No file available
To read the file of this research,
you can request a copy directly from the authors.
Museum Acoustics with the Permanent Exhibition: Assessment Proposals for Acoustical Design
Preprints and early-stage research may not have been peer reviewed yet.
ResearchGate has not been able to resolve any citations for this publication.
Acoustics is a broad field of knowledge that extends branches all over the physics of wave phenomena, psychology, natural sciences, and engineering. It is taught, in general, as part of engineering, physics, or architecture programs, or even in graduate programs specialized in the theme. In Brazil, acoustics was taught in graduate programs, until the creation of Acoustical Engineering in 2009, at the Federal University of Santa Maria, an integral undergraduate program dedicated to acoustics, audio, and vibration (lasting ten semesters). This article presents its complete academic program, its creation process, and the professional establishment of the acoustical engineer. In the following, the program of study and subjects are elucidated and detailed, and the teaching methodologies used are also discussed. The program employs several active learning strategies, like project-based learning, aiming to transform abstract into concrete knowledge. The interaction of the university, the acoustical engineer, and society is also presented and clarified. The placement of graduates in fields and their workplaces are presented as outcomes. As a fundamental part of the engineer's formation, the infrastructure used, whether state-of-the-art or cost-effective equipment, is detailed in the context of teaching and research. Finally, some of the ongoing research projects of the students are described.
The aim of this paper is to conduct a qualitative research to clarify the auditory environment in the museums. Research has taken place in the Erimtan Archaeology and Arts Museum which exhibits more than 2000 objects from Yüksel Erimtan's collection of Anatolian archaeological artifacts. It is located near the main gate of Ankara Castle. While the scale and the façade of the building are connected contextually to its historic environment, the interior provides the contemporary experience to its visitors. Grounded Theory (GT) is used in this paper to create a conceptual framework, and clarify visitors' perception of the soundscape in the museums. Semi-structured interviews are held with participants who are chosen among the visitors after they tour the museum on their own, until the data reaches theoretical saturation. Constant comparison method is used for analyzing the data and creating a conceptual framework which is compared with previous studies and the ISO 12913-1: 2014. According to the findings, context is determined as one of the core categories similarly to other studies. And some categories differ because of the various function of indoor space.
This work deals with the improvement of the room acoustic quality of two medium sized meeting rooms through the investigation of the optimal placement of absorption and diffusive panels on the walls and ceiling. Acoustic measurements have been carried out in the existing untreated rooms with ODEON 13 room acoustics measurement and prediction software, and the Adobe Audition plugins Aurora. Simulations of different combinations of sound absorption and diffusion treatments have been carried out with the updated version of the software, ODEON 15. The panels were positioned in the meeting rooms following the guidelines of the DIN 18041 standard and the scientific literature. The results advise the application of absorptive materials on the ceiling or around the borders, creating a reflective middle area, and on the upper part of one the lateral walls, including the rear wall. Configurations with diffusers do not generally bring significant improvements. The Speech Transmission Index (STI) is a less sensitive parameter for the different acoustic scenarios, compared to Reverberation Time (T) and Clarity (C50). The research also outlined a design workflow, useful to successfully design meeting rooms and rooms for speech in general, which allows to determine the optimal number and location of acoustic panels and to minimize the costs.
Diffusive surfaces are considered as one of the most challenging aspects to deal with in the acoustic design of concert halls. However, the acoustic effects that these surface locations have on the objective acoustic parameters and on sound perception have not yet been fully understood. Therefore, the effects of these surfaces on the acoustic design parameters have been investigated in a real shoebox concert hall with variable acoustics (Espace de Projection, IRCAM, Paris, France). Acoustic measurements have been carried out in six hall configurations by varying the location of the diffusive surfaces over the front, mid, and rear part of the lateral walls, while the other surfaces have been maintained absorptive or reflective. Moreover, two reference conditions, that is, fully absorptive and reflective boundaries of the hall have been tested. Measurements have been carried out at different positions in the hall, using an artificial head and an array of omnidirectional microphones. Conventional ISO 3382 objective acoustic parameters have been evaluated in all conditions. The results showed that the values of these parameters do not vary significantly with the diffusive surface location. Moreover, a subjective investigation performed by using the ABX method with auralizations at two listening positions revealed that listeners are not sensitive to the diffusive surface location variations even when front-rear asymmetric conditions are compared. However, some of them reported perceived differences relying on reverberance, coloration, and spaciousness.
Room acoustic parameters of open-plan offices, e.g. ceiling, wall, screen, and furniture absorption, and screen height between workstations, affect strongly sound pressure level (SPL) of speech. The effect depends on the distance from the speaker. The aim of this experimental study was to examine the effect of ceiling, wall, and screen absorption, and screen height on the spatial decay of speech both at short and long distances from the speaker. Twenty-two experimental conditions were built in an open-plan office including 12 workstations. The conditions were combinations of ceiling absorption (2 levels), wall absorption (2 levels), screen absorption (2 levels), and screen height (4 levels). The spatial decay of A-weighted SPL of speech was measured according to ISO 3382-3 in workstations at distances from 2 to 9 m from the speaker. The A-weighted SPL of speech in the nearest workstation, at the distance of 2 m, varied between 53 and 61 dB, and at the distance of 4 m, between 46 and 60 dB. The spatial decay rate of A-weighted SPL of speech, D2,S, varied between 1.3 and 8.5 dB. The results confirmed that ceiling absorption is the most efficient way to increase spatial attenuation, but the attenuation effect also depends on the absorption of vertical surfaces like screens and walls, and the height of the screens. Our study is unique because it is the first experimental study conducted in a full-scale open-plan office, which investigates the impact of several sound absorbing surfaces and screen height simultaneously.
Previous research has shown that semantic-based tasks are negatively influenced by semantic aspects in background speech. Collaboration is an important task in open-plan study environments and is a semantic task which might be disrupted by background speech. Therefore, the aim of this study was to analyze the influence of irrelevant background speech on student-collaboration. Participants worked in pairs to solve spot-the-difference puzzles, by using the 'DiapixUK' collaboration task, while they were exposed to different background sound scenarios. The composed sound scenarios varied in semantic content (mother tongue and foreign language background speech)and reverberation time (short vs long), the latter affecting speech intelligibility. Although a longer reverberation time decreases the intelligibility of background speech and a foreign language decreases meaningfulness of speech, no significant changes in performance were found. On the other hand, the data show an increased perceived disturbance for a longer reverberation time, which we interpret as an increased difficulty of interpersonal communication in the collaboration task due to the increased level of the background speech. The quiet reference condition was the most preferred sound condition which is in line with both the effect of a low background sound level and the absence of semantic interference.
A round robin was conducted to evaluate the state of the art of room acoustic modeling software both in the physical and perceptual realms. The test was based on six acoustic scenes highlighting specific acoustic phenomena and for three complex, “real-world” spatial environments. The results demonstrate that most present simulation algorithms generate obvious model errors once the assumptions of geometrical acoustics are no longer met. As a consequence, they are neither able to provide a reliable pattern of early reflections nor do they provide a reliable prediction of room acoustic parameters outside a medium frequency range. In the perceptual domain, the algorithms under test could generate mostly plausible but not authentic auralizations, i.e., the difference between simulated and measured impulse responses of the same scene was always clearly audible. Most relevant for this perceptual difference are deviations in tone color and source position between measurement and simulation, which to a large extent can be traced back to the simplified use of random incidence absorption and scattering coefficients and shortcomings in the simulation of early reflections due to the missing or insufficient modeling of diffraction.
Before creating a new activity-based design in retrofit buildings projects the existing acoustic environment is
carefully measured. Not only the acoustical qualities of the existing building and facilities are measured
(according to NEN 5077 for sound insulation, ISO 3382 for room acoustics and ISO 3382-3:2012 for room acoustics in open plan offices), but also the actual behaviour of people is measured as well as this is a very important input for defining the acoustic environment. During a representative week of working
hours, the sound levels are monitored at different locations in the open plan office. To gather information about the character of the sound, sound fragments are also recorded based on a trigger level.
Defining the acoustic environment of (semi-) open plan offices based on building measurements and noise level measurements provides a good starting point for redesigning a diversity of office environments. Often the new design leads to activity-based office plans, where the different activities are carefully projected in a (semi-) open plan office. Practical measurement data of office noise levels are presented and analysed.
This paper proposes a multi-criteria analysis model to acoustically characterize a specific type of building: museums. Significant acoustic objective parameters are used and logically weighted (after analysis of questionnaires and interviews for subjective responses of visitors) to find a representative unique index of evaluation of a room, the IAQM (Index of Acoustic Quality in Museums) rated on a scale from 0 (worst) to 20 (best). The acoustic parameters used are: Reverberation Time (RT), weighted standardized sound level difference of facades (D2m,nT,w), Rapid Speech Transmission Index (RASTI), LAeq (from HVAC background noise) and L'nT,w (weighted standardized field impact sound pressure level). The multi-criteria mathematical model is presented and numerically tested with two museums.
In open-plan offices, employees often have to work despite being distracted by multiple sources of speech close to them. This disturbance is even more severe when the surrounding conversations are intelligible. For this reason the disruptive power of such babble-like noises can be quantified by measuring speech intelligibility. As such, the Speech Transmission Index is usually used to assess the quality of open-plan offices. In this article, the STI is used to evaluate the nuisance potential of sound environments in an open-plan office. 57 subjects were confronted with a serial memory task in four STI conditions (from 0.25 to 0.65). Noise annoyance was assessed objectively by measuring performance, and subjectively using the NASA-TLX questionnaire for measuring cognitive load. A very strong inter-individual variability appeared in performance measurements. Approximately half of the panel was made of high performing subjects, who proved to be insensitive to speech intelligibility. For the other half, performance was reduced when STI increased, as shown in previous studies, with a plateau for STI greater than 0.45. This strong inter-individual variability can explain the discrepancy between results observed in previously published studies. The NASA-TLX questionnaire proved to give useful complementary results. For example, high-performing subjects related a higher workload in the condition of maximum speech intelligibility.
This study used a multi-talker database containing intelligibility scores for 2000 sentences (20 talkers, 100 sentences), to identify talker-related correlates of speech intelligibility. We first investigated "global" talker characteristics (e.g., gender, F0 and speaking rate). Findings showed female talkers to be more intelligible as a group than male talkers. Additionally, we found a tendency for F0 range to correlate positively with higher speech intelligibility scores. However, F0 mean and speaking rate did not correlate with intelligibility. We then examined several fine-grained acoustic-phonetic talker-characteristics as correlates of overall intelligibility. We found that talkers with larger vowel spaces were generally more intelligible than talkers with reduced spaces. In investigating two cases of consistent listener errors (segment deletion and syllable affiliation), we found that these perceptual errors could be traced directly to detailed timing characteristics in the speech signal. Results suggest that a substantial portion of variability in normal speech intelligibility is traceable to specific acoustic-phonetic characteristics of the talker. Knowledge about these factors may be valuable for improving speech synthesis and recognition strategies, and for special populations (e.g., the hearing-impaired and second-language learners) who are particularly sensitive to intelligibility differences among talkers.
The measured impulse response is often used as a true reference of a real room impulse
response and geometrical acoustic simulations are considered to be only a crude representation
of it. However, both approaches have their own challenges and limitations. Geometrical acoustic
models do not include wave phenomena, such as interferences and diffraction, as they simplify
sound propagation by rays. The advantages of acoustic simulations with such models include a
perfectly omnidirectional and impulsive sound source, no distortion problems, full control of the
background noise, and a well-defined onset time of the impulse response. On the other hand,
impulse response measurements include wave phenomena, but they do have their own
weaknesses, which may cause significant errors in the derivation of the ISO-3382 room acoustic
parameters. Due to the presence of background noise in the measured impulse response it is
difficult to evaluate which part is valid. In addition, the sound source used for measurements
often has strong lobes at high frequencies and cannot produce an ideal Dirac function. For this
reason and due to distortion products by the octave-band filtering process detection of the
arrival time of the direct sound from a measured impulse response is of questionable accuracy.
In this paper simulated and measured parameters are compared in a number of well
documented cases and the various sources of errors are discussed. It is concluded that doing
room acoustic measurements correctly may be more difficult than it appears at first glance.
Over the last decades, Virtual Reality (VR) technology has emerged to be a powerful tool for a wide variety of applications covering conventional use, e.g., in science, design, medicine and engineering, as well as in more visionary applications such as the creation of virtual spaces that aim to act real. However, the high capabilities of today's VR-systems are mostly limited to first-class visual rendering. In order to boost the range of applications, state-of-the-art systems aim to reproduce virtual environments as realistically as possible for the purpose of maximizing the user's feeling of immersion, presence and acceptance. Such immersive systems deliver multiple sensory stimuli and provide an opportunity to act interactively, as reality is neither mono-modal nor static. Analogous to visualization, the auralization of virtual environments describes the simulation of sound propagation inside enclosures where methods of Geometrical Acoustics are mostly applied for a high-quality synthesis of aural stimuli that go along with a certain realistic behavior. Here, best results are achieved by combining deterministic methods for the computation of early specular sound reflections with stochastic approaches for the computation of the reverberant sound field. By adapting acceleration algorithms from Computer Graphics, current implementations can manage the computational load of moving sound sources around a moving receiver in real-time - even for complex but static architectural scenarios. In the course of this thesis, the design and implementation of the real-time room acoustics simulation software RAVEN will be described, which is a vital part of the implemented 3D sound-rendering system of RWTH Aachen University's immersive VR-system. RAVEN relies on present-day knowledge of room acoustical simulation techniques and enables a physically accurate auralization of sound propagation in complex environments including important wave effects such as sound scattering, airborne sound insulation between rooms and sound diffraction. Despite this realistic sound field rendering, not only spatially distributed and freely movable sound sources and receivers are supported at runtime but also modifications and manipulations of the environment itself. All major features are evaluated by investigating both the overall accuracy of the room acoustics simulation and the performance of implemented algorithms, and possibilities for further simulation optimizations are identified by assessing empirical studies of subjects operating in immersive environments.
A physical method has been developed for measuring the quality of speech transmission systems. The method quantifies the deterioration of the intensity fluctuations of running speech. The decrease of these fluctuations as a function of the fluctuation rate (Modulation Transfer Function,MTF) is expressed by a single index, the Speech Transmission Index(STI). A hardware device using an artificial test signal determines the STI on the basis of the MTF'smeasured for each of the octavebands from 125 Hz up to 8 kHz. The method has been applied successfully to distortions as bandpass limiting, interfering noise, peak clipping, digital modulation methods, automatic gain control and reverberation. For a set of 200 of such communication channels a high correlation was found between the physical index STI and the PB?word score as determined with talkers and listeners.
Perceptual distances among single tokens of American English vowels were established for nonreverberant and reverberant conditions. Fifteen vowels in the phonetic context (b-t), embedded in the sentence "Mark the (b-t) again" were recorded by a male talker. For the reverberant condition, the sentences were played through a room with a reverberation time of 1.2 s. The CVC syllables were removed from the sentences and presented in pairs to ten subjects with audiometrically normal hearing, who judged the similarity of the syllable pairs separately for the nonreverberant and reverberant conditions. The results were analyzed by multidimensional scaling procedures, which showed that the perceptual data were accounted for by a three-dimensional vowel space. Correlations were obtained between the coordinates of the vowels along each dimension and selected acoustic parameters. For both conditions, dimensions 1 and 2 were highly correlated with formant frequencies F2 and F1, respectively, and dimension 3 was correlated with the product of the duration of the vowels and the difference between F3 and F1 expressed on the Bark scale. These observations are discussed in terms of the influence of reverberation on speech perception.
Although a binaural advantage for understanding speech in adverse listening conditions has long been recognized, little is known about how binaural hearing influences consonant error patterns. The goal of the present study was to examine differences in consonant perception when stimuli distorted by noise and/or reverberation were presented using monaural, binaural, and diotic presentation modes. Results showed significant differences in error patterns among the three listening conditions (noise, reverberation, and reverberation + noise). Differences among the three presentation modes (monaural, binaural, and diotic) were subtle, as were interactions among presentation modes and listening conditions.
Sound field modelling of sequential spaces is required for large-scale public buildings. A FEM model comprising five successive rooms was used for a parametric study considering, i.e., contextual (opening dimension and position, number of rooms), acoustical (absorption coefficient and position) and source (directional radiation of opening and additional source) factors. The results reveal that (i) In rooms with high absorption, the average SPL of the rooms is higher, and the level difference between rooms is smaller with a larger opening dimension. Opening position only has impact for rooms with low absorption. (ii) The level differences between rooms are larger for oblique radiation compared to horizontal radiation, especially for rooms close to source. With an additional source (e.g., HVAC system and human speech) in a room, the level in this room is kept or reduced by approximately 10 dB with increasing opening dimension for rooms with high or low absorption, whereas the levels in other rooms increase for rooms with high absorption. (iii) The changes achieved by absorption are more significant with low absorption. The difference in the SPL between uniform and non-uniform absorption distribution with a given absorption amount is greater with a smaller opening dimension. (iv) The effects on the SPL in the first receiving room, caused by adding more sequential rooms, is less when the absorption in those rooms is high (e.g., with an absorption coefficient of 0.5), and the difference between different absorption coefficients is about 6 dB maximum, depending on the number of rooms.
The purpose of this research is to investigate the accuracy provided by different algorithms for room acoustics simulation. Three software packages are used to simulate the monaural room impulse responses of a medium-sized concert hall. All algorithms deal with geometrical acoustics but with distinct implementations. The simulated room is one of those adopted as a benchmark for the first round robin on room acoustical simulation and auralization, the last international inter-comparison, for which the geometrical and acoustic data were carefully measured. The main novelty is the inclusion of the software RAIOS, which participated in the round robin but did not finish the simulations in time to be included in the publication. Four acoustical parameters, namely the reverberation time, the early decay time, the clarity factor for music, and the definition for speech, are computed for ten source-receiver pairs at octave band frequencies from 125 Hz to 8 kHz. The results obtained by the three software are then compared with the measured data for all source-receiver positions and frequency bands. The relative mean errors to the just noticeable difference (JND) are also computed. The main findings are that there are deviations relative to the measured results for all software, with some of the parameters being better evaluated than others are. Preliminary insights into the advantages and limitations of the software packages with respect to the monaural impulse response are also given.
Enhanced speech intelligibility in realistic acoustic scenarios guarantees effective communication. Noise and reverberation degrade speech intelligibility, but the combined effect of reverberation, informational noise and position of target, listener and noise source on speech intelligibility still needs to insights. This work investigates the effect of real complex acoustic scenarios on speech intelligibility for adults. In two primary-school classrooms with reverberation time of 0.4 s and 3.1 s, receivers were located on axis with the target at increasing distances, and noise sources were both co-located and spatially separated from the target at different distances providing various degrees of energetic and informational masking. Noise level was maintained constant and set at 60 dB at the receiver's position, regardless of the noise source distance. The longer reverberation time resulted in worse speech recognition thresholds (SRT80s) by 6 dB on average. The competitive effect of informational noise resulted in increased SRT80s by 7 dB on average, in comparison to energetic noise. The SRT80 increases by ∼2 dB with doubling the target-to-receiver distance when the noise source is close to the receiver, accounting for both reverberations and noise types. The spatial release from masking resulted in improved speech intelligibility by up to ∼3 dB when the noise source is 1 m far from the receiver for energetic masker for low reverberation and, unexpectedly, for the informational masker in high reverberation. This may indicate that a perceptual segregation mechanism sorts out competing voices of informational masker according to their directions in least favorable listening situations.
To explore the acoustic environment of sequential spaces, the paper discusses the correlations between sound level and five independent dimensions of acoustic environment including loudness, intimacy, reverberation, clarity and spaciousness. Two sequential spaces with similar spatial composition were selected for questionnaire survey and field measurement. The results showed that under the condition of no sound source, sound level only affected loudness. Under the condition of sound source, a higher sound level resulted in a higher loudness, reverberation and spaciousness. Sound level differences between the rooms did not necessarily correspond to assessment difference in these three indicators. Intimacy was negatively correlated to clarity, and there was a threshold of sound level difference related to assessment difference of these two indicators.
This paper presents an approach to analyze visitors' expectations and perceptions of museums' built and auditory environments. It aims to explore visitors' perception of acoustic and built environments in museums, generate a systematic categorization, and create a conceptual framework using the Grounded Theory (GT) approach. We measured the Equivalent Continuous A-weighted Sound Level (L Aeq) and, following the ISO/TS 12913-2/3, conducted semi-structured interviews and questionnaire surveys were conducted to discover the sound environments and capture the subjective responses of visitors in two museums: the Rahmi M. Koç Museum (RMK) and the Erimtan Archaeology and Arts Museum (EAA) (both located in the most historical part of Ankara, Turkey). Although the selected museums offer two different experiences based on themes, exhibitions, and interior designs, we examined whether the museums' soundscapes, connected contextually to the historical environment, could be perceived different from one another. Results show that Museum RMK, which has historical exhibitions and an historical building type, is more appealing to people's preferences and expectations than Museum EAA, which has historical exhibitions and a modern building type. The findings of the study reveal that peoples' perceptions are mostly dependent on the context in which sound is heard, rather than on sound levels in museums. In some areas, where sound was used as a design element, visitors had a better museum experience because they were able to interact with the exhibited objects on exhibit and feel as if they were living in a specific period.
High-attendance exhibitions are often uncomfortable, due to too loud background noise. This is due to several factors, primarily the human voice. Moreover, most of the largest museums are housed in historical buildings which cannot be acoustically corrected, therefore the control of human noise is the only way to manage acoustic comfort. Human noise due to visitors is treated as a non-stationary phenomenon, related to flux of visitors/talkers. In the present study, a predictive model is proposed. Markov-chain theory is proposed to study the temporal behaviour of occupancy and generative algorithms are used to analyse the spatial distribution of visitors. The model has been validated by means of measurements in a highly attended museum in Florence during a free-entrance day. Measurement results agree with the predictive model, returning useful information on vocal effort and feedback processes. Results also show that a maximum number of visitors and a maximum visit time should be established in order to keep a proper acoustic comfort. Excess number of visitors or visiting time, may result in an almost two-fold increase in human noise level.
Irrelevant background speech causes dissatisfaction and impairs cognitive performance in open‐plan offices. The model of Hongisto (2005, Indoor Air, 15, 458‐468) predicts the relation between cognitive performance and the intelligibility of speech described with an objectively‐measured quantity, the Speech Transmission Index (STI). The model has impacted research in psychology and room acoustics as well as the acoustic design guidelines of offices. However, the model was based on scarce empirical data. The aim of this study was to revise the model based on a systematic literature review, focusing on laboratory experiments manipulating the STI of speech by wide‐band steady‐state noise. Fourteen studies reporting altogether 34 tests of the STI−performance relation were included. According to Model 1 that includes all tests, performance begins to decrease approximately above STI = 0.21 while the maximum decrease is reached at STI = 0.44. Verbal short‐term memory tasks were most strongly and very consistently affected by the STI of speech. The model for these tasks showed a deterioration in performance between STI 0.12 and 0.51. Some evidence of an STI–performance relation were found in verbal working memory tasks and limited evidence in complex verbal tasks. Further research is warranted, particularly concerning task‐specific effects.
This study presents a research that is concerned with the indoor soundscape in historical mosque. Hacı Bayram Mosque and its surroundings area of Hamamönü has been selected as the research site due to being the historical centre of Ankara. Although there are studies concerned with the acoustical characteristics of mosques, there is not enough research focusing on user’s expectation and interpretation of the indoor soundscape within a historical space. This study adopts the user-focused grounded theory to capture individuals’ auditory sensation and interpretation of the indoor soundscape within a historical mosque. In-depth interviews are held with congregation of the mosque and with the individuals sitting around the surrounding area. Based on their subjective responses, a theoretical framework is generated to gain an insight on the factors that affect individuals understanding and expectation from mosques. The conceptual framework generated through grounded theory shows how indoor soundscape may influence their individuals’ response to the physical environment of the mosque showing the association between the soundscape elements, spatial function and place identity.
The ability of the Speech Transmission Index (STI) to predict speech intelligibility under noisy conditions is highly dependent on the assumed spectrum of the speech signal. Examination of the literature showed that the long-term average speech spectrum of male talkers differs substantially from the speech spectrum recommended for STI calculations (IEC 60268-16). To explore these issues, the long-term average speech spectrum of forty male British English people was first measured, compared with the available literature and proposed for STI calculations. Then, using several voice alarm systems, the influence of the measured spectrum on STI calculations was assessed and comparisons made with the standard speech spectrum. The results showed significant STI differences under noisy conditions and considerable reductions in the required electrical power with the use of the new proposed male spectrum. This indicated that the current STI method could benefit from a revised speech spectrum.
Previous research suggests that, in open-plan offices, noise complaints may be related to the high intelligibility of speech. Distraction distance, which is based on the Speech Transmission Index, can be used to objectively describe the acoustic quality of open-plan offices. However, the relation between distraction distance and perceived noise disturbance has not been established in field studies. The aim of this study was to synthesize evidence from separate studies covering 21 workplaces (N=883 respondents) and a wide range of room acoustic conditions. The data included both questionnaire surveys and room acoustic measurements (ISO 3382-3). Distraction distance, the spatial decay rate of speech, speech level at 4 meters from the speaker and the average background noise level were examined as possible predictors of perceived noise disturbance. The data were analyzed with individual participant data meta-analysis. The results show that distracting background speech largely explains the overall perception of noise. An increase in distraction distance predicts an increase in disturbance by noise whereas the other quantities may not alone be associated with noise disturbance. The results support the role of room acoustic design, i.e., the simultaneous use of absorption, blocking and masking, in the attainment of good working conditions in open-plan offices.
Sound is becoming an organic element in museum exhibitions, which are increasingly encouraging participation, interaction and experience in communicating cultural information to the visitors. This paper expands on this developing trend in order to address a core question that needs to be considered in museum sound design and exhibition design: what are the possibilities that open up once we recognise the agency of sound and the ways it affects the visitors' experience? Common practices that use sound as an expressive means for conveying messages, for animating exhibits and for stimulating visitors' attention can be enhanced, critically examined and, when appropriate, transformed by attending more closely to the specific features of sound as a distinct modality of cultural communication. Particular emphasis is placed on how the spatiotemporal qualities of sound experience can contribute to the reconsideration of the distance between the visitor and the exhibit, and thus enrich exhibition design as a cultural practice in the production of knowledge.
Objective
The purpose of this study was to explore the differences in speech intelligibility in short-reverberant sound fields using deteriorated monosyllables. Generated using digital signal processing, deteriorated monosyllables can lack the redundancy of words, and thus may emphasize differences in sound fields in terms of speech clarity.
Methods
Ten participants without any hearing disorders identified 100 monosyllables convolved with eight impulse responses measured in different short-reverberant sound fields (speech transmission index >0.6 and reverberation time <1 s), and we compared speech recognition scores between normal and deteriorated monosyllables. Deterioration was produced using low-pass filtering (cut off frequency = 1600 Hz).
Results
Speech recognition scores associated with the deteriorated monosyllables were lower than those for the normal monosyllables. In addition, scores were more varied among the different sound fields, although this result was not significant according to an analysis of variance. In contrast, the variation among sound fields was significant for the normal monosyllables. When comparing the intelligibility scores to the acoustic parameters calculated from eight impulse responses, the speech recognition scores were the highest when the reverberant/direct sound energy ratio (R/D) was balanced.
Conclusions
Although our deterioration procedure obscured differences in intelligibility score among the different sound fields, we have established that the R/D is a useful parameter for evaluating speech intelligibility in short-reverberant sound fields.
1—In a previous paper*, dealing with the importance of properties of sufficiency in the statistical theory of small samples, attention was mainly confined to the theory of estimation. In the present paper the structure of small sample tests, whether these are related to problems of estimation and fiducial distributions, or are of the nature of tests of goodness of fit, is considered further.
The standard tables used for the Kolmogorov-Smirnov test are valid when testing whether a set of observations are from a completely-specified continuous distribution. If one or more parameters must be estimated from the sample then the tables are no longer valid.A table is given in this note for use with the Kolmogorov-Smirnov statistic for testing whether a set of observations is from a normal population when the mean and variance are not specified but must be estimated from the sample. The table is obtained from a Monte Carlo calculation.A brief Monte Carlo investigation is made of the power of the test.
Noise is the most detrimental factor of the indoor environment in open-plan offices. Speech is the most distracting source of noise. Distraction and speech privacy can be estimated objectively by determining speech intelligibility between workstations. Previously used measurement methods have focused on two neighbouring workstations. However, noise complaints are not restricted to the nearest workstation. The aim of this paper is to suggest a new method to measure and characterize the acoustical conditions of the whole office space, including both short and long distances from the speaker. The method should result in compact and expedient single-number quantities to improve the utilization of the method in building design. The measurement is carried out between workstations along a line consisting of at least 4 workstations. Measurements are made for background noise level, spatial reduction of Speech Transmission Index, STI, and spatial reduction of the SPL of normal effort speech. An omni-directional loudspeaker is used. The acoustic performance of offices could be logically described by three single number quantities: distraction distance, rD, spatial attenuation rate of A-weighted SPL of speech, DL2, and SPL of speech at a distance of 4 metres, Lp,S,4m. The method has been validated in 16 offices which varied significantly in room geometry, furniture and absorption. The differences between offices were unexpectedly large. Recommendations are presented for new target values which are already adopted in two Finnish guidelines.
For the 2nd International Round Robin on Room Acoustical Computer Simulation, the acoustical properties of a modern Swedish concert hall were calculated by 16 participants and compared with the data measured by three teams. Within six octave bands the values of T30, EDT, D50, C80, G, TS, LF, LFC and IACC had to be calculated using the geometrical data for the hall as given in maps, photos and descriptions of its surface properties. For a unique data input, a common geometry model and in the second phase the absorption and diffusivity data of the surfaces were given. The comparison of the results of the participants furnishes sources of calculation errors. First of all, the estimation of absorption data by the software user substantially affects the decay times calculated, therefore the acoustic experience of the user plays an important part. Even with uniform absorption and diffusivity data, the differences between the calculation results obtained by the participants can scarcely be attributed to individual software properties. The great error at low frequencies and the inability to handle edge diffraction effects today still limit the application to large rooms without obstructions. Since the accuracy of calculation is only one important point for the quality rating of room simulation software, the potential user of such software is provided with a list of criteria determining the qualification for a certain purpose.
The amount of time visitors spend and the number of stops they make in exhibitions are systematic measures that can be indicators of learning. Previous authors have made assumptions about the amount of attention visitors pay to exhibitions based on observations of behavior at single exhibits or other small data samples. This study offers a large database from a comparative investigation of the duration and allocation of visitors' time in 108 exhibitions, and it establishes numerical indexes that reflect patterns of visitor use of the exhibition. These indexes—sweep rate (SRI) and percentage of diligent visitors (%DV)—can be used to compare one exhibition to another, or to compare the same exhibition under two (or more) different circumstances. Patterns of visitor behavior found in many of the study sites included: (1) visitors typically spend less than 20 minutes in exhibitions, regardless of the topic or size; (2) the majority of visitors are not “diligent visitors”—those who stop at more than half of the available elements; (3) on average, visitors use exhibitions at a rate of 200 to 400 square feet per minute; and (4) visitors typically spend less time per unit area in larger exhibitions and diorama halls than in smaller or nondiorama exhibitions. The two indexes (SRI and %DV) may be useful measures for diagnosing and improving the effectiveness of exhibitions, and further study could help identify characteristics of “thoroughly-used” (i.e., successful) exhibitions.
The intelligibility of speech is known to be lower if the speaker is non-native instead of native for the given language. This study is aimed at quantifying the overall degradation due to limitations of non-native speakers of Dutch, specifically of Dutch-speaking Americans who have lived in the Netherlands 1–3 years. Experiments were focused on phoneme intelligibility and sentence intelligibility, using additive noise as a means of degrading the intelligibility of speech utterances for test purposes. The overall difference in sentence intelligibility between native Dutch speakers and American speakers of Dutch, using native Dutch listeners, was found to correspond to a difference in speech-to-noise ratio (SNR) of approximately 3 dB. The main segmental contribution to the degradation of speech intelligibility by introducing non-native speakers and/or listeners is the confusion of vowels, especially those that do not occur in American English. Vowels that are difficult for second-language speakers to produce are also difficult for second-language listeners to classify; such vowels attract false recognition, reducing the overall recognition rate for all vowels.
The standard tables used for the Kolmogorov-Smirnov test are valid when testing whether a set of observations are from a completely specified continuous distribution. If one or more parameters must be estimated from the sample then the tables are no longer valid. A table is given in this note for use with the Kolmogorov-Smirnov statistic for testing whether a set of observations is from an exponential population when the mean is not specified but must be estimated from the sample. The table is obtained from a Monte Carlo calculation.
The main intent of this paper is to introduce a new statistical procedure for testing a complete sample for normality. The test statistic is obtained by dividing the square of an appropriate linear combination of the sample order statistics by the usual symmetric estimate of variance. This ratio is both scale and origin invariant and hence the statistic is appropriate for a test of the composite hypothesis of normality. Testing for distributional assumptions in general and for normality in particular has been a major area of continuing statistical research-both theoretically and practically. A possible cause of such sustained interest is that many statistical procedures have been derived based on particular distributional assumptions-especially that of normality. Although in many cases the techniques are more robust than the assumptions underlying them, still a knowledge that the underlying assumption is incorrect may temper the use and application of the methods. Moreover, the study of a body of data with the stimulus of a distributional test may encourage consideration of, for example, normalizing transformations and the use of alternate methods such as distribution-free techniques, as well as detection of gross peculiarities such as outliers or errors. The test procedure developed in this paper is defined and some of its analytical properties described in ? 2. Operational information and tables useful in employing the test are detailed in ? 3 (which may be read independently of the rest of the paper). Some examples are given in ? 4. Section 5 consists of an extract from an empirical sampling study of the comparison of the effectiveness of various alternative tests. Discussion and concluding remarks are given in ?6. 2. THE W TEST FOR NORMALITY (COMPLETE SAMPLES) 2 1. Motivation and early work This study was initiated, in part, in an attempt to summarize formally certain indications of probability plots. In particular, could one condense departures from statistical linearity of probability plots into one or a few 'degrees of freedom' in the manner of the application of analysis of variance in regression analysis? In a probability plot, one can consider the regression of the ordered observations on the expected values of the order statistics from a standardized version of the hypothesized distribution-the plot tending to be linear if the hypothesis is true. Hence a possible method of testing the distributional assumptionis by means of an analysis of variance type procedure. Using generalized least squares (the ordered variates are correlated) linear and higher-order
The characteristics of speech, hearing, and noise are discussed in
relation to the recognition of speech sounds by the ear. It is shown
that the intelligibility of these sounds is related to a quantity
called articulation index which can be computed from the intensities
of speech and unwanted sounds received by the ear, both as a function
of frequency. Relationships developed for this purpose are presented.
Results calculated from these relations are compared with the results
of tests of the subjective effects on intelligibility of varying
the intensity of the received speech, altering its normal intensity-frequency
relations and adding noise.
Speech distortion is defined broadly as any operation that evokes inappropriate behavior by a listener in response to speech. Two categories of distorting operations are distinguished: (1) response-independent, in which the transfer function applied to the original speech signal is not determined by the probable response of the listener (e.g., masking, filtering); and (2) response-dependent, in which the distorting operation is related to the probable response of the listener during undistorted transmission and therefore may be specified in linguistic terms (e.g., foreign accent). Two experiments examine the effects and interactions of these two types of distortion. Twenty-four Midwest Americans listened to recorded articulation lists rendered by one American and three foreign-born speakers under eight conditions of masking and filtering. Reducing the speech to noise ratio to 20 dB or the transmission bandwidth to 500 cps yields approximately 50% reduction in word articulation for both native and foreign accent speech. The latter was approximately 40% less intelligible than native speech under all experimental conditions.
C-50 is an early-to-late arriving sound ratio used to assess the influence of room acoustics on the clarity and intelligibility of speech. A just noticeable difference in C-50 values was determined for speech sounds in simulated sound fields. Over a range of C-50 values from -3 to +9dB, representing most situations in rooms for speech, a just noticeable difference was estimated to be 1.1 dB. The corresponding just noticeable difference in Speech Transmission Index (STI) values was 0.03. This is similar to previous related estimates for speech and musical signals. To improve the acoustical characteristics of a room for speech, it is probably necessary to increase C-50 by approximately 3 dB to create a readily detectable improvement in everyday situations.
English consonant recognition in undegraded and degraded listening conditions was compared for listeners whose primary language was either Japanese or American English. There were ten subjects in each of the two groups, termed the non-native (Japanese) and the native (American) subjects, respectively. The Modified Rhyme Test was degraded either by a babble of voices (S/N = -3 dB) or by a room reverberation (reverberation time, T = 1.2 s). The Japanese subjects performed at a lower level than the American subjects in both noise and reverberation, although the performance difference in the undegraded, quiet condition was relatively small. There was no difference between the scores obtained in noise and in reverberation for either group. A limited-error analysis revealed some differences in type of errors for the groups of listeners. Implications of the results are discussed in terms of the effects of degraded listening conditions on non-native listeners' speech perception.
A physical method for measuring the quality of speech-transmission channels has been developed. Essentially, the method represents an extension of the Articulation Index (AI) concept, which was developed mainly to account for distortions in the frequency domain (noise, bandpass-limiting). The underlying concept of the present approach, based on the Modulation Transfer Function (MTF) of a transmission channel, has been adapted to account for nonlinear distortions (peak clipping) as well as for distortions in the time domain (reverberation, echoes, AGC). The resulting index, the Speech-Transmission Index (STI), has been correlated with subjective intelligibility scores obtained on 167 different transmission channels with a wide variety of disturbances. The relative predictive power of the STI, expressed in PB-word score, appeared to be 5%. This accuracy is comparable with results obtained from subjective measurements when about four talkers and four listeners are used. Expressed in terms of signal-to-noise ratio, the accuracy is about 1 dB. Pilot studies have been carried out to evaluate the use of the STI for testing digital-speech transmission channels.