Abstract

The present report has for aim to reflect the achievement of the creative project II by developing the fundamental concepts of the development as well as the key reflexions points made upon completion. The nature of the creative project II being elected by the students, this assignment enabled me to explore a field which had been drawing my attention for the last three years, that is, the science of architectural acoustics. In fact, I decided to devote my project to the science of architectural acoustics for several reasons. I was first introduced to the field of acoustics while studying the SAE Diploma of Technical Production in 2007. The science or architectural acoustics particularly attracted me due to its close connection to the characteristics of musical environments; whether related to music performance in concert halls and recording studios or to listening situations. The fact that the quality of a space could be evaluated in term of musicality by study of the acoustic parameters; and that such parameters could be predicted, measured, or even altered in order to suit a particular usage was particularly appealing. Among the diverse areas regrouped under the science of architectural acoustics, one in particular attracted my attention, that is, the reverberation in enclosed spaces. This phenomenon is commonly quantified under the form of reverberation time (in seconds) and describes the decay rate of sound. As such, this acoustic parameter appeared to me as one of the least subjective and most influent of the perceived quality of musical environments. Indeed, through researches, experiments, and sometimes even mistakes, the reverberation was recognised as being an important parameter in the liveness and enjoyment of musical performances. As such, the music performed in almost anechoic conditions was demonstrated as being thin and weak; although it appears clean-up, distinct, smooth and polished, the performance has no life (Everest 2001; Thompson 2002). On the other hand, excessively reverberant environments were deemed to be a constraint for musical definition and speech intelligibility (Long 2006). This direct impact of the phenomenon of reverberation on the musical quality of an environment thus quickly aroused my curiosity.

On the other hand, while studying distinct aspects of the reverberation time, more ambiguity appeared to me regarding the theoretical and practical approaches of the subject. In fact, several equations were available in order to predict the reverberation time is enclosed spaces; however it seemed to be a strong divergence in the definition of their application to common enclosures. In that way, the so-called classic theories of reverberation time appeared to present a certain limit of accuracy when applied to common environments but the limitation wasnt readily defined in the acoustic literature. On the contrary, the practical approach of reverberation time seemed as a straightforward process whereby the parameter was actually measured rather than predicted. Thus, it was possible to evaluate the degree of accuracy of the theory compared to practical measurements. This aspect of the subject particularly intrigued me, in part due to the scientific process involved, and the creative project II subsequently enabled me to put into practice this investigation. The creative project II was consequently devoted to the comparison of reverberation time prediction methods in enclosed spaced. The first two chapters of the report will address the key concepts involved in the establishment of the project while relating to certain aspects of the process undertaken. A final chapter will support the critical reflexions on the outcome of the creative project II.

Table of Content

Introduction

The growing interest regarding the acoustic parameters of enclosed spaces appears closely related to the propagation of musical culture through the nineteenth and beginning of the twentieth century. During the eighteenth and early nineteenth centuries, music in America was performed primarily by amateurs who made music for their own enjoyment (Thompson, 2002). By around 1850 this local festivity was regularly supplemented by the occasional performances of professional musicians primarily visitors from Europe and touring the larger cities of the United States (Beyer, 1999). The increasing development of musical culture over the 19th century had rendered the act of listening increasingly important. Musical performances quickly began to proliferate in American cities to the point where demand for concert space began to surpass the availability (Meloy, 1916). People began to believe that such establishments would procure several advantages for a city as much for the business as for the pleasure (Thompson, 2002). Thus, the projects of concert hall construction became as much a commercial venture as a cultural one. By 1900, the concert halls became a solemn place, and listening became serious business (Beyer, 1999). The acoustical environment consequently became of great interest to architects and physicists who consequently investigated the overall quality of an enclosure with regards to acoustic parameters. However, the field of acoustics wasnt established as properly speaking but was rather considered as something of a black art (Thompson, 2002); until the end of the ninetieth century where a physicist named Wallace Sabine looked deeper into the science of architectural acoustics.

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Chapter I
Theory of reverberation time
1/ Sabine reverberation time
Wallace Clement Sabine graduated from Ohio State University in 1886 where he studied physics and then moved to Boston to continue his study at Harvard University where he received his M.A. from the Department of Physics in 1888 (Beranek, 1985). Sabine subsequently collaborated with his senior colleague John Trowbridge on a series of studies exploring different aspects of electricity as well as working as assistant professor of Physics (Thompson, 2002). In 1895, Sabine was asked by Harvard University President Eliot to improve the faulty acoustics of a lecture hall in the new Fogg Art Museum. The sound in the room would persist for over five seconds, making a speaker's voice unintelligible to the listeners (Long, 2006). Eliot thus asked Sabine to find a way to reduce the reverberation in the room. He suggested that Sabine develop a quantitative measure of acoustical quality, in order to compare the faulty room with Harvard's Sanders Theatre considered as acoustically superior (Beranek, 1977). Sabine consequently undertook the process of measuring the reverberation time of the hall, that is, the duration to inaudibility of a sound as it propagates through a room after the emitted signal has been stopped (Curtis, 1996). At this time, it was assumed that a sound had dropped to one-millionth of its original intensity to become inaudible which represents a decrease of 10.log(106 ) = 60 decibels in sound intensity (Everest, 2001). As a consequence the reverberation time would later be defined as the time required for a sound to drop by 60dB from its initial intensity after emission of the signal. However, it was not obvious to Sabine what that measure should be, as the measurement of sound was a problem that had long challenged acoustical experimenters (Thompson, 2002).

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Throughout the past century, scientists had approached this problem primarily by attempting to render visible acoustical phenomena. Sabine initially adopted this strategy but found no useful way to interpret the results (Orcutt, 1933). He therefore abandoned all attempts to look at sound, and instead chose the apparent alternative of listening to it (Beyer, 1999). Sabine's technique consisted of using an organ pipe as sound source producing a pitch of 512 Hz, until a steady state of sound was achieved in the room. He then shut off the source and listened to the residual sound, or reverberation, until it was no longer audible. A torsion pendulum silently recorded the duration of audibility to hundredths of a second (Sabine, 1923). Sabine subsequently measured the reverberation times of the Fogg Lecture Hall and the Sanders Theatre, and he studied numerous other rooms throughout the Harvard campus, as well as in Cambridge and Boston (Beranek, 1985). Sabine measured the reverberation times of the rooms in their usual configuration, and he additionally manipulated those reverberation times by introducing different amount of sound absorption with the use of removable seat cushions from the Sanders Theatre (Long, 2006). After several years of experimentation and collection of various data, he was still unable to derive a fundamental mathematical relationship between the architectural properties of a room and its reverberation time. Meanwhile, the Fogg Lecture Hall remained unusable and unused (Thompson, 2002). In 1897, Sabine (1923) finally extracted results from his numerous experiments. He discovered that when he plotted the quantity of Sanders Theatre seat cushions (x) versus the corresponding reverberation time for a room (y), the resulting graph was a rectangular hyperbola, a standard mathematical curve characterized by the equation: x. y = k Where k is a constant. He discovered that the curve, in which the duration of the residual sound is plotted against the absorbing material, is a rectangular hyperbola with displaced origin and that the displacement of the origin is the absorbing power of the walls of the room; and that the parameter of the hyperbola is very nearly a linear function of the volume of the room (Sabine, 1923).

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Sabine realized that his discovery was an innovation for his understanding of reverberation (Thompson, 2002). Sabine's establishment resulted by 1900 in a comprehensive and quantitative analysis of reverberation. He initially represented his hyperbola with the equation: + x .t = k (1)
(Sabine, 1923)

Where

: Absorption of the room (walls, ceiling, etc.) x : Absorption of materials added to the room t : Reverberation time k : Hyperbolic constant

In this form, Sabine's equation differentiated the absorbing power of the room itself (a) from the absorbing power of the materials added to it (x). This distinction reflected his experimental practice, in which he first measured the reverberation time in a room, then introduced additional absorption to alter the reverberation time. Sabine initially expressed the total absorption of each room in terms of its equivalent in Sanders Theatre seat cushions, however this unit of absorption was clearly problematic as a more general scientific standard, and Sabine replaced it with a new "open-window unit" of absorption (Thompson, 2002). This unit was equivalent to the complete absorption of sound energy provided by an open window of one square meter and it consequently represented one square meter of a perfectly absorbent material. As a result, the standard unit of absorption is now called the sabin (or metric sabin) and has units of square feet (or square meter). Sabine then divided the total absorption of a room into its individual components. He expressed the absorbing power of each component with the quantity: An = n Sn (2)
(Sabine,1923)

Where

n : Coefficient of absorption of material n, (sabin or metric sabin) Sn : Total surface area of material n (m2 ).

As a result, the total room absorption could be represented by the equation: A = (1 S1 ) + (2 S2 ) + + (n Sn ) (3)
(Sabine, 1923)

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For any given room, Sabine could experimentally derive the value of this sum by measuring its equivalent in Sanders Theatre seat cushions and making some measurements of the surface area of each different material in the room. His task was then to determine the absorption coefficients of all those different materials. To accomplish this, Sabine's method required a full-sized room possessing a significant amount of the material to be tested (Thompson, 2002). The absorption coefficient of the material was calculated from the reverberation time of the room. Sabine set up systems of equations representing different rooms, each of which contained a different proportion of a range of materials; he was then able to solve the equations and determine the values of the different absorption coefficients (Sabine, 1923). Once determined, the coefficient for a given material was available for any future calculation, and Sabine published tables of these coefficients for others to use. The complete absorption of an open window was represented by a coefficient of 1.00, or 100 percent. By contrast, in a 1912 paper, he showed an absorption coefficient of 1.26 at 1024 Hz for a felt material and, in a 1915 paper, an absorption coefficient of 1.10 at 512 Hz for upholstered settees and 1.12 at 512 Hz for wood sheathing (Beranek, 2006). Sabine's next task was to determine the value of the hyperbolic constant, k, which would be valid for every room. By comparing hyperbolae for different rooms, he determined that the constant was directly proportional to the volume of the room (Sabine, 1923). He ultimately determined that the hyperbolic parameter k was proportional to the volume of a room according to the equation: k = 0.164 V
(Sabine, 1923)

0.164 was a coefficient introduced first empirically depending on the propagation conditions. Therefore in the literature, values equal to 0.16, 0.161, 0.162, 0.163, 0.164 can be found (Neubauer, 2000). The coefficient 0.161 being the most commonly referenced, it was subsequently used in the project. Sabine's equation could now be written in the form:

60 =
Where RT60 : Reverberation time (s)
V : Volume of the room (m3 )

0.164 V A

(4)

(Sabine, 1923)

A : Total absorption of the room (metric sabin).

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Sabine derived his equation based on the experiments he had undertaken using exclusively a frequency of 512Hz as sound source. Thus in 1904, Sabine began to expand on his earlier study of reverberation by examining the frequency dependence of the sound-absorbing powers of materials (Thompson, 2002). If sounds of different pitches were perceived as inaudible at different intensity levels, this difference would somehow have to be taken into account; only then would the equation be valid for all frequencies within the range of human hearing. This study followed the same method as his earlier work, supplementing the data collected at 512 Hz with data for six other frequencies ranging from 64 to 4096 Hz (Sabine, 1923). As a result of these experiments, Sabine discovered that the absorbing properties of materials varied considerably over this range, and since the variations were not simple functions of frequency, he plotted the result for each material as a curve. The theory was finally applicable to the entire frequency range and with his equation; Sabine had finally achieved the fundamental understanding of reverberation that he had long been researching (Beranek, 1977). This formula could now be used to predict the reverberation time of a room in advance of its construction, a benefit long sought by architects but never enjoyed before (Thompson, 2002). However, it is important to establish the circumstances in which the Sabine reverberation time equation has been developed. In fact, Wallace Sabine undertook his experiment in various rooms and lecture halls having the mutual characteristic of being reverberant due to their construction mainly of wood, plaster, and glass (Thompson, 2002). The absorption coefficients that he ultimately derived for these materials ranged from .025 for plaster to 0.61 for hard pine sheathing (Sabine, 1923). The sound in these rooms was therefore reflected off the various surfaces hundreds of times before it died away to inaudibility, resulting in reverberation times ranging from 1.91 seconds to 7.04 seconds (Beranek, 1985). As such, while deriving his equation, Sabine assumed that the sound energy in a room could be characterized as a homogeneous field, distributed uniformly through space, and gradually absorbed by the surfaces to which it was exposed.

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In conjunction to the reverberation time equation, Sabine developed an infinite series to represent the total sound energy in a room. In order to simplify the series, Sabine assumed that the room was reverberant enough so that the sound in it would suffer many reflections before any individual contribution of reflected energy would become negligible. Sabine ultimately used this quantity in his theory and in this way; the liveness of the room was implanted in his equation (Thompson, 2002). Since Sabines discovery, the technology had evolved dramatically, partly generated by the prosecution of the First World War which particularly impacted on the field of acoustics (Curtis, 1996). In the 1930s, Paul Sabine (1939) stated that commonplace equipment of every acoustical laboratory consisted of "linear response microphones, vacuum tube amplifier and oscillators, sensitive alternating current meters, and telephonic loudspeakers". As a result, the technology enabled researchers to extend Wallace Sabines work and propose revised theories which could be applicable to a wider range of enclosed spaces.

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2/ Eyring-Norris reverberation time

Carl Eyring was a physicist from Brigham Young University who left his academic position to work in the Sound Picture Labs of the Bell Labs in the early 1930s (Thompson, 2002). In the course of working in the extremely sound-absorbent environment of the Bell Labs soundstage, Eyring (1930) discovered that Sabines reverberation equation did not accurately describe the behaviour of sound in this room. The absorption coefficient of the thick material that lined the walls of the Bell Labs soundstage was 0.77, much greater than any coefficient with which Sabine had worked (Thompson, 2002). The resulting reverberation time of the soundstage was 0.35 seconds, far less than any time that Sabine had ever measured. In such case, sound energy was absorbed so quickly and completely that Sabine's assumptions about the gradual, diffuse absorption of sound no longer applied. Eyring's working environment constituted an extreme case that Sabine had neither encountered nor considered. Carl Eyring's acoustical environment differed considerably from that of Sabine, and it was this difference that drove him to reformulate the Sabine equation (Thompson, 2002). Paul Sabine however pointed out that Eyring's revision was stimulated by a discussion with R. F. Norris who suggested the modification to Sabine's equation that Eyring subsequently developed (Sabine, 1939). As a result, the revised formula is sometimes referred to as the "Eyring-Norris" equation. Eyring subsequently worked to modify the Sabine equation in order to fit the acoustically dead rooms (very absorbent) as well as the live ones that Sabine hadnt considered. Eyring's technique differed significantly from the method employed by Sabine. He replaced the mechanically sounded musical tone of the organ pipe with an electrically driven oscillator whose pure signal was amplified and then projected from a loudspeaker. The human detector was replaced with an "electro-acoustical ear," a microphone that automatically triggered a recording chronograph to register the instant at which the received sound signal had attenuated by 60 decibels. Eyring thus established the reverberation time as the time required for an electrical signal to suffer a standard degree of attenuation (Thompson, 2002).

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Simply put, Eyring (1930) presented an analysis based on the assumption that image sources may replace the walls of a room in calculating the rate of decay of sound intensity". He determined an abstract source located in free space, surrounded not by walls but by an infinite number of other sources located at increasing distances from the original, all simultaneously emitting sound back toward that original source. Where Sabine had supposed a smooth, gradual, and continuous decay of sound energy, Eyring described a discontinuous process whereby the flow of energy suffered abrupt drops. Eyring (1930) observed that this constant energy flow followed by an abrupt drop, rather than a continuous drop to this same level means a greater absorption during the same interval of time and hence a more rapid decay of sound. With this new understanding of the decay of sound energy, Eyring ultimately derived a new equation for calculating the reverberation time:

RT60 =

0.161 V S ln (1)

(5)

(Eyring, 1930)
Where V : Volume of the room (m3 )
S : Total surface are of the room (m2 ) : Average absorption coefficient

The average absorption coefficient is defined as:

S 1 .1 +S 2 .2 ++S n .n S

(6)
(Long, 2006)

Where S1 , S2 , Sn : Surface area of the materials (m2 )

1 , 2 , n : Absorption coefficient of the respective materials S : Total surface area of the room (m2 )

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Contrary to Sabine who assumed that as a sound wave travels around a room it encounters the surfaces one after another, Eyring assumed that all the surfaces are simultaneously impacted by the initial sound wave and that sound coming from a source in a room is successively reflected by boundaries having an average coefficient (Beranek, 2006). Each time a sound wave strikes one of the room boundaries, a fraction () of the energy is absorbed, and a fraction (1 ) is reflected. The number of reflections per second is equal to the distance sound will travel in one second divided by the average distance between reflections (the mean free path) (Neubauer, 2000). When Eyring defined the above process, he assumed that walls of the same area were hit by the same amount of sound energy per second (Neubauer & Kostek 2000). This assumption implicitly implies a relative diffuse sound field and ideal distribution of sound absorption which makes the Eyring-Norris equation dependent of acoustical and environmental conditions rarely observed in common enclosures. With the new equation, the reverberation time of a fully absorbent room predicted a logical reverberation time of zero seconds, a mathematical criterion that had not been met by Sabine's original equation. In 1900, Sabine and other acousticians had not been concerned with this limitation, as the existence of such an absorbent room was virtually inconceivable at that time (Beranek, 1985). Sabine's equation became consequently perceived as dated and inadequate, acousticians and sound engineers thus turned to Eyring's new equation in order to understand the behaviour of sound in the modern world (Thompson, 2002).

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3/ Eyring absorption coefficient

The Eyring-Norris equation is sometimes presented as using different absorption coefficients than the sabine coefficients in order to determinate the average absorption of a room (Beranek 2006; Neubauer 2000). As such, it is often expressed to employ the Eyring-Norris equation with caution as the eyring absorption coefficients must be less than 1 as implied in the formula itself. However, every absorption coefficients referenced in the literature and material specifications are measured using the Sabine formula in reverberation chamber and coefficient greater than 1 are sometimes obtained due to diffraction of sound from the edges of the sample material which makes the sample appear of greater area than it really is (Everest, 2001). This apparently important concern is rarely, not to say never, assessed in the acoustic literature when the Sabine and Eyring-Norris formulae are presented. In fact the formulae are developed using the same connotation of average absorption coefficient and the difference between the two is rarely expressed. Only Leo Beranek expressed this concern in several papers (Beranek 2000, 2007, 2008) by suggesting that a precise means for transfer from the sabine absorbtion coefficients to the eyring ones is possible because the same procedure for obtaining the average absorption coefficient in a room is followed in both equations. It is nevertheless necessary to develop both Sabine and Eyring-Norris equations in order to establish the relation between sabine and eyring coefficients. Because both Sabine and Eyring-Norris reverberation time formulae predict the same parameter, it is possible to establish the following relation:

RT60 =

0.161V S.sab

0.161 V S ln (1ey )

As expressed above, the sabine average absorption coefficient

sab = ln(1 ey )

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In it then required to solve the equation:

sab = ln(1 ey ) esab = 1 ey ey = 1 esab

(7)

The above expression enables to determine the eyring average absorption coefficient of the room from the already calculated sabine average absorption. As developed by Beranek (2000), It is then required to establish the ratio of ey /sab in order to determine the eyring absorption coefficient of a specific material from its known sabine coefficient:
ey sab

=
ey

S.ey S.sab

S. ey = ey =

sab ey sab

S. sab sab

Thus, the eyring absorption coefficient for a given material can be determined from its respective sabine absorption coefficient by using the expression:

n ey =

ey sab

n sab

(8)

(Beranek, 2000)
Where n ey : eyring absorption coefficient of material n
n sab : sabine absorption coefficient of material n ey : eyring average absorption coefficient sab : sabine average absorption coefficient

This expression originates from Beraneks investigation but has never been mentioned by other author to my knowledge.

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Regarding my project, I therefore wanted to make use of the above expression in order to determine the Eyring-Norris reverberation time using the eyring" coefficients, however after further evaluation, the reverberation times obtained in the three room configuration were exactly the same results than the ones predicted by the Sabine equation (using the sabine coefficients). Furthermore, after further examination it revealed that every reverberation time calculators available online or as software and which use the Eyring-Norris equation make also use of the sabine absorption coefficients in their calculation. After consideration I was still perplex about this aspect of the Eyring-Norris equation so I therefore preferred to use the sabine coefficients in order to predict the Eyring-Norris reverberation times peculiar to my project. By using this approach, it appeared that all the reverberation times obtained for the three room configurations using the EyringNorris equation presented 0.05 seconds shorter results across the frequency range than the Sabine formulae. This observation remains of unknown reason to my knowledge and would demand further personal investigation.

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4/ Fitzroy reverberation time

The concern about the limit of the classic reverberation time formulae when applied to ordinary spaces continued to increase and researchers persisted to develop formulae that better take into account the configuration of common rooms. In 1959 Daniel Fitzroy published a paper devoted to the problem of a more accurate calculation of the reverberation time in rooms with non uniform distribution of sound absorption (Neubauer & Kostek 2000). Fitzroy established a revised theory based on extensive tests in a large number of rooms where the distribution of sound absorption varied widely in uniformity (Kanga & Neubauer 2001). In his paper Fitzroy (1959) stated that the presented new formula afforded results which are closer to those measured in real halls. As such, Fitzroy considered not only the physical, but also geometrical aspects of the sound field in an enclosure. In this way, Fitzroy stated that the sound field tends to settle into three patterns of simultaneous oscillations occurring along the vertical, longitudinal and transverse axes of an enclosure. In a typical rectangular room, each one of the three axes contains a pair of opposite parallel boundaries. This would result in three distinct decay rate occurring along the three axes of a room and each rate being influenced by the specific average absorption of the pair of opposite boundaries normal to the corresponding axe. Since the sound field encounters the total area of a rooms boundaries, the sum of the decay rate specific to each pair of parallel boundaries would represent the total decay rate or reverberation time of the room. As a consequence, Fitzroy derived the first empirical reverberation time equation which takes into account the non uniform distribution of sound absorption (Neubauer & Kostek 2000). The Fitzroy reverberation time is defined as:

RT60 = 0.161

V S2

x ln (1x )

y ln (1y )

z ln (1z )

(9)

(Neubauer, 2000)

Where x, y, z : Total surface areas of a pair of opposite boundaries (m2 )

S : Total surface area of the room (m2 ) x , y , z : Average absorption coefficient of a pair of opposite boundaries

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Fitzroy made use of the Eyring-Norris equation in order to develop his formula, however the difference resides in the fact that Eyring assumed the absorptivity to be equal in all direction while with the Fitzroy formula, in the case of the rectangular room, three different calculations are made by means of the Eyring equation but the average absorption is altered for each pair of opposite boundaries (Neubauer 2000). Fitzroy then established ratios relating each pair of boundary areas to the total room area and thus the actual geometry of the room with regards to sound absorption is taken into account.

5/ Assumption of diffuse sound field

As expressed previously, while deriving their empirical theory, both Sabine and Eyring assumed a diffuse sound field with homogenous distribution of sound energy. As Neubauer (2000) pointed out, the sound field in an enclosure will be, in general, sufficiently diffuse if there are no large differences in the basic dimensions of the room, the walls are not parallel, the sound absorbing material is uniformly distributed, and most internal surfaces are divided into parts. However these conditions are rarely fulfilled in practice, the major source of deviation from these criterions encountered in most enclosures being the irregular distribution of sound absorption. In fact, the absorbing materials are most often placed on the floor and ceiling while the vertical walls remain reflective. As Joyce (1978) expressed, it is logical that sound intensity tends to be greater for paths in the enclosure that mainly involve surfaces with low absorption coefficients. In such rooms, the sound energy between horizontal surfaces would tend to dissipate slowly while the energy between vertical surfaces would be more rapidly attenuated due to the large absorption (Everest, 2001). As a consequence, these enclosures would suffer from a non homogenous repartition of sound energy where the assumption of diffuse sound field cannot apply. Another parameter that directly impacts on the diffuseness of the sound field is the modal resonances of a room (Knudsen 1932; Louden 1971; Schroeder 1964). Modal resonances are frequencies whose energy is increased or decreased depending on the room dimensions (Mayo, 1952). As developed by Everest (2001); in any enclosed space, a sound emitted from a source will propagate and be reflected from all the boundaries. The resulting sound pressure level at any point in the enclosure will consequently be the combination of the direct and reflected sound. The interaction of the two will inevitably produce standing waves at different frequencies relative to the dimensions of the enclosure. Clement Bresson - SAE Institute - BAP260.2 Report 15

The wavelength of the emitted sound and the distance between the boundaries will dictate the frequencies at which this resonant condition occurs. Thus the sound pressure level in the enclosure will be different at different positions and for different frequencies depending on the modal density. In large numbers, resonant modes reduce the field fluctuations because the individual modal resonances become more or less evenly distributed and thus the modal properties combine into a diffuse field (Nelson, 1992). In small number however, modal resonances can create timbral defects caused by acoustic isolations and degeneracy whereby the sound energy becomes dissipated or concentrated within specific frequency ranges (Self et al. 2009). The modal influence is known to dominate up to 300Hz in most enclosures (Everest, 2001), point from which the modal density increases to form a uniform distribution of modal frequencies. This last condition is required in order to establish a diffuse sound field with a uniform distribution of sound energy and random direction of propagation. As can be seen from the above description of modal resonance activity, in most enclosures, the sound energy is distributed very unevenly at low frequencies and the direction of propagation is far from random. This condition is especially observed in small room where the dimensions are comparable to the wavelength of the sound (Knudsen, 1932). As it has been expressed, in practice several limits set by environmental factors prevent the assumption of diffuse sound field to be verified, as a consequence, the accuracy of the reverberation time theories also appears to be dependant of these limits.

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Chapter II
Practical measurement of reverberation time
-OverviewMeasurements of scientific parameters have always been a statistical method to support the theories and develop the accuracy required for the determination of the parameters on the field. One of the first non-subjective measurements of reverberation time was done by E. Meyer in around 1930 who used a microphone and an electrode recording device. The results, however, were difficult to interpret because the measured response was recorded on a linear scale (Bjor, 1973). Around the same time various reverberation time measurement equipments emerged. The common procedure to most of the equipment used was to excite the enclosure with a sound source which was then stopped. A microphone plugged to a preamplifier picked up the resultant decay which was recorded on magnetic tape for later analysis. The recording was then played back through a sound level meter plugged to graphic level recorder which would trace the sound decay in function of the time (Everest, 2001). The reading from these devices was however strongly complicated by the presence of background noise, and fluctuations in the decay curve, among other factors whether set by environmental or material constraints (Bjor, 1973). As a result, in practice several requirements are imposed regarding the equipment used, the measurement process and the analysis procedure undertaken in order to obtain accurate data statistically representative of the overall behaviour of the sound field in the enclosure. One of the main requirements is that the sound source used to excite the enclosure should provide enough energy throughout the entire frequency spectrum so as to obtain sound decays sufficiently above the noise level in order to determinate the reverberation time which by definition requires a full 60 decibels decay. As such, bandpass filters are incorporated in both emitting and receiving equipment, partly to reduce the background noise level, and partly to form the necessary frequency range selectivity when broadband excitation signals are used.

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Thus reverberation time is measured in narrow frequency bands, usually octave or 1/3 octave bands so as to provide a dependable indication of the average acoustical effects taking place within a specific range of the audio spectrum (Everest, 2001). In practice, both steady state and impulse sound sources are used to excite the enclosure to be tested. The initial aim of the project was not only to evaluate the degree of discrepancy between theory and practice, but also to compare the reverberation time measured with different excitation signals. However, the use of steady state sources was revealed impracticable due to a lack of suitable equipment. In fact, the measuring software I have purchased for the project only enables the use of impulse excitation signals as sound source. As such, I concentrated the research on the evaluation of the discrepancy between measurements made by the impulse method. In that way, I wanted to assess the potential differences observed between the results obtained with different combination of sound source and microphone positions. I had at my disposal two measuring softwares presenting the same characteristics, and three room configurations presenting different acoustic conditions. I was therefore able to evaluate if the nature of the divergence observed between measurements was whether due to the acoustic characteristics of the environment or the variability of the measuring equipment. The fact of using the same excitation method carried out by two different measurement systems thus enabled to verify if the discrepancies observed between microphone and sound source positions were of the same nature for both softwares and in all three room configurations. One of the main tasks of the project was to establish a reverberation time of reference for each room configuration in order to assess the accuracy of the reverberation time theories. As such, it was particularly important to identify that the reverberation times measured were well representative of the actual acoustic conditions of the environments and not of the limits inherent in the measurement process. The next sections will consequently focus, without being exhaustive, on the procedure specific to the measurement of reverberation time with the impulse method as it was the only method employed during the project.

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1/ Impulse Response method

The use of an impulse sound source requires to record an enclosures response to a very short impulse whose energy is ideally distributed throughout the frequency spectrum (Ueda, Kon & Iazzetta 2005). A large number of acoustic parameters can then be extracted from the impulse response of an enclosure. The impulse sound sources fall into two categories: impulsive (or mechanical) and nonimpulsive excitation signals (Nelson, 1992). The most common sources of impulsive signals are starter pistol, electrical spark discharges, or pricked balloons producing a sharp transient and high level sound. However, these signals often dont provide the energy required to obtain suitable decays sufficiently above the background noise, especially at low frequencies (Everest, 2001). As a result, non-impulsive signals are most commonly used to obtain the impulse response of an enclosure. These sources feature a better directivity, frequency spectrum and reproducibility than impulsive sounds (Stan, Embrechts & Archambeau 2002); however, they require the use of a loudspeaker and an additional process is necessary in order to extract the impulse response from the recording. As such, a signal of readily known input is played in the enclosure through the loudspeaker and the microphone picks up the signal with the response of the room. The impulse response then requires to be recovered from the recording by applying a deconvolution process whereby the recorded signal is mathematically compared with the original generated one (Schultz & Beranek 1962). The nature of the deconvolution process is dependant of the input signal. The most commonly used non-impulsive excitation signals are deterministic, wide-band signals known as: Maximum-length sequence (MLS) and Inverse Repeated Sequence (IRS), which use pseudo-random binary sequence similar to wideband white noise. Time-stretched pulses, Time-Delay Spectrometry (TDS) and Log Sweep Frequency (LSF), which use time varying frequency signals.

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a) Pseudorandom noise excitation signals These excitation signals are based on a periodic sequence of randomly distributed positive and negative impulses with a flat energy distribution over the frequency range (Ueda, Kon & Iazzetta 2005). As expressed by Stan, Embrechts and Archambeau (2002), pseudorandom noise techniques possess the ability of randomizing the phase of any component in the recorded signal that is not related to the input signal emitted in the acoustical space. Thus any additional noise (white or even impulsive) will be distributed uniformly along the deconvolved impulse response which in term reduces the negative effect of background noise. However, a drawback resides in the appearance of distortion artifacts which are more or less uniformly distributed along the impulse response. The origin of the distortion peaks lies in the nonlinearities inherent in the emitting and receiving equipment (Stan, Embrechts & Archambeau 2002). These distortion artifacts introduce characteristic crackling noise when the measured impulse response is convolved with the anechoic signal in order to realize the auralization process (Guzina, 1982). Regarding my project, the measuring softwares which were to my disposal enabled the use of MLS and LSF impulse signal so I therefore wanted to compare the results obtained with both excitation signals. After several measurement sets, the impulse responses obtained with the MLS technique all presented distortion artifacts as mentioned above and the evaluation of the decay rates was consequently obscured. Moreover the reverberation times obtained after computational processing of the parameters were extremely divergent from the ones obtained with the sine sweep method and no particular pattern in the results could be observed. As a consequence, I decided not to include the analysis and evaluation of the MLS excitation signal in the project in order to avoid deducing erroneous statements misled by an observation that may be the result of interference in the receiving equipment or an operational mistake from my part.

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b) Sine Sweep excitation signals The sine sweep techniques use a sinusoidal wave that has its instantaneous frequency varying in time. In order to perform acoustical measurements over the entire audible range, the excitation signal must extend from 20 Hz to 20 000 Hz. Sweeps can be linear (TDS) or logarithmic (LSF). The linear sweep frequency signal is identical to an impulse in time, but provides equal energy across the frequency spectrum (DAntonio, 1986) while a logarithmic sweep exhibits a pink spectrum, that is, its amplitude decays at a rate of 3dB per octave which means that the signal has the same energy per octave and results in a better energy distribution than with a linear sweep (Ueda, Kon & Iazzetta 2005). The Sine sweep techniques and especially the LSF overcome most of the limitations observed with pseudorandom noise excitation. In fact, by using a logarithmic timegrowing frequency sweep it is possible to simultaneously deconvolve the linear impulse response of the system and to selectively separate each impulse response corresponding to the harmonic distortion orders considered (Stan, Embrechts & Archambeau 2002). The impulse response is finally extracted in the deconvolution process by applying a Fourier Transform, whereby a periodic signal is decomposed into its various harmonic components (Everest, 2001). Fouriers mathematical theorem states that any complex periodic wave can be synthesized from sinusoidal waves of different frequencies, different amplitudes, and different phase (Self et al. 2009). When the complete time response of a system is determined, the frequency response can be calculated directly using the Fourier transform. As Long (2006) highlights, (...) frequency response is by definition the Fourier transform of a system time response obtained by impulse excitation. Following this process, the impulse response can finally be analysed in order to extract the various acoustic parameters of the enclosure.

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2/ Measurement & Analysis Procedure

The impulse response obtained by mechanical or non-impulsive excitation is the function of sound pressure level vs. time of an enclosure (Stan et al. 2009). However the impulse response is not the decay trace as properly speaking. In this sense, the initial impulse will be followed by peaks and dips corresponding to the time-varying energy density, even so presenting an overall decreasing tendency. As such, the impulse response will present severe variations in amplitude due to modal resonance activity and reflections effects among other factors (Stan, Embrechts & Archambeau 2002). In practice, it is then necessary to process the impulse response in order to obtain the proper decay curve representative of the overall decay rate of the sound. The impulse response is subsequently processed using the Schroeder Integration method according to the ISO 3382 Standard (International Standard Organization 1997). It is in 1964 that Manfred R. Schroeder (1964) of Bell Telephone Laboratories presented the Tone burst integrated method where he explained that, ...in a single measurement, yields a decay curve that is identical to the average over infinitely many decay curves that would be obtained from exciting the enclosure with bandpass filtered noise. As developed by Schroeder in his paper, the impulse response recorded is played back in reversed. The output signal from the recorder is squared and integrated by means of an RC-integrating network. The voltage on the capacitor finally yields the decay curve, on a reversed time scale which may then be converted to a logarithmic amplitude scale in order to better evaluate the decay rate. The curve obtained is often referred to as the Energy Time Curve (ETC) or Schroeder curve and is a monotonically decreasing function of time, that is, it decreases without backtracking (Nelson, 1992), in contrast to the impulse response that fluctuate both up and down. In that way, this method better suits the definition of decay of sound, which requires that the sound energy in an enclosure should always decrease with time when no energy is being radiated into the enclosure (Schroeder, 1964). Once the Energy Time Curve is obtained, the reverberation time can be evaluated. In practice however, persisting fluctuations in the decay curve often obscure the determination of the overall decay rate. These random fluctuations depend on, among other factors, the nature of the modal resonances activity at the moment the sound source is switched off (Everest, 2001).

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The initial amplitudes and phase angles of the normal modes are different from trial to trial and thus for the same enclosure and identical sound source and microphone positions, different decay curves are obtained, the differences being the result of the randomness of the excitation signal, not any changes in the characteristics of the enclosure (Schroeder, 1964). This random situation thus requires to record several sound decays for each measurement set in order to achieve an acceptable repeatability of the results. The ISO 3382 standard (1997) prescribes a minimum of three decays. The overall reverberation time can then be determined from the average reverberation time of the individual decays or, more precisely, from the decay curve of the ensemble average of many individual curves (DAntonio, 1986). Such a procedure makes it possible to reveal if the energy dissipation of the enclosure is non-exponential, producing an average decay curve with some significant curvature (Nordtest, 1985). As expressed by Guzina (1984), curved or broken line decays indicate conditions that should be avoided. Such conditions are often the indication of a non diffuse sound field, or it may be the due to energy dissipation from an adjacent room with a longer reverberation time (Schultz & Beranek 1963). In practice, due to such fluctuations arising within a set of decays obtained under exact same condition, it is often required to deduce the average decay rate. This is achieved by fitting a least-square line to the Energy Time Curve; the reverberation time is finally calculated from the slope of the linear regression approximation of the decay curve (Nelson, 1992). However, this regression range is directly related to the decay range available, which in turn is limited by the background noise level (Brel & Kjr 2002). As such, the dynamic range of 60 decibels implied in the definition of RT60 is often impracticable due to ambient noise interfering with the tail of the decay. As a consequence, the reverberation time has to be measured over a smaller range than 60 decibels, and the result is then extrapolated to RT60 rather actually than measured. The range over which the decay will be measured and then extrapolated directly depends on the signal to noise ratio available (Schultz & Beranek 1962). As such, the ISO 3382 standard (1997) defines RT30 as the regression range from -5 to -35dB and RT20 the regression range from -5 to -25 dB.

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One last important aspect of the practical approach of reverberation time is the repeatability of the measurements with regards to positions. In fact, as expressed earlier, the sound pressure level will vary for different positions within the enclosure due to close proximity to absorptive/reflective surfaces and to the influence of modal resonances, among other factors. As a consequence, it is required to capture several decays for different combination of sound source and microphone positions, the ensemble averaging will then provide a more accurate representation of the overall decay rate of the sound in the room with respect to modal resonances activity (DAntonio, 1986).

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3/ Measurement requirements
a) Sound source
To measure the acoustic parameters of an enclosure in compliance with the ISO 3382 standard (1997), an omnidirectional sound source should be used which is usually a dodecahedral loudspeaker. The use of such equipment was however impracticable to my project thus it had to be substituted by a more available speaker presenting a directional pattern. On the other hand, as Figueiredo and Iazzetta pointed out (2005), musical instruments tend to project the sound in a directional way, although the directivity of the source is not properly the same one could observe in a real instrument or in a instrumental ensemble. As such, the use of directional sources could appear to be suitable in order to evaluate the reverberation time of musical performance or listening spaces. The literature presents several opinions regarding the number of sound source positions required. As such, the ISO 3382 Standard (1997) prescribes only two positions, while other authors express the need for three (Nordtest) or four positions (Guzina, 1984). All publications however agree that one of these positions should be in a corner of the enclosure. In fact, by aiming the loudspeaker into a corner of the room, all resonant modes are excited, because all modes terminate in the corners (Everest, 2001). After consideration, I decided to use three sound source positions with two positions being placed in adjacent corners, the speaker facing the room, while the third was placed centred to the north wall. However, after reflexion, it now appears that the speaker should have been placed facing the corners instead of being directed towards the centre of the enclosure. This fact didnt appear to me as I first considered that the sound source positions should be representative of the usual speaker arrangement commonly found in listening rooms (that is turned towards the listeners).

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b) Microphone
The receiving equipment should be composed of an omnidirectional microphone as small as possible preferably having the maximum dimension 13 mm (Nordtest, 1985). The smaller the microphone, the less its directional effects (Long, 2006), microphones up to 26 mm are allowed if they are of the pressure response type or of the free field response type but supplied with a random incidence corrector yielding a flat frequency response at random incidence (ISO 3382 Standard, 1997). As extensively expressed in the literature (Nelson 1992; Guzina 1982; DAntonio 1986; Everest 2001), the divergence in sound pressure level between different points in an enclosure requires to use several microphone positions for each sound source position in order obtain a better statistical representation of the sound field fluctuations with regards to positions. The ISO 3382 standards (1997) recommends between three and four positions. The positions should be located at least /2 apart, where is the wavelength of sound for the centre frequency of the frequency band of interest. Only one microphone should be used at a time. The microphones should be at least 0.5 meters away from any absorber or diffuser and at least 2 m from the sound source (Nordtest, 1985).

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Chapter III
Creative Project II Outcomes
One of the first reasons which incited me to devote my creative project II to the evaluation and comparison of the reverberation time prediction methods was in a way stimulated by the inconsistence in the theory. In this sense, both acoustics classes I attended during my two years of scholarship at SAE College presented the unique application of the theory of Wallace Sabine when dealing with reverberation. This was eligible as the curriculum covered several aspects of architectural acoustics besides the fact of being on a timeline. Due to my interest and intrigue in the subject, I therefore undertook further research in order to identify and study the more specific aspects of reverberation. I subsequently acquainted myself with the existence of the various reverberation time equations derived from Sabines model which were developed throughout the previous century by numerous researchers around the globe. Following these investigations, an aspect of the theory of reverberation time remained recurrent in every equation presented, that is, the assumption of diffuse sound field. In fact, a large part of the acoustics literature devoted to the subject of reverberation express the concern about the assumptions on which the classic theories are based. As such, the first empirically derived equation, namely, the Sabine reverberation time, is essentially acknowledged to depend on the assumption of diffuse sound field and relative low average absorption (Everest 2001). Numerous authors thus expressed the limit of accuracy of the Sabine equation when applied to common spaces that often dont suit the above conditions (Beranek 2006; Eyring 1930; Neubauer 2000). I was subsequently perplex regarding that statement which appears as a scientific reality according to acoustics theoreticians despite the fact that the Sabine equation remains the standard reverberation time equation referred in the acoustic literature and various other publications. I consequently became attracted by the subsequent theories developed in order to refine the limits of the Sabine equation.

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The Eyring-Norris formula is thus frequently referred to as the first revised theory which better took into account the large absorption of certain enclosure. However, additionally to offering a new approach on reverberation time prediction, the Eyring-Norris equation also raised further personal interrogations. In fact, as it has been expressed earlier in the report, the computation of the equation remains somewhat ambiguous to me due to the rather divergent unit of the average absorption coefficient found in the acoustic literature. In my opinion, the use of the sabine coefficients in the Eyring-Norris equation appears limited and mathematically illegitimate due to the logarithm dependence in the expression of the average absorption ln(1 ) which implies that average absorption coefficient smaller than 1 must be used, while sabine Coefficients sometimes present greater values. On the other hand, the fact that no eyring coefficients are listed in the literature and that only one author to my knowledge, Mr Beranek (2000), has made mention of a means for determining the eyring coefficients from the sabine ones seems to me of a great uncertainty as for the use of the Eyring-Norris equation. Unfortunately, this is the main interrogation that still isnt resolved after achievement of the project. Another discrepancy regarding the accuracy of the Eyring-Norris equation resides in the assumption of diffuse sound field with regards to uniform distribution of sound absorption (Neubauer, 2000). In that way, the Eyring-Norris theory is based on Sabines model, and although it doesnt presume the same sound absorption process, it appear to remain dependant of the same acoustic conditions rarely fulfilled in practice (Powel, 1970). As a result, the Fitzroy equation was the third and final formula under investigation. Daniel Fitzroy being the first theoretician who derived an equation that takes into consideration the non uniform distribution of sound absorption, it naturally appeared appealing to me. However, its close connection to the Eyring-Norris equation implies the same limit of absorption coefficients availability. In this sense, it is only rarely specified (Neubauer & Kostek 2000) that the Fitzroy equation uses the eyring coefficients in its calculation. Moreover, although the equation better represents the geometrical aspect of an enclosure with regards to absorption distribution, Fitzroys theory still assumes a diffuse sound field with homogenous repartition of sound energy and random direction of propagation; an assumption common to the Sabine and Eyring-Norris equations (Neubauer, 1999). I nevertheless chose these three equations, in spite of their assumption, because of their innovation in the field, the three equations being the first to assess different acoustical aspects of an environment. Clement Bresson - SAE Institute - BAP260.2 Report 28

The increasing intrigue in the prediction of reverberation time using the classic theories thus demanded further attention from my part in order to determinate their specific level of accuracy when applied to practical environments. The subjectivity of the theories consequently involved resorting to the practical measurement of reverberation time in order to assess this personal vagueness towards the theory of reverberation time. I consequently investigated the process of reverberation time measurement in enclosed spaces. After deeper research, the practical measurements also appeared to present several limits regarding the accuracy of the results. In fact, as it has been expressed through the development of the practical aspect of reverberation time, the precision of the measurements strongly depends on the repeatability of the process imposed by variations in the results obtained. The large influence of modal resonances in the low frequency range of most small rooms, such as the test room used for the project, appears to be one of the main cause of non diffuse sound field (Schultz & Beranek 1963); as such, the averaging of many measurements at different sound source and microphone positions appears to be a necessity in the determination of the statistical behaviour of sound in an enclosure (Prokofieva, 2009). However, such a procedure appears to better take into consideration the actual acoustical conditions of the enclosure and as such, it presented a rather strong basis for comparison to the theories. One of the main aims of the project was to undertake the experiment in different acoustic environments in order to assess the dependence of the theories accuracy to the assumptions on which they are based. However, only one room was used with different distributions and amounts of sound absorption for each configurations. This process, it has to be acknowledged, is not readily representative of the actual changes observed between enclosures of different dimensions and materials constructions. The acoustical changes observed between room configurations nevertheless suggested a noticeable effect on the overall behaviour of the sound field. However, the hypothesis and deductions put forward upon the achievement of the project should be considered as reflecting this particular experiment only, due to lack of repeatability in additional enclosed spaces.

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Through the practical measurements, the project nevertheless enabled me to better perceive the different assumptions on which the theories are based. In fact, the largest discrepancies observed between the theoretical and the measured reverberation times revealed to be in the first two frequency bands, that is, 125Hz and 250Hz. This observation coincides with the evaluation of the modal resonances of the room which suggested a modal influence up to the Schroeder frequency of 222Hz, point from which the modal density increases to form a more or less diffuse sound field (Everest, 2001). The diffuse sound field condition required by the theories to provide satisfactory accuracy was thus unfulfilled in the lower frequency range, this condition was also perceptible through the measurement process where the largest divergences in the decay curve obtained for different sound source and microphone positions were observed in the three first frequency bands. Moreover, the third room configuration qualified as non uniform also provided an interesting comparative analysis between the theories and the practice. In that sense, the Fitzroy equation predicted the closest results to the measurement which would suggest a relatively significant improvement in the accuracy of the theory when applied to enclosures with non-uniformly distributed sound absorption, in comparison to the Sabine and Eyring-Norris equations. Another interesting point regarding the assumption of the theories was assessed in the room configuration 2 which presented the smallest amount of absorption with an average coefficient of 0.15 across the frequency range compared to 0.25 and 0.32 for the other two configurations. As such, the assumption required by the Sabine theory regarding the low average absorption ( 0.2) was verified. The Sabine equation interestingly provided the closest results to the practical measurements. These analyses of the results in conjunction to the assumptions made by the theories thus enabled to verify the limits of accuracy peculiar to each reverberation time theory as stated in the acoustic literature. In that way, this enabled me to better discern the acoustical and environmental factors that prevent the assumption of diffused sound field to be fulfilled. However, the incertitude in the value/unit of the average absorption coefficient used in the Eyring-Norris and Fitzroy equations may impact to some extent on the validity of the observations made throughout the project regarding these two equations.

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As a conclusion, according to the experiment conducted as part of the creative project II, the evaluation of the reverberation time in enclosed spaces appears to remain in the domain of the approximation. In fact, the accuracy of the results at low frequencies appears strongly related to the assumption of diffuse sound field which is generally not fulfilled in common enclosures, especially in small rooms such as the one used for the project. Strong influence of modal resonances and non uniform distribution of sound absorption seem to be the main causes for preventing such acoustical condition to occur. Moreover, despite the fact that the theory tends to better agree with the practice in the last three frequency bands, significant fluctuations remain noticeable and as such, the results obtained with the theory dont appear to present enough consistency with the practical measurements to provide a satisfactory and accurate approximation of reverberation time. However, several other experiments such the one presented here and undertaken in diverse environments would be required in order to establish a statistical conclusion on the accuracy of the theory of reverberation time compared to the practical measurements.

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GLOSSARY
[Extract from The Master Handbook of Acoustics, 4th edn, McGraw-Hill, New York] absorption In acoustics, the changing of sound energy to heat. absorption coefficient The fraction of sound energy that is absorbed at any surface. It has a value between 0 and 1 and varies with the frequency and angle of incidence of the sound. acoustics The science of sound. It can also refer to the effect a given environment has on sound. AES Audio Engineering Society. algorithm Procedure for solving a mathematical problem. ambience The distinctive acoustical characteristics of a given space. amplitude The instantaneous magnitude of an oscillating quantity such as sound pressure. The peak amplitude is the maximum value. analog An electrical signal whose frequency and level vary continuously in direct relationship to the original electrical or acoustical signal. anechoic Without echo. audio frequency An acoustical or electrical signal of a frequency that falls within the audible range of the human ear, usually taken as 20 Hz to 20 kHz. audio spectrum See audio frequency bandpass filter A filter that attenuates signals both below and above the desired passband. bandwidth The frequency range passed by a given device or structure. bass The lower range of audible frequencies. binaural A situation involving listening with two ears. coloration The distortion of a signal detectable by the ear. comb filter A distortion produced by combining an electrical or acoustical signal with a delayed replica of itself. The result is constructive and destructive interference that results in peaks and nulls being introduced into the frequency response. When plotted to a linear frequency scale, the response resembles a comb, hence the name. critical band In human hearing, only those frequency components within a narrow band, called the critical band, with mask a given tone. Critical bandwidth varies with frequency but is usually between 1/6 and 1/3 octave. decay rate A measure of the decay of acoustical signals, expressed as a slope in dB/second. decibel The human ear responds logarithmically and it is convenient to deal in logarithmic units in audio systems. The bel is the logarithm of the ratio of two powers, and decibel is 1/10 bel. delay line A digital, analog, or mechanical device employed to delay one audio signal with respect to another. diffraction The distortion of a wavefront caused by the presence of an obstacle in the sound field. digital A numerical representation of an analog signal. Pertaining to the application of digital techniques to common tasks.

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distortion Any change in the waveform or harmonic content of an original signal as it passes through a device. The result of nonlinearity within the device. distortion, harmonic Changing the harmonic content of a signal by passing it through a nonlinear device. dynamic range All audio systems are limited by inherent noise at low levels and by overload distortion at high levels. The usable region between these two exptremes is the dynamic range of the system. Expressed in dB. echo A delayed return of sound that is perceived ty the ear as a discrete sound image. ETC Energy-time curve. FFT Fast Fourier transform. An iterative program that computes the Fourier transform in a shorter time. filter, bandpass A filter that passes all frequencies between a low-frequency cutoff point and a highfrequency cutoff point. Fourier analysis Application of the Fourier transform to a signal to determine its spectrum. frequency The measure of the rapidity of alterations of a periodic signal, expressed in cycles per second or Hz. frequency response The changes in the sensitivity of a circuit or device with frequency. graphic-level recorder A device for recording signal level in dB vs. time on a tape. The level in dB vs. angle can be recorded also for directivity patterns. harmonics Integral multiples of the fundamental frequency. The first harmonic is the fundamental, and the second is twice the frequency of the fundamental, etc. hertz The unit of frequency, abbreviated Hz. The same as cycles per second. image source A loudspeaker located at an image point. impulse A very short, transient, electric or acoustic signal. intensity Acoustic intensity is sound energy flux per unit area. The average rate of sound energy transmitted through a unit area normal to the direction of sound transmission. kHz 1,000 Hz. level A sound pressure level in dB means that it is calculated with respect to the standard reference level of 20 _Pa. The word level associates that figure with the appropriate standard reference level. linear A device or circuit with a linear characteristic means that a signal passing through it is not distorted. logarithm An exponent of 10 in the common logarithms to the base 10. For example, 10 to the exponent 2 is 100; the log of 100 is 2. loudness A subjective term for the sensation of the magnitude of sound. loudspeaker An electroacoustical transducer that changes electrical energy to acoustical energy. mean free path For sound waves in an enclosure, it is the average distance travelled between successive reflections. microphone An acoustical-electrical transducer by which sound waves in air may be converted to electrical signals. MLS Maximum length sequence. modal resonance See mode.

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mode A room resonance. Axial modes are associated with pairs of parallel walls. Tangential modes involve four room surfaces and oblique modes all six surfaces. Their effect is greatest at low frequencies and for small rooms. noise Interference of an electrical or acoustical nature. Random noise is a desirable signal used in acoustical measurements. Pink noise is random noise whose spectrum falls at 3 dB per octave; it is useful for sound analyzers with constant percentage bandwidths. nonlinear A device or circuit is nonlinear if a signal passing through it is distorted. normal mode A room resonance. See mode. octave The interval between two frequencies having a ratio of 2:1. pink noise A noise signal whose spectrum level decreases at a 3-dB-per-octave rate. This gives the noise equal energy per octave. preamplifier An amplifier designed to optimize the amplification of weak signals, such as from a microphone. random noise A noise signal, commonly used in measurements, which has constantly shifting amplitude, phase, and frequency and a uniform spectral distribution of energy. reflection For surfaces large compared to the wavelength of impinging sound, sound is reflected much as light is reflected, with the angle of incidence equaling the angle of reflection. refraction The bending of sound waves traveling through layered media with different sound velocities. resonance A resonant system vibrates at maximum amplitude when tuned to its natural frequency. response See frequency response. RT60 Reverberation time. reverberation The tailing off of sound in an enclosure because of multiple reflections from the boundaries. reverberation time The time required for the sound in an enclosure to decay 60 dB. room mode The normal modes of vibration of an enclosed space. See mode. sabin The unit of sound absorption. One square foot of open window has an absorption of 1 sabin. Sabine The originator of the Sabine reverberation equation. Schroeder plot A reverberation decay computed by the mathematical process defined by Manfred Schroeder. signal-to-noise ratio The difference between the nominal or maximum operating level and the noise floor in dB. sine wave A periodic wave related to simple harmonic motion. sound absorption coefficient The practical unit between 0 and 1 expressing the absorbing efficiency of a material. It is determined experimentally. sound pressure level A sound pressure expressed in dB above the standard sound pressure of 20 micropascals. spectrum The distribution of the energy of a signal with frequency. standing wave A resonance condition in an enclosed space in which sound waves traveling in one direction interact with those traveling in the opposite direction, resulting in a stable condition. steady-state A condition devoid of transient effects. TDS Time-delay spectrometry.

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timbre The quality of a sound related to its harmonic structure. time-delay spectrometry A sophisticated method for obtaining anechoic results in echoic spaces. tone A tone results in an auditory sensation of pitch. tone burst A short signal used in acoustical measurements to make possible differentiating desired signals from spurious reflections. transient A short-lived aspect of a signal, such as the attack and decay of musical tones. wave A regular variation of an electrical signal or acoustical pressure. wavelength The distance a sound wave travels in the time it takes to complete one cycle. white noise Random noise having uniform distribution of energy with frequency.

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References
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