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Most auditory evoked potential (AEP) studies in echolocating toothed whales measure neural responses to outgoing clicks and returning echoes using short-latency auditory brainstem responses (ABRs) arising a few ms after acoustic stimuli. However, little is known about longer-latency cortical AEPs despite their relevance for understanding echo processing and auditory stream segregation. Here, we used a non-invasive AEP setup with low click repetition rates on a trained harbor porpoise to test the long-standing hypothesis that echo information from distant targets is completely processed before the next click is emitted. We reject this hypothesis by finding reliable click-related AEP peaks with latencies of 90 and 160 ms, which are longer than 99% of click intervals used by echolocating porpoises, demonstrating that some higher-order echo processing continues well after the next click emission even during slow clicking. We propose that some of the echo information, such as range to evasive prey, is used to guide vocal-motor responses within 50–100 ms, but that information used for discrimination and auditory scene analysis is processed more slowly, integrating information over many click-echo pairs. We conclude by showing theoretically that the identified long-latency AEPs may enable hearing sensitivity measurements at frequencies ten times lower than current ABR methods.

Passive acoustic monitoring (PAM) is an increasingly used technique to access the occurrence, distribution, and abundance of cetaceans that may be visually unavailable most of the time. The largest tailings dam failure disaster occurred on 5 November 2015, when the Fundão dam collapsed, releasing over 50 million cubic meters of tailings into the Doce River basin; 14 days later, the tailings plume reached the Atlantic Ocean. PAM was implemented in the concerned area and cetacean species were acoustically identified. Whistles and clicks of visual and acoustic matches were used to predict and classify exclusive acoustic records through random forest models. The identified species were Guiana, rough-toothed, and bottlenose dolphins. Additionally, the franciscana, the most threatened cetacean in the western South Atlantic Ocean, was also acoustically identified. The whistle classifier had 86.9% accuracy with final frequency, duration, and maximum frequency ranked as the most important parameters. The clicks classifier had 86.7% accuracy with peak frequency and 3 dB bandwidth as the most important parameters for classifying species. Considering the potential effect of the increase in turbidity on sound transmission, such as attenuation, the presented classifier should be continuously improved with novel data collected from long-term acoustic monitoring.

Safety of biomedical ultrasound largely depends on controlling cavitation bubbles in vivo, yet bubble nuclei in biological tissues remain unexplored compared to water. This study evaluates the effects of elastic modulus (E) and impurities on bubble nuclei available for cavitation in tissue-mimicking polyacrylamide (PA) hydrogels. A 1.5 MHz focused ultrasound transducer with f# = 0.7 was used to induce cavitation in 17.5%, 20%, and 22.5% v/v PA hydrogels using 10-ms pulses with pressures up to peak negative pressure (p−) = 35 MPa. Cavitation was monitored at 0.075 ms through high-speed photography at 40 000 fps. At p− = 29 MPa for all hydrogels, cavitation occurred at random locations within the −6 dB focal area [9.4 × 1.2 mm (p−)]. Increasing p− to 35 MPa increased bubble location consistency and caused shock scattering in the E = 282 MPa hydrogels; as the E increased to 300 MPa, bubble location consistency decreased ( p = 0.045). Adding calcium phosphate or cholesterol at 0.25% w/v or bovine serum albumin at 5% or 10% w/v in separate 17.5% PA as impurities decreased the cavitation threshold from p− = 13.2 MPa for unaltered PA to p− = 11.6 MPa, p− = 7.3 MPa, p− = 9.7 MPa, and p− = 7.5 MPa, respectively. These results suggest that both E and impurities affect the bubble nuclei available for cavitation in tissue-mimicking hydrogels.

Remote acoustic sensing can be used to estimate the error signals in human ears without placing any physical microphones there. In this paper, the coherence between the signals picked up by physical microphones over a sphere surface and the signal obtained at the sphere center is investigated. Based on the multiple channel coherence formulas in the time domain and frequency domain, the relationship between the coherence and the placement of physical microphones is analyzed by numerical simulations first, then the experimental results obtained in a reverberation chamber and a car cabin are presented to verify the simulation results. Finally, a placement of physical microphones for active control of road noise in car cabins is discussed. Both the numerical and experimental results show that an upper limit frequency exists for accurate sound pressure estimation at the center of a sphere with the sound pressure on the sphere surface. For a sufficiently complex sound field such as that in a reverberation room or in a car, half the wavelength of the upper limit frequency is about the average distance among the physical microphones.

Acoustics is generally defined as the science that deals with the production, transmission, and reception of sound and the understanding and control of its effects. In fact, the fields of acoustics cover an especially broad range of subjects and domains, and comprehensive acoustics textbooks are usually quite thick as a consequence. While they are valuable resources for researchers, these books might appear a little daunting for a young audience or for people who are new to acoustics. This paper is an example of how educational comics can be designed and used to introduce one of the most commonly discussed topics when the basics of acoustics are taught: decibel level. Seven drawn pages constitute a visual support to explain the origin and history of the decibel, together with examples from acoustics and other domains on the use of logarithmic scales and classical decibel calculations. Several comments and comprehensive bibliographical references are also provided for each drawn page to enlarge the range of subjects or exercises that can be discussed in courses and foster further readings.

A substantial fraction of the membership of the Acoustical Society of America are faculty at various types of educational institutions and are actively engaged in educational activities. However, papers focusing on aspects of teaching, pedagogy, demonstrations, student learning, and other education topics are not often published in JASA, even though the Education in Acoustics Committee regularly offers special sessions on these topics at every ASA meeting. This special issue of JASA dedicated to Education in Acoustics includes 41 papers from authors all over the world. This introduction to the special issue briefly describes each of the papers, which have been organized into several broad categories: teaching methods and exercises; project-based learning; use of experiments, demos, and experiential learning; adapting to teaching during COVID-19; circuit models and impedance concepts; software apps and online resources; teaching musical acoustics; and descriptions of acoustics programs at a variety of institutions.

Measurements of wideband acoustic immittance (WAI) rely on the calibration of an ear probe to obtain its acoustic source parameters. The clinical use of WAI and instruments offering the functionality are steadily growing, however, no international standard exists to ensure a certain reliability of the hardware and methods underlying such measurements. This paper describes a reciprocity method that can evaluate the accuracy of and identify errors in ear-probe source calibrations. By placing the ear probes of two calibrated WAI instruments face-to-face at opposite ends of a short waveguide, the source parameters of each ear probe can be measured using the opposite calibrated ear probe. The calibrated and measured source parameters of each ear probe can then be compared directly, and the influence of possible calibration errors on WAI measurements may be approximated. In various exemplary ear-probe calibrations presented here, the reciprocity method accurately identifies errors that would otherwise remain undetected and result in measurement errors in real ears. The method is likely unsuitable for routine calibration of WAI instruments but may be considered for conformance testing as part of a potential future WAI standard.

While the glottal flow is often simplified as one-dimensional (1D) in computational models of phonation to reduce computational costs, the 1D flow model has not been validated in left-right asymmetric vocal fold conditions, as often occur in both normal and pathological voice production. It is unclear to what extent the 1D model approximates the effect of three-dimensional (3D) flow phenomena and fluid-structure interaction. In this study, we performed 1D and 3D flow simulations coupled with the two-mass vocal fold model and compared vocal fold vibration patterns at different degrees of left-right stiffness asymmetry. The flow and acoustic fields in 3D were predicted by solving the compressible Navier-Stokes equations using the volume penalization method and considering the moving vocal fold wall as an immersed boundary. The results showed that vocal fold vibration amplitudes and left-right phase differences in the 3D flow were predicted by the 1D flow model under conditions of small left-right asymmetry, while vocal fold vibration amplitudes were underestimated at conditions of large left-right asymmetry. This indicates that 1D flow models may be sufficient in modeling phonation under left-right asymmetric conditions, although the performance can be further improved by more accurately predicting air pressure on vocal fold surface.

Descriptions of the acoustic characteristics of Swedish voiceless fricatives are scarce and are limited to static measures derived from the speech of a small number of speakers. The current study provides an updated acoustic description of the static (spectral, temporal, and intensity) characteristics of word-initial voiceless fricatives in Central Standard Swedish. In addition, temporal variation of spectral centre of gravity is modelled using a generalized additive mixed model. Results show that fricatives were differentiated in terms of spectral properties, duration, and intensity level, such that sibilant fricatives were generally longer and more intense than non-sibilant fricatives. Spectral centre of gravity differentiated between all places of articulation apart from labio-dental /f/. Gender differences were found for centre of gravity in /s/ but overall, sex/gender differences were small. Dynamic analyses revealed differences in curvature as well as overall level of spectral centre of gravity across the duration of the fricative, associated with place of articulation and mediated by vowel context, fricative duration, and speaker specific patterns. The results from the present study are valuable for future cross-linguistic research, and as reference for investigations concerning children's acquisition of Swedish voiceless fricatives.

This work presents the results of a perception-based study of changes in the local soundscape at residences across India during the last 2 years of the COVID-19 pandemic and their effects on well-being, productivity during work from home (WFH), online education, anxiety, and noise sensitivity. Using emails and social media platforms, an online cross-sectional survey was conducted involving 942 participants. The responses showed that a greater percentage of participants felt that the indoor environment was noisier during the 2020 lockdown, which was attributed to increased home-entertainment usage, video-calling, and family interaction. The outdoor soundscape was much quieter during the 2020 lockdown due to drastically reduced traffic and commercial activities; however, during the 2021 lockdown, it was perceived to be comparable with pre-COVID times. While changes in indoor soundscape were shown to affect peace, happiness, and concentration while increasing annoyance, the reduction in outdoor noise positively impacted these aspects. The responses indicate that indoor soundscape changes adversely affected productivity and online education. Consequently, only 15% of participants now prefer the WFH model, while 62% have reservations about online education. In some cases, the responses demonstrate a significant influence of demography and suggest the improvement of the acoustic design of residences to support work.

Infauna influence geoacoustic parameters in surficial marine sediments. To investigate these effects, an experiment was conducted in natural sand-silt sediment in the northern Gulf of Mexico. In situ acoustic measurements of sediment sound speed, attenuation, and shear speed were performed, and sediment cores were collected from the upper 20 cm of the seabed. Laboratory measurements of sound speed and attenuation in the cores were conducted, after which the core contents were analyzed for biological and physical properties. Since no model currently accounts for the effects of infauna, a deviation from model predictions is expected. To assess the extent of this, acoustic measurements were compared with the viscous grain shearing model from Buckingham [J. Acoust. Soc. Am. 122, 1486 (2007); J. Acoust. Soc. Am. 148, 962 (2020)], for which depth-dependent profiles of sediment porosity and mean grain size measured from the cores were used as input parameters. Comparison of acoustic results with distributions of infauna, worm tubes, and shell hash suggests biogenic impacts on acoustic variability and model accuracy are important in surficial marine sediments. The presence of infauna and worm tubes were correlated with higher variability in both sound speed and attenuation and greater deviation from the model near the sediment-water interface.

Crowd noise is usually the primary noise in large waiting halls, and it is difficult to predict because it is influenced by several factors such as room acoustics and crowd characteristics. This study developed a crowd noise prediction model based on the superposition of direct and reverberant sound energy using the factors of the spatial layout of waiting halls, number and distribution of crowds, behavior ratio (ratio of vocal passengers to the total number of passengers), and average crowd sound power. To verify the model, on-site measurements were conducted in two large waiting halls to obtain the necessary input parameters. The crowd noise levels in one of the waiting halls were obtained from 1-s noise level data after excluding broadcast periods. A method for determining an individual's average sound power based on the model was also presented and found to be approximately 70.6 dB. Finally, the model was verified using measured data, and it showed that the model could accurately predict the average crowd noise level and changing trend of crowd noise in temporal and spatial dimensions with an average R-square of approximately 0.55 and average difference of approximately 1.1 dBA between the predicted and measured results.

A steerable parametric array loudspeaker (PAL) can electronically steer highly directional audio beams in the desired direction. The challenge of modelling a steerable PAL is to obtain the audio sound pressure in both near and far fields with a low computational load. To address this issue, an extension of the spherical wave expansion is proposed in this paper. The steerable velocity profile on the radiation surface is expanded as Zernike polynomials which are an orthogonal and form a complete set over a unit circle. An expression for the radiated audio sound is then obtained using a superposition of Zernike modes. Compared to the existing methods, the proposed expansion is computationally efficient and provides a rigorous transformation of the quasilinear solution of the Westervelt equation without paraxial approximations. The proposed expansion is further extended to accommodate local effects by using an algebraic correction to the Westervelt equation. Numerical results for steering single and dual beams are presented and discussed. It is shown that the single beam can be steered in the desired direction in both near and far fields. However, dual beams cannot be well separated in the near field, which cannot be predicted by the existing far field models.

The measure H1–H2, the difference in amplitude between the first and second harmonics, is frequently used to distinguish phonation types and to characterize differences across voices and genders. While H1–H2 can differentiate voices and is used by listeners to perceive changes in voice quality, its relation to voice articulation is less straightforward. Its calculation also involves practical issues with error propagation. This paper highlights some developments in the use of H1–H2 and proposes a new measure that we call “residual H1.” In residual H1, the amplitude of the first harmonic is normalized against the overall sound energy (as measured by root mean square energy) instead of against H2. Residual H1 may mitigate some of the issues with using H1–H2. The current study tests the correlation between residual H1 and electroglottographic contact quotient (CQ) and compares the ability of residual H1 vs H1–H2 to differentiate statistically across phonation types in !Xóõ and utterance-level changes in phonatory quality in Mandarin. The results show that residual H1 has a stronger correlation with CQ and differentiates contrastive and allophonic phonatory quality better than H1–H2, particularly for more constricted phonation types.

The present study investigated “the” reduction in phrase-medial Verb- the-Noun sequences elicited from 5-year-old children and young adults (18–22 yr). Several measures of reduction were calculated based on acoustic measurement of these sequences. Analyses on the measures indicated that the determiner vowel was reduced in both child and adult speech relative to content word vowels, but it was reduced less in child speech compared to adult speech. Listener ratings on the sequences indicated a preference for adult speech over children's speech. Acoustic measures of reduction also predicted goodness ratings. Listeners preferred sequences with shorter and lower amplitude determiner vowels relative to content word vowels. They also preferred a more neutral schwa over more coarticulated versions. In sequences where ratings differed by age group, the effect of coarticulation was limited to adult speech and the effect of relative schwa duration was limited to child speech. The results are discussed with reference to communicative pressures on speech, including the rhythmic and semantic pressures towards reduction versus the pressure to convey adequate information in the acoustic signal. It is argued that these competing pressures on production may delay the acquisition of adult-like function word reduction.

An autonomous surface vehicle known as a wave glider, instrumented with a low-power towed hydrophone array and embedded digital signal processor, is demonstrated as a viable low-noise system for the passive acoustic monitoring of marine mammals. Other key design elements include high spatial resolution beamforming on a 32-channel towed hydrophone array, deep array deployment depth, vertical motion isolation, and bandwidth-efficient real-time acoustic data transmission. Using at-sea data collected during a simultaneous deployment of three wave glider-based acoustic detection systems near Stellwagen Bank National Marine Sanctuary in September 2019, the capability of a low-frequency towed hydrophone array to spatially reject noise and to resolve baleen whale vocalizations from anthropogenic acoustic clutter is demonstrated. In particular, mean measured array gain of 15.3 dB at the aperture design frequency results in a post-beamformer signal-to-noise ratio that significantly exceeds that of a single hydrophone. Further, it is shown that with overlapping detections on multiple collaborating systems, precise localization of vocalizing individuals is achievable at long ranges. Last, model predictions showing a 4× detection range, or 16× area coverage, advantage of a 32-channel towed array over a single hydrophone against the North Atlantic right whale upcall are presented for the continental shelf environment south of Martha's Vineyard.

The harmonics-to-noise ratio (HNR) and other spectral noise parameters are important in clinical objective voice assessment as they could indicate the presence of nonharmonic phenomena, which are tied to the perception of hoarseness or breathiness. Existing HNR estimators are built on the voice signals to be nearly periodic (fixed over a short period), although voice pathology could induce involuntary slow modulation to void this assumption. This paper proposes the use of a deterministically time-varying harmonic model to improve the HNR measurements. To estimate the time-varying model, a two-stage iterative least squares algorithm is proposed to reduce model overfitting. The efficacy of the proposed HNR estimator is demonstrated with synthetic signals, simulated tremor signals, and recorded acoustic signals. Results indicate that the proposed algorithm can produce consistent HNR measures as the extent and rate of tremor are varied.

Technical and technological advances have revolutionised the architecture, engineering, and construction industries in recent decades. Building information modelling (BIM) methodology has become essential in the process of information management and the development of building projects. This study aims to analyse the potential advantages of the implementation of BIM-based models for the acquisition of theoretical and procedural knowledge about building acoustics. This procedure was implemented as part of a problem-solving exercise in Science, Technology, Engineering, and Mathematics (STEM) university degrees. For this purpose, three-dimensional (3D) BIM models were generated to assess the contribution of their implementation in the process of visualization, comprehension, and analysis of the acoustic behaviour of buildings. The participants' experiences and satisfaction with the BIM models were measured through a questionnaire. The results showed a high level of satisfaction among the participants and good potential for the application of 3D models based on BIM methodology for the acquisition of knowledge and practical skills in building acoustics. These results highlight the potential of BIM models to provide information for understanding the procedure followed during data collection in the experimental analysis and to facilitate the understanding of system behavior.

The Acoustic Laboratory for Marine Applications (ALMA) is a deployable and autonomous acoustic system, designed by DGA Naval Systems, to address problems in underwater acoustics, such as sound propagation in fluctuating environments. In this article, data from the ALMA-2016 at-sea campaign are used to analyze the ocean fluctuation's influence on sound propagation in a shallow-water waveguide. The experiment took place on the continental shelf of the island of Corsica in November 2016. A source and a receiver array were 9.3 km apart in a nearly constant water depth of 100 m. The source emitted a variety of signals from which the chirp (1–13 kHz) is used to extract the waveguide eigenrays. To do so, a time-domain beamforming is performed on the match-filtered received signals with an automatic detection of local maxima in the time of arrival/direction of arrival (TOA/DOA) domain. A 2 min acquisition period of more than 13 h duration shows significant fluctuations in eigenray TOAs/DOAs. Qualitative comparisons with synthetic signals obtained from simulations in two and three dimensions permit reproduction of the observed eigenray fluctuations without including range dependence of the sound-speed profile.

The acoustics research at Tongji University began in the mid-1950s. The Institute of Acoustics (IOA) of Tongji University, which is one of the earliest institutions engaged in acoustics research in China, was formally established in 1984. The IOA has decades of experience in acoustics education, including offering a master's degree program for approximately 40 years, a Ph.D. program for 35 years, and a postgraduate program for 25 years. The IOA is one of the oldest acoustic research facilities in China with outstanding acoustics laboratories. Research at the IOA is performed in many areas of acoustics, which focus on detection acoustics and marine acoustics, laser ultrasonics and photoacoustics, medical ultrasonics and bioacoustics, architectural acoustics, environmental acoustics, noise control, functional and microstructural acoustic materials, aeroacoustics and aeronautical acoustics, and vehicle acoustics. This paper presents acoustics education at the undergraduate and graduate levels of Tongji University and its outstanding acoustics research facilities in detail.

Commercially available broadband echosounders have the potential to classify acoustic targets based on their scattering responses, which are a function of their species-specific morphological and physiological properties. This is particularly important in complex environments with biologically diverse fish assemblages. Using theoretical acoustic scattering models among 130 fishes across six species, we examine the potential to classify reef fish based on the fine-scale gas-bearing swim bladder morphology quantified from three-dimensional computed-tomography models. Modeled echoes of the swim bladder for an incident broadband sound source (30–200 kHz) and across a range of orientation angles (±44°) are acoustically simulated using the boundary element method. Backscatter models present characteristics that are consistent within species and distinguishable among them. Broadband and multifrequency echoes are classified and compared with Bayesian, support vector machine, k-nearest neighbor, and convolutional neural network estimators. Classifiers have higher accuracies (>70%) when noise is not present and perform better when applied to broadband spectra than multifrequency data (42, 70, 100, 132, 160, 184 kHz). The modeling and classification approaches presented indicate that a taxonomic distinction based on morphologically dependent scattering responses is possible and may provide the capacity to acoustically discriminate among fish species.

Sounds to Astound is an acoustics demonstration show, produced for the community twice yearly by the Brigham Young University Student Chapter of the Acoustical Society of America. The free, interactive demonstration show explores the science of sound for a target audience of fifth- to eighth-grade students. Introductory acoustics concepts, such as longitudinal wave motion, wave properties, propagation effects, and standing waves, are taught with live demonstrations, animations, and videos. The goal of this paper is to inspire and encourage readers in their outreach efforts by describing the purposes of Sounds to Astound and technical details of several entertaining and educational demonstrations. Lessons learned from a decade of these student-produced shows serve as an aid for future efforts and highlight the benefits of outreach efforts, particularly for the students involved.

Voice therapy often emphasizes vibratory sensations in the front part of the vocal tract during phonation to improve vocal efficiency. It remains unclear what laryngeal and vocal tract adjustments are elicited in speakers by this emphasis on oral vibratory sensations. Using a three-dimensional phonation model, this study aims to identify laryngeal and epilaryngeal adjustments that might produce maximal oral vibratory sensations during phonation, as quantified by the oral sound pressure level (SPL), and thus are likely to be elicited in voice therapy at different semi-occluded vocal tract configurations. Results show that maximum oral SPL occurs at intermediate vocal fold adduction configurations characterized by a trade-off between glottal gap and vocal fold vertical thickness. Epilaryngeal tube narrowing further increases the oral SPL in an open vocal tract, but has little effect on oral SPL in semi-occluded vocal tracts. Laryngeal and epilaryngeal configurations producing the maximum oral SPL generally have lower peak vocal fold contact pressure when producing a target output SPL. These favorable configurations are more easily identified in open vocal tracts than semi-occluded vocal tracts. However, semi-occlusion increases both the mean and dynamic oral pressure, which may familiarize speakers with oral vibratory sensations and facilitate adoption of favorable laryngeal configurations.

Recent studies have advocated for the use of connected speech in clinical voice and speech assessment. This suggestion is based on the presence of clinically relevant information within the onset, offset, and variation in connected speech. Existing works on connected speech utilize methods originally designed for analysis of sustained vowels and, hence, cannot properly quantify the transient behavior of connected speech. This study presents a non-parametric approach to analysis based on a two-dimensional, temporal-spectral representation of speech. Variations along horizontal and vertical axes corresponding to the temporal and spectral dynamics of speech were quantified using two statistical models. The first, a spectral model, was defined as the probability of changes between the energy of two consecutive frequency sub-bands at a fixed time segment. The second, a temporal model, was defined as the probability of changes in the energy of a sub-band between consecutive time segments. As the first step of demonstrating the efficacy and utility of the proposed method, a diagnostic framework was adopted in this study. Data obtained revealed that the proposed method has (at minimum) significant discriminatory power over the existing alternative approaches.

Bionic camouflage covert underwater acoustic communication has recently attracted great attention. However, we have not found relevant methods or literature to recognize these bionic camouflage communication signals (BCCSs) in the area of anti-reconnaissance. Focused on recognizing the BCCSs, this article proposes a recognition method based on the statistics of inter-click intervals to recognize the camouflaged click communication train (CCCT), which is modulated by time delay difference (TDD). We first analyze the characteristics of TDD distributions of CCCT and real click train (RCT). According to the coding principle, the TDDs of CCCTs present a ladder-like distribution with a fixed time step, and the TDDs are equal to the integral multiple of the fixed time step. On the contrary, the TDDs of RCTs are approximately random distribution within a certain time range. Therefore, based on the different TDD distributions, this article classifies CCCTs and RCTs by utilizing the statistical property of TDD distributions. To measure the TDDs of diverse cetacean clicks accurately, a new click location scheme based on the dynamic window energy ratio is proposed. Next, based on the statistics of TDD distribution, the influences of the TDDs that are caused by multipath interferences are eliminated by iteration. Simulations demonstrate the accuracy of the recognition method under different conditions.

Virtually all undergraduate communication sciences and disorders programs require a course that covers acoustic phonetics. Students typically have a separate phonetics (transcription) course prior to taking the acoustic phonetics course. This paper describes a way to structure an acoustic phonetics course into two halves: a first half that focuses on the source, including basic acoustics (simple harmonic motion, harmonics), vocal fold vibration, modes of phonation, and intonation, and a second half that focuses on the filter, including resonance and tube models, vowel formants, and consonant acoustics. Thus, basic acoustic properties are interwoven with specific examples of speech-related acoustics. In addition, two projects that illustrate concepts from the two halves of the course (one on fundamental frequency and the other on vowel formants) are presented.

Depending on the room type and various other assumptions, the damping distribution of a room will follow a specific “reference distribution,” i.e., any deviation from the distribution should indicate sound coloration. In this paper, sound coloration will be caused by the acoustic feedback when operating sound reinforcement systems at relatively high gain settings. A sound reinforcement system based on measured impulse responses as well as an “emulated” system are studied using two speech and two music samples with system gains ranging from low to just before instability. Using one microphone placed in the audience area, blind estimation of sound coloration is achieved by computing decay times of non-harmonic components in the time-frequency domain. The results show that the computed damping distributions agree well with the chi-square distributions at low system gains. As the system gain increases, the distributions are shifted toward lower damping constants, and their shapes deviate more and more from the reference distribution, thus, giving a clear indication of sound coloration. The suggested objective measures show that deviations from the reference damping distribution can be detected at substantially lower system gains compared to results of related listening tests where audible coloration is evaluated.

The chimney tube, comprising connected body and chimney segments with respective diameters d1 and d2, is ubiquitous in musical acoustics and can be a useful analog for flute embouchure holes, woodwind toneholes, and organ pipes. Most treatments are complex and not readily accessible to lay audiences. Simple expressions for the input impedance of ideal chimney tubes are derived by two methods. Chimneys short compared to the overall length of the resonator are shown to have effective lengths larger than their physical length by the factor ( d1/ d2)2 when the chimney is open and smaller by the same factor when closed. Examples presented range from a simple cylindrical tube open at both ends to the classic Helmholtz resonator. Insights are provided about tonehole placement and the tradeoffs between a tonehole's position and diameter; how the resonance frequencies of an open-open chimney tube with a short chimney do not depend on which end is used as the input; and the interesting way in which a long chimney can change the spacing of the resonances from the 1:2:3 pattern of a cylinder open at both ends to the characteristic 1:3:5 pattern of a cylinder open at one end and closed at the other.

This paper presents reference equivalent threshold sound pressure levels (RETSPLs) for the Wireless Automated Hearing Test System (WAHTS), a recently commercialized device developed for use as a boothless audiometer. Two initial studies were conducted following the ISO 389-9 standard [ISO 389-9 (2009). “Acoustics—Reference zero for the calibration of audiometric equipment. Part 9: Preferred test conditions for the determinations of reference hearing threshold levels” (International Organization for Standardization, Geneva)]. Although the standard recruitment criteria are intended to yield otologically normal test subjects, the recruited populations appeared to have slightly elevated thresholds [5–10 dB hearing level (HL)]. Comparison of WAHTS thresholds to other clinical audiometric equipment revealed bias errors that were consistent with the elevated thresholds of the RETSPL populations. As the objective of RETSPLs is to ensure consistent thresholds regardless of the equipment, this paper presents the RETSPLs initially obtained following ISO 389-9:2009 and suggested correction to account for the elevated HLs of the originally recruited populations. Two additional independent studies demonstrate the validity of these corrected thresholds.

Odontocetes have evolved special acoustic structures in the forehead to modulate echolocation and communication signals into directional beams to facilitate feeding and social behaviors. Whistle directivity was addressed for the Indo-Pacific humpback dolphin ( Sousa chinensis) by developing numerical models in the current paper. Directivity was first examined at the fundamental frequency 5 kHz, and simulations were then extended to the harmonics of 10, 15, 20, 25, and 30 kHz. At 5 kHz, the –3 dB beam widths in the vertical and horizontal planes were 149.3° and 119.4°, corresponding to the directivity indexes (DIs) of 4.4 and 5.4 dB, respectively. More importantly, we incorporated directivity of the fundamental frequency and harmonics to produce an overall beam, resulting in −3 dB beam widths of 77.2° and 62.9° and DIs of 8.2 and 9.7 dB in the vertical and horizontal planes, respectively. Harmonics can enhance the directivity of fundamental frequency by 3.8 and 4.3 dB, respectively. These results suggested the transmission system can modulate whistles into directional projection, and harmonics can improve DI.

Known errors exist in loudspeaker array processing techniques, often degrading source localization and timbre. The goal of the present study was to use virtual loudspeaker arrays to investigate how treatment of the interaural time delay (ITD) cue from each loudspeaker impacts these errors. Virtual loudspeaker arrays rendered over headphones using head-related impulse responses (HRIRs) allow flexible control of array size. Here, three HRIR delay treatment strategies were evaluated using minimum-phase loudspeaker HRIRs: reapplying the original HRIR delays, applying the relative ITD to the contralateral ear, or separately applying the HRIR delays prior to virtual array processing. Seven array sizes were simulated, and panning techniques were used to estimate HRIRs from 3000 directions using higher-order Ambisonics, vector-base amplitude panning, and the closest loudspeaker technique. Compared to a traditional, physical array, the prior HRIR delay treatment strategy produced similar errors with a 95% reduction in the required array size. When compared to direct spherical harmonic (SH) fitting of head-related transfer functions (HRTFs), the prior delays strategy reduced errors in reconstruction accuracy of timbral and directional psychoacoustic cues. This result suggests that delay optimization can greatly reduce the number of virtual loudspeakers required for accurate rendering of acoustic scenes without SH-based HRTF representation.

The recently named Rice's whale in the Gulf of Mexico is one of the most endangered whales in the world, and improved knowledge of spatiotemporal occurrence patterns is needed to support their recovery and conservation. Passive acoustic monitoring methods for determining spatiotemporal occurrence patterns require identifying the species' call repertoire. Rice's whale call repertoire remains unvalidated though several potential call types have been identified. This study uses sonobuoys and passive acoustic tagging to validate the source of potential call types and to characterize Rice's whale calls. During concurrent visual and acoustic surveys, acoustic-directed approaches were conducted to obtain visual verifications of sources of localized sounds. Of 28 acoustic-directed approaches, 79% led to sightings of balaenopterid whales, of which 10 could be positively identified to species as Rice's whales. Long-moan calls, downsweep sequences, and tonal-sequences are attributed to Rice's whales based on these matches, while anthropogenic sources are ruled out. A potential new call type, the low-frequency downsweep sequence, is characterized from tagged Rice's whale recordings. The validation and characterization of the Rice's whale call repertoire provides foundational information needed to use passive acoustic monitoring for better understanding and conservation of these critically endangered whales.

The practicality of the finite-difference time-domain (FDTD) method was confirmed by comparing head-related transfer functions obtained from a three-dimensional (3D) digital model of a bat ( Rhinolophus ferrumequinum nippon) head with acoustic experiments using a 3D printed physical model. Furthermore, we simulated the auditory directionality using a 3D digital model that was modified based on the pinna movement of a bat during echolocation and found that the alternating movements of the left and right pinna result in a binaural sound pressure difference for vertical sources. Using the FDTD method, suitable for simulating acoustics in large spaces, we could analyze in detail the binaural echoes that bats receive and the acoustic cues they use for echolocation.

Phase aberration induced by soft tissue inhomogeneities often complicates high-intensity focused ultrasound (HIFU) therapies by distorting the field and, previously, we designed and fabricated a bilayer gel phantom to reproducibly mimic that effect. A surface pattern containing size scales relevant to inhomogeneities of a porcine body wall was introduced between gel materials with fat- and muscle-like acoustic properties—ballistic and polyvinyl alcohol gels. Here, the phantom design was refined to achieve relevant values of ultrasound absorption and scattering and make it more robust, facilitating frequent handling and use in various experimental arrangements. The fidelity of the interfacial surface of the fabricated phantom to the design was confirmed by three-dimensional ultrasound imaging. The HIFU field distortions—displacement of the focus, enlargement of the focal region, and reduction of focal pressure—produced by the phantom were characterized using hydrophone measurements with a 1.5 MHz 256-element HIFU array and found to be similar to those induced by an ex vivo porcine body wall. A phase correction approach was used to mitigate the aberration effect on nonlinear focal waveforms and enable boiling histotripsy treatments through the phantom or body wall. The refined phantom represents a practical tool to explore HIFU therapy systems capabilities.

Four species of grouper (family Epinephlidae), Red Hind ( Epinephelus guttatus), Nassau ( Epinephelus striatus), Black ( Mycteroperca bonaci), and Yellowfin Grouper ( Mycteroperca venenosa) share an aggregation site in Little Cayman, Cayman Islands and produce sounds while aggregating. Continuous observation of these aggregations is challenging because traditional diver or ship-based methods are limited in time and space. Passive acoustic localization can overcome this challenge for sound-producing species, allowing observations over long durations and at fine spatial scales. A hydrophone array was deployed in February 2017 over a 9-day period that included Nassau Grouper spawning. Passive acoustic localization was used to find positions of the grouper-produced calls recorded during this time, which enabled the measurement of call source levels and evaluation of spatiotemporal aspects of calling. Yellowfin Grouper had the lowest mean peak-to-peak (PP) call source level, and Nassau Grouper had the highest mean PP call source level (143.7 and 155.2 dB re: 1 μPa at 1 m for 70–170 Hz, respectively). During the days that Nassau Grouper spawned, calling peaked after sunset. Similarly, when Red Hind calls were abundant, calls were highest in the afternoon and evening. The measured source levels can be used to estimate communication and detection ranges and implement passive acoustic density estimation for these fishes.

This is an investigation of sound propagation over a muddy seabed at low grazing angles. Data were collected during the 2017 Seabed and Bottom Characterization Experiment, conducted on the New England Mud Patch, a 500 km2 area of the U.S. Eastern Continental Shelf characterized by a thick layer of muddy sediments. Sound Underwater Signals (SUS), model Mk64, were deployed at ranges of 1–15 km from a hydrophone positioned 1 m above the seafloor. SUS at the closest ranges provide measurements of the bottom reflection at low grazing angles (

A polyurethane-based tissue mimicking material (TMM) and blood mimicking material (BMM) for the acoustic and thermal characterization of high intensity therapeutic ultrasound (HITU) devices has been developed. Urethane powder and other chemicals were dispersed into either a high temperature hydrogel matrix (gellan gum) or degassed water to form the TMM and BMM, respectively. The ultrasonic properties of both TMM and BMM, including attenuation coefficient, speed of sound, acoustical impedance, and backscatter coefficient, were characterized at room temperature. The thermal conductivity and diffusivity, BMM viscosity, and TMM Young's modulus were also measured. Importantly, the attenuation coefficient has a nearly linear frequency dependence, as is the case for most soft tissues and blood at 37 °C. Their mean values are 0.61 f1.2 dB cm−1 (TMM) and 0.2 f1.1 dB cm−1 (BMM) based on measurements from 1 to 8 MHz using a time delay spectrometry (TDS) system. Most of the other relevant physical parameters are also close to the reported values of soft tissues and blood. These polyurethane-based TMM and BMM are appropriate for developing standardized dosimetry techniques, validating numerical models, and determining the safety and efficacy of HITU devices.

Substantial evidence suggests that sensitivity to the difference between the major vs minor musical scales may be bimodally distributed. Much of this evidence comes from experiments using the “3-task.” On each trial in the 3-task, the listener hears a rapid, random sequence of tones containing equal numbers of notes of either a G major or G minor triad and strives (with feedback) to judge which type of “tone-scramble” it was. This study asks whether the bimodal distribution in 3-task performance is due to variation (across listeners) in sensitivity to differences in pitch. On each trial in a “pitch-difference task,” the listener hears two tones and judges whether the second tone is higher or lower than the first. When the first tone is roved (rather than fixed throughout the task), performance varies dramatically across listeners with median threshold approximately equal to a quarter-tone. Strikingly, nearly all listeners with thresholds higher than a quarter-tone performed near chance in the 3-task. Across listeners with thresholds below a quarter-tone, 3-task performance was uniformly distributed from chance to ceiling; thus, the large, lower mode of the distribution in 3-task performance is produced mainly by listeners with roved pitch-difference thresholds greater than a quarter-tone.

Changing the balance between the early and late reflections in the impulse response affects the clarity of speech, and also the spatial perception of the sound source is affected when the direction of the early reflections is manipulated. While the effect of noise on early reflections has long been investigated in speech intelligibility studies, it is unclear whether and how the spatial characteristics of the source are altered by noise, and whether this would influence speech intelligibility in any way. The aim of the present work was to analyze the spatial perception of a speech source in noise and its relationship, if any, with speech intelligibility. Impulse responses with specular or scattered early reflections and two different reverberant tails were used to create sound fields with controlled clarity and reverberation. It emerged that noise affects spatial cues compared to the reverberation-only (quiet) condition; ratings are consequently changed, and most percepts are distorted. Speech intelligibility is also sensitive to changes in acoustic variables and the type of reflection, but the direct association between spatial percepts and speech intelligibility is weak.

This study replicates and extends the recent findings of Lee, Keating, and Kreiman [J. Acoust. Soc. Am. 146(3), 1568–1579 (2019)] on acoustic voice variation in read speech, which showed remarkably similar acoustic voice spaces for groups of female and male talkers and the individual talkers within these groups. Principal component analysis was applied to acoustic indices of voice quality measured from phone conversations for 99/100 of the same talkers studied previously. The acoustic voice spaces derived from spontaneous speech are highly similar to those based on read speech, except that unlike read speech, variability in fundamental frequency accounted for significant acoustic variability. Implications of these findings for prototype models of speaker recognition and discrimination are considered.

This study quantified the effects of face masks on spectral speech acoustics in healthy talkers using habitual, loud, and clear speaking styles. Harvard sentence lists were read aloud by 17 healthy talkers in each of the 3 speech styles without wearing a mask, when wearing a surgical mask, and when wearing a KN95 mask. Outcome measures included speech intensity, spectral moments, and spectral tilt and energy in mid-range frequencies which were measured at the utterance level. Masks were associated with alterations in spectral density characteristics consistent with a low-pass filtering effect, although the effect sizes varied. Larger effects were observed for center of gravity and spectral variability (in habitual speech) and spectral tilt (across all speech styles). KN95 masks demonstrated a greater effect on speech acoustics than surgical masks. The overall pattern of the changes in speech acoustics was consistent across all three speech styles. Loud speech, followed by clear speech, was effective in remediating the filtering effects of the masks compared to habitual speech.

An acoustics course for undergraduate music majors should take advantage of the natural affinity between acoustic science and musical practice. In this study, current students and recent graduates of one university's music school were surveyed with the goal of assessing their unique needs in an acoustics curriculum. The results of the survey are reported, and several curriculum recommendations are provided based on the principles of human-centered design. In particular, the acoustics course can harness musicians' intuitive understanding of sound by incorporating musical instruments into classroom demonstrations. Also, acoustics instructors should strive to introduce students to acoustical software, which is also used in the music industry. Finally, the survey findings suggest that the contemporary shift toward active learning and technology-based instruction in acoustics pedagogy is beneficial to music students.

The Ocean Observatories Initiative (OOI) sensor network provides a unique opportunity to study ambient sound in the north-east Pacific Ocean. The OOI sensor network has five low frequency (Fs = 200 Hz) and six broadband (Fs = 64 kHz) hydrophones that have been recording ambient sound since 2015. In this paper, we analyze acoustic data from 2015 to 2020 to identify prominent features that are present in the OOI acoustic dataset. Notable features in the acoustic dataset that are highlighted in this paper include volcanic and seismic activity, rain and wind noise, marine mammal vocalizations, and anthropogenic sound, such as shipping noise. For all low frequency hydrophones and four of the six broadband hydrophones, we will present long-term spectrograms, median time-series trends for different spectral bands, and different statistical metrics about the acoustic environment. We find that 6-yr acoustic trends vary, depending on the location of the hydrophone and the spectral band that is observed. Over the course of six years, increases in spectral levels are seen in some locations and spectral bands, while decreases are seen in other locations and spectral bands. Last, we discuss future areas of research to which the OOI dataset lends itself.

Funding from the ECHO Program of the Vancouver Fraser Port Authority. Long-term passive acoustic monitoring of cetaceans is frequently limited by the data storage capacity and battery life of the recording system. Duty cycles are a mechanism for subsampling during the recording process that facilitates long-term passive acoustic studies. While duty cycles are often used, there has been little investigation on the impact that this approach has on the ability to answer questions about a species' behavior and occurrence. In this study, the effects of duty cycling on the acoustic detection of southern resident killer whales (SRKW) (Orcinus orca) were investigated. Continuous acoustic data were subsampled to create 288 subsampled datasets with cycle lengths from 5 to 180 min and listening proportions from 1% to 67%. Duty cycles had little effect on the detection of the daily presence of SRKW, especially when using cycle lengths of less than an hour. However, cycle lengths of 15–30 min and listening proportions of at least 33% were required to accurately calculate durations of acoustic bouts and identify those bouts to ecotype. These results show that the optimal duty cycle depends on the scale of the research question and provide a framework for quantitative analysis of duty cycles for other marine species. Publisher PDF

The active noise control (ANC) technique has been applied in staggered windows to improve the noise reduction at low frequencies. The control performance of such a system deteriorates significantly at some frequencies where the secondary source cannot radiate effectively due to the reflection at the boundaries of the staggered window. A resonant absorber consisting of a perforated panel and coiled up tubes is proposed to solve the problem. By designing a combination of different absorbers, a proper sound absorption coefficient is achieved around the ineffective frequency. Numerical simulations show that the active sound power reduction increases by 13.5 dB at the frequency with the absorbers attached on one end of the staggered window, and the overall sound power reduction between 100 and 500 Hz increases from 25.9 to 31.2 dB. Attaching the sound absorbers elsewhere in the upstream of the secondary source, for example, on the side walls of the duct also works. The active sound power reduction at 435 Hz increases by 6.3 dB after attaching the absorbers in the experiments, and the noise reduction increment at the evaluation point is 13.6 dB, which agrees with simulation results and demonstrates the feasibility of the proposed sound absorbers.

There is a demand around the world for professional staff to join acoustic consulting companies, and in Australasia this demand is increasing. The extent of acoustics covered in undergraduate programs varies greatly, and there is usually the need to provide additional learning opportunities for new employees. While larger companies may provide training in-house, there are times when they are not in a position to release the senior members to provide such training. By contrast, smaller companies generally do not have this option. Short courses or similar programs can provide the necessary upskilling of new staff, but such courses may not be offered at times or locations that suit the company recruitment program. To overcome the demonstrated need for supplementary education programs in acoustics, the Australasian Association for Acoustical Consultants worked with the University of New South Wales to develop a fully flexible distance learning program, loosely based on the United Kingdom Institute of Acoustics Diploma program. The development, implementation, and experiences from over a decade of offering this program are discussed in this paper.

Native talkers are able to enhance acoustic characteristics of their speech in a speaking style known as “clear speech,” which is better understood by listeners than “plain speech.” However, despite substantial research in the area of clear speech, it is less clear whether non-native talkers of various proficiency levels are able to adopt a clear speaking style and if so, whether this style has perceptual benefits for native listeners. In the present study, native English listeners evaluated plain and clear speech produced by three groups: native English talkers, non-native talkers with lower proficiency, and non-native talkers with higher proficiency. Listeners completed a transcription task (i.e., an objective measure of the speech intelligibility). We investigated intelligibility as a function of language background and proficiency and also investigated the acoustic modifications that are associated with these perceptual benefits. The results of the study suggest that both native and non-native talkers modulate their speech when asked to adopt a clear speaking style, but that the size of the acoustic modifications, as well as consequences of this speaking style for perception differ as a function of language background and language proficiency.

The aim of this paper is to briefly present ways to improve the interest of students who begin to attend physical acoustic courses and give examples of exercises (among many others) dedicated to the deepening of their training. Regarding the interest of students who are just beginning, two particular questions arise: how is the attention of these students gained? and how would the students be properly trained when starting out? An attempt to answer these questions is given at the beginning of the paper. Then, the paper continues with a brief presentation of some problems, which are extracted from a three-volume textbook that was published recently, precisely posed, and solved in detail analytically in an exact or approximate manner, even numerically, according to the three levels (bachelor, master, and Ph.D.) while offering an opportunity to deal extensively with the modeling of important topics in acoustics, which are treated as having several ways to be addressed.

This paper investigates the feasibility of remotely generating a quiet zone in an acoustic free field using multiple parametric array loudspeakers (PALs). A primary sound field is simulated using point monopoles located randomly in a two-dimensional plane, or three-dimensional (3D) space, whereas the secondary sound field is generated by multiple PALs uniformly distributed around the circumference of a circle sitting on the same plane as the primary sources, or on the surface of a sphere for 3D space. A quiet zone size is defined as the diameter of the maximal circular zone within which the noise reduction is greater than 10 dB. The size of this quiet zone is found to be proportional to 0.19 λN for N secondary sources with a wavelength λ when the primary and secondary sources are in the same plane, whereas it is found to be 0.55 λN1/2 for the 3D case. The size of the quiet zones generated by PALs is similar to that observed with traditional omnidirectional loudspeakers; however, the effects of using PALs on the sound field outside the target zone is much smaller due to their sharp radiation directivity and slow decay rate along the propagation distance. Experimental results are also presented to validate these numerical simulations.

Humans are able to follow a speaker even in challenging acoustic conditions. The perceptual mechanisms underlying this ability remain unclear. A computational model of attentive voice tracking, consisting of four computational blocks: (1) sparse periodicity-based auditory features (sPAF) extraction, (2) foreground-background segregation, (3) state estimation, and (4) top-down knowledge, is presented. The model connects the theories about auditory glimpses, foreground-background segregation, and Bayesian inference. It is implemented with the sPAF, sequential Monte Carlo sampling, and probabilistic voice models. The model is evaluated by comparing it with the human data obtained in the study by Woods and McDermott [Curr. Biol. 25(17), 2238–2246 (2015)], which measured the ability to track one of two competing voices with time-varying parameters [fundamental frequency ( F0) and formants ( F1, F2)]. Three model versions were tested, which differ in the type of information used for the segregation: version (a) uses the oracle F0, version (b) uses the estimated F0, and version (c) uses the spectral shape derived from the estimated F0 and oracle F1 and F2. Version (a) simulates the optimal human performance in conditions with the largest separation between the voices, version (b) simulates the conditions in which the separation in not sufficient to follow the voices, and version (c) is closest to the human performance for moderate voice separation.