bluuebunny/crossref_metadata_2025 huggingface dataset:
Noise impact assessment on residential area near highway and noise control measures development
Authors: Tsukernikov Ilya E. , Shubin Igor L. , Tikhomirov Leonid A. , Schurova Natalia E. , Tsukernikov Ilya O.
Published: November 2016

Currently, from 30 to 50% of urban population of the Russian Federation it is affected by the increased traffic noise. A similar situation is observed not only in big cities, but also in rather small settlements. Instrumental and calculated assessment of existing noise levels in the residential area “Zarechie” adjacent to a site of the Moscow Ring Road and having a soundproofing barrier on the border of the territory is performed in the paper. Acoustic situation on the territory without existing soundproofing barrier is modeled with use of the program “ARM Acoustics,” developed by Russian Company “TECHNOPROEKT” and the effectiveness of the noise reducing by the screen is evaluated. A calculation for feasibility of a further increase in the height of the existing barrier is carried out. Noise maps are built which allowed to identify the main sources of acoustic discomfort in the residential area. The analysis of the impact on the acoustic environment of the building being built is conducted. Against this background options for protecting the area from noise are proposed.
Wideband audio systems for hearing aid devices
Authors: Salameh Michael
Published: November 2016

Multi-loudspeaker design has been one of the most widely used methods for improving the sound quality of audio systems. Some of the electroacoustic design concepts defined for typical earphones can be used for other wideband miniature audio systems as well. However, wideband audio for hearing aid devices and smart earphones introduce new design factors. Efficiency and transducer size are among the factors that become important in the electroacoustic design. In this paper, the electroacoustic design factors for woofer-tweeter audio systems are revisited and discussed for hearing aid devices. Simulation and measurement results are presented and analyzed.
The effect of adaptive music playing system on emotion regulation
Authors: Wang Wei-Chun
Published: November 2016

We live in an industrially polluted environment, surrounded with stress and negative emotions. Since music has the marvelous power of arousing different moods and regulating emotions depending on the features of musical elements, people could be prescribed specific kinds of music to help alleviate their symptoms. Therefore, refer to entrainment, iso, and diversion principles of music therapy stated by Campbell, the investigator developed an Adaptive Music Playing System base on the two-dimensional arousal-valence emotion model. The purposes of this study were: (1) to explore the two-dimensional emotional responses of subjects when listening to specific music; (2) to survey the effects of gender, music preference on their perception of musical emotions; (3) to compare the relativities of various musical elements with musical emotions; (4) to detect the healing effect of adaptive music playing system. 56 college students were recruited as subjects in this study. Subjects imported their instant mood value, and the Adaptive Music Playing System would automatically play three music pieces. The results of this study showed that Adaptive music playing system could successfully lead the subjects in a better physical and mental condition. Gender effect wasn’t significant for subjects’ perception. Music preference affected subjects’ perception.
Vocal fry in realistic speech: Acoustic characteristics and perceptions of vocal fry in spontaneously produced and read speech
Authors: Irons Sarah T. , Alexander Jessica E.
Published: November 2016

There has been a great deal of recent interest in vocal fry, both in production and perception. However, much of the scientific literature that has used naturally produced fry has focused on speech elicited through reading, rather than spontaneous speech. The current study compares reading with spontaneous speech elicited in various ways for both male and female speakers, recorded in dyads. The speakers were asked to teach their partner information, give instructions, and describe their qualifications for a scholarship, with reading controls for each task. Surprisingly, we found more vocal fry (in both proportion of words and words/duration) for men than women. Men differed in the amount of fry across task types. Interactions of speaker sex, task type, and fry were examined for measures of F0, pitch range, intensity level, jitter, shimmer, and HNR. Listener perceptions of speech with and without vocal fry were also obtained. Task type affected listener perceptions of speech with vocal fry. Vocal fry in spontaneously produced speech seems to differ from fry in speech produced during reading, in quantity, acoustics, and listener perceptions. Additionally, acoustic measures suggest that vocal fry may be intimately tied to decreased vocal effort across task types and speaker sex.
Natural language dialog system considering speaker’s emotion for open-ended conversation
Authors: Takahashi Takumi , Mera Kazuya , Kurosawa Yoshiaki , Takezawa Toshiyuki
Published: November 2016

To respond appropriately to an utterance, human-like communication system, should consider not only words in the utterance but also the speaker’s emotion. We thus proposed a natural language dialog system that can estimate the user’s emotion from utterances and respond on the basis of the estimated emotion. To estimate a speaker’s emotion (positive, negative, or neutral), 384 acoustic features extracted from an utterance are utilized by a Support Vector Machine (SVM). Artificial Intelligence Markup Language (AIML)-based response generating rules are expanded so that the speaker’s emotion can be considered as a condition of these rules. Two experiments were carried out to compare impressions of a dialog agent that considered emotion (proposed system) with those of an agent that did not (previous system). In the first experiment, 10 subjects evaluated the impressions after watch four conversation videos (no emotion estimation, correct emotion estimation, inadequate emotion estimation, and imperfect emotion estimation). In the second experiment, another 10 subjects evaluated the impressions after talk with both dialog agents. These experimental results and a demonstration of the proposed system will be shown in the presentation.
Universal vs. language-specific aspects in human vocal attractiveness: An investigation towards Japanese native listeners’ perceptual pattern
Authors: Xu Anqi , Leung Shue-sum
Published: November 2016

Studies on Western societies show that male voices with acoustic parameters encoding a big body size (low F0, narrow formant dispersion, and F0 range) were considered to be attractive, while the opposite was true for female voices (e.g., [Xu et al., 2013, PLoS ONE, 8(4), e62397]). The present work investigates whether Japanese native listeners are guided by the same principles in assessing the voices of the opposite sex. We replicated the design in Xu et al. (2013) with the added parameter of creaky voice, which is prevalent in North America nowadays and hotly debated in terms of attractiveness. Thirty-four heterosexual participants (16 female) rated the attractiveness of synthetic stimuli controlling for F0 height, formant distribution, F0 range and voice quality. Results indicate that their preferences for voice quality are similar with studies on Western societies (breathy > modal >creaky >pressed/tensed). Additionally, low-pitched male voice with narrow formant dispersion and high-pitched female voices were also favorable. Interestingly, a narrow F0 range significantly lowered the attractiveness ratings, regardless of the gender of the voice, which contradicts Xu et al. (2013). These various results are discussed in light of the cross-linguistic/cross-ethnic divergences in vocal attractiveness.
Constructing boyishness: A sociophonetic analysis of Japanese anime voice actresses performing male roles
Authors: Redmond Ryan C.
Published: November 2016

The present study investigates the performance of different gendered roles by Japanese female seiyuu [voice actors] through analysis of fundamental frequency (f0). Ten voice actresses who have been known for commonly voicing male characters were sampled for analysis. For each actress, two characters from their repertoire were chosen (one male, one female), and from each character 75 intonation phrases were sampled, for a total of 1500 utterances. For each of these samples, pitch data were extracted and analyzed. Results for both male and female role performances display an average f0 that generally falls inside the category previously defined as hegemonically “feminine” in Japanese. While male role measurements were on average lower than female role measurements, only occasionally did any actress produce pitches in the average expected range for a Japanese man. However, because the female role measurements were significantly higher than that of average Japanese women, the differences between the “male role” and “female role” categories proved statistically significant in terms of minimum pitch, maximum pitch, mean pitch, and pitch range. A theory is proposed that the hyper-femininity of many female roles is what necessitates this difference.
Comparisons of conventional and frequency-difference beamforming
Authors: Douglass Alexander S. , Dowling David R.
Published: November 2016

Conventional beamforming can isolate ray arrival directions from transducer array recordings, provided the array element spacing does not significantly exceed half of a signal wavelength. However, when the signal frequency is high and the array elements are many wavelengths apart, conventional beamforming may break down due to spatial aliasing. Frequency difference beamforming avoids this limitation by utilizing a quadratic product of complex signal amplitudes at different frequencies to extract wave propagation information at a difference frequency chosen low enough to avoid aliasing. For this presentation, the performance of frequency difference beamforming was evaluated two ways. First, simulations of acoustic propagation in a 106-m-deep shallow-ocean sound channel using the BELLHOP ray-tracing code were completed with frequency-sweep signals from 11 kHz to 33 kHz and 1100 Hz to 3300 Hz broadcast from a single source to a 56-m-long 16-element vertical array. Second, lab experiments in a 1.07-m-diameter 0.8-m-deep cylindrical water tank were conducted with pulse signals having center-frequencies of 165 kHz and 15 kHz broadcast from a single projector to a 0.7-m-long 14-element horizontal array. In both cases, frequency difference beamforming utilizing the higher frequency signal produced results comparable to conventional methods applied to the lower frequency signal. [Sponsored by ONR.]
Application of acoustic contrast control for out door music performance: Feasibility study and experiments
Authors: Kim Yang-Hann , Kim Wanjung , Kim Hwan , Kim Jueseok , Kim Sang-Hyeon , Choi Jungwoo , Lee Jong-Hwa
Published: November 2016

Ideal outdoor music hall will be what has invisible but acoustically hard wall. This rather impossible objective can be achieved, at least, in some extent, for example allowing quite acceptable sound level in certain zone or zones where most residence lives and hear unwanted sound. This challenging problem can be tackled by finding the solutions that determines the magnitudes and phases of speakers, which surround the ideal hall's wall, that minimizes acoustic contrast between the listeners zone (bright zone) and residence zone (dark zone). It is interesting to find that acoustic contrast control essentially builds up acoustically hard wall, in other words, it makes large impedance mismatch on the wall. This bright observation, however, has been essentially based on two dimensional simulation and experiment. Next step has to go how well this can be extended to general 3D cases. We attempted to have another layer of speaker arrays, and requiring bright zone inside of the double array with certain height, therefore making a cylindrical volume of bright zone, and dark zone having certain angle, between two lines: one connects between the center and the bottom array and another one is the line between the center and upper array, rotating these two lines makes a diverging volume outside of the wall that has certain angle. Simulations and experiments demonstrated how well it can be practically applied and what are its limitations.
A nested beamforming approach to sound source separation using an array-of-arrays recording system
Authors: Trevino Lopez Jorge A. , Sakamoto Shuichi , Suzuki Yôiti
Published: November 2016

Beamforming is a technique used with microphone arrays to selectively isolate sounds according to their direction of incidence. However, conventional methods struggle to separate sound sources if the spatial resolution of the recording system is not enough to define a recording zone containing only the target. In this presentation, a sound source separation method using multiple beamforming arrays is introduced. The proposal consists of three stages. The first stage applies modal beamforming methods to individual spherical microphone arrays; this results in a virtual array of directional microphones. The second stage distributes the recordings from the virtual microphones onto a new angular coordinate system. This stage transforms the radial coordinates for the different sound sources into pairs of parallax angles between them and the spherical microphone arrays. Finally, the third stage uses a least-squares optimized beamforming method defined on the parallax-angle coordinate system. The combination of these three stages takes advantage of parallax differences between spherical microphone arrays and allows for the isolation of sound sources at specified positions. The new method is evaluated using a computer simulation of two 64-channel spherical microphone arrays. The proposal shows satisfactory results even in the presence of aligned interfering sources, which hinder separation using conventional beamforming methods.
ACOustic keyhole
Authors: Butler Rhett , Duennebier Frederick K. , Howe Bruce M.
Published: November 2016

The Aloha Cabled Observatory (ACO) is located on the seafloor about 100 km north of O‘ahu at 4800 m depth. Seismoacoustic T-waves generated by earthquakes in the South Pacific from Tonga-Kermadec to the Solomon Islands travel 40-70°Δ, passing through an acoustic keyhole—the Ka'ie'ie Waho channel—between O‘ahu and Kaua‘i before reaching ACO. In contrast to ACO and the Pacific seafloor at depths greater than 4500 m, the channel shallows to 1 and 3 km in depth, blocking deeper SOFAR ray/mode paths. T-waves from >50 earthquakes (Mw 6.1 to 8) are reviewed, correcting apparent velocities for near-source P-wave path effects. We examine T-waves crossing the Ka'ie'ie Waho channel and descending to ACO, comparing and contrasting with T-waves propagating unhindered across the North Pacific from events in offshore Japan (Tohoku aftershocks), Kuril Islands, Aleutian Islands, and offshore Alaska. Preliminary review indicates that the T-waves exhibit seismoacoustic modal-coupling to the seafloor, exhibiting a structure observed previously* at the Hawai‘i-2 Observatory (H2O) at 5 km depth between Hawai‘i and California. *R. Butler (2006), J. Acoust. Soc. Am. 120(6), 3599-3606. *R. Butler and C. Lomnitz (2002), Geophys. Res. Lett. 29(10), 57 1-4.
Influence of source frequency spreading on transmission loss patterns
Authors: Brown Elisabeth M. , Pierce Allan D. , Siegmann William L.
Published: November 2016

A classical experiment [Wood and Weston, Acustica 14 (1964)], conducted in Emsworth Harbor at low tide, examined propagation in a 1 m mud layer overlying gravel. Hydrophones on the mud bottom transmitted six source frequencies between 4 and 72 kHz and received signals at ranges up to 50 m. Signal levels show modal fluctuations at lower frequencies but not at higher frequencies, even though the authors did not perform data smoothing. This behavior is hypothesized to arise from a frequency-dependent spread of transmitted frequencies. A Pekeris two-layer model of mud and gravel with a pressure release mud surface is used to idealize the waveguide. Transmission loss is calculated using a weighted frequency average of time-averaged intensity, with a Lorentzian function (Cauchy distribution) modeling the frequency spread. Appropriate values for the Q-factor of the distribution are examined. Compressional wave attenuation values from a recent inversion of the data [Pierce, et al., POMA, accepted (2015)] are incorporated into the calculations. Computational results demonstrate the same behavior in intensity fluctuations as in the data with identifiable modal interference at the three lower frequencies and smoothed patterns at the three higher. [Work supported by the ONR.]
Analytical formulas for incoherent transmission loss in shallow water based on effective approximations of seafloor depth and reflectivity
Authors: Zhang Zhi Y.
Published: November 2016

We derive analytical formulas for modelling incoherently averaged transmission losses in shallow water environments by taking into account the effects of energy spreading, in-water absorption, sea surface scattering, and leakage into the seafloor. The formulas are based on the effective depth approximation to include the effect of sound penetration into the seafloor and use effective two-parameter representations of seafloor reflectivity. Comparison with published measurement data at several shallow water sites worldwide shows that the formulas are capable of reproducing the main features of shallow water sound propagation in environments with different types of sediments. The formulas are useful for quick assessments of incoherent transmission losses in generic yet representative underwater environments where detailed environmental knowledge is lacking or the fine details of underwater sound propagation is not required.
Modeling surface scatter in the channel impulse response with a Helmholtz-Kirchhoff integral
Authors: Richards Edward , Hodgkiss William , Song Heechun
Published: November 2016

Several experimental measurements of channel impulse responses (CIRs) using underwater communication bandwidths have shown distinct striation features arriving up to 10 milliseconds after the main surface bounce, e.g. Badiely et al. [J. Acoust. Soc. Am. 132, EL290 (2012)]. Choo et al. [J. Acoust. Soc. Am. 136, 1046 (2014)] used an intuitive geometric ray model to argue that the source of these arrivals was the reflection from a wave trough near the receiver. This presentation extends Choo's approach and focuses on a quadrature solution of the time domain Helmholtz-Kirchhoff (HK) integral in a half space. This approach provides the same intuitive geometric ray perspective on the surface scattering, but additionally models arrival amplitudes near caustics and higher order catastrophe phenomena induced by surface waves. It is shown that a simple shadowing correction of the HK result produces all of the major features of the full Helmholtz scattering integral solution. The HK integral is computationally tractable, accurate, and can show the path from the surface that contributes to each CIR arrival.
The optimal weight vectors research on the far-field sound field control of the low-frequency phased array
Authors: Lee Haifeng , Ma Li
Published: November 2016

The theory and simulation studies were developed to analyze the far-field sound field control effect of the low-frequency phased array emitting the different weight vectors. First, the relationship between the weight vector and the mean square deviation of time-delay estimation were set up theoretically. Second, using the relationship, the optimal weight vector was solved and discussed. Finally, under the same energy input conditions, the several weight vectors were used to compare the far-field sound field control effects of low-frequency phased array. From the simulating results and the mean square deviation of time-delay estimation exited by the weight vectors, it was showed and proved that the phased array emitting the No.1 normal mode was not always the optimum in the far-field sound field control.
Detection and communication in the “Beaufort Lens”
Authors: Baggeroer Arthur B. , Schmidt Henrik , Carper Scott A. , Collis Jon , Rosenthal Jordan
Published: November 2016

The WHOI Ice Tethered Profilers in the Beaufort has confirmed a “sound speed duct” at depths between 100 to 250 m labeled as the “Beaufort Lens.” It is thought to be caused by warm water intrusion from the Bering Strait. A significant consequence of the duct is the prediction of 10 dB lower transmission losses at ranges of 100 km for sources and receivers both within the duct. We examine how the properties of the duct impact detection and communication. Transmission loss is strongly affected by frequency. Ducted propagation is not supported below a modal cutoff and absorption losses become consequential at higher frequencies. For active sonar, where wideband waveforms are employed, dispersion effects and multipath become important. The shape of the duct, especially boundary gradients, impacts this dispersion. While a vertical line array spanning the duct can potentially exploit the multipath ray/mode coherence and the resolution of the array impacts whether coherent or incoherent combination should be employed to achieve desired gains. Data for validating and verifying long range transmission loss in the duct were acquired by two submarines participating in ICEX-16. This presentation examines the signal processing for detection and communication within the so called “Beaufort Lens.”
The behavioral response of humpback whales to seismic air gun noise
Authors: Dunlop Rebecca , Noad Michael J. , McCauley Robert , Cato Douglas
Published: November 2016

Four major experiments have been conducted off Australia to quantify the behavioral response of migrating humpback whales to various seismic air gun arrays. The first, using a 20 in3 air gun, was used to develop the analysis framework, which was then applied to later experiments. The following two experiments tested a 4-step “ramp-up” procedure (20, 60, 140, and 440 in3 array sequence) and a 140 in3 array. Both studies found a change in movement behavior in response to the air guns, where groups deviated more from their course and made less progress towards the source. There was no evidence of any change in surface behaviors, including respiration rates. The final experiment involved ramp-up to a 3130 in full commercial array. In response, whales decreased their dive time and displayed an elevated respiration rate, more so to the full array phase. In this phase, they also changed their surface behavior, in that breaching rates were elevated but tail and pectoral slapping behaviors were reduced. Consistent with previous studies, whales also changed their movement behavior, more so during ramp-up. Therefore the full array elicited a greater variety and magnitude of behavioral changes than observed with the smaller air gun arrays.
Inter and intra specific variation in echolocation signals among cetacean species in Hawaii, the northwest Atlantic and the temperate specific
Authors: Yack Tina M. , Dunleavy Kerry , Oswald Julie N. , Cholewiak Danielle
Published: November 2016

Odontocete species use echolocation signals (clicks) to forage and navigate. The aim of this study is to explore inter- and intra-specific variation in clicks among odontocete species in the Northwest Atlantic, Temperate Pacific, and Hawaii. Clicks were examined for seven species of delphinids in the western North Atlantic; common dolphin, Risso’s dolphin, pilot whale, rough-toothed dolphin, striped dolphin, Atlantic spotted dolphin, and bottlenose dolphin. Newly developed PAMGuard tools were used to automatically measure a suite of click parameters. Five parameters were compared between species; duration, center frequency, peak frequency, sweep rate, and number of zero crossings. Significant differences in duration, center and peak frequency were evident between species within regions (Dunn’s test with Bonferroni adjustment p<0.05). Geographic variation in click parameters between the three study regions was compared for five species; bottlenose dolphin, common dolphin, striped dolphin, pilot whale, and Cuvier’s beaked whale. Significant differences in several parameters were found for all species between the regions (Dunn’s test with Bonferroni adjustment p<0.05). These results suggest that there are species specific differences in clicks among delphinids and that geographic variation exists for multiple species. The ecological significance of these findings will be discussed along with implications for classifier development.
Determining killer whale (Orcinus orca) call variability from passive acoustic monitoring in the Chukchi and Bering Sea, Alaska
Authors: Madrigal Brijonnay C. , Berchok Catherine L. , Crance Jessica L. , Stimpert Alison K.
Published: November 2016

Killer whales (Orcinus orca) are a highly vocal species that produce three types of vocalizations; pulsed calls, whistles, and clicks. Unlike the Northern and Southern Resident populations of western Canada and the Pacific Northwest, little is known regarding the acoustic behavior of resident and transient killer whale populations north of the Aleutian Islands in the Bering and Chukchi Seas. Acoustic data were analyzed from moored recorders deployed by the Marine Mammal Laboratory at two sites each in the Bering and Chukchi Seas (BOEM-funded). The recorders sampled at 4 kHz on a 7% duty cycle (Bering) or 16 kHz on a 28% duty cycle (Chukchi). Over 1100 calls were identified, and discrete call classification was conducted using an alphanumerical system that distinguished calls by location, general contour, and segment variation. Parameters analyzed included call duration, start/end frequency, and delta frequency/time; periods of call repetition were common. These results will help determine if resident populations occur in the Chukchi Sea, and identify which transient populations are present. This initial work classifying killer whale sounds in the Arctic and along the Bering Sea shelf will also facilitate comparisons of call types within and among transient and resident killer whale populations.
Creation of an acoustic dictionary to classify the vocal repertoire of humpback whale song
Authors: Allen Jennifer , Garland Ellen , Murray Anita , Noad Michael , Dunlop Rebecca
Published: November 2016

Animal vocal repertoires can be varied and complex, and it is often difficult to obtain a comprehensive picture. One of the primary reasons for this difficulty stems from the classification of acoustic signals, which has typically relied on qualitative and subjective methods. Vocal signals (units) were measured from east Australian humpback whale song from 2002-2014 and classified using self-organising maps (SOM). Humpback song was used here to provide an example of a particularly complex display of vocal signals. The use of SOM methodology takes an important step towards a more objective classification system. The resulting classifications were used to create a set of 149 hypothetical “exemplar” units to serve as an acoustic dictionary, representing the song’s repertoire. The intent is to use this dictionary to determine vocal repertoire, as well as to facilitate the transcription and sequential analysis of large numbers of recordings with only a small representative subset of units requiring intensive acoustic feature measurement. Acoustic dictionaries are integral to studies of information content, microstructure, and syntax, and this methodology is thus applicable across the vocalization research community.
Signal processing for optical sound field measurement and visualization
Authors: Yatabe Kohei , Ishikawa Kenji , Oikawa Yasuhiro
Published: November 2016

Accurately measuring sound pressure is not an easy task because every microphone has its own mechanical and electrical characteristics. Moreover, the existence of a measuring instrument inside the field causes reflection and diffraction which deform the wavefront of sound to be measured. Ideally, a sensing device should not have any characteristic nor exist inside a measuring region. Although it may sound unrealistic, optical measurement methods are able to realize such ideal situation. Optical devices can be placed outside the sound field, and some of the sensing techniques, which decode information of sound from the phase of light, are able to cancel optical and electrical characteristics. Thus, optical sound measurement methods have possibility of achieving higher accuracy than ordinary sound measurement in principle. However, they have two main drawbacks that have prevented their applications in acoustics: (1) point-wise information cannot be obtained directly because observed signal is spatially integrated along the optical path; and (2) increasing signal-to-noise ratio is difficult because optical measurement of less than a nanometer order is typically required. To overcome the above difficulties, we have proposed several signal processing methods. Here, those methods are introduced with the physical principle of optical sound measurement.
A trial on the measurement of the acoustic properties by using a parametric loudspeaker
Authors: Sugahara Akiko , Lee Hyojin , Sakamoto Shinichi , Takeoka Shigeto
Published: November 2016

Measurement of acoustic properties of materials, such as absorption and reflection coefficient in a free field, has the results largely affected by diffraction by the sample edge. In order to minimize this, the use of a parametric loudspeaker, which enables to produce a directional audible sound by using the nonlinear effect of sound beams based on the parametric array theory, is investigated. At first, the directivity and the distance attenuation were measured as fundamental characteristics of the sound source. Then, the directivity of reflected sound from a surface, flat and diffuser plate, under oblique incidence conditions was measured in an anechoic chamber. The data were compared with those obtained by an experiment using a conventional loudspeaker, and with a wave-based numerical analysis using the mode matching method. The results show that the narrow sound directivity of the parametric loudspeaker is capable of obtaining the directivity properties of reflection sound from the plates more accurately than a conventional loudspeaker.
Simulation of shear wave elastography imaging using the toolbox “k-wave”
Authors: Prieur Fabrice , Treeby Bradley , Catheline Stefan
Published: November 2016

Elastography is used to map the local elasticity of tissue. It can detect areas inside the body with a different elasticity from that of surrounding tissue indicative of pathologies like tumors. Shear wave elastography imaging produces an elasticity map by analyzing the propagation of shear waves. One source of shear wave is the acoustic radiation force produced by ultrasound. The prediction of the acoustic radiation force and of the shape and amplitude of the ensuing shear displacement is crucial. In this study, we present simulations of the radiation force produced by an ultrasound transducer and of the shear displacement it produces using the software package “k-wave.” Results from simulations in a homogeneous and isotropic medium are compared against analytical solutions and results from a finite element modeling software. The obtained shear displacements from k-wave are very similar to the analytical solution with a root mean square error around the focal zone below 9% and 21% for short and large propagation times, respectively. K-wave appears to be an accurate and efficient tool for simulation of acoustic radiation force and shear wave propagation. It combines the simplicity of finite time differences methods with the flexibility to simulate in any heterogeneous medium.
3-D ultrafast ultrasound strain imaging to improve breast cancer detection
Authors: Hendriks Gijs A. , Chen Chuan , Hansen Hendrik H. , de Korte Chris L.
Published: November 2016

The automated breast volume scanner (ABVS) is used for breast cancer detection and consists of a linear transducer translating over the breast while collecting ultrasound data to reconstruct a breast volume. Although the ABVS shows high sensitivity, specificity remains a limitation resulting in high recall-rates. To improve ABVS specificity, we verified if it is feasible to implement 3-D strain imaging to discriminate between malignant (firmly-bound and stiff) and benign lesions (loosely bound and less stiff) based on lesion-to-surrounding-tissue connectivity (shear strain) and on stiffness (axial strain). Three phantoms containing loosely- (1) and firmly-bound (2) three times stiffer lesions were scanned twice by the ABVS, pre- and post-deformation (0.5 mm). Data was collected by plane-wave imaging to reduce acquisition times below one breath-hold. Displacements were calculated by cross-correlation from which axial and octahedral-shear strains were derived. Shear strains were distributed tightly and globally around the lesion for phantom (1) and (2), respectively. Using axial strain, the lesions could be clearly differentiated from the surrounding tissue in both phantoms (CNRe 30; SNRe 19dB). Therefore, we can conclude that it is feasible to implement 3-D strain imaging in the ABVS, and to improve its specificity by discriminating lesions by elastic properties.
Acoustic field measurement of high-intensity ultrasound source system using acoustic waveguide for calibration of hydrophone
Authors: Igarashi Shigeru , Morishita Takeshi , Takeuchi Shinichi
Published: November 2016

In recent years, high-intensity ultrasound equipments have been used. It is expected that nonlinear calibration and evaluation will be needed more often. It is necessary to develop a high-intensity ultrasound source that transmit a plane wave in far-field for calibration of hydrophone in order to evaluate high intensity acoustic field. We proposed an ultrasound source system using a cylindrical type acoustic waveguide and a concave type piezoelectric transducer. The focused ultrasound wave is emitted from the concave type transducer, propagated through the acoustic waveguide, and the output from that forms high-intensity acoustic beam. We simulated the output acoustic pressure of the proposed source system using axisymmetric 3D acoustic field simulation based finite element method, compared with that of a flat disk type transducer with the same aperture size as the acoustic waveguide, and the simulated data showed the acoustic pressure of our proposed system was higher than that of the flat disk type transducer with the same main beam width. We fabricated the experimental source system using the concave type piezoelectric transducer and the acoustic waveguide using closed cell sponge, and measured acoustic pressure distribution. The obtained results were nearly coincident with the simulated data
Battery outage and portable video recorders
Authors: Yonovitz Al , Yonovitz Joshua , Joe Herbert , Galvin Brittany
Published: November 2016

The advent of user-friendly compact video recorders with large storage capacities has made them ideal for recording investigations and interviews by law enforcement agencies. In addition, they are also useful for surreptitious recordings both by citizens and government services. When these recorders are powered by an internal battery, the battery recording may be interrupted when the battery discharges. The analysis of this premature stoppage of the recorded video file may be important in the determination of the authenticity of the recording for courtroom admissibility. The differentiation between a purposeful edit and the normal shutdown process was investigated with a series of such recordings using a variety of modern recorders and common smartphones. Recordings were made with the battery at 80 percent charge and allowed to deplete. The recorder was then immediately restarted as well as restarted while being immediately attached to permanent power. The resultant recordings were compared and analyzed for any extraneous information related to the shutdown process.
Omni-binaural recording system for 360° video
Authors: Nishino Takanori , Niwa Kenta
Published: November 2016

A binaural recording system for 360° video is developed and evaluated. The 360° images can be stitched by using multiple cameras. A binaural recording generally uses a dummy head microphone or binaural recording microphones; however, the orientation of the dummy head needs to be changed depending on the selected view. Binaural recording corresponding to multiple orientations of a dummy head is very difficult and sometimes impossible. Therefore, we propose a binaural recording system that can simultaneously record the sound for the entire corresponding 360° video. The proposed equipment is cylindrical in shape and has four hollows that correspond to the cavity of the concha on the side. Binaural signals corresponding to a desired direction are generated by a weighted average of the recorded signals. Three types of weights, i.e., linear, spline, and optimum weights, were examined. The degree of suitability between the video and generated binaural signals was evaluated using subjective tests. Nine subjects answered that all the weight conditions were acceptable and the proposed system was very interesting and entertaining.
Deep neural networks for learning classification features and generative models from synthetic aperture sonar big data
Authors: Chen Johnny L. , Summers Jason E.
Published: November 2016

Autonomous synthetic aperture sonar (SAS) imaging by unmanned underwater vehicles (UUVs) provides an abundance of high-resolution acoustic imagery useful for studying the seafloor and identifying targets of interest (e.g., unexploded ordnance or mines). Unaided manual processing is cumbersome as the amount of data gathered by UUVs can be enormous. Computer-vision and machine-learning techniques have helped to automate classification and object-recognition tasks, but often rely on hand-built features that fail to generalize. Deep-learning algorithms facilitated by emergence of graphics-processing unit (GPU) hardware and highly optimized neural-network implementations have recently enabled great improvements in computer vision. Autoencoders allow for deep unsupervised learning of features based on a reconstruction objective. Here, we present unsupervised feature learning applied to seafloor classification of SAS images. Deep architectures are also capable generative models. We illustrate this with generative networks that are capable of generating realistic SAS images of different seafloor bottom types. Deep models allow us to construct algorithms that learn hierarchical and higher-order SAS features, which promise to improve automatic target recognition (ATR) and aid operators in processing the large data volumes generated by UUV based SAS imaging. [Work supported by the Office of Naval Research.]
Acoustic and perceptual differences between novice and professional music theatre singers
Authors: Maxfield Lynn M. , Manternach Brian
Published: November 2016

Research examining contemporary commercial music (CCM) styles of singing has increased significantly over the last ten years. While acoustic analysis has helped define which characteristics define various vocal genres, discrepancy still exists in how those acoustic characteristics are perceived, described, and evaluated. The current study recorded novice and professional musical theatre singers performing belt, legit, and mix vocal samples. Three acoustic analyses were applied to the excerpted recordings from each singer using the Praat voice and speech analysis software: The spectral slope of the long term average spectrum (LTAS) was calculated, as well as the noise to harmonic ratio (NHR), and the dominant harmonic (1st, 2nd, or 3rd) was noted for each sample. Results were compared across pitch, style (belt v. legit), and training level (professional v. novice). Finally, raters listened to the recordings and rated each sample on the basis of style (belt v. legit), roughness (rough v smooth), and tone quality (brassy v flutey, and bright v dark). A 120mm liechert scale was provided to each rater for each of the rating variables. Results of the perceptual assessments were compared with acoustic measures to elucidate discrepancies in how ccm characteristics are perceived.
Vowel duration dependent hidden Markov model for automatic lyrics recognition
Authors: Awata Shohei , Sako Shinji , Kitamura Tadashi
Published: November 2016

Recently, due to the spread of music distribution service, a large amount of music is available on the Internet. Accordingly, it is generally increasing the demand of music information retrieval (MIR). In the field of MIR research, there are several researches to extract meaningful information from music audio signals. However, automatic lyrics recognition is still a challenging problem because the variation of singing voice is much larger than that of speaking voice and a large database of singing voice is not available. In the relevant study, lyrics recognition was performed by extending the framework of speech recognition using hidden Markov model (HMM). However, accuracy rate was not sufficient. To recognize singing voice precisely, one promising approach is utilizing musical features. This study considers the task of recognizing syllable from a cappella singing voice. To respond to the variation of the length of a phoneme, we construct the duration dependent HMM. A large database of singing voice is essential for training the acoustic model. We use synthetic singing voice by HMM based singing voice synthesis system to solve the lack of the database of a cappella singing voice. We confirmed the effectiveness of our method.
Differences of characteristics of music singing between in usual Karaoke and in Hitokara
Authors: Matsumoto Junko
Published: November 2016

Recently some of people in Japan go to Karaoke by oneself. This is called Hitokara (solo Karaoke). We investigated the differences of characteristics of music singing between in usual Karaoke and in Hitokara. Participants were 103 university students. They were asked to fulfill a questionnaire about characteristics of music singing both in usual Karaoke and in Hitkoara. As a result, about one third of participants had at least once experience of Hitokara singing, and they preferred both usual Karaoke singing and Hitokara singing. However, they preferred the music with characteristics of elation or lightness in usual Karaoke singing, in Hitokara singing they preferred the music with various characteristics. In Hitokara singing, people sing actively and they would be able to cope well with various moods by singing various music.
An electromyographic study of left hand between experts and novices in violin playing
Authors: Obata Satoshi , Aiba Eriko
Published: November 2016

We recently recorded string clamping force on the violin during simple (without vibrato) tone production by expert violinists. The results of our studies showed that the peak force exceeded 4.5 N at slow tempi, which decreased to 1.7 N at fast tempi. However, subjective assessment of playing effort indicated an opposite trend; the players felt that playing at faster tempi was more strenuous. We also measured muscular activity along with finger kinematics. We found that the mean left forearm muscle activity increased with tempo. The findings suggest that the speed of finger movement is more related to the arm and hand discomfort when playing a violin, as compared to clamping force. In the present study, we investigated the expert-novice difference in the nature of the string clamping force and related muscles in the left hand and arm while playing a violin. Information regarding the string clamping force and left hand muscle activity can help in teaching novices about the magnitude and timing of the appropriate string stabilizing action during sound production. Surface electromyography (EMG) and string clamping force were measured for this purpose.
Level set-based topology optimization for the design of acoustic metamaterial using two-phase material model
Authors: Noguchi Yuki , Yamamoto Takashi , Yamada Takayuki , Izui Kazuhiro , Nishiwaki Shinji
Published: November 2016

Acoustic metamaterials are artificial materials composed of periodic structures made of elastic and fluid material. They could be used for noise reduction problems and efficient sound propagation control by their unusual properties, such as negative bulk modulus, mass density, and refractive index. The performance of acoustic metamaterial is strongly affected by its designs rather than its material properties. Therefore, topology optimization method could be used to obtain a desired property of acoustic metamaterial. To model acoustic-elastic coupled system, we introduce a two-phase material model where the solid and fluid phases are mixed. While coupling boundary conditions are required in a usual acoustic-elastic coupled analysis, this model allows us to analyze the system without imposing such boundary conditions since their conditions are satisfied naturally. In this research, we propose a level set-based topology optimization method for the design of acoustic metamaterial. First, we introduce the two-phase material model and level set-based topology optimization method. An optimization problem for an acoustic metamaterial that has negative mass density is formulated and sensitivity analysis is performed based on the concept of topological derivative. After verifying the topological derivative by comparing it to numerical differences, we present two-dimensional numerical examples for negative mass acoustic metamaterial.
An application of a parametric transducer to measure the acoustical properties of a living green wall
Authors: Romanova Anna , Horoshenkov Kirill V.
Published: November 2016

Greening of urban spaces provides a number of environmental benefits. Living Green Walls (LGW) is a most typical example of greening which is also known for its ability to absorb unwanted noise. However, this ability of LGW to absorb noise is rather hard to quantify, because there is a lack of reliable experimental methods to measure it in-situ. This work reports on a new method to measure the absorption coefficient of LGW which makes use of a highly directional parametric transducer and acoustic intensity method. This method is tested under controlled laboratory conditions and in a typical street environment. The results of these experiments demonstrate the ability of the method to measure the absorption of a LGW. It also enables us to quantify the effects of the plant type and moisture content in the soil on the ability of the LGW to absorb sound. The proposed method has certain benefits over ISO354-2003 and CEN/TS 1793-5:2003 standard methods.
Adaptive algorithms for combined feedforward-feedback controllers
Authors: Bean Jacob , Fuller Chris , Schiller Noah
Published: November 2016

Traditional approaches to feedback and hybrid active noise and vibration control are generally based on adaptive gradient search methods using estimates of the instantaneous gradient. Most of these algorithms employ internal model control and are derivatives of the least mean squares algorithm commonly used in adaptive filtering. In these algorithms, the gradient calculation neglects the presence of a feedback path. To avoid potential algorithm divergence due to the presence of the feedback path, it is common to update the filter conservatively by including a leakage term, which is in general a trial and error procedure. This paper presents mathematical derivations for the updating mechanisms in feedback and hybrid controllers that make no assumptions regarding the accuracy of the plant model. Rather than deriving update equations assuming an ideal plant model and then modifying the adaptation process to be less sensitive to model errors, the reverse approach is taken. It will be seen that the presence of the feedback loop can be accounted for directly in the update equation. This allows either hard or probabilistic bounds on the uncertainty to be set at the outset of design. Computer simulations are presented to validate the algorithms.