bluuebunny/crossref_metadata_2025 huggingface dataset:
Application of compressive sampling to passive sonar signals.
Authors: Culver R. Lee. , Bose N. K.
Published: October 2009

Recently the compressive sampling (CS) paradigm has generated considerable interest in the signal processing community because it offers the potential to fully characterize signals without satisfying the Nyquist requirement (sampling frequency must be more than twice the highest frequency in the signal). Signal compression itself is not new; it is used in all file compression algorithms. However, it generally requires that all coefficients be generated, many or most of which are discarded and only a few are transmitted. The theoretical basis for CS has been presented in a number of signal processing and statistics journal articles [e.g., Candès and Wakin, IEEE SP (March 2008)]. CS is fixed (non-adaptive) and is efficient in that only the coefficients required for signal characterization are calculated. The key requirement of CS is to find a basis in which the signal representation is sparse and thus can be represented with a minimum number of coefficients. Here we explore possible benefits of applying CS to sonar signals, including signal compression, bandwidth reduction, and exploitation of sparse sampling geometries. [Work sponsored by ONR Undersea Signal Processing.]
Speech localization in any direction using power and frequency signatures, gradients, and differences.
Authors: Barnhill Colin L. , West James
Published: October 2009

Speech localization is a relatively simple task for a human but, often, a difficult task for acoustic arrays. Although arrays can localize impulsive and narrow-band sources through the use of cross-correlation and subspace methods (like MUSIC), these methods break down when there is low SNR, reverberant conditions, or the source is neither impulsive nor narrow-band. Interesting speech sources (for teleconferencing or surveillance) are most often in environments where many of the breakdown conditions exist. New or multiple methods are necessary to reliably localize speech. A new algorithm will be presented that makes use of multiple localization methods to locate the desired speech signal. The localization methods are based on the time and frequency power signatures, gradients, and differences of third order supercardioid beams. Multiple beams are used to spatially segregate power signatures, and the results are clustered to determine position. Using a spherical array, it is possible to localize speech in any direction using these beams. Experimental acoustic and mathematical results in real room situations will be presented.
Update on the reverberation modeling workshops.
Authors: Perkins John S. , Thorsos Eric I.
Published: October 2009

In order to investigate the status of reverberation modeling and establish well-defined benchmark problems, two reverberation modeling workshops have been jointly sponsored by the Office of Naval Research and the Navy Program Executive Office C4I (PMW 120). The first workshop was held 7–9 November 2006, and the second during 13–15 May 2008. This paper presents the approach used in formulating the reverberation problems posed to workshop participants and shows examples of the modeling agreement obtained for some of the workshop problems. Issues that have arisen as workshop participants have collaborated in their attempts to produce consensus solutions will be pointed out. Finally, some of the problems to be addressed at a proposed third workshop will be discussed.
Modeling acoustic propagation in shallow water using finite elements.
Authors: Isakson Marcia J. , Wilson Preston S.
Published: October 2009

Finite element models approach an exact solution of the Helmholtz equation as discretization density increases. Therefore, these solutions are an excellent benchmark model for reverberation studies. This study will present finite element reverberation solutions for two-dimensional shallow water waveguides for times up to 4 s. Five waveguides are considered: rough bottom only, rough surface only, rough surface and rough bottom, rough interfaces with summer (downward refracting) profile, and rough interfaces with winter (upward) refracting profile. [Work sponsored by Office of Naval Research, Ocean Acoustics.]
Emotion and vocalization in mammalian perspective: What do we know, what do we need to know?
Authors: Owren Michael J.
Published: October 2009

Emotional processes can play a central role in mammalian vocal communication, including both in triggering vocalizations and in mediating listener behavioral responses. However, the processes involved have received relatively little attention within the field of animal bioacoustics, at least in comparison to topics such as vocal anatomy, sensory processing, and the possible importance of higher-order cognition. It is therefore useful to take stock of what we currently know about the role of emotion in mammalian vocalization across species. Common elements of the papers in this Special Session will be highlighted, with the goal of identifying commonalities and differences in underlying approaches and overall findings. This overview will focus on key areas and issues that should be addressed both in reaching a better understanding of the role of emotion in vocalization and in raising the profile of this critical component of organismal function in the larger arena of the science of vocal communication.
Ocean acidification and its impact on ocean noise level: An analysis using empirical and physical models of acoustic transmission loss.
Authors: Reeder D. Benjamin , Chiu Ching-Sang
Published: October 2009

Over the next several hundred years, ocean acidification is expected to reduce average ocean pH. It has been reported that, given a 0.3 reduction in pH from 8.1 to 7.8, a 40% reduction in the acoustic absorption coefficient at low frequencies (less than 1 kHz) could result, suggesting a significant increase in ocean noise. Such extrapolations are based on the corresponding percentage change in the absorption loss only and ignore other dominant acoustic energy loss mechanisms, such as seabed attenuation, as well as the effect of ocean waveguide physics and the depth dependence of ocean pH. Presented here is an analysis of the impact of the reduction in ocean pH on ocean noise level using empirical and physical models of acoustic transmission loss. Three ocean acoustic environments are analyzed to elucidate the expected change in ocean noise level as a function of frequency: shallow water, a surface duct, and the deep ocean. Results show a negligible change in ocean noise in the shallow water and surface duct environments for all frequencies. In the deep ocean, the maximum change in ocean noise level occurs at approximately 2 kHz, with no significant change below 800 Hz.
Ocean noise level change in response to ocean acidification.
Authors: Udovydchenkov Ilya A. , Duda Timothy F.
Published: October 2009

It has been known for several decades that reduction in ocean pH (increase in acidity) would have an impact on ocean sound content and propagation loss. Net flux of carbon dioxide into the ocean has already reduced pH in the upper ocean, and as a result has reduced absorption in those same waters. This trend will likely continue. Based on forecasts of oceanic pH profiles and sound sources, changes in ocean sound content (ambient noise) can be computed. In general, sound energy loss can be partitioned into intrinsic dissipation (the chemical relaxation absorption process) and dissipation in the seafloor, with the boundary loss being analogous to sub-unity reflectivity at the boundaries of a room or concert hall. Here, expected deep ocean noise level changes in various frequency bands from 0.1 to 10 kHz are presented. A model is used that includes basic propagation physics (including the sound channel), depth variability of absorption, loss of sound energy into seafloor, and distributed acoustic sources. At low frequency, little sound is absorbed, so absorption effects are muted. At higher frequencies, sound does not propagate far, also minimizing absorption effects. In the center of the band, there can be sizable effects.
Effect of ocean acidity on the mid-frequency ambient noise field in shallow water.
Authors: Rouseff Daniel , Tang Dajun
Published: October 2009

Due to increased acidification, Hester et al. [Geophys. Res. Lett. 35, L19601] estimate that by midcentury the sound absorption of seawater will decrease by almost 40%. In the present study, the expected relative increase in the mid-frequency ambient noise field is estimated. Emphasis is placed on shallow water scenarios with typical summertime sound speed profiles and frequencies from 1 to 5 kHz. A model is developed that assumes shallow water internal waves will redistribute energy from high-order acoustic modes excited by noise sources near the sea surface into ducted, low-order modes that propagate with low loss. Calculations are made as a function of pH level, the strength of the internal waves, and the bottom composition. The modeled noise field is beamformed and the effect on the so-called ambient noise notch is estimated.
Acoustical oceanography and shallow water acoustics.
Authors: Lynch James F.
Published: October 2009

In acoustical oceanography, we use sound to help infer the physical characteristics of the ocean and seabed. There is a rich store of oceanographic, geological, and biological phenomena to be explored in shallow water, and we look at some of these phenomena first, with an eye toward what the current interests and challenges are. In seabed acoustics, the continental slopes and the canyons that cross-cut them are new and challenging areas to address. In physical oceanography, the full four-dimensional description of the internal wave field on the continental shelf is still a challenge. Also, the physical oceanography of the shelfbreak, continental slope, and canyons is another exciting new area. Biologically, the fact that marine life concentrates near the shelfbreak and in canyons is known, but is not well quantified. In looking at these topics, we will first address the forward propagation problem, and then decide how and if the acoustic inverse problem should be addressed. We will also discuss what specific ocean, biological, and seabed properties most need to be measured, and also how to best measure them—whether acoustically or otherwise.
Kidney stone localization in vitro using multiple-signal-classification.
Authors: Kracht Jonathan M. , Barbone Paul E. , Cleveland Robin O.
Published: October 2009

During shock wave lithotripsy, respiration and patient movement result in motion of a kidney stone. It has been estimated that 50% of the shock waves miss the stone. The misshots result in damage to the tissue with no therapeutic benefit. Here we employed a ring array of seven piezo-electric elements with center frequency of ∼600 kHz to track stones in vitro. Each element was used to transmit in turn, and the individual waveforms received on all elements were recorded. The Fourier transform of these data gives the multistatic response matrix, whose largest eigenvalues represent the dominant scatterers in the medium. The associated eigenvectors were employed by the multiple-signal-classification (MUSIC) method to generate a scattering indicator image from which scatterer locations were determined. We tested MUSIC’s applicability to track artificial kidney stones in water or with a tissue phantom. Single targets were tracked with an accuracy of 4 mm. A single stone resulted in three eigenvalues and therefore the seven element array could track two targets. Two targets were resolved for separations of 10 mm. The results indicate that MUSIC can be employed to track kidney stones although more elements would be required to track multiple fragments. [Work supported in part by NIH DK-43881].
Radiation pressure from ultrasound to help kidney stones pass.
Authors: Lu Wei , Cunitz Bryan W. , Kaczkowski Peter J. , Shah Anup , Sapozhnikov Oleg A. , Bailey Michael R.
Published: October 2009

Residual kidney stone fragments often remain months after treatment. These fragments may nucleate new stones and contribute to a 50% recurrence within 5 years. Here, a research focused ultrasound device was used to generate fragment motion with the goal of facilitating passage. Natural and artificial stones 1–8 mm in length were surgically placed in the urine space in pig kidneys. The ultrasound source was a 2.75-MHz, eight-element annular array with a 6-cm radius of curvature. At adjustable focal depths of 5–8 cm, the focal pressure beam width in water was about 2 mm, and peak pressure was about 4 MPa. Targeting was done by ultrasound using B-mode and twinkling artifact that stones produce in Doppler mode. The commercial imaging probe was placed within and oriented down the axis of the therapy probe. Ultrasound and fluoroscopy showed the stones moving in real-time under the influence of the focused ultrasound. Stones moved on the order of 1 cm/s away from the source and several stones moved several centimeters down the ureter. It appeared that stones were affected only when directly in the focal beam, perhaps indicating that radiation pressure not streaming caused the motion. [Work supported by NIH DK43881 and NSBRI SMST001601.]
Dependency of attenuation slope of tumor tissue on animal viability.
Authors: Haak Alexander , Hafes Zachary T. , Hall Timothy J. , O’Brien William D.
Published: October 2009

The attenuation slope (AS) is an acoustic parameter that can be utilized to characterize tissue. Previously, an algorithm was developed and validated on tissue mimicking phantoms that utilized the backscattered ultrasound signal to estimate the AS from single element pulse/echo data. Most AS data reported in literature have been acquired from dead tissue samples. There may be differences in the AS between live and dead tissues due to blood flow. Therefore, the dependency of the AS on animal viability is investigated. Fibroadenomas that developed spontaneously in Sprague Dawley rats were scanned with two single element transducers (3.5 and 7.5 MHz). The animal was then euthanized and the tumor was scanned again using the same parameters as in the previous scan. The data were postprocessed and the AS were estimated. The estimated AS of fibroadenomas for live animals ranged from 0.34–1.4 and 0.53–1.48 dB/cm MHz for 3.5 and 7.5 MHz, respectively. Similar AS results (0.3–1.1 and 0.5–1.4 dB/cm MHz for 3.5 and 7.5 MHz, respectively) were obtained for the euthanized animals. An analysis of variance was performed on the AS data and no significant difference in the AS estimated between live and dead animals was obtained. [Work supported by NIH Grant No. R01CA111289.]
Comparison of ultrasonic backscatter coefficient estimates from rat mammary tumors using single element and array transducers.
Authors: Wirtzfeld Lauren A. , Ghoshal Goutam , Hafez Zachary T. , Miller Rita J. , Sarwate Sandhya , Hall Timothy J. , Oelze Michael L. , O’Brien William D.
Published: October 2009

The ultrasonic backscatter coefficient (BSC) is the fundamental quantitative estimate from measurements that can be parametrized to yield the effective scatterer diameter and acoustic concentration. The ability to accurately estimate the BSC using different imaging systems (i.e., a system independent estimate) is significant for clinical application of QUS imaging. In this study, BSCs were estimated from spontaneous mammary tumors in rats using both single-element transducers and linear arrays from a clinical imaging system. The BSC as a function of frequency was computed from the rf backscattered signals from Sprague Dawley rats that developed either fibroadenoma or carcinoma tumors. The tumors were scanned using three single-element transducers with a collective −10-dB bandwidth of 1.4–18 MHz and two linear arrays from the Ultrasonix RP system with a collective −10-dB bandwidth of 2–8 MHz. For the single-element transducers, a smooth Plexiglas plate was used to acquire a reference scan. For the linear arrays, a well characterized tissue mimicking phantom containing spherical glass beads was used as a reference. Based on the respective reference scans, the BSCs were extracted from the data. The results between a laboratory system and a clinical imaging system showed good agreement. [NIH Grant R01CA111289.]
Quantitative estimation of complex shear modulus of liver and hydrogels through shear-wave imaging.
Authors: Orescanin M. , Qayyum M. A. , Toohey K. S. , Insana M. F.
Published: October 2009

A Doppler-based shear-wave imaging technique is described for estimating the complex shear modulus (μ+iωη) at shear-wave frequencies between 50 and 450 Hz. The developed technique involves a mechanical actuator that harmonically drives a stainless steel biopsy needle placed in the medium. Narrowband cylindrical shear waves are imaged using pulsed Doppler techniques. Shear moduli were computed from Doppler detected velocity. A phase gradient technique is applied to measure shear-wave speed. Spatially averaged speeds were numerically fit to a mathematical model relating dispersion and complex modulus. The proposed method was used to estimate complex shear modulus of the homogeneous three dimensional collagen hydrogels and of fresh and thermally-damaged porcine liver. The elastic shear modulus of 4% collagen hydrogel was measured, μ=640±14 Pa, using a commercial rheometer as a standard. The phase gradient approach yielded frequency-independent moduli μ=570±67 Pa and η=0.16±0.09 Pa s. For fresh liver shear-wave imaging between 50–300 Hz yielded frequency-independent moduli μ=1757±733 Pa and η=1.72±0.48 Pa s and μ=3851±3233 Pa and η=8.9±2.9 Pa s for thermally-damaged liver. Good agreement between the rheometer measurements and the shear-wave imaging approach shows that we can quantitatively estimate viscoelastic properties of the hydrogels and liver.
Generation of long pulses of focused ultrasound by time reversal system.
Authors: Fillinger Laurent , Sinelnikov Yegor , Sutin Alexander , Sarvazyan Armen
Published: October 2009

Ultrasound focusing based on time reversal acoustics (TRA) principles typically produces short ultrasound pulses having relatively low time-average intensity, which highly limits the therapeutic applications of these systems. The aim of this study was to explore theoretically and experimentally the possibility of generating long ultrasound signals by the TRA focusing. The TRA focusing of long signals requires application of broadband signals and depends on the signal bandwidth and the number of transducers. The radiation of the sequence of the TRA focused long signals makes it possible to focus ultrasound in a continuous mode. The theoretical model comprised several point sources located inside a liquid-filled reverberator. The model allowed calculation of the spatial structure of ultrasound field for the various focused signals. The dependence of the focused field spatial structure on the bandwidth of the signal as well as onto the number of ultrasonic transducers was investigated. Experiments were conducted using a water filled plastic bottle reverberator with five attached transducers. The effective TRA focusing of long signals was demonstrated for linear sweep and band-limited noise in the 200–800-kHz range.
Fast MATLAB-based ultrasound simulation software: Time harmonic calculations.
Authors: McGough Robert J. , VanderLaan Donald , Wong Joshua
Published: October 2009

A library of fast C++ routines is created for MATLAB simulations of time harmonic pressures generated by ultrasound phased arrays. These programs are included in the recently released FOCUS software package, where FOCUS is an acronym for “fast object-oriented C++ ultrasound simulator.” FOCUS is intended for large scale phased array simulations, characterizations of beamforming strategies, and analysis of new phased array designs. This software supports nearfield simulations of individual flat circular and rectangular pistons and focused spherical shells. Routines for pressure calculations with flat and curved phased array geometries consisting of circular and rectangular elements are also included. Within the FOCUS program, time harmonic simulations are performed with a combination of the fast nearfield method and the angular spectrum approach. Comparisons of errors and computation times with competing programs are evaluated for single element and phased array calculations, and results show that, for the same peak error value, FOCUS achieves an order of magnitude or more reduction in the computation time in large phased array simulations.
Fast MATLAB-based ultrasound simulation software: Transient calculations.
Authors: McGough Robert J. , VanderLaan Donald , Wong Joshua
Published: October 2009

The FOCUS software package quickly calculates transient pressure fields generated by ultrasound phased arrays. This software is useful for visualizing pressure wave propagation in the nearfield region, and the results are directly applicable to beamforming evaluations and phased array design for diagnostic and therapeutic applications. FOCUS includes support for transient pressure calculations with single transducers, including flat circular and rectangular pistons as well as spherically focused shells, and arrays of flat circular and rectangular transducers. Routines to calculate time delays for focusing the arrays are also included with the software. Transient calculations are performed with a combination of the fast nearfield method and time space decomposition, which achieves rapid convergence to reduce the numerical error while significantly reducing the computation time. Within the FOCUS software, time space decomposition is implemented with a simplified approach that decreases the overhead and reduces the computational complexity. The advantages of FOCUS over competing approaches include short computation times and small numerical errors. Examples of the supported element and array geometries will be shown, and sample codes will be provided. Transient pressure fields computed with FOCUS and with FIELD II are compared, and results show that, for the same peak error value, FOCUS is several times faster for calculations in the nearfield region.
The adoption of the accordion and other bellow-blown free-reed instruments in world cultures.
Authors: Wheeler Paul A.
Published: October 2009

Free-reed mouth organs, such as the Laotian khaen or the Chinese sheng, have been known for centuries in Asia. Several free-reed instruments (such as the accordion, concertina, and harmonium) using bellows rather than the mouth as a wind supply were developed in Europe during the 19th century. As European influence spread throughout the world, so did the adoption of the accordion-like instruments that penetrate local musical cultures around the world. This paper presents an overview of the usage of bellows-blown free-reed instruments in musical cultures of the world. It includes the zydeco accordion of New Orleans, the bandoneon (concertina) of the Argentine tango, and the harmonium of Hindustani music in India.
Characteristics and construction of the accordion.
Authors: Stankus John J.
Published: October 2009

The accordion family has a broad spectrum of members including piano accordions, chromatic button accordions, diatonic button accordions, bayans, buttonbox accordions, concertinas, and bandoneons. The common sound generator in all of these instruments is the free reed. This presentation will focus on the characteristics and construction of the accordion with emphasis on the tunings and the reeds. The configuration of the keys and buttons will be explained using the piano accordion. The application of the concert tuning and musette tuning will be contrasted. The versatility of the accordion will be shown as the combination of the four sets of reeds emulates the various instruments of the orchestra. The effect of tone chambers will be discussed. Accordion construction and its materials will be explained with photos and hands on parts. The materials and construction processes will be described with focus on the reeds and reed blocks. The 19th century construction of the accordion has not really changed. Some ideas will be presented for building a modern accordion utilizing the latest materials and processes. Examples of this are the use of carbon fiber molded case and reed manufacture using electrostatic discharge machining.
Comparison of noise levels between four hospital wings with different material treatments.
Authors: Wiese Cassandra H. , Wang Lily M. , Ronsse Lauren M.
Published: October 2009

Noise levels in four hospital areas of the University of Nebraska Medical Center in Omaha, NE, have been measured and compared to study the relationship between materials present in each of the zones and the resulting noise levels. The four areas evaluated, including three hospital wings and a neonatal intensive care unit, have varying levels of acoustic floor and ceiling treatments in the hallways and nurses’ stations. Sound pressure levels were logged every 10 s over a 24-h period in at least three different locations simultaneously within each area: at the nurses’ station, in the hallway, and in a nearby patient room. The resulting data were analyzed in terms of the hourly A-weighted equivalent sound levels (Leq) as well as various exceedance levels (Ln). Results indicate that the material finishes in a zone are found to affect the ambient sound levels within the nurses’ stations and hallways, while peak levels remain similar in most of the areas. Sound levels in the patient rooms, however, appear less correlated to materials in the hallway and nurses’ station and more impacted by the peak levels coming from those spaces as well as patient equipment.
Numerical model for a hydraulic in-line silencer.
Authors: Marek Kenneth A. , Earnhart Nicholas E. , Cunefare Kenneth A.
Published: October 2009

It is proposed that an in-line hydraulic silencer with a particular engineered lining can provide an effective alternative to other current silencing technologies. As a first step in validating this hypothesis, and to help with the design process, a numerical model is developed for a hydraulic silencer. The silencer consists of an annular dispersive medium with a rigid outer boundary, where fluid flow is directed through the annulus. Silencer characteristics, including input port impedance and acoustic transmission loss, are predicted by this model. Effects of silencer geometry and material properties, including nonlinearities, are discussed.
Angle dependent effects for impulse noise reduction for hearing protectors.
Authors: Murphy William J. , Khan Amir , Zechmann Edward L.
Published: October 2009

The proposed U.S. Environmental Protection Agency regulation for labeling hearing protection devices (HPDs) includes an impulsive noise reduction rating. In 2009, the American National Standards Institute Subcommittee for noise approved a revised standard for measuring the impulsive insertion loss of HPDs, ANSI/ASA S12.42-2009. The exposure at the ear in response to a forward-propagating wave depends strongly on the orientation of the head with respect to the direction of propagation. Furthermore, the insertion loss varies with the peak sound pressure level. This paper reports the results of tests performed using an acoustic shock tube to produce peak impulses of approximately 160-dB peak sound pressure level. Two manikins were evaluated: the GRAS KEMAR manikin equipped with 1/2- and 1/4-in. microphone in a GRAS 711 IEC coupler and the Institute de Saint Louis manikin equipped with a Bruel & Kjaer IEC 711 coupler equipped with a 1/4 in. microphone. The manikin heads were rotated through ±90 deg relative to the direction of the oncoming wavefront and impulsive peak insertion loss was measured according to S12.42-2009. [Portions of the research were supported by U.S. EPA Interagency Agreement No. 75921973-01-0.]
Ultrawideband filter for audible sound.
Authors: Kushwaha M. S.
Published: October 2009

Extensive band structure computation has been performed for two-dimensional periodic arrays of rigid stainless steel cylinders in air, with Bloch vector being perpendicular to the cylinders. It is proposed that the fabrication of a multiperiodic system in tandem could create a huge hole in sound within the human audible range of frequencies.
Robert T. Beyer: Thoughts and reflections from his family.
Authors: Beyer Rick
Published: October 2009

Thoughts and reflections on what made Robert T. Beyer such an inspiring and unforgettable person, through family stories, 50 years of correspondence, and memories that friends and colleagues have shared with his children.
Contributions of Robert T. Beyer to acoustical science: A review.
Authors: Foote Kenneth G. , Letcher Stephen V. , Moffett Mark B.
Published: October 2009

From his appointment as Instructor at Brown University in 1945 until well after his retirement, Beyer made significant contributions to the discovery and dissemination of knowledge about physical acoustics. Beginning with water and electrolytic solutions, Beyer and his students used ultrasonics to study relaxation processes in liquified gases, organic liquids, and single crystals, and in liquid metals. The 1951 review article with Markham and Lindsay [Markham, Beyer, and Lindsay, Rev. Mod. Phys. 23, 353411 (1951)] became a starting point for all ultrasonicists. Nonlinear acoustic phenomena also received special attention by Beyer and his students, who measured absorption and distortion of finite-amplitude waves; interaction of sound with sound, noise, and turbulence, including the first empirical verification of the parametric acoustic array in both water and air; and nonlinearity parameter of fluids as diverse as water, organic liquids, and liquid metals. New apparatus and methods were developed to aid the measurements, as of radiation pressure for the determination of absorption, a radio-frequency pulse system to measure distortion, and a complex wattmeter to measure intensity in a reactive acoustic field.
Robert Beyer’s laboratory in nonlinear acoustics: 1975–1982 and earlier perspectives.
Authors: Stanton Timothy K. , Korman Murray S.
Published: October 2009

Professor Robert T. Beyer’s laboratory at Brown University transitioned in the mid-1970s to conduct experiments (under Navy funding) on the nonlinear interaction of sound with noise (by T.K.S.) and, shortly after, on the nonlinear interaction of sound with turbulence (by M.S.K.). In this presentation, T.K.S. and M.S.K. recall their experiences in Beyer’s laboratory during their tenure as graduate students. Under Beyer’s guidance and support both graduate students (“unofficially” numbers 20 and 21) built up their experiments—mostly from scratch—but were able to use some remnants left over from an earlier era of graduate students, which included some intricately made ultrasonic transducers and precise mechanical staging. Beyer’s earlier graduate students’ dissertations also proved to be an invaluable source of knowledge. T.K.S. and M.S.K. developed considerable apparatus for their research. Prototype electronic equipment in that era involved mostly analog electronics because of the significant challenges associated with building the digital version at that time. The development of equipment including the complex acoustic watt meter (T.K.S.) and the hot film anemometer (M.S.K.) will be presented along with descriptions of their measurements of nonlinear scattering. Professor Beyer’s mentoring—the teacher, the historian, and storyteller—along with his fatherly guidance are heartfelt today.
Robert T. Beyer: A portrait in nonlinear acoustics.
Authors: Korman Murray S. , Muir Thomas G.
Published: October 2009

Robert T. Beyer’s early work in nonlinear acoustics is explored through his publications, his book, Nonlinear Acoustics, and his work with graduate students and colleagues. Beyer’s work was at the crossroads of the fundamental physics of nonlinear acoustics and his dogged pursuit of the B/A constant provided the essential information necessary for the field to develop.
Kinematic modeling and acoustic measures of breathy voice.
Authors: Samlan Robin A. , Story Brad H. , Bunton Kate
Published: October 2009

A variety of disordered vocal fold structural and vibratory parameters have been identified in patients with breathy dysphonia, although the contribution of each parameter to voice quality is unknown. This study is part of an effort to determine the relation between production and voice quality. A kinematic vocal fold model [based on I. R. Titze, Speech Commun., 8, 191–201 (1989)] that emulates the medial surface of the vocal folds was used to ascertain factors that lead to breathy voice. Based on the ratings of four expert listeners, five parameters were selected for further study. Combinations of multiple levels of adduction, bulging, nodal point, epilaryngeal area, and phase were used to generate 180 samples of sustained /ɑ/ and /ɪ/. Seven acoustic analyses were performed for each sample: maximum flow declination rate, maximum area declination rate, two measures of H1-H2, two measures of cepstral peak prominence, and noise to harmonics ratio. All parameters led to systematic changes in acoustic measures reflecting increasing breathiness, with several inconsistencies highlighting the uniqueness of each measure. The H1-H2 measures were inconsistent with the other data. Results will be described in light of how each measure can inform researchers about production. [Work supported in part by NIH R01-DC04789.]
Voice quality variation with fundamental frequency in English and Mandarin.
Authors: Keating Patricia , Shue Yen-Liang
Published: October 2009

Previous research has shown that F0 is positively related to H1*-H2* across male speakers of English [Iseli et al. (2006)] and to H1-H2 (after inverse filtering) within individual male speakers of Dutch [Swerts and Veldhuis (2001)]. That is, males who have overall higher-pitched voices generally have overall higher values of H1*-H2* (cross-speaker relation), and as an individual male’s F0 goes up, H1-H2 generally also goes up (within-speaker relation). The present study investigates both of these relations, cross-speaker and within-speaker, for male and female speakers of English and Mandarin, and extends them to a large set of voice quality measures. The speech samples consist of repeated rising and falling tone sweeps, in which speakers began at a self-selected comfortable pitch, and then swept either up or down in pitch to their highest or lowest comfortable pitch. The beginnings of the sweeps are tested for cross-speaker relations, while the entire sweeps are tested for within-speaker relations. VOICESAUCE, a new program for voice analysis, is used to extract F0, energy, cepstral peak prominence, formants and bandwidths, and a variety of harmonic amplitude measures. Many measures are shown to be strongly related to F0. [Work supported by NSF.]
Modeling segmentation precision and inter-segmenter variability.
Authors: Stojanovic Diana
Published: October 2009

Inter-segmenter differences have been reported in the literature to affect the measurements of segment durations [Allen (1975)] and more specifically rhythm metrics [Mairano and Romano (2007)]. Segmenters may differ in criteria used for segmentation as well as resolution of uncertainty due to gradual change between segments (such as between nasalized vowel and coda nasal). In addition, small variations in the location of segmentation border (one period of the waveform, or 5–10 ms for adult speakers) are likely both between two segmenters and between different segmentation sessions of the same segmenter. This study investigates the effect of segmentation errors on the classification power of rhythm metrics. Errors are modeled to represent (1) small random variation and (2) resolution of larger interval uncertainties (20–30 ms). Simulations using the two scenarios were performed on samples of manually segmented speech. Results show that errors in scenario-1 do not affect the separation of languages in (%V, stdC), (rPVI-C, nPVI-V), and (%V, VarcoV) space, which implies that precision of less than 5 ms is not required in studies that use these measures. Errors in scenario-2 may be large enough to shift a language closer to the group it does not belong, depending on the frequency of such errors.
Aerodynamic modeling for concatenative speech synthesis.
Authors: McGowan Kevin B.
Published: October 2009

Listeners can perceive and use a wide array of fine-grained phonetic details, including the detailed coarticulatory influences of adjacent sounds, when perceiving speech. Details like anticipatory nasalization in can, for example, potentially provide the listener with a rich network of informative cues and are a key to understanding listeners’ ability to disambiguate speech sounds from seemingly ambiguous input. Unfortunately, these coarticulatory cues are generally missing or contradictory in the output of speech synthesis systems. These systems work by concatenating variable-length sound units chosen from a large database of recorded speech. Units are chosen to minimize two functions: the cost of aligning a particular unit with the desired speech output (target cost) and the cost of adjoining the next sound to the most recently selected unit (join cost). Generally, these costs are calculated using features which can be automatically extracted from the acoustic speech signal. A unit selection database is created, automatically segmented and automatically labeled with nasal and oral airflow feature vectors. These aerodynamic features are used as a proxy for articulatory information in the calculation of join and cost functions. Listeners’ mean opinion scores are obtained on output from this system and a baseline acoustic system for comparison.
Normalization for vocal tract differences using long term average spectrum.
Authors: Vitela Antonia D. , Lotto Andrew J. , Story Brad H.
Published: October 2009

Differences in vocal tract size and anatomy lead to substantial differences in acoustic realization of vowels. Much of the variability in vocal tract area functions is captured in the area function of the “neutral” (non-articulating) vocal tract of each talker. A listener could perceptually normalize talker variability substantially by extracting the neutral vocal tract and using it as a referent. Alternatively, the listener could extract the auditory representation of the output of the neutral tract and use it as a referent for the auditory representation of the vowel space. We will present such a model of perceptual talker normalization for vowels and demonstrate its effectiveness for a variety of vocal tract shapes and lengths. Additionally, we will provide evidence that this normalization may be implemented in an auditory process that (i) extracts the long term average spectrum from running speech (providing an estimate of the neutral vowel), (ii) represents new input relative to this spectral average (providing normalization), and (iii) is not specific to speech. Additional analyses and perceptual results will demonstrate how this model can extend to normalization of consonant perception. [Work supported by NIH/NIDCD to A.J.L. and B.H.S. and Diversity Supplement to A.D.V.]
Analysis of the human phonation with different vocal fold geometries by applying a two-dimensional fully coupled fluid-solid-acoustic finite element scheme.
Authors: Zoerner Stefan , Kaltenbacher Manfred , Lerch Reinhard , Dllinger Michael
Published: October 2009

A numerical two-dimensional model is presented simulating the human laryngeal voice production. Air flowing through the larynx interacts with the structural mechanics of the vocal folds forcing them to vibrate and induce sound. In turn, the vibrations of the vocal folds cause the airflow to pulsate which acts as a further sound source inside the larynx. This coupled field problem is modeled by utilizing an enhanced finite element method. The presented approach takes all three involved physical fields consisting of fluid mechanics, solid mechanics, acoustics, and their interactions fully into account. The results of these simulations clearly demonstrate the impact of different vocal fold geometries on the fluid field, e.g., changing the occurrence of the Coanda effect. In addition, influences on the produced sound signal resulting in modified main acoustic frequencies are observed. An additional eigenfrequency analysis on the vocal fold model was performed showing that the first eigenmode and the vibrational frequency in the transient case are identical. The first eigenmode was in the range of about 100 Hz depending on the vocal fold geometry.
Articulatory characteristics of coronal consonants in Argentine Spanish: An electropalatographic study.
Authors: Kochetov Alexei , Colantoni Laura
Published: October 2009

Previous articulatory investigations of Spanish have been largely limited to its Peninsular varieties. This study uses electropalatography (EPG) to investigate articulatory characteristics of coronal consonant contrasts in Argentine Spanish as part of a larger project examining phonetic variation across Spanish dialects. Simultaneous EPG and acoustic data were collected from four female speakers of Buenos Aires Spanish reading sentences with various intervocalic coronal consonants. Results revealed consistent differences in terms of anterior/posterior tongue placement and the amount of linguopalatal contact with the primary distinction between (denti-)alveolar and post-alveolar articulations within the classes of stops/affricates, fricatives, nasals, and laterals/rhotics. Inter-speaker variation was observed, however, in the articulation of some consonants, namely, in the apical or laminal realization of the post-alveolar fricative and in the palatal or palatalized alveolar realization of the post-alveolar nasal. These findings diverge from those reported for Peninsular Spanish [A. M. Fernández Planas, Estudios de fonética experimental, 16, 11–80 (2007)], revealing some dialectal differences in the degree of fronting of denti-alveolars. Additionally, results suggest a sound change in progress in Argentine Spanish, such as the depalatalization of the palatal nasal. [Work supported by Connaught, University of Toronto.]
An acoustic and electroglottographic study of breathy phonation in Gujarati.
Authors: Khan Sameer ud Dowla
Published: October 2009

While it has long been established that breathy and modal vowels in Gujarati can be reliably distinguished based on the H1-H2 measure [Fischer-Jorgensen (1967); Bickley (1982)], and that Gujarati listeners attend solely to H1-H2 when distinguishing phonation types in Gujarati and in other languages [Bickley (1982); Esposito (2006)], new research [Khan and Thatte (2009)] suggests that in more casual speech styles, H1-H2 may play a smaller role in distinguishing breathy-modal minimal pairs. In such speech styles, additional acoustic measures (e.g., H1-A3, CPP, and rms energy) are more effective in distinguishing breathy and modal vowels for some speakers. To more closely examine the phonetics of breathy vowels in casual speech, the current study examines both acoustic and electroglottographic data collected from naturalistic productions of breathy-modal minimal pairs. Preliminary data suggest that while all speakers distinguish modal and breathy vowels, the strategies used by each speaker to produce the contrast vary considerably. [Work supported by NSF.]
An acoustic and electroglottographic study of White Hmong phonation.
Authors: Esposito Christina M. , Ptacek Joseph , Yang Sherrie
Published: October 2009

This study examines the phonation of White Hmong, a language with seven tones (traditionally described as high, mid, low, high-falling, mid-rising, low-falling, and mid-low), five of which are associated with modal phonation, and two of which are associated with non-modal phonation; the low-falling tone is creaky and mid-low tone is breathy. Thirty-three speakers were recorded producing words with all seven tones; 12 also made electroglottographic (EGG) recordings. Acoustic measures were cepstral peak prominence (CPP) and harmonic amplitudes H1* and H2*, H1*-H2*, H1*-A1*, H1*-A2*, H1*-A3*, and H2*-H4*. EGG measures were closed quotient (CQ) and peak-closing velocity (PCV). Measures were made automatically using VOICESAUCE and PCQUIRERX. Results showed that none of the measures tested distinguished all three phonation types. However, several measures distinguished two categories: H1 distinguished creaky versus non-creaky, H1-H2 distinguished breathy from creaky, and the EGG measures CQ and PCV both distinguished breathy from non-breathy. H1*-A1*, H1*-A2*, H1*-A3*, and H2*-H4* did not distinguish any of the phonation types. This suggests that phonation contrasts are realized across several phonetic dimensions in White Hmong. In addition, there was a gender difference in the production of phonation, with females having significantly higher CPP than males, suggesting that female phonations are noisier. [Work supported by NSF.]
Relationships among vowel formant discrimination, discrimination of intensity and frequency, and frequency selectivity.
Authors: Guempel Leah , O’Brien Kathleen , Liu Chang
Published: October 2009

The goal of the present study was to investigate if vowel formant discrimination could be accounted for by intensity discrimination and frequency discrimination as well as by frequency selectivity. Intensity discrimination and frequency discrimination were measured for tonal signals at 250, 500, 1000, and 2000 Hz presented at 70-dB SPL for 21 young normal-hearing listeners with American English as their native language. Auditory filter was measured using notched noise method at the four frequencies above. In addition, thresholds of vowel formant discrimination were examined for F1 and F2 of three English vowels for the same group of listeners. Overall, thresholds of vowel formant discrimination had significantly modest correlation with intensity discrimination, but no significant correlations with frequency discrimination and frequency selectivity. These results indicate that vowel formant discrimination may be associated with intensity discrimination of harmonics in speech sounds, but not determined by frequency discrimination and frequency selectivity. Other factors accounting for vowel formant discrimination will be discussed.
Segregation of co-channel speech signals using a least-squares approach.
Authors: Vishnubhotla Srikanth , Espy-Wilson Carol
Published: October 2009

In this work, an approach to segregate overlapping speech signals from a single-channel recording is presented, and its performance demonstrated for the two-speaker case. The approach relies on a previous algorithm to estimate the pitch frequencies of the participating speakers and then identify their voiced regions. Depending on the voiced-unvoiced combination of the two speakers, an appropriate over-determined system of equations is set up which is then solved to unravel the contributions of the two speakers to the speech mixture. In the voiced-voiced case, the harmonics of the two pitch frequencies are used to set up the basis set. For the voiced-unvoiced and unvoiced-unvoiced cases, a wavelet-based approach is used to construct the basis set. The segregated contributions are then assigned to the appropriate speakers by imposing the temporal continuity of the mel-frequency cepstral coefficients of these separated speech segments. The perceptual quality of the segregated speech is evaluated using the PESQ measure, which shows a significant improvement compared to the mixture signal. On evaluation on a speech recognition task, the algorithm shows comparable or better performance than the state-of-the-art algorithms, even when the target speaker of interest is significantly weaker than the masking speaker.