CHAPTER

6   Audio/Sound

Introduction

This chapter explores the audio production techniques and equipment used to record and control high-quality sound and sound perspectives. Quality audio is extremely important in media production. Poor-quality sounds can destroy the impact of high-quality visuals. High-quality sounds not only enhance accompanying visuals, but they can directly affect emotions and develop additional creative dimensions and responses.

Some directors feel that sounds and visual images should be almost completely independent of one another so that each component could stand entirely on its own, whereas others feel that sounds should reinforce accompanying visual images. The former approach is consistent with modernist aesthetics, whereas the latter reflects a realist approach to production. Some directors combine these approaches and suggest that high-quality sound should function well on its own as well as in combination with visual images.

Except for signal processing, editing, and distribution, no substantial difference exists between analog and digital audio production techniques. Greater care must be taken in all stages of digital audio production because of the increased frequency response and lower level of noise. Digital distribution systems are covered in Chapter 1, Producing: Exploiting New Opportunities and Markets in the Digital Arena, and editing technologies are covered in , Editing.

AESTHETICS OF AUDIO/SOUND

Audio/sound can be approached from the three aesthetic perspectives of realism, modernism, and postmodernism. A realist approach uses sound to stimulate an illusion of reality, reinforcing the temporal and spatial continuity of visual images. Modernist audio develops sound independently of accompanying visual images, breaking down realist conventions and stimulating more abstract impressions and visceral feelings. Postmodernist audio emphasizes listener participation within productions in order to emotionally involve the audience as much as possible.

TYPES OF MICROPHONES

The ability to duplicate quality audio in film, video, and audio-only situations depends on careful mic selection and placement. This means choosing a mic designed for the specific purpose at hand and positioning it properly. A microphone is a type of transducer. Transducers are devices that change one form of energy to another form of energy. Mics convert analog sound wave action into analog fluctuations in electrical voltage. A digital signal must be created by converting the analog signal through an analog-to-digital converter. Sound is created by the rapid vibration of objects and sound waves consist of rapidly contracting and expanding particles of air. A tuning fork, for example, causes air molecules to compress and expand as it vibrates, creating a sound pressure wave. As one arm moves forward, it pushes the air molecules, and as it moves backward, the air molecules, which are elastic or resistant to being pushed, expand again to fill the partial vacuum or void. Rapid vibration creates a pressure wave of alternately compressed and expanded air molecules. This pressure or sound wave moves in a relatively straight line and strikes other objects, such as the human ear or a microphone element. The eardrum vibrates in response to the sound wave and produces an auditory impression in the mind. A mic has an element that is sensitive to these airwaves and converts the wave action into corresponding fluctuations in electrical current. The electrical signal thus becomes an analog, or copy, of the sound wave. Once an electronic equivalent of the audio signal has been created, that signal may be converted into a digital signal for recording, processing, and maintenance of the highest quality during duplication.

Mics can be classified on the basis of the type of transducer element they use into three basic categories: dynamic, ribbon, and condenser. One type of mic element may be better suited to a specific audio situation than another.

Transducer Elements

A dynamic mic consists of a moving coil attached to a vibrating diaphragm or disc suspended between two magnetic poles. As the diaphragm vibrates with the sound wave, the coil moves up and down within a magnetic field and changes the voltage of the electrical current flowing through the coil. In general, dynamic mics are very durable, not extremely susceptible to wind noise, and relatively inexpensive (Figure 6.1).

FIGURE 6.1 The three transducer elements now used in microphones are dynamic, ribbon, and condenser. Dynamic transducer involves the movement of a thin diaphragm moving a coil of wires wrapped around a permanent magnetic. A ribbon transducer involves a thin corrugated strip of metal moving between the poles of a permanent magnet. A condenser transducer involves two thin plates of metal moving within a static-charged capacitance field.

A ribbon mic contains a narrow strip of corrugated foil suspended in a magnetic field. This ribbon vibrates in response to the difference in air pressure in front and in back of it and produces an alternating current along the length of a coil. The ribbon itself is quite fragile and can easily be damaged by simply blowing into the mic, although newer ribbon mics have been designed to be more durable but still are best confined to studio use. A ribbon mic usually produces a smooth, bass-accentuated sound, is preferred by many radio and television announcers for that reason, and it is ideal for digital recording because its warm sound accentuates high frequencies. Most ribbon mics are priced at the top of the range of professional microphones.

Condenser mics are relatively complex, compared to dynamic or ribbon mics. The element is a capacitor that requires two charged plates: a diaphragm and a fixed back plate. As the diaphragm vibrates, the space between it and the fixed plate changes in capacitance, that is, in its ability to pass an electrical current or signal. The strength of the electrical sound signal increases or decreases accordingly. The signal is very weak, however, and a preamplifier is required to boost the signal to a usable level. Additional current may be supplied to the preamplifier by a battery in the mic handle or by a power supply located in the mixer called a phantom supply. An electret condenser mic is constructed with permanently charged plates, reducing the power needed to operate the mic and the need for a built-in preamplifier. Condenser mics vary in price from relatively inexpensive to quite expensive, and some inexpensive cameras and cassette recorders have built-in condenser mics of lesser quality. A condenser mic generally reproduces high-quality sound, and with its built-in preamp can be considered quite sensitive.

Pickup Patterns

Mics can be classified according to their directional sensitivity or pickup patterns, as well as their transducer elements. Different recording situations require the use of mics that pick up sounds from a very narrow or a very wide area. Some mics pick up sounds coming from every direction, whereas others are sensitive to a very restricted area. The three basic categories of pickup patterns are as follows: omnidirectional, bidirectional, and unidirectional or cardioid.

An omnidirectional mic is equally sensitive to sounds from all directions, that is, from the entire 360-degree area surrounding it. A bidirectional mic is sensitive to sounds coming from two opposite directions. Its sensitivity drops off rapidly at 60 degrees on either side of these two opposite directional points. At 90 degrees (perpendicular to the two optimal sound source directions), it is almost totally insensitive to sound. Unidirectional mics are sensitive to sounds from one direction only (Figure 6.2).

FIGURE 6.2 Three basic microphone pickup patterns of omnidirectional, bidirectional, and unidirectional may be modified or combined to create two additional patterns: cardioid or supercardioid or shotgun pattern.

A cardioid mic is a type of unidirectional mic so named because its pickup pattern is heart shaped. A cardioid mic is somewhat more sensitive to sound emanating from directly behind it, but it is very sensitive to sound coming from directly in front of it. A supercardioid mic is somewhat more sensitive to sound coming from the rear of the mic but has an even narrower optimal response area (about 60 degrees as opposed to 120 degrees for a cardioid mic). Shotgun mics are long, narrow tubes; they frequently have a supercardioid pickup pattern.

A multidirectional mic is constructed with more than one pickup head aligned to receive sound from different directions. Such a mic may contain as many as six heads placed for simultaneous recoding of 5.1 audio mounted in one fixture. The mic combination may be mounted directly on a camera or any other mic mounting system. The danger of using such a mic comes from any equipment noise near the mic being picked up by more than one head increasing noise levels (Figure 6.3).

FIGURE 6.3 Recording raw six channel audio simultaneously may be accomplished using a multidirectional mic, which contains at least six different heads pointed in six different directions to provide the various sources needed for 5.1 channel programming. (Courtesy of Holophone Surround Sound Microphone Systems.)

Impedance

A third characteristic of microphones that determines their use and placement is impedance. Impedance is a complex measurement of the property of wires and equipment that determines the ability of a signal to pass through that piece of equipment. It is critical that all audio equipment be designed to match input and output impedances. For microphones, there are two basic impedance choices, high or low.

High-impedance mics are low-cost amateur mics, whereas all professional mics are low impedance. High-impedance mics may be connected with wires that contain one conductor and a shield, which does not provide the maximum protection for the signal from outside interference. Low-impedance mics normally are connected with wires that contain two conductors and a shield, providing maximum protection for the signal.

MIC PLACEMENT AND SELECTION

Mic placement during recording can be either on-camera or off-camera. On-camera mics, such as a reporter’s handheld mic, are visible to the viewer. Off-camera mics are not visible to the viewer. An off-camera mic can be hidden somewhere on the set or under a speaker’s clothing, or it can be situated just outside the camera frame.

On-Camera Mics

Hand mics are the most common on-camera mics. Mics that are to be handheld should be shock mounted; that is, they should be well insulated so that noise is not created as the performer moves the mic. Because the performer can move and control a hand mic, it does not always stay in a fixed position, and it generally has a relatively wide pickup pattern, such as omnidirectional or cardioid. It is wise to use a mic with a durable element, such as a dynamic or an electret condenser mic, in a handheld situation. An inexperienced performer should be instructed in how to keep the hand mic at a relatively constant distance from his or her mouth in order to keep the loudness relatively constant. A problem that frequently arises with the use of a hand mic is controlling the mic cable. Performers must learn to move the mic around without stretching the cable or tangling it (Figure 6.4).

FIGURE 6.4 Hand microphones also may be mounted on a desk stand, floor stand, or boom, but they must be designed to fit comfortably in the hand with reasonable sensitivity. (Courtesy of Shure.)

Desk mics often have less durable elements than hand mics. If a desk mic is placed in a relatively permanent position, it does not have to be shock mounted. If a desk mic is to be removed from its mount and also function as a hand mic, as frequently occurs, it must have some of the same qualities as a hand mic. Most desk mics have cardioid pickup patterns and are placed one to two feet from the speaker. Sometimes a single bidirectional or omnidirectional desk mic can be used for two speakers to limit the number of mics needed (Figure 6.5).

FIGURE 6.5 A desk stand is designed to hold a microphone in position to pick up people seated at the desk. It usually works best for one person but can be placed between two people if their audio levels are nearly the same. (Courtesy of ElectroVoice USA.)

A stand mic is supported on an adjustable pole in front of the performer; thus, it offers a distinct advantage to a person who has his or her hands occupied with a musical instrument. The stand mic can usually be tilted and adjusted to a comfortable height for different performers. In general, more sensitive ribbon and condenser mics are used on a stand to record relatively soft sound sources such as stringed instruments, whereas dynamic mics with omnidirectional or cardioid reception patterns are often used for singers and amplified instruments. Sometimes more than one mic may be attached to a single stand: perhaps a condenser mic positioned from below to pick up the sounds of a guitar and a dynamic mic above to pick up the singer’s voice (Figure 6.6).

FIGURE 6.6 A stand mic is designed to allow one or more people to stand on each side of the microphone, depending on whether the microphone is bidirectional, unidirectional, or omnidirectional. (Courtesy of Sony Corporation.)

A lavaliere mic also leaves a performer’s hands free and does not require a stand that restricts his or her mobility. This type of mic, which is either hung around the performer’s neck with a strap or clipped to a tie or outer garment, is relatively unobtrusive compared with a desk mic or a stand mic. Care should be taken in the placement of a lavaliere mic to ensure that it will not create noise by rubbing against rough clothing or jewelry. Lavaliere mics are often susceptible to cable problems because their cables are relatively thin and fragile. To guard against this on live broadcasts, performers such as newscasters often wear two lavalieres clipped together to create a dualredundancy system, where one mic serves as a backup for the other. Only one mic at a time is live to prevent phasing problems, which are discussed later in this chapter. A lavaliere microphone can be hidden or concealed behind clothing, although this can lead to added rubbing noise (Figure 6.7).

FIGURE 6.7 A lavaliere microphone is designed to be clipped to the clothing of the person speaking. The design of a lavaliere compensates for the microphone resting against the chest of the speaker, and the microphone is located below the speaker’s mouth. A lavaliere should not be used as a handheld microphone away from the body. (Courtesy of Audio-Technica.)

Some handheld and lavaliere mics have battery-powered FM transmitters, which allow the speaker using the mic to move around quite freely without a restrictive mic cable. Wireless lavaliere mics can also be used as hidden mics by concealing the mic and its transmitter under clothing. An FM receiver at the audio input of the recording machine receives the transmitted signal. Although wireless mics can be extremely helpful in many difficult recording situations, they also have a number of pitfalls. Like any FM radio, the wireless receiver can pick up interfering signals, such as noise from CB radios. Batteries can expire in the middle of a recording, especially when performers forget to turn them off. Finally, wireless mics are more expensive to rent or purchase than wired mics (Figure 6.8).

FIGURE 6.8 A wireless microphone may be designed as either handheld, a lavaliere, or on a head mount. The transmitter may be built into the base of the handheld microphone, or the lavaliere and head mount may be connected to a small transmitter fastened to the body of the performer. (Courtesy of Audio-Technica.)

Off-Camera Mics

Off-camera mics may be attached to a mic boom. A mic boom is a long pole that can be placed (usually above the heads of the talent) just outside the camera frame. It can also be hidden on the set. There are three different types of mic booms: fishpole, giraffe, and perambulator booms. A fishpole boom is an aluminum pole with a mic-mounting device at one end. Some fishpoles can be telescoped to allow for maximum extension during shooting and contracted for compact storage.

One disadvantage of the fishpole boom is that the length generally cannot be changed during recording. The boom operator must move as the talent moves. Also, the entire pole must be twisted to change the positioning of the microphone, making it somewhat difficult to alternate the placement of a directional mic between two different performers. The portability and flexibility of a fishpole gain may be increased by using an FM mic instead of a wired mic (Figure 6.9).

FIGURE 6.9 The fishpole’s greatest asset is its portability. A fishpole and the attached mic are usually lightweight enough to be handheld for a relatively long period of time without excessively tiring the operator. (Courtesy of ElectroVoice USA.)

A giraffe boom is somewhat more bulky and less portable than the fishpole, but it allows for greater mobility and flexibility during recording. The giraffe is basically a fishpole gain attached to a three-wheeled dolly. It can be quickly and easily moved around the studio. It also has the advantage of allowing the operator to rotate the mic on a swivel to which the pole is attached. It requires only one operator and can be extended to different lengths during camera setups (Figure 6.10).

FIGURE 6.10 A giraffe microphone is a small boom generally used to pick up the voice of one or two persons in fixed locations.

The perambulator boom is the heaviest type of boom. It has a large pole, which can be telescoped during a camera take; a swivel mechanism for rotating the mic; an operator platform, which can be raised and lowered; heavy-duty rubber tires; a guide pole, which requires the presence of a second operator to push or pull the boom around the studio; and a boom pan-and-tilt control. The perambulator boom is designed primarily for studio use. It is not very portable. It is counterweighted so that it can support a heavy microphone and a mounting device. Some perambulator booms allow an attached microphone to be panned or moved a full 180 degrees, so that a highly directional mic can be used to pick up a moving performer or to switch from one speaker to another.

Boom Operation

Operating a boom demands great care and manual dexterity. Movements of the mic and the boom must be smooth, precise, and carefully planned. Excessively rapid movements of the boom or mic will create objectionable noise. The movement of the talent must be fully anticipated by the boom operator. If the boom operator has not preplanned the movements of the boom so that it can follow the talent, it will be difficult to maintain a constant sound level or to avoid crashing into other equipment on the set. The boom operator’s job is to keep a moving sound source within the mic’s primary pickup pattern. The operator listens to the sounds on headphones, which serve the same function as a viewfinder for a camera operator.

Omnidirectional mics are rarely used on a boom, even though using a boom might make it easier to follow the movements of the talent, because they simply pick up too much unwanted additional noise. Unidirectional mics seem to work best on a boom. They cut down on unwanted sounds by focusing on the sound source, and they provide good reception at a greater distance from the subject than mics with wider pickup patterns. This can be especially helpful when recording long camera shots with an off-camera mic.

Boom Placement

The optimum placement of a cardioid mic on a boom is one to four feet in front of and one to three feet above the speaking subject. In general, the boom operator should keep a uniform distance between the subject and the mic. Sometimes it may be necessary to vary this distance, however. To achieve proper sound perspective in single-camera recording, the mic may have to be slightly closer to the subject for close-ups and farther away for long shots (Figure 6.11).

FIGURE 6.11 A handheld microphone boom may be placed above the speaker’s head or below, depending on the noise at the location or the type of shot. The mic may be placed closer to the speaker’s mouth if the shot is a close-up rather than a wide shot.

An overhead boom can create harsh shadows that disrupt the image. The placement and movement of a boom must be carefully preplanned to prevent objectionable shadows on the set. Sometimes it is simply impossible to place the microphone directly overhead on a boom without noticeably affecting the lighting or camera and performer movements. In these situations, a fishpole boom may be placed at the bottom or side of the frame, or a hidden mic may be used.

Boom operators who are attempting to record the best-quality sound often place the mic as close to the subject as possible without the mic entering the camera frame. In multiple-camera production, the audio operator informs the boom operator when the mic has entered an underscanned TV monitor, which shows a portion of the picture not viewed at home. In single-camera productions, the camera operator carefully monitors the frame area. One strategy boom operators sometimes use to obtain good-quality sound is to place the mic within the camera frame during a rehearsal or blocking session. This forces the director, audio operator, or camera operator to ask for the mic to be raised out of the frame. This strategy ensures that the mic will always be as close as possible to the subject and forces the director to consider whether the camera placement is compromising the quality of the sound. Although directors usually are well aware of these limitations, a periodic reminder can go a long way toward preventing subsequent objections to the quality of the sound recording.

Hidden Mics

There are three different types of hidden or concealed mics: the hanging mic, the prop mic, and the concealed lavaliere mic. Hanging and prop mics are stationary, but the concealed lavaliere moves with the talent to whom it is attached. A stationary hanging mic can be attached to an overhead grid. It is usually an omnidirectional mic capable of covering a wide area of action. Its chief advantage is that it does not require a boom operator. Its obvious disadvantages are that it cannot be moved to vary or improve the audio during visual recording, and it often picks up ambient noises below it, such as footsteps and equipment being moved. Prop mics are microphones that are concealed on the set. A telephone at the center of a table around which several performers are seated can conceal a mic. Because a prop mic is stationary, it often has a relatively wide pickup pattern so that the talent does not have to stand immediately in front of the prop, calling attention to the presence of the mic or making it the focal point of a scene (Figures 6.12 and 6.13).

FIGURE 6.12 In an emergency, a microphone can be hung from a light batten over a sound source, but this works best if the person speaking stands in one position and the mic is a cardioid. (Courtesy of Audio-Technica.)

FIGURE 6.13 If the production requires off-camera mics, a microphone also can be concealed in a set piece or hand prop on a set near where the actors will be talking.

A prop mic can be extremely useful in situations in which it is difficult to use a boom, such as when the camera is shooting an extreme long shot or when the space is so confining that a boom necessarily affects the lighting. A concealed lavaliere mic is frequently used as a hidden mic for extreme long shots and complicated movements of the camera or talent. The concealed lavaliere mic may be wrapped in foam rubber and taped to the subject underneath his or her clothing. It should not be free to rub against garments or jewelry and create noise, and care must be taken to ensure that heavy clothing does not muffle the sound that the mic is intended to pick up.

Another solution to reaching sound without the mic in view of the camera is to use a supercardioid or shotgun mic (Figure 6.14).

FIGURE 6.14 The long reach of a shotgun provides sound pickup for documentaries, news gathering, and interviews in production situations that require quick and efficient operation. (Courtesy of Sennheiser USA.)

Wireless (RF) Mics

As media productions become more mobile, a need for a system of connecting audio sources with recorders and mixers without entangling wires brought about the development radio frequency (RF) wireless microphones. Each RF system consists of a microphone, a transmitter, and a receiver. Mics (usually electret) may be body mounted, head mounted, handheld, stand, or boom mounted. Each mic must be connected to a transmitter. A transmitter may be built into the base of the mic or plugged into the base of the mic, or a lavaliere mic may be connected with a short cable to a body-mounted transmitter. The receiver may be a small, battery-operated unit mounted on a camcorder or a larger A/C-powered unit feeding a mixer, public address system, or recorder (Figure 6.15).

FIGURE 6.15 High-quality, sensitive mics can be mounted directly on camera if the production requires that need, but this is not the best means of gathering sound except in unusual or emergency situations. (Courtesy of Sennheiser USA.)

The transmitters are designed to operate on one of three frequency bands: VHF, UHF, or ultra UHF. VHF equipment offers lower-priced equipment but may suffer a greater chance for interference from taxis, police officers, or other RF users. Units operating on UHF frequencies are designed specifically for radio and TV broadcasters for high-quality communication systems. Newer digital units operate above the UHF band offering the highest quality but are the most expensive. Antenna placement on both the transmitter and receiver is critical. Operation and positioning of all RF equipment should closely follow the manufacturer’s recommendations.

With smaller and quieter operating cameras, a mic mounted directly on the camera provides flexibility in movement, especially for news gathering and documentary productions.

Selecting the Best Mic

Selecting the best mic for a specific recording situation depends on an understanding of sound aesthetics and different mic characteristics. The more versatile and widely used mics are the dynamic and electret condenser cardioids. They have extremely durable elements and a pickup pattern about halfway between a full-range omnidirectional and a narrow unidirectional mic. An on-camera mic can be hand-held (in this case they should be shock mounted) or mounted on a floor or desk stand. An off-camera mic can be suspended overhead on a mic boom just outside the frame.

A cardioid mic works best when it is relatively close to the speaker or sound source; thus, it is not always the best mic to use for long-distance pickup. If an off-camera mic must be used at some distance from the speaker during the recording of an extremely long shot, then a unidirectional condenser mic, such as a supercardioid shotgun mic, may be the best choice. The narrow pickup pattern isolates the primary signal from the surrounding space. The condenser element provides a stronger signal because of its built-in preamplifier. Care must be exercised when using a shotgun mic, however, so that noise coming from directly behind the speaker is not amplified along with the primary voice signal.

A second concern with the dynamic cardioid mic is that it is difficult to make inconspicuous on camera. A lavaliere condenser mic can be the size of a tie tack. It can be placed very close to the speaker without dominating the frame. It can also be completely hidden in a person’s clothing. When connected to a tiny FM transmitter, it can even allow for freedom of movement without mic cables or for extremely long-range camera shots with extremely high-quality voice sounds. This can be an advantage when recording functional sound, but realistic sound perspective is better achieved by using a shotgun mic. The ribbon mic is best left to completely stationary performance situations, such as talk shows, interviews, or dramatic radio productions. The ribbon mic can be versatile in such a situation, because it is capable of producing a resonant sound. It can be set for an omnidirectional, bidirectional, or unidirectional pickup pattern so that several speakers, a single speaker, or two performers facing each other can use it. An omnidirectional dynamic or condenser mic is often used to record several speakers simultaneously. It can be suspended overhead in a fixed position or permanently positioned at a central location on the set.

Using Multiple Mics

Using a single omnidirectional mic is not necessarily the best way to record several different sound sources, such as several talk show performers. For one thing, even if the mic is centrally located, it will probably pick up a good deal of unwanted background sound along with the primary signals or voices. Using a different mic for each sound source provides better control and higher-quality sound recording, provided each mic can be placed close to its sound source. One advantage is that each mic can be selected for the particular characteristics of its sound source. For example, suppose that you are recording a singer on camera while a band is playing off camera. If the singer moves with a handheld mic, the loudness of the band music will vary with the mic direction, unless one or more stationary mics are set up specifically for the band.

These two sound sources should be separately controlled and combined (or mixed together), using two mic inputs on a recorder or a device called a mixer. The music now maintains a constant loudness. The singer can use a dynamic cardioid, while an omnidirectional dynamic or ribbon mic is set up for the band. Better yet, several different mics can be set up for different instruments in the band.

Separate mics can be set up for each individual speaker at a table. Each mic must be carefully placed, however, so that different mics do not pick up the same signals. This can lead to multiple-microphone interference, in which some of the sounds picked up simultaneously by two different mics cancel each other. Such cancellation is a phasing problem that occurs when similar sound waves passing through the same medium are 180 degrees out of phase with respect to each other. Phasing is sometimes used deliberately to create special audio effects, such as the noise of a robot or to disguise a speaker’s identity.

The best way to find out if multiple-microphone interference exists is to set one mic at its proper level and then turn on the other mic. If the volume goes down rather than up when both mics are on, there is interference, which must be corrected by changing the distance between the two mics or their directional placement. Some sophisticated audio consoles allow the audio engineer to eliminate phasing problems electronically. Multiple-microphone interference can be prevented by keeping live mics well separated, using directional mics, and having them directed at different sound sources. If two subjects are seated close together, a single mic should be used for both, either by swiveling an overhead mic or a boom or by placing a stand mic or a desk mic with a relatively wide pickup pattern between them. When more than two people are involved, two or more mics should be set up so that they are at least three times as far apart as the subject-to-mic distances. This three-to-one rule ensures that there will be no phasing problems with multiple mics (Figure 6.16).

FIGURE 6.16 When using multiple microphones, the three-times rule should be followed. The rule indicates that each sound source (person) must be three times the distance from any other microphone as he or she is from his or her own microphone. Any closer and audio phasing may occur, causing the sound to distort.

Another solution is multichannel recording, where each mic is fed to a separate recording channel on a computer sound application or digital audio workstation (DAW). Using multiple mics can also cause problems with excessive ambient noise. Each mic picks up the same ambient noise, and when more than one mic is used, the ambient noise adds up and can become disturbingly loud. Placing mics as close as possible to their sound sources so that loudness levels can be turned down reduces ambient noise in some instances. At other times, different speakers can simply share the same mic.

Stereo Mic Placement

Stereo provides an additional spatial dimension by giving sound a directional placement from left to right. This is accomplished by recording sounds with at least two mics. Two cardioid mics can be arranged so that they crisscross one another, forming a 45- to 90-degree angle.

Each mic picks up sounds from a different direction. This setup works quite well for speech. Using two parallel cardioid mics separated by 10 to 15 feet and well in front of an orchestra or band works well for music. The sounds picked up by each mic are kept separate and recorded on different audio channels, which can then be played back through speakers that are spatially separated from one another. For proper balance, the mics must be adjusted so that a sound coming from a source directly between them creates a signal that is equally strong on both channels.

Cardioid mics are well adapted to stereophonic use because they are slightly more receptive to sounds directly in front of them than to sounds coming from the right or left. Stereophonic sound can be used to bring added realism or simply more spectacular audio effects to a film or television program. But stereo can also bring added production problems. In terms of production logistics, it is often difficult to record stereophonic sound on location. Handling additional mics and audio equipment inevitably leads to greater risks and problems. Stereophonic recording also complicates the postproduction process, because many additional sound elements must be smoothly combined and balanced during final mixing.

Mics with built-in dual pickup heads aligned for stereo recording allow for handheld recording of stereo programming as well as providing alternate means of placing mics for stereo sound pickup. Such mics appear to be the same as standard mono-mics but provide at least two channels of separate stereo audio.

Digital Mic Placement

Most equipment, including microphones, originally designed for analog sound systems emphasizes high frequencies to compensate for losses. Digital systems do not suffer the same problem, so mics designed for analog systems used on digital systems tend to sound strident and shrill. Noises created in preamplifiers or other sources of noise that are not noticed in analog systems may become obvious in digital systems. Therefore, mics must be positioned to avoid emphasizing high frequencies by placing them off-center rather than straight out from a sound source (Figure 6.17). Some mics now are equipped to operate directly into the central processing unit (CPU) of a computer. The mic does not require an adapter of special cable because its impedance is matched to the computer audio board input.

FIGURE 6.17 Mics specifically designed for use with computers are equipped with an USB plug that carries the audio signal to the computer and power to the mic if needed. (Courtesy of Audio-Technica.)

Setting mics close also may cause distortions or noise that would not be obvious in analog systems but that would be heard in digital systems, because there is little or no masking of tape- or amplifier-created noise in a digital system. Some digital systems are also sensitive to overmodulation, creating another type of distortion to guard against.

SOUND-SIGNAL CONTROL

Controlling sound depends on understanding problems of level, signal to noise, and managing the signal as it passes through cables and operational equipment.

Audio Problems: Distortion and Noise

Distortion and noise are two different unwanted changes in an audio or video signal. Distortion is an unwanted change in a signal; noise is an unwanted addition to the signal. In both cases, audio may be distortion or noise in a specific production, or simply an additional audio element. Rock musicians often add distortion to their music for an effect. Someone trying to listen to a country-and-western recording would consider a classical music recording played simultaneously as noise. But, of course, to a classical music fan, classical music is not noise (Figure 6.18).

FIGURE 6.18 There are two types of sound: noise or distortion. Noise is unwanted sound that’s added the original sound, and distortion is an unwanted modification of the original sound.

One of the most common problems encountered in audio recording is distortion. The most common type of distortion encountered by beginning media production students is loudness distortion, which occurs when a sound is recorded at a level that exceeds the limitations of the electronic system. The peaks or high points of the sound wave are flattened, and new, unwanted frequencies of sound are produced. The end result is a reproduction that sounds like there is some kind of variable interference or garble on the line. Loudness distortion is controlled by setting the volume so that it does not exceed the limits of the system. A volume unit (VU) meter, light emitting diode (LED) meter, or peak program meter (PPM) allows the recordist to set the volume controls as high as needed for a good-quality recording without distorting the sound. Digital audio systems are sensitive to loudness distortion. Levels must be monitored carefully while recording and editing.

There are basically two types of noise, ambient noise, discussed earlier, and system noise. Ambient noise comes from open mics fed into an audio console or tape recorder that pick up the sound of air ventilators, lights, cameras, or other devices. (Fluorescent lights frequently cause a hum or buzzing sound, for example.) A second type of noise is called system noise, which can come from the electrical recording system and equipment. Microphone lines placed too close to lights and electrical cables often create system noise, as do worn volume controls or bad circuit boards and cable connections. Tape hiss is inherent in any system using analog tape recordings. Most ambient noise and some system noise can be controlled, but most system noise is simply inherent in the recording equipment. A digital audio system cannot control ambient noise any differently than an analog system can, but a digital system does reduce system noise to a minimum level. Therefore, signal-to-noise ratios are less important in digital systems.

An important determinant of sound quality is a system’s signal-to-noise ratio. This is the ratio of desired sounds to unwanted system noise. Many professional audio systems have signal-to-noise ratios of 55:1 or above; that is, the main signal is 55 times as loud as the system’s noise level. Quality audio production requires the maintenance of high signal-to-noise ratios throughout all stages of the process. At each stage of duplication or reproduction, an analog system’s signal-to-noise ratio will decrease, increasing the noise level. In digital systems, duplication or reproduction will not normally change the signal-to-noise ratio, thereby maintaining the same low level of noise (Figure 6.19).

FIGURE 6.19 The ability to minimize the signal-to-noise ratio is one of the critical measures used to determine the quality of an electronic system. Noise inherent in magnetic-tape systems and noise picked up by cables are the leading creators of poor signal-to-noise ratios.

Sound Intensity Measurement

Many different devices for indicating the volume intensity or loudness of a sound signal are used today. A less expensive tape recorder often has a red light that flickers with volume peaks. Overmodulation or loudness distortion is indicated when the light stays brightly lit for a continuous period rather than flickering intermittently. Other recorders employ a needle device that indicates loudness distortion when the needle enters a red zone. These less-expensive meters are quite small and do not have precise volume scales. More expensive meters are calibrated in specific units of sound intensity, such as volume units or percentages of modulation. There are basically three types of professional sound intensity meters: volume unit (VU) meters, peak program meters (PPMs), and light emitting diode (LED) meters. The VU meter has been the American standard. It is a special type of electrical voltmeter, which reads voltage shifts in electrical current as changes in sound intensity (Figure 6.20).

FIGURE 6.20 The top set of meters are VU meters, which read the average signal level. The set of meters on the bottom are LED meters, and they generally are set to read peak audio voltages. (Courtesy of Logitek USA.)

Needle readings are calibrated in both percentages of modulation and volume units or decibels (dBs). Approximately every 3 dB increase indicates a doubling of sound intensity. (A decibel is a logarithmic unit of sound intensity.) The modulation percentages are usually indicated on the lower scale of a VU meter. They range from 9 percent to 100 percent, the thresholds of signal detection and distortion, respectively. The upper scale indicates volume units or decibels. A reading of 0 dB usually corresponds to 100 percent modulation or peak loudness before distortion occurs, and the scale reads down on the left side and up on the right side (+1, +2, +3, and so on) of 0 dB.

A VU meter provides an electrical analog to human hearing. It does not show instantaneous peaks and immediate distortion, but it does indicate the average sound intensity over a very short period of time. This average reading closely approximates the response of the human ear to peak sound intensities. In general, signals on a VU meter should register between 50 percent and 100 percent modulation, or between 6 dB and 0 dB. Below 50 percent modulation or 6 dB, the signal-to-noise ratio becomes relatively low. Above 100 percent modulation or 0 dB loudness, distortion occurs. Sounds that intermittently peak above 100 percent modulation, or 0 dB, for very short periods of time rarely cause noticeable distortion, but sounds that continuously pin the needle to its maximum above 100 percent modulation, or 0 dB, not only cause distortion but frequently cause meter damage as well. The audio operator continually watches the VU meter and makes minor adjustments in the sound level throughout a recording, using a volume-control mechanism such as a potentiometer (pot) or a sliding fader bar. Volume level adjustments should be made smoothly and slowly. Major shifts in volume level affect the noise levels and background sounds as well as the primary signal and change the sound perspective and dynamics of the recorded sounds.

The PPM (sometimes called a modulometer) is another type of loudness or voltmeter and is the European standard. Rather than averaging sound intensities, a PPM responds immediately to peak sounds. The human ear cannot perceive extremely rapid loudness distortion, but many PPM users believe that such distortion nonetheless affects a sound recording. Obviously an operator using a PPM or modulometer must respond to needle readings on the devices somewhat differently, probably more reservedly and slowly, than one would respond to a VU meter reading. Both types of meters facilitate sound signal control, however.

A third type of level monitoring is a series of light-emitting diodes (LED). The string of diodes lights as the level intensity changes, providing an accurate and easy-to-follow means of monitoring levels. The diodes usually are arranged in a row with at least two colors. A change in color indicates over modulation. LEDs measure instantaneous peak voltages and are considered the most accurate method of determining sound levels. The LED’s small size, absence of moving parts, and minimal voltage requirements make it the ideal tool for measuring audio levels in digital equipment. The same criteria of setting proper levels determined years ago with VU meters also applies to using LED or PPI audio metering systems. Maximum levels of digital audio are more critical than analog, so proper monitoring while recording digital audio is imperative.

Some recorders have automatic gain controls (AGC) or automatic level controls (ALC) for mic input. An AGC prevents loudness distortion automatically. However, it also boosts the ambient noise level when primary sounds are at low levels, such as at pauses in dialogue. To avoid this problem, levels should be set manually using a VU meter with the AGC turned off, if possible. Peak limiters are sometimes more useful than AGCs, as these simply limit the upper level of loudness without automatically setting the basic recording level and running the risk of increasing ambient noise levels. But most professionals prefer to control recording levels manually. Volume levels on a digital recording must also be carefully set. Although digital signals are either off or on, they do vary in intensity. An overmodulated digital recording will suffer uncorrectable distortion or total loss of the recording (Figure 6.21).

FIGURE 6.21 Audio control boards vary from small portable boards (top) for field and postproduction facilities to large multi-input and multi-output boards (bottom) for motion picture mixing theaters and music recording studios. (Courtesy of Soundcraft Studer Group.)

Cables and Connectors

Professional mic cables have two conductor wire lines: a ground and a grounded shield. This type of balanced line is less susceptible to cable noise than an unbalanced line, which has a single conductor wire and a grounded shield. The two conductor lines are usually well insulated from each other, the ground wire, and the cable exterior in balanced lines. Poorly insulated cables are much more susceptible to interference from other electrical cables and devices. Mic cables should never be placed near lighting instruments or electrical power cables, which can cause interference. Nor should they be wound tightly together or twisted in any manner that will reduce their life expectancy and damage the wire conductors. A less-expensive recorder sometimes has a mic attached by an unbalanced line cable with a single-prong miniplug at one end that is inserted into the front or side of the recorder. Balanced line cables are attached to three-prong XLR connectors, which can be plugged into mics, audio consoles, or tape recorders. These connectors have separate prongs for the two conductors and the ground. Male and female connector ends lock into each other so that they do not become disconnected easily (Figure 6.22).

FIGURE 6.22 Audio and video signals are carried through different cable types and connected by a variety of connectors. On the left: three video cables, RF or F, UHF, and BNC. In the center: adapters, RCA to BNC, quarter-inch stereo to miniplug, an XLR male-to-male barrel, UHF to BNC, and RCA to miniplug. On the right: quarter-inch, XLR female, XLR male, RCA to quarter-inch adapter, and a microplug to miniplug adapter.

It is probably a good idea to wrap two cables together in a very loose knot around the connectors, so that any pulling on the cables will pull the connectors together rather than apart. Even though this procedure may place considerable stress on the cables in the case of an accident, such as someone tripping over a cable, it would undoubtedly be more expensive to reshoot the entire sequence in the event of a complete disconnection. Care should be taken, of course, to minimize the amount of twisting and stress that occurs at the juncture of the cable and connector, because this part of the cable is extremely vulnerable to damage and wear. Also cables should be coiled carefully before storing. Two methods of coiling cables are the over-and-under method and the figure-eight method. Each system, when properly carried out, prevents internal twisting and damage to conductors inside the cable (Figure 6.23).

FIGURE 6.23 To properly coil cables, an over-and-under method may be used. This system protects the cable from twisting and provides a clean, straight unwinding when uncoiled for use.

Mixing

An audio console or mixer is designed to combine sounds from several different sound sources, such as mics, tape recorders, and playback units. These devices can vary from an elaborate studio audio console to a simple multiple-input mixer, which allows for separate volume control over each input (Figure 6.24). Basically the audio console routes signals from sound sources or playback units to a control device or recording unit. It can send a signal from a particular mic or a playback unit to a recorder, so that a duplicate copy or dub can be made, for example. It can combine or mix together several sound sources into one (monophonic) or onto two or more (multitrack) soundtracks or channels, which are recorded as a final or master audiotape.

FIGURE 6.24 Audio boards cover a wide range of styles and capabilities. Some are designed for specific production purposes. The top board is designed for radio control room operation, and the board on the bottom is used for television studio production. (Courtesy of Utah Public Radio, Logitek USA, and Wheatstone Corporation.)

In a digital board, the analog signals originating from mics, tape decks, or other non-digital sources are converted to a digital signal as it enters a digital mixer. Digital inputs are then combined with the converted analog inputs. From that point until the signal must be converted back to analog to feed speakers or headphones, the signal may remain in the digital format for processing and editing. Digital boards also provide for digital outputs to feed other digital signals including a digital transmitter. Some digital boards contain their own recording media. A computer hard drive, solid-state memory, or digital tape deck may be built into the board. Because a digital board is in essence a computer with multiple inputs and outputs, the processing of the signals, depending on the software, follows that of computer word processing: cutting, pasting, adding, deleting, and modifying with simple user-friendly controls. There will be more on this subject in , Editing. Some digital boards are labeled digital audio workstations (DAW) (Figure 6.25).

FIGURE 6.25 A digital audio workstation (DAW) is designed for quick editing for talk radio, call-in clips, news actualities, promotional announcements, and commercials. It is designed to edit like a word processor with cut/copy/paste and precision scrubbing functions. (Courtesy of Fairlight, USA.)

An audio console consists of a series of faders, each of which controls the volume level of a single input. The inputs can come from microphones, turntables, analog and digital audiotape recorders, compact disc (CD) players, or audio playbacks from videotape recorders. A single (or dual) master pot controls the output. Each pot or fader often has its own equalization controls for increasing or decreasing bass and treble (low frequencies and high frequencies) directly above or below it. In a digital board, signal control may be by faders or by computer controls or software operations (Figure 6.26).

FIGURE 6.26 Audio mixing board circuits follow the basic pattern of preamplifying low-level inputs, mixing all inputs through a program bus, and monitoring the program output signals by viewing a metering system and listening on headphones or through loudspeakers. A parallel set of circuits carries signals for cueing or monitoring purposes without placing those signals on air.

Most audio consoles and mixers have two types of audio inputs: high impedance and low impedance. Impedance cuts down on the flow of alternating current; it is analogous with resistance in devices operated on direct current (batteries). Impedance and resistance are measured in ohms. High-impedance signals come from some nonprofessional mics, from playback machines, and from some signal-processing equipment. Low-impedance signals usually come from professional-quality microphones and other equipment. Professional mics usually have an impedance of about 50 ohms, whereas playback units and other high-impedance sources are above 600 ohms.

An impedance imbalance or mismatch between the sound source and the mixer or console input will result in a signal that is either too loud and distorted or too soft and weak to be useful for recording purposes. Different sound sources can have different levels of sound intensity or signal strength (volts), as well as different impedances (ohms). These also require separate inputs or an audio console. Mic levels are usually lower than line levels from playback units. Preamplifiers in the audio console car boost a low-level signal to a higher level so that it equals that of other sound sources. When a high-level signal enters the console through a low-level input, distortion occurs. The level output of a mixer can be either high-level or mic-(low)-level, or, if amplified, the speaker level that is generally the highest level (Figure 6.27).

FIGURE 6.27 The three primary audio levels vary in voltage from the very weak signal directly from a microphone, turntable pickup arm, and magnetic recorder playback head, to a middle level of a preamplified signal (called line level), and lastly to the high level of the output of an amplifier intended to power a speaker or speaker system.

Any computer equipped with sufficient memory, a sound-processing board, and an audio-editing program can function as an audio postproduction board. The functions of mixing, equalizing, setting and varying levels, editing, and adding special effects are performed quickly with such a computer system. More detailed information on these topics may be found in , Editing.

Compression

The term “compression” in audio traditionally referred to a process of decreasing the dynamic range (loudest to quietest) of a signal. In the digital world, “compression” refers to a reduction in the amount of bandwidth required to record or transmit a digital signal. A compression system omits certain sounds unimportant or redundant in the overall signal so that the human ear does not recognize the loss. The amount of compression is stated as a ratio of 2:1, which means the bandwidth has been cut in half. The higher the compression ratio, the greater the possibility that the signal will lose enough of the signal that a discerning listener will detect a loss in quality. MP3 recordings are compressed to reproduce a signal lower in quality than a CD that is also compressed, but not as much.

The term codec (COmpressionDECompression) refers to a process or equipment that encodes or decodes data. To save storage space and time in moving files with high bytes of data, repetitious amounts of data are deleted. There are two basic systems in common use: lossy and lossless. In lossy systems, unneeded data are not transmitted or recorded. Tests indicate the deleted data generally are not missed. Lossless systems either do not compress the files or do so in such a manner that the deleted data are replaced or substituted for when decompressed. Lossy systems use far less space than lossless systems but offer a lower quality signal on reproduction.

A variety of mostly noncompatible audio codecs are used to record, store, and process audio files. The quality of the end product depends on the application required for the audio signal. High-quality soundtracks for motion pictures and network television program require the highest uncompressed audio. Mobile and handheld audio players monitored with headsets can operate with higher compression ratios to save on storage space and because the listener will not easily notice the lower quality.

Some Examples of Codecs Now in Use

Console Operation

Once impedance and line levels of source and input match, the console operator can set the proper loudness levels for recording each sound source. To accomplish this on an audio console or mixer with a single VU meter or LED indicators, all of the faders should be closed, except for the one being set. If each input has its own VU meter, then it can be set independently from the others. The level for each input should be set between 80 percent and 100 percent modulation for an optimal signal-to-noise ratio. In some instances, such as background music and sounds, the level may be set somewhat lower for a proper overall balance between the sounds. Balance is an aesthetic concept that refers to the best proportion of sound intensities from the different elements, such as speech and music. Generally speaking, music must be toned or faded down to achieve a proper balance with accompanying dialogue or narration. In addition to balancing sounds, an audio operator should check for multiple-microphone interference by determining if the volume levels of specific sources increase as others are shut off. It is generally a good idea to label each fader with the number or name of the mic or sound source it carries for each sound source and fader in order to eliminate any confusion when adjustments have to be made during actual recording.

Recording and Mixing Commands

To perform well at the audio console or mixer, the audio recordist should be familiar with each of the following audio terms, cues, and commands:

SOUND PERSPECTIVES

The relationship of sound to space parallels that of a picture with three dimensions: left-right, up-down, and in front of or behind the listener. To duplicate audio realistically, the perception of these dimensions must be duplicated. The characteristics to be duplicated are distance and directionality. Sound that appears to be originating close by or far away may be recorded by placing the mic(s) close to or at a distance from the sound source. The sound indicating the size of the environment will be determined by the reflective or absorption values of the walls, furniture, or other objects in the space, and the size of the room. The direction of the sound can only be determined through the use of one of several multichannel systems to give the audience the sense that the sound is to the left, right, in front of, or behind.

Stereo Sound

Both stereo and surround-sound systems and modifications are attempts to duplicate the three-dimensional aspects of a sound environment. Stereo offers two or three channels of sound: left, right, and, depending on the size of the theater, a center channel. Reproduction of sound through the left or right channel to match the objects originating the sound on the screen may add to the realistic effect but also may become confusing if overdone. Now that stereo television receivers are available, the impulse to split sound into left and right must be weighed against the realization that television is a close-up medium that requires audio to be concentrated in the center of the screen or balanced between the left and right channels.

Wide-screen films, on the other hand, offer much greater opportunities to utilize sound originating from the side of the screen matching the source of the sound. Care must be taken not to overbalance sound to one side or another, or else the audience seated on the opposite side of the theater may miss some critical sounds. In the recording industry, music is recorded either to duplicate the physical arrangement of the group (symphonic orchestras) or to enhance the vocals or a solo instrument (rock groups). Today music is recorded with the assumption that it will be reproduced on a stereo rather than monaural system.

Multichannel Sound

Surround sound was the next logical step from stereo in creating the realistic three-dimensional sound environment. Mixing the sound into at least four channels to be reproduced through speakers located in the four corners of the listening space enhances the sense of the original recording, but such a process increases the complexity of mic placement, mixing, and speaker location. Surround sound in the past has not proved practical for the average home or video production. Modification of the theory has opened new avenues for film sound. Most theaters today are capable of reproducing sound recorded on as many as 10 channels using three to four speakers lining the side walls of the theater to supplement the normal stereo and center speakers located behind the screen. Multichannel sound increases the illusion of both the width and depth of the picture.

Dolby Digital 5.1, 6.1, and 7.1 Sound and Beyond

Dolby Digital 5.1, 6.1, and 7.1 sound systems designed to complement advanced television systems requires six, seven, or eight channels of audio and six or seven separate speakers. The usual four-corner speakers (left front, right front, left rear, and right rear) are supplemented by two, three, or four additional speakers: one directly behind or under the screen for bass response, called a subwoofer, and the sixth directly behind or above the screen, called a front channel, and a seventh directly behind the audience (for 6.1) or a seventh and eighth (back left and back right for 7.1) behind the audience, which are called rear channels. The maximum effect of making the audience feel as if they are placed within the program occurs when each audio channel is properly programmed to carry the correct signal. Whether the complexity of Dolby Digital 5.1, 6.1, or 7.1 will discourage consumers from installing such a system to match their HDTV system will be determined in the next 5 or 10 years. As a production situation, a 5.1, 6.1, or 7.1 signal, consisting of six or seven channels, is complicated not only by the differences in apparent direction of the sources but also by the differences in equalization of the individual channels. Editing multichannel systems is complex and requires a deep understanding of the part each channel of audio plays in that particular production. Ultra-high-definition TV, still in development, plans on using 21 channels of audio for exhibition.

Summary

The aesthetic use of recorded sounds demands an understanding of realist, modernist, and postmodernist aesthetics as well as recording devices and their selection, placement, and control. A mic or microphone is a transducer that converts analog sound-wave energy into analog electrical energy. Mics can be classified into three different categories on the basis of their transducer elements: dynamic, ribbon, and condenser.

Mics can also be classified on the basis of pickup patterns: omnidirectional, bidirectional, and unidirectional mics, such as the cardioid and supercardioid or shotgun mic. Mics can be placed in on-camera and off-camera positions. Hand mics, desk mics, stand mics, and lavaliere mics are examples of on-camera mic positions. Mics on booms, such as fishpole, giraffe, and perambulator booms, as well as various hidden mics, such as the hanging mic, prop mic, and concealed lavaliere mic, are off-camera mics.

Selecting the best mic and mic position depends on an understanding of what mic characteristics and placements are best suited to a specific situation. Digital recording of audio requires greater care in mic selection and placement as well as noise reduction.

Sound signal control helps a sound recordist achieve the best-quality recorded sound by eliminating specific audio problems, such as loudness distortion and excessive ambient and system noise. A sound-measuring device, such as a volume unit (VU) meter or LED indicators, can be used to set the sound level as high as possible for optimal signal-to-noise ratio while avoiding loudness distortion. Balanced mic cables should be used to minimize noise and electrical interference.

Audio mixing is done on an audio console, mixer, or DAW. Mixing refers to combining several different inputs, such as different mics or playback machines, into a single (monophonic) or dual (stereophonic) output, which is directed to a tape recorder or some digital recording medium. Faders on the audio console, mixer, or DAW are used to adjust the volume gradually.

Audio operators using a console, mixer, or DAW must be familiar with basic audio terms, cues, and commands so that they can effectively communicate with the rest of the staff and crew. Sound perspectives should match the perspectives of the matching visual as well as the requirements of the drama or music. Perspectives include both distance and dimensionality.

EXERCISES

Additional Readings

Alburger, James. 2007. The Art of Voice Acting: The Craft and Business of Performing for Voice-Over, Focal Press, Boston.

Alten, Stanley. 2008. Audio in Media, eighth ed. Wadsworth, Belmont, CA.

Bartlett, Bruce, Bartlett, Jenny. 2005. Practical Recording Techniques: The Step-by-Step Approach to Professional Audio Recording, fourth ed. Focal Press, Boston.

Bartlett, Bruce, Bartlett, Jenny. 2007. Recording Music on Location, second ed. Focal Press, Boston.

Case, Alex. 2007. Sound FX: Unlocking the Creative Potential of Recording Studio Effects, Focal Press, Boston.

Eargle, John. 2004. The Microphone Book: From Mono to Stereo to Surround: A Guide to Microphone Design and Application, second ed. Focal Press, Boston.

Geoghegan, Michael, et al. 2008. Podcast Academy: The Business Podcasting Book: Launching, Marketing, and Measuring Your Podcast, Focal Press, Boston.

Grant, Tony. 2003. Audio for Single Camera Operation, Focal Press, Boston.

Gross, Lynne S, Reese, David E. 2001. Radio Production Worktext, fourth ed. Focal Press, Boston.

Hausmann, Carl, et al. 2007. Modern Radio Production, seventh ed. Wadsworth, Belmont, CA.

Holman, Tomlinson. 2005. Sound for Digital Recording, Focal Press, Boston.

Holman, Tomlinson. 2008. Surround Sound, second ed. Focal Press, Boston.

Howard, David, Angus, Jamie. 2006. Acoustics and Psychoacoustics, third ed. Focal Press, Boston.

Huber, David Miles, Runstein, Robert. 2005. Modern Recording Techniques, sixth ed. Focal Press, Boston.

Iuppa, Nicholas. 2001. The Complete Guide to Game Audio for Composers, Musicians, Sound Designers, and Game Developers, Focal Press, Boston.

Izhaki, Roey. 2008. Mixing Audio: Concepts, Practices, and Tools, Focal Press, Boston.

Katz, Bob. 2007. Mastering Audio: The Art and Science, second ed. Focal Press, Boston.

Keith, Michael C. 2007. The Radio Station: Broadcast, Satellite, and Internet, Focal Press, Boston.

McGuire, Sam, Pitts, Roy. 2008. Audio Sampling: A Practical Guide, Focal Press, Boston.

Newell, Philip. 2007. Recording Studio Design, Focal Press, Boston.

Newell, Philip, Holland, Keith. 2006. Loudspeakers for Music Recording and Reproduction, Focal Press, Boston.

Nisbett, Alec. 2004. Sound Studio: Audio Techniques for Radio, Television, Film, and Recording, Focal Press, Boston.

Rumsey, Francis. 2003. Desktop Audio Technology: Digital Audio and MIDI Principles, Focal Press, Boston.

Rumsey, Francis, McCormick, Tim. 2005. Sound Recording: An Introduction, Focal Press, Boston.

Watkinson, John. 2002. Introduction to Digital Audio, second ed. Focal Press, Boston.

Webber, Stephen. 2008. DJ Skills: Essential Guide to Mixing and Scratching, Focal Press, Boston.

Weis, Elisabeth. 1982. The Silent Scream: Alfred Hitchcock’s Sound Track, Farleigh Dickinson University Press, Rutherford, NJ.

Weis, Elisabeth, Belton, John. 1985. Film Sound: Theory and Practice, Columbia University Press, New York.

Yale French Studies. 1980. Special issue on “Sound in Film,” 60.

Yewdall, David Rush. 2007. The Practical Art of Motion Picture Sound, Focal Press, Boston.