bluuebunny/crossref_metadata_2025 huggingface dataset:
Why does humming clean your maxillar sinuses of NO gas?
Authors: Sundberg Johan , Granqvist Svante , Weitzberg Eddie , Lundberg Jon
Published: September 2013

Recent measurements have shown that the nitric oxide (NO) produced in the maxillar sinuses can be evacuated by producing nasal murmur; the NO content of the nasal airflow increases under these conditions (Maniscalco et al., 2003). Experiments were carried out to test the hypothesis that this effect is caused by resonance in the nasal tract and sinus cavities. A model was constructed where an airflow from a pressure tank was modulated at different frequencies. This airstream was passed through a tube with a radial hole constituting the neck of a Helmholtz resonator with a gas containing NO. The NO content of the air streaming out of the tube was measured. This NO content varied when the location of the resonator, its air volume, or the modulation frequency of the airflow was changed. The relevance of these three factors was also tested by means of a computer model of the system. The significance of the modulation frequency was also analyzed in human subjects by injecting a modulated airflow through one nostril and measuring NO content of the airflow exiting the other nostril, the results showing a dependence on the modulation frequency that differed considerably between subjects [Maniscalco et al., ‘‘Assessment of nasal and sinus nitric oxide output using single-breath humming exhalations,’’ Eur. Respir. J. 22, 323–329 (2003)].
An application of boundary element method calculations to hearing aid systems: The influence of the human head
Authors: Rasmussen Karsten B. , Juhl Peter
Published: September 2013

Boundary element method (BEM) calculations are used for the purpose of predicting the acoustic influence of the human head in two cases. In the first case the sound source is the mouth and in the second case the sound is plane waves arriving from different directions in the horizontal plane. In both cases the sound field is studied in relation to two positions above the right ear being representative of hearing aid microphone positions. Both cases are relevant for hearing aid development. The calculations are based upon a direct BEM implementation in Matlab. The meshing is based on the original geometrical data files describing the B&K Head and Torso Simulator 4128 combined with a 3D scan of the pinna.
Time reversal interactive objects
Authors: Ing Ros Ki , Quieffin Nicolas , Catheline Stefan , Fink Mathias
Published: September 2013

Time reversal has shown to be a fruitful concept in nondestructive testing in underwater acoustic or in ultrasonic imaging. In this paper this technique is adapted in the audible range to transform every day objects into tactile sensitive interfaces. A quick historical background is presented in the ultrasonic field and specially in chaotic cavity. In all time reversal experiments, it is demonstrated that a wave field spatially and temporally recorded is able to back propagate to its source. In other words, the field contains all the information on the location of the source. In the interactive experiments, it is shown that touching an object like a window, a table or a world globe generates an acoustic field easily detectable with one or two acoustic sensors. Using the concept of time reversal, the source location is deduced in real time. Then, touching objects at specific locations (virtual switches) is used to activate devices. Such devices are for example lights, stereo volume, or computer software. From a technical point of view, all these interactive experiments just use some computation easily performed with a standard personnel computer.
Movement amplitude and tempo change in piano performance
Authors: Palmer Caroline , Dalla Bella Simone
Published: September 2013

Music performance places stringent temporal and cognitive demands on individuals that should yield large speed/accuracy tradeoffs. Skilled piano performance, however, shows consistently high accuracy across a wide variety of rates. Movement amplitude may affect the speed/accuracy tradeoff, so that high accuracy can be obtained even at very fast tempi. The contribution of movement amplitude changes in rate (tempo) is investigated with motion capture. Cameras recorded pianists with passive markers on hands and fingers, who performed on an electronic (MIDI) keyboard. Pianists performed short melodies at faster and faster tempi until they made errors (altering the speed/accuracy function). Variability of finger movements in the three motion planes indicated most change in the plane perpendicular to the keyboard across tempi. Surprisingly, peak amplitudes of motion before striking the keys increased as tempo increased. Increased movement amplitudes at faster rates may reduce or compensate for speed/accuracy tradeoffs. [Work supported by Canada Research Chairs program, HIMH R01 45764.]
Flute ‘‘breath support’’ perception and its acoustical correlates
Authors: Cossette Isabelle A. , Sabourin Patrick
Published: September 2013

Music educators and performers commonly refer to ‘‘breath support’’ in flute playing, yet the term ‘‘support’’ is neither well-defined nor consistently used. Different breathing strategies used by professional flautists who were instructed to play with and without support were previously identified by the authors. In the current study, 14 musical excerpts with and without support were recorded by five professional flautists. Eleven professional flautists listened to the recordings in a random order and ranked (1 to 6) how much of the following sound qualities they judged to be in each example: support, intonation, control and musical expressiveness. Answers to the test showed that musical expressiveness was associated more closely with the supported excerpts than the answers about support itself. The ratings for each sound quality were highly intercorrelated. Acoustical parameters were analyzed (frequency and centroid variation within each note) and compared with the results of the perception test in order to better understand how the acoustical and psychological variables were related. The acoustical analysis of the central part of the notes did not show evident correlation with the answers of the perception test. [Work funded by the Social Sciences and Humanities Research Council of Canada.]
Noise in Mexico City
Authors: Beristain Sergio
Published: September 2013

Mexico City is known to be the largest city in the world, inhabited by some 20 percent of the national population, so noise pollution is not strange to it, particularly in view of the fact that industry is not concentrated, but rather spread throughout the city. The international airport also lies within the city limits, in the midst of residential areas. The heavy traffic during rush hours in the morning and in the evening and the activities of the populace, together with special events, produce a noise problem that is difficult to assess and to solve. Nevertheless, with educational programs begun several years ago and noise campaigns planned for the near future, in addition to existing regulations, the problem is not completely out of control. This paper presents a discussion of the general noise problem and describes how authorities and institutions are dealing with it.
Numerical three-dimensional solution of broadband acoustic pulses through media with power-law attenuation
Authors: Wismer Margaret G.
Published: September 2013

Acoustic waves in tissues and weakly attenuative fluids often have an attenuation parameter, α, satisfying α=α0ωy, in which ω is the applied frequency and y is between 1 and 2. This power-law attenuation is not predicted by the classical thermoviscous wave equation, and recent research has led to a number of modified viscous wave equations in which the third term usually consists of a convolution operator or a fractional spatial or temporal derivative. These wave equations are obtained by taking into account the requisite wave velocity dispersion predicted by the attenuation in order for the signals to be causal. In this paper, acoustic waves undergoing power-law attenuation are modeled by a slight modification to the thermoviscous wave equation, in which the time derivative of the viscous term is replaced by a fractional time derivative. This new equation satisfies the power-law formulation for lossy waves. An explicit time-domain, finite-element formulation leads to a stable algorithm capable of simulating three-dimensional broadband acoustic pulses propagating through attenuative and dispersive media. Results are given for pulse propagation through layered lossy media, and it is shown how attenuation affects the transmission and reflection of broadband signals.
Application of time reversal acoustics focusing for nonlinear imaging ms
Authors: Sarvazyan Armen , Sutin Alexander
Published: September 2013

Time reversal acoustic (TRA) focusing of ultrasound appears to be an effective tool for nonlinear imaging in industrial and medical applications because of its ability to efficiently concentrate ultrasonic energy (close to diffraction limit) in heterogeneous media. In this study, we used two TRA systems to focus ultrasonic beams with different frequencies in coinciding focal points, thus causing the generation of ultrasonic waves with combination frequencies. Measurements of the intensity of these combination frequency waves provide information on the nonlinear parameter of medium in the focal region. Synchronized stirring of two TRA focused beams enables obtaining 3-D acoustic nonlinearity images of the object. Each of the TRA systems employed an aluminum resonator with piezotransducers glued to its facet. One of the free facets of each resonator was submerged into a water tank and served as a virtual phased array capable of ultrasound focusing and beam steering. To mimic a medium with spatially varying acoustical nonlinearity a simplest model such as a microbubble column in water was used. Microbubbles were generated by electrolysis of water using a needle electrode. An order of magnitude increase of the sum frequency component was observed when the ultrasound beams were focused in the area with bubbles.
Characterization of scattering object with the decomposition of the time-reversal operator: Theory and experiment
Authors: Minonzio Jean-Gabriel , Prada Claire , Chambers David H. , Clorennec Dominique , Fink Mathias
Published: September 2013

The decomposition of the time-reversal operator provides information on the scattering medium. It has been shown [Chambers, J. Acoust. Soc. Am. 109, (2001)] that a small spherical scatterer is in general associated with four eigenvalues and eigenvectors of the time-reversal operator. In this paper, the 2D problem of scattering by a cylinder measured by a linear array of transducers is considered. Experimental results are obtained for wires of different diameters and different materials. It is shown how the singular-value distribution and the singular vectors depend on the elastic wave speeds cL, cT, the density and the radius of each wire. These results offer a new perspective towards solution of the inverse problem by determining more than scattering contrast using conventional array processing like that used in medical ultrasonic imaging.
Localization of frequency-separated bands of noise in the median plane
Authors: Ng Gerald , Colburn H. Steven
Published: September 2013

Six subjects reported the perceived location of two sets of noise stimuli presented through speakers in the median-sagittal plane. Two sets of stimuli were generated by passing flat (1 to 16 kHz) Gaussian noise through multiple, equal-amplitude bandpass filters. The control stimulus included all frequencies between 1 and 16 kHz. The first set contained two to six equally spaced, 1/3-octave-wide noise bands. The second set contained five equally spaced noise bands that varied in common bandwidth. Loudspeakers in the medial-sagittal plane were located at six frontal locations (−30, −15, 0, 15, 30, 45, with 0 directly in front) and two rear locations (120, 180). The experiments were conducted in a sound-treated room with the speakers concealed from view. Stimuli were 200 ms in duration and presented from one speaker per trial. Subjects used a pen to mark perceived stimulus locations on preprinted coordinate diagrams. Despite noticeable intersubject differences, most subjects localized stimuli with five or six 1/3-octave-wide bands as well as control stimuli. Many subjects showed steady decreases in accuracy as the bandwidths of five bands were reduced. HRTFs were recorded for each listener, and several model algorithms are being evaluated. [Work supported by NIDCD: Grants R01 DC00100 and P30 DC04663.]
Improved BTE hearing-aid directivity using a directional microphone array
Authors: Jones Douglas L. , Lockwood Michael E. , Lansing Charissa R. , Feng Albert S.
Published: September 2013

Extraction of speech in noise is of great importance to hearing-impaired listeners. Directional microphones are incorporated in some hearing aids to improve noise rejection through increased directivity. A symmetric cardioid response is created either with a single directional microphone, or using beamforming with two omnidirectional microphones. The head-related transfer function (HRTF), however, introduces an asymmetry that cannot be exploited by a linear array of omnidirectional microphones. A new BTE array consisting of a gradient directional microphone with nulls in the front–back vertical plane and two omnidirectional microphones exploits the asymmetry of the HRTF to obtain almost 2 dB better directivity than the best cardioid. HRTFs measured on KEMAR with this array were transformed to the frequency domain, where directivity-maximizing coefficients in each band were derived. The Articulation-Index (AI) weighted directivity gain of this optimal three-element directional array was 6.4 dB greater than a single omnidirectional microphone on the BTE, whereas the directivity gain of the HRTF-optimized two-omni beamformer was 4.6 dB, and the optimal free-field cardioid placed on the head yielded 4.4 dB. [Work partially supported by NIH (NIDCD) under Grant No. 1 R01 DC005762-01A1.]
Effect of source location and listener location on ILD cues in a reverberant room
Authors: Ihlefeld Antje , Shinn-Cunningham Barbara G.
Published: September 2013

Short-term interaural level differences (ILDs) were analyzed for simulations of the signals that would reach a listener in a reverberant room. White noise was convolved with manikin head-related impulse responses measured in a classroom to simulate different locations of the source relative to the manikin and different manikin positions in the room. The ILDs of the signals were computed within each third-octave band over a relatively short time window to investigate how reliably ILD cues encode source laterality. Overall, the mean of the ILD magnitude increases with lateral angle and decreases with distance, as expected. Increasing reverberation decreases the mean ILD magnitude and increases the variance of the short-term ILD, so that the spatial information carried by ILD cues is degraded by reverberation. These results suggest that the mean ILD is not a reliable cue for determining source laterality in a reverberant room. However, by taking into account both the mean and variance, the distribution of high-frequency short-term ILDs provides some spatial information. This analysis suggests that, in order to use ILDs to judge source direction in reverberant space, listeners must accumulate information about how the short-term ILD varies over time. [Work supported by NIDCD and AFOSR.]
Temporal integration of interaural-time differences carried by trains of 500-Hz tone bursts
Authors: Akeroyd Michael A.
Published: September 2013

The just-noticeable difference for the interaural time difference (ITD) of a sound generally reduces as its duration is increased [e.g., T. Houtgast and R. Plomp, J. Acoust. Soc. Am. 44, 807–812 (1968); E. Hafter and R. H. Dye, J. Acoust. Soc. Am. 73, 644–651 (1983)]. The rate of this temporal integration is less than would be expected from a simple multiple-looks application of signal detection theory, as though one part of the sound is given greater weighting. The present experiment measured the contributions of the start, middle, or end of a sound to the detectability of its ITD. The stimulus was a 32-pip train of 500-Hz, 10-ms pips, diotic apart from some (2–16) target pips which carried the ITD to be detected and which were placed either at the start, middle, or end of the train. Integration rates were determined from psychometric functions measured using four normal-hearing listeners. The results showed that the rate was least for targets placed at the start or end but larger for targets at the middle. The data can be described using a weighted multiple-looks approach, based on an integration function of approximately 100-ms time constant together with an emphasis of the start, and end, of the sound.
Comparing spectral resolvability in chinchillas and human listeners using phase discrimination
Authors: Shofner William P. , Sparks Kathryn , Wu Yuanxing Esther , Pham Ellen
Published: September 2013

A tone complex made of harmonic components that are added in cosine-starting phase can be discriminated from complexes comprised of identical harmonics that are added with random-starting phases. Phase discrimination occurs when unresolved harmonics interact within a single auditory channel. When harmonics are resolved, there is less interaction among components resulting in poorer phase discrimination performance. Thus, phase discrimination indirectly reflects spectral resolvability. Performance in a phase discrimination task was measured in chinchillas and human listeners to compare spectral resolvability between the two groups. Subjects discriminated a cosine-phase tone complex from random-phase tone complexes in a go/no-go behavioral paradigm. Tone complexes were comprised of a 250-Hz fundamental frequency and N consecutive higher harmonics, where N was 5, 10, 20, and 40. Performance was evaluated in terms of d’. The results show that the measured d’ increased as N increased, and values of d’ for each N condition were similar between chinchillas and human listeners. Values of the criterion for each N condition were also similar between chinchillas and humans. The results do not support the hypothesis that spectral resolvability is poorer in chinchillas, but suggest that resolvability is similar between the two groups. [Work supported by NIH/NIDCD.]
Mechanisms of forward masking
Authors: Wojtczak Magdalena , Viemeister Neal F.
Published: September 2013

Experiments involving forward masking were run with the aim of elucidating the mechanism underlying this phenomenon. Two possible mechanisms were considered: adaptation following masker offset, and persistence of excitation. In one experiment, detection of a 10-ms noise probe following a 150-ms noise masker was measured as a function of the level of a 20-ms burst of noise temporally centered on the probe. Model predictions were generated assuming either a reduction of gain applied to the probe (adaptation) or additivity of persisting masker excitation and excitation produced by the probe. Both models accounted for the data equally well. In the second experiment, listeners performed an image-centering task involving binaural presentation of a 4-kHz 10-ms probe. In one ear, the probe was preceded by a 150-ms noise forward masker. The level of the probe in the unmasked ear was adjusted until the image of the probe was centered. In general, the image was centered when the probe level in the unmasked ear was much lower (by 10–20 dB) than that in the masked ear. This result could be explained in terms of adaptation underlying forward masking but not in terms of persistence of excitation. [Work supported by NIH/NIDCD DC00683.]
Development of a topic-related sentence corpus for speech perception research
Authors: Helfer Karen S. , Freyman Richard L.
Published: September 2013

A large sentence corpus has been developed for use in speech recognition research. Sentences (n=881, three scoring words per sentence) were developed under 23 topics. In the first phase of development subjects rated each individual scoring word for relatedness to its given topic on a Likert scale. Next, two groups of young, normal-hearing listeners (n=16/group) listened and responded to the recordings of the sentences (spoken by a female talker) presented with one of two types of maskers: steady-state noise (S:N=−13 dB) or two other females speaking random sentences (S:N=−8 dB). Each subject responded to half of the sentences with topic supplied and half with no topic supplied. Data analyses focused on addressing two questions: whether supplementation of topic would be more important in the presence of the speech masker versus the noise masker, and how the degree of relatedness of each key word to the topic influenced the effect of topic on recognition. The data showed little difference in how beneficial the topic was for speech versus noise maskers. Moreover, there was a complex relationship between effect of topic, type of masker, and position of the word in the sentence. [Work supported by NIDCD DC01625.]
Uncertainty and confusion in temporal masking
Authors: Formby C. , Zhang T.
Published: September 2013

In a landmark study, Wright et al. [Nature 387, 176–178 (1997)] reported an apparent backward-masking deficit in language-impaired children. Subsequently, these controversial results have been influential in guiding treatments for childhood language problems. In this study we revisited Wright et al.’s temporal-masking paradigm to evaluate listener uncertainty effects. Masked detection was measured for 20-ms sinusoids (480, 1000, or 1680 Hz) presented at temporal positions before, during, or after a gated narrowband (W=600–1400 Hz) masker. Listener uncertainty was investigated by cueing various stimulus temporal properties with a 6000-Hz sinusoid presented either ipsi- or contra-lateral to the test ear or bilaterally. The primary cueing effect was measured in the backward-masking condition for a contralateral cue gated simultaneously with the on-frequency 1000-Hz signal. The resulting cued masked-detection threshold was reduced to quiet threshold. No significant cueing effects were obtained for other signal temporal positions in the masker nor for any off-frequency signal conditions. These results indicate that (1) uncertainty can be reduced or eliminated for on-frequency backward masking by cueing the signal and (2) the deficit reported by Wright et al. for language-impaired children may reflect uncertainty and confusion rather than a temporal-processing deficit per se. [Research supported by NIDCD.]
Auditory discrimination of binary sequences
Authors: Sheft Stanley , Yost William A.
Published: September 2013

Auditory sequence discrimination was investigated for stimuli generated with binary modulation coding schemes. Both amplitude- and frequency-shift keying (ASK and FSK) were used to convert 16-event binary patterns into a series of pulses. Discrimination was measured as a function of the number of binary events that differed between the standard and comparison sequences with temporal locations of the deviant pulses randomly selected on each trial. Across conditions, pulse duration ranged from 4 to 64 ms. In the ASK scheme, the carrier was wideband noise with off-key amplitudes scaled by a factor ranging from 0.0 to 0.5. Performance improved with increasing either pulse duration, amplitude shift, or the number of deviant pulses from one to eight. These trends are consistent with envelope processing by a modulation filterbank which discards phase information above low modulation rates. In the FSK scheme, binary patterns were coded by pulse-frequency differences centered at 1.25 kHz. As with ASK, FSK results showed performance improving with pulse duration and number of deviant pulses. FSK findings suggest that the low-pass results obtained with ASK envelope coding may reflect a general characteristic of sequence discrimination that is not restricted solely by auditory envelope processing. [Work supported by NIDCD.]
Perceptually related analysis of time-frequency patterns via a hearing model (Sottek), a pattern-measurement algorithm (‘‘relative approach’’) and a window-deconvolution algorithm
Authors: Bray Wade
Published: September 2013

The presence of patterns (time structure and/or tonal structure) in sound situations elicits a different analytic process by the human hearing system than when sound situations are without pattern. In the presence of pattern, two issues can interfere with the effectiveness of conventional measurement methods: the human hearing is not acting as an absolute magnitude measurer so conventional measurements may not, or only poorly, resolve the important criteria; and pattern information that is subjectively highly significant is frequently of low magnitude amid general high-level structure that is not a direct part of the subjective pattern-response problem yet carries significance. Several automotive sound quality and information technology time-data recordings involving various patterns, of different strengths, were analyzed using conventional, psychoacoustic, and specialized pattern-sensitive techniques. In general, the advanced perception-related methods proved superior in quantifying ‘‘patterned’’ sound situations according to subjective impression.
Acoustic cues in the perception of second language speech sounds
Authors: Bogacka Anna A.
Published: September 2013

The experiment examined to what acoustic cues Polish learners of English pay attention when distinguishing between English high vowels. Predictions concerned the influence of Polish vowel system (no duration differences and only one vowel in the high back vowel region), salience of duration cues and L1 orthography. Thirty-seven Polish subjects and a control group of English native speakers identified stimuli from heed-hid and who’d-hood continua varying in spectral and duration steps. Identification scores by spectral and duration steps, and F1/F2 plots of identifications, were given as well as fundamental frequency variation comments. English subjects strongly relied on spectral cues (typical categorical perception) and almost did not react to temporal cues. Polish subjects relied strongly on temporal cues for both continua, but showed a reversed pattern of identification of who’d-hood contrast. Their reliance on spectral cues was weak and had a reversed pattern for heed-hid contrast. The results were interpreted with reference to the speech learning model [Flege (1995)], perceptual assimilation model [Best (1995)] and ontogeny phylogeny model [Major (2001)].
Perceptual effects of preceding non-speech-rate information on temporal properties of speech categories
Authors: Wade Travis , Holt Lori
Published: September 2013

The purpose of the study was to determine whether a general mechanism such as durational contrast can explain the apparent rate-dependent nature of speech perception. Four experiments were performed to test effects of pure tone presentation rate on perception of following speech continua involving duration-varying formant transitions with which the tones shared critical temporal and spectral properties. Results showed small but consistent shifts in the stop-continuant boundary distinguishing /ba/ and /wa/ syllables based on the rate of precursor tones similar in duration and frequency to syllable-initial F1 and F2 transitions, across differences in amplitude of tones and despite variability in their duration. Additionally, the shift was shown to involve the entire graded structure of the [w] category and was not limited to an ambiguous boundary region, affecting goodness judgments on both sides of an estimated best exemplar range. These results are problematic for accounts of rate-dependent processing that explicitly reference speech categories or articulatory events and are consistent with a contrast account.
Spectral analysis of glottal flow models
Authors: Lee Matthew E. , Smith Mark J. T.
Published: September 2013

An analysis of the effects of time-domain characteristics of glottal waveforms in the spectral domain is presented. Studies have shown that the relative difference in amplitude of the first two harmonics of the inverse-filtered voice waveform (H1*−H2*) can be used to predict the value of the open quotient in certain situations [J. Sundberg, M. Andersson, and C. Hultqvist, J. Acoust. Soc. Am. 105, 1965–1971 (1999)]. However, recent experiments have shown that glottal waveform asymmetry is an additional factor that will affect the relationship between the harmonics and open quotient. This work is aimed at establishing a method in which both the open quotient and a glottal asymmetry coefficient can be reliably determined based on the characteristics of the first two harmonics. By deriving analytic formulas for the spectra of two glottal flow models (LF, R++), it is shown that a measure of the relative phase difference of the first two harmonics as well as the amplitude difference can be used in order to determine these perceptually important time-domain parameters of the glottal waveform. A theoretical analysis as well as experimental results are presented.
Determining the relationship between respiratory variables and fundamental frequency declination: A mathematical modeling of the data
Authors: Gelfer Carole E. , Jorgenson Jay
Published: September 2013

Previous research [C. E. Gelfer, unpublished doctoral dissertation, City University of New York Graduate Center (1987)] demonstrated stability in fundamental frequency (F0) declination across uttererance of various lengths that appeared to be derived from a stable underlying subglottal pressure (Ps). Moreover, this stability was maintained even when airflow requirements were varied in response to the varying phonetic composition of utterances. However, although a preliminary mathematical analysis suggested that both the Ps and F0 could be characterized as second order linear systems, no formal testing of these curves was performed. In this study we compare these data employing the following statistical procedure. First, we hypothesize that the data from both sets can be modeled by a curve of the form y=axbecx, with unknown parameters a, b, and c which are computed. We then consider a multivariate confidence interval for the vector (a,b,c) of fitted parameters, as well as the difference of the vectors for two such models. The comparison is made through distributional estimates obtained via Monte Carlo simulation.
Psychoacoustics based gain compensation for low listening levels
Authors: Einarsson Thorvaldur , Espy-Wilson Carol
Published: September 2013

Although the human hearing system is very complex and only understood to a limited degree, many measurements and models exist that explain parts of the hearing system. This paper uses one of these models, the contours of equal loudness, along with DSP techniques, to make music played at low listening levels sound more like it does at the intended listening level. The perceived frequency balance of music varies with listening level. This is especially noticeable at low listening levels, where frequencies below 500 Hz seem attenuated. Moreover, hearing perception exhibits nonlinear dynamic range compression, most evident at low frequencies. Traditional methods add fixed low-frequency gain to compensate for perceptual low-frequency attenuation at low levels. These methods do not consider the dynamic range compression and its nonlinearity and are often characterized by a boomy, unnatural sound. A system is designed where filter banks and power measurements estimate the time-varying power of low-frequency parts of the audio signal. The time-varying power of each narrow frequency band is compared to the contours of equal loudness, and changes are made to get the same frequency balance as at the intended listening level. The paper covers the design, implementation and performance of this system.
Implementation of algorithms for extracting tonal components in underwater noise
Authors: Supriya M. H. , Pillai P. R. Saseendran
Published: September 2013

Underwater target classification and tracking problems utilize the noise signals emanating from the targets as well as the target dynamics features. Target classification is carried out by extracting certain classification clues about the target such as its emission frequencies, and other target specific features from the self noise or active transmissions. The self noise generated by the targets being nonstationary in nature are to be analyzed with short term averaged data segments for extracting the tonal components using various spectral estimation techniques. Though classical spectral analysis techniques are computationally efficient, it suffers from several inherent limitations such as frequency resolution, performance degradation due to implicit windowing of the data, etc. In an attempt to improve the spectral resolution, several modern spectral estimation techniques, which utilize the procedures of indirect Fourier analysis by fitting the measured short data segments to an assumed model, have been evolved. This paper presents the development of procedures for the estimation of power spectral densities using modern techniques based on parameter estimation such as auto regressive, moving average, auto regressive moving average, maximum likelihood, etc. The performance of the estimator has been validated by computing the emission frequencies of a 50-foot vessel.
Exploiting autonomous underwater vehicle mobility for enhanced sonar performance
Authors: Edwards Joseph R. , Schmidt Henrik
Published: September 2013

Autonomous underwater vehicles (AUVs) provide a mobile sonar platform for local environmental characterization and mine hunting missions in areas that are prohibitively dangerous or otherwise inaccessible to manned vessels. Such missions are typically implemented by providing the vehicle with a series of way-points that uniformly sample the region of interest. While these preplanned sampling paths can be effective for many purposes, a higher degree of efficiency and efficacy can be achieved when the AUV is able to react to its perceived environment. Such response to the sensory stimulus represents a preliminary step toward the AUV achieving tasks in the way that a dolphin or human might. Upon sensing an object or feature of interest, the AUV can further interrogate the object by obtaining multiple views from preferred vantage points. Probabilities of detection and correct classification can be greatly impacted by a wise choice of sonar-adaptive AUV behavior. In this paper, the ability of the AUV to improve sonar performance through sonar-adaptive mission planning is investigated in the context of the mine hunting problem. Trade-offs are discussed for continuous versus scripted mission adaptations, single-minded versus multiple-objective mission control, and task-wise versus mission-wise behavior. [Work supported by ONR and SACLANTCEN.]
Continuous wide area monitoring of fish shoaling behavior with acoustic waveguide sensing and bioclutter implications
Authors: Makris Nicholas C. , Ratilal Purnima , Symonds Deanelle T. , Nero Redwood W.
Published: September 2013

Field measurements are used to show that the detailed behavior of fish shoals can be continuously monitored at roughly 1-min intervals over wide areas spanning hundreds of square kilometers by long range acoustic waveguide sensing. The technique was used on the New Jersey Continental Shelf to produce unprecedented video images of shoal formation, fragmentation, and migration. Simultaneous line-transect measurements show the imaged shoals to contain pelagic fish with densities of at least one individual per meter3. The technique relies upon acoustic waveguide propagation in the continental shelf. Here, trapped modes dominate propagation and suffer only cylindrical spreading loss rather than the spherical loss suffered in free-space transmission or short-range propagation in the ocean. In contrast, standard methods for fish surveyance involve line transect measurements from slow moving research vessels that significantly under-sample fish distributions in time and space, leaving an incomplete behavioral picture. The implications of this bioclutter phenomenon on the Navy’s long range active sonar operations in continental shelf environments are discussed.
Acoustic communications and autonomous underwater vehicles
Authors: Freitag Lee , Grund Matthew , Preisig James , Stojanovic Milica
Published: September 2013

Acoustic communications systems used on autonomous underwater vehicles (AUVs) provide supervisory control, access to real-time data, and also allow multiple vehicles to cooperate in undertaking adaptive sampling missions. However, the use of acoustic systems on AUVs presents special challenges because of limited space for optimal placement of transducers, and potential conflicts with other acoustic systems such as side-scan sonars and transponders. In addition, radiated and structure-borne acoustic interference from thrusters and actuators reduces the sensitivity of on-board receivers. Recent work in acoustic communications and AUVs has included combining some navigation functions into communications equipment, development of operating modes that remove conflicts between different subsystems, design of vehicle components to avoid or remove interference, and other approaches to improving performance. While these efforts have been successful for specific installations, many challenges remain. This talk addresses problems and solutions for supervised and completely autonomous multi-vehicle communications to support complex AUV missions. Also presented are recent results which demonstrate that acoustic communications can be used successfully on a variety of AUV platforms for many different applications. [Work supported by ONR.]
Experimental modeling of the acoustic signature of an AUV
Authors: Cuschieri Joseph M.
Published: September 2013

In this presentation test tank results from an experimental analysis of the acoustic signature of an Ocean Explorer class AUV are presented. The results are from measurements performed in a reverberant test tank on an AUV model, and of an AUV under typical operating conditions. The main source of excitation is the propulsion module or ‘‘podule.’’ Inside the podule are the propulsion motor and the motors for the control surfaces with penetrations for the main propulsion shaft and the shafts for the control surfaces. Different operating conditions and different mountings for the main propulsion and control module (podule) inside the AUV are considered. The influence of the propeller, the mounting of the podule, and covering the podule with a compliant layer are considered in the measurements. It is shown that for the type of AUV considered here, the mounting of the podule is not very significant and that significant energy is transferred through the water trapped between the podule and the AUV hull. Furthermore, the propeller has a significant influence on the acoustic signature. [Work supported by ONR.]
24 hours of noise in a large city, problems and solutions
Authors: Blomberg Leslie
Published: September 2013

This paper begins by comparing a contemporary urban noise study to studies done in the 1970s. Several changes in the urban soundscape are noted, as well as areas in which progress has and has not been made in quieting urban soundscapes. Finally, the challenges to quieting urban areas are presented and potential solutions are identified.
Study of the daily environmental urban noise levels in San Juan, Puerto Rico
Authors: Alicea-Pou José A. , Viñas-Curiel Olga , Cruz-Vizcarrondo Wanda , Hernandez-Dávila Daniel , Rocaford Jorge
Published: September 2013

The study of the urban environmental noise levels in the city of San Juan is one of the first steps into understanding the noise behavior of the largest and denser populated zones in Puerto Rico. For the year 2002 San Juan had 36% of all citizen noise complaints reported for the entire island. The aim of the study was to monitor for 24 continuous hours the levels of urban sounds and noise for different locations in the city and describe the general tendencies regarding those levels for residential, commercial, recreational or industrial areas. The sampling was done with four (4) units of the Norsonic NOR-121 sound analyzer that were installed temporarily for 24 hours in multiple locations chosen randomly around the city. The study is expected to continue during the next 2 years. The preliminary data showed that the noise levels in residential areas with a high traffic routine fluctuated in levels that were consistently high all day and most of the night, dropping around 1:00 a.m. and rising again around 5:00 a.m. The results of this and other projects would be used to build and implement a National Noise Plan for Puerto Rico for 2005–2007.
Frequency dependent P-wave speed in a porous medium with aligned fractures
Authors: Brajanovski Miroslav , Gurevitch Boris , Schoenberg Michael
Published: September 2013

In an elastic medium, fractures can be modeled as thin layers, the elastic stiffnesses of which approach zero as the volume fraction of the fractures hf→0. This yields linear slip theory [M. Schoenberg, J. Acoust. Soc. Am. 68, 1516–1521 (1980)], shown to be a robust way to account for the acoustic effect of fracturing. From Norris’ [J. Acoust. Soc. Am. 94, 359–370 (1993)] dispersion relation for alternating porous layers, fractures must be modeled similarly in porous media. However, another fracture parameter of great interest is permeability. If fracture permeability is taken to be O(hf−1) or taken to be independent of hf, one arrives at a dispersion relation for the fast P-wave dependent on porous background properties and a real excess compliance which takes the fractures into account. However, if fracture permeability is assumed to be small, and is taken to be O(hf), the P-wave dispersion dependence on background parameters remains the same, but the term accounting for the fractures is frequency dependent. Only in the zero frequency limit does this result agree with that of the other two cases.
Acoustic correlates of the question–statement contrast in children
Authors: Patel Rupal , Syeda Mariam , Grigos Maria
Published: September 2013

The acoustics of prosodic control in children was studied in 4-, 7-, and 11-year olds using the question–statement contrast. Each child produced the utterances ‘‘Show Bob a bot’’ (voiced consonants) and ‘‘Show Pop a pot’’ (voiceless consonants) ten times each as a question and ten times each as a statement. A total of 40 utterances were analyzed per child. The following acoustic measures were obtained for each word within each utterance: average fundamental frequency (f0), peak f0, slope of f0, average intensity, peak intensity and duration. Preliminary results indicate no significant difference between questions and statements for 4-year olds in both the voiced and voiceless consonant conditions. In contrast, 7- and 11-year olds differentiated questions from statements by increasing average, peak, and slope of f0 as well as increasing the duration of the final syllable. Changes in syllable duration between questions and statements were more pronounced for the utterance with voiced consonants. These findings suggest that the acoustics of prosodic contrasts begins to differentiate somewhere between ages 4 and 7 and is influenced by developmental changes in physiological control and flexibility which may also affect segmental features.
Fundamental frequency and intensity variability in young children’s speech produced at comfortable effort level
Authors: Brown William S. , Shrivastav Rahul
Published: September 2013

Experiments to study speech production and clinical evaluation of speech often require that participants phonate or produce speech at a comfortable level of effort. It is necessary to know the degree of variability in fundamental frequency (F0) and intensity (SPL) across multiple test-sessions when speech is elicited in such a manner. If the intersession variability in these measures is too large, the experimental task may need to be appropriately modified. Although the variability in F0 and SPL produced at comfortable effort level across multiple sessions has been reported for normal adult speakers, no such data is available for young children. A vowel, a sentence, and four words were produced at a comfortable level by 15 males and 14 females between the ages of 3 and 4 years, on three separate days, one week apart. The F0 and SPL for these tasks were compared across the three test sessions. Results show that there were no significant differences in F0 and SPL and suggest that normal-speaking children between the ages of 3 and 4 years show the same degree of variability in their F0 and SPL as adults when asked to set their own level of comfort.
Acoustic characterization of developmental speech disorders
Authors: Bunnell H. Timothy , Polikoff James , McNicholas Jane , Walter Rhonda , Winn Matthew
Published: September 2013

A novel approach to classifying children with developmental speech delays (DSD) involving /r/ was developed. The approach first derives an acoustic classification of /r/ tokens based on their forced Viterbi alignment to a five-state hidden Markov model (HMM) of normally articulated /r/. Children with DSD are then classified in terms of the proportion of their /r/ productions that fall into each broad acoustic class. This approach was evaluated using 953 examples of /r/ as produced by 18 DSD children and an approximately equal number of /r/ tokens produced by a much larger number of normally articulating children. The acoustic classification identified three broad categories of /r/ that differed substantially in how they aligned to the normal speech /r/ HMM. Additionally, these categories tended to partition tokens uttered by DSD children from those uttered by normally articulating children. Similarities among the DSD children and average normal child measured in terms of the proportion of their /r/ productions that fell into each of the three broad acoustic categories were used to perform a hierarchical clustering. This clustering revealed groupings of DSD children who tended to approach /r/ production in one of several acoustically distinct manners.
Learning to talk
Authors: Messum Piers
Published: September 2013

Is imitation a necessary part of learning to talk? The faithful replication by children of such arbitrary phenomena of English as tense and lax vowel properties, ‘‘rhythm,’’ and context-dependent VOT’s seems to insist that it is. But a nonimitative account of this is also possible. It relies on two principal mechanisms. First, basic speech sounds are learned by emulation: attempting to reproduce the results achieved by other speakers but without copying their actions to do so. The effectiveness of the output provides sufficient feedback to inform the child of the adequacy of its performance and to guide refinement. Second, phonetic phenomena such as those above appear through aerodynamic accommodation. Key elements of this are (a) that speech breathing is a complex motor skill which dominates other articulatory processes during acquisition and starts pulsatile before becoming smooth, and (b) that a child-scale production system imposes constraints on talking which do not operate in the adult speaker. Much of ‘‘the terrible complexity of phonetic patterns’’ [J. Pierrehumbert, Lang. Speech 46, 115–154 (2003)] then becomes epiphenomenal: appearing not as a result of young learners copying phonetic detail that is not linguistically significant, but of them reconciling conflicting production demands while just talking.