Snoman R. Dance Music Manual: Tools, Toys, and Techniques

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down by 2 –5 cents it can produce a wide, thick up-front sound. Additionally, if

you move the second vocal track forward by a few ticks so that it occurs slightly

later than the original, this can help to thicken the overall sound.

Many engineers will also recommend applying reverb to a vocal track to help

thicken them in a mix but if it is not applied cautiously and sparingly it will

push them into the background producing a less deﬁ ned, muddied sound.

Nevertheless, if you do feel the need to apply reverb to vocals, use it as a send

effect rather than insert as this will allow you to keep a majority of the vocal

track dry. Also, if reverb is being used on any other instruments within the mix,

use a reverb with a different tonal character for the vocals as this will help to

draw a listener’s attention to them.

The trick behind using any reverb is to set the effect so that it’s only notice-

able when it’s removed, but keep the decay time short for a fast tempo sing

and lengthen it for those of a slower tempo. As a very general starting point,

try setting the reverb’s pre-delay to 1 ms, with a square room and 7 ms decay.

By then increasing the pre-delay you can separate the effect from the dry vocal

to enable the voice to stand out more.

The most widespread processing used to obtain the up-front sound is compres-

sion. If you really want that big, in your face, up-front vocal, then compres-

sion is the real key to acquiring it, but for this it is imperative that you use

compressors with the best sound possible. As every dance genre will use valve

compressors, this means you will need to use them too and whether hardware

or software emulations they should ideally exhibit 0.2% total harmonic distor-

tion (THD) or more for a good sound. Notably, most musicians use more than

one compressor for this and will usually employ two or three. The ﬁ rst is com-

monly a solid-state compressor that’s used to smooth out the overall level with

a ratio of 4:1, a fast attack and release and threshold set so that the gain reduc-

tion meter reads 4 dB. The results of this are then fed into a valve compressor

and this is generally set with a ratio of 7:1, a fast attack and release along with

a threshold set so that the gain reduction meter reads 8 dB. Naturally, these

are general settings and should be adjusted depending on the vocals and the

microphone used to capture them. As we’ve touched upon in earlier chapters,

applying compression will reduce the higher frequency as it squashes down on

the transients of the attack, but if you ﬁ nd that the high-end detail becomes

lost, it can often be brought back up again using a sonic exciter or enhancer.

‘ TELEPHONE ’ VOCALS

The second effect that’s often used in the dance genre is the ‘telephone’ vocal,

called so because it sounds as though it’s being played through a restricted

bandwidth (i.e. a telephone or transistor radio). This effect is quite easy to

achieve by inserting (not sending) the vocals into a band-pass ﬁ lter so that

frequencies above 5 kHz and below 1 kHz are removed from the signal. If you

don’t have access to a band-pass ﬁ lter, the same effect can be acquired by ﬁ rst

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inserting the vocals into a low-pass ﬁ lter set to remove all frequencies above

5 kHz with the results of this inserted into a high-pass ﬁ lter set to remove

all frequencies below 1 kHz. If you can adjust the poles on these ﬁ lters then,

generally speaking, a 2-pole (12 dB) slope will produce the best results but

depending on the mix that the vocals are sitting in, it may be worth using a 4-

pole (24 dB) or, if possible, a single pole (6 dB) slope. Alternatively an EQ unit

can be used to achieve much the same results by cutting all frequencies below

1 kHz and above 5 kHz. Once you have this general effect, it can be expanded

upon by automating the ﬁ lters (or EQ) to gradually sweep through the range to

add movement and interest.

PITCHED VOCALS

Another, more recent, effect to apply to vocals is the unnatural pitch-shift-

ing that’s acquired by using an intonation unit to alter the pitch. How this is

accomplished depends entirely on the effects unit itself, but fundamentally

they all work on the principle that you can analyse the incoming signal and

then select the key signature that it should be in. By using this as an insert

effect so that it affects the entire vocal line and setting it to a key that’s at least

an octave out of the range of the original input, the processor will attempt to

correct the vocals to the selected scale and, as a side-effect, produce a strange

moving pitch very similar to the effect used on Cher’s ‘Believe ’. On top of this,

it is also worth experimenting by running the vocals through the intonation

unit a number of times, each time using a different key setting before ‘comp-

ing’ the different results together.

VOCODERS

One ﬁ nal effect that’s particularly useful if the vocalist is incapable of singing in

key is the vocoder. Of all the vocal effects, these are not only the most instantly

recognizable but also the most susceptible to changes in fashion. The robotic

voices and talking synth effects they generate can be incredibly cliché unless

they’re used both carefully and creatively, but the way in which they operate

opens up a whole host of creative opportunities.

Fundamentally vocoders are simple in design and allow you to use one sound –

usually your voice (known as the modulator) – to control the tonal character-

istics of a second sound (known as the carrier), which is usually a synthesizer’s

sustained timbre. However, as simple as this may initially appear actually pro-

ducing musically useable results is a little more difﬁ cult since simply dialling

up a synth preset and talking, or singing, over it will more often than not pro-

duce unusable results. Indeed, to use a vocoder in a musically useful way, it’s

important to have a good understanding of exactly how they work and to do

this we need to begin by examining human speech.

A vocoder works on the principle that we can divide the human voice into

a number of distinct frequency bands. For instance, plosive sounds such as ‘P’

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or ‘B’ consist mostly of low frequencies, ‘S’ or ‘T’ sounds consist mostly of

high frequencies, vowels consist mostly of mid-range frequencies and so forth.

When a vocal signal enters the vocoder, a spectral analyser measures the sig-

nal’s properties and subsequently uses a number of ﬁ lters to divide the signal

into a number of different frequency bands. Once divided, each frequency

band is sent to an envelope follower which produces a series of control volt-

ages

based on the frequency content and volume of the vocal part. This exactly

same principle is also used on the carrier signals and these are tuned to the

same frequency bands as the modulator’s input. However, rather than generate

a series of control voltages, they are connected to a series of voltage-controlled

ampliﬁ ers. Thus, as you speak into the microphone the subsequent frequencies

and volume act upon the carrier’s voltage-controlled ampliﬁ ers which either

attenuate or amplify the carrier signal, in effect, superimposing your voice onto

the instrument’s timbre. Consequently, since the vocoder analyses the spectral

content and not the pitch of the modulator it isn’t necessary to sing in tune as

it wouldn’t make any difference.

From this, we can also determine that the more the ﬁ lters that are contained

in the vocoders bank, the more accurately it will be able to analyse and divide

the modulating signal, and if this happens to be a voice, it will be much more

comprehensible. Typically, a vocoder should have a minimum of six frequency

bands to make speech understandable but it’s important to note that the num-

ber of bands available isn’t the only factor when using a vocoder on vocals.

The intelligibility of natural speech is centred between 2.5 and 5 kHz, higher

or lower than this and we ﬁ nd it difﬁ cult to determine what’s being said. This

means that when using a vocoder, the carrier signal must be rich in harmonics

around these frequencies since if it’s any higher or lower then some frequencies

of speech may be missed altogether. To prevent this, it’s prudent to use a couple

of shelving ﬁ lters to remove all frequencies below 2 kHz and above 5 kHz before

feeding them into the vocoder. Similarly, for best results the carrier signal’s sus-

tain portion should remain fairly constant to help maintain some intelligibility.

For instance, if the sustain portion is subject to an LFO modulating the pitch or

ﬁ lter the frequency content will be subject to a cyclic change which may push it

in and out of the boundaries of speech resulting in some words being compre-

hensible while others become unintelligible. Plus, it should also go without say-

ing that if you plan on using your voice to act as a modulator it’s essential that

what you have to say, or sing, is intelligible in the ﬁ rst place. This means you

should ensure that all the words are pronounced coherently and clearly.

More importantly, vocal tracks will unquestionably change in amplitude through-

out the phrases which will create huge differences in the control voltages gen-

erated by the vocoder. This results in the VCA levels that are imposed onto the

Control voltages (often shortened to CV) measure the signal at predetermined points

and convert each of these measurements into differing voltages to represent the volume and

frequency of the waveform.

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carrier signal to follow this change in level, producing an uneven vocoded effect

which can distort the results. Subsequently, it’s an idea to compress the vocals

before they enter the vocoder and if the carrier wave uses an LFO to modulate the

volume, compress this too. The settings to use will depend entirely on the vocals

themselves and the impact you want them to have in the mix (bear in mind that

dynamics can affect the emotional impact), but as a very general starting point

set the ratio on both carrier and modulator to 3:1 with a fast attack and release,

and then reduce the threshold so that the quietest parts only just register on the

gain reduction meter. Additionally, remember that it isn’t just vocals that will trig-

ger the vocoder and breath noises, rumble from the microphone stand and any

extraneous background noises will also trigger it. Thus, along with a compressor

you should also consider employing a noise gate to remove the possibility of any

superﬂ uous noises being introduced.

With both carrier and modulator under control there’s a much better chance of

producing a musically useful effect, and the ﬁ rst stop for any vocoder is to rec-

reate the robotic voice. To produce this effect, the vocoder needs to be used as

an insert effect, not send, as all of the vocal line should go through the vocoder.

Once this modulator is entering the vocoder you’ll need to programme a suit-

able carrier wave. Obviously, it’s the tone of this carrier wave that will produce

the overall effect and two sawtooth waves detuned from each other by ⫹ and

⫺4 with a short attack, decay and release but a very long sustain should pro-

vide the required timbre. If, however, this makes the vocals appear a little too

bright, sharp, thin or ‘edgy’ it may be worthwhile replacing one of the sawtooth

waves with a square or sine wave to add some bottom-end weight.

Though this effect is undoubtedly great fun for the ﬁ rst couple of minutes, after

the typical Luke, I am you’re father it can wear thin, and if used as it is in a dance

track, it will probably sound a little too clich é, so it’s worthwhile experiment-

ing further. Unsurprisingly, much of the experimentation with a vocoder comes

from modulating the carrier wave in one way or another and the simplest place

to start is by adjusting the pitch in time with the vocals. This can be accom-

plished easily in any audio/MIDI sequencer by importing the vocal track and

programming a series of MIDI notes to play out to the carrier synth, in effect

creating a vocal melody. Similarly, an arpeggio sequence used as a carrier wave

can create a strange gated, pitch-shifting effect while an LFO modulating the

pitch can create an unusual cyclic pitch-shifted vocal effect. Filter cut-off and

resonance can also impart an interesting effect on vocals and in many sequenc-

ers, this can be automated so that it slowly opens during the verses, creating

a build-up to a chorus section. Also, note that the carrier does not necessar-

ily have to be created with saw waves, and a sine wave played around C3 or

C4 can be used to recreate a more tonally natural vocal melody that will have

some peculiarity surrounding it.

Vocoders do not always have to be used on vocals and you can produce great

results by using them to impose one instrument onto another. For instance

using a drum loop as a modulator and a pad as the carrier the pad will create

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a gating effect between the kicks of the loop. Alternatively using the pad as a

modulator and the drums as the carrier wave the drum loops will turn into a

loop created by a pad!

Ultimately these have only been simple suggestions to point you in a more cre-

ative direction and you should be willing to try out any effects you can lay your

hands on to hear the effect it can have on a vocal. Bear in mind that due to the

very nature of dance music it’s always open to experimentation and it’s much

better to initiate a new trend than simply follow one set by another artist.

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Although many of the timbres within dance music are created with synthesis

and many of the real-world instruments are often sampled from a sample CD –

or more commonly, another record – if you have access to the appropriate

instruments, microphones, direct injection (DI) boxes and pre-amps, recording

your own instruments can be incredibly rewarding. Not only will they be guar-

anteed to ﬁ t with the tempo of your music, they’re also pretty much guaranteed

to be in the right key too.

Recording real-world instruments is not something that should be approached

lightly, however, since to gain the best results you need to not only have the

appropriate environment for the instrument you wish to record but also good-

quality microphones, a good pre-amp and – depending on what you are

recording – a good ampliﬁ er and DI Box.

It must also be stressed that different engineers will use different techniques

to record any instrument, and the judgement on how to record the instrument

will depend on a large number of factors including the instrument’s character,

the room it’s recorded in, its tuning and, most important of all, the player’s

style. Therefore, what follows in this chapter should only be viewed as general

guidelines and you should adapt these to ensure you receive the best recording

possible.

RECORDING PREPARATION

The ﬁ rst priority to record any instrument should be the room’s acoustics. It is

not necessary to have a recording room with a totally ﬂ at frequency response

for a majority of real-world instruments; in fact, if you do record in a room

Recording Real Instruments

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’ In the late Sixties a record producer was a more

hands-on person. A record producer was also a skilled

music arranger or an engineer or both. ...’

Tony Visconte

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such as this, you may ﬁ nd that the instrument is lacking in character. This is

because with any real instrument the sound you hear is a result of a number of

factors. For example, the sound of an acoustic guitar is not a result of only the

sound hole but from the entire guitar vibrating sympathetically along with the

reﬂ ected frequencies within the room. The same applies for brass instruments;

the sound does not come directly from the bell but from the bell’s vibrations

and the reﬂ ected frequencies of the room it is played in. Because of this, it is

vital that you choose a room that has the best acoustics for the instrument you

wish to record. While you can add reverb at mix-down for example, you cannot

remove it if it was captured at the recording stage!

From a general point of view, a typically decorated and carpeted room will

prove suitable for recording most real-world instruments. Garages, basements

and rooms with wooden ﬂ oors should normally be avoided since these create

brighter reﬂ ections which can result in a recorded sound that lacks any bottom

end. However, that is not to say you shouldn’t try playing the instrument in

these rooms and listening to the results. In some instances, they may produce

the sound you’re after, particularly if you’re recording brass instruments.

The best way to know if a room is suitable for recording is to actually play the

instrument in that room. The principle to work with is that if it sounds good

to the performer, then it stands to reason that it will sound good to the micro-

phone, provided of course, that you use a good-quality microphone.

For many instruments, either ribbon or large diaphragm capacitor microphones

are regarded as the best suited for the job. As touched upon in the previous

chapter, larger diaphragm microphones have a heavier mass which produces a

rounder, warmer timbre that is often suited for many instruments.

Ribbon microphones, although not as popular or widely used as large dia-

phragm microphones, are often used since these have a bidirectional polar

pattern in that they capture sound from both sides of the microphone. By

doing so, they cannot only capture the direct sound but also the ambience

of the room. Naturally, this can be accomplished by increasing the distance

from the instrument with a cardioids microphone, but for some engineers, the

sound of a ribbon microphone produces the best results.

Ribbon microphones work similar to a dynamic microphone; however, whereas

the diaphragm in a dynamic microphone is coupled to a coil of wire that’s sus-

pended in a magnetic ﬁ eld, with a ribbon microphone there is an extremely

thin ‘ribbon’ of aluminium suspended at both ends which is vibrated by sound

pressure within a magnetic ﬁ eld. This means that the ribbon can vibrate from

both sides but not from its edges, allowing it to record from both sides of the

microphone and producing what’s known as a ﬁ gure 8 pick-up pattern.

For this reason, ribbon microphones are considered the most natural sounding

microphones and perfect for capturing instruments, but they are also extremely

delicate. Blowing into a ribbon microphone can damage the thin aluminium

ribbon and even carelessly slamming down the lid of the microphone box

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can create a surge of air that will damage the microphone’s ribbon.

Unfortunately, due to this delicate nature, ribbon microphones are

few and far between and those that are in production are expensive

( Figure 8.1 ).

It should also be noted that due to the low output of a typical rib-

bon microphone, you need a powerful and quiet pre-amp to bring

the gain up to an acceptable level (60 dB of ampliﬁ cation to bring it

up to 0 VU!). Therefore, if you plan on using a ribbon microphone,

you’ll need to ensure that you have a powerful and high quality pre-

amp that has a very low noise ﬂ oor.

As previously touched upon, when recording instruments, each

engineer will approach the project differently and this includes

the choice of microphones for the job at hand. Consequently, rec-

ommending microphones can be a mineﬁ eld since whereas one

engineer may choose one particular microphone for a particular

instrument, a different engineer will have a totally different opin-

ion. On top of this, the room’s frequencies and the player’s personal

style will also affect the choice.

Nevertheless, at the risk of leaving myself open, I would personally

recommend the Neumann KM184 capacitor microphone for some

instruments since it produces particularly pleasing results and rolls

off at 200 Hz keeping the instruments free of any low end rumble.

Alternatively, the B&K 4011, the Shure SM81 and the AKG C1000s

can all produce excellent results if set up correctly.

If I were to recommend ribbon microphones (and if you have a large wallet),

Audio Engineering Associates produce some outstanding microphones such as

the R44 or alternatively Golden Age Companies R1 Active and Coles 4038. It is

doubtful that you will ﬁ nd a ribbon microphone for sale in the second-hand

market since they’re often snapped up by studios the moment they appear.

RECORDING ACOUSTIC GUITARS

Before recording an acoustic guitar, it’s often preferable to tune the guitar to

the Nashville System. This consists of using the octave strings from a 12-string

guitar, ﬁ tting them and tuning E and B to their regular pitch but tuning G, D,

A and (the lower) E an octave higher – as they would be on a 12-string guitar.

By doing this, the guitar produces an incredibly bright and ‘jingly’ timbre, rep-

licating the typical pop music guitar sound. This also reduces a proportionate

amount of low-end boom which helps the instrument ﬁ t perfectly into a busy

dance mix.

Ideally, acoustic guitars should be recorded in stereo. While you can record

in mono and apply stereo effects afterwards, although the results are not as

impressive as recording in stereo. Obviously, this means that to record in ste-

reo requires two microphones and you should use a paired set of microphones.

FIGURE 8.1

An early ribbon

microphone

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In other words, they are paired to one another directly from the manufacturer and

sold as a pair. This ensures that they both have the same response which makes

it much easier at mix-down. If this isn’t possible, you should try to use two of the

same make and model of microphone to ensure you receive a consistent record-

ing from the guitar. Of course, you should feel free to use different microphones

if that is all you have; experimentation may provide the best results for you.

The microphones should be placed with the ﬁ rst pointing towards the bridge of

the guitar and the second pointing towards where the fret meets the body of the

guitar. The distance of the microphones from one another and from the guitar

must be carefully considered, however, in order to reduce any phase cancellation.

Phase cancellation is a result of the sound waves from the guitar reaching the

two microphones at differing times. As discussed in Chapter 1, if the waveforms

are captured at different phases of its cycle by the microphones, when they are

combined at the mixing desk it can create phase cancellations between the

two waveforms. To avoid this, you should adopt the 3:1 rule. This means that

the spacing between each microphone should be three times the distance they

are from the guitar. For example, typically a microphone should be placed

approximately 300 mm away from an acoustic guitar; therefore, the micro-

phones should be placed at least 900 mm apart.

Of course, if you don’t have access to two microphones you’ll be left with no

option but to record the guitar with one microphone. One of the most com-

mon mistakes when recording with just one microphone is to place it directed

entirely at the sound hole. The problem with this approach is that rather than

capturing the entire guitar’s sound, it only captures the lower end of the fre-

quencies produced by the guitar.

A preferential approach is to place the microphone around 300 mm away from

the guitar aiming towards where the neck and body meet. This captures both

the sound emanating from the sound hole and the natural vibrations from the

guitar’s body.

This microphone position shouldn’t be considered de facto and you should

always experiment by moving the microphone further away since the room will

have an inﬂ uence on the sound that you may want to capture. It is important,

however, that you monitor through headphones so you only hear what the

microphone is picking up and not frequencies from the room that the micro-

phone may not be picking up.

Depending on the player’s individual style, you may record fret squeals as the

guitarist slides his ﬁ ngers up and down the fret. While some engineers will

attempt to avoid capturing any fret noise whatsoever, I ﬁ nd it removes the real-

ism of the guitar since fret squeal – in limited amounts – is a natural occur-

rence from playing the instrument and often enhances the sound of the guitar.

During recording, you should attempt to keep processing to a minimum since

it cannot be removed afterwards. This means compression should be kept to

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a minimum and only used to prevent any louder transients from clipping the

recording device. A good starting point for compression is to use a 5:1 ratio,

with a 5 ms attack, a 40 ms release and a hard knee setting. Once set up, lower

the threshold so that the compressor only captures any hard transients that

may clip recording. The idea is to set the compressors threshold to ensure that

the guitar is not compressed whatsoever, and it only compresses when abso-

lutely necessary. If it is not set up correctly, you may ﬁ nd that the compressor

pumps the recording which ruins the end result.

ELECTRIC GUITARS

Electric guitars require a slightly different approach to recording acoustic and

for many the best approach is to simply employ a DI box. A DI box is a device

with – most commonly – a jack connection that allows any electric instrument

with a high impedance (such as an electric guitar) to be connected directly into

it. The unit then converts this to a low-impedance signal (like a microphone ’s

output) which can then be connected directly into the microphone input on

a mixing desk. More recently, however, DI boxes have started to appear that

feature a USB connection that directly connects the DI box into a computer,

allowing the user to record the instrument directly without having to connect

the DI box output into a mixing desk. Many of these ‘computer DI boxes ’ also

feature amp simulations, allowing you to set a speciﬁ c amp type which is then

applied to the instrument before it is recorded into the computer. One of the

most celebrated of these is the Line 6 POD, a DI box used by most professional

engineers and very highly recommended ( Figure 8.2 ).

FIGURE 8.2

The Line 6 POD