Basics of the modulation effects

modulation effects

Enhance your music production with effects. Producing music with use of modulation effects can bring to your audio production your personal touch.


Chorusing began as an electronic means of recreating the sound of two virtually identical signals such as a c tracked vocal. Tracked vocals give a natural chorusing effect since even if the singer is extremely accurate there will e be slight variations in pitch and timing which cause some cancellations and reinforcements of the waveform. 12 string guitars are a good example of chorusing in the real world. The strings are tuned in pairs to the same note but they are different gauge so they will never sound exactly the same. Electronically, and now digitally, chorus is created by re-combining a slightly delayed and pitch modulated version of a sound with its source. The mix between the wet and dry signals should be roughly 50:50 or there will be no effect. The pitch modulation of a chorus is achieved using an 110 produces a regular cyclic pattern. This distinguishes it from natural chorus where the modulation is more random and chorusing of this type has become an effect in its own right. It is always better to double track properly if you have the Chorus is commonly used on guitars and keyboards and can give an idea of movement, richness and stereo width. However it can also de-localise the sound so it’s often best reserved for background parts. For example, it can be use good effect to add richness to backing vocals, even if they are already double-tracked! It also famously works well on fretless bass.


Flanging and phasing

Flanging is actually a specific and distinctive type of phasing. It is an effect that originated du the era of the tape machine and which has been copied electronically and digitally. It is created by mixing two identical signals one of which is delayed by a tiny but varying amount. Flanging was first achieved by copying an audio track or second tape machine, playing the two tracks back in sync and mixing the outputs. The engineer would then manually pressure against the flange of the reel on the second machine slowing it down slightly. This would change the pitch ar momentarily place it out of phase and time with the first machine. The outputs of the two machines were then mixed together. The result was a number of constantly changing phase cancellations and reinforcements a bit like a sweeping comb filter. To the ear it sounded like a swooshing ‘drainpipe’ or swishing tunneling sound. In flanging the peaks and troughs of the comb filter are in harmonic series. One of the first instances of flanging on a commercial pop record was the tom fills of the Small Faces 1967 hit ltchycoo Park. Phasing is general perceived as a more subtle effect than flanging. It is created in a similar way. An electronic phaser split the signal into two and shift the phase of one signal (delay it by a tiny amount) by using an all-pass filter and then recombine the signals. The all-pass filter allows all frequencies through at their original amplitude but introduces differ phase shifts at different frequencies. When the filtered and non-filtered signals are recombine peaks and troughs are f along the frequency spectrum. These are not in harmonic series in contrast to flanging. The degree of the phase shift modulated with and LFO causing the characteristic sweeping sound of a phaser.

With flanging and phasing the intensity of the effect (the amount of cancellation) depends on the balance between the original and the delayed sound. The depth control adjusts the amount of variance from the central delay time. The ratio control refers to the speed of the LFO modulating the time delay. This causes the speed of the sweep to change. Fee allows you to recirculate the delay for a more exaggerated effect.

Both Logic and Pro Tools include Modulation Delay plug-ins as part of their standard set. It is possible using this typ plug-in to create chorus, phasing and flanging effects. Logic also helpfully provides dedicated chorus, flanger and phaser plug-ins.

Flanging and phasing can be applied to many types of sounds successfully. The more subtle settings tend to be used on vocals, backing vocals and cymbals. Wilder settings can be commonly heard on bass and electric guitar. They both work well on harmonically rich sounds so overdriven guitars are ideal. In fact many guitarists have flangers and phasers as stomp boxes for that very reason. Flanging is particularly successful on bass, giving a hollow tunnel like effect. This has become something of a cliché so perhaps it’s advisable to steer clear — unless you’re going for that 80s bass sound!


Tremolo and Vibrato

They are both also well known modulation effects. They are simpler than flanging and phasing and originate from being a part of many a musician’s normal repertoire of expression. Tremelo is essentially amplitude modulation and vibrato pitch modulation. Many singers employ both techniques in the course of a performance. On stringed instruments such as guitars and violins vibrato is produced by wobbling one’s finger on the fretboard whilst holding down the string. Confusingly the tremolo arm of a guitar actually produces pitch modulation which is vibrato. As an effect tremolo is not surprisingly great on guitar. Many amps have a tremolo built in like the Vox AC30 for example. Vibrato effects were often built into electric organs such as the Hammond B3. A Leslie rotating speaker cabinet would have the effect of creating both pitch and amplitude modulation simultaneously. These days plug-ins are available to simulate these phenomenon. A chorus with the output set to 100% wet would essentially produce vibrato. Logic Pro 7 includes a Tremelo plug-in, a Scanner Vibrato plug-in and a Rotor cabinet plug-in as standard. Logic’s tremolo plug-in offers a variety of modulation wave shapes and can combine amplitude modulation with auto-panning to create movement in the stereo field, making it a very useful plug-in. Some pitch correction plug-ins such as Autotune include a vibrato option which you can set to kick in after a note has been sustained for a certain amount of time.

Ring modulators

This is an unusual effect which is good for producing experimental and weird sounds. It’s most famous usage is as the voice of the Daleks in Doctor who. It has to be used with care though as the results can often be atonal. A nng modulator works by processing two input signals and combining the sum and difference of their frequencies at the output. If you input two sine waves at 900Hz and 600Hz the output would be made up from two tones at 300Hz and 1500Hz for example. This doesn’t sound very interesting but if the input signals are harmonically rich then all those frequencies will contribute to the sum and difference process and very interesting results ensue. Ring modulators can be great for adding something different to synth patches, for creating special effects voices and they can be interesting too on percussion and cymbals.

Introduction into Delay Effects

delay effect

Delay or echo is one of the oldest artificial effects. A wide variety of musical styles use delay in many different ways. At it simplest it is one or more repeats of a sound played back later in time. We are all familiar with echo in the natural world. It occurs in large spaces enclosed by hard surfaces such as canyons or cathedrals. Perhaps for this reason it has a nature association with the epic and ethereal. It can bring this and much more besides to a musical production and has become an integral part of some musical styles such as dub, trip hop and trance.

Setting a musically relevant delay time

In a musical context, delays are generally required to synchronize in some form to the tempo of the piece so that they enhance rather than clutter the mix. By adding delays one is effectively adding notes to the music but at a later point in time. So you need to take care to ensure that delays that continue over subsequent chord changes don’t create unwelcome clashing notes.

If the music has a fixed tempo it is easy enough to calculate the delay time. There are 60 000 milliseconds in a minute Sc you know the number of beats (ie quarter notes) per minute you divide 60 000 by the tempo to find the duration in milliseconds of one beat. If you use this as your delay time you will have a quarter note delay. You can divide this by 2 ft an 8th note delay time, by 4 for a 16th note delay time and so on.

Here are the figures for 120 bpm.

1/2 note delay – 1000
1/4 note delay – 500
1/4 note triplet delay – 333.33
1/8 note delay – 250
1/8 note triplet delay – 166.67
16th note delay – 125

You can use this Delay time calculator.

These days however most delay or echo plug-ins do the hard work for you. They are able to synchronize to the tempo of the session (set in the software) and allow you to se a delay time from a range of note values. Be aware that the longer the delay time you choose the more likely you are to create clashes with subsequent musical phrases.

Tape delay

Artificial delay first became possible with the advent of tape recording. As noted in the previous chapter, a . machine with a play head after the record head could be used to create a slap back delay which was used to emulate if early reflections present in reverberation. Feedback or “spin” was created by sending the return from the play head bad the record head mixed with the unaffected sound.

space echo re201This was later refined with units such as the classic Roland RE-201 which first appeared in 1973 and worked by means of a constantly circulating tape loop. A sound would be recorded onto the tape played back from several spaced play heads and then erased when the tape came round again. This unit included a spring reverb and had controls for mode, repeat rate, intensity and wet/dry mix amongst others. Adjusting the repeat rate controlled the delay time, the mode controlled which play heads were active and the intensity controlled the feedback i.e. the number of repeats and the amount of self-regeneration. As well as being used as a basic delay this unit could create some pretty strange effects, it had a warm sound and could be rather unpredictable which, was also part of its charm. It can be heard on numerous classic recordings from the 70s, 80s and 90s from Bowie to Portishead via dub gae, The RE-201 is still much sought after even today. Universal Audio now do a software emulation of the RE-201 but ething like Logic’s tape delay which is more readily accessible to many of us can get close to recreating many of its nds. The Pro Tools user can choose from some sophisticated plug-ins like the Line 6 Echofarm and the amazing hoboy by Soundtoys both of which emulate all the classic tape delays.

tape delayTape delays generally tended to have a very poor frequency response which in fact helped to give them their warm sound. is side effect also helps to distinguish the delayed sound musically from the source sound and to some degree reflects at happens in nature. Real echoes are naturally duller and quieter than their source sounds since the sound waves lose ergy as they travel through the air and bounce off reflective surfaces. Psycho-acoustically speaking, the brain is more ely to recognize a delay as such (rather than as a new note in the wrong place) if it has some high frequency roll-off. So plying this principle during mixing can often help to ensure that delays don’t make the mix sound cluttered. Tape delays re also infamous for their dodgy transport systems but again this was part of their charm. Provided they weren’t too onounced slight variations in the playback speed would throw up interesting flutter irregularities in the sound. Logic’s tape lay has a flutter section to emulate this aspect of tape delays and there is also a high cut filter to remove those offending pend frequencies. As you can see the user can sync the delay time to the session tempo and choose from a selection of te values. You can modify these to create dotted or triplet note values by moving the groove slider fully to one end or the er. If you move the feedback control higher than 50% the sound will start to self regenerate. Such controls are fairly typical of most delay type plug-ins.

Ping-Pong and Stereo Delays

stereo delay

A stereo delay is, in essence, two mono delays in parallel and panned hard left and righ In the case of a mono send to a stereo delay the input signal would be the same on both sides. Each side can send feedback to the other side as shown in the diagram below. This allows for the creation of some interesting effects since can set musically relevant but different delay times for the left and right channels and of course differing amounts of crossover feedback.

In a ping-pong delay the signal would arrive only at one input. Having passed through the delay it would then be fed acro to the other side and so on. If the delay times are the same this creates an even bouncing effect from side to side. If the delay times are different then a myriad of interesting configurations are possible. Logic Pro’s Stereo Delay plug-in has a fairly typical set of parameters with separate wet/dry controls for the left and right channels, a high and low cut filter sectii the usual delay time selection options, crossover feedback for each side and internal feedback for each side.


Multi-tap delay

The term multi-tap originates from vintage tape delays like the RE-201 which had more than one playback ad. A delay was considered to be “tapped off’ from the signal at each playback head. This type of delay creates a huge rnber of possibilities for manipulating the sound. These fall into two categories:

  1. If each tap output is panned at even spacings across the stereo image and is set to a sequential and musically relevant delay time (1/2 notes, 3/8th notes, 114 notes 3/16th notes, 8th notes etc) interesting musical delay configurations can be created.
  2. More commonly though multi-tap delays are used with short and slightly random delay times (say 50ms-350ms) with the taps panned at alternating positions across the stereo image and gradually getting further from the centre. The ear then doesn’t necessarily distinguish the taps as separate echoes and so you get a spreading, thickening and reinforcing type effect. If a little feedback is sent from each tap the repeats quickly build up in complexity creating a coarse reverb-type sound. This can sound really effective on solo electric guitar passages for example. Below is an example of this type of configuration with DSP FXs multi-tap delay plug-in.

Multi-tap delays are also often used as the basis for voice-doubling type effects. An 8 tap delay such as the one above Id work very well with extremely short delay times and panned voices to create thickened and widened vocal sound.

multi tap delay

Spot delays

Since delay effects can easily make the mix sound cluttered and busy they make a good candidate for spot For example if there is a point where the music pauses it can sound great to have a delay on the last word of a vocal at tails off into the space. This can be achieved in a number of ways. One is to make a copy of the phrase or word to be eyed on a new channel and set up the delay as an insert on that channel, setting the mix to 100% wet. Another is to put bypassed version of the delay insert on the original channel (this time with a combination of wet and dry in the mix) and le the automation to un-mute it when you want it to activate. Another is to set up the delay as a send and automate the nd so that the delay only receives an input at the desired point in time.

Reverb in the mix

How you use reverbs in your mix to some extent will depend on the musical style, your mixing philosophy and your outboard and DSP resources. Commonly, where resources are relatively limited, perhaps two different reverbs might be used – one mainly with the drum kit in mind and the other for the vocal. If realism is a factor then you might use only one since all . members of a band would generally perform in the same acoustic space. In practice, if you have the processing power, a variety of reverb choices can work well on different areas of your mix.

Drums and percussion

Generally drums and percussion are close miced these days so you will need to add reverb of some kind. The simplest approach is to pick a single reverb and use it on the whole kit in differing amounts. This works best if you use a short reverb setting and leave the kick dry or almost dry. If you have recorded room mics the quality of the sound you’re getting back from these may affect your choice of reverb setting. If you are applying different reverbs to the separate parts of the kit the Mowing guidelines may help. To avoid creating a mushy low end, do not add any obvious reverb to a kick drum. Use a short ambient setting if it is necessary to add anything at all. Snare drums generally benefit from more reverb. Plate settings can be good as they’re bright and don’t produce such distinct early reflections as real spaces. Reverb times for snares can be anything from less than a second to more than 3 seconds but it makes sense to avoid using a long decay time unless the music contains enough space for it to be heard. Generally speaking, the faster or denser the music the shorter the reverb tine should be. This will avoid cluttering the mix. Short bright plates or tiled room ambiences can give breathe life into a dull snare or if you’re looking for a bigger sound try a hall patch and adjust the pre-delay to give a slap back feel. Toms tend to have a long natural sustain so they don’t necessarily benefit from much reverb but a short ambience setting can give them that elusive sense of place. A little ambience is also good for high hats just to give a sense of three dimensional space and lo add some extra high frequency detail. Percussion generally sounds best with just a short ambience-type setting as these are quite dense and tend to reinforce and fatten the sound.


The lead vocal reverb is perhaps one of the most important settings in a mix. A vocal drenched in reverb will sound awful and a completely dry one unnatural and disassociated from the music. A very wet vocal sound can reduce the intelligibility of the lyrics and long decay times can fill up space in the mix unnecessarily. Generally speaking in pop and rock mixes we want the vocal to be very “up front”. Adding a lot of reverb will create the impression of distance — the opposite of what we are looking for. Short reverb and even ambience settings can be pretty successful especially if you add a little pre-delay of between 50 and 80ms to separate the vocal and the reverb. If you are looking for a reverb tail to add a little extra sheen to the vocal try a small hall or chamber but increase the early reflections balance so that the reverb tail doesn’t dominate.

Backing vocals are often meant to sit a little behind the lead vocal so longer reverb times are not necessarily a problem here. If you want to thicken the sound a little a setting with obvious early reflections can help.


Distorted guitars playing heavy chords don’t generally benefit from much reverb. This may depend on how they’re recorded. Sometimes, with amped guitars recorded in a live room, engineers also take a room mic recording, for ambience. However, solo and clean electric sounds and acoustics are a different matter. This very much depends on the style of music. Spring reverbs as we have seen can be used to recreate that classic ‘amp sound and large bright halls can work if you’re after that big sustaining solo guitar sound. Ambience settings are again good for acoustics – adding space and brightness without cluttering the sound with a long reverb tail. Short plates can also be good for this.

[box type=”info”] Less is more: In today’s musical climate very obvious reverb is not always required. Often what people want is a sound that has life and dimensionality without an obvious reverb tail — in effect almost a ‘reverb without the reverb’. This can apply to both band music and dance music, especially the latter where beats are often very dry and clipped. This is why ambience settings can be useful as they create the early reflections of a natural space but without the long decay times. Thus they reinforce and add solidarity to the sound without obviously smothering it. The sense of space is achieved and the sounds sit more comfortably in the mix, whilst a clean and clear sound is maintained.[/box]



  1. Reverb is a natural phenomenon and part of the way we habitually perceive sound.
  2. AU enclosed spaces create reverb as sound waves are multiplied by reflection, and diffraction.
  3. The frequency content of reflected sound waves is modified by the size, shape and absorptive characteristics of the room and its contents.
  4. Adding reverb is an essential part of the mixing process. It will breathe life into your mix but don’t over do it and choose appropriate settings.
  5. Early attempts to recreate reverb included echo chambers, tape delays, plate reverbs, and spring reverbs.
  6. These were superseded by the advent of digital reverbs in the 80s. Such digital reverbs are now highly sophisticated and afford the user a huge amount of control over the reverberant sound.
  7. Convolution or “sampling” reverbs are becoming an increasingly realistic prospect for users of DAWs in home and project studios. These effectively place the dry sound into a “real” space through the application of Impulse Responses recorded in those “real” spaces. They have fewer controllable parameters than conventional digital reverbs but sound highly convincing.
  8. As a general rule of thumb, with reverb, less is often more. If you’re worried about making your mix too wet choose presets with short decays and/or add just a small amount.

Cover photo by Saigo Caltroine

Why do we need dynamics control?

Limiters, compressors, expanders and gates are all devices which control the level of audio signals and therefore the dynamic range of a track or piece of music. They affect the loudness of a sound just as a fader or volume control does. In a dynamics device, however, the device according to parameters set up by the engineer controls the level automatically. In a traditional analogue device this would have been an electrical circuit element that would first sense the level of the signal and then adjust the output as required. Analogue compressors are still regularly used at the input stage, during recording, but in digital systems most dynamics control is now achieved through plug-ins which emulate the analogue world in terms of both performance and parameters. Below is a diagram to show how a traditional analogue dynamics unit would work.

dynamic control

As we noted earlier on, the useful dynamic range of the human ear is from OdB SPL (the threshold of hearing) to 115dB SPL (the threshold of pain). Digital sound equipment is now able to cover this dynamic range with 24bit recording, which has a range of 144dBs (theoretically anyway as the real limit of even the best Digital to Analogue converter is 122dB SPL of dynamic range). CD audio is a 16-bit system and has a dynamic range of 96dBs, which is more than adequate for most music and speech recording. Classical music is most renowned for having very quiet and very loud passages — a wide dynamic range. Even if reproduced in a Hi-Fi environment this would equate to about 65dBs difference between the loudest and quietest parts. Most pop and rock formats area designed to have a much smaller dynamic range since they need to be intelligible in a wide variety of situations – such as cars, factories, kitchens, railway stations etc – where ambient noise might be considerable. Under such conditions the signal must be compressed so that the quietest level is audible above the background noise without the loudness of reproduction being unacceptable for the given situation. Background music, for example, must be reproduced quietly yet should at all times be audible enough to be intelligible. The key to a successful commercial mix lies in an ability to engineer loudness and impact into a mix that is heard on a transistor radio. The limiter/compressor is a tool that makes this possible.



Common applications for automatic gain control

Protecting a system from being overloaded
In order to optimise the signal-to-noise ratio and maintain a high average record level without the fear of accidental overload, a limiter can be inserted to operate just prior to the onset of significant distortion

Increasing loudness
Changing the sound by making it either “denser (by reducing the dynamic range) or lighter (by increasing it).

Noise reducing
Reducing background noise, “spill”, cross-talk and improving the signal-to-noise ratio of analogue tape recorders and other noisy signal paths by the use of elaborate systems such as the Dolby ones.

Reducing sibilance
Extreme “s” and “t” in vocals and other similar problems with the use of de-essers.

Special effects
Voice-over and ducking (as used by disc-jockeys), dynamic equalisers such as the Opti-Mod for increasing loudness in radio broadcasts and a myriad of other inventive applications that we shall describe later.

Cover photo by Pedronchi

Insert or send effect


Effects have been used in recorded, and for that matter live, music pretty much from the word “go”. Classical choral music, for example, makes use of the acoustic properties of churches, as an effect, to enhance the music as does orchestral music in a concert hall auditorium. In the 50s and 60s guitar amps began to feature such effects as tremolo and spring reverbs and it was not long before engineers began experimenting with tape to create many of the effects we take for granted today such as delays, chorus, flanging, and phasing. Today musicians, engineers and producers have a stunning array of effects available to them at the touch of a mouse button. Some programs even make it possible to call up combinations of effects as channel strip presets. The possibilities are virtually endless. However, as with most things in music, whilst random experimentation can often throw up some interesting results there’s no substitute for having at least a basic understanding of the tools you have at your disposal — especially if you want to be able to fine tune and manipulate them to get the results you’re looking for. This can be a daunting prospect if you are new to mixing and particularly if you have no experience of using mixing desks and effects units outside of a computer environment. In this chapter we will investigate the origins, main controls and usages for some of the most widely used effects.

Insert or send effect

An insert effect is inserted directly onto a track or channel strip. Only the audio on that channel is processed. Generally the whole of the sound is processed unless the effect has a wet/dry mix control. The signal is then output to the next insert or it continues down the channel path to the output via its pan pot and fader, Most programs allow the user to change the order of inserted plug-ins as sometimes becomes necessary when adding more processing.

A send/return effect is placed on an aux return or effects bus. Multiple signals can be sent in varying amounts via send pots on individual channel strips. These signals are combined on the send or bus path routed to the aux return or effects bus, they are processed collectively by the effect and then pass on to the output of the aux or bus channel via its pan pot(s) and fader.

When setting up any kind of effect the first question that presents itself is:– “Should I set this up as a send effect or use an Insert point on the channel strip?” There is not necessarily a correct answer. It largely depends on the type of effect, what its role is to be and how much of your processing resources it will take up. Here are some guidelines:

  • If the whole of the sound you are working on is to be processed by the effect then set it up as an insert.
  • If you need a mixture of the effected and unaffected sounds then a send is more appropriate. Although most plug-ins include a wet/dry control so you can still do this on an insert point. It’s worth noting that if you set up an effect (such as a reverb or delay) on a send, you will generally want to set the mix to 100% wet since you will be able to control the wet/dry mix by the amount you send from the channel (don’t forget to make sure the send is post fade).
  • If the effect is to be applied to only one sound then use an insert point.
  • If the effect needs to be accessed by more than one sound then you should definitely set it up as a send. This most obviously applies to reverbs and delays which are time based effects. If you had enough processing resources you could use multiple instances of the same plug-in, with the same setting on different channels but it is extremely wasteful. Using a send is a good way of conserving DSP resources since one effect can be applied to many sounds. Leaving more room for experimentation with other plug-ins later if necessary.
  • If the effect is stereo and the original signal is mono, it may make sense to use a send in order to retain control over the panning and placement of both the effected and dry signal

Expermienting with effects is one of the most fun and rewarding stages of the mix process. With the plenty of possibilities open to engineers today at the touch of a mouse it can be very easy and tempting to keep adding plug-ins until your DSP resources are used up. However, too many effects can make a song busy and over produced so it’s good to keep a bit of perspective. Try to use effects where they seem musically and sonically appropriate and where they enhance rather then detract from the message or mood of the piece.

Reverbnation in the real world

reverb effect

Reverb, of one type or another, is probably the most widely used effect in modern mixing. In today’s recording environments music is usually made from a combination of artificially produced sounds and close-miced recordings. The recordings may have been made at different times and in different places. This approach gives us control over practically all aspects of our mix but if we want it to have any kind of coherence we will need to use some reverb to help to recombine the sounds and bind them into a performance with the impression of an ambient space. Using the correct reverb can make the disparate multi- tracked parts sound whole again as well as bringing colour and three-dimensionality to the sound. Conversely, using too much reverb or the wrong reverb can cause the mix to sound cluttered, messy and distant.

When we make a noise in an enclosed space the sound radiates outwards. It hits obstacles such as walls, tables and chairs and one of the following will occur.

  • Reflection: If sound encounters a hard surface it will generally reflect back into the room.
  • Absorption: Softer surfaces will tend to absorb sound although precisely how much absorption occurs and which frequencies are affected depends largely on the nature of the materials.
  • Diffraction: Sound will bend round or be shadowed by objects. This will spread the sound out in many new directions.

Generally there will be a distinct initial reflection after which further reflections build up quickly as the sound gets rebounded around the room and becomes more diffuse. The time between the sound and its initial reflection depends on the size of the room. On a reverb unit it is usually known as pre-delay. The complex mass of repeats and echoes that follow the initial reflection are what we recognise as reverb. Each line on the diagram below represents the amplitude of a reflection of a discrete sound. There are a number of early reflections followed by a more dense mass of intermingling later reflections. These diminish in amplitude over time.

reverb in the real world

The exact point when reverb ends, is difficult to define, so a measurement known as RT60 is used. This is the time it takes for the reverb to decay by 60dBs from its original value. This is known as the reverberation time. Reverb can radically change the nature of a sound. The sound of a hand clap in a living room, a church or a bathroom, for example, would all be very different. Reverb can be bright or dull, have a long or short decay time, it can be dense or sparse, it may or may not have a distinct early reflection.

These characteristics, and so the resultant sound, are determined by such factors as the size and shape of the enclosed space, the absorptive qualities of the materials that form the room boundaries and the quantity, size, shape and reflectivity of the objects within the room.

To illustrate the last point, if you’ve ever been to a band sound-check and then to the subsequent gig you will have noticed that the room sounds much more reverberant and generally brighter during the sound check than the gig. When the room is full of people high frequencies are much more readily damped and a large part of the reflected sound is absorbed by the mass of bodies making up the audience (assuming they have turned up!) giving a tighter sound.

Mix Delay Into Reverb

This is a great tip for mixing delay using reverb plug-in if you are looking for a different sound, depth and space in the vocal. This tip isn’t for all situations, but sometimes a mix calls for a specific aesthetic that can only be accomplished with some vocal delay run into a reverb. This combination is really cool and worth trying out.

Mono Vocal Delay

A tip called “Mono vocal delay” is used when sometimes a stereo delay only gets lost in a dense mix. Creating a mono aux or mono bus and putting the delay will get in the repeats of a delay but at one fixed location. The significance of a mono vocal delay is that it’s fixed. It can echo and repeat for eternity, but only in one specific location in the stereo spectrum.