metal plate. A transducer fed from the aux send of a mixer (and then amplified) drives the plate into vibration and the resulting reverberation is picked up using contact mics on the surface of the plate. Placing two different contact mics in different positions on the surface of the plate, then feeding these back as two channels panned left and right, creates a pseudo-stereo effect. Plates have a very dense reverb, with a slight metallic coloration, but they're still valued today. In fact, many electronic reverberation devices include plate emulations or, in the case of convolution reverbs, impulse responses taken from real plates.

A plate reverb (above) is a mechanical device that uses contact mics on a resonating metal plate. The plate reverb sound remains hugely popular, and has been painstakingly replicated in the software domain, Universal Audio's Plate140 (left) being one example.

A slightly less well-known early mechanical reverberation device was the Cooper Time Cube. I've never heard one of these working, but the principle involved feeding the sound from a loudspeaker down a long length of coiled hose (folded into a cube-shaped box, hence the name), then picking it up at the other end using a microphone. I'd imagine that some serious EQ would be necessary to make the sound useable, but as with so many mechanical devices, the unusual character was no doubt part of its charm. While apparently simple ideas, what occurs during all these mechanical reverberation processes is actually quite difficult to emulate electronically, though the modern generation of convolution processors can capture them pretty effectively.

Some of these reverb effects can be found in the most unlikely of places. A simple spring reverb system is used in the familiar 'toy' microphone which adds apparent reverb when you sing into it. Here, a plastic diaphragm is forced into vibration by the voice and this, in turn, feeds energy to the internal spring, producing an entirely acoustic reverb that uses the mic body and the plastic diaphragm as soundboards to amplify the sound. In fact, you can use these in the studio to add a quirky kind of reverb to your own mixes, by simply miking the result of singing or playing a sound into the mic. As the plastic body also acts as a soundboard, it is also well worth experimenting with contact mics (the sort of stick-on contact mics that are suitable for guitars should work just fine) as a means of collecting the reverberant sound without getting too much of the direct sound, and this should allow you to add reverb to sounds generated by loudspeakers, as well as the voice and other instruments played close to the microphone.

I mentioned earlier the laser gun sound from Star Wars. Another classic sound effect, which may generate useful ideas for the music studio, is another Star Wars favourite: the light sabre. The humming energy of this fictitious weapon was apparently created by feeding 60Hz US mains hum and buzz from a cheap portable cassette deck (discovered when it picked up interference from a nearby TV set) into a pair of loudspeakers. Then the sound designer simply moved a very directional shotgun mic between the two speakers, in time with the action on screen. What happens here is that the sound from the closest speaker reaches the mic slightly before the sound from the other speaker, so you get some frequencies cancelling and some adding, as you do with a phaser or flanger that uses electronic time delays. As the mic is moved between the speakers, the time differences and audio levels vary, which creates the apparent movement in the sound.

More DIY Tricks

You can obtain an interesting phasing effect by playing a track through a speaker and recording the sound while moving the mic around.

The light sabre effect was great in the film but there are also potential applications in the music studio. I wouldn't normally suggest doing this to a whole mix, but it can be effective on 'trippy' sounding vocals, especially if they already have a bright reverb added, and distorted guitar and complex-sounding keyboard pads also respond well. You can try to approximate the technique using any directional mic and a pair of studio monitors, the main proviso being that both speakers must be fed from the same audio source (ie. a single mono signal — the effect won't work with a stereo one). Record the result from the mic and you should hear a gentle, very organic-sounding phasing effect. It can be controlled fairly precisely by how quickly and how far you move the microphone relative to the two speakers. As the sound level picked up between the speakers will be less then when the mic is close to one of the speakers, you may want to experiment with mic distance to get the most consistent possible level, then use compression and/or mix automation to level up the parts afterwards.

Another trick you could try is to put a basic dynamic mic or pencil-style capacitor mic inside a length of plastic waste pipe so that the sound has to travel down the pipe before it reaches the microphone. This will make the microphone much more directional and also add a lot of tube-style resonance, which can be effective when used creatively. You can tune the tube resonance by moving the mic further up the tube to shorten the amount of tube in front of the mic.

Rotating Phasey Weirdness

Let's cast our minds back to the rotating speaker cabinet mentioned at the beginning of this article. If you can't afford the cost or space of a Leslie speaker, you can create DIY variations on this effect by using a stationary speaker and moving the mic, or by putting something like a slow-moving room fan between the mic and the speaker. If you have ever recorded a guitar combo, you'll already know that the tone of the recording changes as you move the mic across the front of the speaker cone. If you manually wave the mic to and fro in front of the speaker while someone is playing (or while you re-amp a DI part you've already recorded), variation is created in the tone as you record, which can result in a really nice sound that is halfway between a rotary speaker and phasing.

Swingers

Suspending a mic above the speaker and swinging it, rather like a pendulum, gives an effect that is somewhere between a rotary speaker and phasing. By setting up more than one mic in this way and varying the direction and degree of the swing, you can build more complex and interesting effects.

Another variation, this time creating a slow, regular sweep, is placing the amplifier face up, then fixing the mic cable to some point above it that allows the mic to swing, pendulum-style, over the speaker. Set the mic swinging gently and record the results. More complex results can be obtained by setting up two mics that swing at different rates and then summing their outputs to mono, as you'll get phasing characteristics caused by the path-length differences from the speaker to the two mics at any given time. Just make sure the mics don't swing into each other! Simple dynamic models are probably the best for this purpose, as they tend to be more rugged than capacitor mics, and using two identical mics will give the most pronounced effect. Two people can move the two mics independently in a more controlled way, to create a less cyclic modulation effect. If you independently compress the outputs from the two mics, to stabilise their levels as much as possible before summing them, the strength and consistency of the effect should be better.

Another option is to use a slow-moving, large-bladed fan placed between a fixed mic and speaker (or even somebody waving a reflective piece of board around between them!). If you try this, you should put a pop shield in front of the mic, otherwise the strong air stream might cause popping. While a fan will generate mechanical noise and air movement noise, the sound from a typical guitar combo will usually be loud enough to render this insignificant.

Talk Is Cheap

A couple of decades ago, another popular effect was the guitar 'talkbox', and it's still used even today. This simple device feeds the guitar sound to a small loudspeaker, up a length of hosepipe and into the performer's mouth. By changing the shape of the mouth, the instrument can be made to appear to 'talk' in a similar way to a vocoder. Clearly the technique isn't only limited to guitars, as synths or even complete mixes can be treated in this way.

Again, you can take the principles of the talkbox to create DIY alternatives. If you have a small speaker or practice amp, try taping the wide end of a plastic funnel over the speaker, so as to seal any air gaps, then fit the end of a piece of hose over the narrow end of the funnel. Put the end of the hose in your mouth (it's easiest if you tape it to your mic stand), set the audio running and just mouth the words! A mic set up as you would to record vocals will capture the processed sound, ready to record.

The guitar talkbox was famously used by Peter Frampton, Pink Floyd and others. As with other such mechanical effects, you can improvise some interesting DIY alternatives.

This effect is essentially a form of filtering, which is controlled by the resonant shape of the mouth, so it will be at its most effective if you start out with harmonically rich material such as a distorted guitar sound or a complex synth sound.

Summary

These are just a few examples of mechanical sound effects with DIY suggestions that you can try at home without any great financial outlay. There are lots more options to explore — it just takes a little lateral thinking and a willingness to experiment. I'll never forget the effect we discovered during a reader SOS visit, where the studio owner demonstrated a guitar playing technique that he'd developed using a lady's, er, personal massage device to excite the strings (as opposed to the lady). The motor also picked up on the guitar's single-coil pickups to he could use the speed control on the device to alter the pitch of the added drone. Apparently, he got some odd looks in the Ann Summers shop where he auditioned a number of such devices to find the most suitable, but it just goes to show that inspiration can come from any source!

The way interesting sounds can be created or modified never ceases to amaze me, and the best thing is that they usually cost little or nothing to try. The ideas can be very simple, or they can be more complex, such as the ill-fated Gizmo, financed by the band 10cc, which used a series of rotating, toothed, nylon wheels to physically bow the strings of an electric guitar. You may also find inspiration from ancient instruments, such as the Aeolean harp, which is played by the wind. You'll never know just how close I came to adapting my wind chimes to generate MIDI notes, so that I could let the wind choose the tunes while I chose the sounds! This would be easy enough to do using cheap piezo discs glued to each chime, feeding the signals to the pad inputs of a MIDI drum converter, such as an Alesis D5, and then sending the resulting MIDI notes out to the synth or plug-in of your choice. In the end, the results you get from experimenting in this way are usually very different to what everyone else is getting from plug-ins and they can help you to develop a truly innovative and unique sound. If you come up with anything particularly novel, please let us know! 

Published in SOS January 2007

Korg Monotribe - Musikmesse 2011

Tape Delay In Your DAW

Using A Reel-to-reel Recorder For Echo Effects

Technique : Effects / Processing

Second-hand tape recorders are cheap and can provide uniquely warm, organic delay effects. Best of all, it’s a piece of cake to integrate them into a digital mixing setup. We tell you how...

Will Haas

You should be able to obtain a vintage two-track tape recorder such as this Otari for relatively little money.

Legend has it that Sun Studios owner Sam Phillips stumbled upon the technique of creating echo effects using a reel-to-reel recorder by mistake. Whether or not he was the first recording engineer ever to produce this effect, it is safe to say that his studio made the sound famous. Scotty Moore, Elvis Presley’s electric guitar player, liked the sound so much that he even had an echo device built into his guitar amp, so that his echo sound was not just an effect, but an integrated part of the guitar sound. When Elvis moved on to RCA, the RCA studio engineers used this same technique on his hit songs of the time.

The character and depth of sound that was produced from tape echo on these old records is extremely lush, warm and wide, and although recording studio technology has changed so much since the ‘50s, some of the biggest-sounding records being mixed today still use this classic technique. In this article, I’ll be explaining how to integrate a real tape-based delay into a digital mixing setup.

Before we get to the mechanics behind actually creating that beautiful lush echo effect, you’ll need a tape machine. For tape delay you have two avenues to choose from: use a dedicated echo device such as a vintage Echoplex or Copicat, or use a three-head reel-to-reel tape machine. Dedicated echo devices will give you more delay-time options, because most units have variable playback head locations, but these vintage units are relatively expensive and hard to find in good working order. Also, some require proprietary tape cartridges that are difficult to locate. For these reasons, I’ll concentrate on how to achieve echo effects using conventional mono or stereo reel-to-reel tape machines. These give you fewer options for delay settings, but are very reliable and do not require dedicated tape cartridges.

You will require a three-head tape machine — in other words, one with separate erase, record and playback heads. I recommend a quarter-inch two-track machine that can run at either 7.5 or 15 inches per second. You should be able to find a decent used machine that works well and is great for echo for around £150-250. My dedicated tape echo machine is an Otari MX BIII, which I bought for £150 from my local university radio station. You may find some much cheaper, if you don’t mind them being in less than perfect cosmetic condition. You’ll also need some quarter-inch analogue tape, but this is relatively cheap compared to two-inch and other multitrack tape sizes. You may even score some used tape with your machine purchase. 

Using a tape echo as an external effect in Pro Tools: the Echo bus contains an insert called A6, which is routed out of the HD interface, through the tape recorder and back again.

When you’re using a tape machine only for tape echo effects, it doesn’t require perfect calibration. Inconsistencies with EQ and playback reproduction are, to a certain extent, what you are looking for when it comes to delay sound, because they help to give tape delay its unique character. Every tape recorder sounds slightly different, whereas the same two plug-ins sound identical. A little wow and flutter is what you want, and as long as the machine records, erases and reproduces signal properly, you are in business. The only maintenance required will be to clean the heads periodically with alcohol and demagnetise them.

Setting Up Your Tape Recorder

The idea behind getting a tape echo effect to work is quite simple. Most tape recorders have a switch allowing you to monitor either the signal that has actually been recorded to tape, or the source input. When you are monitoring in Tape mode, there is a slight time delay between input and output, because it takes time for the tape to pass from the record head to the playback head. Tape echo is achieved by routing the source signal from your DAW to the tape recorder, monitoring it in Tape mode, and mixing this delayed signal back into the mix bus on your DAW. I am an avid Pro Tools user, so I will illustrate how to go about this operation in Pro Tools, but the concept is the same in all digital systems. As long as you can route analogue signals to and from your DAW, you should be able to use tape echo.

The Echoplex is one of many classic tape-based delay units that now command high secondhand prices.

First off, I create an auxiliary track (mono for most cases) and label that track ‘Echo’. I then bus a post-fader or pre-fader send to that auxiliary track’s input. My tape machine’s inputs and outputs are connected to an analogue patchbay, which is also wired to my Pro Tools HD192’s analogue inputs and outputs. I use an analogue insert on the auxiliary track’s signal path to route that track to and from the tape machine. That’s basically it! Now all you have to do is press Record and Play on your tape machine and you will have a delayed signal appearing on your auxiliary track. Just make sure that you are monitoring Tape on your tape machine and not Source, or you will be hearing just the latency from your DAW converters and not the actual tape echo! Make basic level adjustments and you are off to the races.

More Flexible Than You Might Think

The slower the tape speed setting on your machine, the longer the delay time. I find that for a lot of tunes the 7.5ips delay time is magical, but for faster groove-orientated material, the 15ips speed gives a tighter delay time. Although you are limited to these two delay times, there are lots of ways to carve your sound to give the delay a character that is really cool and unique. This is where you can get experimental. Because your echo effect track is on an auxiliary fader that is output to your main mix bus, you will have full creative control over how you want to further enhance the effect’s sound. Compress the track, EQ it, route that delay to the tape machine’s other channel, use reverb... The options are endless. You can also add other plug-in effects. One of my favourite effects is distorted echo, which I achieve by overdriving the tape machine’s line inputs to get a lovely saturated delay sound. Also, experiment with panning and overall level automation for different parts of the song. Tape selection also changes the sound of your delay. You may want to invest in some high-quality reels, but it’s not essential. Whatever type of tape you choose, each reel will last numerous spins. Don’t throw away all of your degenerated tapes, either: you might just want a really gritty delay sound for certain applications.

If you can find vintage echo units that work, like Michael Brauer’s collection, you are very lucky, so take care of these devices. Fortunately there are lots of reel-to-reel tape machines still available for your echo needs, but in 20 years or so this may not be the case. Get your hands on one now, and get the benefits of a dedicated tape echo machine without the cost, by taking advantage of a used tape machine and making it your own dedicated echo tool.  

Published in SOS August 2007

Adam A77X - Musikmesse 2011

Alesis Mic Tube Solo - Musikmesse 2011