bluuebunny/crossref_metadata_2025 huggingface dataset:
Band importance functions of listeners with cochlear implants
Authors: Bosen Adam K. , Chatterjee Monita
Published: May 2016

Band Importance Functions (BIFs) are measured by dividing the acoustic spectrum into discrete bands and estimating the relative contribution of each band to speech intelligibility. Previous studies have demonstrated strong across-subject consistency for normal hearing (NH) listener BIFs. However, it is unlikely that cochlear implant (CI) listeners will demonstrate similar consistency, because many physiological and psychophysical properties of hearing vary across CI listeners and across electrodes within an ear. Here, we measured BIFs by testing speech intelligibility of IEEE sentences filtered to contain a random subset of bands from a listener’s clinical MAP and regressing across trials to determine each band’s average contribution. BIFs obtained from NH listeners with vocoded speech that either did or did not simulate cross-band interaction followed the characteristic inverted “U” shape with a peak around 1–2 kHz that has been previously observed. In comparison, CI listeners had BIFs that were less similar across listeners and across ears within the same listener. These results indicate that CI listeners use idiosyncratic listening strategies for speech perception that are ear-specific and cannot be fully accounted for by cochlear spread of excitation.
Perceptual speech intelligibility and speech production variability in Mandarin-speaking children with cerebral palsy
Authors: Chen Li-mei , Lin Yu Ching , Hustad Katherine C. , Kent Raymond D.
Published: May 2016

Perceptual speech intelligibility index can provide an objective standard for evaluating how speech deficiencies affect listeners’ judgments. This study investigated the effective variables for assessing speech intelligibility in Mandarin-speaking children with cerebral palsy. Acoustic measurements include vowel working space, vowel duration, pitch variation, and intensity variation of speech samples collected from picture naming tasks in two children with cerebral palsy (CP) and two typically developing children (TD) at four years old. Only clear word productions in terms of perceptual judgment and vowel formant display were incorporated to secure the reliability of data analysis. For speech intelligibility, eight judges were recruited to record the words they heard from the speech samples. Major findings are: (1) CP children had a smaller vowel space area due to limited control of tongue compared to TD children; (2) CP children showed longer vowel duration because they spent more time producing speech due to limited motor control; (3) CP children exhibited greater pitch and intensity variations due to instability in speech production. More data from more participants should be included for analysis to verify the preliminary findings.
Age-related differences in talker segregation and selection: Contributions of attention to voice features
Authors: Bologna William J. , Ahlstrom Jayne B. , Dubno Judy R.
Published: May 2016

Focused attention on expected voice features, such as F0 and spectral envelope, may facilitate segregation and selection of a target talker in competing talker backgrounds. Age-related declines in attention may limit these abilities in older adults, resulting in poorer speech understanding in complex environments. To test this hypothesis, younger and older adults with normal hearing listened to sentences with a single competing talker. For a majority of trials, listener attention was directed to the target by a cue phrase that matched the target talker’s F0 and spectral envelope; on these trials, younger adults out-performed older adults. On the remaining “probe” trials, the target’s voice unexpectedly differed from the cue phrase in terms of F0 and spectral envelope; performance declined for both groups, and younger and older adults performed similarly. Thus, older adults performed poorer than younger adults only when attention could be focused on an expected voice. Moreover, older adults responded more frequently than younger adults with words from the competing sentence rather than the target sentence. Taken together, these results support the hypothesis that declines in the ability to focus attention on expected voice features contribute to speech understanding difficulties of older adults in complex environments. [Work supported by NIH/NIDCD.]
The effect of talker and listener depressive symptoms on speech intelligibility
Authors: Yi Hoyoung , Smiljanic Rajka , Chandrasekaran Bharath
Published: May 2016

It is widely recognized that depression is associated with deficits in communication behaviors (Miller, 1975), but few studies have examined speech intelligibility in talkers and listeners with elevated depressive symptoms. The present study examined intelligibility of conversational and clear speech sentences in the presence of speech shaped noise and one-talker babble noise. Talkers and listeners were young adults varying on the extent of depressive symptoms, and classified as having high depressive (HD) symptoms or low depressive (LD) symptoms based on a self-report scale (Radloff, 1977). The results showed that increased intelligibility through conversational-to-clear speech modiﬁcations was smaller for HD listeners than LD listeners during only one-talker babble noise which is consistent with Chandrasekaran et al. (2014). Intelligibility was scored lower for the sentences produced by HD speakers compared to LD speakers. Acoustic analyses indicate smaller conversation-to-clear speech modifications for speaking rate, energy in the 1-to-3 kHz, F0 mean, and F0 range for HD talkers compared to LD talkers. The results revealed that individuals with elevated depressive symptoms exhibit difficulty producing listener-oriented speaking style changes resulting in lower overall intelligibility. Adverse listening conditions involving informational masking, oft-present in typical social environs, present particularly challenging communication conditions for HD listeners.
Intonation perception problems in advanced age
Authors: Seddoh Amebu , Blay Afua , Madden John
Published: May 2016

This study sought to determine whether intonation processing difficulties for elderly people are attributable to stimulus contextual factors. Participants were 20 old (61–84 years) and 15 young (19–29 years) adult native speakers of English. They were presented auditorily a total of 108 English sentences to decode following evaluation of their hearing abilities. Each sentence conveyed either positive or negative emotional meaning and was accompanied or unaccompanied by contextual information. Both groups of participants demonstrated significantly better outcomes on the decoding of the stimuli presented with contextual information (CS) compared to the decontextualized sentences (DCS). The elderly participants performed poorly relative to the young adults on the perception of positive and negative emotions in the DCS, as well as negative emotions in the CS. On the other hand, their performance on perception of positive emotion in the CS was comparable to that of the young adults. These findings are consistent with data that have shown that intonation meaning is context dependent [e.g., Woodland and Voyer, Metaphor Symbol 26, 227–239 (2011)], and that elderly people have greater amygdala activation for positive than negative stimuli [Mather et al., Psychol. Sci. 15, 259–263 (2004)]. They suggest that intonation perception problems in old age may be influenced by stimulus contextual information, but part of their origin may lie in neurobiological factors such as age related changes in the brain.
Speech recognition thresholds in reverberation: Effects of age and hearing loss
Authors: Srinivasan Nirmal Kumar , Molis Michelle R. , Gallun Frederick J.
Published: May 2016

While young normally hearing individuals can function in moderate amounts of reverberation with a minimal reduction in speech understanding, older individuals with and without hearing loss are much more susceptible to reverberant distortions of the speech signal (Harris and Reitz, 1985; Danhauer and Johnson, 1991; Helfer, 1992). In this experiment, we evaluated the individual contributions of age and hearing loss to speech reception thresholds (SRT) for reverberant speech (T60: 0, 300, 600, and 1000 ms) in the presence of speech-shaped noise. A one-up one-down adaptive-tracking procedure was used to measure SRT in the better ear of each of 27 listeners. Results indicated, as expected, that older-hearing-impaired listeners had higher SRTs as compared to younger and older normally hearing listeners. However, the variability among listeners was very high. Multiple regression analyses predicting SRT with age and hearing loss as factors indicated that hearing loss was the most significantly contributing factor for all four reverberation times tested. The effects of age and hearing loss on speech recognition in reverberation will be discussed in relation to the data available in the literature. [Work supported by VA RR&D I01RX001020.]
Reliability and time to administer visual analog scaling of intelligibility
Authors: Mocarski Samantha T. , Watson Peter J.
Published: May 2016

Measuring speech intelligibility is important in assessing the severity of impaired communication; and is used to demonstrate change overtime, e.g. before and after therapy. Orthographic transcription of speech is considered the clinical-gold standard of assessment, but can be time consuming. Many clinicians opt for quicker, subjective scaling such as Likert-interval, percent-estimate, and more recently visual-analog scaling (VAS). Percentage-estimate and Likert-interval scaling produce lower intra- and inter-judge reliability [Schiavetti (1992)] than transcription. Recently, VAS was used to study intelligibility in persons with dysarthria and found that reliability was good [Stipancic et al. (2015)] for mild deficits. We compared VAS to transcription under 4 SNR levels of background noise (−8, −5, −2, and 5). This approach allowed us to compare the reliability of VAS to transcription throughout a range of intelligibility levels. Twenty listeners both transcribed and visually scaled the intelligibility of audio-recorded IEEE sentences of a man and woman under the 4 SNR conditions. Reliability of VAS was as good as transcription at the extremes of intelligibility and less so in the middle levels. Transcription took 3x the time of VAS to administer.
Analysis of available and listener-received phonetic information in babble-masked consonant identification
Authors: Silbert Noah H. , Zadeh Lina M.
Published: May 2016

Speech communication commonly occurs in the presence of multiple, non-target talkers. Previous work has shown that the amount of glimpsed target speech is a good predictor of overall intelligibility of babble-masked speech (Brungart et al., 2006, JASA) and that an automatic speech recognition system trained on glimpses closely approximates listener accuracy as a function of the number of babble talkers (Cooke, 2006, JASA). The present work uses a number of machine learning models to analyze the auditory information available in babble-masked speech and in available glimpses of babble-masked speech. Regularized Linear Discriminant, Support Vector Machine, and Naive Bayes classifiers were fit to modeled auditory representations of babble-masked CV syllables (with C = t, d, s, z, and V = a). The machine learning models outperformed human listeners substantially (>70% versus 54% accuracy, respectively). Analysis of predicted confusion patterns indicates that the Naive Bayes model most closely approximates human error patterns. The effects of variation in the local and absolute thresholds for glimpse calculation are explored with respect to overall accuracy and error pattern prediction in these machine learning models.
Differential beamforming with circular microphone arrays
Authors: Benesty Jacob , Chen Jingdong
Published: May 2016

Differential beamforming cannot only achieve high directional gain but also can form frequency-invariant beampatterns. Therefore, it has the great potential to be widely used in voice communications to process broadband speech signals. The performance of a differential beamformer in terms of directivity factor (DF), white noise gain (WNG), and frequency invariance of the beampattern depends on many factors including the array geometry, the number of sensors, the inter-sensor spacing, etc. In the literature, most studies on differential beamforming have been focused on linear microphone arrays and not much efforts have been devoted to other array geometries. This paper is devoted to the study of circular microphone arrays. In comparison with linear arrays whose spatial response is a function of the steering angle, circular microphone arrays can have the same spatial response along many different directions. The focal point of this paper is on differential beamforming with uniform circular microphone arrays. The main topics addressed include: (1) major properties of circular microphone arrays, (2) how to design differential beamformers with different beampatterns, (3) how to design differential beamformers with different orders, and (4) how to achieve a good control of the white noise amplification problem, high directional gains, and frequency-independent responses.
Estimating the spatial spectra of Gaussian processes with co-prime sensor arrays
Authors: Buck John R. , Adhikari Kaushallya
Published: May 2016

A co-prime sensor array (CSA) is a nonuniform line array formed by interleaving two undersampled uniform line arrays. The CSA requires fewer sensors to span the same aperture as a densely sampled uniform line array (ULA), allowing the CSA to match the resolution of the ULA for direction of arrival estimation of narrowband planewaves. However, each CSA subarray suffers from aliasing, or grating lobes, due to the spatial undersampling. Vaidyanathan and Pal (2011) proved that if the subarray undersampling factors are co-prime, the aliasing can be unambiguously resolved by multiplying the spatial spectra of the subarrays. This product spatial spectra is the spatial cross-spectral density between the arrays, and is an estimate of the spatial power spectral density (PSD). In this talk, we extend the classic results of Jenkins and Watts (1968) on the periodogram PSD estimator for Gaussian processes to obtain the product processor's bias for spatially wide-sense stationary processes, and the processor's covariance for spatially white Gaussians. Additionally, we demonstrate that the CSA's product PSD estimate is not necessarily positive definite. Consequently, the CSA product spectrum may fail to detect weak signals in the presence of strong interferers. [Work supported by ONR BRC Program.]
Three-dimensional audio immersion: Development and human perception testing of a large scale, 128 loudspeaker surround sound system
Authors: Kern Alexander , Roan Michael
Published: May 2016

Large-scale loudspeaker arrays for immersive audio are beginning to see increased interest in the research and entertainment industries. There are many algorithms to produce 3D sound fields. Although these algorithms are well developed, there is little published work that quantitatively compares their performance in terms of human perception for various numbers of speakers and speaker arrangements. In this talk, we will present listener perception test results using a 3D, 128 loudspeaker array. This loudspeaker array is part of the CUBE theatre at Virginia Tech. The CUBE is a 5-storey 50 ft. x 50 ft. black-box theatre that uses Dante™ audio-over-ethernet to send signals to the 128 loudspeakers. Combined with a 3D motion tracking system and an immersive video vision system. The CUBE provides a very high-resolution test bed for human perception studies. Using speech intelligibility tests with stimulus phrases and various interferers and noise fields, listener tests will be performed for surround sound algorithms such as Higher Order Ambisonics, Vector Based Amplitude Panning, Wave Field Synthesis, and Higher Order HARPEX (High Angular Resolution Plane Wave Expansion). The Higher Order HARPEX decoder is a novel improvement to First Order HARPEX. Also, we will present technical details of the development of a 16-element soundfield microphone.
Azimuth-elevation direction finding using one four-component acoustic vector-sensor spread spatially as a parallelogram array
Authors: Song Yang , Wong Kainam T.
Published: May 2016

An acoustic vector-sensor (a.k.a. a vector hydrophone) consists of three uni-axial velocity-sensors (which are oriented perpendicularly with regard to each other) and one pressure-sensor. Song and Wong have demonstrated how to space these four component-sensors apart in three-dimensional space, in order to extend the overall spatial aperture spanned by them, while improving the accuracy in estimating the azimuth-elevation direction-of-arrival of an acoustic emitter incident from the far field. This paper will focus on a special spatial geometry—where the four component-sensors occupy the four corners of a parallelogram in three-dimensional space.
Complementary arrays above and below the ocean surface
Authors: Hursky Paul
Published: May 2016

An experiment was performed off the coast of San Diego to study propagation from an aircraft at altitude to a microphone array above the ocean surface and to hydrophone arrays below the ocean surface. An acoustic hailing device was mounted on a small airplane. An in-air array was formed by mounting individual microphones to the frame of a small boat. Several hydrophone arrays were deployed from the boat. The airplane flew lawn mower patterns over these arrays transmitting broadband waveforms at several altitudes. We will discuss propagation modeling, array calibration and beamforming for this configuration. Doppler is a significant factor due to the high speed of the aircraft and due to the speed of sound in air being five times less than in water. We will assess matched filter gain on the air and air-water paths and discuss phenomena that degrade such gains.
Target-depth estimation in active sonar
Authors: Mours Alexis F. , Josso Nicolas , Mars Jérôme I. , Ioana Cornel , Doisy Yves
Published: May 2016

This paper presents a new target-depth estimation method that is based on ray back-propagation with a probabilistic approach. This localization algorithm tries to minimize the mean-squared error of elevation angles at the receiver and arrival times between a model and measures. This method was tested on Monte-Carlo simulations of classic active sonar scenarios and using experimental data from a real tank. In active sonar with a point-target model, combined acoustic paths may exist. These have a different path between the sonar-target and the target-array. This paper discusses also about this ray identification. Simulations with vertical array suggest that the target-depth estimation can be realized with a low uncertainty compared to the water column for long ranges in a Mediterranean sound speed profile. However, some environmental parameters as random sound-speed profile, array depth or array tilts could increase the bias and the variance of the target-depth estimator. Results on experimental data with surface noise reveal a good estimation of the target depth and validate our localization algorithm on a constant sound-speed profile with a vertical array.
Target strength observations of wobbly bubbles
Authors: Padilla Alexandra M. , Rychert Kevin , Weber Thomas C.
Published: May 2016

Large methane bubbles released from shallow water seabed are of interest because bubbles facilitate the transport process of gas through their journey in the water column, making them more susceptible to reaching the atmosphere than those released from deep water. Several models that relate bubble radius and acoustic backscattering for ideal, spherical bubbles have been developed and are typically used by researchers attempting to invert acoustic backscatter measurements for bubble size and/or gas flux. However, based on theory and field experimentation, it has been shown that large free gas bubbles in liquids (large Eötvös and/or large Reynolds numbers) are non-spherical. A shallow water (<6 m) tank experiment has been designed to understand the influence of bubble shape on acoustic backscattering. This experiment was conducted for bubbles between 1 and 5 mm in radius (Eötvös number between 0.5 and 13, Reynolds numbers between 300 and 1500, and a Morton number of approximately 7.2 × 10−11) and at frequencies between 10 and 100 kHz. These frequencies are well above bubble resonance but cover the range of typical echo sounder measurements of bubbles in the field. Calibrated bubble target strength for these bubbles will be compared to existing acoustic backscattering models for bubbles.
Frequency response of proud and partially exposed spheres at flat interfaces
Authors: Gunderson Aaron M. , Marston Philip L.
Published: May 2016

The backscattering spectrum of spheres at flat interfaces viewed at small grazing angles depends upon reflections from the sphere's surface and the interface, as well as the sphere's elastic response and Franz wave contributions. For partially exposed spheres, one approach to modeling the reflective contribution is to extend the Kirchhoff approximation [K. Baik and P. L. Marston, IEEE J. Ocean. Eng. 33, 386–396 (2008)] to spheres. Laboratory measurements were carried out using short tone bursts for solid spheres at air-water and sand-water interfaces in such a way that it was typically possible to distinguish between the reflective and delayed elastic contributions viewed in the time domain. For spheres at an air-water interface, the reflective contributions have strong similarities with the Kirchhoff approximation while the strength of the delayed elastic contributions depends strongly on the extent of exposure of elastic guided wave coupling regions. The spectra of proud targets have similarities with image approximations, yielding ridges and valleys in the frequency versus grazing angle domain (associated with interference of contributions of similar magnitude), modulated by elastic contributions. Elastic contributions from small spheres near an air-water interface display a Lloyd's mirror pattern. [Work supported by ONR.]
Sound data for a new format of short-profile fan coil units
Authors: Hoover K. Anthony , Cudequest Brandon W.
Published: May 2016

In recent years, a new format of short-profile fan coil units as offered by several manufacturers has become increasingly popular. The sound data for such units are generally plotted in octave bands on NC curves, and have the appearance of great data. However, any scrutiny raises a variety of concerns. Fundamental methods were used to approximate power levels from the provided pressure levels for use in HVAC system noise calculations on various projects. In July 2015, test lab power level data were serendipitously provided for two fan coil units that had been included in the design for a new project. In due course, test lab data for six units, based on AHRI Standard 260, were obtained. This paper will discuss the approximation methods, ranges of differences between pressure level and power level data, effects on results of calculations, referenced measurement standards, and an issue related to pure tones.
Noise reduction of heating, ventilating, and air conditioning systems in previously constructed buildings, primarily focusing on health facilities
Authors: Pippitt Logan D.
Published: May 2016

Excessive heating, ventilating, and air conditioning system noise is a leading cause of much unwanted noise throughout most commercial buildings, some of which are medical facilities with multiple systems implemented to meet the heating, cooling, and ventilation requirements. Even so, multiple opportunities for correction can be found. Many new buildings are being properly constructed with this problem in mind, though this still leaves many issues in previously constructed buildings and facilities where the noise issues have not been addressed. So what can be done? This paper will explore the possible heating, ventilating, air conditioning, and room acoustic noise issues of a particular nursing home in Olathe, Kansas. The building administrator receives complaints from families, residents, and staff members on multiple noise issues that can be directly attributed to HVAC system noise. Issues such as, but not limited to, exterior noise pollution, supply and return HVAC noise, duct breakout noise, fan coil unit noise, roof top unit source noise, and related room acoustics. In addressing these issues, solutions will aim to meet current ASHRAE acoustical guidelines for medical facilities. [This research was conducted through the School of Architecture, Design and Planning at the University of Kansas.]
Noise analysis of heating, ventilating, and air-conditioning systems: Modeling in the Trane acoustics program
Authors: Russell Jennifer E.
Published: May 2016

Many different acoustical issues can arise in a building as a result of heating, ventilating, and air-conditioning (HVAC) systems. These issues can include excessive duct-borne noise, inadequate sound isolation, loud mechanical equipment, and sound paths between spaces created by connected ductwork. In each case, the Trane Acoustics Program (TAP) software can be used to model sound paths and assess the acoustical impact the HVAC system has on a space. This paper will present several case studies representing common mechanical and architectural noise control issues resulting from HVAC system design that were assessed using the TAPsoftware.
What do acoustic tags placed on the back of echolocating dolphins really measure?
Authors: Au Whitlow , Finneran James J. , Branstetter Brian K.
Published: May 2016

Suction cup deployed acoustic tags have been used to study the echolocation behavior of a number of odontocetes, yet the relationship between the projected biosonar signals and tag recordings are not known. Acoustic data obtained from these tags consist of the number of clicks emitted, the depth at which the clicks are emitted and the inter-click intervals during a biosonar search. In order to understand the relationship between the emitted signals detected in the front of a dolphin and the signals detected by a tag, a spherical hydrophone was mounted on a wooden model of a tag and mounted on the back of a dolphin via a suction cup. The dolphin was involved in a biosonar discrimination task and the signals were measured 1 m from its blowhole along the acoustic axis of the beam and the acoustic tag placed in five different locations on the animal’s back. The signals recorded by the tag were complex with the first pulse being a high frequency resonance-like signal followed by clicks resembling the outgoing clicks but experiencing reflective interference. The peak-to-peak amplitude of the signals measured by the tag was between 40 and 50 dB lower than that of the outgoing signal.
Long-term monitoring of Physeteroidea (sperm whales, dwarf, and pygmy sperm whales) in the Central and Western Pacific
Authors: Merkens Karlina , Simonis Anne , Oleson Erin
Published: May 2016

The superfamily Physeteroidea includes three extant species: the sperm whale (Physeter macrocephalus), the dwarf sperm whale (Kogia sima), and the pygmy sperm whale (K. breviceps). Despite extreme difference in size between the Kogia spp. and their large Physeter relative, all three share ecological and acoustic traits relating to their deep-diving behavior and high rates of acoustic activity. All three species can be found across the Central and Western Pacific ocean, an area that has been monitored using passive acoustics (High-frequency Acoustic Recording Packages, HARPs) for more than 10 years. We identified sperm whale and Kogia spp. signals in the long-term HARP records from 13 locations across the Central and Western Pacific ocean. A combination of automated tools and human analysis were used to record detection events of both types of signals. While sperm whales were found at all 13 locations, the Kogia species (which cannot yet be distinguished acoustically) were detected at approximately half of the sites. Presence of sperm whale signals was modeled to determine if temporal parameters, such as lunar cycle and day of the year, could explain patterns of presence. Across the whole region the best model included the day of the year and the recording site, while sub-regions and site-specific models had slightly different combinations of parameters.
Performance of the phase and amplitude gradient estimator intensity method in interference fields
Authors: Torrie Darren K. , Whiting Eric B. , Rose Michael T. , Gee Kent L. , Sommerfeldt Scott D.
Published: May 2016

The phase and gradient amplitude estimator (PAGE) method improves the frequency bandwidth of estimated acoustic intensity over the traditional p-p method without altering the spacing between microphones [D. C. Thomas et al., J. Acoust. Soc. Am. 137, 3366–3376 (2015)]. For many broadband sources, accurate estimates may be obtained beyond the spatial Nyquist frequency by unwrapping the phase of a transfer function used in obtaining the phase gradient. However, inaccurate phase unwrapping in interference fields, such as those produced by two loudspeakers with equal strengths but opposite phase, has been observed. This results in erroneous intensity vectors. A two-dimensional, multi-microphone intensity probe was employed to investigate this phenomenon. Findings include: (a) the unwrapping error does not occur for all interference nulls, but is more likely to occur for deeper interference nulls where there is reduction in coherence; (b) rotation of the probe in the field alters which pair-wise transfer function unwraps erroneously, but does not significantly alter the direction that contains the inaccurate intensity vectors; (c) removal of the interference nulls by driving the loudspeakers as incoherent sources allows for phase unwrapping to occur multiple times and accurate estimation of intensity vectors beyond 10 kHz. [Work supported by NSF.]
Absorption of axial plane waves by using double- and single-layer perforated liners
Authors: Zhao Dan , Han Nuomin , Ang Linus
Published: May 2016

In this work, the acoustic damping performances of multiple double- and single-layer perforated liners are experimentally and numerically evaluated. These acoustic liners are associated with different open-area-ratios, thus enable the porosity effect to be studied. Controllable joint bias-grazing flow is applied to the liners. This simulates the flow configurations of a real engine system involving both bias and grazing flow. The damping performance of these liners is characterized by power absorption or transmission loss coefficient. Such coefficient is measured, as the tonal noise frequency is changed from 200 to 800 Hz. It is shown that the grazing flow can reduce the maximum power absorption coefficient, while the bias flow can improve the liners damping performance. The double-layer liner is associated with a larger maximum power absorption and a broader frequency range in comparison with that of the single-layer one. Finally, it is found that when the open area ratios of the inner and outer liners are approximately 1.1%, the damping performances of these liners are dramatically reduced, especially at higher frequency range. Increasing the open area ratios of the inner or outer liner leads to the maximum power absorption and the effective frequency range being dramatically improved.
Loudness range of a choir based on choir size and voice range profiles of individuals
Authors: Titze Ingo R.
Published: May 2016

The size of a choir is based on many factors, including availability of singers, size of the performance hall, economics of practice and performance, and the desired ensemble sound. It is generally assumed that a large choir has a greater dynamic range than a small choir. It is also generally assumed that a choir with well-trained singers can produce a greater dynamic range. A theoretical study is presented here that combines sound pressure level and loudness of homogeneous and non-homogeneous groups of singers to produce an overall sound level profile of a choir. Internal constraints are self-to-other ratio and individual voice range profiles of choir members. Results indicate that the dynamic range of a choir is determined mostly by the dynamic range of individuals with wide ranges, assuming that inhomogeneity is allowed. Choir size makes little difference. Singers with small dynamic ranges have little effect on choir dynamics unless louder voices are selectively turned off in pianissimo passages.
The importance of feature selection in supervised machine learning problems in acoustics
Authors: Dieckman Eric A.
Published: May 2016

As machine learning processes begin to gain traction and are applied to problems in acoustics, there is a need to better understand the importance of all stages of the process. Here, we discuss the preprocessing stage, where features are extracted from raw data and the best features are chosen using algorithms such as linear discriminant analysis (LDA). This allows the creation of a small-dimensional, information-dense feature vector to be used in the machine learning process. Applications discussed include classification of ground vehicles based on their acoustic backscatter signature and classification of wall structures based on transmission loss measurements.
On the distribution of impulsive sound events for environmental noise assessment
Authors: Van den Berg Frank , Van der Eerden Frits
Published: May 2016

The noise levels of impulsive sounds are subject to variation, mostly due to changes in the meteorological situation which have a strong influence on the noise propagation. For environmental noise assessment studies the variation in the single events levels as well as the long term (averaged) level should be considered. To calculate these one can use a number of variations for the atmospheric absorption and the excess attenuation, alongside with histograms of for instance the wind speed and wind directions. Just recently, a new approach has been enforced in the Netherlands to assess the sound from shooting ranges. Details on this calculation method will be given in which meteorological classes are used to account for varying wind speed and temperature gradients. This method makes it also possible to describe the distribution of occurring sound exposure levels around shooting ranges. The distribution of impulsive sounds is further illustrated with data obtained from projects carried out on: long range propagation of high-energy blasts, monitoring military training areas with detonations and muzzle blast noise, and monitoring fireworks in an urban area.
Trans-dimensional Bayesian approaches to room acoustic modal analysis
Authors: Beaton Douglas , Xiang Ning
Published: May 2016

In noise control engineering, it is often desirable to assess the modal behavior of mechanical structures or a room and determine what implications any modes present in the system under investigation will have on noise. The application of Bayesian methods to analyze modal behavior in experimentally measured room impulse responses is of main interest in the current work. These methods extract relevant information from the room impulse response in the time domain and can be used to estimate modal parameters such as mode frequency, amplitude, decay time, and phase. Previous efforts in Bayesian modal analysis divided the analysis into two distinct steps: model selection (determining the appropriate number of modes), and parameter estimation (determining the parameters of each mode). This work considers approaches that combine the two steps into a single trans-dimensional operation. The number of modes in a given model is defined as a parameter of that model. By doing so, the task of model selection becomes part of the parameter estimation, which itself becomes a trans-dimensional problem. Approaches to solve this trans-dimensional problem are applied to both simulated and experimentally measured room impulse responses. Results are compared with other Bayesian approaches and also with conventional Fourier analysis.
Singular acoustics: Transfer of orbital angular momentum to matter
Authors: Wunenburger Régis , Brasselet Etienne
Published: May 2016

A general feature of wave physics is the existence of phase singularities. One usually refers to beams carrying phase singularities as vortex beams, which have become more popular in the field of optics than in acoustics. In particular, the first experimental demonstration of transfer of orbital angular momentum of sound to matter came much later than its optical counterpart. Here, we will review our recent results regarding the transfer of acoustic orbital angular momentum to matter in the ultrasonic domain. This includes (i) the quantitative test of acoustic orbital angular momentum transfer to a sound absorbing object, (ii) the introduction of a novel phenomenon named “rotational acoustic streaming,” and (iii) the demonstration a nondissipative sound-matter orbital angular momentum transfer mediated by chiral scattering.
Acoustic radiation and viscous torque for micromanipulation controlled rotation of particles in fluid cavities
Authors: Lamprecht Andreas , Schwarz Thomas , Wang Jingtao , Dual Jurg
Published: May 2016

Our investigations at ETH Zurich aimed at the theoretical analysis of the acoustic torque and its experimental realization of a controlled rotation of spherical and non-spherical particles by ultrasound. Ultrasonic manipulation of particles provides a contactless handling method for particles suspended in a fluid by acoustic streaming and radiation forces. In addition to the translation of particles in all three spatial directions, particles like functional beads, cells, clumps of cells, fibers, etc., can be rotated. Various methods for the rotation of non-spherical particles were developed with the acoustic radiation torque. The necessary varying pressure field, where the orientation of the nodal pressure lines was controlled by two orthogonal standing waves, was achieved by a modulation of one single parameter over time, e.g., amplitude, phase, and frequency. Stable particle and fiber rotations up to 40 rpm were reached. The rotation can be performed continuously or in a stepwise fashion. Moreover, the rotation of spherical objects was realized by the viscous torque. This torque is formed by acoustic streaming, due to two orthogonal standing waves shifted in phase at the same excitation frequency and amplitude. Angular rotations up to 1200 rpm for spherical 35.5 μm copolymer particles were reached.
Acoustic propagation from a helicoidal wavefront source in an ocean environment
Authors: Hefner Brian T.
Published: May 2016

A helicoidal wavefront source generates a pressure field with a phase dislocation along its axis. About this dislocation, the phase varies linearly with angle while at the dislocation there is a null in the pressure field. The simplest source which can produce this field consists of two crossed dipoles driven 90 degrees out of phase with one another. This phased dipole source was previously used to model propagation from a spiral wavefront beacon in an ocean environment (Hefner and Dzikowicz, JASA, 2012). For the spiral wavefront beacon, the dislocation axis is aligned vertically and the phase variation near the x-y plane is used for underwater navigation. This phased dipole source was shown to be related to the point source through a simple transform which made it possible to transform any point source solution in an ocean environment into the solution for a spiral source in the same environment. This transformation is generalized such that the dislocation axis can be oriented in any direction. This makes it possible to examine helicoidal wavefront propagation in an ocean environment as well. Applications of this general transformation are presented for a helicoidal source near the sea surface and in an ocean waveguide.
Radiation force expansions in terms of partial wave phase shifts for scattering: Examples
Authors: Marston Philip L. , Zhang Likun
Published: May 2016

When evaluating radiation forces on spheres in acoustic beams, with or without helicity, the interpretation of analytical results is greatly simplified by retaining the use of s-function notation for partial-wave coefficients imported into acoustics from quantum scattering theory in the 1970s. This facilitates easy interpretation of various efficiency factors [L. Zhang and P. L. Marston, Phys. Rev. E. 84, 035601 (2011); L. Zhang and P. L. Marston, Bio. Opt. Express 4, 1610–1617 (2013); (E) 4, 2988 (2013)]. For situations in which dissipation is negligible each partial wave s-function becomes characterized by a single parameter: the partial wave phase shift that parameterizes possible degrees of freedom [P. Marston and L. Zhang, J. Acoust. Soc. Am. 131, 3534 (2012); P. L. Marston, J. Quant. Spectrosc. Radiat. Transf. 162, 8–17 (2015)]. The s-function and associated phase shifts are associated with scattering by plane traveling waves, and the incident wavefield of interest is separately parameterized. (When considering allowed outcomes, the method of fabricating symmetric objects having a desirable set of plane-wave scattering partial-wave phase shifts becomes a separate issue.) The present analysis illustrates the advantages of the formulation by extending some prior expansions associated with radiation forces. [Supported in part by ONR.]
Modeling dynamic properties of spontaneous otoacoustic emissions: Low-frequency biasing and entrainment
Authors: Vignali Dario , Elliott Stephen J. , Lineton Ben
Published: May 2016

Spontaneous otoacoustic emissions (SOAE) are sounds generated inside the living cochlea and are regarded as by-products of the active mechanism present in the peripheral auditory system. There is still debate whether these emissions are the product of local oscillators located at various positions along the cochlea (local oscillator model) or if they result from standing waves due to a global collective phenomenon that involves different aspects of the cochlea (global standing wave model). This paper uses a global standing wave cochlear model to predict various features of SOAEs. This involves a state-space formulation with a spatially distributed set of nonlinear active micromechanical elements coupled via cochlear fluid coupling. Simulation results have been compared with available experimental data and demonstrate two interesting nonlinear features of the cochlea: first, nonlinear properties of SOAEs modulated by external low-frequency bias tones are easily predicted and can be used to investigate the plausibility of different nonlinear functions incorporated in the micromechanical elements. Second, entrainment patterns can be obtained when the model is stimulated by a swept-tone: results show distinct areas of beating and others of entrainment between the external stimulus and the SOAE, which depend on the level and instantaneous frequency of the sweep.
Out-of-band beamforming and matched field processing
Authors: Dowling David R. , Worthmann Brian M. , Abadi Shima H.
Published: May 2016

Beamforming and matched field processing (MFP) are array signal-processing techniques for estimating wave propagation directions and source locations, respectively. Both techniques can be formulated as spatial filtering operations that weight and combine the array recordings. In conventional plane- and spherical-wave beamforming, the weights are determined from analytical field models for the corresponding wave type. In conventional MFP, the weights are commonly determined from a computational field model that accounts for known propagation complications (reflection, refraction, scattering, diffraction, etc.) in the acoustic environment. Conventional MFP reduces to conventional beamforming when the environment is simple enough to be described by free-space propagation. Even at high signal-to-noise ratios, both techniques have limitations set by the recorded frequencies, the spacing of the array elements, the geometrical extent of the array, and mismatch between the recorded and modeled acoustic fields. Interestingly, these limitations can be overcome through signal processing techniques that recover out-of-the-signal band information from in-the-signal-band recordings. Here, in-band and out-of-band beamforming and MFP results are illustrated and compared using propagation simulations in free-space and multipath environments, and array recordings from a laboratory water tank and ocean propagation experiments. [Sponsored by the Office of Naval Research and the National Science Foundation.]
Particle-like wave packets in complex scattering systems
Authors: Gérardin Benoît , Laurent Jérôme , Ambichl Philipp , Prada Claire , Rotter Stefan , Aubry Alexandre
Published: May 2016

Waves propagating in complex media typically undergo diffraction and multiple scattering at all the inhomogeneities they encounter. As a consequence, a wave packet suffers from strong temporal and spatial dispersion while propagating through a scattering medium. This wave scattering is often seen as a nightmare in wave physics whether it be for focusing, imaging, or communication purposes. Controlling wave propagation through complex systems is thus of fundamental interest in many areas, ranging from optics or acoustics to medical imaging or telecommunications. Here, we study the propagation of elastic waves in a cavity and a disordered waveguide by means of laser interferometry. We demonstrate how the direct experimental access to the information stored in the scattering matrix of these systems allows us to selectively excite stationary scattering states and wave packets that follow particle-like bouncing patterns in transmission through (or in reflection from) a complex scattering landscape. Due to their limited dispersion, these particle-like scattering states will be crucially relevant for all applications involving selective wave focusing and efficient information transfer through complex media.
Localization of aeroacoustic sources by using time-reversal and beamforming techniques
Authors: Marx David , Valeau Vincent , Prax Christian , Brizzi Laurent-Emmanuel
Published: May 2016

The study deals with the localization of aeroacoustic sources in flows by using array processing techniques. In such situations, the source is located in the flow and the propagation model must properly take into account flow effects, with possible temperature gradients. The resolution of the inverse problem can rely: (i) either on the numerical simulation of the propagation by using the time-reversal principle; (ii) or on using the beamforming technique, provided a high-frequency model of the Green function is used, such as that given by a simplified analytical model or by ray-tracing. The advantages and links between the two approaches are discussed. Some comparisons in terms of localization error are then provided, based on simulated data, both in the time and frequency domains. The case of a pulse in a shear flow is considered, showing similar performances for both methods. Then a harmonic source in a shear flow and in a jet flow is studied, with possible temperature gradients, showing the limitations of the beamforming method compared to the time-reversal-based method. Finally, examples of applications to experimental or numerical data are discussed.