Whether you're a beginner or a seasoned pro, there's always something to learn about adding colour to your mix.
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.
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).
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 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.
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
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.
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.
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
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.
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-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.
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?
Have fun. Rob Kemp
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!
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
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.
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.
Decay Settings: Choosing the most appropriate reverb treatment for a song can be surprisingly difficult, especially if you have hundreds of presets to choose from. So, instead of regarding reverb like the glue that holds the mix together, try adjusting its parameters (and in particular the decay time) while listening to the reverb return by itself. If the decay time is too long you'll hear a continuous mush of sound; if it's too short you'll scarcely hear it unless its level is turned right up. Somewhere in the middle you should find a setting that adds rhythmic interest to your song, without overpowering it, making the reverb work for its keep. This is also a useful technique when using several reverbs in a song, to make sure they complement each other. Martin Walker
Pre-delay: No pre-delay? No problem! Some reverb plug-ins, from freeware favourites to tasty convolution types, don't offer pre-delay — a user-configurable gap before the onset of a reverb's early reflections and tail. It's useful to have, though, as it can contribute to the clarity and separation of individual voices and instruments in a mix when large amounts of reverb are used. Using most software DAWs it's straightforward to rig up a pre-delay for a reverb (or any other effect) that doesn't have one. All you do is set up your reverb on an aux track or channel, but place a simple delay plug-in in a slot above it. Set both plug-ins' wet/dry mix parameters to 100 percent wet, and feed them some audio using an aux send on your normal audio tracks. Now the delay plug-in operates as a pre-delay for the reverb: easy! This kind of 'modular' pre-delay actually opens up some interesting possibilities. By using a multi-tap delay, or a simple delay with some feedback, your dry signal can be fed to the reverb several times, making for longer, more complex — or plain weird — reverb tails. Robin Bigwood
Wet Set: If you have a sound that you want to push a long way back in the mix, it can often be better to make your reverb effect pre-fader, and temporarily remove all the dry sound. Then alter the sound's EQ and reverb settings while listening only to the wet reverb sound. Once you've got that sounding good, gradually fade the dry sound back in until you're happy with the wet/dry balance. This approach can often be more effective than simply whacking up the reverb level while you listen to the whole song. Martin Walker
Combining Reverbs: You don't have to generate all of the reverb sound from a single plug-in, and using two different reverbs can also help you to save CPU power. For example, though a nice convolution reverb gives a good, believable sound, long impulse responses tend to eat up CPU. By using the convolution reverb for the early reflections, and then using something like Logic 's Platinumverb or Waves Trueverb to add the reverb tail — which is less critical to our perception of the sound — you should get a convincing but less processor-intensive result. Matt Houghton
Don't forget that you can create audio-style effects purely through MIDI. For example, using a grid-style sequencer, it's very easy to program in echo and delay effects, just by drawing in the repeated notes and then putting a velocity curve over the top to simulate the echoes fading away. By combining this with automated MIDI control of other parameters — reverb send, filter cutoff and resonance, for example — you can alter the timbre of the repeated note and create dubby-sounding, feedback-style delays. Stephen Bennett
They may not always be the first thing you reach for, but the MIDI effects plug-ins that come with most DAW applications like Logic and Cubase often offer something very different from most audio plug-ins. For example, arpeggiators and step sequencers can be great for use in the composition process, and you can use MIDI note to controller data (CC) plug-ins to generate automation data for the parameters of other plug-ins. As they process only MIDI data, and not audio, MIDI plug-ins put very little strain on your computer. Matt Houghton
You don't have to create audio effects in your sequencer. For example, I use the Access Virus synth, which features a simple delay effect, with the added bonus that all its parameters are available in the modulation matrix. One favourite trick involves routing velocity to the delay colour parameter. For parts that get brighter with increased velocity, it adds extra animation and bite if the echoes also get brighter. Unusually, the Virus also features four-way audio panning, so you can position an audio signal anywhere between the main stereo outputs and a second pair. If the second pair of outputs is routed to an external effects unit, you can play with the concept of moving a note around in a space, where its position also determines the treatment it gets. More fun can be had by modulating reverb time and colour via an LFO. The same LFO can then be used to control filter cutoff, EQ frequency and maybe wavetable position too (if your Virus is a TI). In this way, timbral changes happen at the same time as effect changes. Paul Nagle
Vocal Widening: One of the send effects I most frequently use at mixdown has got to be the classic vocal-widening patch that I always associate with the vintage AMS DMX1580 delay unit. From a mono send a stereo ADT-style effect is created using two pitch-shifting delay lines, panned hard left and right. Normally, I set the first channel to 9ms delay, with a pitch shift of -5 cents, and the other channel to an 11ms delay, with +5 cents of pitch shift. That said, though, I will often tweak the delay times a few milliseconds either way, as this can dramatically alter the effect's tonality. Mike Senior
Processed Pitch Shifts: Few pitch-shifting algorithms are transparent enough to allow you to transpose anything by more than a couple of semitones without obvious side-effects. If what you're processing is going through an amp modeller, however, you can get away with much more radical changes. You can even do effective swoops and dives in pitch by progressively increasing the amount of pitch-shifting you apply to a note, and pitch changes of an octave or more can sound good, although they probably won't sound natural at these extremes. Sam Inglis
Modern audio sequencers make it very easy to play around with spot effects — that is, effects which are applied to single notes or phrases within a track, rather than to a pattern or track as a whole. Try using different reverb styles on the snare within drum patterns: a short decay on the '2' and a long decay on the '4' for example. Another idea is to apply spot chorus to individual words within a vocal line, as a way of adding emphasis to the lyrics. The 'freeze' or audio bounce-down function of a typical sequencer allows you to get around any problems your computer might have in running lots of instances of a particular effect. Stephen Bennett
An enduringly popular effect, with all sorts of uses for vocals, drums, guitars and synths, is genuine reverse reverb. This is where a reverb tail appears to increase in volume ahead of the sound that gives rise to it — a completely unnatural sound and something that's impossible to create in real time. It's easy enough to do in a DAW application though. First, find the section of audio you want to treat and reverse it. In the DAW I use, Digital Performer, there's an offline plug-in for this. Now apply conventional reverb to this 'backwards' audio — either by playing it through a reverb plug-in and recording it to another track, or by rendering it using an offline process (if your DAW offers this). In either case make sure you allow enough additional time at the end of the audio to capture the full, final reverb tail. In the screen grab I've done this by allowing 1500 milliseconds of post-roll processing.
Then, reverse the resulting audio. The original audio plays correctly once more, but the reverb you just applied is reversed. You may need to manually realign the audio in your track, to accommodate the extra length of the first reversed reverb tail. Robin Bigwood
For me, the hardest part of mixing is getting the vocals to sit properly. There are a lot of tricks you can apply that can help, but I think one of the most useful is to send the vocal to a bus and insert a compressor there, with a high ratio of around 10:1 or more. Set a low threshold, and a medium attack and release, then, in the next slot, load a distortion plug-in with a warmish sound. Use high- and low-pass filters, set to around 100Hz and 5KHz respectively, and mix a small amount back in alongside the lead vocals. You don't need to add much — it should be almost 'subliminal' — but it can really help to fit the vocal in the track. Nicholas Rowland
A direct user interface can give far more musically rewarding results than dozens of parameters, menus and alpha dials. Often, even a panel of knobs isn't anywhere near as natural to play as, say, a Korg Kaoss Pad. Here's something Kaoss Pad 3 owners can try at home: choose effect DL2 (Smooth Delay) in which the pad controls delay time on the X-axis and depth on the Y-axis. Next route your favourite solo patch through it and set the FX depth to about 12 o'clock. Solo wildly whilst simultaneously stroking the top right-hand corner of the Kaoss Pad with short, circular motions. With practice, you should be able to produce delicate pitch sweeps as the delay shifts in time. As you control depth by vertical motion, practise diagonal upwards sweeps followed by vertical downward ones to smoothly dampen the effect. Hey, it takes years to master the violin, so a few evenings spent waggling your finger over flashing LEDs shouldn't be too arduous. Next try the same technique with lush solo pads: simple yet devastatingly effective! Paul Nagle
If you want to combine the dynamics of a well-recorded drum kit with the pumping excitement you get from heavy compression, send either the overheads only or the entire kit to a buss and insert a nice-sounding compressor there. Set the compressor to a high ratio and low threshold and mix in some of this with the song. You may need to adjust the attack and release controls to get the effect you're after, but you don't need to blend in much of the compressed sound to really add punch and weight to a drum track. Nicholas Rowland