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© Colin Hemphill, 2016 Edition 1.1 July 22, 2016

Live Sound Reinforcement Guide For Church Volunteers

Colin Hemphill

Module 01

© Colin Hemphill, 2016

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1 TABLE OF CONTENTS

1 TABLE OF CONTENTS 2

2 INTRODUCTION 4

2.1 WELCOME 4

2.2 ABOUT THE AUTHOR 4

3 FREQUENCY AND SOUND 4

3.1 FREQUENCY 4

3.2 TYPES OF AUDIO SIGNALS 6

3.3 DYNAMICS 7

3.4 SOUND PROPERTIES 8

4 LIVE SOUND SETUP 9

4.1 MICROPHONES 9

4.1.1 TRANSDUCERS 9

4.1.2 DESIGN AND FUNCTION 10

4.1.3 PICKUP PATTERNS 11

4.1.4 FREQUENCY RESPONSE 13

4.1.5 THE PROXIMITY EFFECT 14

4.2 PERIPHERALS 14

4.3 DI AND MICROPHONE TECHNIQUES 16

4.3.1 DIRECT BOXES 16

4.3.2 CLOSE MIKING TECHNIQUES 16

5 SIGNAL FLOW 18

5.1 THE SIGNAL PATH 18

5.2 GAIN STRUCTURE 18

6 SIGNAL PROCESSING 19

6.1 EQUALIZATION 19

6.1.1 FILTERING 19

6.1.2 SHELVING 20

6.1.3 PEAK 20

6.1.4 ADDITIVE AND SUBTRACTIVE METHODS 21

6.2 EQUALIZATION UNITS 21

6.2.1 GRAPHIC EQ 21

6.2.2 PARAMETRIC EQ 21

6.3 COMPRESSION AND LIMITING 22

6.4 GATING 25


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6.5 DE-ESSING 26

7 FINAL DETAILS AND FUTURE CHANGES 26

8 BIBLIOGRAPHY 26


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2 INTRODUCTION

2.1 WELCOME

Thank you for your interest in learning live audio, and possibly volunteering to work in audio tech! We are honored to have you as a part of the crew, and your work will be a huge blessing on this community of believers. This handbook is an introduction to Live Sound Reinforcement, which takes you through the arts of setting up sound and stage equipment, mixing a live band, and tearing it all down at the end of the services. Knowing what happens behind the sound board is important not only for the engineers, but also for the band members. Knowing how to work with the tech team to get what you need, and to provide them with what they need, is just as important as knowing your instrument and crafting your tone.

As a member of the worship team, the sound tech is as much a worship leader as the band members. You are responsible for making the sound of the performance balanced so that all elements are clear and audible. It is especially crucial that the lyrics are heard in a worship setting, and that your mix does not distract the congregation from a heart of worship. You are also responsible for making sure that sound from any video being played is heard, and that the pastor and other speakers on stage are heard. Otherwise, your goal should be to have fun, make the band sound its best, and take part in the worship yourself.

If you are new to live sound engineering, you can use this handbook by reading straight through at your own pace. It may also be of use to experienced individuals who are looking for ways to fix problem mixes or need some tips on processing settings. The appendices in the back feature some contact information to get in touch with me if you have any questions or note any changes or additions to the manual that would be helpful.

2.2 ABOUT THE AUTHOR

-time software developer in Austin, TX, but I also have around 12 years of professional and volunteer audio experience. I began in middle school, learning from other engineers and band members, and eventually interned for a production company to learn studio recording in high school. In 2013, I graduated from Belmont University in Nashville, TN with a double major in Audio Engineering and Computer Science. While in Nashville, I had the chance to work in world-class recording studios such as Ocean Way Nashville and RCA Studio B, and learn from the

Society) and Joe Baldridge (engineer/producer for DC Talk, Newsboys, Toby Mac, Family Force 5, Kelly Clarkson, Keith Urban, Chris Tomlin, Passion, and many others).

This handbook was created as part of a service learning project for my senior capstone course, and I hope that I can impart some of what I know to help church communities.

3 FREQUENCY AND SOUND

3.1 FREQUENCY

http://www.oceanwaynashville.com/

http://studiob.org/


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Sound is simply made up of vibrations of molecules in the air, but it is a fascinating process that we can capture and manipulate as engineers. Being a good sound engineer starts with a basic understanding of the medium we manipulate, and how the properties of this medium are used to create music. We will begin by defining the audio properties of frequency and dynamics.

Audio waves vibrate through the air at a particular rate, and the wave simply repeats as it travels. We define this rate of repetition as a wave's frequency. We interpret the audio frequency as pitch, where slowly vibrating waves cause low pitches and quicker waves create high frequency pitches. Frequency also affects the physical length of a sound wave, where the slow moving low frequencies have a long wavelength, and high frequencies are very short.

The property of frequency is measured in Hertz, usually abbreviated as Hz. Western music scales are built around the frequency 440 Hz - a note known as A440. This note is the 'A' directly above "Middle C" on the piano, and the frequencies of the rest of the piano are designed to interact appropriately with this frequency. Human hearing generally ranges from 20 Hz to 20,000 Hz, and all instruments sit somewhere in this range of frequencies. We generally abbreviate the upper limit as 20 kHz, where a kilohertz is shorthand for one thousand hertz.

Each sound in the real world is made up of two parts: the fundamental frequency and harmonics. The fundamental frequency is defined as the lowest frequency present in the sound. The fundamental is usually the loudest and is the one that we hear most clearly. Harmonics, also known as overtones, are higher frequencies on top of the fundamental. The combination and balance of these two parts determine what an instrument sounds like. We can tell the difference between a piano playing A440 and a violin playing A440 because of the harmonic content of the instruments. A violin is a much more complex sound, featuring many strong harmonics. Figure 2 displays this fact by comparing the same note on a sine wave and two instruments.

Figure 1. Comparison of a Sine Wave and Instruments of the Same Fundamental.


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We can see that a piano's soft harmonics only slightly alter the shape of the basic 440 Hz oscillation, whereas the many harmonics of the violin drastically alter the signal. Harmonics are built on top of the fundamental in simple mathematical terms. All overtones are multiples of the fundamental - so harmonics of A440 include 2x 440 = 880 Hz, 3x 440 = 1320 Hz, and so on. We can always determine the octave above a frequency by doubling its value. Thus, 880 Hz is an octave above A440, and 1760 Hz or 1.76 kHz is two octaves above A440.

Engineers often simplify the range of frequencies into five broad categories: low (bass), low-mids, mids, mid-highs, and highs (treble). We can roughly enumerate these ranges as:

Low: 20 Hz to 140 Hz (rumble, thump) Low-Mids: 140 Hz to 400 Hz (bass, mud) Mids: 400 Hz to 2 kHz Mid-Highs: 2 kHz to 5 kHz (human voice) Highs: 5 kHz to 20 kHz (present, bright, airy)

It is important to know that the human ear responds to each frequency differently. We are most sensitive to the 2 kHz to 5 kHz range because of the shape and size of our ear canals. Consequently, we hear the human voice very clearly because it sits in this frequency range. We are considerably less sensitive to low frequencies, especially at lower volumes.

3.2 TYPES OF AUDIO SIGNALS

We can roughly define a signal as any fluctuating representation of a physical property. Therefore, when we transmit audio through a wire as a fluctuating voltage, or record audio onto a computer as bits, we call these measurements audio signals. There are several types of basic audio signals, which are represented visually in Figure 2.

Figure 2. Basic Audio Signal Waveforms [1]


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The visual representation of an audio signal is called a waveform. These are images that we would see if recording audio signals onto a computer or viewing them on an oscilloscope. A sine wave is the most basic audio signal, composed of a smooth oscillation at a particular frequency. A square wave usually represents a digital set of bits, where the bit is either on or off, but has no in-between values. Triangle waves ramp up and down linearly rather than on a curve, and sawtooth waves ramp up, then fall immediately back to the starting point.

Each of these signals sounds unique, and they are all used in situations such as calibration and music synthesis. For example, a synthesizer keyboard will use combinations of these basic types of waves to synthesize sounds. A square wave is actually a sine wave with an infinite number of harmonics, and it sounds harsh and "digital" compared to a sine wave of the same frequency.

One other type of common audio signal is noise, the most relevant of which are pink noise and white noise. White noise is the static that you hear when the signal goes out on old TV sets, and it is created by producing random tones at all audible frequencies. All the tones are at an even power, so the high frequencies are as strong as the lows. Pink noise is similar, except that it is filtered to roughly match human hearing. Thus, the low frequencies are boosted to make them more present in the noise, making it sound more "even" to our ears.

3.3 DYNAMICS

The velocity at which an instrument is struck, plucked, bowed, or otherwise played determines the dynamics of the sound produced. We measure many quantities in decibels (abbreviated dB), but in acoustics we mostly deal with dB (SPL), which indicates the sound pressure level. The dB (SPL) unit is on a logarithmic scale, which in practice means that the difference between 40 dB and 50 dB is fairly insignificant, but the difference between 110 dB and 120 dB can be quite significant.

The unit is actually a ratio comparing the sound occurring to a preset level. This preset level of 0 dB is called the "threshold of hearing," which is roughly the quietest sound that humans can hear. The actual threshold can vary between about 1 dB and 35 dB, depending on age and other hearing conditions. Typical human conversation happens in the 60 dB to 70 dB range, while a garbage disposal or blender may be about 85 to 90 dB (SPL). Any point beyond 85 dB has the potential to cause permanent hearing damage, and is regulated by OSHA standards if you happen to work in noisy conditions. Granted, it may take as many as eight hours of exposure to this level to cause damage, but by 95 dB we really must pay attention to the levels we listen to. A typical rock concert might occur in the range of 100 dB to 115 dB, and we call 125 dB the "threshold of pain," which we certainly want to avoid.

Fun Fact: Believe it or not, there is such thing as a loudest sound possible. At 194 dB the atmosphere can no longer sustain the pressure of a sound wave, though instantaneous effects like explosions can exceed this level. The explosion of Krakatoa is thought to be the loudest sound in modern history, and it was heard up to 3,000 miles away.

We use a simple device called a dB Meter to measure the SPL of a particular sound. These

meters are available at music shops and other electronics stores, and a number of places online. You

can also download dB Meter applications for iOS and Android which are fairly accurate. On most of


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these devices, you simply set the dial to the likely decibel range, then it displays the reading. If a device

offers different weightings, we typically use the A-Weighting as it best identifies the decibel rating

based on human hearing.

Because we hear different frequencies at different levels, we make a distinction between the

rating of dB (SPL) and the loudness of a sound. Whereas dB (SPL) is a very objective measurement, the

loudness of a sound is how we perceive it. For example, there may be a very strong 10 Hz sound wave at

160 dB, but this same wave is not loud because it is outside of our hearing range. We will not get into

specifics about the distinctions of loudness (measured in a unit called phons), but if you are interested

you may want to look up the equal-loudness contour and the Fletcher-Munson curves.

Balancing the output dB of the system in a church service is delicate. We usually try not to push

too far beyond 90 dB, as this makes some people uncomfortable, and your goal is to make it a good

worship experience for everyone. However, you will quickly learn that you just cannot make everyone

100% happy. Some people want the music to shake their very souls so they can sing loudly over it and

really "feel" the music, while others want to experience it in a subtler way. While 85 or 90 dB is certainly

a good listening and mixing level for studio recordings, it can a difficult range to balance in churches,

especially larger spaces and gyms. The key is to get it to sound as good as possible during rehearsal,

then if you find that you're pushing the levels too hard, pull back on the entire mix a bit until it sits at

the proper dB levels.

3.4 SOUND PROPERTIES

Sounds have several properties which can affect the way we perceive them. The characteristics we will discuss briefly are reflection/absorption, refraction, and diffusion.

Reflection is quite simple, and we encounter it all the time. Sound has the ability to bounce off of surfaces and redirect itself at whatever angle is perpendicular to the surface. This is why a sound playing from a speaker may bounce off of a wall behind us and be heard as a distinct repeat of the sound. Sound travels through various materials at different velocities, and this velocity also changes based on the temperature. However, we can generally say that sound travels through air at 761 mph (1116 ft/sec). Because of this, the distance from the sound source to the listener may cause the sound to be heard later than it was played, which is why we hear reflected sounds later than the original. Whenever we hear two distinct sounds, the original source of the sound and its reflection off a surface, we call it an echo.

When we are in a room, a sound can bounce all around the room causing a large number of echoes. When these echoes are very quick, numerous, and indistinct, we call it reverberation or reverb for short. The size and materials of the room cause the reverb properties to change, where a large room will have a longer reverb time. Also, surfaces which are more reflective will create a more live room, with longer reverb time, and rooms with softer absorbing materials will cause the reverb to die off more quickly.


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The property of refraction is essentially the ability of sound to bend around surfaces. This is why we can still hear someone if they are around a corner or behind an obstruction. Diffusion is the way that sound spreads out after reflecting. A surface with a number of different angles will cause greater diffusion, while one single wall will create a more direct reflection. These two properties are less important for our purposes than the previous.

In cases where church meets in a gymnasium, reverberation is a major hindrance to creating good sounding mixes. For example, if you measure a reverb time of several seconds at about 60 Hz, which is close to the fundamental frequency of a bass drum, we actually hear a single kick of this drum for several seconds after it plays.

4 LIVE SOUND SETUP

In this section we finally get into the specifics of practical audio engineering, and will discuss a number of topics such as gear and their applications, practical uses of microphones and speakers, and several others. As a basic introduction to this section, we will introduce two terms: front of house and monitors. Whenever we discuss front of house, typically abbreviated as FOH, we are talking about the performance that we hear on the main speaker system. The FOH mix is what the audience and you as the engineer will hear. Conversely, a monitor mix is what the band hears as they perform. These will always be two separate mixes. If your church has a personal mix system such as Aviom units, the band

4.1 MICROPHONES

Microphones are the primary means of getting a sound signal from the stage to the FOH, and there many types of microphones and techniques for using them.

4.1.1 TRANSDUCERS

A transducer is simply a device that converts one form of energy to another. All microphones have a transducer, and its job is to convert sound energy into electrical energy. We can then amplify this electrical energy and reproduce it once again as sound energy. There are three primary types of microphone transducers, and they serve different purposes in recording and live sound. In any microphone, part of the transducer is called the diaphragm. This is generally the piece of the microphone that moves in response to sound vibrations, and is similar in nature to the human ear drum.

Perhaps the most important, and certainly the most common, type of transducer in live sound is the dynamic microphone. A dynamic microphone is one which uses a magnet and a coil of wire to create the electrical energy. Essentially, when sound enters a dynamic microphone, a coil of wire at the diaphragm moves back and forth in response to the motion of the sound. The moving wire causes a difference in electrical potential across a magnet that is positioned just outside of the coil. This electrical movement creates a voltage which is sent out the microphone, through the microphone cable, and on to whatever medium we use them for.

The advantages of dynamic microphones are that they are relatively cheap to make, are very durable and can take a beating on stage, and can handle very loud sounds without distorting. These


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features are especially useful in miking drums, guitar amps, and often vocals. We will discuss common microphone makes and models in a few sections, and will discuss what instruments they are typically used on. There is a subset of dynamic microphones called ribbon microphones. Although they work on the same principles, these are usually very expensive and basically never used in live sound. The ribbon inside is very fragile, and these microphones create a warm, saturated sound that is a bit reminiscent of old recordings on tape.

Another common type of microphone is the condenser microphone. Inside a condenser mike, the diaphragm is actually one plate of a capacitor. The diaphragm is free to move in response to sound vibrations, and the second plate is fixed in place. When the diaphragm moves, it causes the distance between the plates to change, creating the difference in electrical potential.

In all condenser microphones, the capacitor inside must maintain an electrical charge at all times. Early on this was done with a battery or large power supply, but this electricity is now supplied through phantom power. Phantom power is simply a 48-volt DC power source which is sent through the microphone cable and is used to power it. With the simple switch of a button, the engineer can send phantom power from his workstation to the microphone. While all condensers need phantom power, it is unnecessary for dynamic microphones, and can actually harm some ribbon microphones. When in doubt, simply ask another engineer or a musician who might know whether a particular microphone needs phantom power. If you are having trouble getting a microphone on stage to work, it could be that you have not sent it phantom power.

Condenser microphones are great in studio recording, because they are very precise and sensitive to sounds. On vocals and on instruments that are delicate or contain a lot of high frequency information, or on instruments with strong transients (quick bursts of sound energy such as the quick strike of a drum), condensers are often the best choice. They are quite often used on drum cymbals, piano, acoustic guitar, vocals (especially female), strings, and more. However, they are much more delicate than dynamics, and usually cannot take loud sounds, which is why we avoid using them on loud instruments like bass drums, snare drums, and guitar amplifiers. Usually the only condensers on a live stage like will be drum overheads (one or two mikes place above the drums to pick up the sounds of the cymbals), the hi-hat microphone, or the vocal microphones.

4.1.2 DESIGN AND FUNCTION

The microphone design that most people are familiar with is the handheld microphone. These are the kind we see that people hold up to their mouth or are placed on a microphone stand at concerts. This style is used at most churches for the singers and often for speakers on stage. These are meant to be so that the mesh or fabric grille over the top of the microphone (where the diaphragm is located) is pointed towards the sound source. On the opposite end are three prongs where we connect the microphone cable.

Instrument microphones are often designed in a similar way to this. They are designed so that one end is quite obviously where the diaphragm is located, and at the opposite end is the cable connector. Some are shorter and wider, while others are long and skinny. With condensers, we usually break them into two types: large diaphragm (LDC) and small diaphragm (SDC). The LDC mikes are most often used in studio vocal recording. They stand up tall and the vocalist sings into the side. A SDC however, is long and thin, which is why it is often called a pencil condenser. These are most often used on instruments rather than on vocals.


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One other type of common microphone is the Lavalier microphone, or the lav mike. These are very small microphones which are either pinned to a speaker's clothing and pointed towards their mouth, or are fashioned around the ear and out towards the mouth. Over-the-ear lav mikes are very common at churches for the pastor and for other speakers on stage. Although many lav mikes are condensers, they are usually connected to wireless packs which are battery operated, so we don't need to worry about phantom power.

4.1.3 PICKUP PATTERNS

Another property of microphones to consider is their pickup pattern or polar pattern. This describes the directionality of a microphone. For example, a microphone will either be very directional, so it will only pick up sounds that it is pointed at, or it can be non-directional and pick up sounds from all directions.

The most common pickup pattern is the cardioid microphone. If we inspect the figure below, we can see that this pattern attempts to capture only the sounds directly in front of it (the outlined portion is what is being captured, and 0 degrees is directly in front of the microphone). The pattern is called cardioid because its pattern diagram looks like a heart. This pattern is also called unidirectional, because it picks up sound in one direction. These are almost always the proper choice in live sound, because we want to capture the sound of the specific instruments or singers, and avoid other sounds on the stage. One example of a cardioid microphone used in live sound is the very popular Shure SM58 vocal microphone.

Figure 3. Polar Pattern of a Cardioid Microphone [2]

There are two additional versions of unidirectional microphones called supercardioid and hypercardioid. Their patterns are shown below. As the names imply, they are cardioid mikes which are even more directional in nature, but at the cost of introducing a small amount of directionality at the back of the microphone.

http://www.shure.com/americas/products/microphones/sm/sm58-vocal-microphone


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Figure 4. Polar Patterns of Supercardioid and Hypercardioid Microphones [2]

The following two types of polar patterns are quite uncommon in live sound, so we will only briefly discuss them. The first is the omnidirectional microphone, which picks up sound equally in all directions. Figure 5 shows this rather self-explanatory polar pattern. The final pattern is the bi-directional or figure-eight microphone, which is essentially a cardioid pattern at both ends of the microphone. In other words, it picks up sound from the front and back equally, but rejects sound from the sides. This is the natural polar pattern of a ribbon microphone, and can also be used to create a stereo microphone, since it will differentiate sounds from the front versus sounds from the back. A bi-directional pattern is shown in Figure 6.

Figure 5. Polar Pattern of an Omnidirectional Microphone [2]


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Figure 6. Polar Pattern of a Bi-Directional Microphone [2]

4.1.4 FREQUENCY RESPONSE

There are thousands of microphones available. We choose which one to use largely based on its transducer type, its design and function, and its pickup pattern. However, even within a single combination of those properties, there are hundreds to choose from. What differentiates each of these microphones is the way they respond to certain frequencies, which is a property that we call frequency response. For example, many handheld vocal microphones have a presence peak which accentuates the 4 kHz to 5 kHz range, because that makes the human voice louder and more present. When we look at the specifications of a microphone, we may see a chart that looks like the following:

Figure 7. Frequency Response Chart of a Shure SM58 [3]


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This is a frequency response chart for a common handheld vocal microphone called the Shure SM58. The horizontal axis shows the frequency, and the vertical axis represents how much that particular frequency is accentuated or cut. We can clearly see the presence peak between 2 kHz and 5 kHz, and we see that the low frequencies start to fall off at around 100 Hz, while the high frequencies fall off at around 9 kHz.

The frequency response of a microphone is one of the largest influences on how it makes something sound. It is largely the reason we choose the mikes that we do, and why some mikes are highly regarded for the way that they sound on a given instrument or type of voice.

4.1.5 THE PROXIMITY EFFECT

The proximity effect is something that affects the frequency response of a microphone. The short story is that when a microphone gets closer to a sound source, the bass frequencies are accentuated. In particular, moving a mike closer to the source causes it to boost the low frequencies at around 100 Hz. This effect only occurs on cardioid and bi-directional microphones, but not on omnidirectional mikes. Below is a similar frequency response chart to the one we just saw, except that it also shows how the response changes based on the distance of the microphone to the source.

Figure 8. Frequency Response of a Shure Beta58 Based on Distance [4]

Since these types of microphones are held right up to the singer's mouth, they are designed to roll off at the low frequencies because the designers know that these frequencies will be added back in because of the proximity effect. From this effect we know that if something we are miking sounds too "boomy", "muddy", or generally too "bassy", it may help to back it up a bit from the source.

4.2 PERIPHERALS

Peripheral hardware used with microphones and instruments are pretty self-explanatory, for the most part, but we'll quickly look at some of the options.

There are three basic types of microphone stands. The straight stand picture at the left of Figure 9 only telescopes vertically. They are really only useful for vocalists who are not also playing an


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instrument. If the vocalist is playing an instrument or you need to mike some kind of instrument, a stand with a boom attachment, pictured in the center, is the best option. These can be manipulated around obstructions and keep the mike out of the way. The third is a shrunken version of the boom stand, meant for low-lying instruments like bass drum, snare drum, and guitar amplifiers.

Figure 9. Microphone Stands: Straight, Boom, and Low-Profile Boom [5]

You will deal mainly with two types of cables in live sound: XLR and 1/4" Instrument. An XLR cable

is the type used to plug in microphones. A microphone has a male XLR connector, so you will see three

prongs protruding from one end. Plug the female end of an XLR cable into this, and run that wherever it

needs to go. The connectors look like this:

Figure 10. XLR Connectors and Example Cable [6]

Cables can often pick up radio and cell phone transmissions causing buzzing or extraneous signals, and the longer the cable, the more likely it is to pick up such noise. However, XLR cables are wired with positive and negative connections (called a balanced cable), which in practice means that they cancel out any noise picked up across the cable. These cables can be as long as they need to be, and you can extend an XLR cable by plugging two or more of them together.

The other cable you will run into is the 1/4" audio cable. These are basically the same type of connector as headphone cable (which is 1/8"), and the larger 1/4" cables are often used for professional


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audio and instruments. 1/4" cables used for instruments like guitars are unbalanced, which means that they are quite prone to noise. In this case, the shorter the cable, the better. If you hear a lot of buzz from one of your musicians, you might need to suggest swapping cables.

4.3 DI AND MICROPHONE TECHNIQUES

4.3.1 DIRECT BOXES

An important piece of gear onstage is the Direct Box (a.k.a. the DI or Direct Input Box), which is used to take an unbalanced signal, such as a guitar cable, and convert it to a balanced signal. A common use might be to plug the worship leader's acoustic guitar into the DI, then run an XLR cable from the other side and out to your workstation so that you can hear it at the FOH. The same idea is used for keyboards, although sometimes you might use two DI boxes or a stereo DI box so that you can get both the left and right channel of the keyboard to the FOH. If a direct box says that it's active, you will probably need to supply it with phantom power. If it is passive, there's nothing else to worry about. A common DI box is shown below.

Figure 11. A Radial Brand Direct Box [5]

4.3.2 CLOSE MIKING TECHNIQUES

The rest of your instruments and vocalists will need a microphone to be heard over the FOH. The majority will be close miked. As it sounds, this means that the microphone will be fairly close to the sound source. This isolates the instruments so that noise from the stage does not leak into the microphones as much, giving you better control over the level of the individual instruments. The most obvious is that the vocalists will be right next to their vocal mike, but this applies to instruments as well. Slight variations in the placement of the microphone can make a big difference in how it sounds over


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the system. In the following techniques we will assume some knowledge of the instruments themselves for brevity's sake. If you are unclear on these terms, just talk to the band members or other sound engineers, and they should be able to help you get the best mike placement.

An easy way to mike a guitar amplifier is directly on the grille covering the amplifier cabinet (note that on a combo amplifier, the lower half is the speaker cabinet and the upper half is the amplifier circuitry, but on a stack or half stack, the amplifier and the speaker cabinet are two separate pieces). A dynamic, cardioid microphone like a Shure SM57 is common and works quite well for this. As you move the microphone closer to the center of the speaker, the sound gets brighter and often too harsh. Somewhere between the center and edge of the speaker gives you a bit mellower sound. Pointing the mike directly perpendicular to the cabinet is called on-axis and angling it is called off-axis miking. Miking on-axis generally sounds brighter, but also introduces greater proximity effect, causing high and low boosts. Miking a bit off-axis sometimes sounds more natural, and may be a good place to start. In general, however, if it sounds too muddy, you might move the mike closer to the center. If it's too bright or harsh, move it towards the edge. In either case, if there is too much low end, it may help to move it back an inch or two. These techniques work for all electric guitar amps, including bass guitar amps.

Drums can often be the trickiest to mike, though often the drummer at church will handle the initial set up. A good snare setup is to put an SM57 on a short stand and aim it 45 degrees down at the top of the drum head. If you point the mike too close to the center of the drum, it may sound a bit muddy and get in the way of the drummer, so pointing it about an inch away from the edge is a good starting point. Toms are similar, though many churches use clip-on microphones that simply attach to the toms and you're ready to go. If you mike the hi-hat, a small diaphragm condenser pointing down at the edge of the hat an inch or two above it works well. It often helps to point the hat mike off-axis away from the snare drum to prevent snare bleed.

Miking the cymbals can be finicky, and is done with an overhead microphone setup, which is an array of microphones placed a few feet above the entire drum kit. They pick up the kit as a whole, but live are mainly intended to capture the cymbals. The simplest overhead is to place a single SDC somewhere above the middle of the kit. In most cases, the best route is to place two microphones above the kit, one at either side, to form a spaced pair. The advantage is that you pick up more of the kit, but also so that you get a stereo image of the drums, so you can hear distinct left and right sides of the kit playing through the FOH. This option is not always available at churches, as you must use two of the same microphone for this effect to work properly, plus it adds additional time and hassle to the setup and sound check.

The kick drum is the final piece of the kit. Most setups use specialized microphones for kick drums - models such as the Audix D6, the AKG D112, or the Shure Beta 52. These have frequency responses which accentuate the kick drum well, giving it a deep low end "punch" but also boosting the "slap" and "presence" of the beater hitting the kick drum head. If the kick drum has a port cut into the front head, place the microphone on a short stand just a couple inches inside the port, pointing straight towards the beater head. Otherwise, place it about halfway between the center of the kick drum and the edge, and about an inch away.

If you come across an acoustic instrument such as acoustic guitar, banjo, mandolin, violin, etc. which does not have a way to plug in to a direct box, you will need to mike it. Usually a small diaphragm cardioid condenser is the best choice, but if these are unavailable or you are worried about possible damage to the microphone, a cardioid dynamic microphone should suffice. Acoustic guitars are best


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miked pointing at about the 12th fret a couple inches away. With a violin, a condenser microphone is greatly preferable. Place it about a foot away and pointed at the instrument. A banjo essentially has a drum head built into its body, so mike it sort of like a snare. If you have an additional clip-on mike available, these work well for isolating a banjo.

Miking techniques are also discussed on a per-instrument basis in Module 3 of this manual.

5 SIGNAL FLOW

5.1 THE SIGNAL PATH

Signal flow is the trail of connections that establish your entire sound setup. In church stage setups, this practically means the signal path for how you get audio from the microphones on stage to the FOH loudspeakers. Here is a basic signal flow that is common at many churches:

The signal path starts at the microphones and DI boxes. An XLR cable runs from these devices to the snake. The snake is housed on a reel behind the curtain, and has many XLR ports on it. Each port is labeled with a number that corresponds to their channel. These inputs are often called tie-lines. The snake is a very long collection of cables that run all the way from the stage to the FOH booth. The microphone outputs you just plugged into the inputs on the snake run along the length of the cable, so it is essentially like you took each microphone and ran its cable all the way back to the workstation. This all happens conveniently in one cable. The snake is plugged in to the FOH unit by either a large multi-pin connector, or individual cables into each channel. The workstation is often housed in a rack unit, which holds the workstation/mixing board on top, and may also contain processing units, power supplies and power conditioners, and/or drawers for additional equipment.

We typically refer to the workstation as a console. It is where you control all elements of the sound, including levels, mutes, panning, processing effects, and more. Most of your experience with the console will come from hands-on work with other engineers, so this manual will not share many specifics of any specific console or general console use. Most operations are very simple, and any complex tasks you would like to do are covered in the manual.

The connector from the snake feeds all of your microphone outputs to the inputs of the console. Now, the microphone plugged in to channel 1 on the snake is plugged in to channel 1 on the console, and so on through all channels. Depending on your console, the board may have 8, 16, 32, or even more individual channels. When channel 1 is plugged in to the console, it runs to a microphone preamp, which supplies the proper level to a microphone so that it can be heard. By turning up the gain of channel 1 on the console, you can effectively increase how sensitive the microphone is. Turning it up too much causes the preamp to distort, but with too little gain, you will not have all the power you need to be able to set the proper level in the FOH. Once you have the gain set, the signal runs to the channel 1 fader, which controls the level of that particular channel to the overall mix.

The overall mix, combining each of the channels, is sent out of the console and back across the snake. The snake has outputs that connect to the FOH loudspeakers, so that the mix from the board can be delivered to the congregation.

5.2 GAIN STRUCTURE


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This section will be added in the future. Some information is outlined in Module 02.

6 SIGNAL PROCESSING

6.1 EQUALIZATION

Equalization is the act of boosting or cutting a particular range of frequencies to change the tonal balance of a sound. The device to perform equalization is appropriately called an equalizer, and we often use the shorthand EQ to describe both the verb and the noun form. If you want more bass in an instrument, for example, you could use an EQ to boost the low frequencies. EQ units come in the form of external hardware, built-in functionality of the console, or as software. There are several common types of EQ that we will describe, starting with basic EQ functions then to specific types of EQ units.

6.1.1 FILTERING

The most basic type of equalization is a filter. The goal of this type of EQ is to "filter out" a range of frequencies. The most common filters are high-pass and low-pass filters. These names are off-putting at first because the naming scheme essentially describes the frequencies you keep rather than the ones you want to filter out. So for example, a low-pass filter passes or keeps the low frequencies, and filters out high frequencies. This is why a low-pass filter is often called by the less confusing name of high-cut filter, and a high-pass filter is also called a low-cut filter.

It is not mechanically possible to eliminate all frequencies past a certain point, so they are filtered out on a slope. Typically, the slope of a filter is measured in dB/octave, so a typical high or low pass filter might filter out at 6 dB, 12 dB, or 24 dB per octave. Basic graphs of these types are filters are shown in Figure 12.

Figure 12. Generic High Pass and Low Pass Filters [7]

There are two other common filters - the band pass and the notch filter. A band pass is essentially a combination of both a high pass and low pass filter, so it throws out the high and low frequencies and keeps the middle frequencies. It is quite a flexible filter, as you can typically set the range of middle frequencies to keep. A notch filter is the reverse process, where it throws out a small range of frequencies somewhere in the midrange. Band pass filters typically use a very quick slope, so


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they only affect a very tight range of frequencies. You can target one particularly bothersome frequency and will not affect many other frequencies. One common usage would be to reduce the hum at 60 Hz caused by electrical outlets. You can see examples of band pass and notch filters in Figure 13.

Figure 13. Generic Band Pass and Notch Filters [7]

6.1.2 SHELVING

Another common type of EQ parameter is the shelf EQ, which is designated as either a high shelf or low shelf EQ. Instead of gradually reducing a range of frequencies, shelf EQs affect all frequencies above or below the specified frequency equally. Additionally, they can be used to boost, not just cut, a range of frequencies. For example, if you set a high shelf to 5 kHz, you can either boost or cut all frequencies above 5 kHz equally, with only a short slope immediately at 5 kHz. Figure 14 shows examples of each shelf, where the dashed line indicates your option to cut rather than to boost.

Figure 14. Generic Examples of Low and High Shelf EQ [7]

6.1.3 PEAK

Peak EQ is probably the most common parameter you will adjust, and it is quite straightforward. Given a target frequency, you can choose to boost (causes a peak) or cut (causes a dip) that frequency


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by some dB level. As with the filters, you cannot electronically affect only a single frequency, so peaks occur over some range of frequencies. As we will see when we discuss parametric EQs, the range of frequencies is often adjustable. The following figure compares a peak EQ with a shelf at the same frequency.

Figure 15. Comparison of a Peak EQ and High Shelf at 2 kHz [8]

6.1.4 ADDITIVE AND SUBTRACTIVE METHODS

This section will be added in the future.

6.2 EQUALIZATION UNITS

6.2.1 GRAPHIC EQ

Graphic EQs are very simple, and are most common in live sound. Rather than being used on individual instruments, they are typically used to EQ an entire mix, for instance to fix frequency anomalies caused by the dynamics of the room or to balance out the frequency response of the loudspeakers. These EQs are laid out graphically in bands, which affect a small range of frequencies around a center frequency listed on the unit. Each band is typically a peak EQ, though the units vary and may include various filters. In the EQ shown below, you can affect the frequencies 20 Hz, 40 Hz, 63 Hz... and up to 16 kHz, and can boost or cut each band by 6 dB.

Figure 16. A 2-Channel dbx Brand Graphic EQ [5]

6.2.2 PARAMETRIC EQ

The parametric EQ is the most versatile as it incorporates all of the EQ functions we have discussed in one unit, though it may take a bit of explanation. Most parametric EQ units are four or five bands (low, low-mid, mid, mid-high, high), but they all feature the following components:

1. Variable frequency per band


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2. Variable gain per band 3. Variable bandwidth or Q factor per band

The first feature is simple: at each band, you can select the center frequency you would like to adjust. As an example, the first band may let you select between 30 Hz and 400 Hz, the second between 75 Hz and 1 kHz (so there is usually some overlap), the third between 800 Hz and 12 kHz, and the last between 2 kHz and 20 kHz. Once you have selected the frequency of a particular band, you can then adjust the gain in dB. Usually you can cut or boost the selected frequency by as much as -9/+9 or -12/+12 dB, though the unit usually displays the numbers.

The final feature says that at each band, you can also set how wide the bandwidth is. A wide bandwidth indicates that many frequencies around the selected center frequency are affected, and a narrow bandwidth indicates that fewer frequencies are affected. This is also known as the Q factor or quality factor, a term from electrical engineering. In the case of bandwidth, the effect is usually described in octaves, so a wide bandwidth may influence several octaves and a narrower bandwidth half an octave.

Q factor is usually represented by a number with no units, and the number itself is less important than the effect, as most digital parametric EQs display this factor visually across a frequency spectrum graph. It is important to note that traditionally, a high Q factor means a narrow bandwidth, and a lower Q factor means wide bandwidth. However, some units such as the Yamaha LS9 console reverse this, where a high Q factor means a wide bandwidth. Just be wary of this, and pay attention to the display to see how the bandwidth is measured. A typical parametric EQ that you might see on a console is shown below.

In this EQ, take a look at the green section labeled HMF (high-mid frequency). The kHz knob allows you to set the frequency between .6 kHz (600 Hz) and 7 kHz. You may then adjust the gain with the dB knob, though the specific dB levels are unlisted (in this case, just use your ears, the number itself really doesn't matter as long as it sounds good). Then with the Q knob you can adjust it to be either a narrow bandwidth (counter clockwise) or wide bandwidth (clockwise). It is important to note that the HF and LF sections are both shelf EQs, and so the

bandwidth is not adjustable. By clicking the button with the red light and a peak symbol immediately to the right of the red and black dB knobs, you can set these two sections to be peak EQs instead of shelves.

This brings us to the final point on parametric EQs. The type of EQ used, whether shelves, peaks, or filters, is often adjustable as well, especially in digital EQs with full displays. It is a very common feature that you can change the high and low bands to be either peaks, shelves, or filters, or use some combination of those parameters.

6.3 COMPRESSION AND LIMITING

The art of compression takes time and a lot of experimentation to master, so the best thing we can do is give you the basics and let you run with it. The idea of compression is simple: it is like an

Figure 17. SSL Brand Parametric EQ [5]


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automatic fader adjuster. The real heart of compression is that it reduces the dynamic range of the instrument or mix that you put it on. Compression ignores the quietest parts of a track and automatically turns down the loudest parts or peaks in the dynamic range. Every compressor will feature the parameters of threshold, ratio, attack, release, and makeup gain. I usually conceptualize compression and these parameters with the following illustration.

Imagine that there is a group of tiny elves living in your compressor. Their job is to control the level of whatever is going into the compressor. Say we send a bass guitar through the compressor. The bass is playing happily along and the input to the compressor reads -20 dB consistently. But then, the bass plays louder and the input reads -10 dB. If the compressor has a threshold of -15 dB, the bass guitar playing at -10 has crossed the threshold. This triggers an alarm inside the compressor, and the elves rush over to the fader so they can turn down the level of the instrument. The speed at which the elves run over to the fader so they can turn it down is called the attack time, and how much they turn it down is called the ratio. Once the bass player calms down and the level goes back under the threshold of -15 dB, the alarms stop and the elves know they can turn it back up to where it was. The speed at which they allow the level to go back to normal is called the release time.

All goofy illustrations aside, this is essentially how a compressor works. The net effect is that an instrument or vocal sits at a more consistent level throughout the performance. You don't want to take away the groove of the performance by making it exactly the same level throughout, but it is quite helpful to bring down some of the loudest elements automatically so that you don't have to constantly adjust the fader to keep the mix balanced.

The threshold you set depends on the input level of the audio. You should start with it all the way up, then turn it down until it indicates that the parts you intend to reduce are hitting the threshold. Compressors have different ways of showing this. Some are as simple as an LED light turning on to indicate the input is past the threshold, while some display much more detailed information.

The ratio is shown in terms such as 1:1, 2:1, 4:1, 10:1, etc. At 1:1 ratio, no compression is being applied, even when the input is past the threshold. At a 2:1 ratio, if the level goes 2 dB past the threshold, the compressor will turn it down so that it's only 1 dB past the threshold. Similarly, at 8:1, a signal that is 8 dB past the threshold will be pulled down to 1 dB past the threshold. As an example, if we have a threshold of -10 dB and the input is at -6 dB, we are 4 dB above the threshold. With a ratio of 4:1, the compressor will reduce this level by 3 dB (this is called a gain reduction of 3 dB) and output at -9 dB. Any compressor running at a ratio of 10:1 or higher is typically referred to as a limiter, or we say that the output is being limited. A limiter that operates at about 60:1, and often denoted ∞:1, is called brick wall compression. The following figure shows an example of threshold and ratio.


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Figure 18. Compression Example Showing Threshold and Ratio

While it may be good to start with a ratio of 4:1 or 6:1 for most instruments or vocals, this can vary greatly on what you want to accomplish. Similarly, attack and release times are creative parameters that allow you to shape the sound, so there's not necessarily a right or wrong setting. A fast attack time makes the compression kick on sooner, and a fast release time makes it resume uncompressed operation more quickly. These times are listed on compressors in milliseconds or seconds.

A typical rule of thumb is that on transient instruments, you may want to use a slower attack time. Instruments with strong transients have quick, short bursts of energy, so things like a snare drum are very transient, whereas vocals, violins, and bass guitar have fewer transients. Using a fast attack time on a snare would greatly reduce the transients of the instrument, causing it to lose that "crack" that makes the instrument what it's supposed to be. A slow attack time will allow the transient through and then turn down the resonating loudness of the instrument, giving you a sharp sound but more consistent signal level.

It's usually good to have a quicker release time on something like a snare drum, since it's a quick sound, we would like to bring the level back to normal rather quickly. Giving instruments a long release time usually causes them to sustain. Long release time is nice on a bass guitar. Imagine that we are at the last note in a song that's ending, and each band member holds the last note until their instruments die out or decay. If we compress the bass to level out the peak of his last pluck, then use a long release time (maybe several seconds long), the compressor will drop down the level, then very slowly bring it back up to normal. That means that as the bass guitar decays over several seconds, the compressor is doing the opposite by bring it back up over several seconds. This means that you get a nice even sound and the bass sustains for a long time. Figure 19 shows an illustration of attack and release times.


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Figure 19. Illustration of Compressor Attack and Release Times [9]

While compression is sometimes used as an effect, to make sounds "pump" or "move" in a way that sound interesting, compression is usually something that you are not supposed to hear. If you can hear a sound being compressed, it is probably too extreme. Compression, in most cases, should be transparent, meaning that as the engineer you know it is helping to balance the mix, but the listener should not be able to hear levels being changed. It is meant to smooth out mixes and prevent an instrument or voice from leaping out too far above the mix.

You may note that compression only turns down loud sounds - it does not increase the level of quieter sounds. In the end, looking at the waveforms, you would see that compressed sound is flatter and less dynamic than the uncompressed sound, and it is also quieter because the peaks are no longer present. Therefore, most compressors have an adjustable output level, often called the makeup gain so that you can "make up" for the amount of gain reduction you applied.

6.4 GATING

In many senses, gating is the opposite of compression. This effect is also called expansion because it expands the dynamic range rather than limiting it, and it's sometimes specifically called a noise gate. Practically, gating is mostly used as an automatic mute switch. This is great in live sound, because all of the microphones on stage are going at once. If there is an instrumental section of a song, it may be wise to mute the vocal microphones so that we don't pick up the other instruments. Gates have essentially the same parameters as a compressor.

The threshold on a gate determines at what level the gate will "open" and allow sound through. If it's an instrumental section, and the vocal mike is picking up ambient noise at an input level of -50 dB, we could set the threshold to -40 dB, so that the mike is essentially muted. If the singer comes back and the input reads -20 dB, the gate clearly opens and allows the vocal to be heard.

Gates have attack and release times that determine how quickly the gate opens, and how quickly it closes once the sound is lower than the threshold, respectively. They also have a parameter similar to the ratio control, except on a gate it is called the range. Usually we want the range all the way up, because we want to reduce the sound by the full range if it hasn't reached the threshold. This will


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cause it to mute entirely if the gate is not opened, but with a lower range, some sound will get through even when the gate is closed.

6.5 DE-ESSING

This section will be added in the future. Some information is outlined in Module 03 in the vocals section.

7 FINAL DETAILS AND FUTURE CHANGES

I am always happy to answer questions and fill in details where they are needed. This handbook is intended to change over time, so I will occasionally return to it to add details, clarify text, or clean up extraneous information. If you have any comments, suggestions, or questions, please contact me through one of the following methods:

[email protected]

@colin_hemphill

You can always find the latest version of this handbook and its other modules on my website at http://www.colinhemphill.com/resources.

8 BIBLIOGRAPHY

[1] Wikibooks, "Signals and Systems/Periodic Signals," [Online]. Available: http://en.wikibooks.org/wiki/Signals_and_Systems/Periodic_Signals. [Accessed July 2016].

[2] Wikipedia, "Microphone," [Online]. Available: http://en.wikipedia.org/wiki/Microphone. [Accessed July 2016].

[3] American Musical Supply, "Shure SM58 Dynamic Vocal Microphone," [Online]. Available: http://www.americanmusical.com/images/descimages/ShureSM58FreqResponse.jpg. [Accessed 22 July 2016].

[4] P. Dakeyne, "SM58 - Shure Enough The Best," 27 February 2009. [Online]. Available: http://magazine.dv247.com/2009/02/27/sm58-shure-enough-the-best/.

[5] Sweetwater, [Online]. Available: http://www.sweetwater.com. [Accessed July 2016].

[6] Wikipedia, "XLR Connector," [Online]. Available: http://en.wikipedia.org/wiki/XLR_connector. [Accessed July 2016].

mailto:[email protected]

https://www.twitter.com/colin_hemphill

http://www.colinhemphill.com/resources


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[7] J. v. d. Heuvel, "Signal Processing," Fuzzbass - Bass Guitar Amplifier Knowledge Base, 2009. [Online]. Available: http://fuzzbass.fuzzphoto.eu/020703.html. [Accessed July 2016].

[8] R. Dennis, "Signal Processing With Equalization," Recording Institute, 2001. [Online]. Available: http://www.recordinginstitute.com/da154/ARP/chap3Sig/asp3.html. [Accessed July 2016].

[9] Wikimedia Commons, "File: Audio Compression Attack and Release," 31 August 2007. [Online]. Available: https://commons.wikimedia.org/wiki/File:Audio_Compression_Attack_and_Release.svg. [Accessed 2016 July].

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