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Sound reinforcement system

A sound reinforcement system is an arrangement of microphones, electronic signal processors, amplifiers, and loudspeakers that makes live or pre-recorded sounds—usually music or speech— louder, or that distributes the sound to a larger or more distant audience. A sound reinforcement system may be very complex, including hundreds of microphones, complex mixing and signal processing systems, tens of thousands of watts of amplification, and multiple loudspeaker arrays, all overseen by a team of audio engineers and technicians. On the other hand, a sound reinforcement system can be as simple as a small PA system in a coffeehouse, consisting of a single microphone connected to a self-powered 100-watt loudspeaker system. In both cases, these systems reinforce sound to make it louder or distribute it to a wider audience.

Audio engineers and other sound industry professionals disagree over whether these audio systems should be called Sound Reinforcement (SR) systems or Public Address (PA) systems. Some audio engineers distinguish between the two terms by technology and capability, while others distinguish by intended use (e.g., SR systems are for live music whereas PA systems are usually for reproduction of speech and recorded music in buildings and institutions). In some regions or markets, the distinction between the two terms is important. The terms are also considered interchangeable in some areas.

Basic concept

A typical sound reinforcement system consists of three parts: input transducers such as microphones, which convert sound energy into an audio signal; signal processors such as equalizers and amplifiers, which alter the audio signal characteristics; and output transducers such as loudspeakers, which convert the audio signal into sound for an audience.

Sound is taken and converted into electronic signal by an input transducer (such as a microphone or a pickup on an electric guitar or electric bass). A signal processor (such as a mixing console, amplifier, a reverb effect, or other devices) then alters the signal's equalization, balance, effects and amplitude. Finally, an output transducer such as a loudspeaker (or, for audio engineers, a set of 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 some sound engineers, a "Sound Reinforcement System" is a system where the audience could hear the speaker or the singer at the microphone without electronic assistance, such as in the case of a Minister addressing a congregation in a church. The components of the SR system amplify or "reinforce" the sound one hears to make it more intelligible. The "PA", or "Public Address System", also known as "commercial paging systems", tend more to be classified as a system where one could not at all hear the person at the microphone without amplification, as in the case of a school principal addressing all of the classrooms in a school through a PA system. In these PA systems, or distributed sound systems, there are multiple speakers located throughout a business, institution, or large complex.

Signal path

Sound reinforcement in a large format system typically uses a signal path which starts with the directly-connected instrument or a microphone (transducer) which is plugged into the multicore cable (often called a "snake"). The snake routes the signals of all of the inputs to the Front of the House mixer and to the monitor consoles. Once the signal is in a channel on the console, this signal can be equalized, panned and routed to various mix buses. The signal may also be patched into an external effect processor present in the channel (e.g. gate, compressor, reverb).

The signal can be routed internally to a summation bus (also known as a "bus", mix group, subgroup or simply 'group'), to allow the engineer to control the levels of several related signals at once. For example, all of the different microphones for a drum set might be grouped together, so that the volume of the entire drum set sound can be controlled with a single fader. If an engineer was trying to record the different instruments of drumset with ten microphones and mix the sound into a rock song, and they did not have a bus or group function on their mixer, they would have to lower each individual mic fader to reduce the overall volume of the drums; with a mixer that has a "mix group" function, all of the faders in the group can be controlled by a single master fader. From here each signal is 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. Low-frequency sounds are usually sent to subwoofers, and middle- and high-frequency sounds are usually sent to full-range speaker cabinets.

System components

Input transducers

Many types of input transducers can be found in a sound reinforcement system, with microphones being the most commonly-used input device. They can be classified according to their method of transduction, pickup (or polar) pattern or their functional application. Most microphones used in sound reinforcement are either dynamic or condenser microphones.

Microphones used for sound reinforcement can be positioned and mounted in many ways, including base-weighted upright stands, podium mounts, tie-clips, instrument-mounted and headset mounts. Headset mounted and tie-clip mounted microphones are often used with wireless transmission to allow performers or speakers to move freely. Early adopters of headset mounted microphones technology included country singer Garth Brooks, Kate Bush, and Madonna.

There are many other types of input transducers which may be used occasionally, including magnetic pickups used in electric guitars and electric basses, contact microphones used on stringed instruments, and piano and phonograph pickups (cartridges) used in record players .

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. Multiple consoles can be used for different applications in a single sound reinforcement system. In sound reinforcement, the main or FOH (Front Of House) mixing console must be located where the operator can see and hear the action on stage. Some venues with permanent systems installed (e.g. religious facilities and theaters) place the mixing console within an enclosed booth, but this approach is more appropriate for broadcast and recording applications. In sound reinforcement, mixing from an enclosed booth prevents the operator from hearing the combined effect of the artist, the loudspeakers, the audience, the mix and the acoustics of the room.

Large music productions often use a separate stage monitor mixing console, which is dedicated to creating mixes for the performers' on-stage monitors. These consoles are often placed at the side of the stage so that the operator can communicate with the performers on stage. In cases where performers have to play at a venue that does not have a "Front of House" monitor engineer near the stage, the monitor mixing is done by the main sound engineer from from the main console, which is in the audience or in the back of the hall. This arrangement can be problematic, because the performers end up having to request changes to the monitor mixes with "...hand signals and clever cryptic phrases", and because the sound engineer cannot hear the changes that result in the monitors onstage.

Equalizers

Equalizers exist in sound reinforcement systems in two forms: Graphic and Parametric. Both of these are used in conjunction with End-cut filters. Parametric equalizers have knobs that 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.

Graphic equalizers have faders which resemble a frequency response curve plotted on a graph. Sound reinforcement systems normally use graphic equalizers designed on one-third octave centers. End-cut filters restrict a given channels bandwidth extremes, which can prevent 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 feedback suppressor is a specialized type of filter which automatically detects and suppresses feedback by cutting a deep notch on the frequency which is feeding back.

Compressors

Compressors are designed to manage the dynamic range of an input signal. The compressor accomplishes this by reducing the level of a signal above a defined level (threshold) by a defined amount (ratio). For example, a compressor with a 2:1 ratio provides 1dB of gain reduction for every 2dB that a signal rises above the threshold. A ratio of 4:1 would provide 3dB of gain reduction for every 4dB of signal rise above threshold (RAT). Most compressors available today are designed to allow the operator to select a ratio within a selected range (typically between 1:1 and 8:1, with some allowing settings of up to ∞:1 [maximum compression]). A compressor with an infinite ratio is typically referred to as a "limiter".

Portions of the signal that function below the threshold are theoretically unaffected by the process. A theoretically perfect compressor would instantaneously transition from 1:1 to x:1 when the signal rises to any degree above the threshold. Due to design factors analog compressors cannot theoretically accomplish this perfectly. The area of transition between uncompression and full compression is referred to as the "knee" of a compressor. It is in some cases desirable for a compressor to affect a smooth transition from uncompression to full ratio implementation. One example of this principle in application is the "Overeasy" feature provided by dbx. This function provides a subjectively more pleasing sonic character for some signals acting at or near a threshold setting by gradually transitioning the ratio from 1:1 to x:1 rather than affecting an immediate change from 1:1 to x:1 at the threshold point.

Most presently available compressors also allow the user to increase the overall output level to account for losses due to gain reduction in order to ensure optimal signal-to-noise performance for devices receiving the compressed signal. This is typically referred to as "make-up gain". An input signal, compressed and properly "made-up" yields a signal with a higher average level at the compressors output.

Many compressors include user-adjustable "attack" and "release" times. These enable the user to alter the speed at which the compressor reduces (attacks) and restores (releases) gain. These settings are most often ennumerated using milliseconds, although many compressors allow release times in excess of 2 seconds. Techniques for determining ideal time-dynamic criteria are largely subjective, varying with application and operator preference. It is important to note that altering attack and release times alters the compressive process, creating variances in amounts of gain reduction for identical ratio and threshold settings. i.e. - Attack times that are slow when compared with signal composition may not affect compression for all signal transients.

Compressor applications vary widely, from objective system design criterion, to subjective applications determined by variances in program material and operator preference. Typical system design criterion specify limiters for component protection and gain structure control. Artistic signal manipulation is a subjective technique widely utilized by mix engineers in both live production and studio environments.

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. There are also attack and release settings, which work in the same way as a compressor. Noise gates are useful for microphones which will pick up noise which is not relevant to the program, such as the hum of a miked electric guitar amplifier or the rustling of papers on a Minister's podium.

Noise gates are also used with the microphones placed near the different drums and cymbals in the drum kits in many hard rock and metal bands. Without a noise gate, the microphone for a specific instrument, such as the floor tom, will also pick up sounds from nearby drums or cymbals, which bleed into its microphone. With a noise gate, the threshold of sensitivity for each microphone for each instrument from the drum kit can be set so that only the direct striking of the instrument will be picked up by the microphone, and not the nearby sounds.

While the noise gate is sometimes misunderstood as a type of filter, which removes the unwanted sounds, it should be noted that the device does not remove any sounds from the music or speech. Instead, it opens and closes an electronic "gate" when there is no, or very little signal present. In this fashion, when a speaker pauses in a conference and looks through their notes, the microphones will not pick up the much quieter sounds of papers rustling. In a musical context, if there is a guitar amp which is humming, the most noticable point for the hum is usually in between songs, or during the rests when the guitarist is not playing. With the noise gate set to the proper threshold, the hum will not be picked up during the points where the guitarist is not playing.

Other effects and accessories

Reverberation and Delay (audio effect) effects are widely used in sound reinforcement systems to give the effect of natural reverb. Less commonly, modulation effects such as Flanging and Phaser (effect)s are applied to some instruments for an unusual sound effect. The Exciter (effect) "livens up" the sound of audio signals by applying dynamic equalization, phase manipulation and harmonic synthesis of (usually) high frequency signals.

A wide range of accessories are used in sound reinforcement systems. High-end audio cables are shielded to prevent hum and interference. Rack-mount cases, such as the industry standard 19-inch racks are used to store and transport effects units and amplifiers. Some racks have cushioned shock-mounting to protect equipment from impacts.

Power Amplifiers

Power Amplifiers increase the signal level, and provide current to drive the loudspeaker. All output transducers require amplification of the signal by amplifiers, including loudspeakers, monitor speakers, and even headphones. Most professional amplifiers provide protection from overdriven ("clipped") signals, short circuits across the output, or thermal overload. Compression and limiting features are often used to protect loudspeakers and amplifiers from signal overload.

Like most sound reinforcement equipment, professional amplifiers are designed to be mounted on 19-inch racks. Many power amplifiers have internal fans to draw air across the heat sinks. 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

Since transistor power amplifiers used for large venues need to produce a high output, this usually means that most powerful amplifiers are very heavy. Most powerful amplifiers are Class AB amplifiers, which need bulky transformers made of copper wiring and large metal heat sinks for cooling. However, Class D amplifiers, which are much more efficient, weigh much less than Class-AB amplifiers producing an equivalent power output.

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. 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. 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.

Output transducers

Loudspeakers

Loudspeakers (also known as drivers) are the main speakers which project the sound to the audience. A basic, inexpensive PA speaker may only have a single full-range loudspeaker system. Professional PA speakers usually have different drivers that provide the low-, middle-, and high frequency sounds. In the 1960s, most PA speakers were "columns" with several speakers mounted in a tall cabinet, which evolved into cube-shaped speakers mounted on stands by the end of the decade. In the 1970s, PA speakers developed into stacks of "bass bin"s and high-range horns. By the 1980s, PA speakers became easier to use and set up, because they increasingly contained a range of different drivers (woofer and horn) with built-in crossovers.

In the 1990s and 2000s, professional PA speakers were often "actively controlled" with electronic processors that automatically adjust crossover and equalization settings and protect the speakers. In the 2000s, PA speaker cabinets were increasingly built with protection circuitry that protects high and mid frequency drivers from accidental exposure to low frequency sound material. Some PA speakers also protect the high frequency drivers with "current-to-light" conversion circuitry, which takes excess current which would otherwise damage a horn, and uses it to light a small light bulb. Another circuit protection technique used to protect horns is current sensing / self-resetting breakers, which protects the horn in cases of high-volume feedback (e.g., a microphone being accidentally dropped into a monitor).

As well, many PA speaker companies have begun providing Neutrik Speakon multi-pin connectors, instead of the decades-old 1/4" jacks. The Speakon connectors provide a much larger metal contact area for high current PA applications.

The four different types of transducers (woofers, compression drivers, and tweeters) all use the same components: a voicecoil, magnet, cone or diaphragm, and a frame or structure. PA speakers have a power rating (in watts) which indicates their maximum power capacity, to help users avoid overloading them with excessively powerful amplifiers. However, even an amplifier with less power output than a speaker's power rating can destroy the speakers if the amplifier signal becomes heavily distorted, especially if the distortion is from low-range sounds. The power rating of speakers is expressed either as RMS (Root Mean Square) or PGM (Program). The PGM rating became more widely used in the 2000s; it means "do not use more than the indicated power with this speaker." Trapezoidal or "wedge-shaped" speakers have a shape which allows the speakers to be grouped into arrays so that they can be mounted on rigging.

Professional PA speakers are usually designed with internal brace "flying" hardware (e.g., steel eyebolts) for safe suspension of speakers from rigging or ceilings. Many speakers are designed with interlocking corners so that they can be vertically or horizontally stacked for large concerts. Large, heavy PA speakers are often equipped with wheeled dollys, to facilitate on-stage placement.

Monitors

Speakers

Monitor speakers are usually full-range speakers which are directed towards an individual performer, a sound operator or entire sections of a stage. Most monitor speaker cabinets have a wedge shape, so that they will point up towards the performer when they are placed on their sides on the stage. Monitor speaker cabinets usually contain a speaker (driver) for low- and mid-frequency sounds and a horn for high-frequency sounds. Some monitor speakers include an L-pad for controlling the volume of the horn. In the 2000s, PA speaker companies began offering a range of powered monitor speakers, which contain a power amplifier. Another trend of the 2000s was the blurring of the lines between monitor speaker cabinets and regular speaker cabinets; many companies began selling wedge-shaped full-range speakers which they state can be used for monitors or regular Front of House purposes.

Headphones

Headphones are typically used by the sound board operator to monitor specific channels or to listen to the entire mix. Some performers may use headphones as monitors as well, such as drummers in pop music bands. In the 2000s, some bands and singers have begun using small "in ear"-style headphone monitors. In-ear monitors allow musicians to hear their voice and the other instruments with a clearer, more intelligible sound, because the molded in-ear headphone design blocks out on-stage noise. While some in-ear monitors are "universal fit" designs, some companies also sell custom-made in-ear monitors, which require a fitting by an audiologist. Custom-made in-ear monitors provide an exact fit for a performer's ear.

Applications

Sound reinforcement systems are used in a broad range of different settings, each of which poses different challenges.

Church sound

Designing systems in churches and similar religious facilities often poses a challenge, because the speakers have to be unobtrusive to blend in with antique woodwork and stonework. In some cases, audio designers have designed custom-painted speaker cabinets so that the speakers will blend in with the church architecture. Some church facilities, such as sanctuaries or chapels are long rooms with low ceilings, which means that additional fill-in speakers are needed throughout the room to give good coverage. An additional challenge with church SR systems is that, once installed, they are often operated by amateur volunteers from the congregation, which means that they must be easy to run and troubleshoot.

Some mixing consoles designed for houses of worship have automatic mixers, which turn down unused channels to reduce noise, and automatic feedback elimination circuits which detect and notch out frequencies that are feeding back. These features may also be available in multi-function consoles used in convention facilities and multi-purpose venues.

Touring systems

Touring sound systems have to be easy to set up, and they need to use "field-replaceable" components such as speakers, horns, and fuses, which are easily accessible for repairs during a tour. Tour sound systems are often designed with substantial redundancy features, so that in the event of equipment failure or amplifier overheating, the system will continue to function.

Weekend band PA systems are a niche market for touring SR gear. Weekend bands need systems that are small enough to fit into a minivan or a car trunk, and yet powerful enough to give adequate and even sound dispersion and vocal intelligibility in a noisy club or bar. As well, the systems need to be easy and quick to set up. Sound reinforcement companies have responded to this demand by offering equipment that fulfills multiple roles, such as "amp-mixers" (a mixer with an integrated power amplifier and effects) and powered subwoofers (a subwoofer with an integrated power amplifier and crossover). These products minimize the amount of wiring connections that bands have to make to set up the system. Some subwoofers have speaker mounts built into the top, so that they can double as a base for the stand-mounted full-range PA speaker cabinets.

Sports sound systems

Systems for outdoor sports facilities and ice rinks often have to deal with substantial echo, which can make speech unintelligible. Sports and recreational sound systems often face environmental challenges as well, such as the need for weather-proof outdoor speakers in outdoor stadiums and humidity- and splash-resistant speakers in swimming pools.

Live theater

Sound for live theaters, opera theaters, and other dramatic applications may pose problems similar to those of churches, in cases where a theater is an old heritage building; speakers and wiring may have to blend in with woodwork. As well, the need for clear sight lines in some theaters may make the use of regular speaker cabinets unacceptable; instead, slim, low-profile speakers may be needed.

In live theater and drama, performers move around onstage, which means that wireless microphones may have to be used. Wireless microphones need to be set up and maintained properly, to avoid interference and reception problems.

Classical music and opera

A subtle type of sound reinforcement called acoustic enhancement is used in some concert halls where classical music such as symphonies and opera is performed. Acoustic enhancement systems help give a more even sound in the hall and prevent "dead spots" in the audience seating area by "...augment[ing] a hall's intrinsic acoustic characteristics." The systems use "...an array of microphones connected to a computer [which is] connected to an array of loudspeakers." However, as concertgoers have become aware of the use of these systems, debates have arisen, because "...purists maintain that the natural acoustic sound of [Classical] voices [or] instruments in a given hall should not be altered.

Kai Harada's article Opera's Dirty Little Secret states that opera houses have begun using electronic acoustic enhancement systems "...to compensate for flaws in a venue's acoustical architecture." Despite the uproar that has arisen amongst operagoers, Harada points out that note of the opera houses using acoustic enhancement systems "...use traditional, Broadway-style sound reinforcement, in which most if not all singers are equipped with radio microphones mixed to a series of unsightly loudspeakers scattered throughout the theatre." Instead, most opera houses use the sound reinforcement system for acoustic enhancement, and for subtle boosting of offstage voices, onstage dialogue, and sound effects (e.g., church bells in Tosca or thunder in Wagnerian operas).

Acoustic enhancement systems include LARES (Lexicon Acoustic Reinforcement and Enhancement System) and SIAP, the System for Improved Acoustic Performance. These systems use microphones, computer processing "with delay, phase, and frequency-response changes", and then send the signal "... to a large number of loudspeakers placed in extremities of the performance venue." Another acoustic enhancement system, VRAS (Variable Room Acoustics System) uses "...different algorithms based on microphones placed around the room." The Deutsche Staatsoper in Berlin and the Hummingbird Centre in Toronto use a LARES system. The Ahmanson Theatre in Los Angeles, the Royal National Theatre in London, and the Vivian Beaumont Theatre in New York City use the SIAP system.

Lecture halls and conference rooms

Lecture halls and conference rooms pose the challenge of reproducing speech clearly to a large hall, which may have reflective, echo-producing surfaces. In some conferences, sound engineers have to provide microphones for a large number of people, in the case of a panel conference or debate. In some cases, automatic mixers are used to control the levels of the microphones.

Rental systems

Audio visual (AV) rental systems have to be able to withstand heavy use, and even abuse from renters. For this reason, rental companies tend to own speaker cabinets which are heavily braced and protected with steel corners, and electronic equipment such as power amplifiers or effects are often mounted into protective road cases. As well, rental companies tend to select gear which has electronic protection features, such as speaker-protection circuitry and amplifier limiters.

As well, rental systems need to be easy to use and set up, and they must be easy to repair and maintain for the renting company. From this perspective, speaker cabinets need to have easy-to-access horns, speakers, and crossover circuitry, so that repairs or replacements can be made. Since rental gear is often used by fairly inexperienced users, rental companies often rent powered amplifier-mixers, mixers with onboard effects, and powered subwoofers, which are easier to set up and use.

Live music clubs

Setting up sound reinforcement for live music clubs often poses unique challenges, because there is such a large variety of venues which are used as clubs, ranging from former warehouses or music theaters to small restaurants or basement pubs with concrete walls. In some cases, clubs are housed in multi-story venues with balconies or in "L"-shaped rooms, which makes it hard to get a consistent sound for all audience members. The solution is to use fill-in speakers to obtain good coverage, using a delay to ensure that the audience does not hear the same sound at different times.

Another problem with designing sound systems for live music clubs is that the sound system may need to be used for both prerecorded music played by DJs and live music. If the sound system is optimized for prerecorded DJ music, then it will not provide the appropriate sound qualities needed for live music, and vice versa. Lastly, live music clubs can be a hostile environment for sound gear, in that the air may be hot, humid, and smoky; in some clubs, keeping amplifiers cool may be a challenge.

Setting up and testing

Large-scale sound reinforcement systems are designed, installed, and operated by audio engineers and audio technicians. During the design phase of a newly constructed venue, audio engineers work with architects and contractors, to ensure that the proposed design will accommodate the speakers and provide an appropriate space for sound technicians and the racks of audio equipment. During the design phase of a venue, sound engineers provide advice on which audio components would best suit the space and its intended use, and on the correct placement and installation of these components. During the installation phase, sound engineers ensure that high-power electrical components are safely installed and connected and that ceiling- or wall-hung speakers are properly mounted (or "flown") onto rigging. When the sound reinforcement components are installed, the sound engineers test and calibrate the system, so that its sound production will suit the acoustic environment of the venue.

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:

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.

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.

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.

One modern tool for calibrating a sound system using either DSP or Analog Signal Processing is a Real Time Analyzer (RTA). This tool is usually used by piping pink noise into the system and measuring the result with a special calibrated microphone connected to the RTA. Using this information the system can be adjusted to help achieve the desired response.

Footnotes

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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).
Gander, M. & Eargle, J., "Measurement and Estimation of Large Loudspeaker Array Performance", J. Audio Engineering Society, volume 38, number 4 (1990).
Henricksen, C. & Ureda, M., "The Manta-Ray Horns", J. Audio Engineering Society, volume 26, number, pp. 629-634 (September 1978).
Hilliard, J., "Historical Review of Horns Used for Audience-Type Sound Reproduction", J. Acoustical Society of America, volume 59, number 1, pp. 1 - 8, (January 1976)
Houtgast, T. and Steeneken, H., "Envelope Spectrum Intelligibility of Speech in Enclosures", presented at IEEAFCRL Speech Conference, 1972.
Klipsch, P. "Modulation Distortion in Loudspeakers: Parts 1, 2, and 3" J. Audio Engineering Society, volume 17, number 2 (April 1969), volume 18, number 1 (February 1970), and volume 20, number 10 (December 1972).
Lochner, P. & Burger, J., "The Influence of Reflections on Auditorium Acoustics", Sound and Vibration, volume 4, pp. 426-54 (196).
Meyer, D., "Digital Control of Loudspeaker Array Directivity", J. Audio Engineering Society, volume 32, number 10 (1984).
Peutz, V., "Articulation Loss of Consonants as a Criterion for Speech Transmission in a Room", J. Audio Engineering Society, volume 19, number 11 (1971).
Rathe, E., "Note on Two Common Problems of Sound Reproduction", J. Sound and Vibration, volume 10, pp. 472-479 (1969).
Schroeder, M., "Progress in Architectural Acoustics and Artificial Reverberation", J. Audio Engineering Society, volume 32, number 4, p. 194 (1984)
Smith, D., Keele, D., and Eargle, J., "Improvements in Monitor Loudspeaker Design", J. Audio Engineering Society, volume 31, number 6, pp. 408-422 (June 1983).
Toole, F., "Loudspeaker Measurements and Their Relationship to Listener Preferences, Parts 1 and 2", J. Audio Engineering Society, volume 34, numbers 4 & 5 (1986).
Veneklasen, P., "Design Considerations from the Viewpoint of the Consultant", Auditorium Acoustics, pp.21-24, Applied Science Publishers, London (1975).
Wente, E. & Thuras, A., "Auditory Perspective — Loudspeakers and Microphones", Electrical Engineering, volume 53, pp.17-24 (January 1934). Also, BSTJ, volume XIII, number 2, p. 259 (April 1934) and Journal AES, volume 26, number 3 (March 1978).