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Sound Sense!

"Sound Sense" is a live sound engineering manual that focuses on the fundamentals of sound and sound reinforcement, and has been an excellent resource for training and developing budding sound technicians at various skill levels. Please print the manual out for your own or your organization's instructional use only. Feedback and questions can be directed to: Edward Craner, Little Peach Music, Inc. e-mail: ecraner@littlepeach.com

Introduction


Basic Operations

Intermediate Operations

"Sound Sense" is written by Edward Craner and Scott MacGill. Copyright © 1990.


FOREWORD
After many frustrating years of listening to talented musicians sound like they just lost the $1.98 talent show, or a seminar speaker sound as if she was suffering from chronic nasal congestion, an attempt has been made to do something about it. The world of sound reinforcement has always seemed to be filled with two categories of people: 1. Wannabe musicians who have the musical background, but have little or no comprehension as to the technical aspect, and usually have a chip on their shoulder that inhibits their effectiveness; 2. Technically oriented people who understand the principles of sound, but couldn't hear the subtleties in a song's mix if their life depended on it, as well as having the wrong temperament for working with the unpredictable and usually flaky attitude of a musician. Our answer? "Hello Left Brain, meet Right Brain." This manual is designed to assist individuals and organizations in applying the practical elements of sound reinforcement from both the technical view as well as the aesthetical. By combining these two separate but equally important elements of sound reinforcement, a well-rounded and diverse set of skills can be developed. The key, however, is to apply them as you learn them. This course wasn't designed with acoustical engineers in mind, although even they might find it helpful. Instead, this manual was prepared for those of us who, through no fault of our own, have found ourselves in the operator's seat of this thing called a sound system. It's also for those who have an interest in sound but prefer not to get a formal education from a university. Or maybe people who just want to broaden their horizons. At any rate, that's why we created this manual and the coursework it contains: to give people the opportunity to learn and implement the basic elements of sound reinforcement. This manual is broken down into four user sections, catering to the reader's ability at each level. For instance, Section I is designed to introduce you to the properties of sound and how to visualize some of sound's characteristics. This will establish a foundation on which to build on. We recommend that even if you feel confident with your basic sound knowledge, read each section. We are certain something new will be introduced and will be worth your effort. Sections II, III and IV continue to take you step by step through the components of a sound system, the functions of a microphone, equalization fundamentals, and more. The information becomes more detailed and technically oriented as the manual progresses, so again we encourage you to follow the course as it is laid out. We hope this proves to be helpful in fulfilling your engineering needs, developing your skills, and building your knowledge and confidence. Sound is a great tool and can be effective when used properly. Strive for excellence, be open to learning, and always remember to...TURN IT UP!

INTRODUCTION:
As Julie Andrews said so many years ago, "Start at the very beginning, it's a very fine place to start." Truer words have not been spoken. Sound engineers everywhere have been thrown into positions where they don't have the proper background, training, or practical skills. High expectations for a flawless performance are placed on them, resulting in anxieties, frustration, and D.R.A.'s (Dirty Rotten Attitudes). Section I will start by establishing a philosophy of sound, helping you understand the point-of-view of all parties involved. From there, we'll move to the basic characteristics of sound, discovering how to visualize what we hear. Finally, using analogies and comparisons, we'll end up with a brief description of the very heart of sound reinforcement, the sound system itself. The formula for learning is simple: KNOWLEDGE + APPLICATION = PROFICIENCY. Start with that and you'll finish with the skills needed to efficiently perform your sound engineering duties.

I-A: PHILOSOPHY OF SOUND: Having a philosophy when taking on any task provides continuity, purpose, and a sense of accomplishment. Developing a philosophy regarding the sound you produce is important for all of these reasons, and will help maintain a focus on the GOAL OF SOUND REINFORCEMENT, which is: To create an environment where the majority of all listeners can comfortably hear a quality reproduction and blend of the sound sources. Listed below are a couple of points to consider when operating, choosing, or considering a sound reinforcement system, and should be adapted to each situation, despite the size or importance of the occasion. Striving for excellence each and every time sound is considered paramount when dealing with sound reinforcement.

I-A-1: KNOW YOUR AUDIENCE: Let's face it, heavy metal should be run at volumes that are painful and chamber music should be at a level that is comfortable for sleeping. This has been established based on the type of listeners each music attracts, and the demands the listeners have put on the volume. Occasionally though, we find we can't choose who is in the audience, and therefore must adapt each situation accordingly. Here's an example. An venue predominantly attracts singles and young married couples between the ages of 25 and 40. They were raised on The Beatles, The Rolling Stones, and Vanilla Fudge. But also in the audience are people who are more comfortable with Bobby Vinton, Henry Mancini, and Dinah Shore. What a combination. The music performed is mostly contemporary pop, with live drums, keyboards, guitars, and vocals. To do this style of music justice and have it be intelligible over the live drums, the engineer has to bring it up to volume levels that have stirred up some controversy as to "how loud is too loud." Well, it's all based on the vantage point of the listener and his/her personal preferences. This is where philosophy comes in. Not wanting to offend or make anyone uncomfortable is not a consideration, as was discovered. At that extreme, the quality of the sound suffered greatly. So what's the answer? BALANCE. Determine first-of-all what the average age and basic demographics are of your audience. This will largely determine the volume, style of music, presentation, etc. Second, be open to comments and criticism. Each of us have certain strengths and weaknesses regarding how a performance could or should sound. By disregarding another's input, we limit our growth in areas that we might fall short. Of course if a person criticizes with malice, kindly ask them to take a flying leap! Finally, be aware of the importance sound reinforcement has in your venue and the part you play as the sound engineer. A $10,000,000 auditorium with a $10,000 sound system is sure to be disappointing. It isn't necessary to purchase the best of everything and have the ability to accommodate all performance applications, but the dynamic that is present when the sound matches the intensity of the performers is beyond comparison. Today's listeners have higher standards than ever due to the developments in home audio equipment, Hi-Fi V.C.R.'s, stereo T.V.'s, compact discs, and DVD video. To maintain this standard of sound in a live application, higher levels of skill and specialized equipment are necessary.

I-A-2: CONSIDERING THE IMPORTANCE: When an organization requested assistance in their start-up, they requested a budget for a sound system. Housed in a remodeled a 4,000 square foot warehouse, a system that spent almost 25% of their $60,000 building budget was recommended for sound and lighting equipment. The investment was never questioned because of the freedom it gave them to produce quality sound reinforcement for singing groups, bands, drama, and speakers. The sound system helped to provide a relaxed environment free of any distractions due to system "buzz", feedback, hum, piercing highs, or unintelligible lows. In contrast, a nearby venue spent close to $1,000,000 on their building and only budgeted $6,000 for the sound system, waiting months in anticipation for the first service in their new facility, only to be disappointed by the inadequate coverage and low quality of sound. Sound is a powerful tool that can influence, control, and even manipulate the listener. It sets a mood, and can often "make or break" a production, whether it be a concert, speaking engagement, or wedding. Think of the hours of preparation and anticipation that are put into a performance that requires sound reinforcement. When that production goes before the crowd, an inadequate system or a unqualified operator can negate all of the preparation in the world. With this much importance put on one element, it should have the attention it deserves. When considering your next SOUND investment, whether it be in the form of time, money, or effort, take the time to develop a realistic philosophy. This will include an adequate budget, advice from qualified personnel, and training for your sound engineers. You'll find you will have a SOUND understanding, and when given a responsibility, it's just SOUND SENSE, isn't it?

I-B: CHARACTERISTICS OF SOUND: Feedback. Echo. Reverberation. 760 m.p.h. . . All of these terms relate to sound, but what do they mean? This section will help you to better visualize sound and the way it reacts, is generated, and received. It can be vital information for anyone associated with any element of sound, whether it be room design, sound equipment purchases, wall covering selection, or the operation of a complete sound reinforcement system.

I-B-1: THE NUTS AND BOLTS: Throughout this manual we will be referring to the analogy of water in reference to sound and its characteristics. We chose water because of its visible similarities to sound. For instance, the basic element in sound is referred to as a "wave," much like a wave in water. The way sound reacts off of surfaces is also common to water, in addition to the way a "source" (a rock, for instance) initiates waves when thrown into a pool of water, as does a sound source (the human voice) creates waves through the air when used (see diag. A).

I-B-1-a: VIBRATIONS = WAVES: Sound is nothing more than a series of vibrations (waves) traveling through a substance, and then is read by a device that can interpret these vibrations. The best examples are the God-given instruments we are born with, our vocal chords and ears. When our brain wants to communicate a thought, it instructs the vocal chords to vibrate at certain speeds (talking) which along with air from our lungs, produces sound waves and projects these sound waves in the direction we are speaking. A person with a lower voice has vocal chords that vibrate slower, thus producing a lower sound and a bigger sound wave. Someone who has a high pitched voice has vocal chords that will vibrate at a faster rate, producing a higher sound and a smaller (or faster) sound wave. After the sound is initiated, it travels at an average speed of 755 m.p.h. to the receiving instrument, your ear. Sound will travel through any substance that will vibrate. Based on this principle, the more prone a substance is to vibrations, the more accurate the vibration will be when it reaches your ear, and of course the inverse applies as well. For example, a concrete wall that is 10 inches thick will conduct sound (vibrations) less efficiently than a 1/4 inch sheet of plywood. When the sound finally reaches your ear, the waves are reinterpreted by the small bones in your ear which vibrate at or close to the same rate your vocal chords did. These vibrations are turned into signals that are read by your brain and interpret the sound that was originally generated. This concludes the creation and reception of sound. Simple, and yet so amazing, isn't it?!

I-B-2: WHAT IS A FREQUENCY?: We mentioned earlier that the pitch or how high/low a voice or sound was dependent on how fast the sound source vibrated, thus producing either large sound waves (vibrations) for low sounds, and small or fast sound waves for higher sounds. Let's explore this a little closer. In the world of music, different notes of the scale are referred to as a "pitch." When dealing with sound, different pitches are referred to as "frequencies." A frequency is nothing more than terminology given to sound waves. For example, a bass guitar produces low frequencies, a human speaking voice is made up primarily of mid- frequencies, and a piccolo emits high frequencies. All of these refer to the size of wave that is produced by the given source.

I-B-3: REFLECTION AND ABSORPTION: Think back to our water comparison. What happens when you're washing your car and you spray a window or the side of the car head-on? It comes back and gets you wet, creating great discomfort! Sound reacts in the same way when it comes in contact with reflective surfaces, such as cement, metal, glass and hard plastic. These surfaces act as a spring board for the sound, resulting in the uncontrolled dispersion of the frequencies. For special effects in a controlled situation this is fine, but it can be very annoying when trying to control a sound source. These characteristics are the basis for "feedback," that bothersome sound, usually in the form of a screech, that comes about when a microphone is held in front of a speaker, or a microphone is at too high of a level. Do some practical application. Go into an empty room that has a cement or tile floor and concrete walls (a tile bathroom in a commercial building works well). Clap your hands together and listen for the result. You should hear an echoing effect, which is the result of the frequencies produced by your hands, and then bouncing off of the reflective surfaces. The more reflective the surface, the longer and brighter the echo. This is referred to as a "live" or "bright" room because of the reflective characteristics, and is the basis for "reverb," which will be discussed later in this manual. In contrast, find a room with carpeted floors, low ceilings with acoustical tile, and filled with people. Perform the same test. The result? Other than strange looks from those around you, you'll hear the original clap of your hands and little, if any additional reverberations. All of the elements in the room absorb sound opposed to reflecting it. Materials that have deadening qualities include cloth curtains, unfinished wood, and foam. The principle behind the theory is that when sound comes in contact with a porous material, the sound is absorbed into the pores and is trapped. This is why acoustical tile has so many little holes in it. A room that is equipped with materials such as these mentioned has reflective characteristics that make a room "dead" or "dark." This designates little or no reflection. Further detail will be discussed in room ambiance and speaker placement.

II. BASIC OPERATIONS: Let's recap. By now you know that: sound is a vibration referred to as a wave; travels through all types of substances; a series of waves is called a frequency; and sound is reflected and absorbed by certain surfaces. You should also have a general understanding of what a sound system does, in addition to seeing the importance of proper sound reinforcement and the effect it has on the listener. My, what wisdom you have attained! In this section, BASIC OPERATIONS, you will be taken to the next step in developing solid, sound engineering skills. Areas that will be discussed includes how a microphone works, a more detailed look at the elements that make up a sound system, sound equalization, the measurement of sound, and an aesthetical angle on an otherwise technical subject. We recommend that you read all of the sections, as reference will be made to them in the upcoming portions of this manual.

II-A: MICROPHONE APPLICATION
II-A-1: HOW A MICROPHONE WORKS: In this section we'll learn how sound is transformed (transduced) into a "signal," which is simply an electronic impulse that is a reproduction of the sound that initiated it. This is the basic purpose of a microphone, which is also referred to as a transducer. Let's start from the beginning.

II-A-1-a: THE DYNAMIC MICROPHONE: When a sound source (a human voice, for example) is applied to a microphone, the vibrations from the sound source causes a diaphragm in the mic to vibrate at the same speed. The diaphragm is a sensitive object that reacts to sound waves that are directed toward it. This diaphragm is connected to a coil of wire that, when passed through the magnetic field caused by the two magnets surrounding the microphone, causes a low-level electrical current known as a "signal." Since the coil is connected to the diaphragm, it passes through the magnetic field at the same rate the diaphragm moves. This signal is now an electrical replica of the sound source that initiated it. This is the basis of a dynamic microphone. You will find that the dynamic microphone is the most common type of mic due to its reasonable price and its durability. It's commonly used in singing, speaking and instrument micing applications.

II-A-1-b: THE CONDENSER MICROPHONE: A condenser microphone has the same purpose as a dynamic mic, that is to transduce the acoustical signal of the sound source into an electrical signal. But the process in doing it, along with the application of a condenser differs greatly from that of a dynamic mic. A dynamic mic is dependent on the force put forth by the sound source to move the diaphragm, which in turn creates the electrical signal. A condenser mic, however, is assisted by an additional power source (that can be in the form of a battery or low-voltage current sent from the mixing console, known as "phantom power") that makes it more sensitive to subtle sounds. This creates a microphone that will respond quicker and in a more accurate fashion to softer and less noticeable sounds. With this added sensitivity comes an increased danger of damaging the delicate parts. For this reason, condenser mics are used in applications where durability, humidity and high sound levels are not present.

II-A-2: CHOOSING THE PROPER MICROPHONE: Too often we see microphones used in applications they were never designed to be used in. For example, a dynamic mic used as an overhead mic on a choir, or a condenser used on a loud, raspy guitar. Selecting the proper mic for the proper application is imperative to protect your equipment as well as get the sound you desire. Following are some helpful hints that will assist you in this selection. We will go into more detail as the course continues, giving you more insight as to the specifics of mic selection in usage as well as in purchasing.

II-A-2-a: SOUND PRESSURE LEVEL (S.P.L.) Aaaaaaaack! A technical term! Now don't wig-out on us. We hate to do it, but it is necessary to introduce terminology that you're not familiar with, and fortunate for all of us this just happens to be one of the simpler ones. Sound Pressure Level (S.P.L.) refers to the "volume" or "level" a sound is being disbursed. For example, a jet engine during take-off has a high S.P.L., where a flute has a low S.P.L. This, factored with the frequency that is coming from the source that is being miced, are the basics for choosing the proper mic. There, now wasn't that painless? Below is a general rule that you can apply when deciding whether to use a dynamic mic or a condenser. This isn't set in stone, and can be differentiated from as needed. NOTE: Rule of thumb: When HIGH S.P.L.'s are present, use a dynamic mic, and when LOW S.P.L.'s are present, a condenser mic is recommended. As we stated earlier in this section, condenser microphones are more sensitive to outside elements, and therefore have to be treated more carefully than a dynamic microphone. This is also true in reference to the S.P.L., as indicated by the above diagram. A high S.P.L. will damage the sensitive plates that are found in a condenser mic, and decrease its effectiveness in its proper application. Now, that doesn't mean you can use a dynamic mic to pound nails, although sometimes that's all certain ones are good for! And it doesn't mean you have to provide a feather bed for the condenser--just use common sense.

II-A-2-b: FREQUENCY RESPONSE: C'mon guys, two frightening terms in one section! Yes, you guessed it! Frequency response is the second main factor in choosing the proper microphone. And like S.P.L., won't seem so threatening after you've read this section. In fact, you'll find yourself using it the next time you and some friends get together--won't they think you have some "SOUND SENSE." Alright, on with the show. Think back to Section One when we talked about the characteristics of sound (Section I-B-2). We stated that a frequency is nothing more than terminology given to a sound wave and refers to its "speed" or "rate" (high frequency = small and fast waves; low frequency = large and slow waves). When selecting the proper mic, remember that manufacturers of microphones design their products based on its response to certain frequencies, or their "Frequency Response." Example: A microphone that is designed for vocals would respond better to frequencies (sound waves) that are produced by a human voice rather than the frequencies produced by the bass drum of a drum set. That doesn't mean that it won't respond to frequencies other than those it was designed for, but rather it won't be as efficient. To determine the frequency response needed for each application, further detail must be given to what frequencies are being produced by the sound source. And that can wait until later in this manual (thank you!:).

II-A-3: MICROPHONE ALTERNATIVES: The most basic microphone alternative is, of course, yelling. But what a drag! So other designs were considered and alternatives were invented. Remembering back earlier in this section, we described a microphone as a "transducer," changing acoustical energy into electrical energy. The alternatives we will discuss in this portion of the manual will be the basics and will briefly inform you of the options you have at your disposal.

II-A-3-a: DIRECT BOX Without getting into impedance and line-level vs. mic level at this time, the best way to describe a direct box is that it takes a signal from a source (usually a keyboard or some other electronic sound source) and converts it into a signal that is formatted differently. The instrument it converts it from is generally a single-pin connector known as a "phono plug," and it converts it into a three-pin plug known as an "XLR" connector. Chew on that for now and we'll clarify it for you later. How a direct box benefits you is that it cuts down on the amount of microphones you have, which eliminates feedback possibilities; is usually less expensive; and gives a stronger, more complete signal than a microphone would due to the energy lost when the signal goes from the source, say a bass guitar amp, to the microphone. The drawbacks of a direct box vs. a mic is that it doesn't allow for any ambient characteristics that would be present due to the signal having to travel through the air before reaching the microphone, in addition to the reflected sound that would be coming into the mic from around the room. This leaves you with what is referred to as a "dry" signal, meaning no part of the signal has been effected by any outside forces.

II-A-3-b: ACOUSTIC PICKUP (comes in two-wheel or four-wheel drive): Working again off of the vibrations generated by the sound source (usually the strings of a guitar in this case), a "pickup" has sensors that interpret the vibrations much like all other transducers. The difference being that a pickup is designed for the frequencies and S.P.L.'s generated by guitars. This allows a guitarist to simply plug in the guitar to the amp, bypassing the need for a microphone. It also gives the guitarist the ability to alter the pickup's capabilities, thus creating sounds such as distortion, a hollow sound, a clean sound, a thin sound, etc.

II-A-3-c: CONTACT PICKUP Rather than detecting the sound vibrations via the air, "contact pickups" detect the sound waves from a solid substance. In sound reinforcement, they are used almost exclusively for instruments such as an acoustical guitar, piano, or other instruments that have resonant factors (are good conductors of sound). Because they are not dependent on the air to carry the sound waves, they have a low feedback level and therefore work well in live applications. The drawbacks include the varied sound received due to where the contact pickup is placed and the type of material the contact is placed on, and that you never get pure sound quality because the signal has to travel through a substance rather than the atmosphere. However they do have distinctive qualities about them, and are therefore preferred by certain artists and engineers who strive for a certain sound.

II-B: MIXER BASICS The mixer or "Headquarters" is the element in the sound system that most people find the most intimidating. It never fails that at least once every performance a person will come back to the mixer and ask, "How can you memorize all of those knobs?" If that's your line, this section will take some of the fear and mystery out of the operation and understanding of the mixing console.

II-B-1: MIXER/BLENDER: WHAT'S THE DIFFERENCE? Despite some people's initial reaction to the word "mixer" mentioned in connection with a sound system, the word has nothing to do with food preparation, at least not in this manual. A sound system's mixer (or "console," or "board," depending on what dialect you speak) does, however, perform a similar function in that it blends (and just for the record, sound reinforcement mixers are never referred to as blenders. What is this, Home Economics?) different ingredients, in this case sound sources, into one palatable product. Without at least a rudimentary mixer, it would not be possible to have any kind of sound system. For this reason we can say the mixer is at the heart of every sound system.

II-B-2: SO WHAT DOES IT REALLY DO? In most applications, there are more than two sound sources active at one time, or at the bare minimum, two separate events will need to take place at different times (e.g., a person speaking and a singer at different times in the program). This creates the need for two microphones. Even if one microphone was used there are still advantages of going through a mixer. In other words, we can generalize and say that everyone that needs their sound reinforced needs some sort of mixer. It's all a matter of control. Now that we've established the need, let's take a closer look at what it does.

II-B-2-a: Back in section I, we gave an illustration of a basic sound system that was mixing multiple sources together. We could make an attempt to bypass the need for a mixer and try tying all of the different sources together (mics and instruments) and connecting this combined signal to an amplifier, and then to our speakers. This would obviously cut costs, but would also cause numerous other problems, the most important being that it would eliminate our job!, in addition to impedance(1) mismatching (which creates noise problems (and not just a little either)), level(2) mismatching (which will make some sources very loud and others very quiet, and eliminate any type of control short of adjusting the overall volume on the amplifier), grounding problems (which will make for a very audible and annoying hum and buzz that WON'T go away), and insufficient signal to the amplifier (which will cause you to overwork the amplifier causing it to burn out or trigger its automatic, protective shut-down, giving you substantial amounts of hiss). These are the very problems that a mixer solves.

II-B-2-b: From the very beginning, each sound source is unique. Most microphones put our a very weak signal compared to that of an instrument. Even with the same microphone, the signal level will vary if someone is speaking into it rather than singing. Even if it is used for just singing, two different singers in the same mic can sound completely different due to their individual tonal qualities and vocal strength. For this reason, when we want to blend these sources together, it is not possible to do so without being able to control each signal individually. The more control you have over each signal, the better you can provide the best possible final product. There is no limit to how many channels you can have, as manufacturers make consoles ranging from 4 channels to 48 and beyond. The size of the console should be based on your need and how many sound sources you will need to mix together.

II-B-3: WHY ALL THE KNOBS AND JACKS? Just as features and quality workmanship separate an expensive television or automobile from less expensive models, so do features and quality separate mixers, and all sound equipment for that matter. In general, the more features, the more flexible the mixer, and so it can be used for a broader range of applications. More features should mean more ways to modify (and hopefully improve) the end product. As we look at the different features that are included on the majority of good quality mixers, we will be able to see how more options can be beneficial and not just confusing.

II-B-3-a: SIGNAL PROCESSING As a signal passes through the cable and enters the mixer, it goes through a definite pathway, encountering different types of circuits. We call this "SIGNAL PROCESSING".

II-C: EQUALIZATION When we refer to the term "equalization," we are referring to the "shaping" of a signal through electronics. Maybe a better term would be "compensation," as equalization (or E.Q.) is simply boosting or cutting certain frequencies of a signal to accommodate the acoustics of the environment the signal is being sent into. For instance, if a room's acoustics has the tendency, let's say because of the way it is shaped, to absorb frequencies between 1000Hz and 1500Hz, there has to be compensation to make these frequencies the same level as the others. This is done through an E.Q.. Without an E.Q., the frequencies that are either inhibited or have been expanded due to the room characteristics are not at a level consistent with the frequencies that don't need compensated, and therefore the sound is out of balance and displeasing to the ear. The problems faced with equalization is not whether or not it is needed, but rather understanding how it works, how to apply it, and what types and variations are available. Equalization is an important tool in developing a complete and well-rounded sound, whether it's the E.Q. for the entire system, or simply the E.Q. for an individual channel. Becoming proficient at operating E.Q. will propel your career faster than any other one skill, due to its necessity in properly duplicating and reinforcing a sound source.

II-C-1: WHY E.Q.? The term "equalization" stems from the original use of this type of circuitry to boost certain frequencies to make up for losses over long lengths of cable. The term still illustrates the purpose of such devices well, as we will see a little later. Although leaving an E.Q. out of your system will not render your system inoperable, its omission will definitely degrade the performance of your system and cause you many unnecessary headaches. In other words: if you have a sound system, you need an E.Q..

II-C-2: HOW IT WORKS: An E.Q. functions by cutting or boosting the level of different frequencies across the sound spectrum. As a frequency is boosted or cut, that part of the sound passing through the system is either increased or decreased, respectively. Each control is assigned a central frequency, and is often times marked with the frequency it represents. If there are no markings, refer to the manual specifications. We have briefly discussed channel E.Q., so here we'll go over graphic E.Q., which is designed to give added control in shaping the signal. The typical frequency range of a graphic E.Q. goes from a low of 30Hz to a high of 20kHz, which covers the full range of human hearing. For home or automobile stereo listening, most E.Q.'s are between 2 and 9 "bands" (a band is made up of a central frequency and its lower and higher counterparts). This yields enough control over the sound to make listening more pleasing, generally by cutting the midrange frequencies and boosting the low and high frequencies. This works because midrange frequencies are more easily amplified and carry better over distance than high and low frequencies. Remember that and you will go far. This amount of control is adequate when listening to prerecorded music such as a tape or a C.D., but won't cut it in a live application where sound reinforcement is needed. The ideal E.Q. for most live situations is a "1/3 octave" (27 band) graphic E.Q. with a 12dB boost/cut capability. "What's this mean?" you say. As we've already mentioned, frequencies can be assigned note values, and as notes go higher and lower they experience harmonic convergence (no crystals required!) at evenly spaced points, which are called "octaves" (e.g., the span between "middle C" and the "C" above it on a piano is one octave, so the range of human hearing covers approximately 9 octaves). So, "1/3 octave" means, "The band centers are placed every 1/3 octave across a 9 octave range, which takes 27 bands." The word "graphic" E.Q. refers to the type of control used to boost and cut the frequencies. Graphic E.Q.'s are usually vertically-mounted sliders placed adjacent to each other to control the sound, thus as frequencies are boosted or cut, the frequency response of the E.Q. is displayed as a graph, or "graphically."

II-C-3: WHY SO MANY BANDS? When less bands are used, it is possible that a frequency you need to adjust to make a problem go away or just to make the system sound better will not be available in a sufficiently narrow band to be effective, as you will be adjusting other frequencies along with the one that's creating the problem. A 1/3 octave E.Q. is sufficiently narrow for all but the most extreme situations. If more are used, say a 1/6 octave E.Q., there are now twice as many adjustments and the bandwidth may now be too narrow. A bandwidth that is too narrow can result in an inconsistent sound for your system, as humidity and temperature affect the response of the room your are trying to cover.

II-E: AMPLIFIERS Up to this point, we have been working with a signal that has very little power or "volts." This low voltage was initiated by the microphone, assisted by the mixer and its components, and is now being sent on its final leg of the tour. The last stop before the speakers is the amplifier, a boosting station that takes the signal and puts power behind it, giving it the energy to activate the speakers and hurl itself toward the listeners. The output of the amplifier is measured in "watts," and will range from 50 watts on up to 1,000 watts on the average. Smaller and larger amplifiers are available, but this range is common in small to medium sized sound reinforcement systems. To explain in detail how an amplifier electronically boosts the signal, we would have to get into physics, Ohm's Law, mathematical calculations and electrical diagrams. If this interests you, a section titled "Additional Reading" at the end of the manual will benefit you. For this manual, however, we will remain with a simplified look at the purpose and application of amplifiers.

II-E-1: SIZING THE AMPLIFIER An amplifier that will adequately fill a 20'x30' room with sound won't cut it in a gymnasium filled with screaming kids. The power output won't sufficiently cover the room with sound due to the ratio of power compared to the cubic footage of the room. Without the proper ratio, the sound dissipates before it reaches the listeners. Figure # gives a formula that can be used to get a starting point when sizing an amplifier to a given room. For example sake, let's give value to each of the elements involved. FORMULA: 1 watt of power for every 180 cubic feet of empty room. ROOM = 50'W x 100'L x 15'H = 75,000 cubic feet. AMPLIFIER OUTPUT NEEDED: 417 watts of power This would provide the minimum amount of power needed to adequately fill an empty room with sound. But what good is that if there aren't any people to listen? Because of the sound absorption factors involved when you introduce items such as furniture, carpet, wall coverings, people, etc., an adjustment must be made to compensate. A good rule of thumb is 1 additional watt for every 5 people in the audience. This again will only establish a starting point to work from.

II-G: SOUND MEASUREMENT: The measurement of sound is approached from three different angles. The first deals with the measurement of "volume" or "intensity," which is measured in "decibel" or "dB." The second is a measurement of the characteristics of the signal and deals with the electrical current that a signal produces. This unit of measurement is called a "Hertz" or "Hz." These two terms deal with the acoustical properties of sound. Referring back to our previous sections though, we know that after a microphone converts a signal from acoustical energy into electrical energy, a low-voltage current is generated. This type of measurement is called a "Voltage Unit" or "V.U.." Measures the amount of electrical signal that passes though the system, and is not affected by the frequency, but rather is in direct response to the strength of the signal. Let's look at them individually.

II-G-1: THE DECIBEL As we stated, a decibel is a measurement of the "volume" or "intensity" of sound. To explain the mechanics of a decibel, we would have to get into logarithms, Bels and boring mathematical equations. This wouldn't be fun, so we won't do it! Instead, we'll try to confuse you with some basic principles that will help you begin to understand the simple yet abstract idea of decibels.

II-G-1-a: Decibels are based on ratios, so they don't increase/decrease in set increments. For instance, 1 watt of power is equal to 0dB, 10 watts is equal to 10dB, 100 watts is equal to 20 dB, and 2,000 watts is equal to 33 dB. WHAAAAAT?

II-G-1-b: A 3 dB gain is hardly noticeable, but a 10 dB gain appears to have doubled the volume.

II-G-1-c: Below is a chart that will help you get a feel for how many decibels ordinary sounds produce. <> Don't be intimidated by this measurement. I've found the best way to deal with it is accept it, and move on. If you are intrigued by the concept of sound measurement, there is some recommended reading at the end of this manual. Check it out.

II-G-2: THE HERTZ We know that sounds are made up of a series of vibrations known as waves. We're also aware that small waves or frequencies produce a high sound, and large waves or frequencies produce low sounds. A "Hertz" or Hz" is simply another name for a wave. To be specific, a Hertz equals one cycle (wave) per second. For reference this large of a wave produces such a low sound that it is not possible to hear with the human ear. Below is a cycle, which is also a wave, and now a Hertz (you know, like the Trinity, three in one). DIAGRAM If a Hertz is a cycle or a wave, then a frequency must be made up of a lot of Hertz. RIGHT-O!!! For instance, a high soprano voice ranges from 250Hz to 1,200Hz, meaning the sound waves produced by the vocal chord sends out up to 1,200 waves/second. Now that's some quick vibrating! Understanding the concept of frequencies will help you in the upcoming sections. It will assist you in visualizing acoustics, troubleshooting, equalization, and a host of other skills needed to improve your SOUND SENSE.

II-G-3: V.U./V.U. METER: The electrical signal that is generated from a transformed acoustical signal is monitored by a "Voltage Unit Meter" or "V.U. Meter". A V.U. Meter gives reference to how much signal is passing through the component that is being monitored by the meter. The amount of acoustical energy that is generated by the sound source is directly relative to the amount of electrical signal (see section II-A-1). Therefore the stronger the signal, the more current generated, and in turn the more response indicated on the V.U. Meter. In theory, the optimum position of the needle on a V.U. Meter is at "0." This signifies that the signal is passing through the meter at its most efficient level. Anything below "0" V.U. produces a signal that includes "noise," and above "0" V.U. is cramming too much signal into a limited area (the sound board), and will produce distortion.

II-H: CABLES AND CONNECTORS: In any networked system there are the lifelines, the very items that won't allow it to function unless adequately supplied. A heart relies on the arteries supplying it; an appliance on the electrical wires running through the walls; a steam heater on the maze of pipes and boilers supplying it; etc. The lifelines in a sound reinforcement system are no different, but seem to be neglected or not considered at all when putting together a system. Without quality cabling and connectors, a sound system will lack the signal needed to properly produce the sounds generated by the sound source.

II-H-1: THE PART THEY PLAY In any sound system, some type of cabling is required to make the system work. The most expensive and technical equipment is rendered useless when connected by improper or faulty wiring. Though these are the simplest parts of a sound system, their importance is equal to that of any microphone, mixer, or amplifier. We will now look at the two basic elements that make up any cable or cord: 1.) The connector(s) and 2.) The wire itself.

II-H-2: THE CONNECTORS Through standardization of the industry, there are a small number of connector types that all equipment manufacturers use in order to make their equipment compatible with other equipment used for the same purpose, such as sound reinforcement. This reduces the effort on our part in trying to match some obscure connector to a mic or amp. The vast majority of all connectors used in sound systems today are one of three types: The XLR (three pin) connector; 2.) The Phono Plug; 3.) The Phono Pin or RCA Plug.

II-H-2-a: XLR The XLR connector (also known as a "mic plug" and "A3M" or "A3F" connectors) is found on most higher quality equipment and on almost all quality microphones, which makes it a common sight, and for good reason. The XLR connector provides us with many advantages over any other type of connector. When wired properly, the XLR connector has 3 separate wires (or conductors) attached to it: signal, ground, and shield. The signal wire carries the electrical energy that is supplying the sound system; the ground wire completes the circuit by giving the excess electricity someplace to go; and the shield wire absorbs interference electricity from outside sources (a light's dimmer switch, fluorescent lights, radio stations, etc.). This is called a "balanced" circuit(1), which are typically very quiet. In addition to its wiring advantages, the construction of the XLR connector allows it to lock into place so that the connection cannot be broken by an inadvertent pull of the cable, and to be released by simply pushing a tab. It also is constructed to connect the shield wires together first which grounds out the circuit before the audio connection is made, thus eliminating potentially harmful and always irritating "pops" when making a live connection. Although these connectors are more expensive than other audio connectors ($3-$8 ea.), their advantages are well worth the extra cost.

II-H-2-b: PHONO PLUG Our next type of connector, the phono plug, is probably the most used connector in audio reinforcement systems. These connectors can be either 2 conductor or 3 conductor for unbalanced or balanced(1) connections, respectively. Phono plugs are popular because of their price, simplicity and ease of use. They come in various sizes, the most common being the 1/4" and 1/8" diameter plugs. These connectors are easy to attach to a cable and, if properly constructed are quite sturdy and dependable. Although slightly more convenient to connect and disconnect, the phono plug will make quite a noise when this is done to a live connection, and therefore the volume should always be turned down to a minimum level before taking such action. Drawbacks to phono plugs are first that the female connection relies upon a spring-loaded piece of metal to establish the connection, which sometimes loses its spring and weakens the connection, and second, the phono plug isn't secured into its receptacle and can easily be pulled out accidentally, resulting in a potential "pop" and break in the signal-flow.

II-H-2-c: PHONO PIN OR RCA PLUG Phono pins or RCA plugs are found on all types of tape decks, some mixers, and a few types of outboard gear(2). Though common, these connectors are more suited for studio and permanent applications, as they are not as sturdy as the XLR or phono plugs. As with phono plugs, phono pins do not ground out the circuit first, so watch out for noise when connecting or disconnecting a live one. In addition to the drawbacks mentioned in II-H-2-B, phono pins have less of a surface area because of their smaller size, and therefore the connection has more of a chance to not make an adequate connection.

II-H-2-d: OTHER OPTIONS Other types of connectors are the "Banana Plug," which is mainly used for speaker connections at the amp, and bare wires that are screwed to terminals, which are also used primarily for speakers. Though less common, these connectors are just as effective as any other when properly wired.

II-H-3: THE WIRE ITSELF Variety is rampant in the cable industry. Cables of every size, shape, strength and capacity are manufactured, and choosing the right one for your application can be quite confusing. When considering a cable, the connector type is a given, as your equipment will only accept certain types. What determines the quality of the cable over and above the connector quality is the quality and how appropriate the wire is between the connectors. If a wire is too thin, too thick, too stiff, too flimsy, or has the wrong number of conductors or insufficient shielding, you could be in trouble. Make sure the wire you buy fits the application you need it for. If you need a cable for a 3-conductor mic, you need a 3-conductor wire; if you have noise sources close by, you'll want extra shielding; and if the cable will be moved around, it needs to be flexible and well insulated to keep the individual conductors from being damaged. This example shows that care should be taken when selecting a cable. When making a purchase, ask the salesperson about the difference between cables and how the wire differs due to brand and grade. He or she should have sufficient product knowledge to help make the correct choice.

II-J: DEVELOPING YOUR EAR Many have told me that this is an area that can't be taught, and to be truthful, I'd have to agree. Teaching someone how to listen and what to listen for can be described and pointers can be given, but the ability only comes from hours of practical training, like any other skill. However, this manual is designed to develop skills in all aspects of sound reinforcement, which includes the aesthetical skills such as the one we are dealing with in this section. In fact, it is my feeling that too much emphasis is put on knowing and understanding the technical part of sound reinforcement, such as the specifications, mechanics and inner-workings of sound systems and not enough time spent on what is needed to properly blend together all of the ingredients that are introduced into a sound system. This is where a discerning ear comes into play.

II-J-1: WHAT TO LISTEN FOR One of the drills I find to be fun when I drive in my car (of course, before my stereo was stolen) is to listen to music and try to determine what each of the instruments is doing. I attempt to hum the bass guitar line, distinguish how many keyboard sounds and guitar parts there are, listen for how many vocal parts there are, etc. Separating these parts out helps me to recognize the importance that each part has, in addition to understanding how to blend these individual sounds into one end product. This exercise develops a basic sense of how loud a "hi-hat" symbol on a drum set is in comparison to a snare drum; the lead guitar to the rhythm guitar; the vocals to the instruments; etc. When a mix is out of balance, listening will be uncomfortable, and despite how good the talent on stage might be, the majority of the listeners will assume the problem is the musicians and not the sound engineer. From past experiences, I have found that it doesn't matter how good the equipment is, because it all comes down to how well the engineer mixes it all together. Without a discerning ear, the end result is chaotic and lacks structure, which are two items any listener wound find annoying.

II-J-2: RECOGNIZING AND SOLVING A PROBLEM A common problem that is run into when engineering a sound reinforcement system is having a mix that "just isn't right, but can't put my finger on what it is that's wrong." For some unknown reason, things don't sound like they should. This first place to start is the volume. It seems a tendency is that if something is wrong, start adding ingredients. Either more guitar, louder vocals, more high E.Q., etc. The correct response, however, is not to add but rather subtract. When confronted with a mixing challenge, follow the following three-step process: Step #1: Decrease the overall volume. This will bring everything down to a comfortable volume, lessening the S.P.L. and enabling your ears to be more responsive to problem areas and changes that need to be made. At lower volumes you will be able to hear subtleties that weren't clear at higher volumes, due to your ears ability to vibrate more freely and fully. Decreasing the overall volume will also help eliminate the possibility of feedback in the system due to microphonos that are too hot. Step #2: Single out the sound sources. Now that you have a lower, overall volume, start bringing up each channel individually, listening for problem areas such as a signal that is too bright or abrasive; too much low end, making the signal muddy or unintelligible; too hot of an input level, producing distortion and clipping; not enough input signal, which creates "noise;" doesn't sound like it should (e.g., a snare that sounds like you mother's sauce pan, an acoustical guitar that sounds like a mandolin, etc.); feedback present at low levels; etc. Many of these symptoms can be easily corrected by applying the techniques taught in section II-B, II-C and II-D of this manual. Step #3: Remix your sound sources. Start by mixing related groups of channels. For instance, combine all of the drum channels so the drums are balanced. Next, add the instruments, starting with the bass guitar, then adding guitar, piano, keyboards, brass, woodwinds and strings. Finally add the vocals, both lead and back up. As you put all of these together, continue to listen for the separation. If any one instrument is out of balance, refer back to the basics and decrease it instead of increasing the others to compensate. Once you've established a comfortable mix at this lower, overall volume, you can now start to bring up the master channel. Be aware, though, that as you increase the volume, certain frequencies will stand out above others. However, because you have identified each sound and are familiar with how the channels sound individually, you'll be able to quickly recognize and remedy any problem areas. NOTE: The true measure of a sound reinforcement engineer (you!) is the ability to produce a quality mix at a low volume. Get this right, and you're on your way.

II-J-3: OTHER EXERCISES IN EAR TRAINING Following are two suggestions that have helped us in the past, and could very easily assist you in developing your skills. Ear training is a slow process, so if you don't see incredible results immediately, don't be discouraged. You never graduate from the school of ear training; it's a lifetime process!

II-J-3-a: THE BAR ROOM BLITZ My father always said, "If you spend 1% of your time in bars, you'll find 99% of your problems." As true as that is, bars and night clubs can be some of the best classrooms for beginning and intermediate sound engineers. Why? Because when the majority of these establishments were built, the last consideration was the room's acoustics. This creates a unique learning environment due to the challenge of fighting the negative forces brought about by the room, in addition to the dealing with the "club-breed" of musicians, a complete challenge and learning experience in itself!!! Let's look at some examples of typical barroom scenes. A narrow, long room made of brick, with a high ceiling and adjacent wings. How do you get even, quality coverage at a comfortable volume? Or an under powered system in an oversized room. Try getting equal coverage to people on the dance floor as well as on balcony wings! It's a barroom nightmare! If the opportunity arises, or you can coerce a local sound engineer to let you tag along for a couple of gigs, it's great experience (volunteer to be a roadie for a road gig - - you'll almost always be welcome). Ask questions, but primarily watch and learn by taking notes and observing technique. A word of advice, though. If things start to go wrong, step back and offer advice only when asked, or you may find a microphone in a most precarious spot! And remember, 1% of your time, 99% of your problems.

II-J-3-b: SCHOOL? NO WAY! RELAX! It's not that bad. Junior colleges or universities offer classes in music appreciation, music theory, basic piano or guitar, vocal lessons, etc. If they're not available through the school, community associations often times offer something, and there are always private lessons or self taught courses. Correspondence and Internet curriculum is also helpful. Each one of these will broaden your skills, giving you a better foundation to draw from when performing your sound engineering duties.

II-J-4: PROTECT YOUR INVESTMENT After making such a large investment in your ears, and since they're the only ones you're getting, it's imperative to take good care of them. After all, they are tools you can't do without. With this in mind, avoid prolonged high S.P.L. environments (110 dB+); high pitches (e.g., sirens); and loud, sharp sounds (e.g., hammers hitting iron).

III: INTERMEDIATE OPERATIONS Throughout this section, we will be using the terms, principles, and techniques that were taught in the two previous sections. We will take a more in depth look at many of the items we've discussed, such as advanced E.Q. techniques, different methods in micing a sound source, and further developments in speaker selection and application, in addition to some new items that haven't been addressed yet. The glossary at the end of this manual will assist you in locating the sections you need to review if you come up against a term or skill that you find unfamiliar.

III-A: MICING THE SOURCE Microphone placement can be the deciding factor between a good sound vs. a great sound, or feedback vs. no feedback. Too often microphones are placed in the vicinity of the sound source, leaving it up to luck as to whether or not the mic's features are being utilized 100%. It is important to understand your microphone's characteristics and how to take advantage of its features. After reading this section, we encourage you to take some time and experiment with microphone placement and see what kind of a difference placement can make.

III-A-1: CHOOSING THE RIGHT MICROPHONE Before deciding where to place a mic when micing a sound source, you must choose the proper mic. A great mic in the wrong application will give undesirable results and can cause damage to your equipment. To choose the proper mic, we have to five you a crash course on "specification" or "spec" reading. Specs refer to the data the manufacturer puts out in regards to a component's performance and application capabilities, whether it be a mic, mixing console, outboard effect, etc.. Described in the specs o f a microphone is "frequency response," which we touched on in section II-A-2-B. We said that frequency response referred to haw a mic reacts to certain frequencies, and the level or "magnitude" of the signal from input to output. Frequency response is measured by charting and creating a curve It is plotted by feeding the signal processing gear a range of frequencies at a constant level. The signal processor is hooked up to a meter that monitors its reaction to the different frequencies (horizontal numbers). The most desirable curve for a general purpose mic is a "flat" response, meaning the mic responds equally to all frequencies, free of noticeable peaks and valleys. This give and even, smooth delivery in the microphone's performance. NOTE: Do not choose microphones, or any other sound gear for that matter, based solely on the specifications. Use them only as a starting point in equipment selection. Often times microphones will look identical on paper, but have considerable differences in their actual sound and performance results. In section II-A-2-B we showed what frequencies were produced by certain sound sources, establishing a reference point to assist you in recognizing what frequencies were being dealt with. By comparing the specification to the chart, we can match which mics will best serve in micing each corresponding sound source, based on the microphone's frequency response curve. We've now established the first step in micing a sound source.

III-A-2: CHARACTERISTICS AND VARIATIONS Referring back to section II-A, you'll remember we briefly discussed dynamic and condenser microphones and their specific applications. Condensers and dynamic mics are two categories that are most widely used, but even they have variations. An important difference that is characteristic among all mics is a "pickup pattern." This is terminology given to how a microphone "sees" a sound and reacts to it, and the direction the mic must be pointed to get optimum performance.

III-A-2-a: PICKUP PATTERNS The most common pickup pattern is called a "cardioid" pattern, called that because of its heart shaped design. A cardioid pattern rejects sound from the rear of the mic, has minimal pickup on the sides, and has the most efficient response "on axis." The axis is the imaginary centerline that would run from the base of the microphone through the center of the top of the microphone. Simply, "on axis" means head-on or directly in front of the microphone. This type of pattern is generally "tight" and directional, meaning a sound source must be within a few inches and be on axis to have the mic be effective. A tight pattern helps to eliminate feedback potential, and gives the sound engineer needed control be not having the mic pick up sound sources other than the one that's being miced. Another reason a cardioid mic is popular in sound reinforcement is because frequencies react differently when miced on and "off axis" (or with the microphone turned at an angle in reference to the sound source) due to its directional qualities.

III-A-2-b: OMNIDIRECTIONAL An omnidirectional pickup pattern is just as the name implies, picking up the signal equally from all directions. An immediate reaction to this pickup pattern in sound reinforcement is how susceptible it is to feedback and unwanted noise sources, of which bath are relevant. Omnidirectional microphones do, however, have better low frequency response and aren't as prone to breath and wind noise as is a cardioid pattern. Microphones that generally have omnidirectional patterns are lavaliers (mics clipped onto a shirt or tie, usually used in a lecture or interview format); overhead microphones, which are commonly used when micing choirs; certain singing and speaking microphones; and many types of recording microphones.

III-A-2-c: BI-DIRECTIONAL OR FIGURE 8 Although this pattern is not as common as the previous two, there are applications where it is quite useful. A bi-directional microphone picks up sounds equally from the front and back, which are on "on axis," and rejects the sound coming in from the sides. Applications that would benefit from this design includes an interview format, two singers who wish to face each other and don't want to have the crowd heard, or tom-toms on a drum set. It is limited by the fact that the two sound sources being miced can't be controlled individually. Bi-directional mics are generally dynamic, and have a tight and directional pickup pattern.

III-A-2-d: ON/OFF AXIS As we mentioned, a microphone's axis is the direction of application that will make the most efficient use of the microphone's features. When operating "on axis," the full dynamic range of the sound source will be picked up by the microphone. "On axis" gives the signal its highest amplitude or strength due to utilizing the complete surface area of the microphone's diaphragm. This decreases feedback potential and gives maximum gain at the mixing console. The drawbacks of "on axis" micing includes "boominess" due to maximum bass exposure, sibilance problems from the "S" consonant, and occasionally a damaged diaphragm from an excessive S.P.L.. "Off axis" micing gives increased highs, reduces sibilance problems, and picks up additional room ambiance for an added effect. However, because the "off axis" angling of the microphone makes it open to outside signals, feedback potential is increased.

III-A-3: DESIGN VARIATIONS A microphone's casing and design, along with its electronic components, are created with a certain application in mind, whether it be a vocal mic, instrument mic, podium mic, etc.. The following subsections will describe some of the options available, giving purpose to the designs and their functions.

III-A-3-a: HANDHELD As the name suggests, a handheld mic is designed to be held in the hand of the lecturer or performer. This provides mobility, proximity effects that can be dictated by the user, and gives liberty over limited mic placement. A handheld is usually a dynamic mic that utilizes a tight, cardioid pickup pattern, enabling the user to determine what sounds the mic picks up. Keeping the mic free of any vibrations or handling noises is important, thus using the proper mic clips and shock mounts is important. Using a rubber shock-mount is a common way to decrease vibration noise, and making sure that the protective screen or cover is in place will assure diaphragm protection.

III-A-3-b: LAVALIER Lavalier microphones are commonly used in television broadcast applications because of their inconspicuous size. A lavalier can be pinned directly to the clothing, hidden in a prop, or hung around the neck. Although lavaliers were originally dynamic mics designed with cardioid pickup patterns, many have now converted to condensers with omnidirectional pickup patterns. This has given increased response due to the mic being more sensitive; convenience due to the smaller size; and a more consistent signal because it remains at a consistent distance from the users mouth and is not affected by any proximity effect.

III-A-3-c: STAND-MOUNTED Although still available, stand-mounted mics aren't as popular as they were many years ago. Stand-mounted mics were used because the mics weren't as durable, weren't as practical to hold because of their size, and were more sensitive to handling noises. Stand-mounted mics are still made, but are used primarily for broadcast and recording purposes. They come with their special mic clip that usually has some kind of shock-mounting or has the clip built directly into the microphone's body.

III-A-4: SPECIALIZED APPLICATIONS Following are transducers that have limited but necessary applications. Each have characteristics that enable them to serve in specialized situations and fulfill needs that can't be met by microphones that have a wide variety of uses.

III-A-4-a: PRESSURE RESPONSE (PZM)

III-A-4-b: SHOTGUN A shotgun mic has a highly directional pickup pattern that can isolate a sound source from a distance. They are most often used in broadcasting and film work, picking up a selected source from a distance while rejecting other surrounding sounds. Although rarely used in reinforcement, shotgun mics can work when reinforcing dramatic presentations, choirs, and interviews with small children. Easy to use, all that is required is aiming the mic in the general direction of the sound you wish to reinforce. The effectiveness of the mic and the distance it will reach differs between manufacturers.

III-B: CONSOLE INPUTS AND ROUTING This section deals with the options a sound engineer has when plugging into an audio mixing console. Because each manufacturer designs their consoles a little different than the next, we'll be dealing with the most common inputs and their functions. Mixing consoles are commonly described by the number of inputs/outputs they have, thus giving both the consumer and the retailer a way to refer to the size of console that is required. For example, a console that has 16 inputs and 2 outputs (stereo) is referred to as a "16x2" console. Often times there are subgroups involved, so a "16x4x2" would refer to a console with 16 inputs, 4 subgroups, and 2 main outputs.

III-B-1: CHANNEL INPUT The channel input or "jack" is the receptacle where the signal enters the console via a microphone or instrument cable. This initiates the processing of the signal in the console and signal processing of the signal in the console and outboard gear before it is sent to the power amps. The two most common jacks you'll find for the input signal are female XLR (Balanced or Low Z) and 1/4 inch female plug (Unbalanced or Hi Z). Another input jack that is common to most pro-audio boards is an "insert" or "in/out" jack. Its primary functions is to insert an effect into the channel without going through the console's effects loop. It works usually on a 1/4 inch plug that has "tip, ring, sleeve," which gives a loop effect to the signal. This input works well when you want to insert an effect on an individual channel or a specific subgroup. The signal is sent to the signal processor via cabling, then put back in the same jack and back into the console. The diagram below shows the special cable that is needed to accomplish a channel insert.

III-B-2: SIGNAL ROUTING As we've already discovered, a signal goes through many components before it finally reaches the listener's ears. The path the signal takes comes as a function of routing, the directing of the signal to its destination. This is known as "signal flow," referring to the path the signal takes from the time it enters the transducers to the time it exits the speakers. In this section we'll be concentrating on the signal flow through the mixing console as well as the alternate paths the signal can take.

III-B-2-a: ELECTRONIC ROUTING In a mixing console, the signal flow is hardwired to take a designated path through the console to the main outputs. This path is setup to be the most effective and efficient use of the signal, giving the operator optimum control. Along the signal path are options that can be utilized by the operator, "sending" the signal to other signal processors. The used to assign the signal are called "sends" or "pots." These controls simply split the signal, sending it to be processed elsewhere. The primary send the signal encounters is the "auxiliary" send, which can be used to route the signal to stage monitors, effects, or any other type of outboard gear. The signal's strength is adjusted by the send, only allowing as much as the operator desires. Although some mixers don't have any sends, professional consoles have anywhere from 2 sends on up, and are referred to as "auxiliary," "monitor," "effect," "foldback," or any other term the manufacturer chooses. Once the signal is sent from the channel, it is routed internally to the master send, which is an attenuator that establishes the overall output level of the signal. The master send routes the signal to an output on the back of the console, where it can be manually patched to the recipient of choice. Since there's an output, common sense would have it that we have to reintroduce the affected signal back into the mix. If the send was used for stage monitors, the signal must find its way to the monitor amp. But if it was used for an effect, the signal has to be put back into the console where it can continue to be manipulated and eventually reintroduced into the mix. This is where the "return" control comes in and the options that accompany it. The "return" reintroduces the processed signal into the main mix, as we stated. For instance, if the operator sent a vocal signal to a reverb unit, the return would take the signal with the effect on it and put it back into the mix along with the original signal. The "return" is controlled by an attenuator, and therefore can be increased or decreased according to the operator's discretion. "Returns" are usually labeled as such, and have to be manually patched into from the signal processor that needs to be. A good rule of thumb as to how much effect there should be: The processed or "wet" signal should not be louder than the original or "dry" signal. An alternative method to patching into the effect or auxiliary return is to go directly into a channel input if one is available. A return pot usually doesn't have E.Q. as does an individual channel, therefore giving the operator more control as to how the effect sounds, in addition to being in a more convenient location. As we briefly mentioned, the only send that won't have a return is in the case of a stage monitor. This is because once a monitor signal is sent out of the mixer, it is no longer needed for anyone but the performers. However, like the auxiliary sends, the monitors have a master send that establishes the overall monitor level that is sent to the monitor amps.

III-B-3: PANNING, ASSIGNING, AND SUBGROUPING After a signal has been bounced around, equalized, and processed, it now must be routed to the appropriate output. This is done via a "pan" pot or button, which assigns the signal to the desired output. The most common assignment option is "left" or "right," referring to the master outputs. For example, if the channel is panned to the left output, the signal from that channel would be routed to the left master output and vice-versa. If neither the left or right outputs are selected, the signal is sent equally to both the left and right channels. In consoles that have subgroups, a channel can be assigned to one or all of the subgroups, and then the subgroup(s) assigned to the desired main or master output. Note: A subgroup often times has an output jack located on the back panel of the console if a direct line out is desired. Panning and subgrouping can be great tools in organizing the operation of a console. One method is when operating a console that has four subgroups, assigning all of the drums to sub #1, instruments to sub #2, vocals to sub #3, and effects to sub #4. This enables the engineer to increase or decrease the overall volume of a certain group without disturbing the settings that were established in the original mix. Learn to use panning and subgrouping in your favor to increase your engineering efficiency.

III-C. SETTING AN INPUT LEVEL Also known as "gain-staging," setting your channel's input level is the most critical step in ensuring a quality sound mix. As we've already mentioned, this is where the signal is introduced into the channel and consequently, the mixing console. Nowhere does the saying "garbage in, garbage out" better apply. Set a weak input and you'll introduce noise into your mix due to having to overcompensate the weak signal by boosting faders; set too strong of an input level, and you'll introduce distortion into your mix. Because the input level effects so many items down stream (see II-B-1), take your time on this step. The following three-step process will help you get off to a good start: Step #1: Determine the strength of the signal. A keyboard going through a direct box will usually have a stronger initial signal than a nylon string guitar being miced by a dynamic microphone; a strong soprano may be able to overpower a soft bass vocal. Using headphones for this step is recommended to prevent a strong signal from blowing out your speakers. Step #2: Set your fader at zero boost/cut or at about 75%. This gives you a starting place for how much signal you need to introduce into the mix. Take this one step further and set your subgroup masters and sum/main/master out at the same point. Step #3: Increase/decrease your gain based on your mixing needs. This will be a combination of your input level and the strength/weakness of the sound source. If you find yourself driving your gain past 3 o'clock or 75%, the problem is with the sound source. To correct this, have the sound source increase its/his/her volume and make sure you're using the appropriate microphone or transducer. NOTE: The V.U. meters are going to be for reference only when gain-staging your live mix. Because a mixing console "sums," that is adds, each mixer channel together, a "zero V.U." signal added to ten other "zero V.U." signals will produce a distorted "sum" or "master out." Your final result (theoretically) should be all of your channel, subgroup, and master faders at zero boost/cut (every fader about 3/4 the way up and level across the console), and an even mix. Ideally, your V.U. meters will be barely registering (if at all). Simple in theory, next to impossible in application. Don't beat yourself up if you can't get it right the first time, but keep trying - - this is integral.

III-C-1: WHAT AN INPUT LEVEL AFFECTS Now that we have sources for our inputs and have talked about connecting that signal to the board, we now need to know how to manipulate that signal (which we have worked so hard to obtain) in the most effective and efficient manner possible. When we use a supply of water for watering the lawn, the first factor we can control in the flow of the water is how much of that water we will allow to pass through the hose and nozzle. If we use too little water we will be operating inefficiently since we won't be able to cover as much ground as we would if we had the correct amount passing through our simple little system. And, by the same token, if we use too much water we might either damage the foliage we are trying to help out or spring a leak somewhere in the line (i.e., cause problems). It follows, then , that the same principles apply to an audio signal passing through our sound system. The input level control on your mixer (of which there are only a few types, which we will cover in a moment) acts as the main valve on your watering system, as it directly determines how much of the signal will make it to the rest of the system. If our input level is set too low, then we will have to compensate by running our channel fader level up much higher than would otherwise be necessary, creating hiss from trying to draw from an insufficient source. And, if our input level is too high, we will produce distortion in the system, due to feeding too much of a signal through a limited area. The setting you establish for your input level is the main factor in determining how efficient you will be at utilizing the signal you have assigned to that particular channel. When you adjust your input level, it affects every area of the system that utilizes that signal because that is where you introduce the signal to the system (and just as you must exercise caution when introducing one friend to another by not telling one too little about the other (leaving them with an unclear impression of the person) or by telling them too much (which may leave them nothing to talk about or with the wrong impression of each other)), so you must take some care in setting your input level.

III-E: ESTABLISHING A MONITOR LEVEL The monitor volume is the only level that is dictated by a person other than the engineer. The monitor mix enables the performers to hear what they are playing and singing. Rarely is the monitor mix similar to the mix coming out of the mains due to the performers desire to hear a mix that meets their needs opposed to the needs of the engineer. Factors that come into play when setting monitor levels are stage volume, monitor placement, channel input level, the number of monitors, available power, room acoustics, the number of monitor sends and the performer's preference. Let's look at these individually.

III-E-1: STAGE VOLUME The lower the stage volume the better, from an engineer's viewpoint that is. If the monitors are too loud, they reflect off of surrounding surfaces (usually the wall directly behind the performers) and are projected out into the audience. This can potentially create a muddy sound to the listeners due to the unintelligible sounds produced by the reflected monitor signal. In contrast, a quiet stage volume enables the performers to hear more subtleties due to lower S.P.L.'s, gives the engineer more control out in front due to less reflected sound, and decreases potential monitor feedback due to a lower overall gain.

III-E-2: MONITOR PLACEMENT Monitor location is important when expecting optimum performance. Directional qualities of the speakers must be considered as well. The primary purpose is to get even stage coverage without excessive volumes, and doing this with as few monitors as possible. Wedge-shaped floor monitors are the most common, but properly placed side-fills can help to fill-in the dead spots.

III-E-3: CHANNEL INPUT LEVEL As we have already established, the channel input sets the strength of the signal into each individual channel, which includes the available signal at the monitor send. Therefore, the increase or reduction of the input level will directly affect the monitor level. Keep in mind that if you attenuate the channel gain you must compensate the monitor send accordingly.

III-E-4: NUMBER OF MONITORS Ideally, each performer should have at least one monitor and preferably two. Unfortunately that is seldom the case when working on a budget. Monitor coverage is similar to the main speaker's coverage in that the purpose is to eliminate dead spots where certain sounds or frequencies aren't clearly heard. To attain maximum performance from your monitors: 1. Make sure each monitor is unobstructed (no lyric sheets placed on them, etc.). 2. Keep each musician in the "line of sight" of the monitor's horn, enabling maximum clarity. 3. Put similar instruments on the same monitor mix (e.g., bass w/ drums; acoustic w/ electric guitar; etc.). 4.) Minimize stage volume. The more sound you can give to the performers through their monitors, the more control you have on the house mix (less spill-over from the stage). Overall, here's a rule of thumb: monitors are primarily for pitch and keeping rhythm. Anything else in the monitor is for the musician's listening pleasure only. Although many musicians will argue with you on this one (claiming they need to hear a mix including everyone), stage and house acoustic problems or system limitations often present a situation where a compromise must be reached. A reminder that the purpose of the service/presentation/program is not for them (the musicians) but for the audience/congregation. All personal preferences MUST be relinquished to this objective.

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