bluuebunny/crossref_metadata_2025 huggingface dataset:
Dipole projectors for conformal sonar system applications
Authors: Gray Michael D. , Rogers Peter H. , Larson Gregg D.
Published: October 2013

Dipole projectors are being investigated for use in active sonar arrays on compliantly coated, low input impedance hulls. When the projector standoff d is acoustically small (k0d<1), the direct and hull-reflected signals add in phase in the far field, leading to an increase in signal strength on the order of 2 relative to a free field dipole. By contrast, the direct and reflected signals for monopole transducer will have nearly opposite phase, and the monopole will have a reduction in total signal strength on the order of 2k0d. Ideally, both transducer types have cosine directivities in-situ, although the monopole transducer response can be more strongly influenced by hull elastic contributions. Designs for two simple realizations of a dipole projector will be presented, along with beam pattern measurements made with prototype transducers at the Georgia Tech water tank facility.
Determination of noise descriptors and criteria for pyrotechnic noise sources
Authors: Wu Weixiong
Published: October 2013

A noise study was conducted to determine appropriate noise descriptors and criteria for assessing pyrotechnic noise sources. The study was carried out to support an environmental impact statement (EIS) that defined sensitive land uses adjacent to reservoirs in New York City area, where potential noise impacts from avian dispersion measures would occur. The pyrotechnic techniques defined as impulsive noise sources are among the avian dispersion measures that would be used at the reservoirs. Determining appropriate noise descriptors and criteria was critical to the EIS because of the distinctive sound characteristics of pyrotechnic impulse noise sources, the lack of published literature on assessing them, and the absence of corresponding noise regulations. Noise descriptors and criteria used for EIS in the United States were investigated, and noise measurements for pyrotechnic noise sources and some impulsive noise sources were also performed. The study results demonstrate that C-weighted DNL is an appropriate descriptor for assessing noise impacts from the pyrotechnics based on the U.S. Army Environmental Noise Management Program criteria, and peak hour A-weighted Leq(1) is a suitable noise descriptor for determining noise impacts for avian dispersion measures, including the pyrotechnics, based on the New York City Environmental Quality Review criteria.
An outdoor noise propagation study to predict the effect of a power plant expansion
Authors: Brasovan Philip J. , Carney Melinda J. , Cheenne Dominique J.
Published: October 2013

The results of an outdoor noise propagation model using CadnaA were compared to test data obtained on-site. The subject property is the central utility plant of a hospital located in Milwaukee, scheduled to be expanded with the addition of cooling towers. The modeled area was 400 m squared with a resolution grid of 2 m squared. The model was used to validate the observed test data as well as to predict the anticipated noise levels at completion of the expansion. A total of 11 points were investigated and the predicted data were found to match the test values within 2 dB at many locations. The data from the model show that the anticipated noise levels at the East property line will exceed those mandated by local ordinances by 3 dB. The model also predicts that the addition of a three meter absorbing barrier and the use of reduced noise fans for the six cell cooling system will bring the overall noise level from the system into compliance.
Multiparameter studies of surf infrasound
Authors: Garces Milton , Fee David , Caron Pierre , Hetzer Claus , Aucan Jerome , Merrifield Mark , Gibson Robert , Bhattacharyya Joydeep
Published: October 2013

Infrasound stations on islands or near coastlines routinely detect signals associated with breaking ocean waves. Although the source mechanisms of these infrasonic surf signals are not well understood, they might provide useful insight into coastal processes and permit an assessment of wave energy distribution in the littoral zone. Near-shore infrasonic arrays, ocean bottom sensors, an infrared imager, and a video camera were deployed on rocky and sandy coastlines in Hawaii during the 2004–2005 Winter swell season, with the aim of establishing a relationship between the directional ocean swell height and the infrasonic source distribution, intensity, and spectral content. The cameras targeted acoustically active regions for selected time periods to associate the timing of the infrasonic signal arrivals with the breaking and dissipation of ocean wave sets. A wide range of sea states and weather conditions were captured during two separate deployments. The results of our multiparameter analyses and preliminary comparisons of our observations with mesoscale ocean wave model predictions will be presented. [Work supported by the Defense Advanced Research Projects Agency. Support of the State of Hawaii is acknowledged.]
Borrowing a trick from So/ren Buus: Application of a simple quantitative model to data obtained from an impaired population
Authors: Carlyon Robert P.
Published: October 2013

One of the many things that So/ren taught me was to apply a simple quantitative model to psychophysical data, even when studying impaired populations, renowned for inter-listener variability. Here I present a recent application of this approach, applied to cochlear implant patients’ detection thresholds for electrical stimulation. Most implants stimulate the nerve with trains of biphasic pulses, in which a rectangular pulse of one polarity is followed, with a short inter-phase gap (IPG), by an equal pulse of opposite polarity. Together with van Wieringen and colleagues in Leuven, Belgium, we have shown that psychophysical thresholds in humans drop as IPG increases to at least 2900 microseconds implying a much longer time constant than predicted from physiological studies on animals. By applying a simple model, we showed that this long time constant is quantitatively consistent with the threshold drop observed for electrical sinusoids as frequency is lowered to about 100 Hz. The model involves passing the electrical waveform through a low-pass filter and calculating the RMS output. Despite its simplicity, the model successfully predicts the results of several novel manipulations, and helps constrain explanations of the physiological basis of the observed time constants.
Direction selective structural-acoustic coupled radiator
Authors: Seo Hee-Seon , Kim Yang-Hann
Published: October 2013

This paper presents a method of designing a structural-acoustic coupled radiator that can emit sound in the desired direction. The structural-acoustic coupled system is consisted of acoustic spaces and wall. The wall composes two plates and an opening, and the wall separates one space that is highly reverberant and the other that is unbounded without any reflection. An equation is developed that predicts energy distribution and energy flow in the two spaces separated by the wall, and its computational examples are presented including near field acoustic characteristics. To design the directional coupled radiator, Pareto optimization method is adapted. An objective is selected to maximize radiation power on a main axis and minimize a side lobe level and a subjective is selected direction of the main axis and dimensions of the walls geometry. Pressure and intensity distribution of the designed radiator is also presented.
Development of a directivity controlled piezoelectric transducer for sound reproduction
Authors: Bédard Magella , Berry Alain
Published: October 2013

One of the inherent limitations of loudspeaker systems in audio reproduction is their inability to reproduce the possibly complex acoustic directivity patterns of real sound sources. For music reproduction for example, it may be desirable to separate diffuse field and direct sound components and project them with different directivity patterns. Because of their properties, poly (vinylidene fluoride) (PVDF) films offer lot of advantages for the development of electroacoustic transducers. A system of piezoelectric transducers made with PVDF that show a controllable directivity was developed. A cylindrical omnidirectional piezoelectric transducer is used to produce an ambient field, and a piezoelectric transducers system, consisting of a series of curved sources placed around a cylinder frame, is used to produce a sound field with a given directivity. To develop the system, a numerical model was generated with ANSYS Multiphysics TM8.1 and used to calculate the mechanical response of the piezoelectric transducer. The acoustic radiation of the driver was then computed using the Kirchoff-Helmoltz theorem. Numerical and experimental results of the mechanical and acoustical response of the system will be shown.
Behavior of the intermediate and exterior layers of a close fitting enclosure surrounding a cylindrical fluid-loaded acoustic source
Authors: Cuschieri Joseph
Published: October 2013

Previously, results for the Insertion Loss (IL) of a close fitting enclosure surrounding a submerged (in water) cylindrical acoustic source [Cuschieri, J. Acoust. Soc. Am. 115, 2537 (2004); 116, 2521 (2004)] were presented for different intermediate layer materials, different characteristics of the exterior cladding layer material and different source sizes. The IL results were presented as a function of frequency and layer/cladding characteristics. In this presentation, details on the behavior of the intermediate and outside layers enclosing the cylindrical source are presented which demonstrate the coupling between the cylindrical source the external acoustic (water) medium. These results help to explain the IL results previously obtained and presented. Furthermore, the scattering characteristics of the cylindrical source with and without the close fitting enclosure are considered. [Work supported by ONR.]
Examination of voicing onset time during Mandarin tone productions
Authors: Chen Yang , Ng Manwa
Published: October 2013

The voice onset time (VOT) of stop consonant is defined as the time interval between the release of a stop and the onset of the following vowel (Lisker and Abramson, 1964, 1970). Previous studies indicated that VOT could be used as a perceptual cue for identification of both voicing and place of articulation of stops (Kewley-Port, 1983; Ladefoged, 2001). Abramson (1977) suggested VOT could be interpreted as interrelated acoustic consequences of variation in the relative timing of glottal and oral gestures. This study will attempt to extend the probe on this suggestion by comparing the VOT values among different tones. Mandarin, which is a tone language, where a change in the tone of a syllable leads to a change in meaning, will be used in the study. There are four contrastive tones in Mandarin, each of which is realized by changing the vocal pitch during the course of the syllable production. The proposed study will determine if VOT values for each of the six stop sounds (/p,b,t,d,k,g/) will vary with the variation of different tones at which each of the following vowel sounds is produced. The interrelationship between VOT and tone production will be discussed.
Breathiness in Indic languages
Authors: Esposito Christina , Khan Sameeruddowla , Hurst Alex
Published: October 2013

Previous work on breathiness in Indic languages has focused on the acoustic properties of breathy oral stops in languages like Hindi ([bal] hair versus [bhal] forehead) or Bengali ([baSa] house versus [bhaSa] language). However, breathiness in Indic languages often extends to nasals (e.g., Marathi ([maar] beat versus [mhaar] a caste). It is unclear if languages such as Hindi and Bengali have breathy nasals in addition to breathy oral stops. This study addresses the following questions: (1) Are breathy nasals (Nh) acoustically different from N+h sequences, both in languages where they are phonemic and ones where they are not? (2) In sequences of a breathy stop and a modal nasal (e.g., Hindi [udhmi] naughty) where is the breathiness realized, if at all? To answer these questions, audio, aerodynamic, and electroglottographic recordings will be made of Hindi, Bengali, and Marathi speakers. It is hypothesized that acoustically breathy nasals in Hindi and Bengali will not be distinct from sequences of N+ h. We believe that this will also be true for the oral stops. In addition, it is believed that in sequences of breathy oral stop followed by a modal nasal (e.g., ChN), the breathiness will be produced on the nasal.
Performance of mode-based processing in the presence of environmental uncertainty
Authors: Zurk Lisa
Published: October 2013

A critical limiting factor to the performance of many underwater sonar detection and localization techniques is the typically high degree of uncertainty regarding the ocean environment. Several researchers have proposed mode-based processing methods for Matched Field Processing, depth-dependent pre-filtering, and sub-array processing. Provided that an appropriate modal decomposition can be applied, these methods are believed to provide processing that is less sensitive to environmental mismatch, with the added potential advantage of computational efficiency. In this paper, we attempt to quantify the performance of mode-based processing in uncertain environments, and compare this performance to more classic processing approaches. Results are presented from Monte Carlo simulations of range-dependent environments in which the uncertainty is varied parametrically. An analytical framework for describing the uncertainty is presented and discussed.
Broadband source localization using matched correlation processing
Authors: Ebbeson Gordon R. , Matthews Marie-Noël R. , Heard Garry J. , Desharnais Francine , Thomson David J.
Published: October 2013

For many years, model-based signal processing algorithms using Matched Field Processing (MFP) techniques have been analyzed with the goal of improving the capability of passive sonar systems for localizing quiet underwater sources. Recently, researchers at DRDC Atlantic have been investigating Matched Correlation Processing (MCP) as a faster alternative to MFP. In this method, the cross-correlations for a source as measured with a pair of hydrophones in a horizontal array are matched with those generated with a correlation model for many candidate ranges and depths along a candidate bearing. These matches are carried out with a number of hydrophone pairs to form many ambiguity surfaces. The maximum on the average of these surfaces is assumed to yield the best estimate of the source position. By carrying out this procedure over a number of candidate bearings, a full 3-D search for the source location is achieved. Since 2002, a number of localization trials have been carried out east of Nova Scotia, Canada. During those trials, an array was deployed on the sea floor and used to collect acoustic signals from various broadband sources. In this paper, we describe the broadband MCP localization technique and show some localization results from those trials.
Broadband continuous wave source localization via pair-wise, cochleagram processing
Authors: Nosal Eva-Marie , Frazer L. Neil
Published: October 2013

A pair-wise processor has been developed for the passive localization of broadband continuous-wave underwater sources. The algorithm uses sparse hydrophone arrays and does not require previous knowledge of the source signature. It is applicable in multiple source situations. A spectrogram/cochleagram version of the algorithm has been developed in order to utilize higher frequencies at longer ranges where signal incoherence, and limited computational resources, preclude the use of full waveforms. Simulations demonstrating the robustness of the algorithm with respect to noise and environmental mismatch will be presented, together with initial results from the analysis of humpback whale song recorded at the Pacific Missile Range Facility off Kauai. [Work supported by MHPCC and ONR.]
Long term stability and individual distinctiveness in captive orca vocalizations
Authors: Noonan Michael , Suchak Malini
Published: October 2013

With focus on the question of signature calling in killer whales, recordings from five captive orcas (of Icelandic origin) held at Marineland of Canada were compared. For the present analysis, samples of three different call syllables were selected from recordings made five years apart and from instances in which the identity of the calling whale was unambiguous due to temporary isolation, concomitant bubbling, and/or head nodding. The Raven software package was used to ascertain the frequency range, frequency (max), duration, and timing of maximum and minimum power within each sample. For two of the three call syllables, statistically significant differences were found among the five whales for call length and for the timing of maximums and minimums (p<0.01−0.001). This similarly proved true for nearly all pairwise comparisons between whales, including mother-offspring dyads. By contrast, for three of four whales for which we had sufficient samples, no significant differences were found on any measure between samples taken from the same whales five years apart. These findings therefore support the notion that the voices of individual orcas are distinct from one another in ways that are stable over the course of multiple years.
The acoustic repertoire of bottlenose dolphins (Tursiops truncatus) from the southern Gulf of Mexico
Authors: Bazua-Duran Carmen
Published: October 2013

Bottlenose dolphins live in a variety of habitats of the world’s oceans using their acoustic repertoire to communicate and inspect their environment. This work investigates the acoustic repertoire of bottlenose dolphins that inhabit a coastal lagoon of the southern Gulf of Mexico, the Laguna de Terminos and how it may relate to the dolphins’ general behavioral state and herd size, and to the general characteristics of the habitat, such as visibility, depth, and sea state. Preliminary results show that bottlenose dolphins produce by far more clicks than whistles in all behavioral states (feeding, resting, social, and travel) and herd sizes, which may correlate with the decreased visibility and shallow depth of the Laguna de Terminos. Additionally, silence was found during all behavioral states, but very seldom in herds of large size. These preliminary results suggest that bottlenose dolphins are choosing when and where to produce their phonations. Therefore, more detailed studies are needed to understand how these animals are using their acoustic sense to communicate and inspect their environment. [Work supported by CONACyT-Gobierno Edo. de Campeche and PAPIIT, UNAM.]
The glissando illusion: A spatial illusory contour in hearing
Authors: Deutsch Diana , Hamaoui Kamil , Henthorn Trevor
Published: October 2013

In the glissando illusion (originally demonstrated by Deutsch, 1995) a synthesized oboe tone of constant pitch is played together with a sine wave whose pitch repeatedly glides up and down. These are presented through two loudspeakers, placed to the listeners left and right, and are repeatedly switched between loudspeakers, such that when the oboe tone emanates from one, a segment of the sine wave emanates from the other. The oboe tone is perceived correctly as switching between loudspeakers; however the segments of the sine wave appear to be joined together seamlessly, such that a single, continuous tone is heard, which appears to emanate from a source that moves slowly around in space in accordance with its pitch motion. In the present experiment, righthanders (n=22) tended strongly to hear the glissando move between left and right, and also between low and high in space, as its pitch moved between low and high. It was therefore frequently heard as tracing an elliptical path aligned diagonally between a position low and to the left when its pitch was lowest, and high and to the right when its pitch was highest. Nonrighthanders (n=42) perceived the illusion in statistically different ways.
Fractal dimension analysis of complexity in Ligeti piano pieces
Authors: Bader Rolf
Published: October 2013

Fractal correlation dimensional analysis has been performed with whole solo piano pieces by Gyrgy Ligeti at every 50ms interval of the pieces. The resulting curves of development of complexity represented by the fractal dimension showed up a very reasonable correlation with the perceptional density of events during these pieces. The seventh piece of Ligeti’s ‘‘Musica ricercata’’ was used as a test case. Here, each new part of the piece was followed by an increase of the fractal dimension because of the increase of information at the part changes. The second piece ‘‘Galamb borong,’’ number seven of the piano Etudes was used, because Ligeti wrote these Etudes after studying fractal geometry. Although the piece is not fractal in the strict mathematical sense, the overall structure of the psychoacoustic event-density as well as the detailed event development is represented by the fractal dimension plot.
Attack transients on lip reed instruments
Authors: Campbell Murray , Bromage Seona , Chick John
Published: October 2013

The generation of sound in a lip reed (brass) instrument relies on a non-linear coupling between two vibrating systems: the lips of the player, and the air column contained by the walls of the instrument. The nature of this coupling has been studied by filming the motion of the lips of a number of players using a specially constructed transparent mouthpiece, while simultaneously recording the pressure in the mouthpiece and at the bell of the instrument. Attention has been focused on the attack transient, and the way in which its duration and evolution depend on the extent to which the air column modes are harmonically related. For this reason the instrument chosen was the serpent; on this instrument certain fingerings give air column modes whose relationship is close to harmonic, while other fingerings give strikingly inharmonic air column modes. The relationship between measurements of transient character and judgements of the playability or responsiveness of the instrument will be discussed.
Normal-hearing listener preferences of music as a function of signal-to-noise-ratio
Authors: Barrett Jillian G.
Published: October 2013

Optimal signal-to-noise ratios (SNR) for speech discrimination are well-known, well-documented phenomena. Discrimination preferences and functions have been studied for both normal-hearing and hard-of-hearing populations, and information from these studies has provided clearer indices on additional factors affecting speech discrimination ability and SNR preferences. This knowledge lends itself to improvements in hearing aids and amplification devices, telephones, television and radio transmissions, and a wide arena of recorded media such as movies and music. This investigation was designed to identify the preferred signal-to-background ratio (SBR) of normal-hearing listeners in a musical setting. The signal was the singer’s voice, and music was considered the background. Subjects listened to an unfamiliar ballad with a female singer, and rated seven different SBR treatments. When listening to melodic motifs with linguistic content, results indicated subjects preferred SBRs similar to those in conventional speech discrimination applications. However, unlike traditional speech discrimination studies, subjects did not prefer increased levels of SBR. Additionally, subjects had a much larger acceptable range of SBR in melodic motifs where the singer’s voice was not intended to communicate via linguistic means, but by the pseudo-paralinguistic means of vocal timbre and harmonic arrangements. Results indicate further studies investigating perception of singing are warranted.
Analysis of scattering by spheres having a negative acoustical refractive index
Authors: Marston Philip L.
Published: October 2013

Electromagnetic waves having oppositely directed phase and group velocities propagate in metamaterials having a negative permeability and negative permittivity [J. B. Pendry and D. R. Smith, Phys. Today 57(6), 37–44 (2004)]. Such materials are predicted to have unusual electromagnetic scattering properties [R. Ruppin, Solid State Commun. 116, 411–415 (2000)]. If it is possible to fabricate acoustical materials having a simultaneously negative effective elastic modulus and density (in a dynamical sense), the mechanical energy flux will have the opposite direction as the wave-vector associated with phase evolution. Rays descriptive of the energy flux refracted by such hypothetical materials at interfaces with ordinary fluids would be characterized by a negative acoustical refractive index. Partial-wave-series calculations of high frequency scattering by fluid spheres having an acoustical refractive index at (or close to) 1 reveal backscattering enhancements associated with glory rays which, unlike ordinary spheres [P. L. Marston and D. S. Langley, J. Acoust. Soc. Am. 73, 1464–1475 (1983)], require only a single internal chord. Generalized Lamb waves on elastic shells having opposite phase and group velocities also cause enhanced backscattering associated with unusual rays [G. Kaduchak, D. H. Hughes, and P. L. Marston, J. Acoust. Soc. Am. 96, 3704–3714 (1994)].
Sound scattering by clusters of rigid cylinders in air
Authors: Figueres Miguel , Cervera Francisco , Håkansson Andreas , Sánchez-Dehesa José , Llinares Jaime
Published: October 2013

The scattering of sound by a cluster of rigid cylinders in air has been investigated both theoretically and experimentally. The cluster was made by a lattice of cylinders with hexagonal symmetry. The sound scattered in all directions around the cluster is measured and compared with a calculation based on multiple scattering theory. The behavior as a function of the external geometry of the cluster will also be presented. The refraction and reflection of the sound by the boundaries of the cluster is analyzed by a theoretical model that takes into account its inner crystal symmetry. [Work supported by MEC of Spain.]
High resolution auditory perception system
Authors: Alam Iftekhar , Ghatol Ashok
Published: October 2013

Blindness is a sensory disability which is difficult to treat but can to some extent be helped by artificial aids. The paper describes the design aspects of a high resolution auditory perception system, which is designed on the principle of air sonar with binaural perception. This system is a vision substitution aid for enabling blind persons. The blind person wears ultrasonic eyeglasses which has ultrasonic sensor array embedded on it. The system has been designed to operate in multiresolution modes. The ultrasonic sound from the transmitter array is reflected back by the objects, falling in the beam of the array and is received. The received signal is converted to a sound signal, which is presented stereophonically for auditory perception. A detailed study has been done as the background work required for the system implementation; the appropriate range analysis procedure, analysis of space-time signals, the acoustic sensors study, amplification methods and study of the removal of noise using filters. Finally the system implementation including both the hardware and the software part of it has been described. Experimental results on actual blind subjects and inferences obtained during the study have also been included.
Adjustment of interaural time difference in head related transfer functions based on listeners’ anthropometry and its effect on sound localization
Authors: Suzuki Yôiti , Watanabe Kanji , Iwaya Yukio , Gyoba Jiro , Takane Shouichi
Published: October 2013

Because the transfer functions governing subjective sound localization (HRTFs) show strong individuality, sound localization systems based on synthesis of HRTFs require suitable HRTFs for individual listeners. However, it is impractical to obtain HRTFs for all listeners based on measurements. Improving sound localization by adjusting non-individualized HRTFs to a specific listener based on that listener’s anthropometry might be a practical method. This study first developed a new method to estimate interaural time differences (ITDs) using HRTFs. Then correlations between ITDs and anthropometric parameters were analyzed using the canonical correlation method. Results indicated that parameters relating to head size, and shoulder and ear positions are significant. Consequently, it was attempted to express ITDs based on listener’s anthropometric data. In this process, the change of ITDs as a function of azimuth angle was parameterized as a sum of sine functions. Then the parameters were analyzed using multiple regression analysis, in which the anthropometric parameters were used as explanatory variables. The predicted or individualized ITDs were installed in the nonindividualized HRTFs to evaluate sound localization performance. Results showed that individualization of ITDs improved horizontal sound localization.
The Fresnel-Product Approximation, and comments on extending rational function approximation to include flow
Authors: Martinez Rudolph , Ting Carina
Published: October 2013

The first half of this presentation reviews the existing development of the Fresnel-Product Approximation as a replacement of the exact fluid impedance on a two-sided blade-like object. Excellent performance versus exact calculations is demonstrated for vibration patterns above, below, and at coincidence, viz., for flexural wavelengths that are longer, shorter, or equal to that of characteristic sound. The approximate solution is global and thus of the WKB type: (1) It captures the correct edge behavior of diffraction loads on the vibrating blade; (2) Collapses to the Kirchhoff solution away from both edges at high frequencies; and (3) Differs by only a small factor when applied at low frequencies to predict the blades globally entrained fluid mass. The second half of the talk addresses the Rational Function Approximation (RFA) pioneered by Feit and DiPerna. It shows how their spectral technique could be readily generalized to add flow effects, including the influence of shed vorticity as it co-exists with sound. Flowless acoustics will be derived from fully unsteady aerodynamics/aeroacoustics through a delicate limiting process and an integration by parts. The talk ends with a discussion of how current applications of the RFA could be stripped of some of their structural-acoustic artifacts.
Harmony in Gitksan
Authors: Yamane-Tanaka Noriko , Fujimori Atsushi
Published: October 2013

What types of consonants are transparent in vowel harmony in Gitksan has been little understood. Previous studies suggest that it is translaryngeal harmony, where the qualities of unstressed suffixal vowels are harmonized with the ones of last stem vowels, apparently skipping the stem-final laryngeal stop and laryngeal fricative [Rigsby, unpublished manuscript, University of Queensland, Australia, 1986; Rigsby and Ingram, International Journal of American Linguistics 56, 251–263 (1990)]. However, our original data show that harmony takes place when the stem-final consonant is uvular as well. It suggests that it cannot simply be analyzed as laryngeal transparency, but it could be reanalyzed as [+low] transparency. The present study examines cross-consonantal vowel harmony through acoustic experiment. It is predicted that there is a significant difference in formant values between the vowels across uvular or laryngeal consonants versus the ones across the other consonants. New data from an Eastern Gitksan speaker (female, 72) and its analysis will be presented. Implications for locality of vowel harmony as well as for the phonological treatment of consonantal transparency will be discussed. [Work supported by SSHRC.]
Extensible bioacoustical analysis software: Two examples
Authors: Mills Harold G. , Figueroa Harold K.
Published: October 2013

Two extensible software tools for bioacoustical analysis have been developed by the Bioacoustics Research Program at the Cornell Laboratory of Ornithology. Each tool provides a framework for sound visualization and analysis, supporting both manual selection and automatic detection of acoustic events in arbitrarily long recordings, manual annotation and automatic measurement of properties of these events, and management of the created acoustic metadata. The acoustic events of interest might include animal vocalizations and/or anthropogenic sounds. Simple examples of measurable event properties include duration, total energy, and spectrum. More advanced measurements might classify sounds according to the species or individuals that produced them, or estimate the location of a vocalizing animal. Both software tools can be extended in various ways, including the addition of new detectors and measurements. It has been found that this extensibility adds great value to the tools by allowing them to be readily adapted for new applications by bioacousticians or others with programming ability. It has also been found that such tools greatly facilitate the development of new detection and measurement algorithms by allowing prototype algorithms to be rapidly evaluated. Several case studies are presented that illustrate these principles. [Work supported in part by NSF.]
Measurement of biosonar signals of echolocating bat during flight by a telemetry system
Authors: Riquimaroux Hiroshi
Published: October 2013

A telemetry microphone system (Telemike) has been developed, which can be mounted on the bat’s head, to measure acoustic characteristics of emitted pulses and returning echoes. The system allows us to monitor what the bat listens to during its flight. A high-speed video camera system has been also adopted together with the Telemike to trace positions in space of the flying bat for analyzing temporal pulse emission patterns. With those devices, how the CF-FM bats execute parallel time-sharing real-time processing during their flight can be investigated. Doppler-shift compensation, echo amplitude compensation and processing for multiple target detections were observed. Some of the evidence found by the Telemike will be introduced and discussed. [Work partly supported by the Innovative Cluster Creation Project promoted by MEXT and by a grant to RCAST at Doshisha University from MEXT.]
Comparison of intensity discrimination, increment detection, and comodulation masking release in the envelope and audio-frequency domains
Authors: Nelson Paul C. , Ewert Stephan D. , Carney Laurel H. , Dau Torsten
Published: October 2013

In the audio-frequency domain, the envelope apparently plays an important role in detection of intensity increments and in comodulation masking release (CMR). The current study addressed the question whether the second-order envelope (‘‘venelope’’) contributes similarly for comparable experiments in the envelope-frequency domain. One set of experiments examined the relationship between gated intensity discrimination and continuous-carrier increment detection. In contrast to the asymmetry observed in the audio-frequency domain (listeners are more sensitive to increments), AM-depth discrimination thresholds were found to be the same in conditions with a continuous (modulated) carrier and with traditional gated stimuli for AM frequencies ranging from 4–64 Hz. The second set of experiments compared the amount of CMR in a tone-in-noise detection task when slow, regular fluctuations were imposed on the masking waveform in both domains. A significant release from masking of a 32-Hz signal in the modulation frequency domain was obtained only when the venelope fluctuations were slower than 1–2 Hz. Both experiments suggest a relatively weak contribution of venelope cues in the AM domain when compared to those provided by envelope cues in the spectral domain. [Work supported by NIH-NIDCD R01001641 (PCN, LHC) and the Danish Research Council (SDE, TD).]
Factors affecting gap duration discrimination performance
Authors: Grose John H. , Hall Joseph W. , Buss Emily
Published: October 2013

This investigation tests the hypothesis that reduced acuity in temporal tasks is evident relatively early on in the aging process for tasks involving brief stimuli or intervals, across-frequency processing, and/or significant processing loads. In an earlier study of gap duration discrimination (GDD) using fixed 20-ms markers, it was found that young listeners performed significantly better than middle-aged listeners for both within-channel and across-channel configurations. However, this age effect did not emerge in a subsequent study that employed random-duration markers. The purpose of the present study was to clarify stimulus conditions under which early aging effects are evident in GDD tasks. Normal-hearing young (20–25 years) and middle-aged (40–55 years) listeners were tested using both fixed- (20-ms) and random- (40 ms±50%) duration markers in configurations that were either within-channel or across-channel. In addition, task complexity was manipulated by incorporating a rhythm discrimination feature into the GDD task. Results to date indicate that performance declines when marker duration varies randomly, both for within- and across-channel conditions, as well as when task complexity is increased. These findings will be discussed in the context of identifying early aging effects in temporal processing. [Work supported by NIDCD DC001507.]
Modeling comodulation masking release using an equalization cancellation mechanism
Authors: Piechowiak Tobias , Ewert Stephan D. , Dau Torsten
Published: October 2013

Across-filter comparisons of temporal envelopes are a general feature of auditory pattern analysis which plays an important role in extracting signals from noise backgrounds. One example where a comparison of the temporal envelope in different frequency regions can lead to a substantial facilitation of detecting a signal in noise is comodulation masking release (CMR). It has been suggested that in (monaural) across-channel processing, like in (binaural) across-ear processing, an equalization-cancellation (EC) based circuit might be an effective strategy of noise reduction. The first part of the study investigates the relation between CMR and envelope-based binaural masking level differences (BMLD), using narrowband noise maskers and classical across-channel configurations (like N0Spi, N0Sm). In the second part, a model is presented that explicitly simulates CMR whereby the EC mechanism is assumed to take place at the output of a modulation filterbank. In the case of several flanker bands, the equalization is realized effectively by averaging the flanker bands, and cancellation by subtracting the averaged representation from the signal band. A generalized version of the model is presented that considers both across-channel and within-channel contributions to signal detection in CMR. [Work supported by the Danish Research Council.]
A unified view of the temporal-window and the adaptation-loop model in conditions of forward- and simultaneous masking
Authors: Hau Ole , Ewert Stephan D. , Dau Torsten
Published: October 2013

Two different mechanisms have been discussed in the literature as a possible explanation for forward masking: persistence (or temporal integration) and adaptation. In this study, two well established models of temporal processing in the auditory system are compared in a unified modeling framework. The temporal-window model representing a temporal-integration mechanism and the adaptation-loop model as the representative for the adaptation mechanism. The unified modeling framework shared a compressive, non-linear auditory filter stage and a template-based (optimal detector) decision stage. Predictions from both models were obtained in conditions of forward masking at 1 and 4 kHz for masker-to-signal gaps of 0 to 160 ms, and in conditions of simultaneous masking, where the signal was temporally overlapping with the masker. The signal was a 10- or 12-ms raised-cosine ramped pure tone and the masker a 200-ms broadband noise. It was found that when both models are adjusted to behave similarly, the two mechanisms can be considered as being essentially equivalent. In fact, the combination of integration and the signal-to-noise-ratio based detection criterion in the temporal-window model can be interpreted as adaptation. Suggestions for a unified model of auditory processing are presented. [Work supported by the Danish Research Council.]
Response growth with and without a low-frequency suppressor
Authors: DiGiovanni Jeffrey J. , Wojtczak Magdalena
Published: October 2013

In a psychophysical study, Wojtczak and Viemeister (2005) demonstrated that the response to a tone suppressed by a fixed-level, higher-frequency suppressor grows faster than the response to the same but un- suppressed tone. This finding is consistent with a linearization of the response under suppression. This study extends their experiment to suppressors with frequencies below that of the suppressee. Detection of a 10-ms, 4-kHz probe was measured under two forward-masking conditions: one with a 200-ms, 4-kHz masker presented alone, and the other with the same masker/probe paired with a 2.4-kHz fixed-level suppressor. A range of probe levels was used to measure growth of masking with and without the suppressor. The 4-kHz masker level was varied adaptively to find the masked threshold. Suppression was revealed by the difference between the masker levels needed for the masked threshold in the presence and absence of the suppressor. Initial comparison of the rates of masking growth between the two conditions suggests a linearization of the response to the 4-kHz masker in the presence of a lower-frequency suppressor. A few levels of the suppressor were used to determine the rate of suppression growth and its effect on the growth of response at the suppressee-frequency place.
Repetition ‘‘pitch’’ in chinchillas
Authors: Shofner William P. , Whitmer William M.
Published: October 2013

Repetition pitches are evoked in human listeners by rippled noises. Infinitely iterated rippled noise (IIRN) is generated when wideband noise is delayed, attenuated, and added to the original wideband noise through positive (+) or negative (−) feedback. In human listeners, the pitch of IIRN(+) is matched to the reciprocal of the delay, whereas the pitch of IIRN(−) is an octave lower. A stimulus generalization paradigm was used to characterize IIRN repetition pitch in chinchillas. Chinchillas were trained to discriminate IIRN(+) with a 4-ms delay from IIRN(+) with a 2-ms delay. In the generalization task, chinchillas were tested with IIRN(+) having delays between 2–4 ms. The pitches evoked by these IIRNs ranged from 500–250 Hz. The delayed noise attenuation for all IIRNs was fixed at −1 dB. A systematic gradient in behavioral response occurred along the dimension of delay for each animal tested, suggesting that a perceptual dimension of pitch exists. Responses to IIRN(−) evoking pitches between 500–250 Hz (delays of 1–2 ms) were also measured. Responses to IIRN(−) were more variable among animals, suggesting that other perceptual cues such as timbre differences may be stronger than the pitch cues. [Work supported by NIDCD R01 DC005596.]
Modes of vibration and sound radiation from percussion instruments
Authors: Rossing Thomas D.
Published: October 2013

When a membrane or bar or plate is struck, it vibrates in a complex manner, which can be described in terms of normal modes of vibration. We describe the modes of vibration of percussion instruments, such as drums and bells, some ways in which the normal modes are observed, and the way in which they determine sound radiation from the instruments.
Development of an anthropomorphic talking robot and the mimicking speech control
Authors: Fukui Kotaro , Nishikawa Kazufumi , Kuwae Toshiharu , Takanishi Atsuo , Takanobu Hideaki , Mochida Takemi , Honda Masaaki
Published: October 2013

We developed an anthropomorphic talking robot WT-4 (Waseda Talker No. 4) to produce human speech. WT-4 consists of 1-DOF lungs, 4-DOF vocal cords and articulators (the 7-DOF tongue, 5-DOF lips, 1-DOF teeth, nasal cavity and 1-DOF soft palate); the total DOF is 19. The lips and the tongue are made of elastic material to allow large deformation and to prevent from the air and sound leaks, and are controlled by a looped wire mechanism to form their various configurations. The talking robot enables to produce vowel and consonant sounds by mimicking the vocal cords vibration and the fricative and plosive source generation by the air flow as well as dynamically controlled vocal tract acoustic resonance in human speech production. Articulatory control of the talking robot is designed to track the acoustic goals (pitch, sound power, two formant frequencies, and voice-unvoiced timing) of the speech. The robot parameters are determined by minimizing the weighted mean squared error of these acoustic parameters between the human and robot speech sounds. It is shown that this mimicking speech control is effective in producing fluent continuous speech by the talking robot.
A noise-reduction strategy for speech based on phase-opponency detectors
Authors: Deshmukh Om , Anzalone Michael C. , Espy-Wilson Carol Y. , Carney Laurel H.
Published: October 2013

A noise-reduction algorithm was developed based on a neural model for detection that is robust in fluctuating noise. The phase-opponency (PO) neural model correlates the outputs of two different auditory filters that differ in phase by 180 degrees at the target frequency. The PO detector used here consists of a pair of overlapping bandpass filters with phase responses that differ by 180 degrees near the center frequency (CF). The correlation between the filter responses is reduced when a narrowband signal near CF is present in a noisy background. A bank of PO detectors was used to process speech corrupted by additive Gaussian noise. The time-varying outputs of the detectors can be post-processed to retain information-rich regions, such as formants and frication onsets, while greatly reducing noise between formants. The final detector output controlled the gains in a separate analysis/synthesis filterbank. Spectrograms of speech sounds before and after noise reduction illustrate the ability of the system to detect major features in speech down to low signal-to-noise ratios. The quality of the processed signal will be demonstrated. This system is intended as a front-end for speech recognition systems or in hearing-aid noise-reduction algorithms. [Work supported by NSF BCS0236707 (OD,CYE) and NIH DC-001641 (MCA,LHC).]
Mechanical vocal cord model mimicking human biological structure
Authors: Shintaku Eiji , Fukui Kotaro , Nishikawa Kazufumi , Ikeo Shunsuke , Takada Kentaro , Takanishi Atsuo , Takanobu Hideaki , Honda Masaaki
Published: October 2013

We present a mechanical vocal cord model aiming for a talking robot, WT-5 (Waseda Talker No. 5). Unlike a musical reed which has been used in conventional mechanical speech synthesizer, the vocal cord model is formed to mimic the human’s vocal cord in the shape and the biological structure. It is made of a thermoplastic rubber, Septonh (Kuraray Co. Ltd.) of which the elasticity like a human’s, and has 3-DOF mechanisms which is similar to the human structure. 1-DOF link mechanism could change the pitch by stretching the length of the vocal cords. The 2-DOF arm mechanism is used to mimic the abduction and adduction of a human arytenoid cartilage. The vocal cord model was excited by air flow exhausted from a mechanical lung model. The vibration pattern was observed by a high-speed camera, and the glottal volume velocity and the sound pressure were recorded by a mask-type wire screen pneumotachograph and a microphone. It was shown that the lower and upper edges of the vocal cords could vibrate in a different phase and the sound spectrum was similar to the source spectrum of human speech.