A sound reinforcement system is a functional arrangement of electronic components that is designed to reinforce a live sound source.^[1][2] This usually involves making the sound louder or distributing the sound to a larger or more distant audience.^[3] Sound reinforcement systems are a specialized type of professional sound systems, which are in turn a subset of sound systems. Sound systems are anything which amplify, record or transmit sound, such as telephones, sonar or even medical ultrasound systems.^[4]

A sound reinforcement system may be complex and include hundreds of microphones, complex mixing and signal processing, and multiple loudspeaker arrays. They can also be as simple as a single microphone connected to a self-powered loudspeaker system. In both cases it reinforces sound to make it louder or distribute it to a wider audience.^[5] Sound reinforcement systems are designed, installed, and operated by trained audio engineers who are able to reduce the risks inherent in the use of such a system, such as high sound pressure levels which may damage listeners' hearing.

Basic concept

An example of a simple sound reinforcement system

A typical sound reinforcement system consists of three parts: Image File history File linksMetadata SimpleSRsystem. ... Image File history File linksMetadata SimpleSRsystem. ...

• Input transducers -- which convert different forms of energy such as sound into an audio signal
• Signal processors -- which alter audio signal characteristics
• Output transducers -- which convert audio signal into sound

Sound is taken and converted into electronic signal by an input transducer (such as a microphone or pickup). A signal processor (such as a mixing console, amplifier, or other devices) then alters the signal characteristics. This includes equalization, balance, effects and amplitude. Finally, an output transducer (such as a loudspeaker or headphones) converts the electronic signal back into sound, so that the listener can hear the end product. This basic concept of sound reinforcement systems encompasses anything from a simple system with only one microphone, amplifier and loudspeaker, to the complex systems in professional applications including multiple mixing boards, monitors and a vast selection of effects. There is debate on the classification of sound systems as "sound reinforcement systems" and "public address systems" depending on size and application. For more information go to Terminology. A transducer is a device, usually electrical or electronic, that converts one type of energy to another. ... This article or section does not cite its references or sources. ... A microphone, sometimes referred to as a mike or mic (both IPA pronunciation: ), is an acoustic to electric transducer or sensor that converts sound into an electrical signal. ... Three magnetic pickups on an electric guitar. ... BBC Local Radio Mark III radio mixing desk In professional audio, a mixing console, digital mixing console, mixing desk (Brit. ... Mission Cyrus 1 Hi Fi integrated audio amplifier An audio amplifier is an electronic amplifier that works with audio frequencies (generally 20 hertz to 20,000 hertz). ... “Loudspeaker” redirects here. ... // Radio headphones c. ...

Signal path

The signal comes in the top of the board and travels toward the bottom, being adjusted by each dial/fader in turn

Sound reinforcement in a large format system typically uses the following signal path: Image File history File linksMetadata Download high-resolution version (832x1064, 143 KB) A cropped close up on a Soundcraft brand mixing console, taken with a Sony Cybershot by Davidkazuhiro. ... Image File history File linksMetadata Download high-resolution version (832x1064, 143 KB) A cropped close up on a Soundcraft brand mixing console, taken with a Sony Cybershot by Davidkazuhiro. ...

A signal enters the chain via a directly connected instrument or a transducer (microphone) which is plugged into the multicore cable (often called a "snake") where it is split to carry the signal to the Front of the House and to the monitor consoles. Once in a channel on the console, this signal can be equalized, panned and amplified. One might also patch it into an external effect processor present in the channel. A microphone, sometimes referred to as a mike or mic (both IPA pronunciation: ), is an acoustic to electric transducer or sensor that converts sound into an electrical signal. ... It has been suggested that this article or section be merged with snake cable. ... BBC Local Radio Mark III radio mixing desk In professional audio, a mixing console, digital mixing console, mixing desk (Brit. ... Methods and media for sound recording are varied and have undergone significant changes between the first time sound was actually recorded for later playback until now. ...

The signal can be routed internally to either a summation bus, also known as a group, in order to allow the engineer to control the levels of several related signals at once. A good example of this is controlling several individually miked drums or backup singers. From here each signal is typically routed through the stereo masters on a console (left and right, or balance, pan, etc.). Additionally, each signal can be sent to separate outputs from the main console, typically referred to as Auxiliary sends, or "Auxes."

The next step in the signal path generally depends on the size of the system in place. In smaller systems, the main outputs would be sent to an additional equalizer, or directly to a power amplifier, with one or more loudspeakers (typically two) then connected to that amplifier. In large-format systems, the signal is first routed through an additional equalizer then to a crossover. A crossover splits the signal into multiple frequency bands, with each band being sent to separate amplifiers and speaker enclosures for low-, middle-, and high-frequency sounds. Equalizer can mean: Equalizer, an audio processing tool. ... An amplifier is a device which changes a small movement into a larger movement. ... Equalizer can mean: Equalizer, an audio processing tool. ... Audio crossovers are a class of electronic filters designed specifically for use in audio applications, especially hi-fi. ...

Once the signal has reached the amplifiers it is typically no longer considered a "signal" per se, as the voltages and currents now present greatly exceed those experienced throughout the rest of the signal path.

System components

Input transducers

Many types of input transducers can be found at the scene of sound reinforcement, but not all are maintained and operated within a sound reinforcement system. Sound reinforcement systems are mainly concerned with microphones as more so than other types of transducers.

Microphones

Microphones are the foremost input transducers in a sound reinforcement system. They can be classified according to their method of transduction, pickup (or polar) pattern or their functional application.Modern microphones either utilize the electromagnetic principle (dynamic) or the condenser (capacitor) principle for their method of transduction.poo A microphone, sometimes referred to as a mike or mic (both IPA pronunciation: ), is an acoustic to electric transducer or sensor that converts sound into an electrical signal. ...

Pickup patterns

Microphone pickup patterns are either omnidirectional (picking sound equally from all directions) or bidirectional (picking up sound in a figure eight pattern). All existing patterns are based on either or a combination of these two basic patterns. An equal combination of omidirectional and bidirectional is called cardioid, since a graph of the pickup pattern represents a heart. For a full discussion on pickup patterns, see the article on microphones. A microphone with a cord A microphone, sometimes called a mic (pronounced mike), is a device that converts sound into an electrical signal. ...

Functional applications

There are many different functional applications in modern sound reinforcement systems. Stand-mounted microphones are by far the most common in all sound reinforcement applications. Virtually all microphones come with their own stand mount, including studio microphones, which often also have shock-proof suspension. Podium mounts are used for the reinforcement of fixed speech. This can either be mounted on a removable desk stand or be permanently fixed. Personal microphones take the form of tie-clip microphones or head worn microphones so that the performer or speaker can freely move about. The head-worn type was first used in new gathering applications, and finally found full acceptance on the musical stage through the pioneering use by singer Garth Brooks.^[6] This article or section does not adequately cite its references or sources. ...

Hanging microphones are commonly used in religious applications for choral pickup. These are popular because of the need of unobtrusiveness of sound sound reinforcement systems in houses of worship. Boundary layer microphones are commonly placed on the floor at the front of the stage in theatrical applications, as well as conference rooms and telecommunications. These microphones are barely visible yet still provide good sound balance even when placed some distance from performers. Distance pickup microphones, such as shotgun or rifle microphones, are used in applications where unobtrusiveness is a requirement. Stereo mounts are used to pick up a general coverage of large music ensembles. Instrument-mounted microphones are used on many instruments such as drums and acoustic guitars. These can also be plugged into a transmitter so that a moving performer can go wireless.

Electrical specifications

In sound reinforcement, it is important to understand the electrical specifications of microphones to fully take advantage of them. Important electrical characteristics are:

• Microphone sensitivity
• Frequency response and directional data
• Detailed polar curves
• Dynamic range
• Signal padding
• Microphone output impedance

The operating environment

There are different phenomena in the operating environments of sound reinforcement for microphones. Proximity effect is the change in frequency response in relation to the distance between the microphone and the sound source. Since cardioid microphones are susceptible to the proximity effect, wind-screens are necessary to prevent the "p" and "b" sounds from popping when the microphone is near a singer or speaker. Another phenomenon is reflections from nearby surfaces. This is commonly observed with podium mounted microphones, where the microphones picks up the original source as well as the reflection off of the podium. A common remedy is to move the microphone closer to the reflecting surface to minimize the path length difference between direct and reflected sound. Another remedy is to use a more directional microphone instead, and block out the reflected sound. Another phenomenon is multi-microphone interference. If more than one microphone picks up the same signal and is mixed, a delay effect is observed. This can be avoided by either using directional microphones or keeping the different microphones closer together, provided they are intended to pick up the same source. Frequency response is the measure of any systems response to frequency, but is usually used in connection with electronic amplifiers and similar systems, particularly in relation to audio signals. ... Dynamic range is a term used frequently in numerous fields to describe the ratio between the smallest and largest possible values of a changeable quantity. ... In physics, proximity effect is a term used for many different effects in which things behave differently when near, or proximate, to one another. ...

Contact microphones

Contact microphones convert sound waves in dense mediums such as wood or metal into an electronic audio signal. They are not often thought of as microphones because of their different application. Contact pickups are often piezoelectric, and are typically used on stringed instruments and occasionally on drums as noise gate triggers. They are usually of the crystal type, but occasionally are capacitive. A Contact Microphone is a form of microphone designed not for picking up airborne sound waves, but rather to act as a transducer which picks up vibrations through solid materials and converts them into audible sound. ...

Wireless microphones

Wireless microphones have reshaped the way staged events are carried out in sound reinforcement. Typically transmitters are either handheld or bodypacks. Handheld transmitters combine both the microphone and the transmitter in one unit. Controls include a power on/off switch, gain adjustment a microphone on/off switch (mute button). These are powered by batteries which last around 2 hours or more before needing replacement. Bodypacks have the same controls and battery life as handhelds, but are more versatile in that they can be connected to a variety of microphones or line-level sources since the microphone is not built in. Invented by Hung C. Lin, the first wireless microphone went into production in 1962, called the transistophone and put out by the German equipment manufacturer beyerdynamic. ...

Usage of microphones can become quite complicated because of limited bandwidth and the different laws in effect regarding it. Using wireless microphones can be quite complicated if no licenses are obtained, especially in a metropolitan area where many frequency ranges are in use. In larger productions where more than 20 channels may be used, a person is put in charge of wireless activity. This involves ensuring all transmitters have freshly charged batteries and assigning this microphones to actors and performers.

Other transducers

There are many other types of input transducers which sound reinforcement is not directly concerned with, yet may be used occasionally. A sound reinforcement system operator may not usually operate these devices, but will benefit from a knowledge of them, especially in the event of technical trouble related to them. These input transducers include:

• Magnetic pickups. These are used in electronic stringed instruments, such as electric guitars.
• Tape heads
• Laser pickups. These are used with CDs
• Optical pickups. These are used in film.
• Phonograph pickups (cartridges)

A tape head is a type of transducer used in tape recorders to convert electrical signals to magnetic fluctuations and vice versa. ... A Compact Disc or CD is an optical disc used to store digital data, originally developed for storing digital audio. ... Edison cylinder phonograph ca. ...

Signal processors

Mixing consoles

Mixing consoles are the heart of a sound reinforcement system. This is where the operator can mix, equalize and add effects to sound sources. Many different features are included on top of the basic mixing function in different mixing consoles. The most standard features are onboard equalization and preamplificaiton, audio inserts and auxiliary outputs. More advanced mixing consoles include metering, mute groups, voltage controlled amplifiers (VCA), VCA groups, LCR panning, Scenes and Matrix output. Multiple consoles can be used for different applications in a single sound reinforcement systems. There are other types of consoles than a sound reinforcement mixing console. Besides non-SR consoles (such as those for recording, broadcast or post production), there is the stage monitor mixing console. This console is dedicated to creating mixes for the monitors on stage. Subsequently these consoles have a large number of mix group controls. Thus each musician on stage can have a different mix of their own, e.g. the lead singer has her voice louder in her monitor loudspeaker and the keyboardists has his keyboard louder in his monitor loudspeaker. These consoles are normally placed at the side of the stage so that the operator can be in direct communication with the performers on stage. Some mixers have amplifiers built in an are called powered mixers. Though these are easy to set up, they are not as versatile and limit the operator to certain loudspeakers. BBC Local Radio Mark III radio mixing desk In professional audio, a mixing console, digital mixing console, mixing desk (Brit. ... For information about computer bandwidth management, see Equalization (computing). ...

In sound reinforcement it is important for the mixing console to be located where the operator can see and hear the action on stage. This means the console will be ideally placed within the audience. Places with permanent systems installed such as religious facilities and theaters can be tempted to place the mixing console within an enclosure, but this suitable for broadcast and recording, not sound reinforcement. If the system will be serving and live music and dramatic presentations, then a sound reinforcement setup is a superior setup.

Digital mixing consoles

On top of duplicating their analog counterpart, digital mixing consoles offer superior versatility and many new features. Such features include saving multiple mute groups, multiple VCA groups and channel settings into a scene, reconfiguration of signal routing and the incorporation of special effects units into the console. Many digital mixing consoles are designed so that they are only a remote control to an engine which it is connected to. This makes for a light console and a clean operating environment. With digital consoles it is also possible to use a computer as a remote control, yet this is not very popular with operators who are used to analog systems and would like instant access to all controls, a function which a keyboard cannot keep up with let alone a mouse. In professional audio, a digital mixing console, is an electronic device for combining, routing, and changing the dynamics of digital audio samples. ...

Many tasks of an operator are predictable and redundant, such lowering the gain of peaking channels or lowering channels which are causing feedback. Digital consoles can perform automatic mixing for these sorts of tasks. In simple systems such as those used in courtroom, conference, or speech-oriented religious facilities, the automatic mixing function can totally replace an operator. In more complicated applications however it can be used as an aide but no more.

Equalizers

Equalizers exist in sound reinforcement systems in two forms: Graphic and Parametric. Both of these are used in conjunction with End-cut filters. For information about computer bandwidth management, see Equalization (computing). ...

Parametric equalizers

Parametric equalizers have knobs which adjust three parameters: frequency, boost/cut and Q (bandwidth). These equalizers are often found built into each channel in mixing consoles, but are also available as separate units. Parametric equalizers first became popular in the 1970s and have remained the program equalizer of choice since then. Parametric equalizer (or parametric EQ) is an electronic multi-band variable equalizer device used in sound recording and live sound reproduction with Public Address systems (PA systems). Parametric equalization devices allow audio engineers to control the parameters of the internal band-pass filter sections (amplitude, center frequency and bandwidth). ...

Graphic equalizers

Graphic equalizers get their name from the fact that they have faders which resemble a frequency response curve plotted on a graph. These equalizers typically exist as separate units. Sound reinforcement systems normally use graphic equalizers designed on one-third octave centers.

End-cut filters

End-cut filters restrict a given channels bandwidth extremes. This is useful in keeping subsonic disturbances and RF or lighting control disturbances from interfering with the audio system. End-cut filter sections are often included with graphic equalizers to give full control of the frequency range. If their response is steep enough, then high-pass filters (low-cut) and low-pass filters (high-cut) can function as end-cut filters. A high-pass filter is a filter that passes high frequencies well, but attenuates (or reduces) frequencies lower than the cutoff frequency. ... A low-pass filter is a filter that passes low frequencies but attenuates (or reduces) frequencies higher than the cutoff frequency. ...

Compressors and limiters

Compressors and limiters are used to maintain an average signal level by automatically reducing the level of louder sound. Signal compression is necessary because the dynamic range of equipment may be smaller than that of the signal and cause distortion, clipping, or even equipment damage. When the signal is compressed, there is a smaller difference between the volume of the loudest and quietest sounds. Thus the content can still sound like it's louder or quieter but actually have pretty much the same volume. The output level of a compressor can then be turned up to optimize the efficient use of a systems dynamic range. Compressors also have attack and release settings, which determine how long the compressor will wait before turning down the volume on a loud sound, and how long to wait before turning the volume up again. Attack time is usually within the range of microseconds, while release time is in the range of seconds. There is also a threshold setting which determines at what minimum decibel level the compressor should start compressing. Compressors are useful for keeping a vocalist's with a large dynamic range restricted. Professional vocalists know to distance themselves from a microphone when making especially loud noises, but many amateurs do not do this. Compressors provide for safer signal handing by avoiding distorting and clipping and also a more pleasant listening experience. Audio level compression, also called dynamic range compression, volume compression, compression, limiting, or DRC (often seen in DVD player settings) is a process that manipulates the dynamic range of an audio signal. ...

A limiter is essentially a compressor with the most extreme setting in effect, and sets a volume ceiling on a signal instead of a slow curve. These are extremely valuable for suppressing very loud pops (such as when unplugging a guitar or dropping a microphone) which could damage the audio system.

Noise gate

A noise gate sets a threshold where if it is any quieter it will not let the signal pass and if it is louder it "opens the gate." Thus the noise gate's functions are very much opposite to those of a compressor, and can be also thought of as an "expander". There are also attach and release settings, which work in the same way as a compressor. Noise gates are useful for instruments or microphones which will pick up noise which is not relevant to the program (such as the acoustic guitarist readjusting her instrument or the vocalist whispering to another performer). A noise gate is an electronic device or software logic that is used to control the volume of an audio signal. ...

Reverb

• For the full article, see Reverberation

This article is about audio effect. ...

Delay

• For the full article, see Delay (audio effect)

Delay is an audio effect which records an input signal to an audio storage medium, and then plays it back after a period of time[1]. The delayed signal may either be played back multiple times, or played back into the recording again, to create the sound of a repeating...

Flangers and phasers

• For the full articles, see Flanging and Phaser (effect)

Flanging is a time-based audio effect that occurs when two identical signals are mixed together, but with one signal time-delayed by a small and gradually changing amount, usually smaller than 20 ms (milliseconds). ... A phaser is an audio signal processor used to filter a signal by attenuating a series of notches in the frequency spectrum. ...

Exciters

• For the full article, see Exciter (effect)

An Exciter is an audio signal processing technique used to enhance a signal by Dynamic Equalization, Phase manipulation and harmonic synthesis of (usually) high frequency signals. ...

Feedback suppressors

Feedback suppressors automatically suppress feedback from occurring. Audio feedback (also known as the Larsen effect) is a special kind of feedback which occurs when a loop exists between an audio input (for example, a microphone or guitar pickup) and an audio output (for example, a loudspeaker). ...

Amplifiers

Amplifiers increase the signal level to drive the loudspeaker. All output transducers require amplification of the signal by amplifiers, including loudspeakers, monitor speakers, and even headphones. The heavy duty amplifiers used to power loudspeakers and monitor speakers are referred to as power amplifiers. Virtually all professional amplifiers are self-protected from abuse be it overdriving the input or placing a short circuit across the output. Modern amplifiers are designed to prevent internal component failure. As most sound reinforcement equipment, professional amplifiers are designed to be mounted on 19-inch racks. Heat dissipation is an important factor for operators to consider when mounting amplifiers onto equipment racks, since they can generate a significant amount of heat. Mission Cyrus 1 Hi Fi integrated audio amplifier An audio amplifier is an electronic amplifier that works with audio frequencies (generally 20 hertz to 20,000 hertz). ... Equipment mounted in several 19-inch racks. ...

Digital loudspeaker system controllers

Digital loudspeaker system controllers (DLSC), also known as digital crossover networks, are most commonly used to process the final mix being sent from the mixing console to the amplifiers, and in turn to the loudspeakers. They usually carry out the following functions:

Crossover Network

Multiple loudspeakers with a more narrow-band response, tailored to specific frequency bands, can be used together (i.e. lows, mids and highs) to obtain a more accurate reproduction of the input signal. This also makes more efficient use of amplifier power by sending each amplifier only the frequency band that its respective loudspeaker can reproduce. The crossover function of a DLSC does this splitting of the input signal into separate outputs for each speaker.

Compression/Limiting

Most commonly used for, but not limited to, protecting loudspeakers and amplifiers from signal overload.

Time Alignment

Calibration/Testing

Most DLSCs have calibration and testing functions, such as a tone generator, a pink noise generator coupled with a real-time analyzer and automated room optimization. Pink noise spectrum Pink noise ( ), also known as 1/f noise or flicker noise, is a signal or process with a frequency spectrum such that the power spectral density is proportional to the reciprocal of the frequency. ...

Output transducers

Loudspeakers

Loudspeakers (also known as drivers) are the main speakers which project the sound to the audience. They can exist as single full range loudspeaker system, or can be a collection of high, middle, and low frequency drivers, more commonly know as Tweeters, Midrange and Woofers in the consumer electronics field. “Loudspeaker” redirects here. ...

Monitor speakers

Monitor speakers are usually full range speakers which are directed towards an individual performer, a sound operator or entire sections of a stage. Monitor speakers may refer to: Studio monitor loudspeakers The loudspeakers facing the stage during a live performance. ...

Headphones

Headphones are typically used by the sound board operator to monitor specific signals or the entire mix. Performers may use headphones as monitors as well, especially recently as in ear monitors are becoming more popular^{[citation needed]}. // Radio headphones c. ...

Accessories

Cables

• For the full article, see High-end audio cables

High quality coaxial audio cable. ...

Equipment racks

• For the full article, see 19-inch rack

Equipment mounted in several 19-inch racks. ...

Applications

Religious facilities

Tour sound systems

Sports facilities

Live theaters

Lecture halls

Conference rooms

Weekend bands

These systems are small enough to fit into a minivan yet powerful enough to give adequate and even sound dispersion and vocal intelligibility in a small club or bar.

Audio visual (AV) rental systems

Live music clubs

System testing

A sound reinforcement system should be able to accurately reproduce a signal from its input, through any processing, to its output, without any coloration or distortion. However, due to inconsistencies in venue sizes, shapes, building materials, and even crowd densities, this is not always possible without prior calibration of the system. This can be done in one of several ways:

Manual Calibration

The oldest method of system calibration involves a set of healthy ears, test program material (i.e. music or speech), a graphic equalizer, and last but certainly not least, a familiarity with the proper (or desired) frequency response. One must then listen to the program material through the system, take note of any noticeable frequency changes or resonances, and subtly correct them using the equalizer. However, simply relying on a prior knowledge of how the program material "should" sound can be very subjective, and should be avoided if at all possible. A more objective method of manual calibration requires a pair of high-quality headphones patched into the input signal before any processing (such as the pre-fade-listen of the test program input channel of the mixing console, or the headphone output of the CD player or tape deck). One can then use this direct signal as a near-perfect reference with which to find any differences in frequency response. This method may not be perfect, but it can be very helpful with limited resources or time, such as using pre-show music to correct for the changes in response caused by the arrival of a crowd. Rat, Dave. When Hearing Starts To Drift. Retrieved on 2007-04-26. 2007 (MMVII) is the current year, a common year starting on Monday of the Gregorian calendar and the Anno Domini era. ... April 26 is the 116th day of the year in the Gregorian calendar (117th in leap years). ...

Because this is still a very subjective method of calibration, and because the human ear is so dynamic in its own response, the program material used for testing should be as similar as possible to that for which the system is being used.

Electronic Calibration

Since the development of digital signal processing (DSP), there have been many pieces of equipment and computer software designed to shift the bulk of the work of system calibration from human auditory interpretation to software algorithms that run on microprocessors.

System design

Installing, commissioning and documenting a system

Terminology

There is disagreement over when to call an audio system a Sound Reinforcement (SR) system or a Public Address (PA) system. Some distinguish between the two by technology and capability, while others distinguish by intended use. This distinction is important to professionals in some marketplaces, while in other marketplaces the terms are interchangeable.^[7]

See also

• Public Address system (PA system)

A public address system, abbreviated PA system, is an electronic amplification system used as a communication system in public areas. ...

Footnotes

References

• Davis, Gary, and Ralph Jones. Sound Reinforcement Handbook. 2nd ed. Milwaukee: Hal Leonard Corporation, 1989.
• Eargle, John, and Chris Foreman. Audio Engineering for sound reinforcement. Milwaukee: Hal Leonard Corporation, 2002.

Further reading

Websites

• Professor PA: Sound Reinforcement system advice and tips

• ProSoundWeb: Sound Reinforcement - "Your Sound Reinforcement Center"

• LiveSoundWeb -- Help for the Beginner

• DB Acoustics, Inc - Your Eastern Iowa Point of Contact

Journals

• Live Sound International Online

Books

• AES Sound Reinforcement Anthology, Vols. 1 and 2, Audio Engineering Society, New York (1978 & 1996).
• Ahnert, W. & Steffer, F., Sound Reinforcement Engineering, SPON Press, London, 2000. ISBN 04-1921-810-6
• Alten, Stanley R. Audio in Media (5th ed.), Wadsworth, Belmont, CA (1999). ISBN 05-3454-801-6
• Ballou, Glen. Handbook for Sound Engineers, Third Edition. Oxford: Focal Press, 2005. ISBN 02-4080-758-8
• Benson, K. Audio Engineering Handbook. New York: McGraw-Hill, 1988. ISBN 00-7004-777-4
• Borwick, J. (ed.), Loudspeaker and Headphone Handbook (3rd ed.), Focal Press, Boston 2001. ISBN 02-4051-578-1
• Brawley, J. (ed.), Audio systems Technology#2 - Handbook for Installers and Engineers, National Systems Contractors Association (NSCA), Cedar Rapids, IA. ISBN 07-9061-163-5
• Buick, Peter. Live Sound: PA for Performing Musicians, PC Publishing, Kent, U.K. 1996. ISBN 18-7077-544-9
• Colloms, Martin. High Performance Loudspeakers. Chichester: John Wiley & Sons, 2005. ISBN 04-7009-430-3
• Davis, D. & C. Sound System Engineering, second edition, Focal Press, Boston 1997. ISBN 02-4080-305-1
• Dickason, V. The Loudspeaker Cookbook (5th ed.), Audio Amateur Press, Peterborough, NH 1995. ISBN 09-6241-917-6
• Eargle, J. Electroacoustical Reference Data, Kluwer Academic Publishers, Boston 1994. ISBN 04-4201-397-3
• Eargle J. Loudspeaker Handbook, Kluwer Academic Publishers, Boston 1997. ISBN 14-0207-584-7
• Eargle, J. The Microphone Book, Focal Press, Boston 2001. ISBN 02-4051-961-2
• Eiche, Jon F. The Yamaha Guide to Sound Systems for Worship, Hal Leonard Corp., Milwaukee, WI 1990. ISBN 07-9350-029-X
• Fry, Duncan. Live Sound Mixing (3rd ed.), Roztralia Productions, Victoria Australia 1996. ISBN 99-9635-270-6
• Giddings, Philip. Audio Systems Design and Installation (2nd ed.), Sams, Carmel, Indiana (1998). ISBN 06-7222-672-3
• JBL Professional, Sound System Design Reference Manual, Northridge, CA 1999. (ebook) [1]
• Langford-Smith, F. (Ed.), Radiotron Designers' Handbook, 4th ed., Amalgamated Wireless Valve Co. Pty Ltd, Sydney, 1952; CD-ROM, Old Colony Sound Labs, Peterborough, NH; reprinted as Radio Designer's Handbook, *Newnes, Butterworth-Heineman Ltd. 1997
• Moscal, Tony. Sound Check: The Basic of Sound and Sound Systems, Milwaukee, WI; Hal Leonard Corp. 1994. ISBN 07-9353-559-X
• Olson, H., Acoustical Engineering, D. Van Nostrand, New York 1947. Reprinted by Professional Audio Journals, Inc., Philadelphia 1991.
• Oson, H.F. Music, Physics and Engineering, Dover, New York 1967. ISBN 04-8621-769-8
• Pohlmann, Ken, Principles of Digital Audio (5th ed.), McGraw-Hill, New York 2005 ISBN 00-7144-156-5
• Stark, Scott H. Live Sound Reinforcement, Bestseller edition. Mix Books, Auburn Hills, MI 2004. ISBN 15-9200-691-4
• Streicher, Ron & F. Alton Everest. The New Stereo Soundbook (2nd ed.), Audio Engineering Associates, Pasadena, CA 1998. ISBN 09-6651-620-6
• Talbot-Smith, Michael (Ed.) Audio Engineer's Reference Book, 2nd ed. Focal Press, Butterworth-Heinemann Ltd. 2001. ISBN 02-4051-685-0
• Trubitt, David. Concert Sound: Tours, Techniques & Technology, Mix Books, Emeryville, CA 1993. ISBN 07-9352-073-8
• Trubitt, Rudy, Live Sound for Musicians, Hal Leonard Corp., Milwaukee, WI 1997. ISBN 07-9356-852-8
• Trynka, P. (Ed.), Rock Hardware, Blafon/Outline Press, London: Miller Freeman Press, San Francisco 1996. ISBN 08-7930-428-6
• Vasey, John. Concert Sound and Lighting Systems (3rd ed.) Focal Press, Boston 1999. ISBN 02-4080-364-7
• Whitaker, Jerry. AC Power Systems Handbook, Third Edition. Boca Raton: CRC, 2006. ISBN 08-4934-034-9
• Whitaker, Jerry and K. Benson. Standard Handbook of Audio and Radio Engineering. New York: McGraw-Hill, 2002. ISBN 00-7006-717-1
• White, Glenn and Gary J. Louie. The Audio Dictionary. Seattle: University of Washington Press, 2005. ISBN 02-9598-498-8
• White Paul, the Sound On Sound book of Live Sound for the Performing Musician, Sanctuary Publishing Ltd, London (2005).ISBN 18-6074-210-6
• Yakabuski, Jim, Professional Sound Reinforcement Techniques: Tips and Tricks of a Concert Sound Engineer, Mix Books, Vallejo, CA 2001. ISBN 08-7288-759-6

Papers

• Benson, J.E. "Theory and Design of Loudspeaker Enclosures," Amalgamated Wireless Australia Technical Review, (1968, 1971, 1972).
• Beranek, L., "Loudspeakers and Microphones," J. Acoustical Society of America, volume 26, number 5 (1954).
• Damaske, P., "Subjective Investigation of Sound Fields," Acustica, Vol. 19, pp. 198-213 (1967-1968).
• Davis, D & Wickersham, R., "Experiments in the Enhancement of the Artist's Ability to Control His Interface with the Acoustic Environment in Large Halls," presented at the 51st AES Convention, 13-16 May 1975; preprint number 1033.
• Eargle J. & Gelow, W., "Performance of Horn Systems: Low-Frequency Cut-off, Pattern Control, and Distortion Trade-offs," presented at the 101st Audio Engineering Society Convention, Los Angeles, 8-11 November 1996. Preprint number 4330.
• Engebretson, M., "Low Frequency Sound Reproduction," J. Audio Engineering Society, volume 32, number 5, pp. 340-352 (May 1984)
• French, N. & Steinberg, J., "Factors Governing the Intelligibility of Speech Sounds," J. Acoustical Society of America, volume 19 (1947).
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