bluuebunny/crossref_metadata_2025 huggingface dataset:
Detection of signals with expected and unexpected total durations and temporal structures
Authors: Wright Beverly A. , Dai Huanping
Published: October 2013

When trying to detect a tonal signal in a continuous broadband noise, listeners attend selectively to the total duration of the expected signal, but do they also attend selectively to its temporal structure? To investigate this question, a probe-signal method was used to measure the detectability of expected 1-kHz target signals that differed across conditions in total duration (4 or 299 ms) and temporal structure (pure tone or pulse train), and unexpected 1-kHz probe signals that differed from the target in total duration, temporal structure, or both. For all targets, probe detectability decreased as the difference between the target and probe duration increased. However, the pattern of results differed between the two target durations. For the 299-ms targets, probe detectability did not differ significantly between pure-tone and pulse-train probes, regardless of the target type or the probe duration. In contrast, for the 4-ms targets, detectability for each pulse-train probe was significantly higher than for pure-tone probes of the same duration. Thus, listeners appear to attend to the targets total duration more than its temporal structure when the target is long, but do the reverse when the target is short. These results are inconsistent with current temporal-integration models. [Work supported by NIH.]
Development of temporal-interval discrimination during adolescence
Authors: Huyck Julia Jones , Wright Beverly A.
Published: October 2013

The accurate perception of time intervals in the 10–100-ms range plays an important role in sensory encoding, including the processing of speech, yet little is known about the development of this ability, particularly during adolescence. Here, auditory temporal-interval discrimination thresholds were measured in normally developing individuals aged 11 (n=14), 14 (n=8), and 18–26 (n=17) years. Both average performance (mean within-listener threshold) and performance consistency (within-listener standard deviation) were evaluated for each listener with a 100-ms standard at each of two frequencies, 1 and 4 kHz. At the group level, for both frequencies, average performance improved significantly, from ∼48 to ∼22 ms, between 11 and 14 years of age (both p≤0.02), but did not improve further between 14 years and adulthood (both p≥0.92). There were parallel improvements in performance consistency. At the individual level, while the majority of 11-year-olds showed poorer performance than 14-year-olds and adults on both measures, a sizable minority performed within the range of adults. Thus it appears that, both in terms of average performance and performance consistency, the sensitivity to differences in temporal intervals has a developmental time course that can differ considerably across individuals, but, for most, reaches well into adolescence. [Work supported by NIH/NIDCD.]
Phonetic context effects in normal-hearing listeners using acoustic simulations of cochlear implant signal
Authors: Aravamudhan Radhika , Lotto Andrew J.
Published: October 2013

Recent work has indicated that the perception of speech sounds shifts as a function of preceding and following context and that these shifts are due in part to the particular spectral makeup of the context sounds [Holt et al., J. Acoust. Soc. Am. 108, 710–722 (2000)]. Aravamudhan and Lotto [J. Acoust. Soc. Am. 118, 1962–1963 (2005)] studied these context effects in listeners with cochlear implants (CI) and the results demonstrated absent or abnormal context-dependent speech perception. The lack of normal context effects in CI users may have practical implications for situations in which there is substantial coarticulation (e.g., nonlaboratory speech) or talker variability (e.g., switching between multiple speakers). Since the representation of speech in CI listeners differs from NH listeners, the current study was designed to investigate the effects of processing normal signals through a CI processor with NH listeners. In particular, we investigated how the spectral resolution of implant input could limit context-dependent perception even in NH listeners. The results showed that the context effects elicited in NH listeners using CI processed signals were similar to CI listeners. The effects of context were moderated by the processing strategy being simulated as well as some of the parameter settings.
The role of prosody in the foreign accent syndrome: A single-subject study
Authors: Garst Diane M. , Levitt June S. , Katz William F.
Published: October 2013

Foreign accent syndrome (FAS) is an unusual motor speech disorder involving a perceived foreign accent resulting from brain damage. An important question concerning FAS is the role of prosodic deficits, including the extent to which difficulties in stress assignment may contribute to segmental errors. In this study, acoustic analyses were performed on the speech of a 46-year-old female patient who presented with a pseudo-accent described as Swedish. Stop consonant VOT, consonant burst spectra and duration, and vowel formant frequencies and trajectories were analyzed, along with prosodic cues for lexical stress assignment and sentence-level intonation. Results indicated that the patients vowels were centralized and realized with reduced dynamic specifications. There was a strong tendency to realize the English alveolar flap as a full stop, and to produce flaps that had greater-than-normal closure durations. Lexical stress assignment was frequently inaccurate and highly variable, with similar problems noted for non-speech stimuli. Overall, the data suggest that stress assignment deficits were the basis for most of her segmental difficulties. The findings support the view that local prosodic deficits play a critical role in FAS.
The nonlinear relation between articulation and acoustics: Multiple constrictions
Authors: Iskarous Khalil
Published: October 2013

The vocal tract can be usefully approximated as a linear filter, but it has been known since the advent of this approximation that the relation between the area function and the acoustic output is nonlinear, despite linearity with respect to the input. This nonlinearity can be readily seen in nomograms displaying formants changing as a contriction changes its location or degree. It will be shown in this work that the nonlinear relation between area function and acoustics becomes even more significant when several constrictions are superimposed, which is the norm in natural speech production, where constrictions overlap in time. Simulations have been carried out, using a Kelly-Lochbaum digital lumped-loss transmission line, that show a complex relation between the formants resulting from a multiply constricted vocal tract to formants due to the constrictions when each is present by itself. The goal of the simulations is to predict time varying formant patterns when 2 or 3 constrictions are overlapped and all changing in time. A theory is then presented for how time-varying constrictions add in the acoustic domain. Implications for restrictions on compensation in speech production will be discussed. [Work supported by NIH NIDCD 02717.]
Acoustic enhancement in large indoor volumes
Authors: Barbar Steve
Published: October 2013

Large indoor spaces present many unique challenges for controlling sound. If both speech and music must be simultaneously delivered with high fidelity in a large indoor volume, the challenges for sound delivery can become daunting. Carefully designed acoustical treatments and electro-acoustic systems can yield good speech intelligibility. These treatments, however, can be detrimental to the delivery of music. Integrating time-variant electronic acoustical enhancement allows both acoustical treatments and electro-acoustic systems to be used more effectively. This presentation will discuss the electro-acoustic systems installed at the conference center for the Church of Jesus Christ of Latter Day Saints. Impulse response measurements indicate how portions of each system and architecture contribute to the listening experience at different locations in the venue.
Interesting acoustics in the Salt Lake Tabernacle and the Latter Day Saints conference center
Authors: Rollins Sarah
Published: October 2013

The Salt Lake Tabernacle and LDS Conference Center have interesting acoustical properties. Due to the elongated domed ceiling of the Tabernacle, there are audible focusing effects throughout the hall. In order to characterize the Tabernacle’s unique acoustics prior to its seismic renovation, the acoustical properties of the hall were investigated through impulse response measurements and computer modeling. This paper will show how the measurements and model were used to see the focusing effects in the Tabernacle. The acoustics of the LDS Conference Center have also been extensively measured. This paper will show various methods that were used to investigate the sound focusing effects that were audible in the Conference Center.
Computer-aided long distance troubleshooting
Authors: Miller Andrew N. , Coffeen Robert C.
Published: October 2013

A three-dimensional model of a religious worship facility was developed through the use of the EASE software package. Using the EASE Aura module, a binaural impulse response was created for three listening locations in the space. EASERA software was then used to analyze the impulse response data, and potential solutions to the problems observed were proposed. Each solution was then implemented in the model, and impulse responses were made for each scenario. Analysis after all proposed solutions were implemented in the model suggested that their real-world counterparts would prove successful. This paper raises the discussion of the effectiveness of long distance troubleshooting using modeling and analysis software.
An acoustic spread spectrum technique for the study of outdoor noise propagation
Authors: Waddington David C. , Angus Jamie A.
Published: October 2013

An investigation of the properties of a directly carrier-modulated code sequence modulation signal is presented for the study of outdoor propagation. The audible frequency carrier is biphase-modulated using a pseudorandom code sequence specially designed for simultaneous fine-time resolution and large-range ambiguity. An experiment is described in which the correlation properties of the spread spectrum signal are demonstrated, and the amplitude of the transfer functions between the receivers, together with accurate times-of-flight, are calculated from the cross-correlation of the measured acoustic spread spectrum signals. The results illustrate that an acoustic spread spectrum technique can provide significantly improved ways of measuring sound propagation outdoors. In particular, the fine-time resolution for time-of-flight suggests an application in acoustic tomography, for example in the investigation of nocturnal boundary layers, horizontal gradients, and of turbulence structures in the atmospheric surface layer. [Work supported by EPSRC UK.]
High resolution color flow—application of coded excitation in color flow imaging
Authors: Hao Xiaohui
Published: October 2013

Color flow imaging has suffered from the trade off between resolution and sensitivity ever since it was introduced. The requirement of blood flow sensitivity results in application of narrow bandwidth, long time duration pulses. The problem associated with this type of pulse is the severely degraded spatial resolution, which causes wall overwriting, blobby flow aesthetics, loss of detail flow hemodynamics, etc. Coded Excitation has been applied in B mode imaging to solve the similar tradeoff between SNR and resolution, but has not been so successful in color flow imaging. This is due to the fact that current color flow pulses are already at the regulatory limit and the focus of delivering more power for more sensitivity with coded excitation in color flow imaging. High resolution color flow imaging is an effort to introduce Chirp Coded-Excitation to achieve B-mode-like spatial resolution in color flow without sacrificing sensitivity. Wide bandwidth pulse with time bandwidth product greater than 1 is employed to bring in sensitivity/penetration and resolution. As a result, better spatial definitions, less wall overwriting, and detailed flow hemodynamics are achieved with improved flow sensitivity. Corresponding practical application issues are discussed in this paper.
Phonometrical precision measurements during a high end classical music live recording and four standardized recording systems’ signal comparison
Authors: Policardi Franco P. A.
Published: October 2013

Discographical sound recording process is very seldom accompanied by real sound pressure, intensity and level measurements. Usually available data are often referred to mathematical extrapolations calculated on sound recording audio signals. This article investigates the situation defined as ‘‘real’’ from a scientifical point of view, having as a referring base a Swedish precision sound level meter fresh from calibration, during the stereophonic multichannel live recording session of the Philharmonische Camerata Berlin. Comparison among reference signal and eight high quality microphone capsules signals coming from four different stereophonic standardized microphone systems devoted to discographical purpose recording (XY, ORTF, Binaurale and AB) shows big differences among the respective signals and the same recording spot measurement system position. It is noticeable how differences even higher than 6 dB can make microphone positioning particularly critical. A standardizing method is proposed.
Nonlinear time reversal acoustic method of land mine detection: Experiment and modeling
Authors: Sutin Alexander , Libbey Brad , Sarvazyan Armen
Published: October 2013

The nonlinear vibration response of land mines is an effective detection clue. This response can be measured by application of two frequency acoustic waves producing a vibration spectrum above the mine that is different from its surrounding. As an alternative to this approach, this work investigates time reversal techniques as a method of acoustically exciting nonlinear mine vibrations over a broad frequency range. The phase-inversion method used here requires two short broadband time-reversed focused signals with opposite sign, i.e. phase inverted. The focused signal and the inverted focused signal are broadcast sequentially and the responses are summed in post processing. This cancels the linear response leaving the nonlinear component. A theoretical model has been developed that treats the mine stiffness as a nonlinear spring. It predicts that the dominant nonlinear frequencies correspond to the second harmonics of the mine’s primary resonance. Experiments were conducted using six loudspeakers in a box placed directly over a mine. The measured responses confirm most of the model predictions, and illustrate differences between the soil and mine nonlinearities. Work was supported by the U.S. Army RDECOM CERDEC Night Vision and Electronic Sensors Directorate
Prediction of seismic pulses from an outdoor explosive source
Authors: Taherzadeh Shahram , Attenborough Keith
Published: October 2013

Near-surface layering of ground soil can influence propagation of seismic pulses originating from above-surface sources. Furthermore, in a cold climate snow and frozen ground can add to this layering effect. A numerical model called Fast Field Program for Layered Air Ground Systems (FFLAGS), developed originally for continuous sound sources above a porous elastic ground, is used to predict radial and vertical seismic signals from an above ground explosive charge recorded by a geophone. Simultaneous acoustic air pressure recording of the same charge is used to make a prediction of the source pulse shape. This procedure produces an effective linear source. In other words, nonlinear effects are assumed to be small at the ranges of interest here. Subsequently, suitable optimization methods were used to predict a set of best fit parameters for the near-surface ground structure. [Work supported in part by US Army through its European Research Office.]
Complex source imaging using time-reversal (TR): experimental studies of spatial and temporal resolution limits
Authors: Anderson Brian E. , Johnson Paul A. , Ulrich T. J. , Griffa Michele , Ten Cate James A. , Huang Lianjie
Published: October 2013

Large earthquakes are composed of a complex succession of slip events that are nearly indistinguishable on a seismogram. The question, how does an earthquake work? remains largely unsolved. The slip events on the fault plane(s) generally take place at different spatial locations and at different times. TR wave physics can be advantageously exploited to recreate, from measured signals, a spatially and/or temporally complex sound/seismic source. An experimental study is conducted to determine the spatial and temporal resolution limitations in imaging a complex source in solids, as part of our goal to understand earthquake source complexity. TR experiments are conducted on solid blocks of different materials, such as Berea sandstone and aluminum. Arrays of piezoelectric transducers are bonded to the samples for the creation of complex spatial-temporal sources, as well as to record signals. The experimental spatial and temporal resolution limits for complex source imaging will be presented as a function of material physical characteristics (e.g., Q, modulus), as well as source signal characteristics such as pulse width, frequency and repetition rate. [This work was supported by Institutional Support (LDRD) at Los Alamos National Laboratory.]
Environmental sound research as it stands today
Authors: Gygi Brian , Shafiro Valeriy
Published: October 2013

Environmental sound research is still in its beginning stages, although in recent years a body of research has started to accumulate, both on the perception of environmental sounds themselves and on their practical applications in other areas of auditory research and cognitive science. However, several important issues remain unresolved, such as methodologies for environmental sound research, how to coordinate research efforts in disparate areas of environmental sound study, and how best to integrate environmental sound research into the larger body of hearing research. These issues will be discussed along with some of the current and future practical applications of environmental sounds, such as in assistive listening devices and automatic sound source recognition.
Effects of training on the perception of environmental sounds with limited spectral resolution
Authors: Shafiro Valeriy
Published: October 2013

Environmental sounds play an important role in maintaining listeners’ awareness of the sound sources in the environment. They can warn of potential dangers (e.g., alarms, collisions) or provide aesthetic satisfaction (e.g., babbling brook, bird song). Although research indicates that decreased spectral resolution has a negative effect on environmental sound identification, little is known about perceptual adaptation to spectrally sparse environmental sounds. This project investigated changes in the ability to identify spectrally smeared environmental sounds in response to training. In a pretest-posttest design, seven naive normal-hearing listeners, identified the spectrally smeared environmental sounds obtained with a four-channel noise-based vocoder. Five training sessions were administered between the pretest and the posttest. Overall mean identification performance improved from 33% on the pretest to 63% on the posttest, with the largest improvement of 86% obtained for the sound tokens used during training. Increases in identification accuracy were also observed for alternative tokens of the training sounds (36% difference) and sound sources not included in the training set (31% difference). Among those, the greatest improvement was found among sounds with distinct envelope patterns. The implications of these findings for perceptual learning and rehabilitation programs for cochlear implant patients will be discussed.
Voiced geminates in Yamagata and Tokyo Japanese
Authors: Butler Lynnika
Published: October 2013

Voiced geminate stops (/bb/, /dd/, /gg/, etc.) are cross-linguistically rare, due to aerodynamic constraints on voicing during prolonged closure of the vocal tract. Standard Tokyo Japanese prohibits voiced geminate stops in native and Chinese-borrowed vocabulary, but permits them in some recent loanwords (primarily English borrowings), where they reflect stress on the preceding syllable of the word in the originating language. (1)big–/biggu/ (2) headphone–/heddohoN/ Yachi-ben (YB) is a local subdialect of the northeastern (Tohoku) dialect of Japanese, which is characterized in part by voicing of most non-initial stops. Transcriptions of YB speech recorded in the 1970s by Japans National Language Research Institute (NLRI) indicate a sizeable number of voiced geminate stops, most frequently /dd/ but also /gg/ (there are no /bb/ tokens). In this study, voiced geminate stops from the NLRI recordings were compared with voiced geminate stops in English loanwords as pronounced by Tokyo Japanese (TJ) speakers. Closure duration, as well as duration and amplitude of closure voicing, were measured for both sets of data. Overall, voicing ratios (the percent of closure duration in which voicing was present) were found to be much longer in YB voiced geminates than in TJ voiced geminates.
On the phonetics of Cypriot-Greek geminate plosives
Authors: Christodoulou Christiana
Published: October 2013

This research provides evidence toward the existence of geminates in Cypriot Greek (hereinafter, CyG). In the existing literature it is reported that the sole or primary cue to gemination, across languages, is closure duration. This phonetic study verifies that this cross-linguistic phonetic cue is also present in CyG. Toward this end, statistical analyses evidence highly significant durational differences in the voice onset time (VOT) of geminates versus singletons in CyG [F(164)=p<0.0001]. In fact it can safely be argued that VOT is the primary cue to gemination when it comes to CyG geminate plosives. This study contradicts previous claims [Arvaniti and Tserdanelis, 562 (2000); Arvaniti, 602 (1999b)] that argue closure duration to be the primary cue to gemination for CyG geminate stops. Furthermore, durational differences amongst the three voiceless plosives for both VOT and closure duration were highly evident. To the researchers’ knowledge this is the first study on CyG stops that considers durational differences with respect to the place of articulation, not only for singletons but also for geminates. Preliminary statistical analysis suggests an effect of the vowel following the target segment, a fact that could facilitate in resolving the phonological representation of utterance initial geminates.
Processing of fine phonetic detail in American English flaps
Authors: Tucker Benjamin V.
Published: October 2013

Listeners are very good at recognizing casual, reduced speech in their native language. In day-to-day conversation acoustic information is often impoverished or reduced. Reduction in speech can occur in the form of deletion and/or change of segments, syllables, or words [Johnson (2004)]. This study explores differences in processing between reduced/conversational speech and unreduced/careful speech. Two lexical decision studies, using reduced and unreduced pairs of the American English flap [Warner and Tucker (2007)], investigate whether processing differences exist between reduced and unreduced speech. The first study presented target words to listeners in isolation and the second study presented the same targets preceded by a reduced or unreduced frame sentence. The materials contained, among 260 items, 40 word medial /d/ phonemes and 40 word medial /g/ phonemes occurring in a flapping environment [Zue and Laferriere (1979)]; targets were produced in both reduced and unreduced form. These materials, spoken by a native American English speaker, were presented to 55 monolingual American English students from the University of Arizona. Listeners performed the lexical decision task for each item and response time and error rate was measured. Native English listeners process unreduced items more quickly than reduced items.
Nature versus nurture: Children’s development and perception on syllable-final nasals in Taiwan Mandarin
Authors: Hung Renee , Fon Janice
Published: October 2013

In Hung and Fon’s study (2006), it was found that elementary school children in Taiwan also tend to merge /■N/ with /■n/ as the way adults do, and the merging rates found in reading characters and sentences are higher than those in reading Zhuyinfuhao, a local phonetic system. However, while most adults and older elementary school children merge /in/ with /iN/, younger children have much lower merging rates for /i/ no matter under what condition. The first part of this study extends its investigation to preschoolers who have not officially learned Zhuyinfuhao. A picture-naming task was conducted with 20 kindergarten children. The results show that preschoolers have very similar performances with the adults’ for both vowels. The second part of this study is a perception experiment. Sixty Grade-2 and Grade-4 students participated in the task, in which they were asked to write down the characters and phonetic symbols of the monosyllabic words they heard. Preliminary results indicate that these children are apt to treat /in/ and /■N/ words as homophones of /iN/ and /■n/ words, respectively, although such occurrences for /■/ are lower than those for /i/.
Validation study of computer modeling of source directivity
Authors: Vigeant Michelle C. , Wang Lily M.
Published: October 2013

Previous research has focused on the changes in objective and subjective results when changing source directivity in a computer model, but little work has been done to validate these simulated changes experimentally. An investigation was carried out to compare the measured and simulated room acoustics parameters which result from using a directional sound source, a JBL Eon-10 G2 loudspeaker. Measurements were made in a 100-seat lecture-style classroom, using the sine sweep method, with (a) an omni-directional and (b) the directional loudspeaker oriented in four directions. The measured differences in reverberation time (T30) were minimal across the two source types and four orientations, while significant differences resulted for early decay time (EDT) and clarity index (C80). An ODEON v6.5 model of the classroom was calibrated against the omni-directional results to within two just-noticeable-differences (JNDs) across the three parameters of interest: T30, EDT, and C80. Simulations with the directivity of the JBL loudspeaker were then performed and the results differed by less than two JNDs from the measurements for all source-orientation/receiver combinations. In conclusion, ODEON v6.5 does accurately model the changes in room acoustic parameters which result from using different source directivities. [Work supported by the National Science Foundation.]
Increasing the direct-to-reverberant ratio in rooms with a constant-beamwidth, wide-bandwidth loudspeaker array
Authors: Winker Douglas F. , Hixson Elmer L.
Published: October 2013

A constant-beamwidth, wide-bandwidth (CBWB) loudspeaker array was used to improve speech intelligibility in two rooms. The CBWB loudspeaker array maintained a constant beamwidth between 500 Hz to 4 kHz. The array had a narrow vertical beam pattern and a wide horizontal beam pattern. During the study, an increase in the direct-to-reverberant ratio was observed that correlated with an increase in speech intelligibility (measured with the STI method). The correlation between improved speech intelligibly and an increased direct-to-reverberant ratio will be discussed, and the relation between each source’s beam pattern and the direct-to-reverberant ratio will be presented. The array was compared to a typical PA source and showed increases in the direct-to-reverberant ratio. Room simulations were conducted with CATT-Acoustic and compared to real-world measurements. The study showed an increase in the direct-to-reverberant ratio in each room, which corresponded to improved speech intelligibility.
Noise annoyance reduction using active control
Authors: Mangiante Gerard , Canevet Georges
Published: October 2013

In two former studies [G. Canevet and G. Mangiante ‘‘Psychoacoustic Assessment of Active Noise Control,’’ Active 2004 (2004), and G. Mangiante and G. Canevet ‘‘Active Reduction of Environmental Noise: A Sound Quality Evaluation,’’ Internoise 2006 (2006)], the psychoacoustical effects of active noise control were described. One of these effects is the reduction of noise annoyance. This contribution aims at describing this annoyance reduction using the following indicators: loudness, sharpness, unbiased annoyance, and acoustic comfort. These indicators were calculated with a psychoacoustic-analysis software in which the following models were implemented: Moore and co-workers (1997), Zwicker and Fastl (1999), and Glasberg and Moore (2002). Using this software, various noises, selected from our everyday environment, were tested to investigate the effects of the following parameters: the width and the center frequency of the controlled band, the maximum attenuation in this band, and the sound pressure level of the noise before active control. The results showed that, in all cases, acoustic comfort is positively affected by active control. The annoyance reduction is greatly dependent of the shape of the power spectrum of the noise to be reduced. The best results are obtained for broadband noises with an energy contribution concentrated at low frequencies.
Sonoluminescence: Theoretical explanation and comparison to experiment pertaining to the transduction of sound into radiated light
Authors: Woodsum Harvey
Published: October 2013

The subject of sonoluminescence, i.e., the transduction of sound into light via the intermediate step of cavitation bubble collapse, is of general interest and has sparked a great deal of published research. The present work considers the integration of bubble dynamics with the nonlinear generation of light via the Euler-Heisenberg theory [W. Heisenberg and H. Euler, Z. Phys. 98, 714 (1936)], in relation to Schwingers’ conjecture that bubble cavitation is ‘‘squeezing light from the quantum vacuum’’ via the Casimir effect [Schwinger, Proc. Natl. Acad. Sci., USA 90, 4505–4507, (1993)]. Preliminary results of applying this model appear to show good agreement with both radiated spectral shape and efficiency of sonoluminescent light production, as presented by the relevant literature.
Periodic orbit techniques for mode shape approximation
Authors: Ham Christopher J. , Wright M. C. M.
Published: October 2013

At high frequency, numerical methods for calculating mode shape functions in a waveguide are expensive. The effect of modifying the shape of a waveguide in an industrial design context may only be calculated on a case-by-case basis if numerical methods are used. Periodic orbits are closed, repeating ray paths. A significant body of research in quantum physics indicates that periodic orbits provide fundamental insight into mode theory, which can be applied to acoustics. A method to approximate mode shape functions around short periodic orbits proposed by Vergini [J. Phys. A: Math. Gen. 33, 4709–4716, (2000)] and Babic and Buldryrev [‘‘Short-Wavelength Diffraction Theory’’ (Springer, Berlin) (1991)] is implemented in this paper. In particular, mode shape functions in nonintegrable waveguides are approximated. This method links the geometry of the waveguide to the mode shape functions. The technique is extended to waveguides that have partly chaotic ray dynamics and is applied to the oval and eccentric annulus waveguides. The approximate mode shape functions from these two cases are compared with numerical results.
Noise quality of traffic noise
Authors: Genuit Klaus , Guidati Sandro , Rossberg Sebastian , Fiebig Andre
Published: October 2013

Traffic noise is considered as one major source of noise annoyance in urban context indicated by the increasing number of complaints. The neglect of quality parameters regarding pass-by noise appears as a basic cause for the unreduced annoyance. Several parameters besides the SPL have an important influence on noise evaluation, such as low frequency contributions, temporal aspects or noise patterns. The European research project Quiet City Transport (6FP PL516420) was motivated by the EU Noise Directive 2002/49/EC and is dealing with vehicle pass-by noise and its evaluation in order to identify promising noise mitigation measures. In this framework a quantitative description of noise annoyance using psychoacoustic descriptors was achieved. By means of the detected metric certain perceptual-relevant aspects, such as the difference between diesel and gas, can be reconstructed. Furthermore, the development of a synthesis tool makes the simulation and binaural auralization of pass-by noise, traffic noise as well as specific vehicle noise sources possible. Based on the presented technique the detailed investigation of causes for noise annoyance is carried out. The results will be presented.
Simulating bipolar and monopolar cochlear-implant stimulation in speech recognition by normal-hearing listeners
Authors: Bingabr Mohamed , Espinoza-Varas Blas , Loizou Philip C. , Hawkins Kenneth
Published: October 2013

An algorithm was developed to simulate cochlear-implant (CI) bipolar and monopolar stimulation (BS and MS) in speech recognition by normal-hearing listeners. In MS, the active electrode is inside and the return is outside the cochlea, and electrical current decays at 0.5 dB/mm. In BS, both electrodes are adjacent inside the cochlea and current decays at 4.0 dB/mm to either side of the active electrode [Bruce et al., IEEE Trans. Biomed. Eng. 46, 617 (1999)]. Owing to the faster decay rate, spatial (spectral) resolution is higher with BS than with MS. In recognition of CNC sentences, decay rate was simulated as spectral smearing. Speech was processed through 6, 11, or 22 pass bands, rectified, and low-pass filtered (200 Hz). The passbands amplitude envelopes modulated the amplitude of noise bands equal in center frequency to the passbands. The spectral envelope of each noise band decayed to either side of the center frequency at a rate matching the MS or BS current decay (i.e., noise-band bandwidth was proportional to decay rate). With the summed noise bands, recognition was studied as a function of number of pass bands. [Work supported by NIH.]
Speech intelligibility in noise with single-microphone noise reduction implemented in 9-, 16-, and 24-channel compressors
Authors: Nilsson Michael J. , Hicks Michelle L. , Ghent Robert M. , Bray Victor H.
Published: October 2013

Reception Threshold for Sentences as measured by a modified version of the HINT test [M. J. Nilsson et al., J. Acoust. Soc. Am. 95(2), 1085–1099 (1984)] were collected on hearing-impaired listeners fit binaurally with digital hearing aids incorporating a 9-channel, 16-channel, and 24-channel spectral subtraction-like technique of single-microphone noise reduction. Thresholds were measured in noise presented in a two-dimensionally diffuse soundfield with the subjects listening unaided, aided without noise reduction, and aided with noise reduction. The ability of noise reduction to reduce the level of steady state noise was quantified using 2 cm3 coupler measures with flat linear fittings (to measure the maximum attenuation possible) as well as using each of the listeners’ prescribed fittings (to better relate to the changes in performance measured in the sound field). Previously measured benefit from noise reduction [(M. J. Nilsson et al., J. Acoust. Soc. Am. (2000)] is hypothesized to occur from frequency-specific gain manipulations that increases gain in frequencies where speech is the dominant signal and decreases gain in frequencies where noise is the dominant signal. Additional benefits from smaller, independent channel structures are therefore expected from the 16 and 24 channel systems.
Validation of the noise reduction index (NRI) as an estimate of the signal-to-noise ratio change of a mixed signal by a hearing aid
Authors: Ghent Robert M. , Nilsson Michael J. , Bray Victor H.
Published: October 2013

The NRI is an extension of an idea discussed by Licklider [J. Acoust. Soc. Am. 20, 150–159 (1948)] and implemented in practice by Hagerman and Olofssen [Acta Acoustica 90, 356–361 (2004)] to investigate the conventional wisdom that asserts that a hearing aid cannot fundamentally alter (improve) the input SNR using a single-microphone signal processing algorithm such as digital noise reduction. The present research explores the validity of the NRI by using it to estimate the change in input SNR through a variety of audio devices, including linear hearing aids (with and without directional microphones), a directional microphone designed for noise rejection on a concert stage, an ear trumpet, and a multi-channel hearing aid with a digital noise reduction algorithm both engaged and disengaged. Results indicate that the change in SNR through an audio device can be negative as well as positive, that the NRI is a robust and valid method for estimating this change, and that single-microphone noise reduction algorithms can improve the input SNR when properly designed. The test setup and methodology, along with some pitfalls to avoid when performing these measurements, will be discussed.
Effect of level variation on intensity resolution in intensity-discrimination and increment-detection paradigms
Authors: Jesteadt Walt , Neff Donna L.
Published: October 2013

In studies of profile analysis, the level of the pedestal and increment have been varied together from interval to interval by up to 40 dB to reduce the likelihood that subjects could perform the task by comparing the overall level in the two observation intervals. This range of rove was chosen based on a calculation by Green [Profile Analysis (Oxford, New York, 1988)]. In the present study, adaptive thresholds for intensity resolution were obtained in intensity discrimination (pedestal and increment 100 ms) and increment detection (pedestal 100 ms, increment 300 ms) paradigms as a function of the range of interval-to-interval level variation (0 to 50 dB in 10-dB steps). In roving conditions, the levels of pedestal and increment varied in parallel. In jittered conditions, only pedestal level varied from interval to interval. Increments were 2-kHz sinusoids. Pedestals were 2-kHz sinusoids or narrow-band noises centered on 2 kHz. Effects of level variation were greater in roving than in jittered conditions. Increment detection was unaffected by roving level, whereas thresholds for intensity discrimination were poorer than Green’s best-case prediction by a relatively constant amount for all rove ranges. [Work supported by NIH-NIDCD.]
Vowel identification and vowel space characteristics
Authors: Neel Amy , Andruski Jean
Published: October 2013

Acoustic characteristics of ten vowels produced by 45 men and 48 women from the Hillenbrand et al. (1995) study were correlated with identification accuracy. Global (mean f0, F1 and F2, duration, and amount of formant movement) and distinctive measures (vowel space area, mean distance among vowels, f0, F1 and F2 ranges, duration ratio between long and short vowels, and dynamic ratio between dynamic and static vowels) were used to predict identification scores. Global and distinctive measures accounted for less than one-fourth of variance in identification scores: vowel space area alone accounted for 9% to 12% of variance. Differences in vowel identification across talkers were largely due to poor identification of two spectrally similar vowel pairs /ae/-/eh/ and /uh/-/ah/. Results of acoustic analysis and goodness ratings for well-identified and poorly identified versions of these vowels will be presented. Preliminary analysis revealed that well-identified vowels differed from poorly identified tokens not only in static formant frequencies but also in duration and amount of formant movement over time. A talker’s ability to create distinctions among neighboring vowels is more important in determining vowel intelligibility than overall measures such as vowel space area.
Exploring the natural referent vowel hypothesis in infant perception experiments
Authors: Bohn Ocke-Schwen
Published: October 2013

Directional asymmetries in infant vowel perception studies have led Polka and Bohn [Speech Commun. 41, 221–231 (2003)] to posit the Natural Referent Vowel hypothesis (NRV) according to which vowels that are relatively more peripheral in the F1/F2 space are perceptually privileged vis-a-vis less peripheral vowels. This bias has been observed in preference and in change/no change experiments with infants and with adult non-native listeners. NRV was further tested in headturn (change/no change) experiments with Danish-learning infants. With one important exception, results support the NRV by showing that the predicted asymmetries exist in areas of the vowel space that have not been previously explored. The exception was observed for the Danish front vowel pair /e/-/o//, where NRV incorrectly predicts that a change from the more peripheral /e/ to the less peripheral /o// is harder to discriminate than a change in the opposite direction. Interestingly, the unexpected asymmetry was observed only with infants up to the age of 7.5 months. Older infants showed a nonsignificant trend for an asymmetry in the opposite direction, as predicted by NRV. A modification of NRV to incorporate this finding will be proposed. [Research supported by Grant 25-01-0557 from the Danish Research Council for the Humanities (FKK).]
The value of F0, F3, and F4 in identifying disguised voices
Authors: Andruski Jean E. , Brugnone Nikki , Meyers Aaron
Published: October 2013

This study examines the value of F0, F3, and F4 for identifying speakers from a group of ten male and female speakers when the speakers deliberately changed their F0 and/or their articulatory patterns. The speakers were recorded producing a short passage in their normal speaking voice, a lower than normal speaking voice, and using a vocal disguise of their choice. Although F1 and F2 are strongly affected by articulation, the higher formants show relatively little effect of articulation. This relative stability may make them more useful than F1 and F2 for examining speaker identity. While formants reflect vocal tract cavity dimensions, F0 reflects a different aspect of speaker identity, namely the mass and stiffness of the speaker’s vocal folds. Like F3 and F4, F0 shows relatively little effect from articulation. However, speakers can voluntarily shift F0 into a range that is different from what they normally use. Measurements of F0, F3, and F4 were taken on selected stressed vowels from each recording condition. Measurements from the normal voice condition were used to train a discriminant analysis, which then classified vowels from the low F0 and vocal disguise conditions by speaker. Accuracy of the classification results will be presented and discussed.
Tongue action in silence: Japanese voiceless vowels
Authors: Archangeli Diana , Baker Adam , Twist Alina , Watanabe Yuko
Published: October 2013

The study presents acoustic and articulatory analyses of the Japanese voiceless vowels [(i)] and [(u)] arguing Japanese listeners have internal representations for the vowels and perceive them from articulatory information while physical properties of vowels are absent. Ogasawara (2006) reports that there is no formant band in Japanese voiceless vowels. It is an open question as to whether there is an articulatory gesture associated with the voiceless vowels, and whether this gesture is the same or different as that of voiced vowels. To answer the question, we investigated production of voiceless vowels by a native Japanese speaker. We created 11 pairs of nonsense words such as [hogito] vs [hok(i)to] and [hoguto] vs [hok(u)to]. We recorded the speakers production of the stimuli with the voiceless vowels and their voiced counterparts using the ultrasound device to capture tongue images. The same recording was examined for acoustic analysis. An investigation of the ultrasound images reveals there are tongue gestures for the voiceless vowels, movements differ depending on which voiceless vowel it is, and the movements parallel those of the voiced counterparts. These results argue in favor of the view that Japanese speakers internal representations of words include a representation of distinct voiceless vowels.
Vocal tract normalization for vowel recognition
Authors: Houde Robert A. , Hillenbrand James M.
Published: October 2013

The vocal tract normalization problem arises from the fact that the spectrum envelopes of like vowels spoken by talkers with different vocal tract lengths appear as the same spectral pattern shifted along a log frequency scale. A narrow-band model of vowel perception was developed in which vowels are identified by comparing input narrow-band vowel spectra with broad-band pitch-independent vowel templates [J. M. Hillenbrand and R. A. Houde, J. Acoust. Soc. Am. 113, 1044–1055 (2003)]. An earlier evaluation of this model was carried out on a large database of vowels spoken by men, women, and children. In this evaluation the vocal tract normalization problem was bypassed by constructing separate templates for men, women, and children. The present study extended this work by examining two approaches to the vocal tract normalization problem: (1) An explicit normalization approach in which the spectral pattern of each input token was normalized by a factor based on the pitch of the token and (2) an approach in which templates were assembled by summing together spectral patterns that were sufficiently similar, thus creating multiple templates for each vowel. Performance of the model with these two approaches to vocal tract normalization was compared to listening tests on a large database of vowels spoken by men, women, and children.
Wigner approximation for filtered random functions and wave propagation
Authors: Loughlin Patrick J.
Published: October 2013

We consider the time-varying (Wigner) spectrum of a filtered signal and derive an approximation that shows the effects of the magnitude and phase of the filter on the time-varying spectrum of the original signal. Both the deterministic and random cases are considered. Application to wave propagation with dispersion and damping is discussed.
Ensemble-based Bayesian detection and tracking
Authors: La Cour Brian
Published: October 2013

This paper introduces a novel approach to Bayesian tracking in which the posterior distribution is represented by a random sample or ensemble of possible target states. Such a representation may be viewed as a special case of the traditional particle filter approach wherein all particles maintain strictly uniform weights. Measurement updates are performed using a Markov chain Monte Carlo technique, which has been adapted to use multiple chains and a variable pseudo temperature akin to that used in simulated annealing. The general formalism is illustrated in an example of sonar-based target tracking for antisubmarine warfare. For this example, specific motion models and likelihood functions are developed for both target and clutter hypotheses. The technique is examined in the context of results from a recent multistatic seatrial using an echo repeater target and compared against those of a traditional particle filter. [This work was supported by ONR Contract No. N00014-00-G-0450-21.]