bluuebunny/crossref_metadata_2025 huggingface dataset:
Psychoacoustic audio analysis in frequency domain.
Authors: Osses Vecchi Alejandro , Catalán Víctor Espinoza , Ángel Takaoka Hernán
Published: October 2010

When evaluating the loudness sensation of an acoustical signal, the concept of energy summation in terms of the frequency components of the signal must be reformulated. There are 2 standardized methods for the calculation of the loudness of a signal. Both methods are based on the decomposition of the signal into frequency bands and consider an amplitude axis in Sones. As an approximation to those methodologies, a model is used, which conjugates the concepts of equal-loudness-contours and the Terhardt masking model for obtaining an amplitude weighting in Phons; meanwhile the energy summation is carried out using a Riemann sum approximation to integrate the specific loudness curve. Some results presented for this methodology are applied to audio signals and compared with the traditional one-third-octave band analysis.
Connecting the Kernel for a music language (KL) with a new graphic component.
Authors: Peña-Guerrero Maximino
Published: October 2010

One of the problems facing our Kernel for a music language (KL) is that it does not have a graphic interface that allows showing music scores on screen with a professional quality. KL is a compiler designed for music digital processing. It contains functions for the lexical, syntactic, and semantic analyses. The KL input is a file (or console) with a command sequence (f.e., do do sol do do reb) with which a song is described. The KL output is in essence sound, MIDI, and graphic music elements for console and printer. In this document a computer methodology is presented, which adds to the KL a new semantic function that allows drawing a new musical figure into a graphic screen. The result is a staff that leaves drawing with these new agreement elements with the instructions that it contains a KL data file.
Active noise barrier minimizing pressure gradient.
Authors: Hart Carl R. , Lau Siu-Kit
Published: October 2010

Minimization of the sound pressure field within the shadow zone of a noise barrier is achieved by reducing the pressure gradient along a line, at the top of a barrier, via active noise control. The noise control effectiveness of a barrier is increased by this strategy, especially for specific system configurations. The proposed method was evaluated by numerical simulation. Results indicate that system orientation has little effect on minimizing the pressure gradient at the top of the barrier when the error sensors are invisible to the primary noise disturbance. Highly effective control within the shadow zone and close to the barrier is possible when the system is oriented at an angle where two or more error sensors are in line with the first diffracting edge and the primary noise disturbance. Increasing the spatial extent of the quiet zone is possible by increasing the number of control sources, where the error sensors have a line of sight with the primary noise disturbance. [Work supported by a fellowship (CRH) and grant (SKL) from the University of Nebraska - Lincoln.]
A new speech corpus for studying subglottal acoustics in speech production, perception, and technology.
Authors: Lulich Steven M. , Morton John R. , Sommers Mitchell S. , Arsikere Harish , Lee Yi-Hui , Alwan Abeer
Published: October 2010

Subglottal resonances have received increasing attention in recent studies of speech production, perception, and technology. They affect voice production, divide vowels and consonants into discrete categories, affect vowel perception, and are useful in automatic speech recognition. We present a new speech corpus of simultaneous microphone and (subglottal) accelerometer recordings of 25 adult male and 25 adult female speakers of American English (AE), between 22 and 25 years of age. Additional recordings of 50 gender-balanced bilingual Spanish/AE speaking adults, as well as 100 child speakers of Spanish and AE, are under way. The AE adult corpus consists of 35 monosyllables (14 “hVd” and 21 “CVb” words, where C is [b, d, g], and V includes all AE monophthongs and diphthongs) in a phonetically neutral carrier phrase (“I said a ____ again”), with 10 repetitions of each word by each speaker, resulting in 17 500 individual microphone (and accelerometer) waveforms. Hand-labeling of the target vowel in each utterance is currently under way. The corpus fills a gap in the literature on subglottal acoustics and will be useful for future studies in speech production, perception, and technology. It will be freely available to the speech research community. [Work supported in part by the NSF.]
Quantifying prosodic boundary strength using functional data analysis of articulatory movement.
Authors: Parrell Benjamin , Lee Sungbok , Byrd Dani
Published: October 2010

Prosodic boundaries act to locally slow speech, resulting in lengthened articulatory and acoustic durations. The amount of lengthening has been shown to differ based on the juncture strength; however, previous studies disagree on the quantitative characteristics of and qualitative number of prosodic boundaries. Piecewise durational analyzes of articulatory trajectories give only one view of the nature and types of phrasal junctures. Recent work has proposed an alternative to this approach, based on functional data analysis (FDA). [Lee et al., J. Acoust. Soc. Am. (2006).] FDA offers a route to move beyond examining piecewise intervals in the articulatory kinematics, such as constriction duration, to examining the entire time course of a kinematic trajectory to derive a time warping function that characterizes the local non-linear deformations of articulatory trajectories as a boundary is approached and as it recedes. The use of FDA, and particularly of the deformation index—a measure of local time warping developed by our team—has been shown to distinguish boundaries of different strengths in speech generated using articulatory synthesis [Parrell et al., Speech Prosody (2010).] In the current study, we extend this technique to non-synthetic articulatory data, using the deformation index to distinguish and quantify different types of prosodic boundary. [Work supported by NIH].
Exploring the role of production in predicting vowel inventories.
Authors: Padgett Jaye , Sanders Nathan
Published: October 2010

Since Liljencrants and Lindblom (1972), researchers have attempted to predict vowel inventories from assumptions about auditory perception [e.g., Lindblom (1986); Schwartz et al. (1997); de Boer (2000)]. Our point of departure is Schwartz et al.’s dispersion-focalization theory (DFT), which calculates the optimality of a vowel system as a weighted combination of dispersion and focalization. Like other dispersion-based models, this model has considerable strengths, but also some shortcomings. One is its inability to derive “vertical” vowel systems or even systems with the phoneme schwa. We explore whether this problem can be resolved by extending DFT to include a production-based parameter that models the articulatory cost of vowels. This attempt depends on several variables, e.g., what articulatory cost is assigned to each vowel; how total articulatory cost is weighted with respect to perceptual parameters. On the one hand, the articulatory parameter does not improve the model’s overall match rate with attested inventories. On the other, a qualitative improvement is that three-vowel inventories now include the Kabardian “vertical” system, and four-vowel inventories now include schwa, as in Paiwan and Yupik. However, larger systems still lack schwa, a problem remaining that can hopefully be solved by further exploration of the model’s properties.
KlattWare tools for acoustic analysis of speech signals.
Authors: Truslow Eric O. , Hanson Helen M.
Published: October 2010

Computer-based tools are indispensable for acoustic analysis of speech signals in the time and frequency domains. One of the oldest of these tools was developed by Dennis Klatt for use at the MIT Speech Group. Other programs, both commercial and shareware, are now in popular use on Windows and Mac platforms. Although these programs provide commonly needed functions, Klatt’s program, now called XKL, has useful functions not provided by these other tools. These functions include spectra that are averaged throughout a time window, spectra that are averaged across waveforms, and smoothed spectra. In addition, his spectrogram algorithm produces particularly nice spectrograms. Therefore, it has been a goal to make XKL more easily available to the speech community. To this end, XKL has been ported to the Mac platform, and labeling and F0 contour functions have been added. Other changes will be described and information on accessing the program will be provided.
The importance of optimal parameter setting for pitch extraction.
Authors: Evanini Keelan , Lai Catherine
Published: October 2010

In this study we present a performance comparison for five pitch extraction algorithms: auto correlation (AC), cross correlation (CC), and sub-harmonic summation (as implemented in PRAAT [Boersma and Weenick (2010)]), the robust algorithm for pitch tracking (RAPT) implemented in ESPS [Talkin (1995)], and SWIPE’ [Camacho (2007)]. Recent research showed that SHS and SWIPE’ outperformed the other algorithms on two speech databases with EGG reference values [Camacho (2007)]. That study, however, used a fixed search range of 40–800 Hz for all speakers, regardless of sex or speaker-specific pitch characteristics. In the current study, we adopt the parameter optimization strategy from De Looze and Rauzy (2009) to calculate specific pitch floor and ceiling values for each speaker. Our results show a substantial improvement in accuracy of the AC, CC, and RAPT algorithms when the optimized parameters are used (especially for the female speakers), and all five algorithms show similar performance. The gross error rate for all five algorithms ranges from 0.1% to 0.3% (N=18 098) on the FDA database [Bagshaw (1994)] and from 0.2% to 0.4% (N=11 527) on the KPD database [Plante et al. (1995)]. Our study thus highlights the importance of pre-processing the speech signal to determine optimal speaker-specific parameters for pitch extraction.
The voices of Antoni Gaudi’s architecture.
Authors: Campos Arturo
Published: October 2010

Antoni Gaudi is one of the most important architects in the history of architecture. His architecture has increased the beauty of Barcelona. The civil and religious architecture of Gaudi involves mystic sounds, real or imaginary sounds, whispers almost imperceptible, and sonorous surprises in the journeys destined to peace and spiritual calm. In this research, which come from the Ph.D. thesis “The voices of Gaudi,” the sound of his buildings is explained. Their acoustical phenomena are produced by his architecture and the materials. The acoustics and sounds of three important buildings “La Sagrada Familia,” “La Pedrera,” and “Park Güell” will be explained in the presentation. “The Sagrada Familia” in Barcelona is a church where the sonorous itineraries exist and where the acoustic design of choirs, bells, and bell towers was an essential idea of Antoni Gaudi. “La Pedrera” is a living building with its voice singing with the people and “Park Güell” with the sounds of people, fountains, and rustic materials. Architecture and acoustic design transform Gaudi’s architecture in a daily murmur, highly spiritual.
Open-source software for modeling sound propagation in the ocean.
Authors: Porter Michael B.
Published: October 2010

In the early 1990s web browsers were just emerging, leading us to the now obvious conclusion that the stand-alone computers in offices around the world could be woven into a distributed web of resources to form a globally shared library. With the support and encouragement of the U.S. Office of Naval Research, we established the Ocean Acoustics Library to serve as a means of publishing software of general use to the international ocean acoustics community. This paper will survey the available tools, with an aim also to explain the evolutionary process that has led to a set of models with quite different, but survivable characteristics.
Integrating techniques for identifying individual sperm whales acoustically.
Authors: Sidorovskaia Natalia , Tiemann Christopher O. , Ioup George E. , Ioup Juliette W.
Published: October 2010

Individual sperm whales may be identified acoustically from the properties of individual clicks or from the characteristics of their click trains. Thus far, our group has used cluster analysis (mainly self-organizing maps) and click change detection to identify whales by emphasizing the properties of individual clicks. Click change detection addresses the problem of click changes due to changing aspect (a turning whale) or other factors. Cadence analysis has been used to separate the clicks of individual whales based on click train properties. Localization has also been used to separate the clicks of individual whales. Manual click train identification employs both individual click properties and sequence analysis to identify individual whales. Each of these techniques has shown promise. Some have been compared on the same data set. In this work, multiple methods are applied to the same data sets in order to verify the methods and to develop a multi-tiered approach to identifying individual sperm whales. [Research supported by SPAWAR and ONR.]
Noise inversion: Making use of the signed impulse response.
Authors: Harrison Chris
Published: October 2010

The impulse response of sediment layering can be obtained by cross correlating ocean ambient noise from the steered beams of a vertical array. Given a moving array the envelope of this signal (i.e., the Hilbert transform) maps out a sub-bottom profile. However, there is still additional information in the signed noise amplitude, and this will be demonstrated with experimental data from a number of sites at which a vertical array was moored. Essentially the impulse response is equivalent to that of any active system (the effective transmit pulse being determined by the user-defined processing band) and, consequently, this process is sensitive to changes in acoustic impedance. The quality of the signal can be improved by adaptive beam forming but although it has been shown that, strangely, this reverses the sign [Harrison, J. Acoust. Soc. Am. 125, 3511–3513 (2009)] here we show it does not alter its fidelity. In fact, if the number of layers is constrained as often done in inversion, it is possible to resolve their separations too even if smaller than the effective transmitted pulse length. More information is also available in the comparison of up/down beam powers at angles other than vertical.
Cross-correlation of diffuse surface noise between horizontally separated sensors.
Authors: Fried Stephanie E. , Walker Shane C. , Kuperman William A.
Published: October 2010

The use of ambient noise in the ocean to extract the time of arrival structure of the time domain Green’s function (TDGF) between horizontally separated sensors was first demonstrated in deep water using shipping noise [Roux et al., J. Acoust. Soc. Am. 116, 1995 (2004)]. Subsequently, surface noise recorded along a vertical array was used in the passive fathometer application [Siderius et al., J. Acoust. Soc. Am. 120, 1315 (2006)]. A noise field from primarily volumetrically distributed biological noise sources recorded along a horizontal array was also shown to extract a shaded TDGF in very shallow water [Fried et al., J. Acoust. Soc. Am. 124, EL183 (2008)]. This same noise correlation function (NCF) processing is now demonstrated along horizontally separated hydrophones using diffuse, low sea-state surface noise, where the spectrogram of the noise reveals no significant shipping component. These results are presented along with a discussion of the relevant signal processing as well as the theory and simulation used to understand them. Of particular interest is the build-up time of the NCF under these relatively homogeneous environmental conditions as opposed to what other less homogeneous situations might produce.
Retrieval of deterministic ray travel times from noise cross-correlations.
Authors: Godin Oleg A. , Zabotin Nikolay A.
Published: October 2010

Two-point correlation functions of diffuse wave fields are known to approximate the deterministic Green’s functions that describe wave propagation between the two points. The deterministic ray travel times retrieved from cross-correlations of ambient noise recorded on vertical line arrays of hydrophones serve as the input data for passive acoustic thermometry and tomography of the water column. Accuracy of the sound-speed and water temperature inversions is largely controlled by the accuracy of the ray travel time measurement. Achieving oceanographically relevant results requires the relative error of the passive travel time measurements to be less than 0.1%. Often, the absolute error of the travel time measurement needs to be small compared to the acoustic wave period. In this paper, stationary-point analysis of an asymptotic, high-frequency integral representation of the noise cross-correlation function is used to determine ways for accurate retrieval of ray travel times. Limitations on the accuracy of the travel time retrieval due to noise anisotropy, medium non-stationarity, source and receiver motion, and finite noise averaging times are discussed. [Work supported by ONR.]
On the use of three-dimensional Doppler acquisition for real-time volume flow estimation.
Authors: Desai Tej , Pinter Stephen Z. , Fowlkes J. Brian , Rubin Jonathan M. , Zhang Man , Hall Anne L. , Kripfgans Oliver D.
Published: October 2010

Clinical volumetric blood flow estimation relies on several assumptions. Among them are cylindrical vessel geometry, symmetric flow profile, and Doppler angle. None of them are known well enough to obtain clinically relevant estimates. 3-D color flow acquisition circumvents these assumptions, posing a viable tool for in vivo blood volume flow analysis. A 4-D cardiac scanner operating a 2-D array for real-time 3-D color flow imaging [GE Healthcare, Milwaukee, WI] was used. The array was positioned to fully intersect a 2-cm-diameter flow tube with the constant-depth plane (CPlane). Blood mimicking fluid was circulated at up to 6 l/min using a cardiac bypass pump (60 and 80 beats/min). A trigger source synchronized the pump and scanner. Data volumes were acquired equally spaced throughout the cardiac cycle. Temporally resolved volume flow was derived from CPlane data integration using Doppler power partial volume correction. Results show less than 7% mean flow error for temporally resolved volume flow (100 points per cardiac cycle and 50 averages). Single points in the cardiac cycle (10 max total), including systolic and diastolic flow, can be acquired within 50 heartbeats. Average, nontime-resolved flow can be estimated within 5 s.
Principle and performance of the transverse oscillation vector velocity technique in medical ultrasound.
Authors: Jensen J. A. , Pihl M. J. , Udesen J. , Pedersen M. M. , Hansen K. L. , Nielsen M. B.
Published: October 2010

Medical ultrasound systems measure the blood velocity by tracking the blood cells motion along the ultrasound field. This is done by pulsing in the same direction a number of times and then finding e.g., the shift in phase between consecutive pulses. Properly normalized, this is directly proportional to the axial blood velocity. A major drawback is that only the axial velocity component is found. Often the lateral component is most important as blood vessels run parallel to the skin surface. The talk presents the transverse oscillation approach, which also can find the lateral velocity component by using a double oscillating field. A special estimator is then used for finding both the axial and lateral velocity components, so that both magnitude and phase can be calculated. The method for generating double oscillating ultrasound fields and the special estimator are described, and its performance revealed for a flow rig setup. Several examples from the clinical use of the approach are shown. From these it is seen that both velocity magnitude and angle varies temporally and spatially across the cardiac cycle, and it is, thus, important to estimate both continuously over the image region and time.
Graduate acoustics at University of Texas at Austin.
Authors: Blackstock David T. , Hall Neal A. , Hamilton Mark F. , Hixson Elmer L. , Wilson Preston S. , Haberman Michael R. , Isakson Marcia J. , Knobles David P. , Penrod Clark S.
Published: October 2010

While graduate study in acoustics takes place in several colleges and schools at the University of Texas at Austin (UT Austin), including Communication, Fine Arts, Geosciences, and Natural Sciences, this presentation focuses on the acoustics program in Engineering. The core of this program resides in the Departments of Mechanical Engineering (ME) and Electrical and Computer Engineering (ECE). Acoustics faculty in each department supervise graduate students in both departments. One undergraduate and seven graduate acoustics courses are cross-listed between ME and ECE. Instructors for these courses include staff at Applied Research Laboratories at UT Austin, where many of the graduate students in acoustics hold research assistantships. The undergraduate course, offered every fall, begins with basic physical acoustics and proceeds to draw examples from different areas of engineering acoustics. Three of the graduate courses are offered every year, a two-course sequence on physical acoustics and a course on acoustic transducers. The other four graduate courses, offered in alternate years, are on nonlinear acoustics, underwater acoustics, architectural acoustics, and ultrasonics. An acoustics seminar with invited speakers is held most weeks during the academic year, averaging over 10 per semester. The ME and ECE departments both offer a Ph.D. qualifying exam in acoustics.
Sound synthesis of the harpsichord pluck using a physical plectrum-string interaction model.
Authors: Perng Chao-Yu J. , Smith Julius O. , Rossing Thomas D.
Published: October 2010

The harpsichord, in a broad sense, refers to the family of plucked keyboard instruments. The sound of the harpsichord tone is unique and is not mistaken for other plucked string instruments, such as the guitar. The synthesis of harpsichord sound has been done previously [Valimaki et al., \ EURASIP J. Appl. Signal Process. 7, 934–948 (2004)], where excitation signals are extracted through recorded tones. In this work, we present a revision to the harpsichord plectrum model that we proposed earlier [Chao-Yu J. Perng et al., J. Acoust. Soc. Am. 127, 1733(A)–(2010)], one which incorporates basic friction models. The harpsichord plectrum-string interaction model is used for the sound excitation, which is then implemented with a digital waveguide to produce the synthesized plucked string tone. Our harpsichord plectrum-string interaction model allows for controllability of physical parameters that allow for a range of expressiveness. The effects in which changing the physical parameters have on the synthesized pluck sound are compared and discussed.
Noise propagation through open windows of finite depth into an enclosure.
Authors: Sieck Caleb F. , Lau Siu-Kit
Published: October 2010

Predicting the insertion loss of an opening backed with an enclosed space is important for building noise control. Recent research in sound transmission through apertures of finite depth in infinite rigid baffles has included the effects of propagating and evanescent modes within the aperture in order to extend models to higher frequencies. The present study extends the model to the case of the aperture backed by a cavity as opposed to sound radiating into half-space. The role of coupling between the aperture modes, radiation modes, and cavity modes in the transmission was investigated. The results were compared to those of previous models which neglected the depth of the aperture and finite element modeling using COMSOL Multiphysics. Comparisons show that the current model is effective at predicting the sound transmission loss through the aperture and the acoustic field within the cavity for an obliquely incident plane wave. By changing impedance conditions on the half-space side of the aperture and within the aperture, the model has been used to evaluate passive noise control techniques.
Revaluation of urban sounds in Barcelona.
Authors: Jimena De Gortari
Published: October 2010

This paper summarizes the results of the ongoing Ph.D. thesis “The revaluation of urban sounds in the Gothic Neighborhood, Barcelona, Spain,” which will be presented by September 2010. Acoustic urban space quality can be read from multiple indicators such as the configuration of the built environment, the residents’ perception, and perceived sounds, among others. In order to find those indicators, several elements where put together on a single methodology, so they could be tested at certain public spaces within Barcelona’s gothic neighborhood. One of the main results is the importance given by the people to some of the analyzed areas as places to rest from the noise of the city, although these places exceed the recommended noise values. Furthermore, the variety of nice sounds that people perceive, despite the prevailing background noise during most part of the day, should be highlighted.
Acoustic scattering of a high-order Bessel vortex beam of fractional type α incident upon a rigid immovable sphere.
Authors: Mitri F. G.
Published: October 2010

A closed-form analytical solution for the acoustic scattering of a new class of Bessel beams, termed non-diffracting Bessel vortex beams of fractional type α, is derived. This new class of Bessel beams preserves the same non-diffracting feature of conventional high-order Bessel vortex beams of integer order. The far-field acoustic scattering field by a rigid sphere centered on the beam axis is expressed as a partial wave series involving the real number α, the scattering angle relative to the beam axis θ, and the half-conical angle β of the wave number components of the beam. Unlike the acoustic scattering properties of conventional high-order Bessel vortex beams of integer order [F. G. Mitri, Ann. Phys. NY, 323, 2840–2850 (2008); P. L. Marston, J. Acoust. Soc. Am. 124, 2905–2910 (2008)], both the acoustic forward and backward scattering of Bessel vortex beams of fractional type α by a rigid sphere do not vanish unless α becomes an integer number. The novel properties of non-diffracting Bessel vortex beams of fractional type α may lead to the development of an “acoustic blender” with possible applications in particle rotation, mixing, and manipulation. [Work supported by LANL-LDRD# X9N9.]
Analytical and experimental study on active control of structurally radiated sound from an elastic cylindrical shell.
Authors: Jin Guoyong , Shi Shuangxia
Published: October 2010

This paper presents an analytical and experimental study on active control of structurally radiated sound from an elastic cylindrical shell. An analytical model is developed for the active structural acoustic control (ASAC) of the cylindrical shell. The structural response of a cylindrical shell to the harmonic point force is derived by applying the modal supposition method. Expressions for the far-field radiated sound pressure and radiated sound power are derived by using the Kirchhoff–Helmholtz integral and the appropriate Green’s function. Both global and local control strategies are considered. The optimal control forces corresponding to each control strategy are obtained by using the linear quadratic optimal control theory. Numerical simulations are performed to examine and analyze the control performance under different control strategies. Results show that using point force as the control input of the ASAC system can achieve the global sound attenuation of the cylindrical shell at resonance frequencies. Better control performance can be obtained under the control strategy of minimization of the radiated sound power in terms of the radiated sound power. However, control spillover may occur at off-resonance frequencies with the structural kinetic energy minimization strategy. ASAC experiment was implemented, good agreement was observed between the numerical and experimental results, and successful attenuation of structural vibration and radiated sound was achieved.
Cross-modal talker familiarity: The search for amodal lexical episodes.
Authors: Sanchez Kauyumari , Rosenblum Lawrence D.
Published: October 2010

Familiarity with lip-read (visual-only) speech from a talker facilitates perception of heard (audio-only) speech-in-noise from that same talker [L. Rosenblum et al., Psychol. Sci. 18, 392 (2007)]. This could suggest that stored speech episodes are composed of lexical items that retain amodal talker-specific characteristics. However, the Rosenblum et al. study used sentential material which differed between the lip-reading and speech-in-noise tasks. That study did not examine whether cross-modal talker facilitation was based on lexical items used in both tasks. The current experiment addressed whether amodal talker-specific characteristics are stored as lexical episodes by modifying Rosenblum et al.’s design. Subjects were first asked to lip-read words from a talker and were then asked to listen to words embedded in noise spoken by the same or a different talker. Preliminary evidence showed that speech-in-noise words were identified better when spoken by the same talker from the lip-reading task. In addition, preliminary results showed a lexical frequency by word interaction, where low-frequency old words (presented at training) were better identified than low-frequency new words (not presented at training) during the speech-in-noise task. These preliminary results suggest that cross-modal talker facilitation can be based on the retention of amodal lexical episodes.
Speech intelligibility in speech noise: Single versus multiple-talker training.
Authors: Van Engen Kristin J.
Published: October 2010

Difficulty understanding speech in noise is a primary complaint for listeners with hearing loss. However, a person’s language experience can also affect the extent to which s/he is affected by noise when listening to speech. Non-native listeners, for example, have greater difficulty coping with noise than native speakers, and several studies have shown that native listeners are more detrimentally affected by interfering speech/babble in their native language versus a foreign language. Since experience-related factors can modulate the effects of noise on speech perception, the present study aims to identify training parameters that can enhance speech intelligibility in noise. Listeners were trained on English target sentences produced by either a single talker or by multiple talkers in the presence of English 2-talker babble, Mandarin 2-talker babble, or speech-shaped noise. These training conditions were compared by administering a common speech-in-speech post-test. It was predicted that target talker variability would lead to more generalizable learning. However, preliminary results indicate that single-talker training may be more beneficial than multiple-talker training, at least in some noise conditions. When the task of target speech perception is made easier, listeners may have more opportunity to learn to tune out competing speech noise.
Performance-intensity functions for dichotic speech recognition.
Authors: Roup Christina M. , Lamoreau Katie
Published: October 2010

Psychometric function characteristics of threshold and slope are well defined for monaural speech recognition performance for both normal hearing and hearing impaired listeners. Psychometric functions for dichotic speech recognition, however, have not been characterized in the literature. Dichotic listening refers to the simultaneous presentation of two differing stimuli to the ears. Percentage scores are typically obtained for both right and left ears, as well as an ear advantage score. The purpose of the present study was to characterize the psychometric function attributes of threshold and slope for dichotic speech recognition among young adults with normal hearing. Performance-intensity (PI) functions were obtained for three-pair dichotic digits and monosyllabic words for 26 normal hearing young adults 18–30 years of age. Results revealed significantly lower mean thresholds and significantly steeper mean slopes for dichotic digit versus dichotic word stimuli, reflecting the differences in difficulty between the two types of stimuli. Relative to monaural PI functions, dichotic PI functions exhibited higher thresholds and more shallow slopes. The differences in threshold and slope between dichotic and monaural PI functions likely reflect the greater difficulty associated with processing simultaneous yet different speech stimuli.
The effect of disrupting intonation patterns on the perceptual segregation of competing voices.
Authors: Iyer Nandini , Brungart Douglas S. , Simpson Brian D.
Published: October 2010

Relative differences in fundamental frequency and prosodic features have been extensively studied as cues to voice segregation in multitalker listening tasks. However, little is known about the effects of disrupting the intonation characteristics of a talker within a single phrase. The current experiment measured color-number keyword identification scores with the coordinate response measure (CRM) sentences under two conditions: a “normal prosody” condition, where the target and masker phrases were spoken contiguously by a single talker; and a “disrupted prosody” condition, where the target and masking phrases were generated by concatenating together individual words that were originally spoken in different sentences. The results show that the elimination of normal prosodic cues had the greatest impact on performance when the target sentence was masked by a single CRM masking phrase that was presented at the same level as the target talker. The elimination of prosodic information had only a very slight impact on performance when the target talker was masked by more than one simultaneous talker, suggesting that the use of prosodic cues for segregation may be impaired when more than one interfering talker is present in the mixture.
Experimental evidence for phoneme-level inhibition in spoken word recognition.
Authors: White James , Sundara Megha
Published: October 2010

The TRACE model of spoken word recognition [McClelland and Elman (1986)] contains phoneme-level inhibition whereas the MERGE model [Norris et al. (2000)] does not. Little has been reported in the experimental literature to bear on the question of phoneme-level inhibition, so the current study aims to fill this gap. An auditory lexical decision task was conducted, where participants were tested on English words preceded by isolated English sounds as primes. The relatedness of the prime sound to the final sound of the target word was manipulated. Phoneme-level inhibition is demonstrated when participants respond more slowly in the Similar condition (e.g., /sh/ before “bus”) than in the Identical (e.g., /s/ before “bus”) and Unrelated (e.g., /m/ before “bus”) conditions. This task revealed that phoneme-level inhibition was active at ISIs of 250 and 500 ms, but not 50 ms. A second lexical decision task indicated that inhibition begins after 125 ms and ends by 1000 ms. These results support models of word recognition that contain phoneme-level inhibition. Finally, a single-word shadowing task, in which participants repeat words primed by isolated English sounds as quickly as possible, is expected to replicate these findings and to extend them by adding speech error data.
Speaker recognition infrastructure for legal context in Mexico City.
Authors: Menchaca Garcia Felipe Rolando
Published: October 2010

Speaker recognition is actually a mature technology; it is part of the alternatives considered highly reliable for biometric identification, control access of computer services, and so on. However, applications to legal ambient, particularly as part of juridical processes, are still problematic, particularly in an ambient like Mexico City. High-quality infrastructure to determine with high precision whether a voice recording comes from a specific person is a topic which has to be regulated. Expert studies of this kind have to be regulated. This paper is to fulfill the need to develop a local regulation of infrastructure characteristics and standard test methods for this topic. Standard test has to take into account local language mainly.
Speech enhancement using adaptive spectral-spatial filter.
Authors: Mizumachi Mitsunori
Published: October 2010

Speech enhancement can be achieved in temporal, spectral, and spatial domains. For practical use, spectral subtraction is a popular filter in the spectral domain, and beamforming is a representative spatial filter. However, the former assumes stationary background noise, which should be known or estimated, and the latter requires arrival directions on sound sources. The author has proposed a robust approach to the simultaneous estimation of both a direction-of-arrival (DOA) of a target signal and an amplitude spectrum of background noise [Mizumachi, Proc. 158th ASA meeting]. The proposed method has also given confidence for each DOA estimate. In this paper, a spatial filter is adaptively designed and is integrated with a spectral filter in the adaptation process on spectral-spatial space. The confidence for each DOA estimate plays an important role for designing a reasonable and efficient spectral-spatial filter under time-varying acoustic scenes. In other words, simultaneous adaptation can be achieved time by time based on the reliability of each DOA estimate for the 2-D spectral-spatial filter. [Work supported by NEDO, Japan.]
Scattering from an arbitarily shaped rough interface embedded in heterogeneous fluids.
Authors: Tang Dajun , Jackson Darrell R.
Published: October 2010

The general formula based on first order perturbation theory for sound scattering from a rough interface separating two fluid media is derived. Possible areas of application of the formula are scattering from buried, uneven sediment rough interfaces and reverberation in a waveguide with changing water depth. Each of the two media is allowed to have space-dependent sound speed, density, and attenuation coefficient. The rough interface is assumed to be comprised of two components, a slowly varying, arbitrarily shaped surface considered as the unperturbed background, and a small roughness component superimposed on the slowly varying surface. The formula is intended for both the scattered field back to the medium where the incident wave originates and the penetration field. Properties of the formula are discussed, including reciprocity and symmetry with respect to the scattered and penetration fields. Numerical examples are presented for several scenarios which are of common interest to the underwater acoustics community.
Navigation on an unmanned surface vehicle using an acoustical spiral wave front beacon.
Authors: Dzikowicz Benjamin R. , Hefner Brian T. , Leasko Robert A.
Published: October 2010

An acoustical spiral wave front beacon described previously [Dzikowicz and Hefner, J. Acoust. Soc. Am. 127, 1748 (2010)] is tested using an unmanned surface vehicle (USV). The beacon transmits from two transducers, one with a signal whose phase varies with aspect and a reference signal whose phase is constant with aspect. A remote hydrophone’s aspect relative to the beacon’s can be determined by comparing the phase of the two signals. The beacon is positioned at a fixed depth, location, and orientation in the acoustic test pond at the Naval Surface Warfare Center, Panama City Division. A remotely operated USV is equipped with a hydrophone at the same depth as the beacon. The vehicle is also equipped with a differential global positioning system (DGPS) receiver to determine its exact position. The USV is driven in several routes including areas of varying water depth and reverberation levels. The DGPS positions are then compared to the aspect as calculated by the spiral wave font beacon. Single, dual, and swept frequency beacon signals are tested and compared. [Work supported by the Office of Naval Research.]
Modeling the response of an acoustical spiral wave front transducer.
Authors: Hefner Brian T. , Dzikowicz Benjamin R.
Published: October 2010

An acoustical spiral wave front transducer radiates a field whose phase varies linearly with the azimuthal angle around the transducer. The phase of the field, therefore, carries information about the location of a receiver relative to the spiral transducer. There are two transducer designs that can produce this type of wave front. The first is a “physical spiral” transducer that creates the phase change by physically deforming the active element of the transducer. While this is the simplest design, the physical deformation produces a discontinuity in the active element that affects both the amplitude and phase of the outgoing field. The effect of this discontinuity is examined through both a finite element model and a baffled source approximation. The second technique uses an array of elements, each driven with an appropriate phase to produce a “phased spiral” transducer. The output of this transducer depends on the wavenumber of the outgoing field as well as on the radius and number of the elements of the transducer. Simulations and approximations show that by tuning these parameters, the variations in either the amplitude or the phase of the outgoing field can be minimized. [Work supported by the Office of Naval Research.]
Eccentricity discrimination of hyperbolic localizations to minimize positioning uncertainty.
Authors: Vallarta J.
Published: October 2010

Conventional hyperbolic positioning locates a source at the intersection of hyperbolas that are based on time difference of arrival (TDOA) of a signal at each receiver pair. There are times when the intersections of these hyperbolas may delineate regions of uncertainty rather than a point, causing inaccurate locations due to the ambiguity of source positions. Localization accuracy can be improved by selecting hyperbolas based on their eccentricity (i.e., curvature). Since the eccentricity of a hyperbola is greater than 1, localization curves with eccentricity equal to or less than 1 are discarded as candidates of greater uncertainty. Curves with eccentricity tending to infinity are also discarded only when no other curves intersect with it. If the separations between receivers and the sound velocity are known, the hyperbolic eccentricity can be used as a function of the TDOA to minimize the ambiguity of source localizations. This method was applied to shallow water scenarios where 2-D localization approximation was sufficient (minimum three receivers required). Validation results of this method are shown based on bowhead (Balaena mysticetus) vocalizations recorded in the Chukchi Sea in autumn 2009.
Behavioral reactions of cod and sole to playback of pile driving sound.
Authors: Mueller-Blenkle Christina , Gill Andrew B. , McGregor Peter K. , Metcalfe Julian , Bendall Victoria , Wood Daniel , Andersson Mathias H. , Sigray Peter , Thomsen Frank
Published: October 2010

The effect of anthropogenic underwater sound on fish has become an important environmental issue. Pile-driving noise during construction is of particular concern as the very high sound pressure levels could potentially prevent fish from reaching breeding or spawning sites, finding food, and acoustically locating mates. This could result in long-term effects on reproduction and populationparameters. Additionally, avoidance reactions might result in displacement away from potential fishing grounds and lead to reduced catches. However, reaction thresholds and therefore the impacts of pile driving on the behavior of fish are completely unknown. Pile-driving noise was played back to cod and sole held in two large (40 m) net pens located in a quiet bay. Movements of the fish were analyzed using a novel acoustic tracking system. Received sound pressure level and particle motion were measured during the experiments. The results show significant movement responses to the pile-driving stimulus in both species at relatively low received sound pressure levels. This might indicate a rather large area of avoidance during real pile-driving operations. The results of the study have important implications on regulatory advice and the implementation of mitigation measures in the construction of offshore wind farms.
Three-dimensional acoustic propagation through crossing ocean waves.
Authors: Shmelev Alexey A. , Lynch James F. , Newhall Arthur E. , Lin Ying-Tsong
Published: October 2010

Internal waves, bottom sand waves, and ocean surface waves are known to have an impact on low-frequency 3-D acoustic propagation in shallow water. It has been shown that at certain horizontal grazing angles, acoustic normal modes can be ducted in between such waves and thus propagate along these waves with less spreading loss. In coastal areas, internal waves are often seen as packets with curved wave fronts. Different internal wave packets sometimes superimpose, creating a crossing structure. This work is focused on 3-D acoustic propagation through crossing waves that are the result of either internal wave packets crossing or combinations of internal waves, bottom sand waves, and surface waves crossing at various angles. An idealized shallow water column model is used for understanding the mode coupling physics and acoustic horizontal refraction. 3-D parabolic equation numerical modeling of acoustic propagation through a more realistic environment with crossing waves is then performed for quantitative analysis of the mode coupling effects.
Energy propagation in nonlinear internal wave ducts from radiative transport theory.
Authors: McMahon Kara G. , Lynch James F. , Lin Ying-Tsong , Siegmann William L.
Published: October 2010

Nonlinear internal waves have 3-D random variability arising from smaller scale physical oceanographic processes. Internal wave fronts form acoustic waveguides, and we examine propagation in directions nearly parallel to the fronts. Intensity fluctuations arising from the variability are treated by 3-D radiative transport methods, which provide insight into the parametric dependence on waveguide randomness. We focus on situations where the frozen field approximation applies, the acoustic modes represent the vertical energy distribution in the waveguide, and the modes propagate adiabatically. Under these conditions, horizontal modal variations are specified by 2-D radiative transport equations. With additional assumptions, the transport equations simplify to diffusion-type equations. An alternative energy scattering interpretation describes the duct in terms of a 2-D Galton’s box, assuming discrete scattering events occur as modes propagate down the duct. The resulting equations also reduce to diffusion-type equations in the continuous limit. We use appropriate spatial variability scales from the SW06 experiment to investigate properties of acoustic intensity. [Work supported by the ONR.]