Sound Reinforcement

Sound reinforcement refers to the devices and processes used to amplify sound. A sound reinforcement system—or sound system for short—can be found in venues like halls, theaters, arenas, and stadiums. They are used for musical performances, speeches, sporting events, and other forms of public address. A sound system typically consists of microphones, mixing consoles, loudspeakers, and various computer-based components.

There are three main parts of sound reinforcement: (1) input transducers, (2) signal processing, and (3) output transducers.[1]

There are three main events in the sound-reinforcement process: transduction, signal processing, and loudspeaker amplification.

Input Transducers

Transducers occupy the first part of a sound-reinforcement process, which entails getting musical signals into your electronic equipment. A transducer is a device that turns one form of energy into another. A microphone is a transducer because it turns sound-wave energy into electrical energy, and a loudspeaker is a transducer because it turns electrical energy into sound-wave energy.

A transducer is the operative device at the heart of electric guitars and basses, which create musical electricity via magnetic pickups. A typical magnetic pickup consists of a set of magnets and a coil of copper. If such an apparatus is placed in proximity to the vibrating strings of an instrument, then the vibrations will be converted (transduced) into voltage.

Voltage is the potential for a water-like flow of electrons. The water-like flow of electrons itself (not the potential for flow, but the actual flow) is called amperage. To use an analogy, you can think of voltage as water built up behind the spigot and amperage as water flowing through the garden hose.

Analog audio is the term audio engineers use for amperages resulting from voltages transduced by magnetic pickups, microphones, and other transducers.

A Hammond organ is another instrument that uses magnetic pickups. The device at the heart of a Hammond is a tonewheel, which sees a magnetic pickup placed in proximity to a spinning, toothed disc that hums a pitch—not unlike a vacuum-cleaner motor. To clarify the tonewheel concept in your mind, imagine the sound of a vacuum; then, imagine converting that sound into electricity with a coil and a magnet—that’s a tonewheel.

A piezo pickup is another common transducer. Piezo pickups, or piezos for short, are designed to be in contact with dense substances like the wood of a guitar, violin, or mandolin; hence, they are often called contact pickups. Piezo pickups operate via piezo electricity, which is an electrical phenomenon produced by crystalline substances held under pressure. Here’s how it works: If you apply pressure to a piezo, you’ll get a voltage. If you modulate this pressure, say, by playing different notes on an instrument to which a piezo pickup has been installed, then you’ll get a voltage that’s modulated to the same degree as the notes.

Piezo pickups and magnetic pickups are quite common in the pro-audio world, but other pickup varieties—like the phonograph pickup and the tape head—are increasingly rare. A phonograph pickup is to be found on record players, and a tape head is to be found on tape decks. Obviously, there are not too many record players and tape decks involved in modern sound reinforcement. So, you can safely set these transducers on some dusty shelf in the back of your mind. It’s likely that we won’t be needing them, but they deserve mention for historical purposes. Once upon a time, phonograph transducers were the only ones that existed.

In the realm of movie projection, there is a type of transducer that interprets variations in density and transparency of film. Such a transducer is known as an optical pickup. Like the phonograph and the tape deck, optical pickups no longer figure prominently in sound reinforcement or professional audio of any kind.

Devices that do figure prominently in modern sound reinforcement, however, are digital pickup systems—which are found inside audio interfaces like those made by Presonus, M-Audio, Focusrite, and so on. These pickup systems are designed to operate between analog equipment and computers and feature capabilities for both analog-to-digital conversion and digital-to-analog conversion. CD players are also a variety of digital pickup, but they are basically obsolete nowadays. We can safely set them on the same dusty shelf along with our record player, tape deck, and film reader.

Signal Processing

Once sound has been converted to audio via transducers, the next step is signal processing. Signal processing is a self-defining term: it means to process the signal with devices that amplify and alter sound.

Preamplification is the first step in signal processing, for transduced audio needs to be beefed up to be useful. Beefed up in this context means “increased in voltage.” The problem is that the voltages produced by pickups and microphones are quite weak and easily polluted by interference and noise; hence, the need to enlarge their voltage. This process of enlargement is performed by a device called a preamplifier. Preamplifiers—preamps for short—come in several guises. The most common guises are the microphone and instrument inputs that are found on mixing boards, studio consoles, and audio interfaces. Preamps can also be found in guitar amplifiers, bass amplifiers, direct boxes, and other mechanisms designed to turn weak signals into stronger ones.

Once an audio signal has been sufficiently amplified to be useful, its frequency profile is usually altered via equalization. Equalization refers to processing sound at the level of its frequency. Some audio has too much bass, some has too much treble, some has too little bass and treble, some has too little midrange, and so on. The solution to these problems—and many others like them—is to process the deficient signal with an equalizer, which is an electronic tone control device that operates like a volume knob for specific frequencies or swaths of frequencies. Equalizers are sometimes called spectrum processors, filters, or just EQs.

Mixing is perhaps the most critical stage of the whole sound reinforcement process, for mixing entails balancing volume, pan, dynamics, reverberation, and other effects. If all areas of the sound reinforcement system are in good working order, properly set up, configured beautifully, winning awards for installation brilliance, etc., but the mix is bad, then the whole operation will be ruined and everyone in the audience will be dissatisfied. Complaints like “it was too loud,” or “I couldn’t even hear the singer,” or “they sounded nothing like their album” are evidence of a bad mix. A coherent, easily digestible, and not-too-loud mix must be constructed for any successful event, concert, or recital, especially if people are paying admission. In my experience, I’ve been disappointed by a poor mix on many occasions. My number one complaint is almost always “it was way to fucking loud and everything sounded like shit.”

The mixing console is the nexus of the PA system because it’s where all the signals are fed, summed, manipulated, perfected, and sent out for loudspeaker broadcast. Most of the critical adjustments that are made to professional audio are made via mixing consoles. A mixing console is adorned with the power to equalize, pan, route, and process audio signals as well as adjust overall volume and individual channel volume.

Effects are an important component of the sound reinforcement process. In the realm of pro audio, effects include things like dynamics, reverberation, and delay. Dynamic processors control difference between loud and soft sounds, reverberation processors simulate performance spaces with overlapping ambient echoes, and delay processors copy a sound and play it back later. Just think of your voice echoing repeatedly in a canyon after yelling—that’s delay.

Effects processors work in conjunction with mixing consoles and typically operate via an effects loop, which is an electrical hook-up that routes signals from a mixing board to a signal-processor and back to the mixing board. Nowadays, signal processing is done with onboard, digital signal processors (DSP) that reside within the mixing board itself—so effects loops are increasingly rare. In any case, once the signal has been processed and returned, it is summed with the primary signal, balanced for volume, pan, etc., and then broadcast along with everything else through the loudspeakers. The amount of signal being sent through the effects loop (both onboard and outboard) is controlled by the mixer’s auxiliary send, which is a knob variably labeled Bus, Aux, or FX send.

Pro tip:  When using effects, it makes more sense to route your affected signal into a new input instead of routing it into the auxiliary return jack. This configuration will allow you more control of the affected/unaffected balance because channel strips have EQ, most Aux returns do not. Also, with your affected sound occupying its own track, you get full control over the volume and EQ of your original signal. Conversely, when using an aux return, the unaffected and affected signals have been summed into one.

Before audio signals are sent to the output transducers, they must undergo two additional signal processes: crossover and amplification. A crossover is a device that splits an audio signal into separate channels demarcated by frequency profiles—typically, bass, midrange, and treble. The purpose of which is to send the appropriate signal to the appropriate loudspeaker enclosure. Not all systems feature crossovers, but all professional, large-scale installations do. It’s the only way to ensure high-fidelity loudspeaker broadcast. The last signal processing maneuver is amplification, which sees analog audio beefed up again—this time to truly epic proportions. Everything that happened before amplification in the sound-reinforcement process—that is, equalization, mixing, signal processing, etc.—was being performed with voltages of about 0.775, which is known as line level. Amplifiers for public address systems can create analog-audio voltages surpassing 100 volts. High voltages like these are needed to operate the mechanism at the heart of output transducers, which is where we’ll turn to next.

Output Transducers

Output transducers are devices that convert analog audio back into physical sound. Loudspeakers, or drivers, are the most common type of output transducer. There are many varieties of loudspeaker. Most speaker systems are multiway, meaning they consist of several speakers of various sizes working together in concert.

Below are the most common types of loudspeakers:

Woofer — a woofer is a low-frequency loudspeaker. They are typically large—ranging from 12, 15, 18, or 24 inches—and are designed to handle frequencies below 320 Hz. If a woofer can handle sounds below 30 or 40 Hz, then they are called subwoofers.[2]

Midrange loudspeaker — a midrange loudspeaker is one designed to manage midrange frequencies from 320 Hz to 2,560.

Tweeter — a tweeter is a type of loudspeaker designed to handle high-frequency sounds, typically ones above 5,120 Hz.

Full-range loudspeaker — this variety of speaker is designed to handle as much of the sound-frequency spectrum as possible. Full-range speakers are common in radios and personal computers and operate using a whizz cone, which is a variety of speaker cone featuring more than one diaphragm attached to the voice coil.[3]

Supertweeter — a supertweeter loudspeaker is for reproducing extremely high frequency sounds—between 8kHz and 40 kHz.[4]

Monitor — a monitor is a loudspeaker commonly found in a control booth of a recording studio, the floor of a performance stage, or on the desk of a project studio. It is designed to allow the sound technicians, recordist, producer, etc. to hear the performance being delivered by the musicians—either live or reproduced via storage medium. Sometimes monitors are called foldback speakers because of their floor-based design featuring a distinctive tilt.

Headphones — headphones are a specialized set of loudspeakers designed for personal use and are meant to be worn.

Loudspeaker Nomenclature

The loudspeakers used for live sound are categorized as either mains or monitors. The mains are the speakers that face the audience, and the monitors are the speakers that face the performers.

The mains are typically spoken about in terms of “tops,” which handle the midrange and treble frequencies, and “subs,” which handle the lower frequencies. Sometimes the tops include “horns,” which are tweeters designed for handling frequencies above 5,000 Hz. 

Multiway speaker systems require a crossover. A crossover splits the audio signal into low, mid, and high frequencies to be distributed to the ideal speaker destination. A crossover frequency is the point along the sound frequency spectrum in which the speaker duties are handed off to one another.

Speaker Arrays

Large-scale sound reinforcement systems often feature loudspeakers arranged in a configuration called an array. A speaker array is an arrangement of loudspeakers spaced to give full-range, equally distributed sound amplification in specific acoustic environments like halls, theaters, etc. Nowadays, speaker arrays are usually configured as line arrays.

Line arrays are often hung from the venue’s ceiling, or some other support structure, and feature modular, full-range speakers configured in one of several arrangements. Common arrangements include (1) straight, meaning drivers hung in a straight column, (2) arcuate, meaning drivers hung in a curve with a fixed radius, and (3) J-shaped, meaning drivers hung in a way that combines straight and arcuate modalities. There are other methods for hanging speakers, but the above are the most common.

Each of these loudspeaker configurations possess strengths and weaknesses for frequency coverage and projection distance. For example, straight arrays are “virtually unusable at high frequencies” because, as the listener gets farther away from the array, the high-frequency sounds begin to diminish significantly. On the other hand, the same set of speakers arrayed with a 5-degree splay between each speaker cabinet will maintain a near-uniform frequency coverage.[5] The advantage of a straight array is that this kind of speaker configuration can project the sound a greater distance. A compromise between the two configurations is the J-array, which is partly straight and partly curved: thereby, providing decent frequency coverage and decent distance projection. One drawback to the J-shaped array is that the straight speakers and the curved speakers require different levels of signal processing to maintain a consistent time alignment.

Time alignment refers to matching the speed at which sound reaches a listener when the sound is coming from a multi-driver system (aka, an array). Besides the time discontinuities introduced by J-shaped arrays, similar deformities are created by the interaction between low- and high-frequency speakers. Since woofers tend to have a slower sound response than tweeters, delay is often used to ensure simultaneity between lows and highs for the listener. Placing the tweeters farther away than the woofers is another strategy for loudspeaker time alignment.[6]

The process of configuring speaker arrays for frequency and time to function ideally for any given venue is referred to as tuning. If you hear audio engineers talk about tuning a system, this is what they are referring to. Some of the factors that affect the tuning and alignment process are destructive interference, constructive interference, phase curves, and transfer functions.


Understanding sound reinforcement is a prerequisite for being a music technologist, music producer, or audio engineers. Becoming adept in this field can provide you with résumé bullet points and freelance audio work. Also, knowing how to run live sound is a good skill to have for any situation requiring a microphone and a set of speakers. This includes events like weddings, bingo nights, bat mitzvahs, political rallies, house parties, and so on. Even if your career destinations lie elsewhere, being fluent with sound reinforcement will give you a leg up in many endeavors. Thanks for reading.

Appendix: Setting up a Simple PA System

First, place the gear in a useful configuration. The speakers should be spaced evenly and parallel (or slightly inward-facing) to the dance floor. If there is a sound person, then place the mixing board in front of the speakers and behind the dance floor. If the sound person is also the performer, then place the mixing board at the performance station. Usually, the speakers are placed upon stands, and the mixing board is placed upon a table. The main speakers should be about head level.[7]

After that, place microphones in all the useful positions. Useful positions include (but are not limited to) downstage for vocalists, and upstage for the guitar amps, auxiliary percussion, and drum set.

Then, run all the cabling for the mics, instruments, effects, and speakers. Be sure to use balanced cables for mics, unbalanced cables for instruments and effects, and speaker cables for speakers.

Next, turn on the power to the mixing board, then turn on the power to the amplifier (or powered speakers). If the mixing board is powered on after the amplifier and speakers, then a loud popping sound will issue forth from the system and damage the speakers. When setting up, turn on the board, then turn on the amplifier. When tearing down, turn off the amplifier, then turn off the mixing board.

Finally, do a line check and dial in the sounds.

Begin your line check by turning down all the volume levels. Get a clear, strong signal from each source, then adjust each for volume, pan, dynamics, effects, and equalization. After first getting individual signals, have the band play a song while you adjust the mix.

No matter what you do, don’t make it too loud.

If the stage is far away from the mixing station, then a special piece of equipment known as a snake is needed to transport the signal from the stage to the board. A snake consists of a plug box—one that accepts mic, instrument, and line inputs—and a bundle of long, unbalanced, and balanced cables. Despite its simplicity, a snake is an expensive piece of equipment.


[1] Davis, Gary, and Jones, Ralph. The Sound Reinforcement Handbook, (2000): 7.

[2] White, Glen D. and Louis, Gary J. The Audio Dictionary, 3rd edition, (2005): 424.

[3] A voice coil, by the way, is the coil of wire attached to the speaker cone that introduces the speaker-level voltage to a magnet; thereby, initiating the electromotive force that moves the whole apparatus.

[4] “Super Tweeter Speaker.” Aperion Audio., para. 1.

[5] Engebretson, Mark. “Advanced Loudspeaker Tuning Techniques.” (2007): 2 – 3.

[6] White, Glen D. and Louis, Gary J., (2000): 394.

[7] Elliot, Rod. Public Address Systems for Music, (2009): para. 8.


Alten, Stanley R. Audio in Media 9th ed. Wadsworth, Cengage Learning: U.S.A., 2011.

Behringer. Quick Start Guide: Xenyx 1202/1002/802/502. Music Group IP Ltd., 2001.

Davis, Gary, and Jones, Ralph. The Sound Reinforcement Handbook. Hal Leonard Corporation, 2000.

Eisele, Andrew. Live Sound 101: Sound System Design and Setup for a Live 15 June 2016.

Elliot, Rod. Public Address Systems for Music 2009. 6 June 2016.

Engebretson, Mark. “Advanced Loudspeaker Tuning Techniques.” QSC White Paper, 2007. “Super Tweeter Speaker.” Aperion Audio.

White, Glen D. and Louis, Gary J. The Audio Dictionary, 3rd edition. University of Washington Press, 2005.


  1. Which of the following best illustrates the concept of a transducer?
    • A device used for singing, rapping, and instrument playing
    • Professional audio equipment found on stage and in the studio
    • The conversion of mechanical sound energy into analog audio
    • Studio equipment invented to further the fortunes of the record industry
  2. Which signal processor is responsible for manipulating the tone and timbre of an audio signal
    • Equalizer
    • Pan knob
    • Delay
    • Reverberation
  3. Sound signals are processed with preamplifiers because
    • The voltage produced by transducers is too small
    • The voltage produced by preamplifiers introduces noise
    • The voltage produced by preamplifiers filters below 100 Hz
    • The voltage produced by transducers need reverberation
  4. Which one of the following is an example of a transducer?
    • Mixing console
    • Magnetic pickup
    • Alignment rig
    • FX send
  5. In the realm of speaker arrays, which of the following outcomes is most desirable?
    • Monotonic narrowing of coverage
    • Near field filtering and wide low frequency beamwidth
    • Far field coherence and even frequency coverage
    • Aberrant far field coherence and narrow high frequency beamwidth
  6. Which of the following speaker arrays feature a consistent splay angel between components?
    • Convoluted
    • Spital
    • J-arrays
    • Arcuate
  7. What is the job title given to a person who runs live sound?
    • Audio engineer
    • Mix coordinator
    • Remote broadcast mixer
    • Sound commander
  8. If you were setting up a PA system, your first step would be to
    • Turn the mixing board on
    • Increase all channel faders to maximum
    • Place the equipment in a useful configuration
    • Line check the microphones
  9. This type of pickup reacts to changes in pressure applied to a crystal.
    • Magnetic
    • Digital
    • Optical
    • Piezo
  10. When configuring multiway speaker systems, engineers must grapple with differences in low and high frequency response times at the level of the loudspeakers. What is the preferred method for dealing with this problem?
    • Delaying the signal to some speaker cabinets
    • Moving the subs
    • Configuring a convoluted speaker array


  1. The conversion of mechanical sound energy into analog audio
  2. Equalizer
  3. The voltage produced by transducers is too small
  4. Magnetic pickup
  5. Far field coherence and even frequency coverage
  6. Arcuate
  7. Audio engineer
  8. Place the equipment in a useful configuration
  9. Piezo
  10. Delaying the signal to some speaker cabinets

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