Submitted Abstracts Sorted by Session

Acoustical oceanography began in the early 1900’s after the sinking of the steamship Titanic (Medwin and Clay). Since then, there has been a lot of research and development of acoustical methods and technologies that enhance our knowledge and understanding of the physical properties of the marine seafloor. Without the application of acoustics, our knowledge of benthic systems and the resident species is limited to anecdotal reference from industry fishers, although often lacking the spatial resolution necessary for stock assessment and management. Single beam acoustics, the simplest means of generating seafloor backscatter, has been used by the geoduck stock assessment program at Fisheries and Oceans, Canada, to quantify suitable seabed sediments for fishery quota calculation. The variance in the amplitude and duration of the seabed echo backscatter is relative to the variance in the consolidation and complexity of the seabed surface sediments. This acoustic backscatter data is segmented into discrete classes representing known seabed sediment types, ranging from fine unconsolidated sands to bedrock, and consequently, patterns of sediment deposition and seafloor bathymetry that reflect the complex and dynamic physical structure of marine ecosystems can be discerned. Single beam hydro-acoustic data is logged using echo digitization hardware (Quester Tangent Corp.). Spatial analysis software (Idrisi, Clark Labs) is used for geo-statistical interpolation and clustering of hydro-acoustic data to both quantitatively and visually describe the sub-tidal physical systems that are necessary for the structure and interdependence of infaunal, benthic, and demersal species.

Acoustic seabed classification with single-beam sounders involves discriminating among bottom types using attributes of their echoes. One attribute that is rich in sediment information is the duration of these echoes, but it needs to be interpreted with care because it depends on water depth, seabed slope and sounder beamwidth, as well as on seabed characteristics which we aim to measure, particularly roughness and attenuation. A ray-trace model of echo duration utilizes these factors and shows a linear relationship between water depth and echo duration. The linear increase in echo duration with water depth must be compensated for statistical seabed classification methods in order to make maps of seabed classes, rather than maps of classes influenced by bathymetry. An extensive series of experiments has been performed over a well documented site in Patricia Bay, BC. In one experiment a sounder was lowered toward the seabed over several homogeneous regions of sediment and echo durations measured as a function of altitude, simulating varying the water depth without varying anything else. These data showed the echo duration model is an effective basis for depth compensation. Seabed slope strongly affects echo durations, shapes and amplitudes. Echoes were collected with various tilts on the sounder, which is equivalent to seabed slope. Applying the simulated slope in the echo duration model was found to compensate echo duration effectively. Echo attributes measured from slope-and depth compensated echo time series discriminate seafloor sediments well up to a degree of seabed slope that is related to beamwidth.

Between 2000 and 2005, the Irish National Seabed Survey mapped the area from the outer margins of Ireland's territorial seabed to the shelf. Since 2006, INFOMAR (INtegrated mapping FOr the sustainable development of Ireland's MArine Resource) has succeeded INSS, with a focus on nearshore surveys. Using several research vessels equipped with a variety of instruments, ocean, seabed and sub-seabed properties have been measured. During this time, INFOMAR has been systematically producing regional seabed classification charts using multibeam backscatter data on the shelf and coastal areas. Multibeam backscatter returns at high frequencies are generally a good indicator of surface seabed properties, however, they are not suitable for the subsurface discrimination because of the minimal penetration. Furthermore, multibeam backscatter returns below nadir are not generally a good seabed discriminator, leaving unclassified gaps in most of these charts. Single beam echoes are precisely imaging most of this region, making the two sonar systems complementary in an integrated approach to seabed characterization. INFOMAR and the Canadian Marine Acoustic Remote Sensing (C-MARS) Facility partnered in 2009 to examine the utility of using both multibeam and single beam data in tandem for improved seafloor and shallow seabed classification and characterization. Results from two case studies will be presented. Multifrequency single beam echosounder (SBE) data at 38 kHz and 200 kHz from Dunmanus Bay were analysed to explore the potential utility of multifrequency classification. Comparisons between physically related SBE features and other geophysical data sets were made with the Malin Sea 12 kHz data to assess acoustic seafloor characterization.

It is certainly a commonplace among acousticians the fact that the research on acoustics represents an important segment of science and a significant necessity of society . In many situations, however, it is only marginally incorporated into the curricula of universities. This is particularly the case in the Brazilian educational and research institutions, mainly due to the lack of a comprehensive curriculum that accounts for basic education in acoustics, despite the enormous demand for acousticians in Brazil. To meet this demand, the Federal University of Santa Maria has conceived, in the end of 2008, a dedicated 5-year undergraduate course in Acoustical Engineering to provide students with more than basic knowledge in acoustics, vibrations and audio. Within the first 4 semesters the curricula is structured to provide fundamental knowledge for enginners with disciplines for engineering like math, physics, mechanics, electrical circuits and alike, acommpanied by disciplines concerning musical education for acoustical engineers (first-forth semester) and an introduction into the relation between humans and noise. Starting at the fifth semester education in acoustics, vibration and audio is broadend each semester with disciplines like room acoustics, noise control, environmental acoustics, electroacoustics, numerical methos in acoustics, psychoacoustic and so on. The structure of the course is similar to that used in the German and Chilean acoustic engineering schools and is intended to provide the undergraduate student with knowledge to work in the main fields of acoustics

CSA has made the decision to disband the Technical Committee Z107 TC on Acoustics and Noise Control. There was not enough support in CSA to maintain the environmental noise standards and the review and endorsing of international standards, including this area, is being transferred to a new CAA Standards Committee. However, the occupational noise control and audiometric standards from the group of standards overseen by the Z107 TC are maintained within CSA.The CSA OHS Standards Steering Committee, had decided to expand the mandate of the existing Hearing Protection TC and to transfer the occupational noise control standards under a new Committee – the TC Z107 Occupational Hearing Conservation. The Committee shall be responsible for developing and maintaining standards related to: a. Hearing conservation management systems; b. Workplace noise and vibration measurement and surveys; c. Determination, measurement, and assessment of occupational exposure to noise and vibration; d. Strategies for reducing exposure to noise and vibration (design, engineering, work procedures); e. General noise and vibration control systems in the workplace; f. Personal hearing protection devices; and g. Audiometric testing for early detection of occupational hearing loss. This presentation will provide details on the new TC as well about its members.

Non-traditional work shifts such as 12 hours shifts with 3 days on and 2 days off are changing the workplace, yet noise regulations all use a standard 8 hour shift to evaluate noise. This means that employees with identical long term exposures can have, in the example case, a 2 dB difference in daily noise exposures. This paper presents the issues both in terms of the difference in measurements, the regulations and also in terms of the difference in the effect of such shifts on employees’ hearing.

CSA made the decision to disband their Technical Committee Z107 TC on Acoustics and Noise Control which has acted as a clearinghouse for noise standards activity in Canada. They are keeping the occupational noise standards from Z107 and transferring them to a renewed Occupational Hearing Conservation Technical Committee which will also include the Z94.2 committee on hearing protection. CSA will be transferring their standard Z107.10 to the Canadian Acoustical Association’s newly formed Standards Committee. This standard is a listing of acoustical standards from Canada, the US, ISO and elsewhere which are deemed useful for Canadian use. Each listing includes a brief description f the endorsed standard and any recommended changes or concerns for its use in Canada. For example, some ISO standards do not adequately discuss the use of instrumentation in cold weather or the effect of snow on ground absorption and this would be flagged and recommendations made for dealing with these issues. In addition, the committee is expected to help coordinate the Canadian representation on a variety of ASTM, ISO, IEC and other standards groups. This was one of the original reasons that the CAA was set up and this function has now returned to its original home.

An extensive ambient noise monitoring program was conducted in rural Ontario under a wide range of meteorological conditions to assess theoretical noise predictions for an Energy Facility. It is standard practice for acoustic consulting professionals to evaluate the environmental noise impact from proposed or existing industry on adjacent sensitive land uses for compliance assessment and permitting using modelling software or a spreadsheet based on ISO 9613-2 “Acoustics – Attenuation of Sound During Propagation Outdoors – Part 2: General Method of Calculation.” ISO 9613�2 is based on the principle of "predictable worst�case" where the off-site environmental noise impact is evaluated based on downwind propagation for all sources to all receiver locations simultaneously under wind speeds of between 1 to 5 metres per second. An Energy Facility in Ontario was evaluated using a standard acoustical model that predicted off-site noise impacts that were at or below the regulatory noise limit. The model results proved to under-predict the off-site noise impact when compared to the 24-hour noise monitoring program at critical receivers. The baseline acoustic model was adjusted based on the worst-case receiver-specific meteorological conditions including atmospheric stability class, wind speed and wind direction that were determined to be appropriate from a historic meteorological data analysis. Acoustic professionals must be aware that the unadjusted ISO 9613-2 equations will under-predict the environmental noise impact significantly and that appropriate receiver-specific meteorological conditions should be considered to evaluate the predictable worst-case.

The organisation of sound for auditory monitoring is often considered from a perspective of superimposed simultaneous layers of sound: The grouping of auditory elements into streams, or the deconstruction of auditory scenes into different sound sources. A temporal perspective of sound is highlighted in Murray Schafer’s investigation into soundscapes - in which the elements are regarded in their spatio-temporal relationship as part of an environmental ecology. In this paper, we would like to highlight the perspective of the human perceiver on the temporality of sound - our own contribution to the way sound occurs to us and how we can enable us as listening beings to derive more meaning from what we hear. Here, we would like to focus specifically on the temporal aspects. Marshal McLuhan highlights the nature of media as “Extensions of Man”, while George Lakoff emphasises the importance of metaphorical structures to the way the world occurs to us. Temporal metaphors we could apply in this bi-directional information exchange between our approach to the world and the way the world occurs to us can come from a variety of origins (the time of music, familiar temporalities in our environment and everyday experience, temporal structures of communication and our own body, among many others). As concrete implementations scientific sonifications and audifications of seismological data are presented and evaluated: - A perceptual analysis of audified seismograms. - sonification of earthquake event catalogues under the application of different temporal metaphors This will guide us through an investigation of sound generation, how temporal structuring of sounds can target our specific perceptual abilities better, and finally, the potentials of our own openness of listening “for". This will provide us with concrete handles to make auditory applications in the field of discovery-oriented science more relevant to a human listener.

A novel concept for dynamic hearing protection has been developed based on an automatically opening and closing gate. Using miniaturized electronics this solution can be worn in the ear with a very low energy consumption meeting international standards for hearing protectors. The underlying physics of dampening sound through a varying orifice will be discussed in relation to attenuation data obtained in the lab and with test persons. Current CE certification methods of dynamic hearing protectors pose an issue to test the devices compared typical usage conditions. A lab based test method is proposed to ensure quality and performance.

While speaking, a human subject will perceive his own voice mostly through the free-air radiation path from his mouth to his eardrum. Nevertheless, the vibration of his vocal tracks and mouth cavity are also inducing a bone (skull) conducted vibration that will excite directly the cochlea, but that will also induce vibration of the earcanal walls. When the earcanal is occluded by an in-ear device (like an earplug), assuming that the device is providing a good acoustical seal (from the external environment) while leaving the earcanal walls free to vibrate (device shallow inserted) the regenerated sound pressure level inside the occluded earcanal can be clearly perceived by the human subject (it is called the “occlusion effects”). While the occlusion effect is often perceived negatively, since it is changing the normal perception of one's own voice, it also enables the pickup of one’s own voice from within its occluded earcanal. A microphone placed in a well occluded ear will provide a strong signal, free from external disturbance, but the resulting voice pick-up will be perceptually different from the one from the free-air path. Usually, the resulting voice signal will indeed be colored and will never sound as “clear” as a free-air voice taken under the same acquisition conditions, unless extra signal processing is applied. The purpose of the current research study is to characterize the differences in signal characteristics of the occluded ear voice pickup relatively to the free-air voice pickup, in the form of a transfer function between the two signals. The results obtained from a coherence-based method will be presented, together with possible applications of this research.

Communication headsets are increasingly used in the workplace. In some of the most challenging environments (e.g. military, law-enforcement tactical operations), the device must protect hearing against hazardous continuous and/or impulse noise while maintaining good situational awareness (e.g. sound localization, speech communication) within the immediate surrounding and during radio communications with remote locations. These objectives must be met despite the presence of noise-induced and other types of hearing losses among users. Several analog and digital headsets are commercially available with adaptive level-dependent passive/active noise reduction, user-adjustable talk-through or surround volume, in addition to radio capabilities. Unfortunately, the technical specifications supplied by manufacturers are very limited, even for high-end products, and do not readily allow for a systematic analysis of the best devices or parameters to use in specific communication scenarios given the noise characteristics, task demands and hearing status of the user. This is in contrast to the hearing aid industry, where product testing and specification sheets are very extensive and highly standardized, and where systematic fitting procedures exit to optimize communication. In the paper, the characteristics of two high-end tactical communication devices are reported. The devices are the Peltor Powercom Plus (circumaural) and the Nacre QUIETPRO (intraaural). The test battery included measurements of (1) the passive sound attenuation, (2) the insertion gain and compression parameters at various control settings of the talk-through/surround modes, and (3) the speech intelligibility in two military noises with subjects covering a wide range of hearing profiles. [Work supported by DRDC Toronto].

The use of air attack on wildfire causes fixed-wing aircraft and helicopters to operate in close proximity within a designated airspace. The Bird Dog crew members are charged with the responsibility to co-ordinate the direction of air traffic over and in the immediate vicinity of wildfire alongside the ground firefighting. The crew of the Birddog consists of a pilot and a specially trained forest fire fighting person. These workers wear communication headsets during their entire mission, can receive transmissions through as much as 9 to 10 channels and can fly over long period of time ranging from 3-4 hours to 8-9 hours. There are thus great concerns regarding the noise exposure for such activities. This paper presents the comparison of two techniques for measuring the exposed levels (noise under the headset) for different workers in real flight conditions. The first technique, based on the F-MIRE technique, uses dual miniature microphones inserted into the ear muff to measure the noise outside and inside the ear muff. The second technique uses a probe-tube microphone placed close to the tympanic membrane to measure the exposed signal. A communication headset was instrumented with the necessary equipment for both techniques and simultaneous time recordings are performed during entire flights missions to allow systematic comparisons of the two approaches. Results for different workers and different flight conditions are presented and discussed.

Common methods for predicting speech understanding, the speech transmission index (STI) and the speech intelligibility index, fail when the speech signal is corrupted by nonlinear distortion [Yu G., et al., J Acoust. Soc. Am. 127, 1903 (2010)]. Models based on the STI have been constructed to predict the intelligibility of speech subjected to center clipping, with clipping thresholds from 2% to 98% of the cumulative magnitude histogram. Subjects (3 male, 3 female) with normal hearing and confirmed understanding of American English were seated in an anechoic chamber. The psychoacoustic paradigm was the Modified Rhyme Test (MRT). Speech was reproduced by a small, high-fidelity loudspeaker located on the ear-nose plane, 2.4 m from the center-head position. The models employed unperturbed speech as the input (probe) signal. The intensity modulation spectrum input/output transfer index (from 0.63 to 12.5 Hz), MTIk, incorporated the coherence between the signals. In addition, the index of intelligibility was adjusted to take into account the correlation between speech sounds in different octave frequency bands from 125 Hz to 8 kHz (k = 1, ..7). This was achieved by a summation of inter-band contributions of the form MTIk * MTIj (where j = k+1, ..7) for the kth octave band. The ability of the preferred model to predict the intelligibility of distorted MRT words will be discussed. [Work supported by NIOSH grant R01 OH008669.]

Musical noise is a term used to describe short-duration narrowband artifacts present in speech processed by noise suppression systems. These artifacts are so distracting and un-natural sounding that listeners generally prefer the original noisy signal to enhanced speech with musical noise. While the effect is well known, little has been done to understand musical noise on a theoretical basis. In this work we use statistical noise models to demonstrate how these artifacts arise in speech enhancement, and to explain why different algorithms exhibit significantly different artifact levels and why there is a limit to artifact-free suppression.

The performance of a digital feedforward active noise reduction (ANR) circumaural hearing protector device (HPD) is strongly affected by the design of the ear cup. The selection of an optimal ear cup design must account for the passive attenuation and the predicted active attenuation if a feedforward ANR control system is introduced. Several commercially available ear cup designs have been evaluated to predict performance if they were converted into feedforward ANR HPDs. Measurements were performed in a pseudo-diffuse sound field using a head and torso simulator (HATS) mounted with an artificial ear (Brüel & Kjaer Type 4128C). Active attenuation of the ear cup was estimated via the coherence function between the sound pressure from a reference microphone outside the ear cup and an error microphone at the desired point of cancellation, the “eardrum” of HATS. Results demonstrate a need to balance the passive attenuation, which generally improves with increased volume underneath the ear cup, with the need to develop a coherent signal between the reference and error microphones, which generally improves as the microphones are moved closer together. The ear cup selected for optimal predicted performance was then modified to form a feedforward ANR HPD and its attenuation was measured. The results suggest guidelines to follow if a custom ear cup solution is explored for future feedforward ANR HPDs. [This work was supported by NIOSH research grant R01OH008669.]

This paper shows the change of sound transmission due to adding a floating concrete topping on a concrete floor, as it was investigated in a small study at the National Research Council Institute for Research in Construction (NRC-IRC). Furthermore, it looks at the effects of changing the stiffness of the resilient interlayer and compares measurement results to theoretical results, obtained by a simplified mass-spring model. The assumptions made with this model are only applicable to the measured assembly for low frequencies, where the wavelengths are big. The assembly was evaluated using the standard airborne test (ASTM E90) and three standard impact sources: ISO tapping machine (ASTM E492), ball (JIS A 1418-2, ISO 140-11) and bang machine (JIS A 1418-2, KS F 2810-2). The measurement results of the assembly with floating topping display, as expected, an overall improvement compared to the bare concrete floor, except in the 50Hz and 63Hz frequency bands, where the system has its fundamental mass-spring resonance. In order to achieve a further improvement, this resonance was shifted to a lower frequency by replacing some of the rubber interlayer with air. The shift in the resonance frequency coincides well with the in-advance calculation using the simplified mass-spring model.

As part of a performance verification exercise reverberation times (RT) were measured in several newly constructed school gymnasia, rectangular in plan with two variations in room size, all with similar finishes and constructions. Due to architectural constraints the rooms are absent of acoustically absorptive finishes below a height of 3 m. The room finishes are primarily acoustically reflective with the exception of continuous bands of absorptive upper wall paneling around the full perimeter of the rooms (exposed unpainted Tectum over mineral fibre insulation) and painted acoustic metal deck ceilings (fiberglass insulation in the perforated deck flutes). The initial RT measurements exceeded the performance requirements. Modeling using ODEON room acoustics prediction software was conducted to determine the quantity and placement of additional absorption required to bring the RT into compliance. After installation of an additional continuous band of absorptive paneling in the rooms at a height below the existing panels, the RT were re-measured and found to be significantly higher, particularly in the critical mid-frequencies. Further investigation lead to the hypothesis of a lack of sound diffusion in the lower portion of the room contributing to the unexpected results. RT were subsequently re-measured under five different conditions; an empty gym, addition of 5 people, three levels of diffusion. Diffusion was varied by simply adding sheets of plywood (5, 10, 15 sheets) leaned against posts or each other. The addition of as few as 5 people or 5 plywood sheets was found to significantly reduce the measured RT, closer to the modeled predictions.

This presentation will review experimental approaches to measuring the sound-absorption characteristics of vegetative roofs or their components, and will summarize laboratory and field-test results. An impedance-tube method was used for the determination of the effects of the composition and micro-climatic condition of the green-roof substrate. Evaluation of six substrates and three constituents indicates that the percentage organic matter, the volumetric water content and the compaction affect the impedance and hence the absorption coefficient of the substrate in the non-vegetative state. A spherical-decoupling method has been developed for the determination of the effect on the impedances of various green-roof samples of substrate depth, plant-species diversity and biomass establishment; preliminary results from 18 green-roof research plots will be presented.

In open-plan offices speech is often the most distracting cause of noise that can disturb the workers. Therefore, appropriately designed noise-masking systems are often used to increase the speech privacy and to enhance the general level of acoustical comfort and productivity in the workplace. The aim of the present study was to evaluate the existing noise-masking system on one of the two floors of a community-health-care facility in North Vancouver, by determining its effects on background-noise levels, speech privacy, and on the workers. To achieve that, acoustical measurements were carried out on the two floors to determine background-noise levels when the system was on and off, and speech levels. Moreover, a questionnaire was developed for the investigation of the effects of the system on the workers, and their satisfaction with the workplace acoustical and general environments. The database contains a total of 31 respondents (18%) from the two floors. Statistical analysis was used to find a relationship between the results of the physical measurements and the questionnaire responses. Final results showed that the system increased the background-noise levels by 4 dBA; it provided reasonable speech privacy at short distances between workstations, while it was not effective at long distances. Employees on both floors have trouble remembering things and concentrating due to intermittent noises and speech. Because of the small sample size, there was no statistically-significant difference between the two floors regarding dissatisfaction with the overall environmental quality, the acoustical conditions and the speech privacy. Overall, this study emphasises the importance of considering the acoustical conditions to achieve satisfaction with the workplace overall environmental quality in a health-care facility.

ASTM E2638 "Standard Test Method for Objective Measurement of the Speech Privacy Provided by a Closed Room" describes a test method suitable for use in enclosed rooms of nearly all sizes. It defines a measure called Speech Privacy Class (SPC) that can be used to accurately rate speech privacy over a very broad range, from no privacy to very high secrecy. SPC is the sum of the measured average noise level at the position of a potential eavesdropper outside the room, and the measured level difference between a source room average and the transmitted levels at the same potential eavesdropper location. For a given situation, the likelihood of transmitted speech being audible or intelligible can be related to the probability of higher]speech levels occurring in the meeting room. Increasing speech privacy can be defined in terms of increasing SPC values. The measurement standard provides guidance on interpretation and use of the result (SPC), but does not set criteria. This paper will describe the new E2638 measurement method, and an approach for setting criteria using SPC.

To rate the acoustical performance of floor-ceiling assemblies excited by impact sources (e.g., footsteps), an objective metric must correlate well with the subjective judgments of listeners hearing the radiated sounds. For each of a series of full-scale lightweight floor-ceiling assemblies, physical measurements and sound recordings were made of the sounds generated by standard impact sources (tapping machine, rubber impact ball, bang machine), and by adult walkers (with and without shoes). The physical measurements were used to calculate standard and non-standard metrics. The recordings were played back for listening test participants, who subjectively rated the radiated sounds. The correlations among the objective and subjective ratings were calculated. Results indicate that some standardized sources and metrics are not optimal for rating subjective performance.

Providing “Good”, “Better”, or “Best” proposals for acoustical noise and vibration isolation of piping systems to cost sensitive clients When it comes to the costs associated with effective acoustical isolation of a plumbing or piping system there are a variety of choices available. Though effective isolation materials and methods are available for very modest costs, there is still often the challenge of “Value Engineering” to face. As in most facets of building construction, there are a variety of quality levels available when it comes to plumbing and piping system acoustic isolation options. Become familiar with “Good”, “Better” and “Best” materials and methods, in order to assist you during the planning and budgeting stages of a building project. Learn to provide valuable input during the Cost–Benefit analysis phase of a building project. Base your input to your client upon solid laboratory test data arranged by specific plumbing or piping system applications, such as through-stud isolation, riser clamp isolation, shower head attachments, etc. This paper will provide fundamental presentation points and cost analysis templates easily customized to any plumbing or piping system. These analytical tools take into account both labor and material factors, in order to generate a true “Installed Cost Analysis”, while assisting you to specify a proven engineered system for your client.

Green roofs on top of commonly used roofing system design are becoming mandatory in major cities around the world. One of the potential benefits of green roofs is touted to the noise reduction of the added mass of the green roof structure. The transmission class of the green roofs has become a major research project in acoustics. A model of an extensive green roof was investigated as to its noise reduction potential as part of an undergraduate thesis. The noise reduction of different layers of the green roofs was measured in a mock-up 2 m X 2 m green roof model. The results of the noise measurement will be presented.

Considerable research has been conducted in understanding as well designing acoustics of performance spaces of conventional auditoria used for music, drama and speech. However, not much is known about the acoustics of spaces used for temporary performances. Typical examples of such spaces are the large group of venues used for jazz festivals. Large single arena, temporary of course, is used for the jazz festivals in Toronto and Halifax. On the other hand Montreal uses a number of venues, including one large space, for the Montreal Jazz Festival held every July. The acoustics of the three main arenas of Toronto, Montreal and Halifax was investigated. Computer simulations of the three venues were conducted. In addition, simple surveys of the stakeholders of the three venues were undertaken to compare the simulation with audience satisfaction. The results of the acoustics of temporary performance facilities will be presented.

Considerable theoretical research has been conducted in understanding the design constraints as well as on the locations horn speakers in reverberation rooms. Many of the research methods applied simple sinusoidal source functions (tones) to evaluate the design criteria of horn speakers. The understanding of the behaviour of horn speakers, when band limited random noise signatures such as pink noise and white noise are used as input sources, is still not clear. An hyperbolic horn with cut off frequency of 75 Hz was used in a medium sized reverberation room to study the horn behaviour. Some of the basic questions to be studied were the influence of horn location on the cut-off frequency, as well as the influence of the horn location on the diffuse sound field in the reverberation room. In addition, the influence of the input sound source on the room sound levels was also studied. The results of the experimental results will be presented in this paper.

An apparatus and procedure, based on ISO 17497-1, were developed for measuring random incidence scattering and absorption on a limited budget and with limited facilities. Three original wooden architectural surfaces – one flat, one with sinusoidal corrugations and one comprising rectangular sections of different heights – have been characterized in terms of their scattering and absorption coefficients. The two non-flat surfaces have been modified for increased sound absorption and scattering, and re-characterized to confirm the improved performance. The measurement method proved sufficient for determining the general behaviour of the surfaces.

This paper discusses the development and testing of sound-source localization methods which use a hemispherical array of microphones and source-localization analysis algorithms to estimate the directions of the sound energy incident from sources in an enclosed environment. Two analysis algorithms were implemented, based on time delay of arrival (TDOA) analysis and on delay and sum beam-forming (DSBF). The structure of the microphone array and the theory behind it are presented. Tests were performed in anechoic and reverberant environments with single and multiple sound sources. The effectiveness of the hemispherical microphone array and the two algorithms to estimate the directions of real and virtual sound sources was investigated. It was concluded that the TDOA method works relatively well for one source in anechoic rooms or in rooms with weak reflections, but the algorithm requires too much computational complexity for applications with several sources. The DSBF method, on the other hand, works for one or more real or virtual sources and estimates their directions to within 3-4 degrees of accuracy.

This paper presents the development of a wave-based, triangular beam-tracing model for predicting sound fields in rooms. The model is based on the one developed by Wareing and Hodgson [J. Acoust. Soc. Am., 118 (4), 2321-2331]. The original model predicts the steady-state response of empty rooms with specularly-reflecting multilayer surfaces. In addition to improved computational efficiency, the new model predicts the transient response of rooms; i.e. the pressure impulse response between the source and receiver. Moreover, the model has been further extended to include sound diffraction around wedges based on the theory developed by Svensson et al. [J. Acoust. Soc. Am., 106 (5), 2331-2344]. This broadens the application of the model to situations with more realistic features such as sound propagation in fitted rooms or in long enclosures with bends, and evaluation of screen barriers in open-plan offices. The improved model is used to compare the effects of different surface-reaction models on the transient response of empty rooms and various room-acoustical parameters that correlate with subjective perception of sound (reverberation time, for example). In addition to investigating the significance of extended and local reaction of surfaces, effects of phase changes due to surface reflections have been studied by considering real and complex reflection coefficients. Moreover, energy impulse responses and room-acoustical parameters have been compared with those of conventional energy models which ignore all phase effects. Finally, the diffraction model has been validated by comparing prediction results with experiments on a finite-length barrier over a flat surface in an anechoic chamber.

One of the minimum design requirements in healthcare facilities is to assure satisfactory acoustical and privacy environments. A satisfactory acoustical environment in a healthcare facility is one in which all the sounds are compatible with the intended use of the space. The inclusive acoustical quality assessments address the environmental and mechanical noise, vibrations, and required acoustical performance within each area considering applicable codes and regulations. Providing satisfactory acoustical environment in healthcare facilities can be ensured by applying recommended minimum design requirements provided in Sound and Vibration Design Guidelines for Hospital and Healthcare Facilities. The objective of this project was to provide acoustical conditions in compliance with the Healthcare design guidelines when a giant chiller and six pumps were going to be installed in a mechanical room, below an auditorium within a hospital. The job involved conducting measurements in a mechanical room with the same type of equipment and within an auditorium; reviews the floor/ceiling assembly and all penetrations due to these additions. The result was a satisfactory acoustical condition for the occupants after overcoming all challenges.

In the case of a retrofitting project for large open-plan offices, it was appropriate to test a prototype of integrated furniture before completing specifications for different floors. As acoustical environment was initially good in the building, it was requested to assure that the new ceiling and the new system of office furniture could offer similar performances regarding speech privacy or even better results (especially for a call center). Four modular workstations have been placed in a large room to accomplish various acoustic measurements. Speech privacy has been compared between 19 configurations of the same prototype, which included the type of acoustical tiles for the ceiling, the material and the height of screens, or by adding acoustic baffles, localized absorption, and sliding doors. As expected, results show that the choice of materials for the ceiling and screens has the greater impact on speech privacy, and that height of screens could be reduce when baffles are used.

Effect of Some Floor-Ceiling Construction Changes on Flanking Transmission Frances King, Stefan Schoenwald and Brad Gover National Research Council, Montreal Road, Ottawa, K1A 0R6 A large study was carried out by the National Research Council-Institute for Research in Construction (NRC-IRC) to evaluate a wide range of floor-ceiling designs to control the transmission of impact sound. The study has identified a number of construction changes that are effective in reducing direct impact sound. However, the system performance of floor-ceiling assemblies depends not only on the direct transmission but also on flanking transmission. Further studies were carried out in the Flanking Facility to examine the effect of these construction changes on flanking transmission. The construction changes examined in the Flanking Facility include adding a heavy topping or decoupling the ceiling from the floor assembly, either by adding resilient channels or by having separate ceiling joists. The test results show that certain construction changes, while reduce the direct transmission effectively, can cause flanking transmission to become more important. It is necessary to examine the overall system performance when construction changes are made to floor-ceiling assemblies to improve impact sound insulation.

The passive sonar systems that are often used to localize and track marine mammals by detecting their vocalizations are also triggered by transient sounds generated by sources other than marine mammals, leading to a large number of false alarms. Even in the case of a successful detection, accurate classification of the genus and species is often required and this relies on expertise that is not typically readily available on the vessel. Perceptual signal features, similar to those employed in the human auditory system, have successfully been shown to reduce false alarm rates in active sonar research by automatically discriminating between target and clutter echoes. Many of the signal features employed were inspired by research focused on discriminating the timbre of different musical instruments – a passive classification problem – suggesting this method might be applied to the classification of marine mammal vocalizations. To test this hypothesis, the automatic aural classifier was trained and tested on a small set of marine mammal vocalizations from four species: the northern right whale, sperm whale, humpback whale, and bowhead whale [J. Acoust. Soc. Am. 127, No. 3, Pt. 2, p. 1970 (2010)]. The present paper provides an overview of the aural classifier's architecture and expands the data set tested with the classifier.

The Snohomish Public Utility District of Washington State is applying for permits to install up to two in stream tidal turbines in Admiralty Inlet, Washington State, as part of a pilot project to determine the feasibility of marine tidal energy generation. To inform the permitting process, acoustic recordings from a vertical hydrophone array were made in Admiralty Inlet as Southern Resident killer whales transited through the study site to determine depth of whales in this area. The vertical array consisted of four hydrophones at ten meter offsets with the shallowest at a depth of 10 meters. A total of 682 calls and echolocation clicks were localized using hyperbolic localization techniques. The Time Of Arrival Difference (TOAD) was calculated using cross-correlation techniques for all calls and click trains. TOAD values were calculated for individual clicks by hand picking the times of arrival. Where present, surface reflections were also incorporated into the localizations of single clicks. We validated our techniques by generating signals at known depths (10 to 60 meters) and distances (100 to 500 meters). Measurement errors were calculated for each localization. Results of the validation study, error estimates, and depth measurements will be presented.

Automatic detection of signals in noise is a common problem in many areas of acoustics. In the field of passive acoustic monitoring of marine mammals, the signals to be detected are vocalizations. The noise originates from natural (wind, waves, rain) and man-made sources (e.g. shipping, construction, seismic surveys). Signal characteristics vary broadly: frequency ranges from a few Hz to 200kHz, duration from milliseconds to seconds to hours. Noise characteristics vary by similar orders of magnitude. While specific automatic detectors have been designed to successfully find specific calls in specific environments, the challenge is to find a large variety of calls in a large variety of noise. An exploitable difference between calls and noise is that most noise is a result of stochastic processes (wind, waves, rain, cavitating propellers & seismics generate gas bubbles underwater of varying size + resonance frequency), while many animal signals are a result of deterministic processes (vibrating strings & cavities of predetermined/fixed size). Shannon entropy was computed for power spectrum density functions of underwater recordings. Noise yielded high, signals low entropy. Results are presented from passive acoustic surveys of marine mammals. The benefits and limitations of entropy applied to automatic signal detection are discussed.

There are a number of sources of bias in estimating the number of marine mammals likely to be affected by underwater noise. These include interspecific and individual variation; variability in sound propagation conditions; the interaction of uncertainty with the non-linear decline of received sound levels with distance; interchangeable use of data reflecting most sensitive individuals, statistically significant results, and 50% effect thresholds; studies based on subsets of populations least sensitive to noise; sociality; and extrapolations from experiments on species that tolerate captive relatively well. Sensitivity analysis was performed to determine the importance of bias in parameters used to estimate numbers of individuals likely to be influenced when individual variation is considered. Data from a variety of platforms used to observe marine mammal behavior during seismic surveys were used to demonstrate bias due to the area sampled. Sociality can result in individuals being affected due to behavioral changes by the most sensitive member of the group. Controlled experiments with captive marine mammals have generally been limited to the species that exhibit good survivorship in captivity, but experiments with stranded porpoises indicate that species that have survived more poorly may be more sensitive to noise. Increasing uncertainty leads to increasing bias toward underestimating the number of individuals to be affected. These results have important implications for mitigation monitoring, identifying habitat-specific risks, estimation of cumulative effects, and the data needed for more accurate estimation of the effects of noise (primarily the level at which 50% of individuals respond in many species).

Mosquitoes hear through stimulation of their Johnston’s organs, complex sensory structures in the second segment of their antennae. These specialized structures detect movement of the distal part of the antenna as it is displaced by sound waves. Thus, the antennae act as mechanosensory transducers. In some species, the long fibrils on the male flagellum are erect when the insects swarm and mate at dusk and dawn, and recumbent during the rest of the day. It has been claimed that when the fibrils are recumbent males cannot hear. To test this, we used laser vibrometry to examine the responses of Aedes togoi antennae with erect and recumbent fibrils to sound stimuli of frequency sweeps. From the sweeps, we determined the resonant frequency and calculated the quality factor, or damping, of the antennae. Our results show that antennae with recumbent fibrils respond differently from those with them erect, but do respond to sound. Therefore, male Aedes togoi, like the females, do not need long, erect fibrils to hear.

Naval exercises include the use mid-frequency active sonar (MFA), which emits high intensity sound between 2 and 9 kHz into the ocean. The returning echoes provide an acoustic picture of the environment, often reaching ranges of several thousands of square kilometers. Given that the frequency band used by the MFA falls into the hearing sensitivity of the marine mammals, and some correlations found between naval exercises and anomalous strandings of beaked whales, there is currently a debate about whether the MFA affects cetaceans and if so, to which extent. To address this question, we used High-Frequency Acoustic Recording Packages (HARPs) to continuously record sounds between 10 Hz and 100 kHz, and deployed them in two different sites in Southern California for over one year. The broadband recording enabled us to pick up the MFA signals and most calls produced by cetaceans in the area. The analysis of the anthropogenic signals showed a variety of signal design, ranging from constant frequency to frequency modulated signals, or combinations of them at varying frequencies. The power spectral density of 1 hour windows revealed a significant contribution of the MFA to the overall ocean noise level at mid-frequencies. To assess the possible effects of the MFA on cetaceans, correlations between different parameters of the MFA and the acoustic presence of cetaceans will be discussed.

Acoustic detection and localization of marine mammal clicks using multiple synchronized sensors can be achieved through many techniques. Time Difference of Arrival (TDOA) techniques require precise time information on detected events. Marine mammal click localization is further complicated in that marine mammals commonly generate a sequential pattern of clicks; it may be difficult to identify corresponding clicks across multiple sensors. In this work, a set of clicks is selected from a single sensor for a certain time period. The time series data are transformed and filtered before being cross-correlated with time series from the other sensors in order to obtain estimated candidate values for each TDOA. The resultant time delays are then input into a hyperbolic localization algorithm. This paper will discuss a data set collected in the Strait of Georgia where marine mammal clicks were synthetically generated and transmitted. Acoustic signals were measured using fourteen sonobuoys and post-processed to localize the source. A comparison of actual source locations (as measured by GPS) with estimated source positions will be included.

Current ultrasound contrast agents (UCAs) used clinically have not been optimized for high frequency imaging (10 – 70 MHz). UCAs models used at high frequencies that are based on the Rayleigh-Plesset equation or variants can only predict resonance frequencies at which the UCAs undergo radially symmetric oscillations. They cannot account for features such as asymmetric bubble oscillations and the interactions of bubbles with their surroundings. In this work, we introduce a finite element scattering model that allows asymmetric bubble oscillations and models the interaction of the bubble shell with the incident pressure wave using the constitutive stress-strain relationship. The COMSOL Multiphysics software package was used to develop a 2-D axi-symmetric finite element model to study scattering from contrast agents subject to high frequency ultrasound. The Helmholtz equation was used to describe the propagation of sound waves in the UCA gas core and the surrounding medium. The constitutive equation for the elastic material was used to describe the stress-strain relationship in the UCA shell. Initially, the far-field backscatter response of a 5 ìm contrast agent BR-14 having a 3 nm phospholipid shell thickness was considered. Then, the scattering behaviors of 25 nm, 125 nm, and 250 nm shelled UCAs were studied at 10 – 70 MHz. A good agreement (error < 5%) was found between the finite element and analytical solutions (Church model) of the UCA resonance frequencies (the radially symmetric monopole resonance). Increasing the shell thickness increased the monopole resonance frequency (5.4 MHz, 12.1 MHz, and 17.6 MHz for the 25 nm, 125 nm, and 250 nm shelled UCAs, respectively) and broadened the resonant peaks. The finite element model revealed the presence of two resonant peaks of comparable magnitudes for these UCAs. The frequency of the second peak also increased with shell thickness. Only one dominant resonant peak was found for the 3 nm UCA within the frequency range studied. The developed 2-D model has many advantages over the previously developed 3-D models for studying UCAs behaviors. It requires much less computational resources and execution times and can be used to calculate all quantities of interest, such as stresses and strains at the surface of the UCA, surface modes, etc. The increase in the shell thickness of the UCA produces a shift in their resonant peaks toward higher frequencies, which well agrees with the Church model predictions. The presence of other resonant peaks in the backscatter from the 25 nm, 125 nm, and 250 nm shelled UCAs may be used to enhance the effectiveness of the ultrasonic imaging systems at high frequencies. The implications of these results on the use of ultrasound contrast agents for therapeutic ultrasound and small animal imaging are discussed. Future work includes the use of the developed model for the optimization of UCAs for high frequency ultrasound imaging.

The marine soundscape is made up of natural ambient sounds (e.g. wind and waves), biological sounds (e.g. animal calls) and anthropogenic sounds (e.g. ship noise). Acoustic ecology studies the relationships—mediated through sound—between organisms and their environment. As ocean water conducts light very poorly yet sound very well, marine mammals rely heavily on acoustics for communication and navigation. Since the onset of the industrial revolution, man-made noise in the ocean has steadily increased. The effects of noise on marine animals can be short-term or long-term, transient or chronic, negligible to biologically significant, where the survival of a population is at risk. This article gives an overview of the components constituting the marine soundscape, of the use of sound by marine mammals and of the effects of noise. The acoustic ecology of animals other than mammals and the effects of noise on animals other than mammals are less understood.

Acoustic communication is important in many insect groups. This overview covers the mechanisms and diversity of acoustic behavior found in psyllids. Psyllids (Psylloidea) are small (2-8mm) plant feeding insects related to aphids, scales, and whiteflies. Their sound making apparatus consists of a "simple" stridulatory system. However, I found a much greater diversity of sounds than may have been expected from a stridulatory mechanism. The role of acoustics in psyllid mating and speciation is illustrated with data from allopatric versus sympatric groups and morphologically cryptic species. This acoustic research was undertaken in one of the centres of species diversity for this group: Australasia and the Pacific Hawaiian Islands.

A Bayesian travel-time inversion method was developed for 3D localization and tracking using multipath arrival times of knock-type sounds produced by walruses. The method accounts for data errors and uncertainties in environmental and geometric parameters. Data were collected on three seafloor sound recorders, arranged in an equilateral triangle with approximately 500-m sides, deployed in 30-m of water in the Chukchi Sea near the Hanna Shoal, west of Barrow, Alaska. Pacific Walrus (Odobenus rosmarus divergens) calls were recorded by these recorders for 2.5 months in the summer of 2009. A regularized, linearized inversion algorithm was used to estimate 3D tracks and track uncertainties for these calling walruses. Regularization incorporates prior information (expected values and uncertainty estimates) for the recorder locations and acoustic environment (water depth and sound speed), as well as preferred track characteristics. Inverting for the smoothest (simplest) track consistent with the acoustic data and prior information mitigates the risk of over-interpreting track structure due to data errors and environmental and geometric uncertainties. Hence, this approach provides more plausible sequences of source positions than a non-regularized solution. Tracks for walrus dives from the 2009 dataset will be presented.

Click-evoked oto-acoustic emissions (OAEs) from human ears are useful especially in the case of patients with whom communication is impossible. If the outer hair cells (OHCs) in the cochlea are damaged, these emissions are weak or absent. In the present analysis of human emissions documented in the literature, it is shown that the time-dependence of the instantaneous frequency of the emissions is consistent with a cochlear model [1] involving two resonators, namely the IOCR (internal organ-of-Corti resonator; spring = OHCs and surrounding structures) and the BMR (basilar-membrane resonator; spring = fibres of the BM). At given distance-from-base in the basal half of the BM, the resonance frequency of the local IOCR is lower than that of the local BMR by about one octave. The IOCR oscillations are thought to cause, during a sine-tone, the OHCs to feed mechanical energy into the cochlear travelling wave and thus to give rise to the "active" response peak. A fraction of the OHC-generated mechanical energy is conjectured to be carried back to the stapes by a backwards-travelling surface wave and thus to form the evoked OAEs. Emissions of a different type, namely the spontaneous OAEs, are attributed to feedback-generated BMR oscillations involving evanescent (standing) liquid sound-pressure waves. [1] R. Frosch, Old and New Cochlear Maps, Canadian Acoustics Vol. 37 No. 3 (2009) 174-175.

It is a well known fact that workers in the field get less protection from their HPs than the predicted from laboratory measurements. This is particularly true for the earplugs that are prone to be improperly fitted. The only way of knowing the actual attenuation provided by the device is by fit testing the individual. There are other advantages for such a procedure, e.g. raising awareness, training workers in the proper fit of protectors, etc. New technologies have been recently developed to allow for attenuation tests to be performed quickly without the use of sophisticated installations. This presentation will focus into devices and techniques used to perform quantitative and qualitative individual attenuation tests. Results are expressed as PARs (Personal Attenuation Ratings), valid for a particular worker, wearing that particular protector. Subtracted from the dBA level of the noise in the workplace it can be used to calculate the noise level of the protected ear. It has to be noted that so far there are no standards (either national or international) to validate there tests, nor can the results be compared to those obtained through normalized procedures

Noise measurements were conducted in over 50 nightclubs in the Greater Toronto Area as part of the licensing process. Sound levels were gathered in a manner representative of the exposure which would typically be experienced by patrons and some wait staff. Those sound levels ranged from below 80 dBA to above 100 dBA on a energy average (Leq) basis. Sound level distributions and spectral data are presented. The results are discussed with regard to damage risk potential considering recent changes to the Occupational Health and Safety Act of Ontario.

As part of the application process for extended hours of operation, the City of Vancouver requires local pubs and nightclubs to demonstrate compliance with its noise bylaw. Since 2007, BKL Consultants Ltd. has provided this service to over 40 entertainment venues in Vancouver. Part of the methods used to establish an entertainment venue’s contribution to the sound level at noise sensitive locations include the installation of an overnight noise monitor inside the facility. The collected data reveals that both patrons and staff are exposed to potentially hazardous sound levels. This paper will present measurement results from the collected data set in terms of the equivalent continuous A-weighted sound pressure levels for hourly and 8-hour time periods.

The popularity of hardwood flooring in new condominium buildings and the replacement of carpeting with hardwood flooring in existing buildings can result in inadequate impact noise insulation between vertically adjacent units. In Victoria, many condominium developers and strata councils are grappling with the decision of whether or not to allow hardwood floors. This typically proves to be a difficult decision. If the developer or strata council do not permit the installation of hardwood flooring they risk turning away prospective buyers or dissatisfying strata members. If they do, they risk the creation of a serious noise problem which, in Wakefield Acoustics Ltd.’s experience, can lead to persistent noise complaints and, in some cases, threats of legal action. Wakefield Acoustics Ltd. has recently conducted a series of FIIC tests in a concrete condominium building in Victoria to assist a developer in deciding whether or not to allow the use of hardwood flooring in the building. Nine FIIC tests were conducted in which the same hardwood flooring was placed upon nine different resilient underlayments. A tenth test was conducted where a different type of hardwood flooring was placed upon one of the nine underlayments previously tested. The methodology used to conduct these ten FIIC tests will be discussed and the results of the tests will be compared. This presentation will also discuss what range of FIIC values might be expected to be acceptable in multi-family buildings.

Hospital noise levels around the world have increased steadily over the past 50 years. High noise levels are among the top complaints of both patients and hospital staff members. High noise levels disturb patients and staff members, hinder speech intelligibility and raise the risk of medical errors. Noise levels often remain high during night and can interfere with patients sleep. Wakefield Acoustics Ltd. has recently conducted a survey of the noise environment within the West and Royal Blocks of Royal Jubilee Hospital in Victoria, B.C. The noise monitoring was conducted continuously over an eight day period at four locations within each of the patient care wards. To identify the sources of noise within the hospital, one of the four sound level meters employed within each patient ward also recorded a digital audio file and several hours of attended monitoring was conducted in each ward. The results of the noise monitoring were used to identify the most significant sources of noise and their respective noise levels. At a later date, this noise survey will be replicated within the new Patient Care Center at the Jubilee Hospital to see if noise control measures implemented within the new facility have been effective.

Large bridges require expansion joints to accommodate thermal expansion and contraction and seismic events. Different styles and sizes of expansion joints are employed depending on bridge type, length and other factors. Virtually all expansion joints create some additional noise over and above normal that due to normal tire-pavement interaction. Many joints create “banging” or “booming” noises due to the impact of tires on leading edges of the joint, which often feature some vertical misalignment and/or mechanical looseness. These impulsive noises can be annoying to nearby residents. It is often possible to control this noise by reducing or eliminating such misalignment and/or looseness or by installing overlapping “finger joints” which reduce tire impact forces on the joints by essentially eliminating transverse leading edges. Wakefield Acoustics Ltd. has recently been involved in assessing and controlling noise from two types of expansion joints which produce a totally different sort of noise - one that results from the excitation of resonances inside the cavities temporarily created between the rolling tires and the gaps between transverse joint elements. The two bridges employ quite difference types of expansion joint. The first utilizes a modular expansion joint consisting of series of transverse “lamella” beams (I-beams) with v-shaped rubber seals between them, while the joints on the second bridge featured corrugated, “saw-tooth” like surfaces (expansion mats) constructed from rubber-encased steel strips. In both cases a similar noise is produced which has been variously described as sounding like a “croaking frog” or a “zipper”. The characteristic frequency of this croaking sound was not found to be directly related to vehicle speed as would be expected if the noise was caused simply by the sequential impact of tires on the transverse joint elements. In both situation cases, these unfamiliar intermittent noises have resulted in ongoing complaints. A series of field tests have been carried out to explore the mechanisms behind these noises and to evaluate potential control measures. The outcomes of these field trials will be discussed as will the Helmholtz resonator analogy believed to explain how these noises are created and may be controlled.

The many sound cartography techniques that exist all face two main challenges when used in the field which are high installation cost and large equipments not suitable for field interventions. Therefore, these large and expensive systems are more effective for use in laboratories. Based on the sweeping method with the intensity probe (which is an approved technique for measuring sound power) a new approach for 2D cartography was developed using machine vision as a tracking device. A high performance DSP is used for signal processing and a powerful measuring and post-processing software was developed to produce fast and accurate intensity mapping. The result is a powerful tool for diagnosis providing a graphical render for acoustical data with exceptional quality. This equipment proved its usefulness as much in laboratories where high precision is needed as in the field where fast diagnosis and ease of use are required.

A simple analytical model which can predict accurately the sonic boom pressure signature on the ground in the vertical plane below a supersonic aircraft flying at constant horizontal speed in the real atmosphere is presented. The proposed model employs geometrical acoustics to propagate the boom and combines a non linear treatment of its amplitude with a quasi-linear calculation of its distortion, supplemented by the "area rule" for shocks fitting; ray-tracing equations are derived to calculate the rays trajectory and the ray-tube area, considering the standard atmosphere. Although simple, this combined method allows a very efficient and accurate prediction of the boom propagation starting from a given pressure signal in the near-field and can be considered a useful tool for the aerodynamic design and multi-objective optimization of low-boom supersonic aircrafts via CFD methods. Comparisons with recent experimental data and previously published results obtained via well established sonic boom propagation codes are provided.

Air injection vacuum blowers are a widely used series of blower utilized in applications with the need of high vacuum level, over 15” Hg. These high pressure blowers can be utilized in applications including: wet and/or dry waste removal (industrial, municipalities, etc.); central vacuum systems; aeration systems; and pneumatic conveyors. Air injection blowers are a lobe rotary configuration that produce high vacuum pressures through using atmospheric air injection to the blower itself to reduce overheating of the gas medium and power absorbed by the blower. A significant complaint related to the use of high flow blowers is the noise. This paper details an investigation to: (1) Analyze, define, and quantify the problematic noise; (2) Engineer, design, and quantify noise control performance and effect of the noise control to reduce the overall noise; (3) Manufacture prototype noise control equipment; and (4) Install and conduct post-installation performance and noise measurements to validate the noise control equipment and reduced blower noise compared with the predicted performance

One dimensional models as well as pseudo two-dimensional models have been successively applied in the past to study the acoustical propagation in HVAC system ducts fitted with splitter-silencers. Even though comparisons between experiment and theoretical finite-element models of the above formulations have been shown to be reasonably accurate within an engineering perspective, the one-d and two-d models were limited since elbows and staggered splitters were not amenable with the above modeling techniques. COMSOL multi-physics provides a powerful three-dimensional application software to solve a variety of different passive silencer designs. As a first step, COMSOL was used in a two-d model mode to solve the splitter silencers whose results have been published in the literature from later 1980s and early 1990s. The results of the above exercise will be presented in this paper.

Roadside noise barriers are an effective way to reduce unwanted traffic noise reaching nearby areas. Making noise barriers sound absorptive will decrease reflections from a barrier and avoid any amplification that may occur between parallel barriers. The effects of absorptive surfaces on the performance of a noise barrier were studied in an anechoic chamber. A reflective stud wall was built on a plywood floor and cotton acoustic baffles were placed on the wall in different configurations. The insertion loss was measured for different baffle configurations at several source and receiver positions, and the total A-weighted IL was calculated using a typical traffic-noise spectrum. The reflective wall was found to provide 12-18 dB of attenuation, and the improvement in overall IL due to absorption was 1-2 dB. It was seen that placing the absorptive material on the top half of both sides of the wall produced the highest insertion loss. Receivers were also placed on the source side of the wall to determine the effect of the absorptive material on reflections from the barrier. Covering the full source side of the barrier with baffles decreased reflections by 1.5 dB.

It is well known that several source mechanisms contribute to low frequency sound emitted from wind turbines. The infra-sound regions, that is frequencies below 10 Hz is dominated by 'lift noise' associated with the virtually steady aerodynamic forces. At higher frequencies, mechanical noise is dominant. Boundary layer and trailing edge noise also contribute. Propagation through the non-uniform atmosphere, introduces another complexity. Multi-point real-time measurements have been conducted in an attempt to provide detailed quantitative description of various mechanisms. The sound generated by the blade-tower interaction, as well as the elusive 'swoosh' have been extracted from the overall sound signal. To this end, well-known, but seldom used signal processing techniques such as cross-correlations and signal averaging have been used with great success.

For various reasons, some HVAC installations fail to meet the design specifications. For the most part exceedances are found in the 100Hz to 300 Hz range. Space constraints often pose considerable obstacles to conventional noise mitigation. Less conventional means may offer a cost-effective solution. This paper describes the design and testing of a passive band-pass acoustic filter system, suitable for installation near supply and return air openings. The system is compact and tunable. Level reductions of the order of 12 dB can be readily achieved.

This paper considers approaches to computing the evidence (Z) in Bayesian inference problems for model selection in geoacoustic inversion. Bayes' theorem combines the likelihood function, model prior, and Z to form the posterior probability density (PPD). Z is difficult to compute for general problems and a common approach is to avoid its computation entirely by calculating an unnormalized estimate of the PPD which is sufficient for moment estimates. However, estimating the normalized PPD, including Z, allows for moment estimates as well as quantifying the likelihood of the model parameterization. This is commonly referred to as model selection and poses a natural way to quantifying the most appropriate model parameterization for a given data set (Bayesian razor). Several approaches for computing Z have been developed in the statistics community. Here, annealed importance sampling is applied to the geoacoustic inference problem. Annealed importance sampling follows an annealing approach and computes weighted averages along cooling trajectories. Both methods also give parameter estimates which are compared to Metropolis-Hastings results. [Work supported by the Office of Naval Research]

This paper applies a general trans-dimensional Bayesian approach to geoacoustic inversion. Trans-dimensional inverse problems are a generalization of fixed dimension inversion and include the number of model parameters as an unknown in the problem. A model is considered to be any particular choice of physical theory, its appropriate parameterization, and statistical representation for the data errors that are used to explain the physical system under examination. By including the dimension of the search space in the inversion, a joint posterior distribution is obtained that characterizes the state of knowledge about parameters, including effects due to limited knowledge about the parametrization of the underlying environment and error processes. The inversion is implemented using a reversible-jump Markov chain Monte Carlo algorithm and the environment is parametrized with a partition modeling approach. Data errors are assumed as unknown and addressed by including a data-error model in the inversion. Jumps between dimensions are addressed with a birth-death methodology that allows the algorithm to wander dimensions by adding or removing interfaces from the seabed partition while maintaining detailed balance of the Markov chain. The approach presented can generally be applied to geoacoustic inverse problems. Here, seabed reflection-coefficient data as function of frequency and angle are considered. [Work supported by the Office of Naval Research]

This paper describes simultaneous non-linear Bayesian inversion of ocean acoustic seabed scattering and reverberation measurements for marine sediment parameters, assuming both diffuse and specular acoustic reflection. Measurements of geoacoustic properties are necessary to model and understand acoustic propagation, reverberation, and scattering in the ocean, as needed for many sonar applications. Significantly sonar performance predictions rely on accurate measurements of bottom scattering strength. The geoacoustic attributes of the seabed also can be used to determine the geological properties more efficiently and over larger areas than direct measures (such as coring). These properties are important for geotechnical applications, such as laying pipelines or cables, and for geological applications, such as understanding sedimentation processes. Bayesian inversion techniques are the ideal choice for such inversions, as they allow for the rigorous treatment of data uncertainties, as well as the flexibility to incorporate varying and updatable sources of information. In addition, Bayesian information measures prevent over-fitting of data without creating arbitrary limits to the complexity of a problem. Bayesian inversion creates a numeric estimate of the posterior probability density, i.e., the probability that the parameters have a particular value given the data and prior information. The estimation of the posterior distribution is conducted using an efficient Markov Chain Monte Carlo method.

In large regions of the Pacific Ocean the ocean bottom is characterized by a thin layer of sediment over an elastic solid crust. The presence of the solid basement close to the sea floor generates additional losses that affect long range acoustic propagation. This paper describes geoacoustic inversion of reflection loss versus angle data at very low frequencies (~16 Hz) to obtain estimates of the compressional and shear wave velocities in uppermost oceanic crust, Layer 2A. The data were obtained in experiments using small explosive charges and a horizontal line array on the thinly-sedimented deep water sites in the Pacific. The measurements were made at sites at increasing distance from the ridge spreading centre to determine the effect of crustal age of the basalt on the seismic velocities. The inversion used a Bayesian approach based on a simple model of the ocean bottom consisting of a thin-layer of sediment over basalt to calculate the reflection loss for comparison with the measured data. The results for very low frequency data are sensitive to the thickness of the sediment layer, which increases with distance from the spreading ridge, resonances caused by interference of compressional and converted shear waves, and possibly Stoneley waves excited at the basalt interface. At the youngest sites, the compressional wave velocity of the solid crust increased by about 8% from 2500 to 2700 m/s over the age span of 1.5 million years (Ma) from the spreading centre. The reflection loss inversion also provides an estimate of the shear wave velocity of the crust. The basalt shear wave velocity increased by nearly a factor of two, from ~725-1320 m/s over the same age span of very young crustal ages. These results show a decreasing trend of Poisson’s ratio with age, from the relatively high value of ~ 0.41 near the spreading centre, for the young basalt. Seismic velocity continued to increase gradually with crustal age at older sites where the sediment cover was thicker.

This paper considers simultaneous localization of multiple acoustic sources when properties of the ocean environment (water column and seabed) are poorly known. A Bayesian formulation is developed in which the environmental parameters, noise variance, and locations and complex strengths (amplitudes and phases) of multiple sources are considered unknown random variables constrained by acoustic data and prior information. Two approaches are considered for estimating source locations and strengths. The first approach, referred to as focalization, maximizes the posterior probability density (PPD) over all parameters using an adaptive hybrid optimization. The second approach, referred to as marginalization, integrates the PPD to produce marginal probability distributions for source positions and strengths, which quantify localization uncertainties. In this approach, two-dimensional Gibbs sampling is applied to source ranges and depths, and Metropolis-Hastings sampling is applied in principal-component space for environmental parameters. In both approaches, closed-form maximum-likelihood expressions for source strengths and noise variance allow these parameters to be sampled implicitly rather than explicitly, reducing the dimensionality of the inversion. Examples are presented of both approaches applied to single- and multi-frequency localization of multiple sources in an uncertain shallow-water environment

This paper considers matched-field tracking and track prediction for a moving ocean acoustic source when properties of the environment (water column and seabed) are poorly known. The goal is not simply to estimate source locations, but to determine track uncertainty distributions, thereby quantifying the information content of the tracking process. A Bayesian formulation is applied in which the posterior probability density (PPD) is integrated over unknown environmental parameters to obtain a time-ordered sequence of joint marginal probability surfaces over source range and depth. Marginal PPDs are computed numerically using Metropolis-Hastings sampling over environmental parameters (rotated into principal components) and two-dimensional Gibbs sampling over source locations to provide an ensemble of track realizations drawn from the PPD. Applying a probabilistic model for source motion to each of these realizations produces a sequence of source range-depth probability distributions for future times. These predictions account for both the uncertainty of the source-motion model and the uncertainty in past source locations. The approach is illustrated using Mediterranean Sea data, and tracking information content is considered as a function of data quantity (number of time samples and frequencies processed), data quality (signal-to-noise ratio), and level of prior information on environmental parameters.

Shallow gassy sediments are found in marine environments all over the world. The contained gases may lead to physical sediment instabilities, increasing the likelihood of underwater landslides. Furthermore, trapped gas in sediment can be a hazard for blowouts during offshore drilling. Since the theory on gassy sediment implies a direct relationship between the geoacoustic properties and the frequency of resonance of the embedded gas bubbles, predicting the impact of the gas on acoustic properties requires accurate information about the size distribution and the concentration of the bubbles. Direct measurements of the gas and sediment properties are normally done using computerized-tomography (CT) scans of pressurized cores and acoustic sensing equipment embedded in the sediment. The main objective of this study was to estimate the values of the geoacoustic parameters of the deep central basin of St. Margaret’s Bay, using a mobile underwater acoustic source (signal’s below 500 Hz), through an inversion scheme that does not rely on prior knowledge of the sediment type and properties. Inversion of simulated data sets showed that with a minimum level of knowledge about the environment, the method can distinguish between a gassy and non-gassy seabed, as well as reasonably estimate the geoacoustic properties of the gassy sediment layer. The inversion of the field data showed that the Bayesian inversion method can provide a first appraisal of the geoacoustic properties of gassy sediments, without the use of expensive pressure cores or in situ measurements.

If we were to look for music in other species what would we look for? What would we measure? Since first traveling to Alaska and hearing the Humpback Whale's feeding call in 1996 I have often wondered if these animals have a root sense of music. The variations within their calls and the interplay of notes and pulses seemed too musical to my ear to discount this possibility. When I later traveled to their mating grounds and heard the ever-evolving winter song, my curiosity was further piqued. Was this animal musical? As a composer and musician it is my job to bring human songs into being. I have spent the last 15 years analyzing my own creative process to better understand music and music-making in my species. Why do I choose certain notes over others? Why and when do I change the patterns I’m playing? What drives me to create? Human music is much more than a sum of its parts – the individual notes, phrases or themes. Somehow they combine to loft the resulting mix into the realm of music and away we go - our bodies and minds get wrapped up in it and we are compelled to dance and sing. If we are open to the potential of another creature having music we need to better understand the essence of music, not just the aesthetics. How is a song brought into being? Why is it brought into being? Why do composers compose? Why do I compose?

This paper reports on a lingual ultrasound study of different articulations used in playing the didgeridoo, and of the resulting acoustic effects of these articulations. The didgeridoo is a very simply-designed musical instrument; it is essentially a straight tube with neither holes nor keys of any kind. To make up for the structural (and hence acoustic) simplicity of the instrument, proficient players use a wide range of complex tongue articulations to achieve the sounds typical of the didgeridoo. For the uninitiated, it is difficult to understand these articulations because they are not visible, being inside the mouth. This study elucidates the role of the tongue in producing the sounds of the didgeridoo, using lingual ultrasound. A single player was instructed to play the didgeridoo, using five typical tonguing techniques, as well as circular breathing – a method that involves simultaneous playing and breathing, allowing for an uninterrupted sound to be produced. While he played, his tongue articulations were tracked and recorded for future analysis via the ultrasound transducer, which was fixed under his chin. Tongue articulations were analyzed in terms of target tongue position and movement, and related to their resulting acoustic signal. Results show systematic tongue articulations associated with each tonguing technique, which were termed a) undulation, b) piston, c) velar ejective, d) alveolar ejective, e) alveolar trill, and f) circular breathing. Results of the study provide an excellent visual teaching tool for those learning to play the didgeridoo and/or wanting to understand how its sounds are produced.

We assessed tonal perception in low-frequency harmonic tones that varied the extent of interaction between partials within critical bands. Hol(e)y tones minimize the likelihood of interaction by incorporating holes (missing harmonics) in the spectrum (Mathews, 1999). Unhol(e)y tones have fewer holes, leading to more interaction. Tonal perception was assessed by the probe-tone method. A listener is presented with a key-defining context followed by a probe tone drawn from the 12-tone chromatic scale. For each probe tone, the listener rates the goodness-of-fit with the preceding context. The correlation between the set of 12 ratings, called the probe-tone profile, and a standardized profile (Krumhansl & Shepard, 1979) is referred to as the "recovery score" (Russo, Cuddy, Galembo & Thompson, 2007). Our previous work (e.g., Cuddy, Russo & Galembo, 2007) examined recovery for synthesized harmonic and nonharmonic piano tones throughout the tessitura and identified critical roles for both temporal and spectral processes. Based on this work, we made three predictions for the current experiment: (1) recovery scores should be superior for unholy tones due to interaction between components; (2) consistent with temporal models, recovery scores for unholy tones should decrease with increasing frequency due to components falling beyond the upper limit of phase locking; and (3) consistent with spectral models, recovery scores for holy tones should increase with increasing frequency due to the increased spacing between harmonics. Results fully supported these predictions and will be discussed in the context of pitch processing models.

The overwhelming presence of electroacoustically reproduced sound within the modern soundscape has altered the relationship between listeners and their acoustic environment. With electroacoustic sound signals having become commonplace, the soundscape articulates not only the acoustic byproduct of economic globalization, but its character has also cultivated the practice of perceptual indifference. Consequently, aural desensitization and habituation typify modern listening habits, whereby the differences between acoustic and electroacoustic signals are seldom considered or even noticed. The concern then lies in first, the historical, cultural, and functional significance of localized forms of soundmaking, and second, in how to approach negotiating these differences amidst the cacophony of the modern soundscape. By employing the discourse of soundscape studies, in particular cross-cultural approaches to acoustic communication, a starting point emerges from which aural reintegration can be negotiated. In this setting, the engaged aural correspondent can positively employ personal perceptual sensitivities, listening habits, and one’s own unique history with sound, in order to renegotiate acoustic space with the ability of soundscape competence.

This paper examines the relationships between memory, the senses and movement from the viewpoint of embodied sound cognition and the practice of soundwalking. It offers a link between such interdisciplinary fields—soundscape studies, anthropology, psychology, education, health and medicine, and gerontology—in the understanding of both individual and collective remembering. I ask: How are memories created and remembered in the mind and felt within the body? What happens to our perception of self, home, and knowing as we move through spaces and places of significance? I aim to explore the subject of memory and movement within the context of soundscape studies; these notions require an understanding of how we “hear” the past and re‐evoke our acoustic memories as we move and act through our environment. Traditional methods for recalling the past involve mainly visual cues and focus on materiality (Bäckman, Small, Wahlin & Larsson 2000)—we look to photographs and hold personal objects, etc.—while remaining visual‐centered and localized. I suggest that it is the physical act of moving our body through meaningful environments that unifies the senses, places and knowing and that brings together the local past into the present experience (Casey 1987). This paper explores how the production of memory and act of remembering are evoked during the process of memory walks (or soundwalks) as a way of understanding and engaging with the world.

Audio technologies offer affordances for forms of representation in musical practices unavailable before their advent. The recording ‘chain,’ from microphone to storage medium to amplified reproduction allows the incorporation of ‘real-world’ sounds into musical, or organized sound, contexts. These electroacoustic technologies frame environmental sounds in seemingly neutral or naturalized ways, although they are by no means neutral. It is important to remember that environmental sound recordings are transduced, mediated representations of environmental sound(s), and that both human and technological agency are always implicated in these processes of mimesis. The temporal and spatial displacement of environmental sound recordings allows the creation of “…surrogate environments…” (Truax, Acoustic Communication) both in terms of our mediated everyday soundscape, as well as in the potentialities afforded to composers. Mimetic electroacoustic practices will be considered in terms of embodied forms of transduction and prosthesis. The environment will be considered as a form of music, and music, as a kind of niche building, will be considered as a form of environment made possible through audio technologies. Finally, both listening and sound-making will be considered as technologies of the self and forms of awareness practice affording us a more nuanced understanding of our world through the soundscape.

Although humanistic and artistic approaches to soundscapes have flourished over the last half-century and particularly in Canada, there is very little that is known about the psychology of urban soundscapes. To this end, we have recently captured over 200 soundscapes from the metropolitan Toronto area. Each soundscape consists of a 2-minute binaural recording with accompanying sound-level measurements. The recordings will serve as stimuli for a new program of research that aims to generate knowledge about the psychology of the urban soundscape. Progress on three of the initial studies in this program will be described. Study 1 is a multidimensional scaling study in which participants will be asked to rate the similarity of soundscapes presented in pairs. The scaling solution will provide insight into the underlying cognitive representation of urban soundscapes. Study 2 combines electrophysiological and behavioral methods to examine stress response and recovery to/from commonly experienced soundscapes that have been described as aversive. Study 3 examines the influence of soundscapes on the useful field of view. In all studies, soundscapes will be presented over headphones in a double-walled IAC chamber at levels that are calibrated with the accompanying SPL measurements. We expect that physical and psychophysical dimensions that are independent of sound intensity will influence the various questions under investigation. In addition to the planned experimental work, we will report on the development of a website (torontosoundmap.com) that provides map-based navigation of the soundscapes and public dissemination of the research findings.

A Raven croaks. In the course of the research project "The Sounding" museum I went on a field trip to Alert Bay, BC in October 2009 to make recordings for a cultural soundscape production to be pre-sented at the Sound Chamber of the Northamerica Native Museum in Zurich, Switzerland. For two weeks I had the honour of being a guest of the Namgis, one of the 17 tribes of the Kwakwaka'wakw, and I could bring back over 35 hours of recordings that I condensed into a surround composition depicting my impressions from that journey into the contemporary life of this Pacific Northwest Coast Native Community. In May 2010 I returned to Alert Bay for a presentation of my work. I had promised not to put anything I had taken with me on display without the explicit approval of the people that had agreed to let me take a peek into their world. The collaboration with the U'mista Cultural Cen-tre of Alert Bay proved fruitful; my work may now be presented to public audiences. On the theoretical level I deal with the representation of the Other, performances of memory, and cultural comparison by means of soundscape implementation in ethnographic museums. The project "Sound Chamber - Acoustic Worlds of the Indigenous People of North America" has been approved by the Swiss UNESCO commission as a contribution to the International Year for the Rapprochement of Cultures. It's scientific / artistic implementation is conducted by the Sounding Museum. word count: 244 A short version of the composition and the story how Raven stole the sun will be part of the presentation.

Mobile technologies and the web - particularly social networking - have in the last several years "democratized" the collection and sharing of soundscape recordings. While audio recorders are now a standard feature in many personal and mobile computing devices, social networking sites like facebook, twitter, and audio-specific sharing sites such as soundcloud, audioboo and woices, not only make it possible, but encourage the "mapping" of soundscapes in relation to geographic locations, images of the surrounding and built environment, as well as sharing all of these mappings with a larger online community - thus building a "world soundscape project" of a different kind. For owners of smartphones there are now accessible and reasonably accurate sound level meters and related real-time analysis applications. In this work I will trace some of the available tools for soundscape analysis, mapping, recording and sharing, and present the beginnings of a study that aims at better understanding listening practices and the role of urban soundscapes in people's lives through mobilizing these tools in the purpouseful mapping of the soundscape and reflection of sounds and listening.

A test of the effectiveness of voice acoustic parameters in distinguishing speakers with stroke-related dysarthria from controls could form the basis of a larger investigation. This study reported acoustic analysis of sustained vowel phonation on discrepancy across trials and group comparisons. The results indicates that (1) the stroke group exhibited significantly larger discrepancies in certain Multi-Dimensional Voice Program (MDVP) parameters, such as fundamental frequency and its standard deviation (f0 and STD), very long-term f0 variation (vf0), very short-term variability of the peak-to-peak amplitude (ShdB and Shim), and noise-to-harmonic ratio (NHR); and (2) stroke effects on acoustic voice parameters are mainly on the standard deviation and range of f0 (STD and PFR), very long-term f0 variation (vf0), and long-term peak-to-peak amplitude variation (APQ and sAPQ).

A pervasive assumption in speech motor behavior is that units of speech production correspond to preplanned motor routines. In a possible counterexample, Browman and Goldstein (1986) showed that, kinematically, a single glottal opening event can span two adjacent segments in s + stop clusters. However, because the segments are adjacent, it has not been clear whether this is a case of preplanned motor efficiency or simply a local concatenation of two separate motor events. Here, we demonstrate that the two movements in a heterosyllabic sequence of English flap allophones correspond to a single motor event. Previous work (Gick 2006) found an unexplained preference for up-down movement pattern in sequences of flap allophones. We propose that the first movement, an initial "up-flap" into a retroflex 'r' is controlled, while gravity and tongue elasticity provide the second "down-flap" upon tongue muscle relaxation. This sequence is reproduced using ArtiSynth, a 3D computer simulation of the human vocal tract. The results demonstrate disparity between phonology and motor behavior in that one motor event can encompass the production of two segments spanning a syllable boundary. Browman, C. P. and Goldstein, L. (1986). Towards an articulatory phonology. Phonology Yearbook, 3:219-252. Gick, B. (2006). Errors and strategy shifts in speech production indicate multiple levels of representation. JASA 119(5), Pt.2: 3302.

Japanese coda nasal (henceforth N) is susceptible to coarticulation with a following consonant (Vance 1986, Nakajima 2003), and has often been called "placeless" (Ito 1986, Trigo 1988). One x-ray study, Uemura (1990), describes prepausal N as having a general dorsal place. However, previous articulatory studies of Japanese N have been inconclusive, being limited to a single subject and lacking quantitative measurement. A lingual ultrasound study was conducted to test whether N shows evidence of a stable place of articulation. The constriction location and degree were recorded, comparing intervocalic N with (a) flanking /a/ vowels and (b) the velar consonant /k/ and the guttural consonant /h/. Seven native speakers of standard Japanese participated in the experiment. 10-12 examples of each token (e.g, aNa, ahha, akka) were used for the current study. Midsagittal tongue contours were produced using EdgeTrak software (speech.umaryland.edu/software.html). Results showed well-defined but highly idiosyncratic place targets for N. For each participant, N showed: (a) a constriction degree greater than that of surrounding /a/ vowels, and (b) a tongue constriction location as tightly constrained as that measured for other consonants. Unexpectedly, however, the locations of these tightly constrained constrictions for N varied dramatically across speakers. Places of articulation for N ranged from palatal (significantly anterior to /k/) to velar (identical to /k/) to postvelar (significantly between /k/ and /h/) to uvular/upper pharyngeal (identical to /h/). This high degree of idiosyncrasy in place of articulation accounts for the variability observed across previous single-subject studies.

This experiment examines whether context effects can be induced by auditory imagery. How a sound is perceived often depends on what was heard immediately prior – the 'context' of the sound. One well-known context effect was reported by Mann (1980) who found that a sound which is ambiguous between /da/ and /ga/ tends to be categorized as /da/ if immediately preceded by /ar/, but as /ga/ if immediately preceded by /al/. The reason for this effect is believed to be that the low F3 of /ar/ makes the F3 of the following ambiguous sound seem higher (so more /d/-like) by comparison. The high F3 of /al/ has the opposite effect, making the following F3 seem lower and so more /g/-like. The current experiment tests whether auditory imagery of the context sound can induce the same effect; if so, it would suggest that auditory imagery has detailed phonetic content (including formant structure). The experiment compares the strength of the effect across 2 conditions: normal speech and imaged speech. In both conditions subjects produce (externally or internally) one of the context sounds (/ar/ or /al/) in a rhythm, and after several repetitions, a /da/~/ga/ ambiguous target sound is played which they must categorize. The experiment is currently being run (8 participants so far) and initial results suggest that imagery does indeed induce the context effect. As not all subjects experience this context-effect, it is interesting to note that susceptibility to the effect seems to be strongly correlated across the two conditions.

Gick & Derrick (Nature, 2009) found that during auditory speech perception, perceivers integrated tactile information in the form of light air puffs delivered cutaneously on the hand or neck. The present study investigates whether visual evidence of speech aspiration can influence speech perception in the same way that tactile information does. Participants were shown short video clips of a speaker producing the sequence "pom" and "bomb" in a noisy bar setting with multi-talker babble. In some tokens, a candle was visibly perturbed by the aspiration from "pa", while in others the candle was either not perturbed, or was not directly in front of the speaker's mouth, and so visually not in the line of the expected airstream. We predicted that tokens with an accompanying visible perturbation of the flame will be more likely to be perceived as the aspirated "pa," where ones without perturbation will be perceived as "ba." Preliminary results indicate that subjects are more likely to correctly identify "pa" in the presence of visible perturbation of the flame, suggesting that ambient information about speech is integrated in perception even when that information is presented via a modality seldom available in the perceiver's prior experience.

Implicit accommodation studies have shown that subjects shift their speech in the direction of an interlocutor (Goldinger, 1998; Namy, 2002; Nielsen, 2008; Babel, 2009). This phenomenon is frequently interpreted as support for exemplar-based theories, and may provide insight to the basic processes inherent in language acquisition. Research has not yet, however, distinguished the relative importance of particular acoustic features within accommodation (e.g. preferential accommodation to VOT vs. vowel quality); furthermore, results from experiments designed to discover the effect of interlocutor characteristics, e.g. gender, that influence degree of accommodation have proven inconclusive (Namy et al, 2002; Pardo, 2006). The present study seeks answers to both these questions. The experiment involves a blocked-shadowing task, consisting of two production phases and one exposure phase. Stimuli were adapted from the Nielsen study (2008), and include /p-/ and /k-/ initial tokens. Words were also selected based on vowel quality (/ɪ/, /u/, /a/ and /ae/), and were controlled for lexical frequency. During the exposure phase, subjects hear two productions of the stimuli from both a male and female model talker. Half of participants are assigned to one condition, in which they are presented with a male voice whose initial VOTs have been extended by 40 msec; the other half are exposed to a modified female voice. Automatic imitation is tracked through differences in the subject’s VOT and vowel quality production from the initial to final production phases. Finally, participants are asked to complete a survey documenting their implicit and explicit stance toward each model talker’s voice.

Pitch is an important speech parameter in speaker recognition, speech synthesis, coding, and articulation training for the deaf. The pitch estimation task becomes very difficult when the speech observations are heavily corrupted by noise. Most of the methods proposed in the literature are capable of estimating pitch from clean speech. In this paper, residual and temporal information of speech are utilized for pitch estimation in a noisy environment. For a voiced speech, the major excitation of the vocal tract within a pitch period occurs at the instant of glottal closure (GC). Pitch period can be determined by careful analysis of the speech signal with the help of GC instants. Some characteristics of the GC instants can be better observed in a residual signal (RS) of speech in comparison to the speech signal itself. In order to overcome the bipolar fluctuations of RS around the GC instants, we derive a Hilbert envelope (HE) of the RS, which presents a unipolar nature at the GC instants. Under a severe noisy condition, the time difference of successive peaks of the HE of the RS may not provide an accurate estimate of the true pitch period. Hence, we propose a circular average magnitude sum function (CAMSF) of the HE that exhibits more prominent peaks even in a heavily degraded condition. Simulation results testify that the global maximization of the temporal function, CAMSF, yields an accurate pitch estimate compared to the state-of-the-art methods in white or multi-talker babble noise under low levels of SNR.

The present study examined how listeners use one prosodic parameter, pitch, and its role in emotional processing. All listening studies (Expts 1-3) had a 2 (Pitch Height: High vs. Low) x 3 (Pitch Steps: 1 vs. 2 vs. 3) design and used the same set of sentence materials, which was constructed in the following way. F0 values for every .01 second were extracted from each of the 120 neutral sentences recorded in the production study. Each sentence was then systematically manipulated using Praat (Boersma & Weenink 2009) so that High conditions had enlarged pitch height and Low conditions had compressed pitch height. To create Pitch Steps, 5Hz was added to or subtracted from the manipulated F0 values for each of the High and Low conditions, respectively. In Expt 1 participants indicated for each sentence which emotion, happy or sad, was being portrayed. In Expt 2 participants were asked to choose which emotion best fit with the sentences they heard from seven basic emotion categories: happy, surprise, fear, neutral, disgust, angry, and sad. Expt 3 tested listeners’ perception of subtle differences in pitch height by asking them to rate the stimuli on a 1-7 scale. Overall the present study supports the view that high pitch elicits happy emotion. Listeners are sensitive to an extreme difference in pitch height (for both High and Low) (Expt 3), while an extreme condition in one direction (Low 3 vs. High 3) leads to a difference in listeners’ choice of emotion (Expt 1 vs. 2).

The labial-velar alternation affecting /f/ and /b/ is a common feature of widely dispersed Spanish dialects. The bilabial fricative [f] alternates with its velar counterpart [x] when preceding [u] and the diphthongs [we] [wi]. For instance, a fuego ‘fire’ is realized as [x]uego instead of [f]uego. Likewise, the approximant [β] alternates with its velar counterpart [ɣ] when preceding [u] and the diphthongs [we] [wi]. Thus, a word such as burro ‘donkey’ is realized as [ɣ]urro and abuelo ‘grandfather’ as a[ɣ]uelo by children and speakers with low education. This phenomenon is intriguing, since it occurs cross-dialectally and cross-linguistically. For instance, there is diachronic evidence of English labial-velar alternation in words such as ‘enough’ which were originally realized with final [x]. The purpose of the present work is to investigate the motivations of labial-velar alternation. I hypothesize that this phenomenon is triggered by the perceptual similarity between [f] - [x] and [β] - [ɣ], which arises in the context of round vowels and diphthongs. As stated by Ohala (1989, 1993) similarity causes confusion and confusion can lead to synchronic variation and eventual sound change. I argue that the propagation of this variation is blocked by literacy, since orthography can help to resist perceptually driven variation. To test this hypothesis, twenty two native speakers (15 literate and 7 illiterate) of Argentine Spanish participated in sociolinguistic interviews and perception experiments. Results showed consistency in perception and production: speakers with lower literacy levels have more discrimination errors in perception and more labial-velar alternation in production

The project to be reported on aims at studying boundary signaling in languages with contrastive quantity. In particular, the role of duration is being examined, for segmental quantity tends to put a constraint on the degree of durational increase in boundary positions. The objective of the cross-linguistic project is (i) to quantify the degree of constraint on the increase of duration in relation to the prosodic system of languages where duration has a linguistic function, (ii) to explore the possibility of predicting the characteristic suprasegmentals compensating for the lesser degree of durational increase, and (iii) to identify the relationship between durational patterns driven by prosodic factors and durational changes relating to preboundary lengthening. The experiment consisted of recordings of six discourse units from 10 languages, five with contrastive quantity and five languages with non-contrastive quantity. The acoustic analysis of the material confirmed the relevance of preboundary lengthening in languages with contrastive quantity despite the linguistic function of duration. There is a constraint however on the durational variation associated with preboundary lengthening in those languages, evidenced by (i) a lesser degree of durational increase in comparison with languages with non-contrastive quantity, and (ii) in addition to the durational increase, an obligatory presence of another suprasegmental appears to have a contributing effect to the signaling of grammatical boundaries. It will be further argued that durational variations within a grammatical unit driven by prosody are separate from preboundary lengthening.

This study investigated the extent to which listeners store in memory the acoustic cues to non-native phonetic contrasts. Native English listeners were trained to identify the voices of Thai speakers, from a series of individual words, in three separate training conditions. In one training condition, the words produced by each voice consistently bore one of Thai’s five distinctive lexical tones. In a second training condition, each voice was consistently associated with one of Thai’s three distinctive VOT categories. After three days of training, listeners were then presented with words in a generalization test, in which the previous associations between words and phonetic properties no longer held. In a third, control condition, the voices were not consistently associated with any particular phonetic property. Evidence from both training and generalization indicated that listeners used both tone and VOT properties to learn to identify voices, including particular VOT values and tone contours that are not meaningfully contrastive in English. Talker identification accuracy in both the tone and VOT training conditions was significantly better than in the control condition; talker identification accuracy also decreased significantly in generalization testing for the tone and VOT listeners, but not for listeners in the control condition. Since listeners showed greater perceptual dependence on tone-talker associations than on VOT-talker associations, listeners may be more sensitive to longer, prosodic cues than to shorter timing cues in speech. Overall, these results indicate that listeners do store in memory low-level phonetic details, including acoustic distinctions they might normally perceive categorically.

Lotto and Kluender (1998), following Mann(1980) studied the perception of stops in the four syllables like /arga, alga, arda, alga/. These experiments and several others used a /-ga/ to /-da/ varying along an F3-transition continuum. All experiments reported finding more /d/ responses following /ar-/ than /al-/ precursor syllables. We replicated part of Lotto and Kluender's Experiment 2 in preparation for more elaborate experiments manipulating /r/ and /l/ as well as /d/ and /g/. Preliminary findings with 16 listeners (native speakers of English) show effects broadly similar to those previously reported, with /al-/ precursors leading to more /d/ responses. However, mixed-effects logistic regression analysis (Laplace approximation) suggests that more than simple boundary shifts are involved. For stimuli in the /al-/ context, the estimated /g-d/ response curve shows a steeper slope than that in the /ar-/ context. The nature of this interaction may have consequences for evaluating competing perceptual accounts (auditory contrasts vs. compensation for coarticulation) of this phenomenon. Simple logistic regression analysis of individual listeners data revealed that only eight of 16 showed significant (p<.05) effects of /ar-/ context in the expected direction, while three showed significant effects in the wrong direction and 5 showed non significant effects. (All but one listener showed significant main effects in the expected direction for the d to g continuum.) Further analysis of these results and results from listeners who are non-native speakers of English will be reported.

We have conducted a series of experiments involving a seven-step vowel continuum along a line in an F1 F2 space. F0 and F3 (with higher formants) were varied independently (three steps each) in ways that affected apparent speaker characteristics and vowel quality. In an initial experiment [S. Barreda & T.M. Nearey, J. Acoust. Soc. Am. 127, 2019, (2009)], we asked participants to report the gender and size of the speaker. We found that these judgments are strongly correlated with simultaneously reported vowel quality independently of the effects of the physical properties of the sound. In this paper, we report on a second experiment using the same vowel stimuli and response interface but with a change in the instructions given to participants. In this experiment we told participants that all of the voices were male whereas in the previous experiment they had been told that the voices might be males or females. The change in instructions resulted in a dramatic weakening of the effect of F3 on vowel quality. Possible explanations for this result will be discussed.

This paper reports on a preliminary ultrasound study investigating possible covert contrasts in first language acquisition. Covert contrasts (Scobbie et al., 2000) refer to cases in which a speaker distinguishes two speech sounds articulatorily, but does so in a way that is not perceptually salient enough for the sounds to be distinguished by listeners. This study focuses on one normally developing 2.5 year old girl – AC – whose pronunciation of sequences of /s/ + labial consonant (e.g. /sp/) are not auditorily distinct from her pronunciation of /f/. For example ‘spoon’ sounds like ‘foon’. The goal of the study is to determine whether or not /s/+labial consonant sequences are indeed the same as /f/ articulatorily. To this end, AC was recorded using lingual ultrasound pronouncing word sets like sponge – sun – fun, so see how /sp/ (or /sw/ or /sm/ in other word sets) compared to /s/ vs. /f/. Specifically, ultrasound videos were analyzed for presence vs. absence of tongue tip raising during the initial consonant(s). Preliminary results show no evidence of a covert contrast; rather /s/+labial consonant sequences have exactly the same articulatory (and acoustic) properties as /f/. Given that /f/ does not represent a compromise on any one phonetic plane between /s/ and labial consonants, it is proposed that the observed mergers (/sp ~ sw ~ sm/ > [f]) are phonological rather than simply phonetic: to ease articulation, AC is merging the manner of articulation of /s/ (fricative) and the place of articulation of labial consonants.

Acoustic Diagnostics of Prosodic Phrasing in SENĆOŦEN (Saanich, North Straits Salish). It is widely accepted that languages organize grammatical information into prosodic units (e.g. Selkirk 1986,) and that evidence for prosodic structure should be reflected in the acoustic signal (e.g. Shattnuck-Hufnagel and Turk 1995). Building from previous work investigating the acoustic correlates associated with Salish prosodic structure, this paper seeks to examine the relationship between the acoustic signal and prosodic structure in SENĆOŦEN. In particular, a set of acoustic correlates associated with the phonological phrase are determined (see Beck 1999, Koch 2008) by examining the acoustic properties of a corpus of SENĆOŦEN sentences elicited during fieldwork sessions with two fluent speakers. The results of this study illustrate that 1) pauses in pitch, along with variations in F0, coincide with the predicted boundaries between phonological phrases, 2) high F0 coincides with the predicted head of a phonological phrase and 3) longer vowel duration coincides with the end of a phonological phrase. This paper contributes to the growing body of literature investigating the nature of the relationship between phonology and phonetics in SENĆOŦEN. Beck, D. (1999). Words and Prosodic Phrasing in Lushootseed. In T. Alan Hall & Ursula Kleinhenz, (eds,). Studies on the Phonological Word (pp 23-46). Amsterdam: Benjamins. Koch, K. (2008). Intonation and Focus in Nlhe7kepmxcin (Thompson River Salish). UBC PhD Thesis Selkirk, E. (1986). On derived domains in sentence phonology. Phonology Yearbook, 3 371-405. Shattnuck-Hufnagel, S., and Turk, A. E. (1996). A prosody tutorial for investigators of sentence processing. Journal of Psycholinguistic Research, 25(2): 193-247.

Vocal affect is an essential part of human speech; it is not only central to understanding and expressing emotion, but also allows for interpretation of how a speaker is feeling. Although notably important, little research has been done examining the role of vocal affect in emotion-recognition tasks. The present study investigated the ability of persons to recognize emotion in vocal expressions in the absence of other cues (gesture, facial expression) and whether this ability was improved when presented with natural versus acted expressions of emotion. Natural expressions were taken from reality television shows and acted expressions were obtained from speakers recording their voice saying selected reality phrases. 7 speakers (3 M; 4 F) were audio-recorded while saying the same semantically neutral phrases taken from the reality television shows. 31 listeners were presented with randomized acted and natural phrases that were spoken in a happy, angry, or fearful tone of voice and asked to identify the emotion. Overall, our hypothesis that listeners would be more correct at identifying emotions that were natural expressions was only found in expressions of anger, while more accuracy was shown in happy and fearful expressions when they were acted. These results suggest that listeners are able to make use of the acoustical information when given affect information only. Given that natural expressions were only better identified for expressions of Anger, the natural physiological changes that underlie the acoustics associated with each emotion may not be necessary for listeners to identify a specific emotion category.

Native English speakers rely on F0, duration, and intensity in the perception of lexical stress. Second Language Acquisition studies have examined the use of these acoustic cues by non-native speakers in the perception of English lexical stress. A problem common to these SLA experiments was that words used for the perception tests were either in citation forms or excised from focused position in a sentence. The lexical stress on these words was, thus, confounded by phrasal accent, also cued by F0. Little is known about how well non-native speakers would discriminate lexical stress on words that bear no phrasal accent and how their performance would differ from native English speakers. In this experiment, Mandarin Chinese learners of English (CE) and native English speakers (NE) were compared in the perception of lexical stress on words excised from both ACCENTED and UNACCENTED conditions in an oddity test. Participants listened to a triad with three accented tokens or with three unaccented tokens, and decided whether the tokens in a triad all have the same stress pattern or not. An analysis of error rates showed no significant difference between CE and NE in the accented context but CE made significantly more mistakes than NE in the unaccented condition. CE¡¯s comparable performance with NE in perceiving lexical stress in accented contexts may be ascribed to their tonal background and sensitivity to F0, whereas their difficulty with unaccented words may be due to their insensitivity to duration and intensity.

While it is not surprising that phonetic realizations of /d/, a plosive, would feature a closure phase followed by a release burst, for realizations of /r/ to feature these is somewhat less intuitive. Weak-plosive realizations of /r/ are described in Kawakami’s (1977) Nihongo Onsei Gaisetsu as well as Vance’s (1987) An Introduction to Japanese Phonology. Outside of these works, /r/ in Japanese is typically described as an alveolar or post-alveolar tap subject to a wide degree of variation encompassing trills, lateral flaps, as well as lateral- and non-lateral approximants. Given this wide range of possibilities, it seems odd that weak plosive variants should co-occur with the voiced plosive /d/ insofar as both are phonetically similar. This paper examines /d/ and weak-plosive realizations of /r/ in a 30-minute conversation between two female speakers of (Kansai) Japanese from the JST/ATR ESP-C corpus. One speaker frequently realized /r/ as a weak plosive while the other realized /r/ more often as a tap or approximant. Assuming that within-speaker phonetic habits act to preserve phonological contrasts, the hypothesis tested is that the first speaker’s /d/s will have exaggerated durations to maintain acoustic distinctiveness from her weak-plosive /r/s, whose acoustic properties closely resemble those of /d/ but for their shorter duration and less intense release burst. Her interlocutor’s /d/s are predicted to have shorter durations, as this speaker used the weak-plosive /r/ less often and there is, therefore, less chance that a short /d/ and a weak-plosive /r/ would cause neutralization of the /d, r/ contrast.

According to Flege’s (1995) Speech Learning Model (SLM), inaccuracies in the production of second language (L2) sounds result from interactions with first language (L1) categories. Research supporting this claim often relies on native speaker evaluation of L2 productions. However, listener variables that might affect these judgments remain little understood. Levi, Winters and Pisoni (2007) claim that the lexical frequency of L2 speech tokens, and access to orthographic representations of the target items, influence listeners. Unfortunately, incommensurate measures used to investigate listener vis-à-vis speaker effects renders Levi et al.’s (2007) results inconclusive. The present study builds upon Thomson and Isaacs (2009), examining the impact of speaker and listener variables on English vowel intelligibility, comparing three speaker groups: L1 English, L1 Mandarin and L1 Slavic. Speakers were asked to repeat a word list comprising ten target English vowels, each embedded in three separate monosyllabic verbs and varying in terms of lexical familiarity for speakers, and lexical frequency for listeners. Speech recordings were obtained in two counter-balanced conditions: 1) after hearing an auditory prompt and 2) after hearing an auditory prompt accompanied by the written form of the word. In a third condition, speakers read the same words, with no auditory prompt. L1 English listeners identified the recorded vowels in two conditions that included lexical information, and one condition in which the vowel portions of the recorded words were presented in isolation. Results indicate an interaction between lexical familiarity, speaking prompt type and intelligibility scores. Furthermore, listening task type also affects intelligibility scores.

One long-deliberated question in speech processing is the extent to which it involves linguistic-specific neural mechanisms or reflects the processing of general acoustic properties, or how these functions are interconnected. The present study addresses this question by examining the neurophysiological processing of speech and non-speech tonal information by tone and non-tone language speakers. We hypothesize that, if tonal processing were linguistic-specific, neural sensitivity for tone in speech would be different for native and non-native tone language speakers, whereas processing for non-speech tonal information would be comparable. Using electroencephalography (EEG/ERP), native Mandarin Chinese (tone language) and English (non-tone language) adults were tested in two stimulus conditions: Mandarin tone word (speech) and hummed tone (non-speech), both involving contrastive rising and falling pitch contours. An oddball paradigm was employed for each condition where stimuli were presented binaurally to the participants. Initial results showed that, first, across groups, the hum condition elicited larger mismatch negativity (MMN) responses than the tone condition. Moreover, in both the tone and hum conditions, the MMNs were present for both Chinese and English listeners, but the magnitude was much smaller for the English group. These results reveal higher neural sensitivity in Chinese listeners in both speech and non-speech conditions in detecting a tonal pattern change. The results indicate that language experience may lead to automatic transfer of processing from speech to non-speech stimuli in the MMN responses to tonal pattern extraction for rising versus falling pitch information, suggesting a degree of interconnections of brain functions for speech and non-speech.

This study examines the production and perception of laryngeal constriction in the vocalizations of Bai and English infants in the first year of life. The study is grounded in Esling’s (2005) model of the vocal tract, which characterizes the laryngeal vocal tract as a separate articulator, distinct from the oral vocal tract. The study of Bai and English infants’ production identifies universal and language-specific patterns in infants’ development of laryngeal constriction. Bai and English infants produce mostly constricted sounds in the first months of life, and produce an increasing proportion of unconstricted vocalizations as the year progresses. However, by the end of the first year, the infants’ babbling begins to reflect the use of laryngeal constriction in their ambient language. The English infants’ babbling is mostly unconstricted, while the Bai infants’ babbling is mostly constricted or alternates between constricted and unconstricted voice qualities, reflecting the use of laryngeal voice quality in English and Bai, respectively. English and Bai adults’ perceptions of infant utterances mirror the patterns found in infants’ production. Adults from both language groups assign lower ratings to constricted non-babbled sounds than to unconstricted non-babbled sounds, which predominate earlier and later in development, respectively. However, in their ratings of babbling, Bai listeners show a preference for constricted sounds, while English adults prefer unconstricted sounds. These results suggest that adults are attuned to laryngeal voice quality in infants, and that, in turn, infants become attuned to the use of laryngeal voice quality features in their ambient language early in development.

The contrast between /f/ and /θ/ is notoriously difficult to differentiate acoustically and Ladefoged and Maddieson (1996: 173) attribute this difficulty to individual differences in production. Such a claim would predict that when presented with auditory tokens of /f/ and /θ/, listeners would exhibit improved performance when faced with the voice of a single talker. We report on a 2AFC experiment in which listeners were presented with the voices of ten talkers (5 = male) producing /f/ and /θ/ in CV, VC, and VCV contexts. Listeners were assigned to conditions with either mixed talker presentation or blocked by talker. An analysis of listener sensitivity showed that listeners were more sensitive to the contrast in the mixed talker block, contra the hypothesis above. However, reaction time analyses showed increased latencies in the mixed condition, as well as an effect of talker gender. The increased response latencies for the mixed talker condition replicates classic findings within the talker normalization literature (e.g., Nusbaum & Morin, 1992). However, the increase in listener sensitivity in the mixed talker condition counters the claim made by Ladefoged and Maddieson that f/θ production is wholly talker-specific. Listeners’ sensitivity to each talker’s f/θ contrast was improved when tokens were presented in mixed talker blocks, which seemed to enhance the acoustic difference.

Talkers have been found to spontaneously accommodate acoustically to a model talker in single word shadowing tasks (Babel, in press, 2009; Goldinger, 1998; Namy et al., 2002; Nielsen, 2008; Shockley et al., 2004) This behaviour has been termed implicit phonetic imitation, as talkers have no awareness of having modified their speech. In this study, we compare implicit phonetic imitation to explicit phonetic imitation, focusing our analysis on the first and second formant frequencies of vowels. Two groups of participants (n = 20) completed an auditory naming task which included 50 monosyllabic words with the vowels /i æ ɑ o u/ produced by a male model talker. With the exception of task instructions, the procedure was identical for both groups; one group, however, was given the instruction to explicitly imitate the model talker, while the other group was simply instructed to repeat the words naturally and clearly. Acoustic analyses of participants’ productions indicate that the qualitative nature of implicit and explicit imitation is similar. The crucial difference between tasks appears to be the amount of imitation. For example, talkers exhibit the same preference for imitating low vowels in explicit imitation as they do in implicit imitation (Babel, 2009), but the amount of imitation was greater across all vowels in the explicit task. This suggests that low-level implicit imitation and higher-level explicit imitation are susceptible to the same cognitive biases. An AXB perceptual task comparing listeners’ sensitivity to implicitly and explicitly imitated tokens is currently ongoing.

Our work examines the acoustic contributions of the epilaryngeal tube to the vocal tract as both a source and a filter. We present a range of empirical data, collected using laryngoscopy and videofluoroscopy, which demonstrates the oscillation of various parts of the tube in voiced sounds to produce a complex vocal source. From this empirical evidence, we demonstrate that aryepiglottic oscillations have a dynamic influence on the vocal folds by continually changing the acoustic loading from the vocal tract. Under conditions of glottal voicelessness, the aryepiglottic folds contribute a source that only weakly energizes the vocal tract due to the highly irregular oscillations of the aryepiglottic folds, potential for considerable air leakage through one or both aryepiglottic apertures, and turbulent losses. We propose that aryepiglottic trilling is a phonetic enhancement that constitutes a complementary acoustic cue to sounds that depend upon turbulent noise sources generated in the lower pharynx (namely pharyngeals/epiglottals). We also provide evidence that the epilaryngeal tube is an important resonator in the production of pharyngeals in general. Based on empirical observation, the epilaryngeal tube can be constricted to extremely narrow dimensions, which have not been adequately studied in previous research. To assess the acoustic contribution of the epilaryngeal tube to vocal tract resonance, we analyze audio recordings of careful phonetic performances involving the manipulation of the epilaryngeal tube volume during several canonical vowels. Our results indicate that constriction of the epilaryngeal tube primarily has an effect on the third formant, typically lowering and strengthening it at the same time.

In younger adults, different types and levels of acoustic distortions on sentence context have differing effects on semantic priming (facilitation vs. inhibition) of sentence-final target words (Pelletier, Goy, Coletta, Giroux and Pichora-Fuller, under review; Aydelott and Bates, 2004). In the Pelletier et al. (under review) study, the acoustical distortion used were low-pass filtering, time compression and masking by multi-talker babble. Each distortion type was tested at two levels, which were selected to yield percent correct scores of 50% and 76% on a word recognition test. Their results showed that compared to intact sentence contexts, the amount of facilitation was similar across the three distortion types. A small inhibition effect was present only in the time compression condition. Their results also revealed more facilitation when the contexts were less distorted. In the present study, we presented the same low-pass filtering and time compression stimuli to older adult listeners aged 65 and over with levels of distortion adjusted to match the degree of difficulty experienced by younger adults. As expected, older adults had longer lexical decision times compared to younger adults. However, unlike younger adults, reaction times of older adults did not differ between intact and distorted sentence contexts, but instead interacted with type of acoustic distortion and priming effect. These results show that older and younger adults differ in their responses to different types and levels of acoustic distortion, possibly due to age differences in the use of contextual information and greater difficulty with spectrally and temporally distorted information.

In this study, the properties of the scattered acoustic vector fields generated by simple rigid spheroids are investigated. Analytical solutions are derived from general acoustic pressure scattering models, and analyzed for wave numbers in the resonance region. Of particular interest is the understanding of the characteristics of energy flow of the scattered acoustic vector field in the near to far-field transition region. The separable active and reactive components of the acoustic intensity are used to investigate the structural features of the scattered field components. Numerical results are presented for the near and transition region for a rigid sphere. The ability to extract scattered field features is illustrated with measurements obtained from a recent in-air experiment using an anechoic chamber and acoustic vector sensor probes to measure the scattered acoustic vector field from rigid spheroids.

With the impact of noise from recreational activity being an ever increasing concern, the accurate measurement/classification of sound emissions from off-highway vehicles (OHVs) for the purposes of controlling sound levels at noise-sensitive off-track locations is becoming more and more critical. Conventionally, sound emission testing for this purpose is conducted according to SAE J1287 (or equivalent) which uses a stationary vehicle in neutral gear with the engine at approximately half-throttle. However, the poor correlation between results using this test method and measured sound levels at “near-track” and “off-track” locations is well documented. A new method has been proposed by the Fédération Internationale de Motocyclisme (FIM) which purports to provide improved correlation between the stationary vehicle sound emission level and “near-track”/”off-track” sound levels, and as such would be expected to be an improved method for qualifying or disqualifying a vehicle from operation in a given locale. A direct comparison of the measured sound emission levels as determined using the SAE method and the proposed FIM method are presented, along with comparisons with “near-track” and “off-track” measurements in order to assess the potential for improved correlation and thus, improved control of environmental noise levels.

At an increasing rate, new hospitals and renovations to existing hospitals are incorporating helicopter pads (helipad’s) into the design of their facility. Hospitals contain many sound and vibration sensitive spaces, which the operation of a helicopter may affect. The vibration generated from a helicopter landing has potential to impede on the operation of sensitive diagnostic equipment in the hospital and the noise generated by the helicopter turbines and propellers can disturb patients. This paper presents sound and vibration measurements of a Sikorsky S-76 medivac helicopter landing and taking off from St. Michael’s Hospital in Toronto Ontario and provides an analysis of architectural and structural design considerations to minimize the affect of helicopter operation at medical health centres.

Micro-acoustic tags have been used to monitor the fine-scale three-dimensional behavior and survival of fish and other aquatic life in the Pacific Northwest and Central California for years. Three-dimensional tracks are obtained at dams, lakes, open rivers, estuaries, and in marine environments. Resolution of three-dimensional positions are sub-meter with some resolutions as fine as 20 cm. In addition to juvenile and adult salmonids, other species tracked include eel, lamprey, sturgeon, shad, crab and shrimp. Salmonid smolts as small as 92 mm have been tagged and tracked with acoustic tags, which weigh as little as 0.5 g. Today’s acoustic tag technology allows researchers to view fish behavior and passage in real-time. Tags operated at 307 kHz with a user-specified pulse width of 0.5-5 msec. Recent innovations include the development of smaller tags weighing 0.5 g in air, longer life tags, remote access via smart-phone, as well as various data display options. A number of advances in the analysis methods, techniques and software have been made over the past several years. Some of these improvements include the development of various fish density algorithms and advances of three-dimensional animation programs. Three-dimensional tracks of fish approaching various structures will be presented. Examples of fish tracks and fish densities will be superimposed over bathymetry, water velocities and structures. Recent Sacramento-San Joaquin River examples will be featured.

Viscoelastic material damping and equivalent mass were compared as treatments to representative aircraft carbon fiber composite fuselage structure by themselves and as part of sound packages. The sound packages were made up of fiberglass or foam. The viscoelastic material damping used in this study is constraining layer damping and abbreviated as CLD. Equivalent mass is a layer of impervious screen that is positioned at different locations within foam or fiberglass layers. The acoustic parameters of ABIL (airborne insertion loss), radiation efficiency, TBLIL (turbulent boundary layer excitation insertion loss) and absorption coefficient were used to compare CLD to equivalent mass layer on the representative carbon composite fuselage structure. The equivalent mass produces a double wall effect and hence becomes an effective acoustic insulator as part of a sound package at mid to high frequencies. The drawback is the occurrence of the double wall resonance at lower frequencies which compromises the effectiveness of the equivalent mass layer. However, overall frequency averaged acoustic performance of equivalent mass is superior to CLD.

The main objective of the current research is to develop analytical models for the prediction of turbulent boundary-layer-induced noise in the interior of aircraft cylindrical cabins. The mathematical model represents the structural-acoustic coupled system, consisted by the aircraft cabin section coupled with the fuselage structure. The aircraft cabin section is modeled as a cylindrical acoustic enclosure, filled with air. The fuselage structure, excited by external random excitation or by turbulent flow, is represented through two different models: (1) as a whole circular cylindrical shell with simply supported end caps, and (2) as a set of individual simply supported open circular cylindrical shells. This paper presents the results obtained from the developed analytical framework, and its validation through the successful comparison with several experimental studies. The developed analytical model can be used to study a wide range of different systems involving a cabin coupled with vibrating panels, excited by the TBL. The properties of the external flow, acoustic cabin, and panels, as well as the number of vibrating panels, can be easily changed to represent different systems. These abilities of the model make it a solid basis for future investigations involving the implementation of noise reduction techniques and multidisciplinary design optimization analyzes.

Active sonar systems are used to detect underwater manmade objects of interest (targets) that are too quiet to be reliably detected with passive sonar. In coastal waters, the performance of active sonar is degraded by false alarms caused by echoes returned from geological seabed structures (clutter) found in these shallow regions. To reduce false alarms, a method of distinguishing target echoes from clutter echoes is required. Research has demonstrated that perceptual signal features similar to those employed in the human auditory system can be used to automatically discriminate between target and clutter echoes, thereby improving sonar performance by reducing the number of false alarms [J. Acoust. Soc. Am., 122(3), 1502-1517 (2007).]. An active sonar experiment on the Malta Plateau was conducted during the Clutter’07 sea trial and repeated during the Clutter’09 sea trial. Broadband sources were used to transmit linear FM sweeps (600-3400 Hz) and a cardioid towed-array was used as the receiver. The data set consists of over 30,000 pulse-compressed echoes returned from two acting targets and many geological clutter objects. These echoes are processed using an automatic classifier that quantifies the timbre of each echo using a number of perceptual signal features. Using echoes from 2007, the aural classifier is trained to establish a boundary between targets and clutter in the feature space. Temporal robustness is then investigated by testing the classifier on echoes from the 2009 experiment. The dependence of signal-to-noise-ratio on classification performance is also examined.

A shallow water reverberation model based on normal modes has been developed and refined at DRDC Atlantic over a number of years. Originally, the model handled range independent boundary reverberation in monostatic or bistatic geometries, including source and receiver beam patterns for comparison with measured data, and was later extended to handle target echo. A formulation [Ellis et al., in Proceedings of International Symposium on Underwater Reverberation and Clutter, Lerici, Italy, 2008] was developed to extend this model to range-dependent environments using adiabatic normal modes. The computations presented at that time used a Matlab/Fortran hybrid model with modes evaluated on a rectangular grid. While the scattering strength and echo at any point on the grid could be arbitrary, a constant water depth was still required. In 2010, a model (implemented in Fortran) has been developed to handle both sloping bathymetry, and towed array beam patterns in bistatic geometry. The model is computationally efficient and its capabilities are evolving. Model-data comparisons of several range-dependent problems from the ONR Reverberation Modeling Workshop, and of towed array clutter data obtained on the Malta Plateau, are underway.

A wave number integral based computer model was used to investigate the effectiveness of bubble curtains at reducing underwater sound pressure levels received to the sides of airgun arrays operated in the Chukchi Sea and Gulf of Mexico. The curtains were modelled as uniform layers 4.3 m in thickness, generated by air released from manifolds towed at 20 m depth and separated by 36 m. The model was applied to compute received sound levels as a function of receiver depth and distance from the airgun array along a path oriented perpendicular to the tow direction including interactions with the curtains. Source levels from the airguns inside the bubble curtain were modelled by using the method of images to account for reflected rays. The results showed that for both shallow and deep water, at all modelled ranges and receiver depths, mitigation using the bubble curtain consistently lowered the SEL, peak, and 90% rms sound pressure levels. The reduction in sound levels using the air curtains was at least 10 dB, and in some cases, up to 30 dB.

We consider a submerged fluid-filled elastic cylindrical shell subjected to two consecutive shock waves generated at the same source. Our focus is on the structure of the hydrodynamic field induced inside the shell by such loading, and we are particularly interested in the differences that the internal field exhibits in comparison to the better studied case of a single shock wave. The latter is known to produce the internal field of a rather complex structure, and it seemed reasonable to expect that the interaction with two consecutive shock waves would result in an even more interesting hydrodynamic pattern. The interaction was simulated numerically using the semi-analytical approach developed in our earlier work, and it was observed that the internal field in the present case indeed is dramatically different from its single-wave counterpart. In particular, it was observed that the hydrodynamic structures associated with the internal reflection induced by each individual shock wave superpose, producing a significantly more complex and phenomenologically diverse pattern. Although the results obtained are of interest on their own, a broader goal we aimed at was to facilitate the investigation of a more complex scenario, namely the interaction between a shock wave and a shell in the presence of reflective surfaces.

We consider a submerged fluid-filled cylindrical shell responding to an external acoustic pulse, and examine the changes that occur in the acoustic field when the system in structurally enhanced by means of placing a rigid co-axial inside the shell. It is known that the external field outside a fluid-filled shell has a considerably more complex structure than that outside its evacuated counterpart, and we demonstrate that this complexity is further increased, and significantly so, when the core is added. In particular, we show that all the internal waves resulting from the reflection off the core are transmitted into the external fluid, thus producing the external field of geometrically complex, multi-front structure. We also demonstrate that a particularly complex and interesting acoustic field is observed around a shell with the core that dominates the internal volume, in which case the shell starts to show some characteristics of a (relatively) high-frequency, quasi-steady source.

Evanescent liquid sound-pressure waves (i.e., standing waves, of limited spatial extension, with variable pressure and liquid-particle velocity but negligible density variation) appear to play a fairly important role in the human cochlea, e.g., in the generation process of spontaneous oto-acoustic emissions. These waves can be studied with the help of underwater resonators. Mainly because of the kinetic energy in the generated evanescent waves, a tuning fork submerged in water oscillates at a frequency below 440 Hz (typically at ~415 Hz, lower than 440 Hz by about a semitone). In the case of tapped drinking glasses, the corresponding frequency reduction is greater than an octave. In the present paper, idealized two-dimensional forms of the just mentioned resonators are treated. Analytic calculations of the lines of constant liquid sound-pressure amplitude and of the evanescent-wave streamlines (along which the liquid particles oscillate linearly) are described, and the predicted frequency reductions caused by submerging the resonators in water are compared with experimental results.

Radiated acoustic levels for ten cruise ships that frequently travel Southeast Alaska waters were measured at the U.S. Navy’s Southeast Alaska Acoustic Measurement Facility near Ketchikan, Alaska to quantify their underwater acoustic levels. This group of ships included diesel-electric, diesel-electric/gas turbine-electric, direct-diesel, and steam turbine propulsion plant ships ranging in size from 90 to 294 meters in length. Peak one-third octave levels for 10-knot ship speeds ranged from 158 to 172 decibels relative to 1 microPascal at 1 yard. Propulsion system type and cavitation performance were important factors in cruise ship acoustic level and spectral character. Diesel-electric ship acoustics were dominated by noise energy from diesel generators and from electric propulsion motors in combination with frequency converters and diesel generators. Propulsion diesel and reduction gear noise were important contributors in the direct diesel ships’ underwater acoustic characteristics. Turbine generator, propulsion turbine, and reduction gear noise were the most significant noise items for the steam plant ship. Each ship was tested at two speeds. The sound levels of some ships were strongly speed dependent while others exhibited less speed dependence. Differences in acoustic levels between speeds were typically dependent on propulsion system and propeller cavitation noise contributions.

Range-dependent model predictions of underwater acoustic propagation rely on a series of input sound speed profiles (SSPs) at different ranges. Models then interpolate between the input SSPs to calculate the SSP at each range step. A variety of interpolation schemes exist, not all of which are suitable for including measured sound speed profiles in propagation models. "Closely-spaced" SSPs measured at range intervals between 700 m and 2000 m along a straight-line track were obtained during two separate field trials off the southwest coast of Nova Scotia. In addition, acoustic transmission loss measurements were made along the same tracks. In order to determine which type of interpolation is appropriate under different circumstances, the measured SSPs were compared with profiles interpolated using a variety of interpolation schemes, including linear, triangular, and trapezoidal interpolation. The effects of the different interpolation methods on modelled range-dependent transmission loss were compared with the measured transmission loss obtained along the same tracks.

Interest in understanding atmospheric acoustic propagation over water has increased in recent years, driven primarily by concerns about noise from offshore wind farms. In addition, there is interest in evaluating directional acoustic hailing devices for use at sea, in determining potential environmental impact of naval gunfire exercises, and in understanding the in-air acoustic footprint of maritime-based military assets. Atmospheric acoustic propagation is strongly affected by the environment, including surface roughness and atmospheric parameter profiles (temperature, wind speed, humidity, turbulence). However published measurements and realistic modelling of the effect of environmental parameters on over-water acoustic propagation are sparse. We report on two experiments designed to measure atmospheric acoustic transmission loss over water. In both experiments, the transmission loss was measured as a function of range by using an acoustic source on a small boat and receivers on a larger vessel. Point measurements of atmospheric parameters and observations of sea state were made in the vicinity of both experiments. Additional data were available for the second experiment: directional wave spectra and atmospheric parameter profiles (measured radiosonde profiles and modelled profiles generated by Environment Canada’s Global Environmental Multiscale [GEM] model). For both experiments the measured values of transmission loss were compared to results obtained with an atmospheric acoustic propagation model developed by the US Army (Sensor Performance Evaluator for Battlefield Environments, or SPEBE).

Ambient noise has been a topic of study in underwater acoustics for over sixty years. In the absence of rain and biological noise sources, wind-generated noise is the dominant source of underwater ambient noise in the 1 kHz to 25 kHz band. The extent to which this noise can be estimated using remotely-sensed data is an area of interest, in particular for the purposes of Rapid Environmental Assessment. Here, synthetic aperture radar (SAR)-derived wind fields measured in littoral areas off the coast of Nova Scotia using RADARSAT-1 and RADARSAT-2 satellite imagery are compared with data measured using shipboard instruments and by the Gulf of Maine Ocean Observation System buoy ‘N’ in the Northeast Channel. Using an algorithm to predict underwater ambient noise generated due to wind, the noise predicted from satellite and other instruments is then compared with in situ measurements recorded on hydrophones.

Sonar performance prediction models are a key tool for planning and carrying out sonar operations in littoral environments. Less well understood are the impacts that variability and uncertainty in the environment may have on propagation predictions. A sensitivity model was previously developed [Dosso et al, J. Acoust. Soc. Am. 121 (1), 2007] to determine the effects of limited sampling of, and uncertainty or variability in, oceanographic and geo-acoustic information on acoustic propagation modeling. Here, the model is applied to the conditions found during two field trials off the coast of Nova Scotia, using environmental data collected both initially from historical databases and then in-situ. The effects of the improved in-situ environmental sampling are demonstrated using the results of the sensitivity analysis. The sensitivity of the propagation and uncertainties in modeled propagation loss are then compared to the propagation loss data collected during the field trials.

Performance prediction provides an important and essential link between the underwater acoustics research community and the ever-elusive customer, be it environmental or military. Nominally, performance prediction involves the use of an acoustics model to generate field predictions for a particular ocean environment given locations of particular sensors and targets; these data are then combined with some detection criteria and assumed (or computed) noise background to produce and display a measure of the performance of sensor against target. Currently, no single acoustics model can handle the wide range of frequencies implied by either military or environmental applications. The Java Acoustic Model Interface (JAMI) is under development with the intent of providing a framework for a Client-Server approach to performance prediction in which the problem configuration and display reside on a potentially thin Client and the models and other computational engines are centralized on a more powerful back end computer or Server. In JAMI, the Client is programmed exclusively in Java whereas the back end consists of a conglomerate of C and FORTRAN code designed to take advantage of as many existing and publically available codes as possible. An interface layer has been written that provides relatively seamless interaction of all three languages. This same interface also provides the links to public domain environmental databases for bathymetry (Gebco), sound speed via temperature and salinity (World Oceanographic Atlas, WOA), and bottom composition (Deck41). Some numerical results and displays obtained by applying the DRDC Atlantic clutter model in a realistic environment are presented.

Vibration shakers are used in primary vibration calibration. Vibration shakers generate sinusoidal accelerations in a main axis precisely defined, normally perpendicular to the mounting surface for accelerometers. Ideally, there should be no acceleration in other directions. In practice, other than transverse motion, tilting and tumbling cannot be avoided. Such movements are occurred under asymmetrical loading conditions primarily at higher frequencies. The accelerometer under test responses to these movements while the laser interferometer only senses the acceleration in the main axis. This becomes a contribution to the measurement uncertainty. For this reason, the international standard ISO 16063-11 requires to keep tilting sufficiently small to prevent excessive effects on the calibration results. This paper presents the development of a laser position sensing system for studying the tilting of vibration shakers. A dummy accelerometer was fabricated that integrated seven mirrors and an electrical connector. The dummy accelerometer has the same physical size of a commercial accelerometer and is made of solid steel to eliminate the rocking motion due to accelerometer housing. A laser beam is incident on one of the mirrors of the dummy accelerometer. The reflected laser beam is received by a lateral effect position sensitive detector (PSD) to measure the tilting. A two-channel waveform recorder is used to capture the PSD output signals and displays the tilting in real time. With this system adjustments can be done in real time to reduce the tilting. Examples of the adjustments for different acceleration amplitudes and frequencies are given in the paper.

The report is devoted to the development problems of the monitoring and diagnosis system of the industrial equipment. Modern systems of stationary type ordinary use some different autonomous subsystems of the equipment monitoring running in the synchronous mode. The structure and the set of these subsystems depend upon the functional purposes of the monitoring and diagnosis system’ using. Each of these subsystems use their own set of physical methods for the detection of the equipment characteristics. It leads to arising of the development problem of multi-parametrical monitoring and diagnosis methods. Using of the simultaneous monitoring of the same technical object by the different physical methods leads to the possibility of the essential generalization of the monitoring and diagnosis problems and to the formulation of the interesting set of general scientific problems. One of the main problems of this set is the problem of the non-uniform information processing. Data acquiring by the acquisition and saving subsystem is a non-uniform one both at the problem of the detection of the current state of the monitoring object and at the problem of the prediction for the dynamics of object characteristics. The approach to the processing of data based on the methodology of Group Methods of Data Handling (GMDH) and statistical data processing is proposed at this report. It makes possible to apply the methods of self-organization of the mathematical models to the problems of the monitoring of complex technical systems. Proposed method was realized as software and applied for the problems of monitoring of turbine engines.