bluuebunny/crossref_metadata_2025 huggingface dataset:
The effect of buffering noise in passenger car
Authors: Iglesias Miguel , Lengua Ismael , Dunai Larisa
Published: May 2015

This paper presents the effect of buffering noise in passenger cars in real conditions at variable speed. At 25% of opened back window, the effect of buffering noise is not perceived. When increasing the aperture of the window to 50%, is perceived the creation of the buffering noise effect with resonance frequency = 15.6 Hz at speed = 80 km/h, but is not irritating because the pressure level is low, pressure level = 110.2 dB. At 75% of the window aperture appears two air flows in the inside car: the one coming from the superior part of the superior zone of the inside and another one from the central zone of the inside. The effect of buffering noise is clearly perceived with resonance frequency = 16.4 kHz at speed = 80 km/h and pressure level = 119 dB and 19.5 Hz at speed = 110 km/h and pressure level at 122 dB. At speed higher than 103–140 km/h, the buffering noise disappears, depending on the car model. For Opel Corsa, the effect of buffering noises appears at the same speed range, whereas for Peugeot and BMW models, it appears three ranges, one from the buffering noise creation, persist in another range, and disappear in the third range.
A feedback active noise control for in-ear headphone with robust on-line secondary path modeling
Authors: Park Young-cheol , Lee Keun-sang , Ji Youna
Published: May 2015

The active noise control (ANC) technique can be used to improve the comfortness and quality of the music listening with an in-ear headphone. For a compact headphone design, the feedback ANC operated with the filtered-x LMS (FxLMS) algorithm is preferred. In practice, the error in the estimated secondary path can be problematic for the FxLMS algorithm, since it can lead the algorithm to an unstable state, and thus cause degradation of sound quality with the in-ear headphone. In this study, an adaptive residual music canceler (RMC) is proposed for enhancing the accuracy of the reference signal of the feedback ANC. Since RMC is designed to track the bias of the current secondary path estimate, the secondary path is continuously updated by combining the previous estimate of the secondary path with the current weight vector of RMC. In addition, variable step-size schemes are developed for both the control and secondary path estimation filters, which enable the ANC system to adapt quickly and robustly to the variation of the physical secondary path. Simulation results show that the secondary path can be accurately estimated and high quality of music sound can be consistently obtained in a time-varying secondary path situation.
Wind-induced noise based upon stability dependent turbulent velocity spectrum models
Authors: Hart Carl R. , Wilson D. Keith
Published: May 2015

The ambient sound level is an important datum for noise control, quantifying soundscapes, and remote detection. For low acoustic frequencies and fast enough wind speeds, wind-induced noise contributes largely to the measured ambient sound spectrum. Predictive models for wind-induced noise are based upon an empirical relationship between fluctuating pressure, mean wind speed, and the fluctuating streamwise wind velocity. A wind noise model developed by Van den Berg [Van den Berg, G. P., J. Acoust. Soc. Am. 119(2), 824–833, 2006] is derived from a turbulent velocity spectrum, which is applicable for neutral atmospheric stability and flat homogeneous terrain. In this work, several other wind noise models are formulated based upon different turbulent velocity spectrum models. The turbulent velocity spectrum models take into account atmospheric stability, with varying degrees of complexity. Differences in predicted wind noise levels among the models show nearly a 3 dB spread for some atmospheric conditions. A recommendation is given for a specific wind noise model based on intercomparisons.
Statistical properties of the mirror flow model of boundary-layer turbulence
Authors: Lyons Gregory W. , Murray Nathan E.
Published: May 2015

The mirror flow model is a construction of inhomogeneous, anisotropic turbulence from the superposition of a homogeneous, isotropic turbulent field with itself in transformed coordinates. This model has been used for atmospheric turbulence to predict spectral contributions to low-frequency wind noise [Yu et al., J. Acoust. Soc. Am. 129(2), 622–632 (2011)]. Here, analytical solutions are reported for the longitudinal and vertical turbulence spectra as a function of elevation. The correlation functions, integral turbulence scale, and profiles of mean-square velocity components are computed. The premultiplied spectra show remarkable agreement with observed scaling in the atmospheric boundary layer. Implications for computation of wind noise spectra are discussed.
A comparison of sintered materials for the reduction of wind noise in acoustic sensors
Authors: Solomon Latasha , Gonski David , Tenney Stephen , Sim Leng
Published: May 2015

Wind noise reduction has been of particular interest in the acoustic sensor arena for many years. Ideally, a desirable noise-mitigating medium is resilient to extreme temperatures and dust, as well as insects/animals. Open cell polyurethane foam is the most widely chosen material for acoustic wind noise reduction. These windscreens tend to provide protection from humidity and moisture, as well as reduction of wind flow noise at the frequencies of interest (500 Hz and below). However, after prolonged use in the environment, degradation of the windscreen and its performance is inevitable. Exposure to high levels of UV rays tends to make the windscreens brittle, while sand and dirt may become lodged in the pores requiring frequent cleaning or replacement, and they can be vulnerable to wildlife. Ideally, one would like to replace the current foam windscreen with a more durable, smaller, lower-profile medium providing similar wind noise rejection to that of foam windscreen. This research will compare the effectiveness of porous metal and plastic windscreens fabricated from sintered material to that of the conventional open cell foam windscreen in reducing wind generated flow noise.
Reduced-size windscreens for infrasound
Authors: Alberts W. C. Kirkpatrick , Ludwig William D. , Talmadge Carrick
Published: May 2015

The long wavelengths of infrasound necessitate large, often complex, wind noise reduction apparatus, e.g., porous hose, pipe arrays, or wind fences. For the infrasonic sources of military interest that fall between approximately 2 and 20 Hz, 6-m radius porous hose rosettes are often sufficient windscreens. However, the number of hoses required for a typical 4-element infrasound array (16–24) reduces its portability. Based upon recent research demonstrating that porous fabric domes could outperform porous hoses in the 2–20 Hz region and a need for more compact windscreens, 1.2-m diameter domes covered in three types of porous fabric and one folding perforated aluminum dome were investigated for their suitability as alternatives to porous hose. Evaluations of wind noise reduction, transfer functions, and environmental durability will be presented for these alternative windscreens.
Additions to a single CI to improve speech understanding
Authors: Gifford René H. , Loiselle Louise , Cook Sarah , Davis Timothy J. , Sheffield Sterling W. , Dorman Michael F.
Published: May 2015

Speech understanding was assessed in a simulated restaurant and cocktail party environment for 39 adult cochlear implant recipients (18 bilateral, 21 hearing preservation). Speech was presented as either collocated (S0N0) or spatially separated from the noise (S0N0-360 or S0N90&270). We calculated the benefit of adding a second CI, acoustic hearing from the non-implanted ear, and the implanted ear as well as spatial release from masking. The addition of either a second CI or binaural acoustic hearing yielded mean benefit of 13-percentage points. The additional benefit afforded via acoustic hearing in the implanted ear in addition to the bimodal configuration was an additional 10-percentage points. Little-to-no spatial release from masking was noted for either the simulated restaurant or the cocktail party environments. Spatial release from masking was not found to be significant at the group level nor was it different across the hearing configurations. We draw two primary conclusions: (1) adding either a second CI or binaural acoustic hearing to a single CI yields similar levels of benefit, but with different underlying mechanisms, and (2) the lack of spatial release is likely related to the presence of noise at both ears directed toward the ports of the HA and CI microphones.
Language emergence in early-implanted children
Authors: Geers Ann E. , Nichols Johanna G. , Tobey Emily A. , Davidson Lisa
Published: May 2015

This study develops a model to differentiate early-implanted children with preschool language delays that persist into the elementary grades from children who achieve normal language as they gain auditory, linguistic, and academic experience. A nationwide sample of 60 children with congenital profound deafness, auditory-oral education, and early cochlear implantation (12–36 months) was assessed at average ages of 3.5, 4.5, and 10.5 years. Children were classified with: Normal Language Emergence (NLE: N = 19), Late Language Emergence (LLE: N = 22), or Persistent Language Delay (PLD: N = 19) based on standardized language test scores. Children with PLD did not differ from children with LLE on a number of variables that have previously been associated with post-implant outcome, including pre-implant hearing, age at first CI, gender, maternal education and nonverbal intelligence. Logistic regression analysis identified variables that predicted whether children with early language delay would catch up with hearing age-mates: (1) preschool speech and language level, (2) ear placement of first CI, (3) upgrades in speech processor technology, and (4) CI aided thresholds. A majority of the 60 children in this study reached age-appropriate levels of language, reading, and verbal reasoning and were placed in primary grades with hearing age-mates.
Pitch perception and representations of frequency in the peripheral auditory system: What’s missing in cochlear implants?
Authors: Oxenham Andrew J.
Published: May 2015

Pitch is a crucial feature of auditory perception. It conveys information about melody and harmony in music, and conveys prosodic and (in tone languages) lexical information in speech. Although pitch has been studied formally for many decades, there still remains considerable uncertainty regarding how frequency and pitch information is coded and represented in the auditory periphery. This uncertainty means it is unclear whether and how pitch information can best be conveyed via cochlear implants. Cochlear-implant users have been shown to be sensitive to changes in the place of stimulation within the cochlea (“place” pitch) and to changes in the pulse rate presented to the electrode (“rate” pitch), at least up to rates of around 300 Hz. Place pitch in cochlear implants is probably best compared to “brightness” and rate pitch has most in common with the relatively weak pitch produced by temporal-envelope fluctuations in normal-hearing listeners. Neither type of pitch seems to provide the salient and accurate pitch sensations associated with normal hearing. Studies in both normal-hearing listeners and cochlear-implant users will be reviewed to shed light on the mechanisms of pitch perception and on the challenges involved in restoring pitch to cochlear-implant users. [Work supported by NIH grant R01DC005216.]
The sound of a cochlear implant: Lessons learned from unilaterally deaf patients fit with a cochlear implant
Authors: Dorman Michael , Cook Sarah , Zeitler Daniel
Published: May 2015

Starting about 10 years ago in Europe, patients who had one normally hearing ear and one deafened ear were fitted with cochlear implants in an effort to suppress tinnitus in the deafened ear. That effort was very successful. More recently, single sided deaf patients without tinnitus have been implanted in an effort to improve health-related quality of life. This effort, too, has been successful. This patient population, with one normal hearing ear and one implanted ear, offers an unprecedented window into the sound quality of a cochlear implant. That is, these patients can tell us whether a candidate stimulus presented to the normally hearing ear sounds like the stimulus presented to the cochlear implant. In this talk, we describe our efforts to reproduce the sound of a cochlear implant.
Transforming acoustic sensing with high refractive index metamaterials
Authors: Yu Miao , Chen Yiongyao
Published: May 2015

Acoustic sensors play an important role in many areas, such as safety (e.g., sonar arrays), public health (e.g., ultrasonic imaging), surveillance (e.g., underwater communication and navigation), and industry (e.g., non-destructive damage detection). However, conventional acoustic sensors inevitably suffer from the fundamental pressure detection limit, which hinders the performance of current acoustic sensing technologies. Here, we design high refractive index acoustic metamaterials that have strong wave compression effect to obtain direct amplification of pressure fields in metamaterials. This enables a novel metamaterial enhanced acoustic sensing system that can help overcome the detection limit of conventional systems. Through analytical, numerical, and experimental studies, we demonstrate that this novel acoustic sensing system can help achieve over an order of magnitude enhancement in acoustic pressure detection limit. This will allow the detection of weak acoustic signals below the noise floor of the conventional acoustic sensing systems. This work is expected to impact many fronts that require high performance acoustic sensing.
Ultrasonic subwavelength focusing above two dimensional membrane metasurface using time reversal
Authors: Lani Shane W. , Sabra Karim , Degertekin F. Levent
Published: May 2015

Surface acoustic waves can propagate above immersed membrane arrays, such as of capacitive micromachined ultrasonic transducers (CMUTs). Similar waves on metamaterials and metasurfaces with rigid structures (typically in the kHz range) have been studied and used for tunable band gaps, negative refraction, and subwavelength focusing and imaging. This work demonstrates through simulation and experiments that a 2D membrane array can be used for subwavelength focusing utilizing a time reversal method. The studied structure consisted of the focusing region, which is a dense grid of 7x7 membranes (6.6 MHz resonance) that support the slow surface acoustic waves. Eight additional membranes are located on the same surface outside the focusing region. Subwavelength focusing was performed by using a time reversal method in which the external eight membranes were used as excitation transducers. Modeling results were verified with experiments that were performed with the membranes being actuated electrostatically and the membrane displacements were measured with a laser Doppler vibrometer. Subwavelength focusing (lambda/5) was achieved on the metasurface while a modal decomposition of the spatial focus from an iterative time reversal method was done to illustrate that optimal focusing resolution requires efficient excitation of the mode shapes containing subwavelength features.
Enhanced acoustic transmission through a slanted grating
Authors: Norris Andrew
Published: May 2015

It is known that an acoustic wave incident on an infinite array of aligned rectangular blocks of a different acoustic material exhibits total transmission if certain conditions are met which relate the unique “intromission” angle of incidence with the geometric and material properties of the slab. This extraordinary acoustic transmission phenomenon holds for any thickness of the slab, making it analogous to a Brewster effect in optics, and is independent of frequency as long as the slab microstructure is sub-wavelength. Here, we show that the enhanced transmission effect can be obtained in a slab with parallel grating elements oriented obliquely to the slab normal. The dependence of the intromission angle θi is given explicitly in terms of the orientation angle. Total transmission is also achieved at incidence angle -θi although there is a relative phase shift between the transmitted amplitudes of the +θi and −θi cases. These effects are easily understood when the grating elements are rigid. In particular, any angle of intromission can be obtained by with thin rigid elements by orienting them to the desired value of θi.
Effect of waveguide impedance on radiation in underwater acoustic leaky wave antennas
Authors: Naify Christina J. , Guild Matthew , Calvo David C. , Orris Gregory J.
Published: May 2015

Leaky wave antennas (LWAs) have been examined for decades as a way to steer electromagnetic waves as indexed by input frequency, enabling rapid video scanning of an environment with a simple compact device. Recently, an LWA device constructed of a rigid waveguide with open shunts was shown to produce a similar steering effect for acoustic waves in air [Naify et al., Appl. Phys Lett. 102, 203508 (2013)]. The shunts serve two purposes, to couple acoustic energy from the waveguide to the surrounding area, as well as control directionality of the radiated wave by changing the impedance, and thus modulating the phase speed of the waves in the waveguide. While this impedance contrast of the waveguide wall relative to that of the surrounding fluid is very large for an air background, when designing a structure for use in water the material of the waveguide can drastically change the directionality of the radiation profile. This study examines the effect of waveguide wall impedance on directionality of an acoustic LWA using both numerical and finite element methods. Three material impedance conditions are examined, including acoustically rigid, and two materials with finite impedance. [Work sponsored by the Office of Naval Research.]
Hysteresis in an extension of Titze's surface wave model and contact with phonation onset/offset experiments
Authors: Fulcher Lewis , Scherer Ronald
Published: May 2015

Three modifications of Titze’s surface wave model are required to incorporate the property of hysteresis. (1) Adding a nonlinearity to the damping term of this model allows it to undergo sustained oscillations above the onset threshold, and thus the model becomes more robust. (2) Adding a nonlinearity to the stiffness term gives the model a frequency dependence that allows contact with the dynamic viscosity measurements carried out by Chan and Rodriguez [J. Acoust. Soc. Am. 124, 1207–1219 (2008)]. These measurements observed a reduction in the dynamic viscosity as the frequency of oscillation increased. In order to include this property in the surface wave model, (3) the size of the damping constant is reduced whenever the energy of the oscillating vocal fold exceeds a certain value. Thus, the nonlinear properties introduce a dependence of the damping term upon the oscillation history of the vocal fold and set the stage for hysteresis effects. Calculations with the extended surface wave model are compared with the observed onset/offset pressures measured by Titze, Schmidt, and Titze [J. Acoust. Soc. Am. 97, 3080–3084 (1995)] and by Chan and Titze [J. Acoust. Soc. Am. 119, 2351–2362 (2006)]. The extended model readily provides an explanation for why the observed phonation offset pressures are always lower than the onset pressures as well as several other trends in the data.
Fundamental frequency movements in one-word imperatives
Authors: Robles-Puente Sergio
Published: May 2015

F0 contours in Spanish declarative and imperative sentences have traditionally been described as identical and it has not been until recently that several phonetic variables have been identified as markers of imperativity. These include higher overall F0s, early peak alignments in pre-nuclear pitch-accents, upstepped nuclear pitch-accents and marginal use of high boundary tones. Since previous analyses have concentrated on utterances with multiple pitch-accents, not much is know about productions with just one word. This study focuses on one-word utterances given the tendency of imperatives to be short and taking into account that some of the aforementioned phonetic markers cannot be used with a sole pitch-accent. The analysis of 117 declarative sentences and 256 imperatives produced by eight Spanish speakers demonstrated that in cases where declaratives and imperatives share the same contour (L+>H*L%), the latter tend to have higher F0 peaks (Wilcoxon signed-rank test; p<0.000). Besides, 32.4% of the imperatives showed F0 contours ending in high boundary tones, a marker not used in declaratives and only marginally found in imperatives with multiple pitch-accents. Current findings suggest that when segmental material limits F0 movements, speakers look for alternative phonetic strategies to distinguish between declaratives and imperatives.
Effect of “only” on prosodic focus marking
Authors: Chalmers Elizabeth , Howell Jonathan
Published: May 2015

This study seeks to determine how the inclusion of the focus-sensitive word “only” affects the focus-marking prosody of a sentence. Twenty-one native English speaking adults read 8 question-answer pairs. The questions were written to elicit focus on an object NP (e.g., Who did you see?). In one condition, the answer contained only (e.g., I only saw Mary); in the other condition, “only” was omitted (e.g., I saw Mary.). The recordings were annotated in Praat using forced alignment. We performed linear residualization of F0, amplitude and duration (cf. Breen et al. 2009) to remove effects of item and participant. Statistical models of residual pitch and duration on object-NP and verb failed to show any significant differences between the sentences that contain “only” and those not containing “only”. These results fail to support theories of utterance-level prominence, which posit a categorical distinction between presentational and contrastive focus (e.g., Katz and Selkirk 2011).
Fathers' use of fundamental frequency in motherese
Authors: VanDam Mark , De Palma Paul , Strong William E.
Published: May 2015

Studies of motherese or child-directed speech (CDS) have paid scant attention to fathers' speech when talking to children. This study compares mothers' and fathers' use of CDS in terms of fundamental frequency (F0) production, examining natural speech from a very large database of hundreds of hours of family speech including mothers, fathers, and preschool children. The day-long recordings are collected with specialized recording software and body-worn hardware, then analyzed with automatic speech recognition (ASR) technology (LENA Research Foundation, Boulder, CO). CDS is defined here as speech in a conversational exchange between the parent and child, and adult-directed speech (ADS) is speech between adults or speech in which the child is not a (vocal) participant. Results confirm many reports in the literature of mothers' increased F0 during CDS. Results fail to show a difference in the F0 characteristics between fathers' CDS and ADS speech. This shows that children's linguistic experience with fathers is different than with mothers. This result could be useful to improve ASR techniques and better understand the role of usage in natural language acquisition and the role fathers play in the language acquisition process.
Real-time three-dimensional ultrasound imaging of pre- and post-vocalic liquid consonants in American English
Authors: Rhodes Brandon , Berkson Kelly , de Jong Kenneth , Lulich Steven
Published: May 2015

Speech sound articulation is typically characterized in the mid-sagittal plane. However, lateral segments, such as the /l/ category in most varieties of English, are usually identified in introductory phonetics courses as exceptions to this rule. On the other hand, many productions of post-vocalic /l/ in varieties spoken in the lower Mid-west U.S. are noted as being non-lateral, involving only a dorsal articulation very different from the canonical coronal occlusion. Furthermore, a large body of literature indicates multi-constriction articulations, which vary by syllable position, for liquids like American English /l/ and /r/. This research presents results from a study of constriction location and laterality in pre- and post-vocalic /l/ and /r/ using real-time 3D ultrasound images of tongue motion, digitized impressions of the palate, and time-aligned acoustic signals.
Kinematic properties of concurrently recorded speech and body gestures and their relationship to prosodic structure
Authors: Krivokapic Jelena , Tiede Mark , Tyrone Martha E.
Published: May 2015

Previous research has shown that body gesturing, such as hand and torso movements, is related to prosodic structure (e.g., Esteve-Gibert & Prieto 2013). However, the details of the relationship are not well understood and specifically the effects of prosodic structure on gesturing duration have received little attention. We present an experiment investigating the effects of stress and prosodic boundaries on the temporal properties of body gesturing and gestures of the vocal tract. Two speakers read 6 repetitions of 12 sentences examining the effect of boundary (no boundary, ip boundary, IP boundary), stress (no stress, stress) and their interaction, phrase finally and phrase initially (144 sentences total). They were observed through concurrent recordings of speech audio, vocal tract gestures using electromagnetic articulometry, and body gesturing using Vicon motion capture. We test the hypothesis that manual movements lengthen under prominence and at boundaries and that the lengthening at boundaries is cumulative, parallel to the behavior of gestures of the vocal tract. Analyzing speech prosody and body gesturing in this manner will allow us to investigate whether prosodic control extends to body gesturing and to identify coordinative patterns that are relevant to fields such as language pedagogy and speech pathology. [Work supported by NIH.]
The contribution of temporal modulations to speech intelligibility: Some thoughts about the speech envelope
Authors: Grant Ken W.
Published: May 2015

Rosen [Phil. Trans. R. Soc. Lond. B 336, 367–373 (1992)] laid out an extremely useful framework for discussing temporal information in speech. Three broad categories were defined: temporal envelope (2–50 Hz), periodicity (50–500 Hz), and temporal fine-structure (600–10,000 Hz). The slower modulations associated with the speech envelope convey segmental cues for voicing and manner-of-articulation, and prosodic cues for syllabification and stress. These cues are manifest in the attack and fall times as well as the overall duration of the envelope segments. The mid-rate fluctuations associated with periodicity convey manner cues (periodic vs aperiodic) as well as voice pitch (an essential cue for intonation and stress). The faster rate modulations associated with temporal fine-structure (modulations within each pitch period) convey place-of-articulation cues and spectral-shape cues for manner-of-articulation and voicing. Studies that try to separate the effects of each of these temporal categories on speech perception often use signal processing strategies that filter out the temporal modulations of interest and present this information without periodicity or temporal fine-structure to listeners. The processing details of how the temporal envelope is extracted are critical to whether cues more readily associated with periodicity or fine-structure are inadvertently presented to listeners. Examples will be provided.
Contributions of the temporal envelope in adverse listening conditions
Authors: Fogerty Daniel
Published: May 2015

Amplitude modulations of the temporal envelope provide important cues for intelligibility, particularly in quiet listening environments. However, the speech understanding of listeners based on this temporal speech signal is often impaired when temporal interruptions or maskers interfere with the transmission of this important information. Performance in these adverse listening conditions is variable, based, in part, upon the properties of the interfering masker and the degree to which the modulation properties of the target speech signal are preserved. In some cases, listeners may be required to shift perceptual weighting to alternate, preserved modulation cues in order to maximize intelligibility of the speech signal. However, modulation properties of the speech signal are related to specific speech events. Therefore, different modulation properties are likely to contribute diverse information for speech understanding. The ability to access and process temporal modulation properties of the speech signal that unfold over different timescales is essential for maximizing speech intelligibility in everyday listening conditions. [Work supported by NIH/NIDCD and ASHA.]
Segmental contribution to the intelligibility of noise-suppressed speech
Authors: Chen Fei
Published: May 2015

Many studies suggested that most speech enhancement algorithms do not improve the subjective intelligibility scores of noise-suppressed speech when presented to normal-hearing listeners. Nevertheless, the reason for lacking intelligibility improvement is still unclear. This study assessed the segmental contribution to the intelligibility of noise-suppressed speech. Mandarin sentences were corrupted by steady-state speech-spectrum shaped noise and multi-talker babble. Stimuli were synthesized by using noise-suppressed speech processed by three types of speech enhancement algorithms and a noise-replacement paradigm. The noise-replacement paradigm preserved selected speech segments (i.e., vowel-only and vowel-plus-consonant-onset), and replaced the rest with noise. Listening experiments were performed to collect the intelligibility scores from normal-hearing listeners. Experimental results showed that the selectively synthesized stimuli (i.e., vowel-only and vowel-plus-consonant-onset) from noise-suppressed speech may be more intelligible than those from noise-corrupted speech. However, this benefit was only observed when the speech signal was corrupted by speech-spectrum shaped noise but not by babble masker. This work suggested that the segmental distortion caused by speech enhancement algorithms may affect the information integration from noise-suppressed speech, and high-level top-down processing may account for the deficiency of the present speech enhancement algorithms for improving subjective intelligibility performance.
Using speech intelligibility metrics to predict the performance of hearing-impaired listeners
Authors: Payton Karen
Published: May 2015

When fitting a hearing aid to a hearing-impaired listener, it would be very beneficial for an audiologist to be able to use an objective metric to determine the best signal processing algorithm to compensate for the user’s loss and improve the user’s understanding of speech in degraded acoustic environments. Several metrics have been considered to predict impaired listeners’ speech recognition, including modified versions of the Speech Intelligibility Index (SII) [e.g., Ricketts, Ear Hear. 17, 124–132 (1996)] and the Speech Transmission Index (STI) [e.g., Humes et al., J. Speech Hear. Res. 29, 447–462 (1986)]. For impaired listeners using linear amplification hearing aids, there are metrics that work fairly well. For listeners using hearing aids that include nonlinear processing, such as amplitude compression, results have not been as successful. This talk will review the current status of metric capabilities, approaches that have been proposed to mitigate metric limitations and which metric features appear most promising for future research.
Investigating the effectiveness of Globalvoice CALL software in native Japanese English speech
Authors: Obari Hiroyuki , Kojima Hiroaki , Itahashi Shuichi
Published: May 2015

Many variables exist in the evaluation of the speech of non-native speakers of English. In previous research, rhythmic accent and pauses are more salient than segmental features in English utterances to make one’s speech more intelligible. In this regard, prosodic features such as intonation and rhythm are crucial in comprehensible speech. One of the main goals of English education in Japan is to help Japanese students speak English more intelligibly in order to be more clearly understood while taking part in international communication. In this talk, several key parameters such as speech duration, speech power, and F0 (pitch), are all introduced to help determine to what extent Japanese students are able to improve their English pronunciation through the use of Globalvoice CALL software. Globalvoice CALL enables students to input English words or sentences for the purpose of practicing their pronunciation and prosody in order to reduce their Japanese-accented speech. The purpose of this paper is to investigate the effectiveness of Globalvoice CALL software to improve the English pronunciation of Japanese EFL learners. Student productions of English sentences were analyzed with regard to speech duration, speech power, and F0 (pitch) to determine how much progress was attained in their English pronunciation and overall proficiency. The results showed improvement in acoustic properties between the pre- and post-recorded readings, indicating the software helped the students to improve their English pronunciation.
Contribution of the acoustic cues to the non-native accent
Authors: Laprie Yves
Published: May 2015

This communication concerns the contribution of acoustic cues to the perception of non-native accents and presents some solutions to help learners to master the acoustic cues of the foreign language. Elementary acoustic cues (F0 frequency, formants…) will be presented by using copy synthesis used to pilot the Klatt formant synthesizer. This enables their perceptual contribution and that of more complex acoustic cues, which play a major role in the identification of speech sounds, to be evaluated easily. The non-native accent is mainly due to the:—realization of incorrect tones, stress patterns, and more generally prosody,—use of the mother tongue phonemes instead of those of the second language,—use of inappropriate acoustic cues to realize a phonetic feature. This is often the case with the voiced/unvoiced feature which covers a range of acoustic cues. These deviations depend on the phonetic specificities of the mother tongue with respect to those of the second language. The first aspect of corrections is the automatic diagnosis of non-native deviations. Learners can be made aware of them, trained to learn new phonetic contrasts, or guided by providing them with appropriate articulatory information, or acoustic feedback which is particularly efficient for prosody.
Paul Langevin’s contributions to the development of underwater acoustics
Authors: Sabra Karim G.
Published: May 2015

Paul Langevin made significant contributions to the understanding of piezoelectricity and the development of piezoceramics materials. For instance, Professor Langevin's invented the quartz sandwich transducer in 1917 for underwater sound transmission. He subsequently used this to develop the first underwater Sonar for submarine detection during World War I, and his work was extensively used the French and American Navy. After world war I, SONAR devices he helped developed were used in several French ocean-liners. We will review Paul Langevin’s most noteworthy contributions to the development of underwater acoustics.
Underwater acoustics research at the Woods Hole Oceanographic Institution, 1930–1960
Authors: Lynch James , Newhall Arthur , Frosch Robert
Published: May 2015

The Woods Hole Oceanographic Institution (WHOI) was founded in 1930, and throughout its history has had a strong involvement in research into the science and applications of sound in the ocean. In terms of a brief history, three eras stand out: (1) pre-WWII, (2) WWII, and (3) the postwar years. Though the most colorful pictures and stories come from the war years, the other two eras also hold much interest. Many of the names associated with research at WHOI also commonly appear associated with other laboratories, and the interplay between various institutions is a fascinating part of the history. Personal reminisces, technical report details, photos, audio, and even some video records will be presented, courtesy of the WHOI Archives.
Columbia Division of War Research and the work of Ewing, Worzel, and Pekeris
Authors: Knobles D. K. , Westwood Evan K. , Muir Thomas G.
Published: May 2015

Three University laboratories were established in WWII to conduct underwater acoustics research, in response to widespread attacks on allied shipping by German submarines; the Columbia University Division of War Research, Harvard Underwater Sound Laboratory, and the University of California Division of War Research, and the last two of these are being discussed by other speakers in this session. During the war, outstanding work was done at Columbia by Maurice Ewing, J. Lamar Worzel, and C. L. Pekeris, which has been summarized in the book, Propagation of Sound in the Ocean. Experiments utilizing explosive sources were done at sea in 1943 and 1944, in both shallow and deep water, off the east coast of the United States, with the USS Saluda (IX87) and other vessels, in cooperation with the Naval Ordnance Laboratory, Woods Hole Oceanographic Institution, and the U.S. Navy Underwater Sound Laboratory. Pekeris is noted for his wave theoretic analysis of the shallow water propagation results and is credited with being the first to apply normal mode theory to shallow water waveguides. Ewing and Worzel are noted for their development of some of the first long range ocean acoustics experiments, which continue to this day in similar context and importance. Some highlights of this work are presented to illustrate the beginnings of long range underwater acoustic propagation research in the United States.
The behavioral impacts and spatial dynamics of alarm fatigue in reverberant healthcare treatment spaces
Authors: Barach Paul
Published: May 2015

Medical device alarms are deliberately designed to alert attention. Conditions found in hospitals produce the unintended consequences triggered by alarms called “alarm fatigue” that is not attributable to individually alarmed devices but rather to the aggregate conditions in which the alarms occur. “Alarm fatigue” is a condition of recurrent noises from myriad uncorrelated medical devices, set at maximum loudness, occurring in hard-walled, reverberant spaces and produces elevated stress, sleep impairment, disorientation and dangerously irrational behavior. Four conditions cause “alarm fatigue": (1) burgeoning use of alarmed devices none of which are prioritized or correlated with others and all of which are set to maximum sound pressure levels; (2) advanced technologies that have caused a steady increase in hospital noise levels; (3) small enclosed spaces that surround dense clusters of healthcare workers and patients with hard, sound-reflecting surfaces and high levels of noise; and, (4) codes and standards that require alarms to be significantly louder than background, ambient sound. These conditions produce an “acoustic feedback loop” in which noise inevitably and rapidly escalates to intolerable levels and interferes with behavior. Understanding these conditions and developing practical and effective ways to interrupt “acoustic feedback loops” requires trans-disciplinary approaches and mixed implementation methods that include health service researchers, signal-processing, bioacoustics, and room acoustics.
Assessing hazardous noise pollution in the United States
Authors: Neitzel Richard L. , Hammer Monica S. , Swinburn Tracy K.
Published: May 2015

The risk of noise-induced hearing loss (NIHL) from noise exposure has been known for hundreds of years, but Karl Kryter’s seminal work on NIHL represented a major step forward in addressing this disease. In recent years, a growing body of evidence has linked noise to additional non-auditory health effects, including cardiovascular disease, sleep disturbance, and stress. The United States has no national plan to address noise pollution, although standards designed to protect human health from noise do exist. National estimates of U.S. exposures were last created in the early 1980s. We have updated these estimates using current census and research data, and estimate that 104 million individuals in the United States had annual equivalent continuous average levels >70 dBA in 2013 and were at risk of NIHL and other effects. We describe approaches to the assessment of noise exposures at the level of the individual and the community, and outline ways in which noise can be more comprehensively assessed. Based on the number of exposed individuals in the United States and the impact of noise-related health effects, greater emphasis on noise reduction will likely improve public health and quality of life. This increased focus will allow us to continue and expand upon Kryter’s outstanding work.
A challenge to Articulation Theory: Narrow-band acoustic signals and visual speech cues
Authors: Grant Ken W.
Published: May 2015

Given Kryter’s extensive work on the physiological and subjective effects of noise on the human auditory system it is not surprising that at some point he would turn his attention to the impact of noise on speech intelligibility. In 1962, Kryter published “Methods for the Calculation and Use of the Articulation Index” greatly simplifying the methods originally proposed by Fletcher in the 20s and 30s and French and Steinberg in 1947, ultimately leading to the ANSI 1969 standard. Over the years, the AI and the revised Speech Intelligibility Index (SII) have become the leading metric for predicting speech recognition in noise and in quantifying the audibility provided by amplification. However, the scope of the AI/SII is still limited to communication conditions, which do not include multiple, narrow bands of speech or noise. Similarly, predictions for conditions of auditory-visual speech are only roughly approximated. Data from a variety of studies suggest that rather than ignore these communication conditions they should be studied closely for what they reveal. In this talk, I will show data that expose chinks in the armor of our understanding of speech perception in noise and under conditions where visual speech cues are present.
The effects of noise on learners of English as a second language
Authors: Rogers Catherine L.
Published: May 2015

In the 65 years since the publication of The Effects of Noise on Man, as the world has grown ever noisier, it has also grown more diverse and more connected than ever before. Today, nearly a fifth of U.S. residents speak a language other than English at home. Consequently, exploring the effects of noise on people communicating in a second language is a crucial component of understanding the effects of noise on mankind in our increasingly pluralistic society. This presentation will overview some of the challenges faced by second-language learners as speakers and listeners, particularly focusing on the extraction of phonetic information from a noisy or impoverished signal. Results of ongoing research investigating vowel identification by native and non-native English-speaking listeners will be presented. Overall accuracy, the slope of the function relating signal-to-noise ratio and intelligibility, and effects of noise on listeners’ ability to benefit from phonetic enhancement strategies will be compared across native English speakers and second-language learners with differing ages of immersion in an English-speaking environment.
Frequency power-law attenuation and dispersion in marine sediments
Authors: Buckingham Michael J.
Published: May 2015

Acoustic waves in marine sediments, including the fine-grained muds and clays, often exhibit an attenuation in the form of a frequency power law in which the exponent is close to unity. The frequency dependence of the wave speed, or dispersion, associated with such a power law has long been a subject of debate. Causality arguments, often in the form of the Kramers-Kronig dispersion relations, are usually applied to the problem, but these are not sufficient to characterize the dispersion fully. Following an alternative approach, a wave equation will be introduced which predicts the dispersion when the attenuation is of the frequency power law type. Based on this wave equation, it will be shown that only certain values of the frequency exponent are allowed, otherwise causality is violated, indicating that the associated attenuation and dispersion are not physically realizable. In effect, for these prohibited values of the frequency exponent, the Fourier components in the dispersion and attenuation cannot combine to give a zero response for negative times. Emerging from the theory is an expression for the complex wavenumber that is complete and exact, and from which all the properties of the dispersion and attenuation may be derived.
Tentative acoustic wave equation that includes attenuation processes for mud sediments in the ocean
Authors: Pierce Allan D. , Siegmann William L.
Published: May 2015

Compressional wave attenuation in mud sediments is less than in sandy/silty sediments and comparable to that of sea water. Experimental results by Wood and Weston (Acustica,1964) yield 0.07 dB/m for a frequency of 1 kHz, while the extensive data analysis of Holmes, Carey, Dediu, and Siegmann (JASA-EL, 2007) suggests a value of 0.33 dB/m for sandy/silty sediments. The linear frequency dependence over the range of 4 to 50 kHz reported by Wood and Weston suggests that the cause of the attenuation is relaxation processes rather than viscosity processes. The tentative appropriate relaxation process is suggested, in accord with previous ideas reported by Mahmood et al. (Colloids and Surfaces A, 2001), to be the breaking and reforming of van der Waals-force-related bonds between clay platelets that are connected edge to face. The present paper's theoretical development of a single wave equation with relaxation included parallels that discussed on pp. 550–553 of the text by Pierce (Acoustics, 1981) and follows the ideas of Liebermann (Phys. Rev., 1949). Predicted sound speed is in accord with the Mallock-Wood equation, and the attenuation term appears as an integral over past history, in accord with the envisioned frequency-dependent attenuation.
Modeling transmission loss over stratified muddy sediments based on chirp sonar imagery
Authors: Tang Dajun , Hefner Brian T. , Yang Jie , Jackson Darrell R.
Published: May 2015

The Office of Naval Research is sponsoring a shallow water acoustics experiment offshore New Jersey where the sediment consists nominally of a layer of mud over sand, and where the layer thickness is about 10 m. A preliminary chirp sonar survey shows that there exists fine layering within the mud layer. Nearby core data also reveal the existence of fine layers within the mud. One of the interesting acoustic questions is what impact such fine layering within the muddy sediments has on acoustic propagation. Based on the chirp sonar imagery, various plausible models of sediment layering and material composition are proposed and the transmission loss is estimated for those models for the purpose of assisting field experiment planning.