# Reverberation

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Reverberation, in psychoacoustics and acoustics, is a persistence of sound after the sound is produced. [1] A reverberation, or reverb, is created when a sound or signal is reflected causing a large number of reflections to build up and then decay as the sound is absorbed by the surfaces of objects in the space – which could include furniture, people, and air. [2] This is most noticeable when the sound source stops but the reflections continue, decreasing in amplitude, until they reach zero amplitude.

Psychoacoustics is the scientific study of sound perception and audiology—how humans perceive various sounds. More specifically, it is the branch of science studying the psychological and physiological responses associated with sound. It can be further categorized as a branch of psychophysics. Psychoacoustics received its name from a field within psychology — i.e., recognition science — which deals with all kinds of human perceptions. It is an interdisciplinary field of many areas, including psychology, acoustics, electronic engineering, physics, biology, physiology, and computer science.

Acoustics is the branch of physics that deals with the study of all mechanical waves in gases, liquids, and solids including topics such as vibration, sound, ultrasound and infrasound. A scientist who works in the field of acoustics is an acoustician while someone working in the field of acoustics technology may be called an acoustical engineer. The application of acoustics is present in almost all aspects of modern society with the most obvious being the audio and noise control industries.

In physics, sound is a vibration that typically propagates as an audible wave of pressure, through a transmission medium such as a gas, liquid or solid.

## Contents

Reverberation is frequency dependent: the length of the decay, or reverberation time, receives special consideration in the architectural design of spaces which need to have specific reverberation times to achieve optimum performance for their intended activity. [3] In comparison to a distinct echo, that is detectable at a minimum of 50 to 100  ms after the previous sound, reverberation is the occurrence of reflections that arrive in a sequence of less than approximately 50 ms. As time passes, the amplitude of the reflections gradually reduces to non-noticeable levels. Reverberation is not limited to indoor spaces as it exists in forests and other outdoor environments where reflection exists.

In audio signal processing and acoustics, echo is a reflection of sound that arrives at the listener with a delay after the direct sound. The delay is directly proportional to the distance of the reflecting surface from the source and the listener. Typical examples are the echo produced by the bottom of a well, by a building, or by the walls of an enclosed room and an empty room. A true echo is a single reflection of the sound source.

A millisecond is a thousandth of a second.

Reverberation occurs naturally when a person sings, talks, or plays an instrument acoustically in a hall or performance space with sound-reflective surfaces. [4] The sound of reverberation is often electronically added to the vocals of singers and to musical instruments. This is done in both live sound systems and sound recordings by using effects units. Effects units that are specialized in the generation of the reverberation effect are commonly called reverbs.

Singing is the act of producing musical sounds with the voice and augments regular speech by the use of sustained tonality, rhythm, and a variety of vocal techniques. A person who sings is called a singer or vocalist. Singers perform music that can be sung with or without accompaniment by musical instruments. Singing is often done in an ensemble of musicians, such as a choir of singers or a band of instrumentalists. Singers may perform as soloists or accompanied by anything from a single instrument up to a symphony orchestra or big band. Different singing styles include art music such as opera and Chinese opera, Indian music and religious music styles such as gospel, traditional music styles, world music, jazz, blues, gazal and popular music styles such as pop, rock, electronic dance music and filmi.

A musical instrument is an instrument created or adapted to make musical sounds. In principle, any object that produces sound can be considered a musical instrument—it is through purpose that the object becomes a musical instrument. The history of musical instruments dates to the beginnings of human culture. Early musical instruments may have been used for ritual, such as a trumpet to signal success on the hunt, or a drum in a religious ceremony. Cultures eventually developed composition and performance of melodies for entertainment. Musical instruments evolved in step with changing applications.

A sound reinforcement system is the combination of microphones, signal processors, amplifiers, and loudspeakers in enclosures all controlled by a mixing console that makes live or pre-recorded sounds louder and may also distribute those sounds to a larger or more distant audience. In many situations, a sound reinforcement system is also used to enhance or alter the sound of the sources on the stage, typically by using electronic effects, such as reverb, as opposed to simply amplifying the sources unaltered.

Whereas reverberation normally adds to the naturalness of recorded sound by adding a sense of space, reverberation can reduce speech intelligibility, especially when noise is also present. Users of hearing aids frequently report difficult in understanding speech in reverberant, noisy situations. Reverberation is a very significant source of mistakes in automatic speech recognition. Dereverberation is the process of reducing the level of reverberation in a sound or signal.

Speech recognition is a interdisciplinary subfield of computational linguistics that develops methodologies and technologies that enables the recognition and translation of spoken language into text by computers. It is also known as automatic speech recognition (ASR), computer speech recognition or speech to text (STT). It incorporates knowledge and research in the linguistics, computer science, and electrical engineering fields.

Dereverberation is the process by which the effects of reverberation are removed from sound, after such reverberant sound has been picked up by microphones. Dereverberation is a subtopic of acoustic digital signal processing and is most commonly applied to speech but also has relevance in some aspects of music processing. Dereverberation of audio is a corresponding function to blind deconvolution of images, although the techniques used are usually very different. Reverberation itself is caused by sound reflections in a room and is quantified by the room reverberation time and the direct-to-reverberant ratio. The effect of dereverberation is to increase the direct-to-reverberant ratio so that the sound is perceived as closer and clearer.

## Reverberation time

Reverberation time is a measure of the time required for the sound to "fade away" in an enclosed area after the source of the sound has stopped.

When it comes to accurately measuring reverberation time with a meter, the term T60 [5] (an abbreviation for Reverberation Time 60dB) is used. T60 provides an objective reverberation time measurement. It is defined as the time it takes for the sound pressure level to reduce by 60  dB, measured after the generated test signal is abruptly ended.

The decibel is a unit of measurement used to express the ratio of one value of a power or field quantity to another on a logarithmic scale, the logarithmic quantity being called the power level or field level, respectively. It can be used to express a change in value or an absolute value. In the latter case, it expresses the ratio of a value to a fixed reference value; when used in this way, a suffix that indicates the reference value is often appended to the decibel symbol. For example, if the reference value is 1 volt, then the suffix is "V", and if the reference value is one milliwatt, then the suffix is "m".

Reverberation time is frequently stated as a single value if measured as a wideband signal (20 Hz to 20 kHz). However, being frequency dependent, it can be more precisely described in terms of frequency bands (one octave, 1/3 octave, 1/6 octave, etc.). Being frequency dependent, the reverberation time measured in narrow bands will differ depending on the frequency band being measured. For precision, it is important to know what ranges of frequencies are being described by a reverberation time measurement.

In the late 19th century, Wallace Clement Sabine started experiments at Harvard University to investigate the impact of absorption on the reverberation time. Using a portable wind chest and organ pipes as a sound source, a stopwatch and his ears, he measured the time from interruption of the source to inaudibility (a difference of roughly 60 dB). He found that the reverberation time is proportional to room dimensions and inversely proportional to the amount of absorption present.

Wallace Clement Sabine was an American physicist who founded the field of architectural acoustics. He graduated from Ohio State University in 1886 at the age of 18 before joining Harvard University for graduate study and remaining as a faculty member. Sabine was architectural acoustician of Boston's Symphony Hall, widely considered one of the two or three best concert halls in the world for its acoustics.

A stopwatch is a handheld timepiece designed to measure the amount of time that elapses between its activation and deactivation. A large digital version of a stopwatch designed for viewing at a distance, as in a sports stadium, is called a stopclock. In manual timing, the clock is started and stopped by a person pressing a button. In fully automatic time, both starting and stopping are triggered automatically, by sensors.

The optimum reverberation time for a space in which music is played depends on the type of music that is to be played in the space. Rooms used for speech typically need a shorter reverberation time so that speech can be understood more clearly. If the reflected sound from one syllable is still heard when the next syllable is spoken, it may be difficult to understand what was said. [6] "Cat", "Cab", and "Cap" may all sound very similar. If on the other hand the reverberation time is too short, tonal balance and loudness may suffer. Reverberation effects are often used in studios to add depth to sounds. Reverberation changes the perceived spectral structure of a sound but does not alter the pitch.

Basic factors that affect a room's reverberation time include the size and shape of the enclosure as well as the materials used in the construction of the room. Every object placed within the enclosure can also affect this reverberation time, including people and their belongings.

## Measurement of reverberation time

Historically, reverberation time could only be measured using a level recorder (a plotting device which graphs the noise level against time on a ribbon of moving paper). A loud noise is produced, and as the sound dies away the trace on the level recorder will show a distinct slope. Analysis of this slope reveals the measured reverberation time. Some modern digital sound level meters can carry out this analysis automatically. [7]

Several methods exist for measuring reverb time. An impulse can be measured by creating a sufficiently loud noise (which must have a defined cut-off point). Impulse noise sources such as a blank pistol shot or balloon burst may be used to measure the impulse response of a room.

Alternatively, a random noise signal such as pink noise or white noise may be generated through a loudspeaker, and then turned off. This is known as the interrupted method, and the measured result is known as the interrupted response.

A two-port measurement system can also be used to measure noise introduced into a space and compare it to what is subsequently measured in the space. Consider sound reproduced by a loudspeaker into a room. A recording of the sound in the room can be made and compared to what was sent to the loudspeaker. The two signals can be compared mathematically. This two port measurement system utilizes a Fourier transform to mathematically derive the impulse response of the room. From the impulse response, the reverberation time can be calculated. Using a two-port system allows reverberation time to be measured with signals other than loud impulses. Music or recordings of other sounds can be used. This allows measurements to be taken in a room after the audience is present.

Reverberation time is usually stated as a decay time and is measured in seconds. There may or may not be any statement of the frequency band used in the measurement. Decay time is the time it takes the signal to diminish 60 dB below the original sound. It is often difficult to inject enough sound into the room to measure a decay of 60 dB, particularly at lower frequencies. If the decay is linear, it is sufficient to measure a drop of 20 dB and multiply the time by 3, or a drop of 30 dB and multiply the time by 2. These are the so-called T20 and T30 measurement methods.

The RT60 reverberation time measurement is defined in the ISO 3382-1 standard for performance spaces, the ISO 3382-2 standard for ordinary rooms, and the ISO 3382-3 for open-plan offices, as well as the ASTM E2235 standard.

The concept of Reverberation Time implicitly supposes that the decay rate of the sound is exponential, so that the sound level diminishes regularly, at a rate of so many dB per second. It is not often the case in real rooms, depending on the disposition of reflective, dispersive and absorbing surfaces. Moreover, successive measurement of the sound level often yields very different results, as differences in phase in the exciting sound build up in notably different sound waves. In 1965, Manfred R. Schroeder published "A new method of Measuring Reverberation Time" in the Journal of the Acoustical Society of America. He proposed to measure, not the power of the sound, but the energy, by integrating it. This made it possible to show the variation in the rate of decay and to free acousticians from the necessity of averaging many measurements.

### Sabine equation

Sabine's reverberation equation was developed in the late 1890s in an empirical fashion. He established a relationship between the T60 of a room, its volume, and its total absorption (in sabins). This is given by the equation:

${\displaystyle T_{60}={\frac {24\ln 10^{1}}{c_{20}}}{\frac {V}{Sa}}\approx 0.1611\,\mathrm {s} \mathrm {m} ^{-1}{\frac {V}{Sa}}}$.

where ${\displaystyle c_{20}}$ is the speed of sound in the room (for 20 degrees Celsius), ${\displaystyle V}$ is the volume of the room in m³, ${\displaystyle S}$ total surface area of room in m², ${\displaystyle a}$ is the average absorption coefficient of room surfaces, and the product ${\displaystyle Sa}$ is the total absorption in sabins.

The total absorption in sabins (and hence reverberation time) generally changes depending on frequency (which is defined by the acoustic properties of the space). The equation does not take into account room shape or losses from the sound traveling through the air (important in larger spaces). Most rooms absorb less sound energy in the lower frequency ranges resulting in longer reverb times at lower frequencies.

Sabine concluded that the reverberation time depends upon the reflectivity of sound from various surfaces available inside the hall. If the reflection is coherent, the reverberation time of the hall will be longer; the sound will take more time to die out.

The reverberation time RT60 and the volume V of the room have great influence on the critical distance dc (conditional equation):

${\displaystyle d_{\mathrm {c} }\approx 0{.}057\cdot {\sqrt {\frac {V}{RT_{60}}}}}$

where critical distance ${\displaystyle d_{c}}$ is measured in meters, volume ${\displaystyle V}$ is measured in m³, and reverberation time ${\displaystyle RT_{60}}$ is measured in seconds.

### Absorption Coefficient

The absorption coefficient of a material is a number between 0 and 1 which indicates the proportion of sound which is absorbed by the surface compared to the proportion which is reflected back . the room. A large, fully open window would offer no reflection as any sound reaching it would pass straight out and no sound would be reflected. This would have an absorption coefficient of 1. Conversely, a thick, smooth painted concrete ceiling would be the acoustic equivalent of a mirror and have an absorption coefficient very close to 0.

### Reverberation in Music Composition and Performance

Several composers employ the reverberation effect as a main sound resource, having a comparable relavance as the solo instrument. For example, Pauline Oliveros [8] , Henrique Machado [9] and many others. In order to employ the reverberant properties of the room, the composers are supposed to investigate and probe the sound response of that particular ambient, that will affect and inspire the creation of the musical work.

## Creating reverberation effects

A performer or a producer of live or recorded music often induces reverberation in a work. Several systems have been developed to produce or to simulate reverberation.

### Chamber reverberators

The first reverb effects created for recordings used a real physical space as a natural echo chamber. A loudspeaker would play the sound, and then a microphone would pick it up again, including the effects of reverb. Although this is still a common technique, it requires a dedicated soundproofed room, and varying the reverb time is difficult.

### Plate reverberators

A plate reverb system uses an electromechanical transducer, similar to the driver in a loudspeaker, to create vibrations in a large plate of sheet metal. The plate’s motion is picked up by one or more contact microphones whose output is an audio signal which may be added to the original "dry" signal. In the late 1950s, Elektro-Mess-Technik (EMT) introduced the EMT 140; [10] a 600-pound (270 kg) model popular in recording studios, contributing to many hit records such as Beatles and Pink Floyd albums recorded at Abbey Road Studios in the 1960s, and others recorded by Bill Porter in Nashville's RCA Studio B.[ citation needed ] Early units had one pickup for mono output, and later models featured two pickups for stereo use. The reverb time can be adjusted by a damping pad, made from framed acoustic tiles. The closer the damping pad, the shorter the reverb time. However, the pad never touches the plate. Some units also featured a remote control.

### Spring reverberators

A spring reverb system uses a transducer at one end of a spring and a pickup at the other, similar to those used in plate reverbs, to create and capture vibrations within a metal spring. Laurens Hammond was granted a patent on a spring-based mechanical reverberation system in 1939. [11] The Hammond Organ included a built-in spring reverberator.

Spring reverberators were once widely used in semi-professional recording and are frequently incorporated into Guitar amplifiers due to their modest cost and small size. One advantage over more sophisticated alternatives is that they lend themselves to the creation of special effects; for example rocking them back and forth creates a thundering, crashing sound caused by the springs colliding with each other.

### Digital reverberators

Digital reverberators use various signal processing algorithms in order to create the reverb effect. Since reverberation is essentially caused by a very large number of echoes, simple reverberation algorithms use several feedback delay circuits to create a large, decaying series of echoes. More advanced digital reverb generators can simulate the time and frequency domain response of a specific room (using room dimensions, absorption, and other properties). In a music hall, the direct sound always arrives at the listener's ear first because it follows the shortest path. Shortly after the direct sound, the reverberant sound arrives. The time between the two is called the "pre-delay."

Reverberation, or informally, "reverb" or "verb", is one of the most universally used audio effects and is often found in guitar pedals, synthesizers, effects units, digital audio workstations (DAWs) and VST plug-ins.

#### Convolution reverb

Convolution reverb is a process used for digitally simulating reverberation. It uses the mathematical convolution operation, a pre-recorded audio sample of the impulse response of the space being modeled, and the sound to be echoed, to produce the effect. The impulse-response recording is first stored in a digital signal-processing system. This is then convolved with the incoming audio signal to be processed.

## Related Research Articles

Distortion is the alteration of the original shape of something. In communications and electronics it means the alteration of the waveform of an information-bearing signal, such as an audio signal representing sound or a video signal representing images, in an electronic device or communication channel.

Audio power is the electrical power transferred from an audio amplifier to a loudspeaker, measured in watts. The electrical power delivered to the loudspeaker, together with its efficiency, determines the sound power generated.

Room acoustics describes how sound behaves in an enclosed space.

An echo chamber is a hollow enclosure used to produce reverberation, usually for recording purposes. For example, the producers of a television or radio program might wish to produce the aural illusion that a conversation is taking place in a large room or a cave; these effects can be accomplished by playing the recording of the conversation inside an echo chamber, with an accompanying microphone to catch the reverberation. Nowadays effects units are more widely used to create such effects, but echo chambers are still used today, such as the famous echo chambers at Capitol Studios.

An anechoic chamber is a room designed to completely absorb reflections of either sound or electromagnetic waves. They are also often isolated from waves entering from their surroundings. This combination means that a person or detector exclusively hears direct sounds, in effect simulating being inside an infinitely large room.

Frequency response is the quantitative measure of the output spectrum of a system or device in response to a stimulus, and is used to characterize the dynamics of the system. It is a measure of magnitude and phase of the output as a function of frequency, in comparison to the input. In simplest terms, if a sine wave is injected into a system at a given frequency, a linear system will respond at that same frequency with a certain magnitude and a certain phase angle relative to the input. Also for a linear system, doubling the amplitude of the input will double the amplitude of the output. In addition, if the system is time-invariant, then the frequency response also will not vary with time. Thus for LTI systems, the frequency response can be seen as applying the system's transfer function to a purely imaginary number argument representing the frequency of the sinusoidal excitation.

Audio system measurements are made for several purposes. Designers take measurements so that they can specify the performance of a piece of equipment. Maintenance engineers make them to ensure equipment is still working to specification, or to ensure that the cumulative defects of an audio path are within limits considered acceptable. Some aspects of measurement and specification relate only to intended usage. Audio system measurements often accommodate psychoacoustic principles to measure the system in a way that relates to human hearing.

Thiele/Small parameters are a set of electromechanical parameters that define the specified low frequency performance of a loudspeaker driver. These parameters are published in specification sheets by driver manufacturers so that designers have a guide in selecting off-the-shelf drivers for loudspeaker designs. Using these parameters, a loudspeaker designer may simulate the position, velocity and acceleration of the diaphragm, the input impedance and the sound output of a system comprising a loudspeaker and enclosure. Many of the parameters are strictly defined only at the resonant frequency, but the approach is generally applicable in the frequency range where the diaphragm motion is largely pistonic, i.e. when the entire cone moves in and out as a unit without cone breakup.

Loudspeaker acoustics is a subfield of acoustical engineering concerned with the reproduction of sound and the parameters involved in doing so in actual equipment.

Digital room correction is a process in the field of acoustics where digital filters designed to ameliorate unfavorable effects of a room's acoustics are applied to the input of a sound reproduction system. Modern room correction systems produce substantial improvements in the time domain and frequency domain response of the sound reproduction system.

Speech Transmission Index (STI) is a measure of speech transmission quality. The absolute measurement of speech intelligibility is a complex science. The STI measures some physical characteristics of a transmission channel, and expresses the ability of the channel to carry across the characteristics of a speech signal. STI is a well-established objective measurement predictor of how the characteristics of the transmission channel affect speech intelligibility.

Critical distance is, in acoustics, the distance at which the sound pressure level of the direct sound D and the reverberant sound R are equal when dealing with a directional source. In other words, it is the point in space at which the combined amplitude of all the reflected echoes are the same as the amplitude of the sound coming directly from the source. This distance, called the critical distance , is dependent on the geometry and absorption of the space in which the sound waves propagate, as well as the dimensions and shape of the sound source.

In a reverberant space, the sound perceived by a listener is a combination of direct and reverberant sound. The ratio of direct sound is dependent on the distance between the source and the listener, and upon the reverberation time in [the room]. At a certain distance the two will be equal. This is called the "critical distance."

Loudspeaker measurement is the practice of determining the behavior of loudspeakers by measuring various aspects of performance. This measurement is especially important because loudspeakers, being transducers, have a higher level of distortion than other audio system components used in playback or sound reinforcement.

In audio signal processing, convolution reverb is a process used for digitally simulating the reverberation of a physical or virtual space through the use of software profiles; a piece of software that creates a simulation of an audio environment. It is based on the mathematical convolution operation, and uses a pre-recorded audio sample of the impulse response of the space being modeled. To apply the reverberation effect, the impulse-response recording is first stored in a digital signal-processing system. This is then convolved with the incoming audio signal to be processed. The process of convolution multiplies each sample of the audio to be processed (reverberated) with the samples in the impulse response file.

The sound reduction index is used to measure the level of sound insulation provided by a structure such as a wall, window, door, or ventilator. It is defined in the series of international standards ISO 16283 and the older ISO 140, or the regional or national variants on these standards. In the United States, the sound transmission class rating is generally used instead. The basic method for both the actual measurements and the mathematical calculations behind both standards is similar, however they diverge to a significant degree in the detail, and in the numerical results produced.

Underwater acoustics is the study of the propagation of sound in water and the interaction of the mechanical waves that constitute sound with the water, its contents and its boundaries. The water may be in the ocean, a lake, a river or a tank. Typical frequencies associated with underwater acoustics are between 10 Hz and 1 MHz. The propagation of sound in the ocean at frequencies lower than 10 Hz is usually not possible without penetrating deep into the seabed, whereas frequencies above 1 MHz are rarely used because they are absorbed very quickly. Underwater acoustics is sometimes known as hydroacoustics.

An Audio Analyzer is a test and measurement instrument used to objectively quantify the audio performance of electronic and electro-acoustical devices. Audio quality metrics cover a wide variety of parameters, including level, gain, noise, harmonic and intermodulation distortion, frequency response, relative phase of signals, interchannel crosstalk, and more. In addition, many manufacturers have requirements for behavior and connectivity of audio devices that require specific tests and confirmations.

Diffuse field acoustic testing is the testing of the mechanical resistance of a spacecraft to the acoustic pressures during launch.

3D sound reconstruction is the application of reconstruction techniques to 3D sound localization technology. These methods of reconstructing three-dimensional sound are used to recreate sounds to match natural environments and provide spatial cues of the sound source. They also see applications in creating 3D visualizations on a sound field to include physical aspects of sound waves including direction, pressure, and intensity. This technology is used in entertainment to reproduce a live performance through computer speakers. The technology is also used in military applications to determine location of sound sources. Reconstructing sound fields is also applicable to medical imaging to measure points in ultrasound.

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