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    Computer Applications in Building and Environmental Acoustics

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    aking and teen NOL 2. No 3. 1-28, 197 (© 198 Pepemon Joa Computer Applications in Building and Environmental Acoustics D. J. OLDHAM* M.A. ROWELL * An account is given ofthe use of computers in building and environmental acoustics, Three areas fare examined. The fist relates 10 the use of computers in acoustic measurements with particular ‘emphasis on thelr use 1 alr impulses and 10 proces impulse response. The second i the use of computers o generate syshete sound fields for research purposes. The thirds the use of computers to eraluate the acoustic design of auditor and acoustics of urban areas. tis concluded that the Tavier area will be the foeut of new developments 1. INTRODUCTION FOR MANY years research in the field of building and ‘environmental acoustics has progressed with the basic objective being to quantify those physical parameters which determine our response to sound so that this infor- mation can then be applied to the design process. Although digital computers were available in all but the very earliest stages of this work, until very recently their application was limited. In the early work crude descrip- tors of sound fields were typically employed. However, recognition of the need for more and more precise descriptors of sound fields coincided with developments in computing such that there has been a greater and greater use of computers in recent years. In this paper we examine some of the ways in which computers are applied to the study of building and ‘environmental acoustics. Three areas can be identified firstly there is the application of computers to acoustic, experiments; secondly, computers are being employed to synthesize sound fields in order to better relate subjective response to objective parameters and finally computers are being employed increasingly in the design process. 2. APPLICATION OF COMPUTERS TO ACOUSTIC EXPERIMENTS: For many years computers have been used to process and tabulate experimental data in all scientific and engin- cering disciplines. However, in the field of acoustics, the computer is now regularly used to generate atest signal, to control experiments, and to perform any necessary processing on the resultant acoustic signal, as well as processing and tabulating the results The advent of the microprocessor has resulted in a whole new generation of acoustic instruments that are really dedicated microcomputers. Its now possible, for ‘example, to purchase a single instrument which is capable of automating all the routine measurements typical of “Department of Building Scicnee, University of Sheffield, Sheffield $10 27N, UK, 189 00m acoustics (reverberation time, background noise spectra, sound transmission, etc.) Similarly, in environ- ‘mental noise measurement itis possible to purchase equip- ‘ment which will perform a statistical analysis on traffic or aireraft noise measurements and express the results in terms of currently accepted units. OF more relevance to this paper, however, is the use of computers as research tools for the investigation of complex sound fields. 2. The fst Fourier transform ‘The most significant event in the last 25 years as far as acoustics signal processing is concemed is the devel- ‘opment of algorithms for calculating Fast Fourier Trans forms (FFTS) ‘A waveform can either be described in terms of its amplitude as a function of time oF in terms ofthe ampli- tudes of the frequencies present in the signal, The two are related by the following relationship x) -f xem dr o where X(o) is the Fourier transform of x(0) ois angular frequency ‘The time function of a given spectrum can be deter- mined by the inverse Fourier transformation x( if ioe, Afro a8 For all but fairly simple functions, the solution of Eqs (2) and () by analytical means is virtually impossible. An alternative approach, one utilizing the power of the computer, is to obtain the discrete Fourier transform (DFT). The calculation is performed on a sampled ‘waveform with a finite summation replacing integration, ‘The expression for the DFT is: 1 = Hm XO" @ 190 D. J. Oldham and M. A. Rowell where W © and (a) is @ sampled signal, with samples spaced T'seconds apart. The function X(m) represents a discrete spectrum with ‘m serving the same purpose as min the time domain. The frequency increment between successive components is Fe UM ‘An inverse discrete Fourier transform can also be defined as follows ¥ xomw-m 4 x ‘The calculation of the DET requires approximately N" computations. Since the resolution is inversely pro- portional to .V, to achieve high resolution requires con- siderable computing power. The Cooley-Tukey algorithm for performing a fast Fourier transform requires only WV log, N computations which results in significant saving in computer time for long records [I]. The existence of FFT algorithms, originally developed for mainframe computers but now implemented on micro-computing systems, has enabled experimenters to work in both time and frequency domains with relative ease, 2.2. Impulse response measurements “The power of the computer in acoustic measurement is demonstrated in the considerable amount of work that has been carried out using impulse response techniques. TThe impulse response describes the behaviour of a system when excited by a very short burst of sound. The true impulse response is obtained using a pulse which is @ Dirac delta function. The Dirac delta function has an infinitesimally short duration and cannot be realized in practice. An approximation to this ideal pulse can be achieved using an explosive device or a spark discharge. Loudspeakers are also sometimes employed as the source of sound in impulse testing and they may be used either to radiate an approximation to the Dirac delta function or to radiate a tone burst (a few cycles of an acoustic ‘waveform of a known frequency). It can be shown by performing a Fourier transform ‘that the spectrum of a short pulse of sound is broad band. ‘The broad band nature of impulse spectra means that information regarding the frequency response of a system can be obtained from a comparison of the Fourier trans- form of a signal used (0 excite that system and its response. This method has been applied {o measure the sound absorption coefficient and transmission loss of acoustic treatment in situ (2, 3}. It is a very powerful technique that is only realizable as a result of develop- ments in computer technology. 2.3. Correlation techniques ‘Another result of advances in computing technology is that correlation techniques have also become impor- tant tools for acoustic research. The techniques of cross- correlation, for example, can be used to identify difer- ent paths when investigating the transmission of sound ‘between two points in a room. The cross-correlation function between two stationary random functions of time x(0) and y() is defined as: F Fig. Example of room impulse response (after Harton 9) Rule) = Clot)» 6) where + represents a variable time delay Tf signals x(¢) and (0) are detected at different points ina sound field and the excitation signal is random noise, the cross-cortelation will be zero unless the value of tis equal to the difference in time between sound from the source reaching the two microphone positions. Thus fis systematicaly varied itis possible to identity different sound paths since they will result in different acoustic delays [4] Although analogue correlators have been constructed the true power of this technique can only be realized using digital techniques applied to sampled waveforms. Once an acoustic waveform has been digitized and cap- tured the necessary mathematical operations present no dificult 24. Examples of applications Tn room acoustics a considerable amount of work has been carried out on developing techniques to measure the impulse response of a room [5]. In a room, however, there are thousands of reflections such that itis impos- sible to measure the true impulse response (ie. perform an FFT on all reflections) and so a simplified technique has evolved, ‘A short duration acoustic signal is enerated (typically using an explosive device or spark source) and then the arrival of the sound at a microphone position is recorded ‘on an oscilloscope sereen. Sound travelling directly between source and receiver will arrive atthe microphone firs, to be followed by sound reftected from the internal surfaces of the room. A signal corresponding to the sound pressure or sound pressure squared is displayed on an ‘oscilloscope screen. This visual impulse response (some- times called an echogram) gives a clear indication of the arrival time and relative evel of early reflections and can also be used to identity echoes (see Fig. 1). Itis essential to employ shorter duration acoustic signals in order to avoid interference between reflections in close temporal proximity ‘A number of objective criteria have been established. based upon measured impulse responses. For example, Thiele (6] has suggested that acoustical definition (Deut- lichkeit) is determined by i “Late de — «100%. (6) D f {oor dr Computer Applications in Building and Environmental Acoustics 191 (a) (b) (ce) Fig. 2. Use of computer to tailor a pulse (after Davie eal {10}: (@) earget function, () input funetion and (@) resultant ‘acoustic function, The denominator i the squared impulse response inte- grated over $0 ms following the arrival of the direct sound and the numerator is squared impulse response integrated over “infinite” time. In other words, Deut- lichkeit isthe ratio of early energy received to total energy received Other criteria based upon the relationship between early and late energy which can be caleulated from the impulse response of a room include clarity, centre of aravity, time and spatial impression [7-9] Although the impulse response can be squared and integrated by analogue means the use of a digital com= puter results in greater flexibility when the impulse response is being examined. Ifa loudspeaker is used to radiate the test signal, it is, impossible to radiate an impulse that contains enough energy and is of short enough duration to approximate to the ideal Dirac delta function. The loudspeaker will ring because of is inertia and suspension system. Davies, Melntosh and Mulholland overcome the problem of loudspeaker ringing by using the technique of Fourier analysis to determine the response of a loud= speaker to a well defined electrical impulse {10]. The loudspeaker response was such that the radiated acoustic impulse was distorted. Davies er al. demonstrated that it ‘was possible using Fourier techniques to “distort” the electrical impulse to the loudpscaker in such @ way as to cancel out the distortion induced by the loudspeaker so that an almost perfect pulse was radiated (sce Fig. 2). The ideal impulse signal has a flat spectrum over a frequency range of 0 Hz (dc. to infinity. However, since in acoustics it is usually only necessary to consider the behaviour ofthe system under test over a finite frequency range it is not essential to employ a true Dirac delta function. Recognizing this fact Aoshima {11} used a com- puter to generate atest signal that was flat over a defined a nen enamia| 4 7 } "taf tS os y ! 06) Y Time mse) Fig. 3, Wave form of tailored tone burst (after Atal ea (12). frequency range and zero elsewhere. He first defined the desired spectrum of the signal and then, by applying the inverse Fourier Transform, he found the corresponding time function of a suitable signal. This signal was still of short duration and in order to avoid the problem of distortion due to the loudspeaker response and to improve the signal-to-noise ratio in the measurement, be used the computer to “stretch” the test signal. The response of the system to this “stretched” signal was then processed by the computer and “compressed” so that the response of the system to the original “unstretched” signal could be determined, ‘The bandwidth of an impulse decreases as its duration, increases. An alternative to the Dirac delta function is a tone burst, ie. number of cycles of a sinusoidal signal. ‘The bandwidth of a tone burst is determined by the frequency of the sinusoidal signal, the number of cycles radiated and the overall envelope. ‘Aneearly example of the use of a computer to produce a tone burst with well defined acoustic spectra was reported by Atal, Schroeder, Sessler and West [12]. The test signal they used, was a sinusoidal toneburst, with an amplitude envelope which was a displaced cosine func- tion on a de. pedestal (See Fig. 3). This signal was ‘employed because its bandwidth is determined by the duration of the envelope and its centre frequency is deter- mined by the frequency of the sinusoidal signal Using this signal Atal et al. measured various room acoustic parameters of the Avery Fisher Hall as a fune- tion of frequency. They achieved an excellent signal-to- noise ratio with this measurement system because all of| the energy from the amplifier and loudspeaker com- bination used to emit the pulse was limited to the fre- ‘quency range of interest, This was probably the fist case of the computer being used to generate a test signal, detect the response of an acoustic system and perform any nevessary processing on the resultant signal Schroeder has described the use of a continuous digi- tally generated test signal as an alternative to the use of a short duration acoustic signal to measure the impulse response of a room [4]. He used a computer to generate pseudo-random noise based on maximum length Sequences. Using this asa test signal, Schroeder obtained the impulse response of a room by using the computer to cross correlate the signal picked up by a microphone ‘with the original signal, Schroeder noted that, as the test signal is actually per= iodic, the response of the room to the test signal is also 192 D. J. Oldham and M. A, Rowell a = = Integrated tone burst decoy ‘Sconce Fig. 4. Comparison of reverberation dcay curves obtained 3) simple tone burst and (b) integrated impulse response (after Schroeder [13). The double slope nature ofthe decay can be leary seen in () pide ig pling sis ita hs fea onthe pe pone ot Sino so ned tas tc mtn tow ose Shas co Mou ep pe met mete 12 Tie wei! erat Soy of aed fm ay ning apa con wen fatanoetnre heunion tatty te Sonnet hte hte tt here tacts om uta eden ey we smdhcarroupatneea cba ae et i tron tania Sst we a ‘nse off nun fuel Steyr costed spline ors [oo 0 The tinal ington tien any asta Rowe! oes ao Sion ache anloge maneutrae cx ste dpa compu Se mp eps si pil gs ng ol ce, ed itmmoo, ohare unin agp Tepe tes lac ets ger hin Saft sctaton dy cre nee tea te anon aie ib dy ssaUuiermaisiaieate ft thence resent ero es ont on a he eas ca 3, SUBJECTIVE RESPONSE TO COMPLEX SOUND FIELDS. ‘The building designer or urban planner is interested in that part of the sound spectrum to which human beings respond, This is generally taken to cover a range of fre- {quencies extending from 20 Hz to 20 kHz and a range of sound pressure levels from the threshold of hearing at 0B to the threshold of pain at 140 dB. However, & more realistic dynamic range encompassing the typical quiet and loud environments encountered in daily life would bbe around 90 dB. The response of human being 10 & given sound field is determined not only by the level of | the sound and ils spectrum but also by temporal vari ations (this is particularly truc in the ease of speech and music). It is the inter-elationship between sound level spectral content and time history that makes it difficult bby means of simple (o characterize an acoustic fel ‘measurements, In the field of environmental acoustics a considerable amount of work was carried out in the 1960s and early 1970s to establish complex units to rate tralfie, aireraft and industrial noise. Although computers played a part in this work their role was typically limited to data analy. sis. For example, they were used to investigate statistical relationships between field measurements of noise fev and the results of sound surveys carried out to assess dissatisfaction with noise conditions Although current units used to measure environmental noise are erude, representing what was achievable with the technology available at the time the basic research was carried out, they are probably adequate given the large uncertainties invariably associated with any attempt to assess subjective response. [Lis for this reason that research activity in this area has decreased in recent years Inthe field of building acoustics (and in particular the acoustics of auditoria) however, the past few years hhave seen an enormous increase in the use of computers. Until recently the parameter assumed to be most important in determining the acoustic quality of a room was the reverberation time, The classical theory of Sabine was based upon the assumption of a perfectly diffuse sound field and predicted an exponential decay of the sound field with time when the source ceased to operate [15]. This exponential decay becomes a lineat decay when the sound field is measured logarithmically in decibels. ‘The reverberation time is defined as the time taken for the sound field to decay by 60 dB and, assuming linear decay of sound pressure levels, can easily be obtained from the slope of the decay curve. ‘One ofthe earliest recorded uses of @ computer in room acoustics research was by Schroeder and Logan in an investigation ofthe subjective perception of room reve beration, They recognized that a simple delay line with Feedback did not result in a convincing artificial rever: berator [16]. The regularity ofthe impulse response when ‘transformed into the iequency domain revealed that this simple system would aet as.a comb filter and hence colour the resultant sound. However. they demonstrated that by adding some undelayed signal to the output of the delay line the system became an all pass filer (see Fig. 5) Schroeder and Logan used an IBM 7090 mainframe com puter to generate the appropriate delays and to produce eolourless” artificial reverberation. This technique was later used by Atal, Sesser and Schroeder to demonstrate that the subjective assessment of reverberation time was determined by the initial part of the decay slope [17]. Values of reverberation time measured over the full 60 dB range for rooms with non-linear decays could lead to an inaccurate assessment of the reverberation characteristics of those rooms as can be scen from the resulls presented by Atal et al. (see Fig. 6). Reverberation time is now recognized as being only fone of a number of parameters affecting the subjective Computer Applications in Building and Environmental Acoustics 193 Oy “LLL, os ave tape out Gain ©) Fig. 5. Simulation of reverberation: (a) simple delay line: a ‘single impulse atthe input produces a train of impulses separated by the delay time and results ina comb fier effect and (b) use of direct signal fed to output to give fat frequency response (aller Schroeder fal (16). assessment of a sound field in a room. A considerable amount of attention has been given to the relationship between the direct sound from source to a receiver and sound arriving via reflecting surfaces. It is the intensity, and time delay of the reflected sound relative to the direct sound that is important. This information can be ‘obtained for an existing hall from an echogram showing the impulse response of that hall. ‘A number of investigators have attempted to obtain “objective criteria for auditoria from an examination of n : au m 2 oe te 20 te 2892 Cbjective reverberation time (eee) Fig, 6, Subjective reverberation times found in paired com- parson tests (after Atal eal (17). The objective reverberation time is based upon the early part of the decay. The objective feverberation time based on the entire decay proces i shown for one extreme case (2). impulse responses and subjective assessments for seats in various halls In order to assess the physical parameters that govern the subjective attributes of auditora it is necessary to try and make a direct comparison between the acoustics of different halls. The obvious way in which to do this is to ask several subjects to sit and listen to a concert in a chosen hall. After the performance they can then be asked to comment on the acoustical performance of that hall. However, any comments would have little value unless they could be compared to those made in respect of @ ‘number of halls, so that a “scale” of some description could be established. It is with this comparison that the problem lies, Any comparison of judgements on the “acoustics” of different halls would be unreliable as the judgements would be based on listening to pieces of ‘music, played at different times, by different orchestras under different conductors and with each visit toa difer- ent concert hall being separated by days, weeks, or even months. An alternative approach is to reproduce the characteristics of the sound ficld of a hall in the lab- ‘oratory so that observers can be presented with a suc- cession of samples of music played in different acoustic conditions and can thus make rclative judgements more casily. (One way of achieving an artificial sound fel, is to first ‘measure the impulse response of a hall and t0 use this to determine the relative level and delay time of the impor- lant early reflections with respect to the direct sound, A computer can then be programmed to generate the required delay times and to modify the spectral content of the delayed signals so that by the use of multiple loudspeaker and amplifier combinations and multiple ‘outputs from the computer, the correct sound field can be created around the observer in an anechoie chamber. ‘The computer can also be programmed to generate some general reverberation to complete the required artificial sound field The system described above requires a very complex computer program and a very powerful computer to ‘generate ll the required delays in real time. The difficulty arises from the vast amount of information contained within a sample of specch or music. In order for an audio signal to be in a form suitable for a computer to process it must frst be digitized using 194 D. J. Oldham and M. A. Rowell Reverteration free signet pt) Fig. 7. Room simulation system (after Ando eta. (19,20), aan analogue to digital converter. The dynamic range of ‘an analogue to digital converter is given by: DR in ® where mis the numberof bits employed in the conversion. Thus a 16-bit converter would have a dynamic range of 96 dB. For a signal to be specified digitally, the sampling theorem states that it must be sampled at twice the rate of the highest frequency present, In the case of a signal covering the full audio spectrum this would mean a sampling period of 25 ns. Thus a mere I s of digitally en- coded music would occupy forty thousand 16-bit mem- ‘ory locations in the computer. For the signal to be pro- cessed in real time all necessary mathematical operations ‘must be performed in the 25 ys sampling cycle. In the early 1970s, in order to achieve the necessary hhigh data throughput rate combined with the ability 10 process the signal, Blesser, Baeder and Zaorski had to esign a special purpose computer to simulate acoustic fields [18]. Towards theend of the decade, however, Ando and Imamura were able to use a general purpose com- puter for this [19, 20]- Figure 7 shows the system that they employed. Their artificial reverberation was based upon the Schroeder system discussed above ‘simpler system has been implemented using a com- puter by Schroeder and Atal 21]. They managed to suc- cessfully program the computer to generate an artificial sound field, using just two loudspeakers positioned in Fig. 8. Pre-fltering scheme for free fisld reproduction method (after Schroeder (21). front of the observer in an anechoic chamber, as shown in Fig.& To reproduce the require sound field each loud- speaker radiates sound previously recorded at the ears of 2 dummy head situated inthe test room. For accurate reproduction of the room acoustics itis essential that only the signal from the right loudspeaker reaches the Fight ear and that only the signal from the left loud- speaker reaches the let ear. To avoid the problem of “ross talk” from cach loudspeaker to its “far” ear Schroeder and Atal devised very clegant scheme where by means ofa system of filters the unwanted signal was prevented from reaching each ea. “The first stage inthe process involves obtaining the response at each ear for an impulse generated by cach loudspeaker. The Fourier transform of each impulse response can then be used to determine the required filter characteristics The experimental arrangement is shown in Fig. 8 where S = S(w) which is the Fourier transform of S(?), the impulse response at the near ear, and A = A(w) which is the Fourier transform of A(f), the impulse response of the far ear Cis C(o) and is equal to ‘A(o)/S(a). The circles labelled (1/1 C2) and 1/5 rep resent ites wth the appropriate characteristics. “The listeners able to perceive the acoustic conditions appropriate tothe original location of the dummy heed with proper free field acoustical coupling of the ear canals. ‘A notable application of this technique was. by Schroeder, Gottlob and Siebrasse [22] who replayed a passage of anechoic music (ie. music recorded in dead Acoustic conditions) from the platform of 25 major European concert halls and re-recorded it using two microphones in the ears of a dummy head, They then presented the resultant “performances” to a series of observers in an anechoic chamber via a par of loud speakers. Using this method they were able fo "measure" subjective parameters, such as “warmth” and “clarity” of the various halls. Statistical analyis ofthis information along with an examination of the impulse responses of tach ball (oblained at the time of recording) enabled Computer Applications in Building and Environmental Acoustics 195 relationships between the subjective attributes of a room and objective measurements based on the impulse response to be established. 4, APPLICATION OF COMPUTERS TO ACOUSTIC DESIGN EVALUATION At the beginning of this paper it was stated that the uikimate objective ofall the work on acoustic measure- ment and subjective response to complex sound fields was to provide the designer with information which would enable him to achieve good acoustic conditions. To this end relationships have been sought between subjective attributes and objective parameters of sound fields. IF these objective parameters can be predicted at the design stage then the probable subjective response can be assessed and the design modified to achieve optimum acoustic conditions. 4.1. Room acousties ‘A number of objective measures of acoustical per- formance have been proposed which can be calculated from the impulse response of a room. ‘The impulse response of an enclosure can be simulated by calculating the magnitude and delay time of reflected sound reaching a reception point relative to direct sound, ‘There are two approaches. The first involves the location of a series of images in the reflecting surfaces ofthe hall ‘This method is difficult to employ with halls of complex shape as it requires considerable computing power to calculate the latice of images and to verify that they are “visible” from the reception point. The second method involves following the path taken by rays of sound eman- ating from a point source, positioned in the room and noting the sequence in which the rays, and their “reflec- tions” arrive ata given observation point, One of the earliest ray-tracing studies was performed by Atal and Schroeder (23). When this work was per- formed the computing power available was limited, hence they restricted their study to two-dimensional rooms. ‘They examined the effect of different room shapes and absorber arrangements on the reverberation charac- teristics of their “rooms”. They programmed the com- puter to calculate the paths taken by 300 equal energy rays, spaced at intervals of 1.2° around the source, as they traversed each room. The “history” of each ray was recorded by the computer and an ‘impulse response” was constructed for the transmission path between the source and a chosen point in the “room”. From this ‘impulse response a decay curve was calculated using the integrated impulse method, ‘They reported significant discrepancies from the rever- beration times calculated using the accepted rever- beration time formulae due to Sabine [15] and Eyring [24] neither of which is sensitive to room shape oF position of absorbent treatment, They also observed that the decay rate was a function of the shape of the enclosure. ‘A ray-tracing program of more practical use, i. simu lates the transmission of sound in a three-dimensional space, was developed by Krokstadt and his co-workers over a period of years (25-27). This utilized a computer to perform the necessary three-dimensional vector cal- culations. (A number of other workers have also pro- Energy 0 * 500 meee Tine Fig. 9. Examples of computer calculated impulse response for two diffrent reception points in a hall (after Krokstadt er al en.

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