bluuebunny/crossref_metadata_2025 huggingface dataset:
A psychophysical measure of level-dependent shifts in the peak of the traveling wave
Authors: Brown Christopher A. , Jeung Changmo , Bacon Sid P.
Published: October 2013

Thresholds for detecting a 50-ms pure-tone probe in the presence of a 200-ms pure-tone forward masker were measured as a function of masker level (50, 70, or 90 dB SPL). The masker frequency was 250, 1000, 4000, or 6000 Hz, and the probe frequency was 1.00, 1.10, 1.20, 1.30, 1.40, or 1.60 times the masker frequency. For subjects 1–3, a 50-ms delay was used. For subjects 4–6, subject-dependent delays were chosen to keep the amount of masking to about 10 dB or less, to restrict the influence of off-frequency listening. These delays ranged from 50 to 110 ms. Subjects 1–3 showed the greatest amounts of masking on frequency in almost all conditions, whereas subjects 4–6 tended to show the most masking at frequencies above the masker frequency, but only at the highest masker levels and only at masker frequencies above 250 Hz. Although inconclusive, these results are in general agreement with both psychophysical and physiological studies that suggest a level-dependent basalward shift in the peak of the traveling wave along the basilar membrane. [Work supported by NIDCD.]
Theoretical models of non-native speech perception: Implications for perception of foreign-accented speech
Authors: Best Catherine
Published: October 2013

Linguistic experience systematically affects speech perception, specifically for non-native phonemes and contrasts. Native language influences are strong in adults, and emerge in infancy. Several theoretical models of non-native speech perception have been proposed, including the Perceptual Assimilation Model (PAM: Best), Speech Learning Model (LSM: Flege), and Native Language Magnet model (NLM: Kuhl). Findings that infants learn statistical distributions of phoneme sequences, and phonetic details, converge on the notion that exposure to specific phonetic patterns affects the perception of phonological structure. A reasonable extrapolation is that perception of even native utterances should be influenced by their pronunciation with either familiar or unfamiliar phonetic structure. That is, if a listener is inexperienced with a given foreign accent, or another native language dialect, perception should be predictably affected when native phonemes/words are produced with those accents. This issue has received modest research attention, which has focused on sentences/words rather than phonemes. Findings suggest that unfamiliar accents do reduce recognition and intelligibility of native sentences and words. The implications of non-native speech models, particularly PAM and SLM, for understanding and studying the impact of foreign accent on perception of native phonemes and contrasts, as well as longer utterances, will be discussed. [Work supported by NIH.]
An introduction to time-delay estimation, localization, and tracking in Acoustics
Authors: Sibul Leon H.
Published: October 2013

Acoustic source localization and tracking has a long, productive history in both passive and active sonar signal processing. More recently, microphone arrays have been used to localize and track air acoustic noise sources. An overview of basic localization and tracking techniques of acoustic radiating sources and scatters is presented. For a wide range of applications, many ad-hoc passive and active source localization techniques have been investigated. Concepts from the statistical estimation theory, such as maximum likelihood estimators and Cramér-Rao lower bounds (CRLB), are useful benchmarks for evaluation of these practical ad hoc systems. In both air and underwater media, bearing and range estimates are determined by time-delay estimates; hence, properties of the time-delay estimates are of critical importance. In the case the of narrow-band active sonar, localization and tracking is based on time delay, Doppler, and direction of arrival estimation. In the case of wideband sonar and tracking of high-speed air acoustic sources, tracking and range estimates are determined from time-scale dilation estimates. CRLBs for active sonar are determined by signal-to-noise ratio, signal properties of signal ambiguity functions such as effective duration and mean-square bandwidth of the transmitted signal, and array parameters. [This work has been supported by ONR.]
Passive detection, data association, and localization of marine mammal calls using traditional 3D hyperbolic tracking algorithms
Authors: Moretti David , Jarvis Susan , DiMarzio Nancy , Morrissey Ron
Published: October 2013

The Office of Naval Research Marine Mammal Monitoring on Navy Ranges (M3R) project has developed a toolset for passive detection and localization of marine mammals using the existing infrastructure of Navy undersea ranges. The tools are designed to work across a broad class of calls including clicks, sweeps, and whistles. Using vocalizations received by widely spaced omni-directional hydrophones, the M3R tools can detect and localize marine mammals in real-time. Signal detection algorithms, data association algorithm, and tracking algorithm have been developed, and demonstrated. The tools have also been successfully fielded to conduct extended monitoring of a 500 square nautical mile area located in the Tongue of the Ocean. This presentation will present the passive acoustic monitoring and localization tools developed under M3R. It will also present results of the application of these tools to the real-time detection and tracking as well as extended monitoring of various toothed whale species at the Atlantic Undersea Test and Evaluation Center (AUTEC), Andros Island, Bahamas.
Experimental measurements of normal-mode statistics at megameter ranges
Authors: Wage Kathleen
Published: October 2013

The Acoustic Thermometry of Ocean Climate, Alternate Source Test, and North Pacific Acoustic Laboratory experiments generated extensive vertical array data sets for studying normal-mode signals at megameter ranges. These data sets include receptions for 28-, 75-, and 84-Hz sources at ranges of 3.5 and 5 Mm. Recent analysis of the low-mode arrivals in these receptions produced estimates of important mode statistics, including the modal time spread, cross-mode coherence, and temporal coherence. This talk will present the experimental results and place them in context, making comparisons to theoretical predictions about long-range mode propagation and to numerical simulations. The implications for future research will be discussed. [Work sponsored by an ONR Ocean Acoustics Entry-Level Faculty Award.]
Mean field of a sound wave propagating in an oceanic waveguide with random inhomogeneities
Authors: Voronovich Alexander G. , Ostashev Vladimir E.
Published: October 2013

The mean field is an important statistical characteristic of a sound wave propagating in an oceanic waveguide with random inhomogeneities such as internal gravity waves. A closed equation for the mean sound field propagating in a 3D random waveguide can be obtained with the use of the diagram technique. Using the Fourier transform, this equation is reduced to a one-dimensional integro-differential equation with respect to the vertical coordinate. The latter equation determines acoustic modes which are generally different from those in a regular waveguide. For low frequency sound waves, the integro-differential equation is solved approximately. This results in an explicit formula for the mean sound field as a sum of modes which attenuate exponentially with the propagation distance. The extinction coefficient of each mode is linearly related to the spectrum of random inhomogeneities. Therefore, measuring the extinction coefficients for different frequencies and different mode numbers, one can try to reconstruct this spectrum. The mean field of a sound wave propagating in a 2D random waveguide is also calculated. The result obtained is compared with that for a 3D random waveguide. This allows us to study the range of applicability of 2D approximation.
Interference wave associated with a multitude of cusped caustics in long-range acoustic propagation in the ocean
Authors: Grigorieva Natalie S. , Fridman Gregory M.
Published: October 2013

In long-range propagation experiments, the observed time-of-arrival patterns show early geometrical-like arrivals followed by a crescendo of energy that propagates along the sound-channel axis and is not resolved into individual arrivals. In these experiments the sound source and receiver are placed close to the depth of the axis. As a result, rays emerging from the source at sufficiently small angles intersect the axis many times and form in its vicinity a large number of caustics with caustic cusps located repeatedly along the axis. It is well known that a description in terms of geometrical acoustics is not valid in neighborhoods of cusped caustics because such a description presupposes that the waves associated with individual ray paths do not interfere with one another. These neighborhoods of interference grow with range and at long ranges they overlap. The interference of the wave fields that correspond to near-axial rays results in the formation of a complex interference wave, or axial wave, that propagates along the waveguide axis being a component of the axial crescendo. In this paper for a range-independent ocean the axial wave formation is discussed as well as its main properties in the frequency and time domains. [Work supported by ONR.]
Speech privacy for healthcare
Authors: Lewitz Joel A.
Published: October 2013

The Healthcare Insurance Portability and Accountability Act, also known as HIPAA, requires healthcare providers to implement reasonable safeguards to protect the privacy of healthcare information. This includes privacy for oral communications. Architects and facility managers are aware of this mandate, and are calling for these capabilities to be designed into current projects. HIPAA contains no objective, measurable criteria for speech privacy, leaving the design team to select specific criteria for their project. Criteria options will be discussed including Privacy Index and other descriptors that have been developed and correlated with subjective evaluation of privacy. This paper describes several recent projects, which began with evaluations of existing facilities and led to establishment of criteria and design solutions. These studies included various aspects of the healthcare facility such as offices, exam rooms and conference rooms as well as open plan reception, waiting and registration areas. Field conditions, measurement techniques, measurement results, criteria selection and design solutions will be presented.
Vocalizations of blue whales in the Antarctic: Implications for the use of passive acoustics for population studies on baleen whales
Authors: Rankin Shannon , Ljungblad Don , Clark Chris , Kato Hidehiro
Published: October 2013

Stereotypical calls of baleen whales can be a diagnostic field characteristic for determining species and stocks. Repeated series of these calls may be used for long-term population studies using passive acoustics. Vocalizations were recorded from 15 groups of blue whales (consisting of 42 individuals) during the 2001–02 and 2002–03 International Whaling Commission-Southern Ocean Whale and Ecosystem Research (IWC-SOWER) surveys in Antarctic waters north and east of the Ross Sea. Long-duration vocalizations attributed to Antarctic blue whales included 28-Hz ‘‘tones,’’ 28 Hz ‘‘downsweeps,’’ and the more complex ‘‘3-unit’’ calls (considered to be song units). The 28-Hz component was stable for all three calls, regardless of signal strength. The 3-unit calls varied in their 2nd and 3rd units and were heard in only 10 out of the 15 groups. There was no evidence of repeated patterns (song phrases) of 3-unit calls. The results of this study suggest that the 28-Hz peak frequency may be used as a diagnostic feature for acoustic identification of Antarctic blue whales in the field. Additional research on the gender, group size, and associated behavior of these animals is necessary to ground-truth long-term acoustic population studies of baleen whales.
Further investigation of acousto-optical imaging in highly diffusive media using a photorefractive crystal based detection system: Sensitivity and contrast
Authors: Sui Lei , Bossy Emmanuel , Roy Ronald A. , Murray Todd W.
Published: October 2013

Acousto-optical imaging takes advantage of good optical contrast and high ultrasonic resolution to facilitate imaging and characterization of biological tissues. By detecting ultrasound modulated optical signals, the local opto-mechanical properties can be extracted. Among the state-of-the-art detection techniques, there is a trade-off between the axial resolution (or ultrasound bandwidth) and signal-to-noise ratio. Recently, a photorefractive crystal (PRC) based system was shown to enhance the detection of ultrasound-modulated optical signals [Sui et al., Proc. SPIE 5320, 164–171 (2004)]. This system achieved improved axial resolution by using short-duration acoustic pulses [Sui et al., J. Acoust. Soc. Am. 115, 2523 (2004)]. Previous experiments were done in tissue-mimicking phantoms whose optical diffusivities were less than that of most biological tissues. In this paper, the PRC-based detection system is applied to acousto-optical imaging in highly diffuse media and in biological tissues in vitro. Both transmission and right-angle detection geometries are studied. The relationship between the amplitude of the detected signals and acoustic driving pressure is determined. The ability of this system to detect both optical absorption contrast and optical scattering contrast is explored. [Work supported by the Center for Subsurface Sensing and Imaging Systems via NSF ERC Award No. EEC-9986821.]
Precision calibration of measurement microphones
Authors: Rasmussen Gunnar
Published: October 2013

The calibration of measurement microphones is often carried out as a rough check using a comparison between the measurement microphone and another microphone. A calibration with lower uncertainty involves either a complete reciprocity calibration which is hard to carry out in the field or the use of a pistonphone, where the calibration is related to well known physical quantities easy to measure. The largest contribution to the uncertainty is often the barometric pressure and the use of less accurate barometers. The use of a precision barometer built into the pistonphone greatly reduces the uncertainty due to human error and due to varying atmospheric conditions independent of altitude. Normally the ambient temperature at which the calibration takes place is also reported. Practical experience shows that it may be more relevant to report the actual temperature of the transducer under calibration. The final working temperature for a microphone is reached several minutes after the start of the calibration. It is therefore practical to measure and report the temperature of the actual calibration coupler and microphone combinations in order to minimize the uncertainty. Direct data transfer of all relevant data calibration level, barometric pressure, and temperature to a data system to ensure immunity to human error is also practical.
Reduction of airflow noise in telephone handsets
Authors: Stinson Michael R. , Daigle Gilles A. , Quaroni John F.
Published: October 2013

The bursts of airflow that accompany plosive sounds such as a ‘‘p’’ or ‘‘t’’ can cause significant noise in a telephone handset when the talker’s mouth is close to the mouthpiece of the handset. Measurements of the generated noise have been made on a series of modified handsets, using various arrangements of ports and tubes between microphone and the mouthpiece exterior. The use of one, two and more exterior holes, and the introduction of different geometrical coupling between these holes and the interior microphone were explored. The noise could be reduced by over 20 dB by using an array of small (less than 0.5 mm diameter) exterior holes coupled with an offset to a sound tube leading to the microphone. [Work supported in part by Mitel Networks.]
Additional modeling details for the suspension of a moving-coil loudspeaker
Authors: Leishman Timothy , Dix Gordon
Published: October 2013

The suspension of a typical moving-coil loudspeaker incorporates both a surround and a spider of dissimilar construction. Nevertheless, classical lumped-element models simplify this composite system by representing it with a single mechanical compliance and a single mechanical resistance. This paper presents a more detailed lumped-element model of the suspension to account for individual contributions of the surround and spider, their effective masses, and their primary resonances. It subsequently proposes an expanded equivalent circuit for a loudspeaker and discusses experimental methods to characterize new circuit elements.
Optimal configurations for active control of axial fans
Authors: Monson Brian B. , Sommerfeldt Scott D.
Published: October 2013

The minimum power radiated from a single monopole noise source surrounded by four symmetrically spaced control sources has been shown analytically. A similar analysis has also been completed for a monopole noise source surrounded by three symmetrically spaced control sources. At Brigham Young University, a multichannel active noise control system has been developed for a small axial cooling fan employing four symmetrically spaced miniature loudspeakers as control actuators. This arrangement of control loudspeakers has exhibited good global control of the fan’s tonal noise, achieving close to the predicted ideal for some harmonics of the blade passage frequency. According to theory, a three-source control configuration should be able to achieve nearly the same power reduction as a four-source configuration. While this is so, increasing the number of control sources to greater than four gives little to no increase in predicted power attenuation. A three-source control implementation, then, appears to be a comparable solution for the fan noise control problem. The experimental results of a three-source control and fan configuration will be discussed. An assessment will be made of the three- and four-source arrangements, including also a comparison of the experimental results versus the theoretical ideal.
Models of commodulation masking release
Authors: Berg Bruce G. , Kong Ying-Yee
Published: October 2013

Three models of commodulation masking release (CMR) are investigated in the context of a task in which masking and flanking bands each consist of five equal-intensity tones spanning 20 Hz. Two are multiple-channel models [J. Hall et al., J. Acoust. Soc. Am. 76, 50–56 (1984); S. Buus, J. Acoust. Soc. Am. 78, 1958–1965 (1985)] and one is a single-channel model [B. G. Berg, J. Acoust. Soc. Am. 100, 1013–1023 (1996)]. Thresholds do not provide an adequate test of the models because all three outperform listeners, thus reducing the solution to arbitrary assumptions about internal noise. However, the models can be discerned with a molecular psychophysical technique. A small intensity perturbation is added to each component of the stimulus. Correlating the trial-by-trial, simulated responses to the perturbations of each component yields a set of spectral weights. It is shown that the three models predict highly distinct patterns of spectral weights.
Skull simulator for design of hearing protection systems for bone-conducted sound
Authors: May B. Scott , Dietz Anthony J. , Clavier Odile , Bilski Steve
Published: October 2013

A skull simulator was developed for an investigation of bone-conducted sound transmission, which can be a significant source of sound in extreme noise environments. The simulator was built from a human skull, with silicon gel used to model internal organs and silicon and latex used to model the skin. Accelerometers attached to the skull bones were used to measure the skull vibration response. Mechanical point impedance measurements on the simulator were compared to results reported in the literature for humans, cadavers, and skulls. Reasonable agreement with these results served to validate the simulation. The link between skull vibration measured on the skull simulator and bone-conducted sound at the cochlea of a human subject was determined by comparing the response at the mastoid measured on the skull with similar measurements on a human subject. The response of the skull simulator in a sound field was then measured. The frequency response peaked from 1 to 3 kHz, which is where the bone-conduction limit to hearing protection attenuation is a minimum. The simulator was used to determine the effectiveness of various hearing protection components in reducing skull vibration. The results are compared to attenuation measurements of these components on human subjects.
Perception of Thai tones in citation form and connected speech
Authors: Zsiga Elizabeth C. , Nitisaroj Rattima
Published: October 2013

F0 contour shapes of the five tones of Thai differ from citation form to connected speech [Potisuk et al., Phonetica 54, 2242 (1997)]. ‘‘Falling’’ tones have a rise-fall contour in citation, but may be realized as a rise in fluent speech. This study investigates whether Thai listeners can reliably distinguish tones in connected speech, and whether any cues to tone identity remain stable across contexts. In four experiments, ten Thai listeners identified naturally produced and digitally altered tones in a forced-choice task. In experiment one, listeners identified naturally produced ‘‘falling’’ tones with 100% accuracy in citation forms and 96% accuracy in sentences, despite differing contours. In experiments 2–4 (replicating and extending Abramson [Lg&Sp 21, 319325 (1978)]), F0 onset and offset, peak height, and peak alignment were systematically modified on syllables in citation and sentence contexts. In all contexts, tones identified as high or low had an F0 peak or valley aligned to the right edge of the syllable. Mid tones had no F0 inflection. Tones identified as falling or rising necessarily had a pitch inflection at the syllable midpoint (end of the first mora). These findings support the view that tones are aligned with moras in Thai.
Developmental processes in production of Korean /s/ and /s*/
Authors: Chung Hyunjoo , Schmidt Anna M.
Published: October 2013

The purpose of this study was to investigate development of Korean children’s productions of tense and lax Korean s. Korean alveolar fricatives are acquired late; /s*/ is generally later than /s/. However, little is known about developmental processes in acquiring the contrast between tense and lax Korean alveolar fricatives. Twenty typically developing monolingual Korean children between 4 and 8 and ten monolingual Korean adults were recruited in Korea. All children fell within normal limits on both a Korean articulation test and a standardized Korean vocabulary test. A picture-naming picture task elicited productions of initial fricatives, alveolar stops, and affricates followed by /a, i, u/ in monosyllabic words. Production error patterns will be analyzed to describe general developmental trends in production. More importantly, the contrast between productions of /s/ and /s*/ will be presented based on acoustic parameters including frication durations, centroid frequency, and harmonic-related amplitude measurements. Finally, similarities and differences in each acoustic parameter will be compared between adults and children. Results will discuss which acoustic parameter(s) effectively describe the distinction between productions of /s/ and /s*/ as well as differences between productions of adults and children between age 4 and 8.
Diphthongization as cue for the automatic identification of British English dialects
Authors: Ferragne Emmanuel , Pellegrino Franois
Published: October 2013

The degree to which a given speaker of British English diphthongizes her/his vowels has been known for decades to be a good indicator of this speaker’s dialectal origin: for instance, some Northern varieties tend to have variants perceived as monophthongs where Standard Southern English has a perceived diphthong in the word GOAT. A portion of the recently released Accents of the British Isles database (14 dialects recorded throughout the British Isles, 20 speakers per dialect) is investigated to determine whether this view still holds at least acoustically for contemporary dialects. Several metrics (based on formant-frequency estimation) are proposed in an attempt to capture a ‘‘degree’’ of diphthongization. In a preliminary and rather oversimplified study, the standard deviation of F2 was measured over the whole duration of the vowel in the word BOAT for all speakers and dialects. A nonparametric one-way analysis of variance was performed to test the effect of ‘‘dialect’’ on diphthongization. The effect turned out to be significant at the 0.01 level. More elaborate procedures (including all first three formants) will be shown in the final presentation.
Effects of jaw perturbation on cortical activity during speech production
Authors: Tourville Jason A. , Guenther Frank H. , Ghosh Satrajit S. , Bohland Jason W.
Published: October 2013

Functional magnetic resonance imaging (fMRI) was used to investigate cortical activation caused by introducing a perturbation to the jaw during speech. The DIVA model of speech production [e.g., Guenther and Ghosh (2003), Proceedings of the ICPhS. Barcelona, ICPhS Committee] predicts that jaw perturbation will cause a mismatch between learned target (expected) sensations and actual sensory inputs in higher-order somatosensory cortical areas in the supramarginal gyrus, and that this mismatch will drive compensatory activations of motor cortical areas. To test this hypothesis, we performed an fMRI experiment in which articulation of pseudo-words was perturbed by the rapid inflation of a small, stiff balloon held between the subjects molars. The perturbation condition occurred pseudo-randomly on 1/7 of the trials. After controlling for brain activation due to normal speech production (no perturbation) and due to the sensation of balloon inflation alone (no production), two regions responded preferentially to the perturbed speech condition: one region along the boundary of the angular gyrus and posterior supramarginal gyrus in inferior parietal cortex, and another in the ventral posterior inferior frontal gyrus extending into the frontal operculum and anterior insula. These results provide evidence for the type of speech motor control processes hypothesized by the DIVA model.
Sound propagation through the large front-cavity volume of American English rhotic sounds
Authors: Zhang Zhaoyan , Espy-Wilson Carol , Boyce Suzanne , Tiede Mark
Published: October 2013

Rhotic sounds in American English (AE) are produced with a large front-cavity volume between the palatal constriction and the lip constriction which is the source of the characteristic low F3 of /r/’s. Depending on its shape, this volume may be modeled as (a) a single tube or (b) a single tube with a side branch. Both models produce a low F3, but through different mechanisms that involve differences in wave propagation direction. In this paper, we describe finite element studies of the acoustics of the front cavity, using dimensions derived from MRI studies of different subjects producing AE /r/. In particular, we discuss the case of a front cavity with minimal sublingual space. The sound-pressure distribution in the front cavity and the possible sublingual cavity was calculated at various harmonic excitations. The results show that at low frequencies (up to 2000 Hz, depending on specific dimensions) the front cavity is better modeled as a single tube, and at higher frequencies as a single tube with a side branch. The implications of accounting for wave propagation patterns (and thereby acoustically critical points along the vocal tract) on extraction of dimensional measures from MRI data will be discussed.
Using phonetic constraints to improve inversion
Authors: Potard Blaise , Laprie Yves
Published: October 2013

The goal of this work is to recover articulatory information from the speech signal by acoustic-to-articulatory inversion. One of the main difficulties with inversion is that the problem is under-constrained and inversion methods generally offer no guarantee on the phonetical realism of the inverse solutions. A way to adress this issue is to use additional constraints: in this work, contraints on phonetically relevant characteristics of the vocal tracts. Knowledge of the phonetic characteristics of most French vowels enables the derivation of reasonable articulatory domains in the space of Maeda parameters: given the formants frequencies (F1, F2, F3) of a speech sample, and thus the vowel identity, an ‘‘ideal’’ articulatory domain can be derived. The space of formants frequencies is partitioned into vowels, using either speaker-specific data or generic information on formants. Then, to each articulatory vector can be associated a phonetic score varying with the distance to the ‘‘ideal domain’’ associated with the corresponding vowel. Inversion experiments were conducted on two different speakers (male and female), using different implementations of these phonetical constraints. The articulatory trajectories were compared with those obtained without using these constraints and to articulatory trajectories measured from x-ray data.
The role of principal component analysis in vowel acquisition research
Authors: Zajdó Krisztina , van der Stelt Jeannette , Wempe Ton G.
Published: October 2013

Traditional formant analysis techniques have yielded questionable results, due to a high fundamental frequency in young children’s speech. Therefore, an alternative approach for researching vowel acquisition appears desirable. A band-filtering analysis method was developed that minimizes the dependence of the results on F0 to measure the spectral envelopes in children’s utterances automatically. In PRAAT scripts, first the criteria are set with regard to sound quality (to avoid clipped fragments) and F0 (to avoid vowel fragments with F0 above 425 Hz). Secondly, the spectral envelopes covering a range from 0 to 7 kHz, resulting in 40 filter values per spectrum, are calculated. A matrix is produced, representing intensity in each of the 40 filters. Data reduction is achieved via principal component analysis. Previous research (Pols, 1977) suggests that the first two principal components are related to the F1 and F2 values. Analyses of unlabeled vowel productions in 2-year-old Hungarian- and Dutch-speaking children show that the first two eigenvectors account for 55% of the variability. By using these two eigenvectors, a reference plane is created. Mapping labeled vowel measures onto the plane suggests that children produce language-specific vowel categories at this young age. [Work supported by UW and UvA/ACLC.]
Source localization in shallow water using artificial neural network
Authors: Too Gee-Pinn James , Lin Eshine
Published: October 2013

This study presents a method of passive localization for shallow water acoustic sources based on ray acoustic theory. ray acoustic theory is used to establish an underwater acoustic channel and to analyze physical quantities (arrival time and arrival angle) received by sensors, and to determine mutual relationship of rays between sources and receivers by using an artificial neural network for source localization. This ray-acoustic-based method of passive localization features simplicity, less calculation, and robust performance to environment variations. Hence, the method is more practicable and more valuable for source localization. In this study, the ray-acoustic-based method of passive localization is not only used in shallow water, but it is also verified by using a nontrained source to prove its accuracy and reliability. [Work sponsored by National Science Council.]
Underwater sound source with tunable resonator for ocean acoustic tomography
Authors: Morozov Andrey K. , Webb Douglas C.
Published: October 2013

A high Q tunable resonator makes possible the construction of an underwater sound source with a very high efficiency. Tuning this narrow-band resonator to match the instantaneous frequency of a reference signal enables control of sound projection over a wide spectrum width. At any given time the source has one fundamental resonant frequency equal to the instant frequency of the frequency modulating signal and the sound radiates with maximum efficiency. Such sound sources find application in ocean acoustic tomography and deep-penetration seismic profiling. The analysis of the simplest known Helmholtz resonator shows its ability to instantaneously switch its frequency or to quickly adjust to a changing reference frequency. From this simplest lumped resonant source the consideration then turns to a complicated distributed system, such as a tunable organ pipe. The analysis shows that the organ pipe can also be quickly swept and instantly switch the radiated signal frequency. The 200–300-Hz swept frequency sound source was designed for ocean acoustic tomography. Test results for this sound source are presented showing that it has a high efficiency, up to 50%, 3 dBi directivity gain, and depth independent parameters.
Recent experience with the voice intelligibility recommendations for fire alarm systems in large athletic spaces
Authors: Moore Matthew
Published: October 2013

Beginning with the NFPA 72 National Fire Alarm Code, 1999 edition, there has been a recommendation for voice intelligibility with fire alarm systems. The code appendix references a Common Intelligibility Score (CIS) of 0.70 or greater. This paper discusses how the design and construction process for some recent sports facility projects has been affected, and some of the difficulties with meeting this score. Example projects and measurements will be used to illustrate the challenges in these unique environments.
Prediction of speech intelligibility in rooms—a comparison of five methods
Authors: Hodgson Murray
Published: October 2013

In this paper, five methods for calculating speech intelligibility at listener positions in rooms, from measured impulse responses and speech and noise levels, are presented and compared. All methods involve combining a measure of room reverberation with the received signal-to-noise level differences in the occupied room. The first method involves measured octave-band C50s and signal-to-noise level differences to determine octave-band U50s. The remaining four methods comprise detailed (involving octave-band values) and simplified (involving single- and/or combined-frequency values) versions of two approaches. The first approach (second and third methods) involves measured background-noise levels and values of the Transmission Index measured for infinite signal-to-noise level difference. The second approach (fourth and fifth methods) involves 1000-Hz octave-band early-decay times and A-weighted signal-to-noise level differences. The five methods are briefly presented. Then, predictions by the methods for a number of classrooms are presented and compared, and the differences, the merits, and the disadvantages of the various methods are discussed.
Comparison of three methods for determining speech transmission index (STI)
Authors: Coffeen Robert C.
Published: October 2013

Speech transmission index (STI) is a commonly used descriptor whose goal is to quantify speech intelligibility. STI can be calculated from computer modeling of a particular architectural space, calculated from an impulse response of an existing space, and measured by introducing modulated noise in an existing space. This paper will present calculated and measured STI ratings for a particular and existing architectural space so that these three methods for determining STI can be compared.
Comparison and evaluation of animal vocalization enhancement techniques
Authors: Tao Jidong , Johnson Michael T.
Published: October 2013

Animal vocalization recordings are often corrupted by wideband background noise and interfering signals; however, signal processing methods for enhancement of these waveforms has not received as much attention in the literature as has human speech enhancement. In order to improve vocalization intelligibility and quality, a variety of enhancement methods taken from the field of speech processing are investigated here. These techniques range from traditional approaches such as spectral subtraction to more advanced ones such as Ephraim Malah log-spectral estimation and wavelet denoising. Results, measured by improvement in signal-to-noise ratio (SNR) and subjective perceptual tests, are given for several noise environments. Signal enhancement is demonstrated across a variety of species, including African elephant, Beluga whale, and ortolan bunting vocalizations. [Work supported by NSF.]
Perception of noise from diesel-engine powered vehicles
Authors: Hastings Aaron , Davies Patricia , Surprenant Aimee M.
Published: October 2013

A program was developed to simulate diesel engine sounds to have control over timing and amplitude variations in combustion events. The simulation includes this combustion-related noise, together with a noise floor and families of tones from auxilliary components. Additional shaping filters to control the spectral balance of the sounds are also included. Sound and vibration measurements from ten vehicles powered by diesel engines were taken and combustion event timing and amplitudes were extracted to determine what ranges and types of variations typically occurred. It was found that there were random, deterministic and individual cylinder variations present in one or more of the vehicles. From this analysis, the simulation program was used to generate 199 sounds for a listening test in which 40 subjects participated. Spectral balance, tonalness, and, fluctuations all affected the subjects’ responses and, above certain levels, as they increased so did subjects’ annoyance ratings. Based on the results of this test, modifications to the psychoacoustic annoyance model [Zwicker and Fastl, Psychoacoustics: Facts and Models, 2nd ed. (Springer, New York 1998)] are proposed. These modifications include the incorporation of a term that quantifies how the tonal character of the sound affects annoyance. [Work supported by Isuzu Motor Company.]
Target detection using a beam space time-reversal operator
Authors: Edelmann Geoffrey F. , Lingevitch Joseph F. , Fromm David M. , Gaumond Charles F. , Calvo David C.
Published: October 2013

Time-reversal operator methods such has DORT have recently garnered interest as promising techniques to ensonify ocean targets. However, measuring the time-reversal operator in the ocean by independently firing each element of a time-reversal mirror is difficult due to low signal-to-noise levels. This talk will discuss a portion of the shallow-water TREX-04 experiment in which energy was focused on a target using a vertical array of 64 source/receive transducers. The time-reversal operator was measured by transmitting energy from the full aperture of the time-reversal mirror instead of from just a single element. The time-reversal operator, made in beam space, was then used to successfully focus energy on a target 1 km away. [Work supported by the Office of Naval Research.]
Modeling of angular and range acoustic energy distribution from seismic exploration array in the Northern Gulf of Mexico
Authors: Tashmukhambetov Arslan , Sidorovskaia Natalia A. , Ioup George E. , Ioup Juliette W.
Published: October 2013

Broad-band experimental recordings (up to 25 kHz) of the acoustic energy arrivals from surface seismic exploration array by a bottom-moored hydrophone were gathered by the Littoral Acoustic Demonstration Center (LADC) in the Northern Gulf of Mexico during the summer of 2003. Experimental data are available for a wide range of horizontal distances (up to 7 km) and arrival angles. The numerical reconstruction of the acoustic radiation pattern and the acoustic energy decay law in range subject to the actual propagation conditions in the Northern Gulf of Mexico based on the calibrated recorded data by the identification of the zero-to-peak value in the direct pulse arrival is presented. The comparison of the measured energy losses with the expected ones due to spherical and cylindrical spreading is given. The identification of different types of arrivals (direct pulse, surface ghost, bottom reflected, etc.) and evaluation of their frequency content depending on the azimuth and emission angles and the distance to the array are conducted. The complexity of propagation and source models required to predict the identified arrivals is discussed.
An airgun array source signature model for environmental impact assessments
Authors: MacGillivray Alexander O. , Chapman N. Ross , Hannay David E.
Published: October 2013

Environmental impact assessments for seismic surveys often include estimates of radially propagating sound levels, which are used to determine marine mammal impact zones. Sound levels may be estimated using computer-based acoustic modelling techniques but these require an accurate description of the survey source signature—arrays of airguns, in particular, have complex, highly directional source functions that depend on the array layout. To address this requirement, an airgun array source signature model has been developed for the purpose of underwater noise level prediction. The source model is based on published descriptions of the physics of airgun bubble oscillations and radiation [A. Ziolkowski, Geophys. J. R. Astron. Soc. 21, 137–161 (1970)] and includes the effects of port throttling, motion damping and bubble interactions. The output of the model is a collection of notional signatures which may be used to compute the source function of the array in any direction. Free parameters in the model have been fit to a large collection of existing airgun signature data, for airguns ranging from 5 to 185 in3. The output of the model is suitable for estimating sound levels resulting from airgun survey activity. [Work supported by NSERC.]
Classification of natural landmark with biosonar
Authors: Wang Maosen , Zell Andreas
Published: October 2013

Echolocating bats can make nocturnal flights in acoustically cluttered environments with the use of echolocation. Their marvelous ability to evaluate natural targets in complete darkness provides us an opportunity to learn target detection, classification, and identification with similar biomimetic platforms. In this work, natural landmark classification with a binaural system, a sequential sensing strategy, and a frequency after reconstruction algorithm are adopted to provide sequential acoustic images for target classification. Experimental results suggest that considerable improvements in classification accuracy can be achieved by the use of this sequential classification method.
Acoustic scattering by benthic shells: Dominant scattering mechanisms and applications
Authors: Stanton Timothy K. , Chu Dezhang
Published: October 2013

When benthic shells occur in sufficiently large numbers, they can dominate acoustic backscattering by the seafloor, especially at angles of incidence away from normal. In order to use sound as a tool to remotely detect and quantify the shells, the scattering properties of the shells need to be understood, both in free-space as well as when placed on the seafloor. Through laboratory experimentation, it has been determined that the edges of certain types of shells (such as bivalves and sand dollars) can dominate the scattering over an important range of grazing angles. The surfaces of these shells and others dominate under other conditions. The dominant scattering effects are discussed in the context of interpreting acoustic backscatter data in terms of meaningful parameters such as numerical density of the shells.
Time-domain calculations of acoustic interactions with rigid porous surfaces
Authors: Wilson D. Keith , Aldridge David F. , Symons Neill P. , Ostashev Vladimir E. , Collier Sandra L. , Marlin David H.
Published: October 2013

In previous work, a set of time-domain equations for sound propagation in a rigid porous medium, including viscous and thermal dissipation effects, was derived [Ostashev et al., J. Acoust. Soc. Am. 115, 2624 (2004)]. From those equations, time-domain counterparts to frequency-domain impedance boundary conditions (BCs) were also derived. In this paper, several approaches to computational implementation of time-domain acoustic interactions with rigid porous surfaces are discussed that are based on these equations. Most of the approaches involve convolutions between relaxation or transformed impedance functions and the acoustic wavefield variables. For these approaches, when the relaxation times for viscous and thermal diffusion in the pores (which are inversely proportional to the static flow resistivity) are large, the convolution integrals must be evaluated over many periods of the acoustic disturbance and therefore become very computationally demanding. Some example finite-difference time-domain (FDTD) calculations show the significantly increased attenuation within the porous medium resulting from the convolution terms. Computational instabilities are observed when the wave is strongly attenuated over the FDTD spatial grid interval. Alternative approaches, based on Pade and series approximations to the time-domain impedance BC, are also considered that provide more compact numerical solutions.
Structural acoustic modeling issues for surface ship hull structures
Authors: Martin Nathan , Chandiramani K.L. , Brown Richard , Haberman Robert , Dees Robert
Published: October 2013

The complex 3D character of surface ship hull structures provides a challenging configuration for detailed analytic predictions of hull response and radiation. This paper reviews key structural-acoustic response mechanisms for doubly-framed surface ship hulls and assesses the implications of these mechanisms for the prediction of hull response and radiation.
Acoustic radiation from arrays on coated prolate spheroids
Authors: Boisvert Jeffrey E. , Van Buren A. L.
Published: October 2013

An important aspect in the design of conformal sonar arrays is the effect of an outer coating on the array response. The design of the coating is subject to the competing requirements of acoustic transparency, toughness for impacts, and flow noise rejection. The present study considers an array of rectangular pistons, conformal to a rigid prolate spheroidal baffle, covered by an outer coating. The coating is modeled as a fluid, i.e., it is assumed to have a vanishingly small value of the shear modulus, typical of many elastomers. Unlike a coating on a spherical or cylindrical baffle, where a uniform thickness coating can be defined by simply increasing the value of the radial coordinate, the corresponding operation in prolate spheroidal coordinates results in a confocal layer, i.e., a layer that is not of constant thickness. In this paper a new method is described that allows for a uniform thickness coating in prolate spheroidal geometry. This method employs the spheroidal addition theorem to satisfy the boundary conditions. Comparisons of the beamformed array response for a confocal coating versus a uniform thickness coating are given for a sample array with various coating properties. [Work supported by the NUWC ILIR Program.]