bluuebunny/crossref_metadata_2025 huggingface dataset:
The effect of classroom capacity and size on vocal fatigue as quantified by the vocal fatigue index
Authors: Banks Russell E. , Bottalico Pasquale , Hunter Eric J.
Published: June 2017

Previous research has concluded that teachers are at higher than normal risk for voice issues that can cause occupational limitations. While some risk factors have been identified, there are still many unknowns. To gain more understanding regarding some of these unknowns, a self-reported survey was distributed electronically with more than 500 female respondents. The survey quantified vocal fatigue using the Vocal Fatigue Index. The areas investigated with the survey included the amount of potential risk involved for teachers of varying classroom sizes. Teachers’ responses from several different school districts throughout the United States were analyzed to compare grade level and classroom size on the teachers’ reported experience of vocal fatigue. Results indicated a significant effect of the physical size and capacity of classrooms on teachers reported amounts of vocal fatigue. Teachers of larger classrooms experienced significantly more vocal fatigue. Age related factors were also examined in this study. These research discoveries will have a great effect on the precautions taken by educators and school administrators to avoid vocal fatigue, and, thus, occupational risk from short- and long-term voice issues. Many additional factors which may affect perceived vocal fatigue must be explored in future research.
Hearing thresholds and the use of delayed auditory feedback provided by SpeechEasy® in patients with Parkinson’s disease
Authors: Wang Emily , Hulse Eric S. , Verhagen Metman Leonard A. , Shafiro Valeriy
Published: June 2017

The wearable in-the-ear SpeechEasy® device is registered with FDA as an anti-stuttering device. It was first investigated as a therapeutic option for hypokinetic dysarthria in Parkinson’s disease (PD) by our group in 2008. While the results demonstrated positive treatment effects on hypokinetic dysarthria in PD, the question remains whether long-term use of the device could lead to detrimental effect on hearing in a typically aging population. Since the SpeechEasy® device has the capacity to provide various levels of increased volume to the wearer, the two main goals of the current study were to determine 1) whether the long-term use of this device caused a significant shift in our subjects’ hearing thresholds; and 2) the various output levels of the SpeechEasy® CF-BTE devices used in the study. All fifteen subjects who wore the SpeechEasy® device for one year in the Wang et al. 2008 study were included. Seven met inclusion criteria for determining if a potential standard threshold shift occurred during device usage. The results indicate that the SpeechEasy® device will not cause hearing loss if programmed correctly. However, because SpeechEasy device is capable of producing potentially high output levels if programmed incorrectly, a standardized device fitting protocol must be considered.
Perception of coarticulation in listeners with cochlear implants and other spectrally degraded conditions
Authors: Gianakas Steven P. , Winn Matthew
Published: June 2017

Hearing loss can lead to not only decreased overall word recognition, but also poor access to cues that help us identify words more quickly in context. One such cue is coarticulation, or the overlap of articulatory gestures in neighboring sounds, which is utilized by listeners to more quickly an upcoming word. This study measures benefit of coarticulation when the incoming speech signal is spectrally degraded, as with the use of a cochlear implant or other degradation. In a visual world eye-tracking paradigm, listeners looked to four pictures of named objects while listening to speech stimuli in which the vowel preceding a target word contained natural cooperating coarticulation cues, conflicting cues (for a different word), or neutral cues (no coarticulation). The benefit of coarticulation was measured as reduction of latency of eye movements elicited by the cooperating cue compared to neutral cues, or the increase in latency resulting from conflicting cues; both situations would show evidence of sensitivity to coarticulation. Preliminary results suggest that coarticulation perception is partially robust to degradation for normal-hearing listeners but is highly variable/deficient in cochlear implant listeners, suggesting a disadvantage in the speed of word recognition that would not be evident in conventional word recognition scores.
Disability rights aspects of ambient noise for people with auditory disorders
Authors: Fink Daniel
Published: June 2017

The United States and European Union (EU) guarantee people with disabilities certain rights, with goals of full enjoyment, active inclusion, and equal participation in society. This approach is also found in the United Nations Convention on the Rights of Persons with Disabilities, adopted by EU. Noise is a disability rights issue for individuals with hearing loss. Many cannot understand speech in noisy places, with or without hearing aids. Ambient noise levels below 60 A-weighted decibels with a reverberation time under 0.50 seconds are needed to allow those with partial hearing loss to follow normal conversations. Noise worsens symptoms for those with tinnitus and hyperacusis. Noisy restaurants, stores, and other places deny full enjoyment and equal participation in public life to those with hearing loss, tinnitus, and hyperacusis. Legislative and regulatory action is needed to provide quiet environments, with established noise standards vigorously enforced. Technologies and environmental modifications to control noise are well known, readily available, and relatively inexpensive. The simplest modification, which costs nothing, is merely turning down the volume of amplified sound. Quiet facilitates communication for everyone and prevents development of hearing loss, tinnitus, and hyperacusis in those without auditory disorders.
Measuring the “glottal waveform” in laringectomized patients
Authors: Beckert Otto Andressa , da Silva Andrey R.
Published: June 2017

There are several noninvasive and well-devised techniques for measuring the glottal waveform of human subjects, most of which rely on source-filter separation by digital inverse filtering. A more direct and simple approach relies on the use of the Sondhi tube technique to record the subjects’ voice, from which the glottal waveform is extracted, assuming that the effect of the vocal tract is negligible on the recorded sound. Although these techniques can capture the characteristics of the source waveform from normal speakers with reasonable accuracy, they have limitations when used to capture the source waveform of tracheoesophageal and esophageal speakers. This is due to the fact that these types of speech are highly aperiodic, which may compromise the accuracy of inverse filtering techniques. Moreover, the acoustic coupling between the vocal tract and the source in tracheoesophageal and esophageal speech is not well understood yet, and the assumptions implied by the Sondhi tube techniques cannot be taken. This work proposes a new experimental technique to capture the source waveform of tracheoesophageal and esophageal speakers based on the layer-peeling technique using a open-closed tube with two microphones and a sound source. Results for the source waveform of both normal and tracheoesophageal speakers are captured with the new technique and compared with traditional methods based on both inverse filtering and the Sondhi tube.
The effect of hearing acuity on using semantic expectancy in degraded speech
Authors: Simeon Katherine M. , Bicknell Klinton , Grieco-Calub Tina M.
Published: June 2017

Speech is degraded by extrinsic factors (e.g., background noise), intrinsic factors (e.g., hearing loss), or a combination of both. Listeners can compensate for degradation with semantic expectancy, which is the ability to predict speech from surrounding linguistic information during spoken language processing. Though listeners with hearing loss use expectancy for speech understanding (Lash et al., 2013; Smiljanic & Sladen, 2013), little is known about whether their reliance on expectancy competes with their processing of acoustic input. This project examines how acoustic degradation, from background noise and hearing loss, influences listeners’ use of expectancy and how this processing affects speech perception. Adults were presented with sentences containing concrete, monosyllabic words in sentence-final position in speech-shaped noise in different SNRs (Bloom & Fischler, 1980). These words were interchanged to create congruent expectancy sentences (i.e., the final word was semantically related) and conflicting expectancy sentences (i.e., the final word was not semantically related). For conflicting expectancy sentences, normal-hearing listeners had poorer speech recognition accuracy in noise, suggesting increased use of semantic expectations when processing degraded speech. Preliminary data show listeners with hearing loss have similar performance with no background noise present. This presentation will discuss results from adults with and without hearing loss.
Hourglass arrays: Planar sparse arrays with hole-free coarrays and reduced mutual coupling
Authors: Liu Chun-Lin , Vaidyanathan Palghat
Published: June 2017

Linear (1D) sparse arrays such as nested arrays have hole-free difference coarrays with O(N2) virtual sensor elements, where N is the number of physical sensors. This property implies that O(N2) monochromatic and uncorrelated sources can be identified. For the 2D case, planar sparse arrays with hole-free coarrays having O(N2) elements have also been known for a long time. These include billboard arrays, open box arrays (OBA), and 2D nested arrays. Their merits are similar to those of the 1D sparse arrays mentioned above, although identifiability claims regarding O(N2) sources have to be handled with more care in 2D. In this presentation, we propose hourglass arrays, which have closed-form 2D sensor locations and hole-free coarrays with O(N2) elements just like the OBA. Furthermore, the mutual coupling effect, which is the undesired interaction between sensors, is reduced since the number of sensor pairs with small spacings such as λ/2 decreases. Among the planar arrays mentioned above, simulations show that hourglass arrays have the best estimation performance in the presence of mutual coupling.
Estimation of surface impedance using different types of microphone arrays
Authors: Richard Antoine , Fernandez-Grande Efren , Brunskog Jonas , Jeong Cheol Ho , Haddad Karim , Hald Jorgen , Song Woo-Keun
Published: June 2017

This study investigates microphone array methods to measure the angle dependent surface impedance of acoustic materials. The methods are based on the reconstruction of the sound field on the surface of the material, using a wave expansion formulation. The reconstruction of both the pressure and the particle velocity leads to an estimation of the surface impedance for a given angle of incidence. A porous type absorber sample is tested experimentally in anechoic conditions for different array geometries, sample sizes, incidence angles, and distances between the array and sample. In particular, the performances of a rigid spherical array and a double layer planar array are examined. The use of sparse array processing methods and conventional regulariation approaches are studied. In addition, the influence of the size of the sample on the surface impedance estimation is investigated using both experimental data and numerical simulations with a boundary element model. Results indicate that the small distance between the planar array and the sample favors a more robust estimation.
Spectrum-dependent bandpass beampattern modeling and spatial filtering with coprime linear microphone arrays
Authors: Bush Dane R. , Xiang Ning
Published: June 2017

Coprime linear microphone arrays consist of subarrays, each with inter-element spacing of half the wavelength observed times an integer factor. These two factors, M and N, corresponding to each subarray, are coprime, which ensures that their sensitivity completely overlaps only in the direction of the main beam. This implies a single observable wavelength, thus frequency; however, the grating lobe-mitigating effect can also be achieved for broadband sources [D. Bush, and N. Xiang, J. Acoust. Soc. Am., 138, 447-456 (2015)]. A modified Laplacian function provides a phenomenological model for broadband noise array responses, but real-world signals vary in spectral content making it prudent to develop a model which incorporates finer-resolution frequency dependence. This work also explores spatial filtering/source separation techniques for coprime linear microphone arrays. Multichannel experimental impulse response measurements with differing angles of incidence are convolved with independent speech signals. Subsequently, coprime beamforming is applied to the results in order to directionally filter, thus separating the source signals.
Statistical characterization of coprime sensor arrays: Array gain vs. spatially correlated noise
Authors: Bautista Radienxe , Buck John R.
Published: June 2017

Sensor arrays suppress spatially-uncorrelated noise through conventional beamforming (CBF). Array Gain (AG) quantifies the SNR improvement at the CBF output compared to the sensor-level SNR. The AG for a CBF Uniform Line Array (ULA) in uncorrelated noise is the number of sensors. A Coprime Sensor Array (CSA) is a sparse array geometry interleaving two undersampled ULAs, multiplying the subarray CBF outputs to achieve the same resolution as a fully populated ULA of the same aperture using fewer sensors [Vaidyanathan & Pal, 2010]. The CSA product process AG in uncorrelated noise is asymptotically equal to the number of sensors for large input SNR [Adhikari & Buck, 2015]. This research derives the AGs for CSAs and ULAs using the traditional SNR definition and deflection statistics [Cox, 1973] for a spatial first-order autoregressive process. This process introduces spatial correlation and is a simple model for turbulent flow noise over a towed array. Although the CSA AG is lower than the ULA for uncorrelated noise, the CSA's AG degrades more slowly than the ULA's AG with increasing noise correlation, due to larger spacing in the CSA subarrays. The CSA is more robust to correlated noise than the ULA AG. [Funded by NUWC & ONR.]
Sound speed and attenuation in marine mud seabottom
Authors: Li Zhenglin , Zhang Renhe
Published: June 2017

Acoustic propagation in shallow water is greatly influenced by the properties of the bottom. Comparing the acoustic characteristics in the sediments of sand or silty are well inverted or directly core measured, sound speed and attenuation in the mud sediment still need investigated further. An experiment was performed in the Yellow Sea in 2002 to inversion for the acoustic parameters. In the experiment, six different sediment types: Fine Sand, Silty Sand, Sand Silty, Sand-Silty-Clay, Silty Clay and Mud, are included in an area. A hybrid geoacoustic inversion scheme, which combines several inversion methods together to invert for the bottom parameters, is proposed based on the fact that the bottom acoustic parameters have different sensitivities to the different physical parameters of acoustic field. The inverted bottom parameters could distinguish the atlas marked bottom type quite well. The sound speed and attenuation parameter in the marine sediments of silty clay and mud is mainly discussed. The nonlinear frequency relationship of attenuation in the mud is compared with that of silty sand. [Work supported by the National Natural Science Foundation of China under Grant No. 11434012 and Grant No. 41561144006.]
A computational method for the time-domain intensity scattered from rough interfaces and volume heterogeneities in deep water
Authors: Olson Derek R. , Holland Charles W.
Published: June 2017

A model for the time-domain scattered intensity from a heterogeneous layered seafloor due to a point source has been recently developed by Tang and Jackson [J. Acoust. Soc. Am. Suppl. 4, 140, 3363]. To accurately model measurements taken in the deep ocean with source / receiver altitudes on the order of 5 km, the computational cost of this model is quite high, with computational cost scaling as O((kH)^2), where k is the acoustic wavenumber, and H is the source / receiver altitude. We present a fast method of numerical integration based on the work of Levin that greatly reduces the computational cost for deep water scenarios. In the absence of a complex resonant structure in the sediment, the proposed numerical integration scheme reduces the scaling to O(1). If this resonant structure is present, then the scaling is approximately O((kD)^2), where D is the total sediment thickness. Model results from deep ocean environments, including turbidite seafloors are presented and discussed.
Forward acoustic scattering analysis from solid objects immersed in the water
Authors: Soubsol David , Leon Fernand , Decultot Dominique , Chati Farid , Maze Gérard , Renou Ygaäl , Audoly Christian
Published: June 2017

Forward acoustic scattering of an immersed solid LINE (cylinder bounded by hemispherical endcaps) in water is investigated in our study. The object is made of stainless steel and its L/2a ratio is equal to 2 (L: Length of the cylinder part and a: the radius is equal to 60 mm). An impulse measurement method is used in the experimentation. Most of results are obtained experimentally, in bistatic configuration. Mobile receiver transducer is located in a distinct position from the emitter. The polar diagram patterns of the scattered pressure shows an important amplitude of pressure in the shadow side of this object. Analysis of this phenomenon is based on theoretical and experimental results obtained for a sphere with the help of elasticity theory. Moreover, this study relies on the grey-level representation of the angular position of the receiver in function of recorded time signals. Thus on forward acoustic time signals, it is possible to identify echoes due to propagation paths of waves on this object.
Accuracy of the far-field approximation for the sound radiated when an immersed steel pile, its toe made non-reflective (for clarity), is driven by a harmonic 1-kHz axisymmetric force
Authors: Hall Marshall V.
Published: June 2017

A steel pile immersed in cold seawater, its toe made non-reflective (for clarity), is driven by a 1-kHz axisymmetric force. Axial and radial vibrations travel from head to toe. Radial vibration, which radiates sound into the surrounding medium, is computed using Membrane thin shell vibration theory. In cold seawater, 1-kHz Mach waves radiate at 73° from the axis of a typical construction steel pile. A Mach wave is received at any position providing a 73° line from that position intersects the pile (at the emitting point). The vibration energy at this emitting point has travelled from the head. For any slant range (R) to a receiver, there is a minimum colatitude (COMIN) below which a Mach wave is not received. For a typical steel pile it is found that as R increases beyond the pile length (L), COMIN increases rapidly from 0°, passes through 68° at 10 L, and asymptotes to 73° as R increases further. Radiated SPLs were calculated using both far-field and all-field radiation theories as functions of colatitude, at slant ranges from 10to 1000 m. The far-field approximation (which omits the Mach wave) underestimates SPL by up to 20 dB if the receiver’s colatitude exceeds COMIN.
A brief history of underwater construction noise
Authors: Reyff James A.
Published: June 2017

Underwater sound from marine construction became an important issue around the year 2000 because of the potential impacts to marine mammals and fish that are protected by the Endangered Species Act. While protections were in place for marine mammals, there was no guidance for protecting fish. Early on, pile driving was found to visibly injure fish, leading to efforts to reduce sounds, develop protective regulatory thresholds, and research the effects of sound on aquatic species. Over the last 16 years, extensive measurements have been conducted and compiled, protective thresholds have been developed and updated, noise attenuation systems or methods have been improved and tested, and research has been conducted. This paper provides an overview of the different types of sounds characterized, thresholds for protecting animals and the types of noise mitigation strategies employed. Sounds characterized include various types of pile driving in and near water, use of explosives in and near water, mechanical demolition, and dredging-type sounds. The various types of thresholds and methodology for applying those thresholds are discussed. Finally, sounds measured from the various types of attenuation systems are summarized.
Radiated noise levels from marine geotechnical drilling and standard penetration testing
Authors: Erbe Christine , McPherson Craig
Published: June 2017

Geotechnical drilling is a common part of site investigations prior to marine construction. A small, solid core is extracted from shallow depth for examination at the surface. During standard penetration testing (SPT), a sample tube is hammered into the ground at the bottom of the borehole. The number of blows needed for the tube to penetrate a fixed depth relates to the hardness of the ground and is termed the standard penetration resistance. Recordings of the drilling and SPT operations of a jack-up rig situated in 12 m of water were obtained with a mobile recording system at 10-50 m range and 10 m depth below the sea surface. Geotechnical drilling (120 kW, 83 mm diameter drill bit, 1500 rpm, 16-17 m drill depth below the seafloor consisting of sand then mudstone) had a radiated noise level (using geometric spreading) of 142-145 dB re 1 μPa rms @ 1 m (30-10,000 Hz). SPT (50 mm outer diameter of the test tube, 15 mm wall thickness, 100 kg hammer, and 1 m drop height) exhibited a radiated noise level of 152-160 dB re 1 μPa2s @ 1 m.
Noise reduction of pile driving and unexploded ordinance detonations at offshore wind farm installation sites
Authors: Wochner Mark S. , Lee Kevin M. , McNeese Andrew R. , Wilson Preston S.
Published: June 2017

Pile driving noise during the installation of monopile foundations for offshore wind farms can produce very high noise levels, and strict regulations on this underwater noise exist around the world. In addition, the controlled detonation of unexploded ordinance (UXO) at offshore wind farm installation sites in European waters is an additional challenge and reduction of this noise is a major challenge. This paper discusses a tunable acoustic resonance-based underwater noise abatement system for use on marine pile driving, controlled UXO detonations, and other applications. The system consists of arrays of underwater air-filled resonators, which surround the noise source and are tuned to optimally attenuate noise in a frequency band of interest. System demonstrations that were conducted at two offshore wind farm sites in the North Sea will be discussed, in which peak sound pressure level reduction of nearly 40 dB was measured, and almost 20 dB sound exposure level reduction was measured. Laboratory testing on noise generated by a combustive sound source, used to simulate UXO noise, will also be discussed. The method of deploying these resonator arrays in a simple collapsible framework, operational advantages of this approach, and future projects using this technology will be shared.
Underwater sensing and surveying with autonomous vehicles
Authors: Siderius Martin , Muzi Lanfranco , Küsel Elizabeth T. , Nielsen Peter L.
Published: June 2017

The increasing capabilities of autonomous underwater vehicles is leading to a variety of new possibilities for underwater sensing and surveying. Portland State University and the Centre for Maritime Research and Experimentation (CMRE) have been working for several years on developing capabilities for underwater sensing with autonomous vehicles- mainly gliders or hybrid glider/powered vehicles. These vehicles have long deployment durations (weeks to months) but move relatively slowly. Two applications have been the focus of the work, seabed characterization and marine mammal population density estimation. For seabed sensing, the ambient noise field (e.g., breaking wave sounds) is being used as the sound source but one of the challenges for this method has been the limitations on the size of the hydrophone receiver array that can be deployed from the vehicle. For the marine mammal population density studies a glider has been customized with two hydrophones on the wings and various species are localized in bearing using time difference of arrival methods. Previous density estimation methods used a single, fixed hydrophone. In this presentation, experiments, challenges, and results will be described from several studies using autonomous vehicles for seabed characterization and marine mammal population density estimation.
Bistatic continuous active sonar processing using arrays towed from unmanned underwater vehicles
Authors: LePage Kevin D. , Canepa Gaetano , Bates Jeffrey , Tesei Alessandra , Micheli Michele , Munafo Andrea
Published: June 2017

Continuous active sonar (CAS) offers the possibility to continuously illuminate targets in underwater detection scenarios. The Cooperative Antisubmarine Programme at the Centre for Maritime Research and Experimentation has developed a real-time continuous active sonar signal processing algorithm and tested it at sea during two recent sea trials. These sea trials, undertaken with partners of the Littoral Continuous Active Sonar multinational project, were conducted in littoral environments for the purpose of evaluating the effectiveness of CAS against a traditional pulse active sonar benchmark. Data collected during these sea trials have been collected on both ship-towed triplet arrays and smaller arrays deployed from AUVs. Some details of the algorithm are given and results at the beamforming, detection and tracking level are presented for both CAS and PAS for an echo repeater target. [This work was made possible by the LCAS Multi-National Joint Research Project (MN-JRP), including as Participants CMRE (NATO), DSTG (AUS), DRDC (CAN), Dstl (GBR), CSSN (ITA), FFI (NOR), DTA (NZL), and ONR (USA). Funding for CMRE was provided by the NATO Allied Command Transformation.]
Observations from the field: Active acoustic systems, architecture, performers, and audience
Authors: Skalevik Magne , Ellison Steve
Published: June 2017

Active acoustic systems provide the potential for acoustic adjustability with far greater parametric control than with architectural systems. Settings and adjustments may be made and compared in extremely short time frames. With almost limitless combinations possible, how do we quantify and qualify these settings? Observations made during the course of tuning multiple Meyer Sound Constellation acoustic system installations are presented. The importance of adjusting spatial energy balance and early reflections and reverberation parameters, subjective observations and objective measurements, performer and audience impressions, and user interface will be discussed.
The future of high quality sound reinforcement
Authors: Engel Gunter
Published: June 2017

Combining advanced sound reinforcement techniques and room enhancement features provides impressive new possibilities for a wide variety of concerts, shows, and acoustic environment design. The achieved natural impression gives the sound designer the means to grip the listeners in the audience with a perception which is acting partly below the limit of consciousness and by this even more effective than all striking showmanship. Preserving the correct localization in shows or distributing sound sources around the listener instead of summing all signals to the same position achieves not only a clear and transparent sound. It dramatically improves clarity, intelligibility, and again the naturalness of perception. The lecture describes the underlying techniques to achieve these effects. Various examples of applying the concept with the acoustic enhancement system Vivace for opera and drama performances are explained in detail.
Amplified music and variable acoustics—Shorter reverberation times at low frequencies
Authors: Adelman-Larsen Niels W.
Published: June 2017

A survey from 2010 among professional musicians and sound engineers revealed that a long reverberation time at low frequencies in halls during concerts of reinforced music such as pop and rock, is a common cause for an unacceptable sounding event. Lower frequency sounds are, within the genre of popular music, rhythmically very active and loud, and a long reverberation leads to a situation where the various notes and sounds including vocals cannot be clearly distinguished. This reverberant bass sound rumble often partially masks even the direct higher pitched sounds. In an article from 2011, it was hypothesized that mid- and high-frequency sound is seldom a reason for lack of clarity and definition due to a 6 times higher absorption by audience compared to low frequencies, and a higher directivity of speakers at these frequencies. A survey from December 2016 among 25 professional musicians and sound engineers confirms that a longer reverberation at higher frequencies in the empty hall can advantageously be projected and that the 125 Hz octave band is probably the single most important band to control. Details from this survey and results regarding the author’s most recent developments in the field of variable and mobile absorption in favor of both amplified and classical musical genres are presented.
Array source for fast sequential matched multiple input multiple output (MIMO) room impulse measurements
Authors: Klein Johannes , Berzborn Marco , Vorlaender Michael
Published: June 2017

Multiple input multiple output (MIMO) room impulse responses are the base for the directivity related analysis and realistic reproduction of room acoustics. To obtain measurements with a high spatial resolution specialized loudspeaker and microphone arrays have to be used for the measurement. Due to the sensor size high-order microphone arrays can be constructed as one physical element, whereas for the respective loudspeaker arrays sequential measurement techniques have to be employed. Based on previous work concerning a dodecahedron approach [Behler 2010] a high-order loudspeaker array for measurements in a wide frequency range has been developed [Klein 2012]. It has since been thoroughly investigated [Klein 2015] and used in concert hall MIMO room impulse measurements [Noisternig 2016]. A major drawback of sequential room impulse measurements is their susceptibility to even minor time variances of the medium [Guski 2015]. Systematic limitations in the joint source and receiver array design [Morgenstern 2017] confine specific source receiver combinations to a narrow frequency range. Moreover, high-order wide frequency range sources require a large mechanical support structure leading to unwanted reflections. Considering the past experience and studies a new source for fast sequential MIMO room impulse measurements in a receiver-matched frequency range is developed. The considerations and the source will be presented.
Best approaches to modeling audience absorption
Authors: Jaramillo Ana M. , Olson Bruce C.
Published: June 2017

Multiple studies have looked into the best methods for measuring, estimating and predicting audience and seat absorption. Especially noticeable are those from L. Beranek. With simulation software the question remains about what is the best way to build those sections geometrically, and what type of absorption coefficient is appropriate to use. We created a series of model variations to understand the differences between the proposed methods and their influence in predicted room acoustics results both statistical as well as through ray tracing.
First international round robin on auralization: Concept and database
Authors: Brinkmann Fabian , Aspöck Lukas , Ackermann David , Opdam Rob , Vorlaender Michael , Weinzierl Stefan
Published: June 2017

For evaluating the performance of room acoustical simulations or numerical simulations in general, these are usually compared to corresponding measurements as a benchmark. Previous studies indicated that differences may result from neglecting wave effects (scattering, diffraction, attenuation at grazing incidence). However, it also proved to be a challenge to provide a precise representation of the primary and secondary structure (geometry, source and receiver characteristics, absorption and scattering coefficients) of the measured ground truth to be re-modeled in the simulation. The round robin on auralization aimed to overcome such shortcomings by generating a ground truth database of room acoustical environments provided to developers of room simulation software. The database includes a selection of acoustic scenes such as “single reflection,” or “coupled rooms” which isolate single acoustic phenomena, as well as three complex “real-world” environments of different size. Simulated monaural and binaural impulse responses were evaluated by comparing them to the corresponding measurements on the basis of acoustical parameters as well as perceptual qualities. We introduce the concept of the round robin along with the description of the acoustic scenes, the acquisition of monaural and binaural impulse responses, and the identification of the boundary conditions.
First international round robin on auralization: Results of the acoustical evaluation
Authors: Aspöck Lukas , Brinkmann Fabian , Ackermann David , Vorlaender Michael
Published: June 2017

The first round robin on auralization allows developers of room acoustical simulation software to provide results of their simulation model for typical scenarios of sound propagation and room acoustics. In contrast to the past round robins on room acoustics simulation, it includes a perceptual evaluation of the simulation results in addition to the evaluation based on room impulse responses and room acoustic parameters. In this talk, we will present the acoustical analysis of the monaural simulation results, based on differences in the temporal and spectral features of the transfer function as well as differences in standard room acoustical parameters for the nine scenes of the round robin. These indicate shortcomings in the corresponding simulation algorithms, provide indications which effects and which acoustical situations are particularly challenging for the different simulation approaches, and predict potential perceptual differences in the listening tests.
Recording and implementing multi-source anechoic material for use in auralizations
Authors: Miller Gregory A. , Mayell Marcus R.
Published: June 2017

Live theater or musical performance rarely involves a single performer on stage. Indeed many of the acoustic cues leading to one’s impression of a performance space are the result of hearing multiple performers interacting with one another across a stage. When using an auralization as a design tool, the interplay of different sources can be extremely valuable both for the acoustic designers and building owners. Some multi-source anechoic material is publicly available, but the material is limited both in the quality of performance and the range of genres. The authors sought to improve upon available material by recording professional actors, singers, and instrumentalists for use in auralizations. This paper will report on the recording techniques, strategies for allowing musicians to play in unison (in tempo, tonality, and dynamics), and implementation in computer modeling and ambisonic playback.
Hybrid modeling and auralization for complex structures
Authors: Mooney Jon W. , Knorr Hannah D.
Published: June 2017

When a design is too complex for direct computer simulation, a hybrid physical/computer model can be an effective tool. Such was the case of the concert stage for Grand Junction Park in Westfield, Indiana. The stage façade comprises an opus spicatum of 6x6-inch stone caps on square aluminum tubing. In addition to the dramatic look, the complicated surface provides an opportunity to control diffusion, reflection and absorption. But the effect that the irregular 6-inch tier construction would have on the stage acoustics was a troubling unknown. With nearly 10,000 closely-packed surfaces—each with dimensions straddling audible wavelengths—direct computer modeling and auralization of the pavilion’s acoustics was felt to be unreliable. Therefore, a 1/12th scale physical model was built and used to obtain high frequency impulse responses for the design. Post processing of the responses showed a dip in the frequency response, most notably for the flute section as heard at the conductor’s position. Additional post processing was used to create auralizations of a 25-piece orchestra to demonstrate the level of the effect. This presentation will describe in detail the hybrid modeling and auralization used for this project, the results of the testing, and the recommendations made to the architect.
The Model versus the room: Parametric and aural comparisons of modeled and measured impulse responses
Authors: Hochgraf Kelsey , Sacks Jonah , Markham Benjamin
Published: June 2017

Computer modeling and auralization have proven their value in the acoustical design of performance venues. However, achieving and evaluating parametric accuracy and perceptual plausibility continues to be a challenge. Advances in our measurement techniques have given us new opportunities to test our models against reality in several recently completed spaces. Impulse responses were collected from the built spaces using a B-format (multichannel) microphone, and these were analyzed in terms of numerical acoustical parameters, and for directionality and timing of reflected sound arrival. These multichannel impulse responses were also convolved with speech and music, and listening comparisons were made with the same speech and music convolved with simulated impulse responses. This session will present the results of these tests and lessons learned regarding strengths and limitations of these techniques.
How they changed: Therukoothu performace in a noisy city
Authors: Manthravadi Umashankar
Published: June 2017

When it first evolved several hundred years ago, the Kerala Koothambalam, a theatre attached to a temple, heard performances of Sanskrit drama in a quiet, almost rural surroundings. Mizhavu, a very loud and important part of the performance, did not drown the words. But now it does. Many performances these days use microphones and loudspeakers to overcome the cacophony outside. The space we are measuring is in the heart of Thrissur, a thriving city in Kerala. This is a study of the acoustics of the Koothambalam, a wooden structure which has not changed much in the last several centuries. We will also be recording a koothu, one of the plays performed in this space. As well as take accurate measurements and photographs. We will also record the performance in a studio, to process with the ambisonic impulse responses recorded in the theatre, to recreate the sound of the performance in quieter times. The presentation will be a talk with brief audio and visual samples.)
Acoustic virtual reconstruction of the Roman theater of Posillipo, Naples
Authors: Iannace Gino , Berardi Umberto
Published: June 2017

On the Hill of Posillipo, during the Imperial times, a sumptuous villa overlooking the sea was built. This villa contained an odeon and a theater. Over the centuries, the villa was destroyed and only towards the end of the 800, the discovery of some of the archaeological heritage of this site started. Today, the theater has the lower part of the cavea, while the scene and the stage have not been reconstructed yet. In the central part of the orchestra, there is a pool that was perhaps used for aquatic events. The purpose of this work is to reconstructi the acoustics of the theater as it appeared during the Imperial period. For this scope, a software for room acoustics was used. The acoustic measurements were carried out in situ for the evaluation of acoustic characteristics in the current state, and then the virtual model was tuned to the results of impulse response measurements. The acoustic analysis is finally carried out using the acoustic properties of the auditorium in its present state.
Toward opera houses: Acoustic analysis of the ancient literature
Authors: D'Orazio Dario , Bo Elena
Published: June 2017

From XVI Century to the present, the demand for buildings where the opera took place followed the development of the melodrama. Several authors wrote about the optimal design of an opera house. While some of them did not cross the boundaries of the theoretical research, as Milizia, others went on building theaters following their particular theories, such as Nicolini with the San Carlo theater in Naples. Early authors researched on a merely architectural systems and principles, following the classical approach—as Carini-Motta in relationship with Vitruvius. Late authors, from the XIX Century, widened the scope including social reason in the theather design. The aim of this article is to focus on the publications, regarding theatre design, of the early authors, analyzing them by highlighting the consistencies and discrepancies in respect to contemporary literature findings.
Falling coins, striking matches, and whispering voices to demonstrate the acoustics of an open air amphitheatre
Authors: Hak Constant , Wenmaekers Remy , Hoekstra Niels , Nicolai Bareld
Published: June 2017

Part of the Ancient Acoustics research project was to measure a dense grid of impulse responses in three ancient amphitheatres in accordance with the ISO 3382 standard. In this conference paper, the measurements performed in the theatre of Epidaurus at 264 receiver positions are used to study the “signal” to “noise” ratios (SNR) for typical low energy sound sources such as falling coins, striking matches and whispering voices, described in travel guides and used by tourists to confirm or disclaim the exceptional acoustical quality of open air amphitheatres. The “signal” is obtained from sound power measurements of these sources in a reverberation room, combined with the measured Sound Strength G at every position in the theatre. To determine G, asynchronous impulse response measurements were carried out using two dodecahedron sound sources playing stimulus signals simultaneously. The measured background noise was used as the “noise” in the SNR. Besides, speech intelligibility measurements have been carried out in accordance with IEC 60268-16 using a special “speech source,” that could be used to determine the SNR of a whispering voice. Using the sound sources mentioned, no evidence of the often claimed exceptional sound transmission was found.
Acoustic communication in fishes: Effects of temperature
Authors: Ladich Friedrich
Published: June 2017

Ambient temperature affects peripheral and central mechanisms of signal production and detection in ectothermic animals. The effects of temperature on sound production have been investigated in representatives of at least 8 families of teleosts, namely, piranhas, thorny catfishes, toadfishes, gurnards, sculpins, gobies, croakers, and gouramis, mostly under laboratory conditions. Temperature affects calling behaviour and characteristics of vocalizations, but only a few general trends are evident. Calling activity may increase with rising temperature (toadfish, sculpins, catfish), or may not be affected (triglids) or even drop (toadfish), indicating that behavioral contexts and seasonal factors other than temperature influence activity. Temperature affects sound duration differently in different species. In contrast, pulse period usually decreases and the fundamental frequency of drumming sounds therefore increases as temperature rises. The dominant frequency increases as well, whereas sound pressure level may increase or remain unaffected. Studies in otophysines (cyprinids, catfishes) showed that auditory sensitivities increase at higher temperature in both eurythermal (temperate zone) and stenothermal (tropical) species. Sensitivities typically increased with higher frequencies by maximally 10 dB with one exception (36 dB, ictalurid catfish). A temperature-dependent sensitivity was described in toadfishes during the breeding season and, together with hormonal changes, presumably influences the attraction to conspecific calls.
Comparison of the Saccules and Lagenae in six macrourid fishes from different deep-sea habitats
Authors: Deng Xiaohong , Wagner Hans-Joachim , Popper Arthur N.
Published: June 2017

The structures of inner ears were compared between six species of Macrouridae (grenadiers and rattails) that live at different ocean depths ranging from 200 to 5000 meters. The goal of this comparison is to find out if there are structural differences in their inner ears related to the depth of habitat. The size of the saccular otolith relative to fish head length varies considerably among the six species; with the largest otolith found in Nezumia aequalis and the smallest in N. parini, a mesopelagic species. N. aequalis is a species with potential sound production and auditory dominance by its drumming muscle on the swim bladder. Its saccular sensory area is four times larger than those from the two species live in shallower depths and have vision as their dominant sense (N. parini and Coryphaenoides rupestris). The differences found in the saccule and lagena of these species reflect the sensory advantages of habitats that are related to the benefits and constraints at different depths. It also reflects the fish’s particular life style and the trade-off among different sensory systems. The most obvious trade-off among sensory systems is found between vision and hearing.
Complex acoustic signaling in the toadfish, Opsanus tau
Authors: Mensinger Allen F. , Van Wert Jacey
Published: June 2017

Male oyster toadfish, Opsanus tau, produce long duration (250 to 650 ms) sexual advertisement calls or “boatwhistles” during the breeding season. When males are in close proximity, they employ multiple vocalization strategies. They normally alternate the production of boatwhistles to avoid temporal overlap. When the fundamental frequencies of multiple callers vary by more than 10 Hz, the frequency is correlated strongly with water temperature. However, when conspecifics with similar frequencies vocalized, the individuals shifted their fundamental frequencies independent of water temperature in a jamming avoidance response. Males also emit a single, short duration pulse or “grunt” (~100 ms), which are emitted almost exclusively during the boatwhistle of a conspecific male which can jam the signal. The disruptive grunt specifically targeted the second stage or tonal portion of the boatwhistle, believed to be the primary acoustic attractant for females, and its brevity and precision may allow its emitter to remain undetectable. While the acoustic repertoire of teleost fishes may be less diverse compared with terrestrial species, the toadfish can detect conspecific signals and employ different strategies to avoid temporal and frequency overlap, thus displaying a capacity for complex acoustic interactions.