bluuebunny/crossref_metadata_2025 huggingface dataset:
The role of acoustic microscopy in bone research
Authors: Raum Kay , Schrof Susanne , Schneider Johannes , Iori Gianluca , Kilappa Vantte , Du Juan , Pumberger Matthias , Putzier Michael , Zhou Jinming , Isaksson Hanna
Published: November 2016

Scanning acoustic microscopy (SAM) has been introduced 3 decades ago with the hope to open a new dimension in the microscopic analysis of biological tissues. However, only during the last decade this technology has emerged from a qualitative imaging modality to a quantitative measurement tool that provides fast and nondestructively elastic maps of acoustic and elastic properties with microscale resolution. Particularly, the spatial registration of parameter maps obtained by SAM with those obtained by complementary modalities, e.g., synchrotron microcomputed tomography, Raman spectroscopy, and inelastic micromechanical testing provided unprecedented inside into structure-composition-function relations, tissue changes with respect to adaptation, ageing, pathologies, and healing. Moreover, elastic maps generated by acoustic microscopy can serve as direct input for numerical simulations. This talk will review the key theoretical principles, experimental clues but also the limitations for the reconstruction of calibrated maps of elastic tissue properties and will present recent findings obtained by correlative imaging and SAM-based numerical sound propagation models. In particular, the role of matrix stiffness heterogeneity as a potential indicator for bone brittleness will be discussed.
Ultrasound multiple scattering for the assessment of bone micro-architecture
Authors: Gardner Mason , Haupert Sylvain , Renaud Guillaume , Muller Marie M.
Published: November 2016

Retrieving bone micro-architectural parameters non-invasively using ultrasound would enable the monitoring of bone loss, and the follow up of the efficacy of treatments. When ultrasound propagate in bone in the MHz range, multiple scattering is non-negligible, and we demonstrate that it is possible to take advantage of it. We propose methods based on ultrasound multiple scattering for the quantification of micro-architectural parameters in cortical and trabecular bone. Using linear transducer arrays at 5 and 8 MHz, we present both finite differences simulations and experimental results in 3D printed phantoms of trabecular bone, in equine trabecular bone and in human cortical and trabecular bone. We show that the framework of the Independent Scattering Approximation can be used to model the attenuation in both cortical and trabecular bone. We demonstrate that measuring the diffusion constant is feasible in cortical and trabecular bone, and that it is correlated to micro-architectural parameters. Preliminary results indicate a significant correlation between the trabecular anisotropy and the anisotropy of the diffusion constant (r = 0.87, p = 7.6e-8). In cortical bone, a significant quadratic relationship was observed between the mean pore size and the diffusion constant. The cortical pore density was highly and significantly correlated to the diffusion constant (r = 0.98, p = 0.001).
Pathology and immune effects of magnetic resonance imaging-guided Boiling Histotripsy in murine tumor models
Authors: Adema Gosse J. , Hoogenboom Martijn , van den Bijgaart Renske , Eikelenboom Dylan , Wesseling Pieter , Heerschap Arend , den Brok Martijn , Futterer Jurgen
Published: November 2016

In situ tumor ablation techniques are successfully applied for the destruction of local, often inoperable tumor masses. Following ablation tumor antigens become instantly available for immune cells, but systemic abscopal effects have only occasionally been reported after ablation monotherapy. Which ablation technique combines optimal local destruction with effective antigen release for induction of anti-tumor immunity is largely unknown. We study non-invasive MRI-guided high intensity focused ultrasound-ablation (MRgHiFU) in murine tumor models for local destruction by heating or mechanical disruption using Boiling histotripsy (BH). BH mechanically fragmentizes soft tissue into submicron fragments that are absorbed as part of a physiological healing response. BH treatment was performed using a MR compatible animal HIFU system (Image Guided Therapy, Pessac, France) with a 3 MHz transducer (focal spot size 0.5 x 0.5 x 2.0 mm). A 7T animal MR scanner was used for treatment guidance and evaluation. Here, we will present the pathological response and efficacy of BH treatment in three mouse models with different tumor characteristics: a soft-tissue melanoma (B16OVA), a compact growing thymoma (EL4), and a highly vascularized neuroblastoma (9464D) tumor. Furthermore, the impact of the type of ablation on immune cell infiltration and immune cell activation will be discussed.
Investigation of mechanical and dynamic characteristics for multifunctional contrast agents using atomic force microscopy and acoustic assessments
Authors: Guo Gepu , Ma Qingyu , Tu Juan , Zhang Dong , Feng Shaotong
Published: November 2016

Although multi-parameter fitting algorithms are often used for the characterization of coated-bubbles, it is inevitable to introduce uncertainty into the results. Therefore, it is urgent to develop some improved techniques to analyze the mechanical properties of the multifunctional microbubbles (MBs) accurately and systematically. By combining the measurements of atomic force microscopy, optical and acoustic detection with the simulations of coated-bubble dynamics, a comprehensive technology was developed to determine the size distribution, shell thickness, elasticity, and viscosity of multifunctional MBs precisely. Moreover, the impact of magnetic nanoparticles (MNPs) concentration on the multifunctional MBs' dynamic properties was studied systematically. It is demonstrated that, with the increasing MNPs concentration, the MB mean diameter and shell stiffness increased and ultrasound scattering response and inertial cavitation activity could be significantly enhanced. Although the shell thickness does not depend on the MNPs concentration, the increased MNPs concentration would generally result in enhanced bulk modulus and reduced bulk viscosity. The results of current studies demonstrate that the proposed single-parameter evaluation method could be helpful for further design and development of MB agents in clinic applications.
Investigation of carbon nanotubes for acoustic transduction applications
Authors: Howarth Thomas R. , Huang Dehua , Schumacher Christian R. , Mayo Nathanael K. , Cox Donald L. , Boisvert Jeffrey E. , Aliev Ali E. , Baughman Ray H.
Published: November 2016

Traditional acoustic transduction sources typically begin with the generation of an electrical excitation pulsed through an amplifier into an electroacoustic material to create a mechanical vibration which is then converted into an acoustic wave to produce sound. The lower the preferred transmitting frequency (and hence, longer acoustic range) desired, the larger the required size of the source. Often this means that for acoustic projectors producing sound at frequencies below a few kHz, that the electroacoustic device will need to be very large in order to produce very long sound waves. This has a limitation for incorporating low frequency sonars on smaller autonomous underwater vehicles (AUVs). The topic of our presentation is an acoustic source technology that relies on the conversion of thermal energy to acoustic energy. Though the concept was first demonstrated in 1917, the recent advent of carbon nanotubes (CNT) now makes it possible to transmit acoustic waves in small and affordable packages using the thermophone approach. The presentation begins with an overview of thermophones and a method for incorporating the CNTs into a useable transduction device. Discussions will include detailing on-going efforts for the encapsulation, packaging and further insight of the thermoacoustic energy conversion process. Recent 2016 experimental data will be shown. New start numerical and analytical modeling efforts and future development thrusts will be discussed.
Thermophone modeling
Authors: Cox Donald L. , Howarth Thomas R. , Schumacher Christian R. , Mayo Nathanael K.
Published: November 2016

The use of carbon nanotube (CNT) films encapsulated within a pouch design pressurized with an inert gas has demonstrated promise for underwater sound projection. To augment this experimental demonstration, a modeling approach has been undertaken to capture the requisite physics and eventually provide a design tool to optimize performance. The modeling approach makes use of the finite element method as programmed in the commercially available COMSOL software. A multi-physics approach is taken where models are coupled that include: electrically stimulated heat, heat transfer to a compressible, viscous fluid, structural response of a thin shell to this computational fluid domain, and external acoustic propagation from the outer surface of the structure. The intent is to capture the physical response of the sensor tested and understand the interaction dynamics of the various media. The multiphysics solution approach taken in this work involves the use of a segregated, time dependent solution of coupled, linear and nonlinear problems. As a result, the process is complicated and computationally intensive. Modeling results will be presented and compared with experimental data.
Underwater parametric array source transducer composed of PZT rods and thin polymer plate with high power efficiency for wideband sound generation
Authors: Ahn Hongmin , Hwang Yonghwan , Moon Wonkyu
Published: November 2016

Nowadays 1-3-type PZT composite transducers are widely used for underwater parametric array (PA) sound source since they can generate highly directional acoustic beam with high intensity, which is the requirements for the PA acoustic phenomenon that results from nonlinearity of a medium. Thanks to the relatively small mechanical quality factor of 1-3 PZT composite, the transducer using it can generate high intensity sounds by relatively low input voltage in the relatively wide frequency band around its resonance frequency. The low mechanical quality factor, however, increase the internal mechanical loss inside the transducer. Consequently, the acoustic radiation efficiency may be lower than expected. In this paper, a dual-resonant-frequency PZT rods are integrated into a thinner polymer plate. Then, the radiation surface increases to enhance radiation power efficiency. The PZT rods with two different lengths, are integrated in the process of molding a polymer plate. The fabricated transducer are operated with the out-of-phase driving method for a dual resonance transducer to generate direct sound beam with a wide frequency bandwidth. The sound pressure levels (SPLs) of the primary waves of the first and second resonant frequencies were 197 dB and 203 dB (re = 1μP), respectively, at 9 m. The acoustic radiation efficiencies at 98 kHz and 135 kHz were 50% and 33%, respectively. The DFW generated by the PA had an SPL of 150 dB (re = 1μP @ 30 kHz) and a high directivity of 3.4° half-power beam width.
Audio scene transformation using informed source separation
Authors: Marchand Sylvain
Published: November 2016

Giving the listener the freedom to transform the audio scene in real time is a challenging task. An important example of transformation is (re)spatialization: moving sound sources in space. This is equivalent to source separation. Indeed, moving all the sources of the scene but one away from the listener separates that source. And moving separate sources then rendering from them the corresponding scene is easy. But allowing this spatial transformation/source separation with a sufficient quality is a (too) challenging task for classic approaches, since it requires an analysis of the audio scene with inevitable (and often unacceptable) estimation errors. We introduced the informed approach, which consists in inaudibly embedding some additional information in the audio signal. This information, coded with a minimal rate, aims at increasing the precision of the analysis/separation. Thus, the informed approach relies on both estimation and information theories. Since the presentation at the ASA meeting in 2010, several informed source separation methods were proposed. Among the best methods is the one based on spatial filtering (beamforming), with the spectral envelopes of the sources (perceptively coded) as additional information. This allows the manipulation of the sound sources with an unequaled (controllable) quality.
Evaluation of singing voice by amateur singers from various aspects
Authors: Ito Akinori
Published: November 2016

Many research works have been conducted for evaluating singing voice so far. Most of the works aimed to investigate the voice of professional vocalists. Besides, evaluation of amateur singers is useful for developing entertainment applications such as scoring of Karaoke. In this presentation, two of such applications are introduced: an evaluation of singing enthusiasm and the detection of mistakes of sung lyrics. The evaluation of singing enthusiasm is based on three kinds of features: the A-weighted power, the vibrato-related feature, and the fall-down feature. Using these features, perceived singing enthusiasm can be estimated relatively accurately. In addition, singers’ intention of enthusiasm can also be estimated from the fluctuation of the segmental power of the voice. Detection of mistakes of sung lyrics is based on the dynamic time warping between the input singing voice and the reference singing voice. To normalize the singer-dependent variation of voice features, we employed singer normalization based on linear regression. Based on the DTW alignment, frame-by-frame distances between the two voices are obtained. After temporal smoothing of the distances, the lyrics errors are detected based on the peak detection of the distance. High detection accuracy is obtained when the errors are based on phrases.
Recent extensions to topological and nonlinear aspects of wave digital filter theory
Authors: Werner Kurt J.
Published: November 2016

The Wave Digital Filter approach can be used to create computational models of lumped reference systems, including rectilinear mechanical, rotational mechanical, acoustical, and electronic systems. When they are applicable, Wave Digital Filters enjoy properties including modularity, accuracy, and guaranteed incremental passivity that make them attractive in the context of musical instrument and audio effect simulation. However, the class of reference systems that can be modeled with Wave Digital Filters has historically been restricted to systems with simple topologies (which can be decomposed entirely into series and parallel connections) and which contain only a single “nonadaptable” element (e.g., a diode, ideal source, or switch). This talk details recent Wave Digital Filter advances from Stanford University's Center for Computer Research in Music and Acoustics (CCRMA), which broaden the applicability of Wave Digital Filters to reference systems with any topology and an arbitrary number of nonlinearities or other nonadaptable elements. These advances have enabled many new and previously intractable musical circuit simulations at CCRMA, including active tone stages and transistor/diode/feedback clipping from guitar distortion pedals, the electromechanical Hammond organ vibrato/chorus system, guitar tube pre-amplifiers and tone stacks, circuits involving operational amplifiers, drum machine circuits, and transistor amplifiers.
Music signal processing by the human brains: Studies on the strategies used by professional pianists to efficiently sight-read music
Authors: Aiba Eriko
Published: November 2016

The human brain is an ingenious signal processing system. When musicians play instruments, their brains must process a huge amount and various types of information in parallel. Particularly, the sight-reading of piano music requires the processing of an enormous amount of information as piano music includes many chords and is written on the great staff (or grand staff). Pianists have to read the score, interpret the music, search for the keys to be played while planning the motions of fingers, and control their fingers. In addition, they must adjust the sound intensity and usage of the sustaining pedal according to the output sound. All these are performed simultaneously and successively. The sensory information or signals to be processed include many different modalities, such as visual, auditory, tactile, and motion sensing. In order to complete this complicated task, it is important that the pianists efficiently process this information. In this presentation, I introduce our experimental studies investigating the strategies used by professional pianists to improve the efficiency of information processing while sight-reading music.
Applications of automatic equipment identification to studies of rail noise and vibration in North America
Authors: Reuter Eric L.
Published: November 2016

When conducting studies of noise and vibration impacts of existing rail lines, it is useful to identify and log train movements. Since the early 1990s, all railcars and locomotives in North America have been outfitted with automatic equipment identification (AEI) tags. AEI is an RFID system that allows for automated electronic identification of passing equipment with unattended trackside readers. These data reveal the makeup, direction, length, speed, and other useful parameters of each passing train. In order to assess the feasibility of using AEI readers in noise studies, a field study was conducted using a temporary trackside reader and correlated sound and vibration monitors. This paper will present a technical overview of the AEI system and the results of the field study.
Modeling the generation of supersonic turbulent jet noise by large-scale coherent structures
Authors: Schmidt Oliver T. , Colonius Tim , Brès Guillaume A.
Published: November 2016

Large-scale coherent structures, or wavepackets, are a salient feature of turbulent jets, and the main source of jet mixing noise at aft angles. They are extracted from a high-fidelity Mach 1.5 LES database as spectral POD mode estimates. These most energetic wavepackets obtained via POD and their acoustic far-field radiation patterns are compared to solution to the one-way Euler (OWE) equations recently introduced by Towne & Colonius (AIAA Paper 2013-2171, 2013; AIAA Paper 2014-2903, 2014). Within the OWE framework, the linearized Euler equations are modified such that all upstream propagating acoustic wave components are removed from the solution. The resulting spatial initial value problem can be solved in a stable and computationally efficient manner by downstream marching the solution. Additionally, the scenario of stochastic forcing of wavepackets by the surrounding turbulence is considered in a resolvent analysis. The resolvent analysis allows for the computation of optimal forcing distributions and corresponding responses. It is based on a singular value decomposition of the transfer function of the governing linear operator. The results of the both methods, OWE and resolvent analysis, are compared to the most energetic POD modes with a special focus on far-field radiation patterns and computational efficiency.
Wavepacket source modeling of high-performance military aircraft jet noise from cross-beamforming analysis
Authors: Harker Blaine M. , Gee Kent L. , Neilsen Tracianne B. , James Michael M.
Published: November 2016

Wavepacket models provide a convenient representation of jet noise source phenomena because of the extended, partially correlated nature of the turbulent mixing noise. When treated as an equivalent source model, they are useful to estimate features of both the radiated noise as well as the source characteristics to assist in jet noise reduction efforts. In this study, advanced cross-beamforming techniques are applied to measurements in the vicinity of a high-performance military aircraft. These results are then decomposed into an azimuthally-averaged multi-wavepacket representation of the data, which can then be treated as an equivalent source. Estimates of the field levels and coherence properties using the equivalent source are compared with measurements, and results from the multi-wavepacket model show good agreement with benchmark measurements over a range of frequencies that contribute significantly to the overall radiation. The capabilities and limitations of the model to estimate field properties are quantified with respect to benchmark measurements in the mid and far fields as a function of frequency. Results indicate that the multi-wavepacket representation is an improvement over single-wavepacket models, which do not incorporate spatiotemporal features of the radiation. [Work supported by ONR and USAFRL through ORISE.]
In defense of the Morfey-Howell Q/S as a single-point nonlinearity indicator: An impedance-based interpretation
Authors: Ohm Won-Suk , Gee Kent L. , Reichman Brent O.
Published: November 2016

Since the Morfey-Howell Q/S was proposed as a nonlinearity indicator for propagation of intense broadband noise [AIAA J. 19, 986-992 (1981)], there has been considerable debate as to its meaning and utility. Perhaps the most contentious argument against Q/S is about its validity as a single-point nonlinearity indicator: the importance of nonlinearity is often judged by observing cumulative effects over some propagation distance, whereas Q/S is based on a pressure waveform at a single location. Studies to address these criticisms have emerged over the years, most recently by Reichman et al. [J. Acoust. Soc. Am. 139, 2505-2513 (2016)] in support of Q/S. In this talk we show that the Burgers equation (from which the Q/S was originally derived) can be recast in terms of specific impedance, linear absorption and dispersion coefficients, and normalized quadspectral (Q/S) and cospectral (C/S) densities. The resulting interpretation is that Q/S and C/S represent the additional absorption and dispersion, introduced by the passage of a finite-amplitude wave to the existing linear absorption and dispersion. In other words, a nonlinear wave process alters the apparent material properties of the medium, the extent of which can be used as a single-point indicator of the relative strength of nonlinearity.
Superresolution using an acoustic vortex wave antenna
Authors: Guild Matthew D. , Naify Christina J. , Martin Theodore P. , Rohde Charles A. , Orris Gregory J.
Published: November 2016

Acoustic waves, like their electromagnetic counterpart, exhibit a diffraction limit, which restricts the ability to resolve or generate fine structures in the pressure field. Previously proposed methods to overcome this limit, and therefore achieve superresolution, have mainly been limited to operating within the near-field region of the aperture. In this work, we describe how far-field superresolution can be achieved using shaped acoustic vortices, created by utilizing the topological diversity of an acoustic vortex wave antenna. Theoretical and numerical results will be presented for the design of an acoustic vortex wave antenna with arbitrary planar arrangement which is capable of generating superresolved far-field features in the radiated acoustic pressure. Variation of the antenna geometry enables different shaped acoustic vortex patterns to be achieved, which propagate from the near-field into the far-field through an arrangement of stable integer mode vortices. Despite the total aperture size being less than a wavelength in diameter, the proposed acoustic vortex wave antenna is shown to achieve far-field superresolution with feature sizes 4-9 times smaller than the resolution limit. Several examples will be presented and discussed in detail. [Work supported by the Office of Naval Research.]
Demonstration of acoustic holographic rendering with two-dimensional metamaterial-based passive phased array
Authors: Xie Yangbo , Shen Chen , Jing Yun , Cummer Steven
Published: November 2016

Acoustic holograms in analogy with optical holograms are useful for a variety of applications, such as multi-focal lensing, multiplexed sensing and synthesizing three-dimensional complex sound fields. We previously presented the designs and simulation results of several metamaterial-based acoustic holograms (J. Acoust. Soc. Am. 138, 1751 (2015)). Here, we demonstrate the experimental realization of two of them: one projects a complex pattern on the image plane, while the other focus energy onto multiple circular spots of different sizes. We showcase the holographic reconstruction measurements carried out in an anechoic chamber. The demonstrated passive holograms, without phase-shifting circuitry and transducer arrays, are advantageous for applications where simplicity, robustness, and small footprint are preferred over adaptive control. We also discuss the higher frequency versions of such holograms and their applications in robotic sensing and wireless power transfer. Such metamaterial-based holograms can serve as versatile platforms for various advanced acoustic wave manipulation and signal modulation, leading to new possibilities in acoustic sensing, imaging and energy deposition.
New frontiers in elastic metamaterials for controlling surface waves
Authors: Colombi Andrea , Roux Philippe , Colquitt Daniel , Guenneau Sebastien , Ageeva Victoriya , Clark Matt , Craster Richard
Published: November 2016

In this presentation, we show recent results obtained by our research group on two types of elastic metamaterial that are capable of controlling the propagation of surface elastic waves at various physical scales: from tens of meter in geophysics to millimeters in ultrasonic. We begin from the geophysical scale introducing a metamaterial that uses soft soil inclusions in the ground to create a lens for detouring seismic surface waves around an obstacle at frequency <10 Hz. We then move to the smaller scale presenting a metamaterial that exploits the local resonances in a subwavelength collection of resonators (vertical rods) attached to a finite (plate) or semi-infinite (halfspace) substrate to control either the A0 mode in the plate or Rayleigh waves in the halfspace. We focus on the halfspace case and we illustrate how Rayleigh waves can be (1) mode converted into S-waves or stopped and trapped by a graded array of resonators (seismic rainbow); (2) focused, or detoured using a resonant metalens (Luneburg, Maxwell, and Eaton type) made with the vertical rods. The metamaterial are presented using results from time domain numerical simulations, laboratory and geophysical experiments to highlight the broad applicability of these objects across the wavelength spectrum.
Metamaterial ground cloak with anisotropic solid inclusions
Authors: Beck Benjamin , Hanford Amanda , Kerrian Peter
Published: November 2016

There has been much research on the use of metamaterials within ground cloaks to hide objects on a rigid surface. Within the literature, ground cloaks made with metamaterials based on unit cells with solid inclusions have required the use of at least two distinct materials within the unit cell to achieve the necessary anisotropic material properties calculated from a transformation method. Here, we present a method to calculate the material properties of an anisotropic solid inclusion for use in a metamaterial ground cloak. The generated materials properties are then shown to be realized through an additively manufactured anisotropic solid inclusion. Finally, numerical results of cloaking performance of the metamaterial ground cloak are presented.
Voicing contrast in Punjabi Singletons and Geminates
Authors: Hussain Qandeel
Published: November 2016

Cross-linguistic studies have shown that voiceless geminates are more frequent than voiced geminates (Blevins, 2004). From an articulatory perspective, maintaining voicing and long closure duration in voiced geminates is challenging (Ohala, 1983). Punjabi, an Indo-Aryan language, has been reported to contrast voiceless/voiced singletons and geminates. However, it is not known if voiced geminates in Punjabi are completely voiced and how their durational properties differ from voiceless geminates. The aim of the present study was therefore to investigate the acoustic characteristics of Punjabi voiceless/voiced singleton and geminate stops. Five native speakers of Punjabi participated in the experiment (24-26 years, M= 24.6). The stimuli consisted of nonce words with word-medial (C2) voiceless/voiced singleton and geminate stops (C1V1C2V2). The C2 duration was measured in Praat, by using the broadband spectrograms and visual inspection of the waveforms (Ridouane, 2007). The durational analysis of C2 indicated that unlike voiceless singletons/geminates, voiced singletons/geminates showed complete voicing during the closure phase. These findings differ from Tokyo Japanese where voiced geminates are partially devoiced (Kawahara, 2016). However, voiceless singletons/geminates are slightly longer in duration compared to voiced singletons/geminates. The current study raises questions about the cross-linguistic differences in voiceless/voiced singletons and geminates.
Prosodic analysis of storytelling speech in Japanese fairy tale
Authors: Saito Takashi
Published: November 2016

This paper presents a prosodic analysis of storytelling speech in Japanese fairy tale. Recent advances in TTS technologies bring us high quality synthetic speech especially on acoustic aspects. On prosodic aspects, however, there is still room for improvement in the expressiveness since most systems use one-sentence speech synthesis scheme. For instance, storytelling applications expect speech synthesis to be capable of having a control mechanism beyond one sentence. In this paper, a prosodic database is built for real storytelling speech of professional narrators on Japanese fairy tale aiming at investigating the actual storytelling strategies and ultimately reflecting them on the expressiveness of speech synthesis. After conducting a baseline speech segmentation for phones, words, and accentual phrases in a semi-automatic way, a multi-layered prosodic tagging is manually performed to extract information on various changes of “story states” relevant to impersonation, emotional involvement, and scene flow control. Storytelling speech materials of six professional narrators for Japanese fairy tale are prepared to analyze their storytelling strategies. In particular, the dynamics of pitch contours is investigated in terms of expressiveness control beyond sentence considering narrator dependency as well. Based on the findings obtained, pitch control schemes are discussed for storytelling speech synthesis.
Formant trajectories in the realization of 3 Malayalam rhotics
Authors: Kochetov Alexei , Howson Phil
Published: November 2016

Malayalam has a typologically rare set of three rhotic consonants—an alveolar trill, an alveolar tap, and a retroflex/postalveolar approximant. The first one is described as velarized, while the other two are somewhat palatalized. Previous acoustic work by Punnoose et al. [Phonetica 70, 274-297 (2013)] has established that the contrast can be differentiated by formant values around the constriction, as well as at the midpoint of adjacent vowels. Less clear, however, is the temporal extent of these formant differences and their overall dynamics. In this study, we analyze three rhotics produced by 10 male Malayalam speakers. Measurements of F1-F4 were taken at 10 points throughout the VCV interval and subjected to a series of Smoothing Spline ANOVAs. The results revealed differences in F2 (tap, approximant > trill) and F3 (tap, trill > approximant) among the rhotics extended through most of the preceding and following vowel duration, while differences in F1 (trill > tap, approximant) were less extensive. While similar to the tap in its F2 trajectory, the approximant showed a more abrupt fall during the following vowel, seemingly indicative of a rapid tongue posture change at the release. [Work supported by SSHRC.]
Temporal and spectral characteristics of the intervocalic consonant clusters in Korean
Authors: Park Hansang
Published: November 2016

Park(2008) showed that there is no significant difference in specific segment duration and their temporal structure between VCC[Asp]V and VC[Asp]V, and between VCC[For]V, VCC[Len]V, and VC[For]V, where V stands for vowel, C for consonant, [Asp] for aspirated, [For] for fortis, and [Len] for lenis. He also showed that there is no significant difference in consonant duration between a singleton and a geminate when the following C is [For] or [Asp], but a significant difference between them when the following C is [Len]. On the other hand, Park(2005) showed that there is a significant difference in phonation type index k (PTI k) across consonant types, where PTI k presents a single and simplified measure of the spectral tilt, which is free from the effects of fundamental frequency and vowel quality. This study, by applying Park(2008)’s and Park(2005)’s methods, investigates temporal and spectral characteristics of the consonant clusters of the other manners and places of articulation. The results showed that temporal characteristics present a similar pattern to those of Park(2008), and that spectral characteristics are associated with the phonation identity of the final consonant of the intervocalic consonant clusters.
Three-dimensional reduction for three-way contrast: Conversational stops in Korean
Authors: Sung Jae-Hyun , Brenner Daniel
Published: November 2016

Phonetic reduction in conversational speech is widely acknowledged across different languages and dialects. Previous experimental studies have reported that spontaneous conversations yield greater variability and reduction in acoustic signals when compared to careful speech. This study investigates the acoustic reduction in intervocalic Korean stops—phonemically lenis, fortis, and aspirated stops—from conversational speech, and adds to the increasing body of reduced speech research. Comparisons of Korean stops from careful and conversational speech confirm greater acoustic variability in conversational speech than in careful speech. Furthermore, conversational stops in Korean show that phonetic reduction occurs in multiple acoustic dimensions. Lenis stops exhibit phonetic reduction represented by both a smaller intensity difference and a higher pitch, fortis stops do so by a higher pitch only, and aspirated stops do so by both a smaller VOT and a higher pitch. Results from this study leave open the possibility of manner-specific strategies in phonetic reduction within a language, and merit further cross-linguistic examination of conversational speech.
Glottalization, reduction, and acoustic variability in function words in American English
Authors: Dilley Laura , Arjmandi Meisam K. , Ireland Zachary , Heffner Chris , Pitt Mark
Published: November 2016

Function words vary widely in pronunciation; understanding this variation is essential for accurate cognitive modeling of lexical perception and production, as well as for computer speech applications. This study analyzed acoustic variation in pronunciation of vowel-initial words a, are, our, and or, as well as her, which has a vowel-initial variant, [ɚ]. Sentences were constructed in which each function word was preceded by a phonetically similar open syllable rhyme, with which the word was expected to sometimes blend spectrally. Twenty-nine participants uttered 100 sentences in a casual speaking style. The degree of clarity of function word onsets was measured, focusing on evidence of onset discontinuities: glottal onset gestures (e.g., low fundamental frequency, F0; amplitude dips) for vowel-initial words a, are, our, and or, and, for her, of /h/ (e.g., root-mean-square amplitude, amplitude differences of first and second harmonics). Results showed that a substantial proportion of the time, function words lacked any detectable acoustic onset discontinuity. When discontinuities did occur before vowel-onset function words, there was a high degree of acoustic variability in those realizations. These findings highlight the challenges for speech perception to identify word onsets and/or to overcome variability in recognizing words. [Work supported by NSF Grant BCS-1431063.]
The relationship of VOT and F0 contrasts across speakers and words in the German voicing contrast
Authors: Bang Hye-Young , Sonderegger Morgan , Clayards Meghan
Published: November 2016

Recent studies on tonogenesis in progress in Seoul Korean (Kang, 2014; Bang et al., 2015) find that the size of the VOT contrast and the f0 contrast between aspirated and lax stops “trade off” across speakers (e.g., male speakers have greater/smaller VOT/f0 contrasts), as well as words (e.g., different frequencies, following vowel heights). We examine whether this parallelism across speakers and words occurs in a language not undergoing tonogenesis by examining the size of the fortis/lenis contrast in VOT and f0 in German, using speech from the PhonDat corpus (Draxler, 1995). Mixed-effect regression models show that the size of the VOT contrast, but not the f0 contrast, is affected by properties of words (e.g., frequency, vowel height), unlike the parallelism observed in Korean. We further investigated whether VOT/f0 parallelism would be observed across speakers by partialing out linguistic factors affecting VOT/f0, and performing one logistic regression per speaker (n = 76) of fortis/lenis class as a function of residualized f0 and VOT. The cue weights of F0 and VOT are negatively correlated across speakers (r=-0.213, p = 0.0213), suggesting parallelism exists across speakers, but not across words.
Voice activity detection in noisy environment using dynamic changes of speech in a modulation frequency range of 1 to 16 Hz
Authors: Dwijayanti Suci , Miyoshi Masato
Published: November 2016

Voice activity detection (VAD) is usually placed at the preprocessing stage of various speech applications. There have been a lot of studies to find acoustic features that are effective in distinguishing speech/non-speech segments on a recorded signal. In this report, we utilize speech characteristics in a modulation frequency range of 1-to-16 Hz, which seem beneficial to avoid misjudgment. The signal amplitudes are evaluated in common logarithms not to miss small changes in speech energy that may have relations with the starting or ending point of an utterance. In order to capture such changes, the first and second derivatives of the signal are calculated. In finding the starting point, the positive second derivatives seem effective. In finding the ending point, the combination of the negative first derivatives and the positive second derivatives seem effective. And these features should be calculated for individual subbands. On these conditions, Deep Neural Network (DNN) is successfully trained to determine the speech/non-speech segments. In this report, we will evaluate this method in comparison with conventional ones for various noisy utterances.
Analysis of voice imitation by professional/non-professional impersonators based on Kullback–Leibler divergence between acoustic models
Authors: Iwano Koji , Horihata Takuto
Published: November 2016

Considering the practical use of speaker verification systems, it is important to investigate the effect of spoofing attacks by professional/non-professional voice imitation. This research proposes an analysis method of voice imitation using the distances between three GMM-based acoustic models trained from cepstral features of “impostor's original voice,” “target person's voice,” and “impostor's imitated voice.” The distance measure is defined by the Kullback-Leibler (KL) divergence. The analysis uses Japanese imitated voice produced by a male professional impersonator and six male non-professional impostors. Each impostor imitated five or six target persons who have never been tried to imitate by the impostors before the experiments. The analysis results show that 1) although the non-professional imitators drastically change their voice features by the imitation, the averaged acoustical distance between the imitated and target voice is still large, 2) whereas the professional imitator approaches their voice characteristics towards the target voice; the distance between the imitated and target voice is approximately 70% of the original distance. The experiments of speaker verification using HMM-UBM-based framework show that the professional imitation certainly yields higher equal error rates than that of non-professional imitation.
Optimization of topic estimation for the domain adapted neural network language model
Authors: Hagiwara Aiko , Ito Hitoshi , Ichiki Manon , Mishima Takeshi , Kobayashi Akio , Sato Shoei
Published: November 2016

We present a neural network language model adapted for topics fluctuating in broadcast programs. Topic adapted n-gram language models constructed by using latent Dirichlet allocation for topic estimation are widely used. The conventional method estimates topics by separating the corpora into chunks that have few sentences. While the n-gram model uses several preceding words, the recurrent neural network and long short-term memory can learn to store huge amounts of past information in the hidden layers. Consequently, chunks for language models trained by using neural networks may have a longer optimal length than the chunks for language models trained by using the conventional methods. In this paper, the length of chunks and topic estimation process are optimized for the neural network language models. For the topic estimation, k-mean clustering, latent Dirichlet allocation, and word2vec were compared. On the basis of the results of comparison, we designed a neural network language model.
End-to-end neural network modeling for Japanese speech recognition
Authors: Ito Hitoshi , Hagiwara Aiko , Ichiki Manon , Mishima Takeshi , Sato Shoei , Kobayashi Akio
Published: November 2016

This study proposes end-to-end neural network modeling to adapt direct speech-to-text decoding to Japanese. End-to-end speech recognition systems using deep neural networks (DNNs) are currently being investigated. These systems do not need intermediate phonetic representation. Instead, many of them utilize Recurrent Neural Networks (RNNs) trained by using much more data than ever before. The end-to-end approach makes acoustic models simpler to train. Typically, previous works have dealt with phonogram labels such alphabetic characters. Ideograms such as Kanji, however, make end-to-end speech recognition more complex. A single Kanji can have multiple readings, such as On-yomi (Chinese reading) and Kun-yomi (Japanese reading). In addition, whereas alphabets have at most 100 labels, Japanese has over 2000 labels to predict, such as Kanji, Hiragana, Katakana, the Roman alphabet, digits, and punctuation marks. To resolve this problem, we attempt to make end-to-end neural network modeling allows speech recognition of Japanese without phonetic representation. This method trains RNN and adopts the connectionist temporal classification (CTC) objective function. The proposed method was able to deal with a large amount of character labels. We also analyzed the decoding results and examined ideas for improving the accuracy of word error rate (WER).
Automatic estimation of extra-linguistic information in speech and its integration into recurrent neural network-based language models for speech recognition
Authors: Toyama Shohei , Saito Daisuke , Minematsu Nobuaki
Published: November 2016

When one talks with others, he/she often changes the way of lexical choice and the speaking style depending on various contextual factors. For example in Japanese, we often use gender-dependent expressions and, when we speak to elderly people, we use polite expressions. It can be said that in any language, formality or informality of speaking changes drastically depending on the situation where speakers are involved. The aim of automatic speech recognition (ASR) is to convert speech signals into a sequence of words and the above fact indicates that the performance of ASR can be improved by taking the various contextual factors into account in the ASR modules. In this study, at first, we attempt to estimate those contextual factors as para-linguistic or extra-linguistic information and then, to integrate the results into language models based on Recurrent Neural Network (RNN) for speech recognition. In experiments, from an input utterance, i-vector and openSMILE features were extracted to represent speaker identity and speaking style. These acoustically driven features were integrated into the reranking process of RNN-based language models. Reductions of perplexity of the language models were shown to be 1 to 2% relative.
Robust multi-frequency sparse Bayesian learning: Data results
Authors: Gemba Kay L. , Nannuru Santosh , Richards Edward , Hodgkiss William S. , Gerstoft Peter
Published: November 2016

Compressive sensing based techniques have been applied successfully to underwater acoustics problems. We present data results using a robust multi-frequency sparse Bayesian learning (SBL) algorithm that can account for model mismatch leading to errors in the replica vector dictionary. We use the SWellEx-96 Event S5 data set to demonstrate SBL capabilities for the beamforming and matched field processing (MFP) application. Beamforming results using the entire 64 element array (design frequency of 400 Hz) and source frequencies ranging from 50 to 400 Hz indicate that multi-frequency SBL outperforms Bartlett and WNC processors in identifying multipath arrivals over the approximately 10 km source track. MFP results in a multiple-source scenario indicate that SBL offers a degree of robustness in the presence of data-replica mismatch when tracking a quiet source. The data-replica mismatch is especially pronounced at the closest point of approach due to array tilt of approximately 2 degrees. Data results further indicate that the two SBL tuning parameters (diagonal loading of the replica vector covariance matrix and number of sources for the algorithm’s noise estimate) do not require excessive calibration.
Some comments on detecting unmanned underwater vehicles using passive acoustics
Authors: D'Spain Gerald L.
Published: November 2016

Passive methods of detection rely upon the presence of a field radiated by the phenomenon of interest (in this case, unmanned underwater vehicles—UUVs). This talk illustrates the challenges of using passive acoustic methods to detect UUVs. Two types of vehicles are discussed; a 20-ft wingspan, buoyancy-driven, flying wing autonomous underwater glider and a 21-in-diameter, prop-driven UUV. At-sea measurements of the acoustic field radiated by the glider's large (30 liter) buoyancy engine are presented. Whereas a prop-driven propulsion system operates continuously, a buoyancy engine has very low duty cycle: a few percent. Onboard self noise from the glider's internal fluid-based roll control system (which provides control authority near neutral buoyancy) far exceeds that from an aileron-based system. For the 21-in-diameter prop-driven UUV, modifications were made to its propulsion and steering systems to reduce radiated noise by 20-50 dB. The resulting self noise levels recorded at sea by a hull-mounted hydrophone now are below typical background ocean noise levels above 200 Hz, with the noise below 200 Hz being primarily vibration induced. Highly directional passive acoustic receiving systems are required to detect the transits of these UUVs. [Work supported by the Office of Naval Research and BP.]