bluuebunny/crossref_metadata_2025 huggingface dataset:
Limitations of current sound system intelligibility performance measurement techniques and metrics
Authors: Mapp Peter
Published: October 2013

The current and increasing trend to employ sound systems for emergency warning and evacuation purposes has led to an increased need to verify and assess the intelligibility performance of such systems. Applications range from passenger cabin PA systems in aircraft through to paging and emergency evacuation systems in industrial complexes, train stations, public buildings, and stadiums. Although a number of methods are available, these have either been borrowed from the fields of architectural and auditorium acoustics or one-dimensional communication networks. The paper shows that such approaches exhibit a number of fundamental inherent limitations that can lead to significantly reduced accuracy when assessing typical sound system performance. Although important factors such as background noise and reverberation can be accounted for, it is shown that the typical, nonlinear amplitude and frequency response characteristics of many systems are not adequately addressed. Furthermore, common temporal distortions such as discrete specular reflections or later arriving echoes are also shown to result in inaccurate assessments. The need for a simple, easy to use, and yet robust measurement technique is highlighted.
Improving the intelligibility of aircraft PA systems
Authors: Mapp Peter , Warnaka Glenn , Shimazu Bruce
Published: October 2013

The speech intelligibility of commercial aircraft audio systems is of great importance. The authors have noted however, that subjectively, the intelligibility of such systems can vary considerably. Measurements made on a number of commercial flights indicate that signal to noise ratios are often around 0 dBA and frequently negative. As aircraft loudspeakers are located relatively close to the passenger, the collapsing narrower dispersion and beaming of the high frequencies from conventional cone loudspeakers can cause loss of intelligibility for passengers not on the axis of the loudspeaker and produce uneven sound distribution within the cabin. Distributed mode loudspeakers (DML) have demonstrated intelligibility superior to conventional loudspeakers in other applications such as public address systems and the opportunity was taken to evaluate their performance within an aircraft cabin environment. DMLs are very light in weight and can even be built into the internal trim, resulting in a near zero weight loudspeaker. Their essentially flat configuration also requires less volume. This paper presents a comparative evaluation of the speech intelligibility of current aircraft loudspeakers and DMLs as determined by both conventional word score tests and Speech Transmission Index measurements using both full STI and RaSTI. The tests were performed under both quiet and representative in-flight cabin noise conditions.
A spherical microphone array-based system for the determination of spatial information in reverberant sound fields
Authors: Gover Bradford N. , Ryan James G. , Stinson Michael R.
Published: October 2013

A measurement system has been developed for analysis of the directional and spatial variations in reverberant sound fields. Two spherical, 32-element arrays of microphones are used to generate narrow beams over two different frequency ranges, together covering 300–3300 Hz. Using an omnidirectional loudspeaker as excitation in a room, the pressure impulse response in each of 60 steering directions is measured. Through analysis of these responses, the variation of arriving energy with direction is studied. The system was first validated in simple sound fields in an anechoic chamber and in a reverberation chamber. The system characterizes these sound fields as expected, both quantitatively through numerical descriptors and qualitatively from plots of the arriving energy versus direction. The system was then used to measure the sound fields in several actual rooms. These sound fields were highly anisotropic, influenced greatly by the direct sound and early-arriving reflections. The rate of sound decay was not independent of direction, sound being absorbed more rapidly in some directions than in others.
Acoustic diffusers II
Authors: Bidondo Alejandro
Published: October 2013

Two lines of research are presented in this work about acoustic diffusers: A microscopic one which looks into its temporal operation presenting a new phenomenons vision, showing its memory property. The other, a pragmatic one, is very useful in the project advance control process, where a diffusion coefficient named Sound Field Diffusivity (SFD) is established for the quantitative evaluation of real diffuse fields. Further research suggestions are presented in both lines, one looking for a more complete surface diffusion coefficient that incorporates the time energy spreading, and on the other a frequency resolution of the SFD for precise design.
IMMOTUS*, sound insulation system
Authors: Schroer Daniel R. , Deblander Jean-Philippe
Published: October 2013

In 1999, a breakthrough sound insulation system was launched in the building industry. This system, developed with IMMOTUS*, a new polymeric foam from Dow, provides a unique answer to the critical issue of efficient sound insulation improvement with minimal space loss in dwellings and other building families. IMMOTUS R+ is a plasterboard laminate made of a 20 mm foam board and a gypsum board of 12.5 mm. When attached to a wall according to the system principle, a ceiling or a roof, this sound insulation system considerably reduces the airborne and impact noise transmissions. First in France and in the United Kingdom, then in other regions of the globe, a vast amount of data have been collected through Dow Laboratory experiments, Acoustic Institutes testing sessions, and successful commercial projects. During the course of this presentation, the originating concept and the main applications for this lightweight, high performance system will be described. Both the IMMOTUS* foam material and integrally associated systems properties will be reviewed. Acoustical behavior prediction and several configurations of full-scale experiments will be discussed. (* Trademark of the Dow Chemical Company.)
Morphometric analyses of hearing in two families of pinnipeds
Authors: Marsh Sarah E. , Fischer Iris , Arruda Julie , Ketten D. R.
Published: October 2013

Pinniped (seal and sea lion) auditory systems operate in two acoustically distinct environments, air and water. Otariids (sea lions and fur seals) generally divide their time evenly between land and water while phocids (true seals) spend the majority of their time in water. These pinniped families therefore offer an exceptional opportunity to investigate aquatic versus terrestrial hearing mechanisms. Recent reports indicate differences in hearing ranges and sensitivities among species from these families [D. Kastak and R. Schusterman, J. Acoust. Soc. Am. 1303, 2216–2228 (1998); P. W. B. Moore and R. Schusterman, Marine Mammal Sci. 3, 31–53 (1987)]. In this project, ear anatomy of three representative pinniped species (Otariidae: California sea lion, Zalophus californianus; Phocidae: northern elephant seal, Mirounga angustirostris; and harbor seal, Phoca vitulina) was examined using computerized tomography (CT scans) and gross dissection. Three-dimensional reconstructions of heads and ears from CT data were used to determine inter-aural dimensions and ossicular chain articulations. Ossicular weights and densities were measured using conventional and displacement techniques. Results strongly support a canal-centric system for pinniped sound reception and localization. Further, true seals show adaptations for aquatic high-frequency specialization. [Work supported by ONR.]
The Office of Naval Research program on the environmental consequences of underwater sound: Approach, results, and applications
Authors: Gisiner Robert
Published: October 2013

The Office of Naval Research (ONR) is a basic and applied research funding organization established by the U.S. Navy in 1946 to stimulate research in a wide range of topics of relevance to naval and ocean science. In the mid- to late 1980s, ONR was one of the first research funding agencies to respond to the concern about possible adverse biological impacts from manmade underwater sound. Since 1991, ONR has had a large and growing research program on marine animal hearing and sound use, the effects of manmade sound on physiology and behavior, and the development of research tools for better assessing the effects of sound. Most recently, the program has increased emphasis on predictive models of underwater noise effects as a management and decision-making tool. The presentation will include specific examples of ONR-sponsored research and describe the mechanisms by which research results are made available to the government and the public.
Distributions and densities of marine mammals and their role in modeling acoustic effects
Authors: Hain James H.
Published: October 2013

Describing, modeling, and predicting acoustic effects on marine mammals depends on accurate inputs of various kinds. One required input is the distribution and density of marine mammals for a given area and time period. The quality of input is especially important for endangered species and critical habitats. Often, acousticians are requested to proceed with modeling in the absence of credible marine mammal data. This may produce unsatisfactory results for use in support of meeting environmental compliance and regulatory requirements. For any given period and area, it is unlikely that we will ever know the true population or density. Therefore, we are left to estimate it. Examples are provided of recent efforts to calculate marine mammal densities in two proposed Navy east coast shallow water training ranges. How many marine mammals of each species occupy a 500 square nautical mile area by season and depth strata? The effects of survey planning, sample size, data collection protocols, and analyses on the resulting estimates are shown. The inescapable conclusion is that increased interaction, collaboration, and understanding of methods and requirements between acousticians and survey biologists are not only desirable but essential.
The pressure gun, a wireline implosive source
Authors: Sauter Allan W. , Dorman LeRoy M.
Published: October 2013

We report development of an implosive source which is not based on stress failure. It is triggered by an electrical signal carried by armored cable, allowing precise timing. It can be used on the sea floor or in the water column. The trigger releases a small valve, allowing ambient pressure to operate a shuttle which opens a volume to flooding by seawater. It is recocked by returning to the surface. The strength of the source is best described by equivalent earthquake magnitude, which is proportional to volume change (a depth-independent quantity, in contrast with an explosive source). The 20-l device we have made has a moment magnitude of about −0.9. The duration of the signal is controlled by the time required to fill the volume, so the radiation of high frequencies is limited. This concentrates the energy radiated in seismically useful frequency band, and reduces effects on marine life. We have used this source to excite Scholte waves at a depth of just over 1100 m. At this depth the peak pressure occurs at about 20 Hz. It has the same source moment as a half-pound of explosive at about 700 m.
Extension of the long-line hydrophone calibrator to higher frequency
Authors: Luker L. D.
Published: October 2013

The usable frequency range of the long-line hydrophone calibrator (LLHC) has been increased for endfire fields in a portion of the calibrator. The LLHC is a water-filled chamber for performing acoustic calibration of towed or other line arrays under operational environmental conditions. Projectors distributed along its length can produce an acoustic field equivalent to that seen by the array in open water from a sound source at any desired bearing and range. The original LLHC configuration was capable of measuring the sensitivity and directivity patterns of line arrays or modules up to 49 m long at temperatures from 2°C to 35°C, pressures up to 6.9 MPa, and frequencies from 5 Hz to 700 Hz. The LLHC has recently been modified in several ways to increase the usable frequency range to 2000 Hz for endfire fields over a 5 m long region inside the chamber. The modifications include changes to the way projector drives are computed, installation of thin-walled collars to make the acoustic impedance (and propagation speed) down the chamber more uniform, and the addition of four nonuniformly-spaced hydrophones. Results are presented that show maps of the sound field uniformity inside the calibration chamber. [Work supported by ONR Code 321.]
Theoretical and experimental investigation of acoustic dyadic sensors
Authors: Silvia Manuel T. , Cray Benjamin A.
Published: October 2013

The theoretical directivity index of a single combined acoustic receiver, a device that measures acoustic pressure, each component of particle velocity and velocity gradients at a collocated point, is presented. This receiver, described here as an acoustic dyadic sensor, is highly-directional, achieving as much as 9.5 dB of directivity gain. In practice, the higher-order velocity gradient components are obtained via a finite difference approximation of measured particle velocity. The statistical behavior of the errors in estimating the pure and mixed second-order partial derivatives are given. Experimental in-water measurements were completed on a prototype dyadic sensor, developed by EDO Corporation of Salt Lake City, Utah, and confirmed theoretical predictions.
Tone hole radiation directivity measurements
Authors: Scavone Gary P. , Karjalainen Matti
Published: October 2013

Measurements have been conducted in an anechoic chamber for a comparison to current acoustic theory with regard to radiation directivity from a cylindrical pipe with toneholes. Several difficulties arise in measurements of this sort, including (1) the generation of sufficient driving signal strength at the pipe input for pickup by an external microphone; (2) external source-to-pickup isolation; (3) measurement contamination due to nonlinear driver distortion. Time-delay spectrometry using an exponentially swept sine signal was employed to determine impulse responses at points external to the experimental air column. This technique is effective in clearly isolating nonlinear artifacts from the desired linear system response along the time axis, thus allowing the use of a strong driving signal without fear of nonlinear distortion. The experimental air column was positioned through a wall conduit into the anechoic chamber such that the driver and pipe input were located outside the chamber while the open pipe end and tone holes were inside the chamber, effectively isolating the source from the pickup. Measured results are compared to both transmission-line, frequency–domain simulations, as well as time–domain digital waveguide calculations.
Wavelet-based automatic recognition of musical instruments
Authors: Kostek Bozena , Zwan Pawel
Published: October 2013

The objective of the present work is to automatically extract information from monophonic sounds. This process consists of several stages, namely, preprocessing, parameterization, and classification. This paper shows a thorough study on the wavelet-based parameterization of musical instrument sounds and automatic recognition by means of artificial neural networks (ANNs). First, an engineered method of pitch detection is presented and exemplified by several analyses. A short discussion on error associated with automatic pitch tracking is also included. Then, examples of time-frequency analyses of various musical instrument groups are presented. The analyses are performed employing a database containing musical sounds recorded at the Sound and Vision Engineering Department, Technical University of Gdansk. On the basis of such analyses a set of parameters is derived. Feature vector properties are then discussed. For that purpose Fisher statistics is used. It allows checking the separability between musical instrument pairs. In addition, for the purpose of automatic recognition of musical instrument groups artificial neural networks are used. Various structures and training methods of the ANNs are examined. Exemplary results obtained in the carried out investigations are provided and analyzed. Concluding remarks concerning further development of such experiments are also included in the paper.
Auditory roughness estimation of complex spectra—Roughness degrees and dissonance ratings of harmonic intervals revisited
Authors: Vassilakis Pantelis
Published: October 2013

Examination of musical instrument construction and performance practice indicates that sound variations involving the sensation of roughness are found in most musical traditions. A new roughness estimation model is proposed that better represents the theoretical knowledge and experimental results on sensory roughness. The model is based on previous studies [E. Terhardt, Acoustica 30, 201–213 (1974); W. A. Sethares, Tuning, Timbre, Spectrum, Scale (Springer-Verlag, London, 1998)] and introduces an additional term to account for: (a) the contribution of the amplitudes of interferring sines to the roughness of a sine-pair and (b) the difference between amplitude modulation depth and degree of amplitude fluctuation. Compared to earlier roughness estimation models, the proposed model demonstrates the best agreement between estimated and observed roughness. Possibilities for further improvement are discussed. A hypothesis linking dissonance and roughness ratings of harmonic intervals within the chromatic scale is tested. Clear presence or absence of roughness appears to dominate dissonance ratings. In other cases, decisions on dissonance often ignore roughness and are culturally and historically mediated. The results suggest that, in the Western musical tradition, the consonance hierarchy of harmonic intervals corresponds to variations in roughness degrees. Further study should include cross-cultural empirical investigations.
What people want to hear
Authors: LoVerde John J. , Dong Wayland
Published: October 2013

As communities increase in population density, more multi-family dwellings appear in the marketplace. With the advent of acoustical legal action against developers and growing sophistication of owners, the need for the developer to communicate to the potential buyer the level of sound isolation provided has become important. The legal cases have created a renewed debate over the criteria and standards for these dwelling types. The Department of Housing and Urban Development (HUD) started the discussion of Acoustical Criteria in Multi-Family dwellings [U.S. Department of Housing and Urban Development, A Guide to Airborne, Impact, and Structure Borne Noise Control in Multi-Family Dwellings (1963)]. HUD later released minimum acceptable acoustical standards [U.S. Department of Housing and Urban Development, HUD Minimum Property Standards: Multi-Family Housing (1973)]. States, counties, and cities, through code requirements, have also provided minimum acceptable standards. This paper shall compare the existing standards. The comparison will include the types of noise sources and isolation systems that are addressed, and a discussion of the adequacy of the standards based on interaction with other consultants in legal cases.
The ‘‘focusing’’ of transient response in structural networks: The development of ‘‘hot spots.’’
Authors: Ertel John , Dickey Joe
Published: October 2013

Vibration induced in a structural network will reverberate in and among the component systems and may, through constructive interference, create ‘‘hot spots,’’ points of anomalously high amplitude. Time domain responses of networks consisting of 2 and 4 one-dimensional wave bearing systems are calculated and displayed in the time/spatial domain and illustrate this phenomenon. Hot spots may be important in networks where they occur in fragile systems placing that system at risk for failure. Examples of physical structures where this model applies and numerical validation of this phenomenon are presented. An elementary illustrative example is a single physical beam supporting several wave types (thus several dynamic systems which interact with each other at the ends of the beam) driven, e.g., in compression but failing in flexure.
Intrinsic pitch in German: A puzzle?
Authors: Mooshammer Christine , Hoole Phil , Alfonso Peter , Fuchs Susanne
Published: October 2013

Intrinsic pitch (IF0) differences have often been attributed to a physiological consequence of articulation, i.e., higher tongue-jaw position yields higher F0. However, Fischer–Jorgensen showed that German tense-lax pairs are produced with a similar IF0 despite the lower tongue positions of lax vowels. She also found better agreement between IF0 and jaw position than for tongue height. The first aim of this study was to replicate Fischer–Jorgensen’s study using EMMA (seven speakers). Second, the contribution of jaw position on IF0 was examined by a bite-block condition (three speakers). The results of Fischer–Jorgensen were confirmed concerning similar IF0 of tense-lax pairs, where the jaw rather than the tongue better correlated with IF0, because jaw position differences between tense-lax vowel pairs were smaller than the tongue position differences. However, it seemed unlikely that this could be any kind of casual explanation, because frequent cases were found of somewhat higher F0 in the lax member of a pair, despite somewhat lower jaw position, and because of the fact that the same patterns of IF0 differences were found for the bite-block and normal condition. a)Currently at Dept. of Audiol. and Speech, University of Tennessee, Knoxville, TN. b)Also at Queen Margaret Univ. College, Edinburgh.
Perception of tremor rates in pathological voices
Authors: Kreiman Jody , Gerratt Bruce R.
Published: October 2013

Vocal tremors characterize many pathological voices, but acoustic-perceptual aspects of tremor are poorly understood. To investigate the perceptual importance of the rate of frequency modulation, we created synthetic copies of ten pathological voices. For half the voices, tremors were approximately sinusoidal, and for the other half, tremors were irregular in rate and amplitude. Tremor rates in the copies were varied in nine steps of 0.25 Hz. Listeners heard each synthetic copy paired with the natural voice sample, and judged which of the copies best matched the target voice. Small differences in tremor rates were easier to hear when tremors were sinusoidal than when they were irregular, but overall, listeners were not especially confident of their judgments. For sine wave tremors, differences in rate were easier to hear for faster rates, but wider extents of tremor significantly reduced the ability to detect rate differences. Similar effects of rate manipulations were not observed for irregular tremors. These results suggest that the perceptual importance of tremor rates depends on the total pattern of frequency variation. In the most complex cases, listeners apparently respond to overall levels of F0 variability rather than to precise patterns of frequency change. [This research was supported by the NIH.]
A high-speed methodology for comparing phonosurgical outcomes in an excised larynx model
Authors: Tayama Niro , Montequin Douglas W. , Berry David A.
Published: October 2013

Many phonosurgical techniques have been used to treat hoarseness resulting from vocal disease, such as unilateral vocal fold paralysis. However, because of difficulty in documenting changes in mucosal wave propagation, comparing the outcomes of different phonosurgical techniques presents a formidable challenge. The purpose of this investigation is to compare two phonosurgical treatments of unilateral vocal fold paralysis, thyroplasty type 1 and arytenoid adduction, in terms of their impact on the mucosal wave. For thyroplasty type 1, the pushing force was systematically varied. For arytenoid adduction, the pulling forces were systematically varied. Across all experiments, systematic changes were also made in the subglottal pressure, glottal airflow, pre-phonatory posturing, and in the size of the glottal chink. Because stroboscopic methods could not image properly the aperiodic vibrations associated with paralysis, a digital high-speed imaging system was utilized (the Kodak EktaPro 4540). Using this system, the 3-D vibrations of the mucosal wave were imaged across all experimental conditions. In this presentation, we focus on the methodology employed to collect these measurements.
Potential application of c
l
e
a
n algorithm for ultrasonic NDT
Authors: Mahmud Khan Mohammad , Ohba Ryoji
Published: October 2013

The detection of close reflection peaks in ultrasonic pulse compression system has been studied by implementing the clean algorithm, which is popular in image processing of radio astronomy data. The algorithm has been adjusted so that it is applicable to ultrasonic data. It has been applied to some practical data with satisfactory results. It is found to be effective for the detection of close peaks and removing sidelobes. The algorithm has been implemented successfully in the detection of close holes in metal plates and some potential use of the method has been described.
Acoustic propagation in a shallow water waveguide with a rough surface: Comparison of results from PE and coupled unperturbed normal mode methods
Authors: Reynolds Stephen A. , Thorsos Eric I. , Henyey Frank S.
Published: October 2013

The PE method is an accepted standard for accurate acoustic propagation modeling. We have developed an alternative method (Henyey et al., this session) based on flat-surface normal modes, with coupling due to surface waves. Results for the complex field obtained with the new mode coupling method are compared with the solutions from a rough surface PE code [A. D. Rosenberg, J. Acoust. Soc. Am. 105, 144–153 (1999)]. A variety of rough surfaces are considered in the comparison. Mode coupling is most easily tested using a single mode starting field with a sinusoidal surface whose wavelength is chosen to produce a resonance with another mode. As an example of a randomly rough surface, a modified Pierson–Moskowitz spectrum of surface waves is considered. Comparisons with the PE method include closeness of the results, accuracy of energy conservation, computational speed, and ease of use. [Work supported by ONR.]
Rough boundary effects on shallow water propagation: Some comparisons between GRAB and PE
Authors: Williams Kevin L. , Thorsos Eric I. , Elam W. T.
Published: October 2013

The Gaussian Ray Bundle (GRAB) computer code is an accepted Navy Standard for calculating propagation. GRAB uses an effective reflection loss to approximate the effect of ocean surface roughness. Results for GRAB and rough surface PE are compared for a shallow water waveguide (50 m depth with a sand bottom) for several wind speeds and for ranges up to 5 km. A special case examined is a 3.2 kHz source at mid depth with a 10° beamwidth and 10° grazing angle. The GRAB results use the Bechmann-Spezzichino reflection coefficient module. The PE results are calculated using 50 realizations of a Pierson–Moskowitz surface and either coherent or incoherent averaging. GRAB with Bechmann–Spezzichino is most appropriately compared to the PE coherent averages. This comparison shows differences on the order of 3 dB in the intensity averaged across water depth at 5 km and a wind speed of 5 m/s. The general implications of these differences as well as differences seen in the intensity as a function of depth at discussed. Work supported by the ONR.
Parabolic wave equation based rough sea surface scattering
Authors: Ryan Frank
Published: October 2013

A physics based technique is described for computing both forward scattered and backscattered (i.e., reverberation) fields from realistic rough sea surfaces at cm wavelengths. The method is based on a novel split-step Fourier parabolic wave equation (PWE) method incorporating a stochastic wind and swell ocean wave spectral model which includes both gravity and capillary scales. Comparisons between the PWE based scattered fields and results using the standard Miller–Brown rough surface reflection coefficient method will be shown. Examples of the rough surface scattered field vertical angle-of-arrival spectra under a variety of ducting conditions will be shown, and the impact on high precision spatial tracking (beamforming) discussed.
Acoustic wave detection and analysis in a sandy sediment
Authors: Chotiros Nicholas P.
Published: October 2013

Coherence, wave front curvature, and direction of arrival provide important clues for analyzing acoustic signals in ocean sediments. Measurements were made using a buried acoustic receiving array and a mobile sound projector to study the underlying physical processes in the penetration of sound into sandy ocean sediments, particularly at shallow grazing angles, as part of the Sediment Acoustics Experiment (SAX99). The buried array was insonified by a wide-band sound source carried on a remotely operated vehicle (ROV). The discrimination between refraction and scattering processes was of particular interest. A method was used that employed coherent superposition to distinguish between refracted and scattered signals. Variable focusing was used to sharpen the distinction between refraction and scattering. Sound waves entering the sediment at steep angles were clearly refracted. At shallow angles, other processes come into play. This is a component of a multi-disciplinary effort to study sediment acoustics. [Work supported by ONR, Ocean Acoustics.]
Centimeter scale seafloor relief measurement using an ROV
Authors: Chotiros Nicholas P. , Piper James N. , Moore James , Weigl Daniel F.
Published: October 2013

For high-frequency acoustic backscatter modeling, in the band 1 to 100 kHz, centimeter scale seafloor roughness is an important parameter. A measurement method was devised for deployment aboard a remotely operated vehicle (ROV) consisting of lasers and a video camera. It was designed to measure small-scale roughness in the range of spatial wavelengths from 0.01 to 1 m. The ROV is well suited to this application because it can hover above the seabed, and it can cover a relatively large area. A cylindrical lens was used to spread the laser beam from a low-power laser into a fan, and direct it downward to produce a stripe on the sediment. Viewing from an oblique angle with a video camera, the profile of the sediment directly under the stripe was obtained. The most difficult problem was precisely determining and tracking the position of the ROV. In the SAX99 experiment, positioning was achieved by using small shells and other objects on the seafloor as landmarks. [Work supported by ONR, Ocean Acoustics.]
Spatially coincident measurements of acoustic backscatter and forward scatter at shallow grazing angles
Authors: Hines Paul C. , Osler John C. , Stewart Shawn A. , MacDougald Darcy J.
Published: October 2013

Direct measurement of acoustic scattering from the seabed at shallow grazing angles presents a considerable challenge in littoral waters. Specifically, returns from the air–water interface typically contaminate the signal of interest. To address this issue DREA has developed a pair of sea going research systems for measuring acoustic scatter from the seabed in shallow water environs. The first system is a bottom mounted parametric transmitter with a 6-channel superdirective line array receiver. The second system is a vertical line array of 8 hydrophones and 2 transmitters. Operating the systems in tandem allows measurements of backscatter and forward scatter from the same scattering patch at the same time. In this paper the acoustic attributes of both systems will be outlined, the experimental geometry will be described, and measurements of acoustic backscatter and forward scatter collected during one such experiment will be presented.
Bill Watkins listens to all the whales in the North Pacific Ocean
Authors: Gisiner Robert
Published: October 2013

In the early 1990s portions of the Navy’s classified underwater listening systems (IUSS) were made available for research. Bill Watkins had long been familiar with this system and eagerly went to work listening to whales in the North Pacific Ocean. A decade later the work that Bill Watkins has amassed, and its impact on marine mammal research, is truly remarkable. Each month Bill’s team produces a summary of hundreds, if not thousands, of whale calls detected and localized using the Navy’s listening arrays. These data have pinpointed seasonal centers of vocal activity in unexpected places, leading to changes in our strategies for monitoring several populations of endangered large whales. Bill has tracked the migratory paths of blue, fin, and humpback whales and has charted the shifts in migratory timing caused by climatic phenomena such as ENSO. Bill has taken a complex technology created for other purposes and crafted a process to get it to deliver simple, clear, and unclassified information for an entire ocean: we have not yet plumbed the limits of what we can do with this information.
Blue whale call intensity varies with ambient noise level
Authors: Wiggins Sean M. , Oleson Erin M. , Hildebrand John A.
Published: October 2013

Blue whales (Balaenoptera musculus) vary the intensity of their sound production level in response to varying ambient noise levels. Blue whales produce tonal, type B calls with a fundamental frequency of 16 Hz, and a strong third harmonic at 48 Hz. These calls, along with other low-frequency baleen whale vocalizations, ambient sea noise, and shipping noise have been monitored and recorded on continuous seafloor acoustic recorders near Cortez and Tanner Banks, offshore Southern California, since August 2000. By separately comparing the intensity in the 16 and 48 Hz spectral level bands with the intensity of ambient noise at and near those frequencies, we observe a correlation between the sound intensity level of blue whale type B calls and ambient noise levels. This is verified by the inspection of individual call intensity in various noise environments. Preliminary analysis indicates a less than unity proportional increase in the call intensity with increasing ambient noise. A trend of increased call intensity with an increased noise level has implications for the effect of shipping, and other natural and artificial noise sources in the marine environment on the behavior of blue whales.
Tranmission loss of a ribbed composite vessel
Authors: Vipperman Jeffrey S. , Li Deyu , Avdeev Ilya
Published: October 2013

The tranmission loss behavior of a new type of composite structure has been characterized through experimental structural and acoustic modal analyses as well as finite element analysis (FEA) modeling. The tapered cylindrical structure, which is being considered for expendable launch vehicles, is fabricated graphite-epoxy and has a series of internal tri-directional rib stiffeners. Light coupling between the structure and interior acoustic cavity exists, permitting separate analyses of the ‘‘structurally dominant’’ and ‘‘acoustically dominant’’ modes for the system. The modal analyses and FEA modeling efforts are complimentary since the former helps to determine appropriate FEA model parameters and the latter predicts the shape and ordering of the modes to expect in the experiments. These modes can be subsequently coupled through the modal-interaction approach for prediction purposes. In addition, they can be used to determine the structural acoustic coupling coefficients or radiation modes of the system in order to assess the relative contributions of various structural modes to the internal acoustic levels. A comparison between in situ transmission loss measurements and the prediction capabilities of the FEA models will be assessed. [Work supported by AFRL Space Vehicles Directorate, and University of Pittsburgh.]
Piezoelectric sensors on internally fluid loaded elastic shells
Authors: Stepanishen Peter R. , Cox Donald L.
Published: October 2013

A general analytical approach is presented to evaluate the harmonic response of electro-mechanical–acoustical systems consisting of external piezoelectric sensors on internally fluid loaded elastic shells with electrical, mechanical and/or acoustical excitations. The approach utilizes the in-vacuo eigenvectors of the shell as the basis functions for the solution of the fluid loaded shell and sensor interaction problem. Fluid loading on the shell is included in the approach via internal mode and frequency-dependent radiation impedances and multiport models are used to represent the sensors. As a specific case of interest, the response of conformal piezoelectric sensors on an internally fluid loaded spherical shell is addressed using both the analytical approach and the FEM method. Numerical results are presented to illustrate the importance of acoustic and elastic interaction effects among sensors on the shell with and without internal fluid. Agreement between the FEM and analytical results is excellent over a broad frequency range. General characteristics of the electro-mechanical–acoustical responses are discussed. [Work supported by ONR.]
Backward projection of harmonic pressure measurements using the generalized internal source density method and SVD
Authors: Stepanishen Peter R. , Brodjoneboro Irsan S.
Published: October 2013

The Generalized Internal Source Density (GISD) method [P. R. Stepanishen, J. Acoust. Soc. Am. 102, 1955–1963 (1999)] is used to backward project acoustic harmonic pressure fields from an external measurement surface of revolution enclosing the finite three-dimensional vibrating body that generated the field. Both the normal velocity and pressure fields on the vibrating surface can be readily reconstructed using the method. The GISD method is based on first decomposing the acoustic field on the measurement surface into a modal summation of circumferential orders, where each modal pressure field is associated with an internal linear distribution of ring sources on the axis of revolution of the measurement surface. Each modal linear distribution of ring sources is evaluated using Singular Value Decomposition (SVD) methods. A brief review of the method is presented and discussed and then numerical results are presented to illustrate the accuracy of the method for a spherical shell and finite cylindrical shells with hemispherical end caps. The results show excellent agreement with the prescribed normal velocities and associated pressure fields over a wide frequency range.
Low-frequency resonances and scattering enhancements of water-filled cylindrical shells open at each end
Authors: Osterhoudt Curtis F. , Marston Philip L.
Published: October 2013

At low frequencies, the contents and termination of cylindrical steel shells can significantly affect the scattering properties. We have observed a giant resonance associated with the limiting case of a water-filled cylindrical shell open at both ends. The resonance, which is analogous to the fundamental mode of an open organ pipe, occurs because of the strong low-frequency reflection of internal acoustic waves from the open ends of the shell [H. Levine and J. Schwinger, Phys. Rev. 73, 383–406 (1948)]. Since the wavelength is approximately twice the length L of the cylinder, the wave number radius product ka may be much less than unity when L/a is large. The mode was excited by an external acoustic transducer and was detected either as a backscattering enhancement or (in the case of a sufficiently large cylinder) with a hydrophone within the cylinder. By tuning the frequency of an incident tone burst to the mode frequency, the echo is like the response of a high-Q harmonic oscillator, showing a gradual buildup and decay. [Work supported by ONR.]
Speaker identification with dynamic synapses
Authors: Dibazar Alireza Afshordi , Namarvar Hassan Heidari , Liaw Jim-Shih , Berger Theodore
Published: October 2013

In this paper, we have proposed a new method of presenting a speech signal to the dynamic synapses neural network (DSNN) and optimal selection of input neurons for the speaker identification task. The DSNN developed by Liaw and Berger (1996) was employed to capture the speaker information from the action potentials, which was generated by wavelet filter banks. In order to sufficiently capture the speaker information, the time-delayed temporal patterns of action potentials are used as the inputs of the network. The optimal number of input neurons obtained by minimizing the total output error of the network while identifying a target speaker and rejecting the other unknown speakers. The genetic algorithm was employed to design a supervised learning rule for training of the network. The TI-46 dataset was used in both the training and testing phase. Comparative results with previous methods showed a better performance of the correct classification for the model-based dynamic synapses speaker identification system. [Work supported by DARPA and ONR.]
VUI aide to character recognition
Authors: Lin Qiguang , Munich Mario , Xia Kun
Published: October 2013

Recognizing characters over the telephone is a challenging task, not only for computers but even for human beings. The reason is because there exist several highly confusable pairs/groups, e.g., the E-set, M/N, F/S, and A/J/K. Human beings resolve errors by saying phrases like: ‘‘F as in Frank.’’ In this report, a new character recognizer is first described, and then a novel voice-user-interface (VUI) is suggested to achieve better user experience with the task. The recognizer provides a hidden Markov model (HMM) for each character, augmented with two context-independent silence models. Furthermore, state-tying is made so that only discriminative states in individual HMMs are unique, while other states are shared. The next step is to design an effective VUI to facilitate character recognition. The VUI asks the user to say one character at a time. It then plays back the best result of recognition for confirmation. If the result is correct, the user proceeds; otherwise, the user says a keyword such as ‘‘correction,’’ and the recognizer retrieves an alternative candidate. Candidate selection can be made, for instance, according to an N-best list or a prior confusion matrix. It is found that the proposed VUI effectively disambiguates pairwise confusions such as F/S.
An introduction to the Buckeye corpus of conversational speech
Authors: Raymond William D. , Makashay Matthew J. , Dautricourt Robin , Johnson Keith , Pitt Mark
Published: October 2013

Words in connected speech are subject to many types of phonological variation. Variation is both widespread and conditioned by an array of factors, including speech rate and context. However, little is known about the frequencies of variants or their acoustic realization. The variation-in-conversation (ViC) group at Ohio State has set about to quantify and measure the distributional properties of variants, with an eye toward their implications for spoken word perception. The current study reports on a corpus of approximately 40 h of conversation-style speech recorded during interviews of 43 midwestern speakers, stratified for age and sex. The speech was paired with text transcription, and is being phonetically transcribed using a combination of automatic and hand alignment. Details of the collection and transcription methodology will be presented, along with some parameters of variation that have emerged from the data. For example, in the corpus t is more likely to flap (16% of all underlying t’s) than d (8%). However, d is more likely to be deleted (36%) than t (18%), and almost all deletions of d are in the word final position (33%) unlike t (11%).
Concatenative text-to-speech synthesis based on waveform interpolation (a time frequency approach)
Authors: Morais Edmilson , Dogil Grzegorz
Published: October 2013

The time domain pitch synchronous overlap and add (TD-PSOLA) is the technique most used in comercial concatenative text-to-speech (TTS) synthesis systems. However, it is well known that TD-PSOLA presents several drawbacks. In order to overcome some drawbacks of the TD-PSOLA, this work presents a method based on time frequency interpolation (TFI) [Yair Shoham]. The method introduced here is a pitch-synchronous time-frequency approach of the waveform interpolation technique (WI) [Bastian Kleijn]. The goal of this work is to show that the TFI technique presents some important advantages to concatenative TTS synthesis. It allows pitch scale modification (PSM) independent of time scale modification (TSM) in a quite straightforward manner, and with high quality. TSM and PSM can be done in a continuous way, without any limitation of pitch period resolution. Moreover, the TFI technique allows simple, flexible, and efficient procedures to smooth diphone (or any other kind of unit) boundaries. The proposed system was evaluated using diphones and prosodies generated by the Festival system [Alan Black, Paul Taylor]. Subjective tests were performed, between the proposed TFI system and the standard TD-PSOLA system, highlighting the superior quality of the proposed system in comparison with TD-PSOLA.
Extraction and generation of aperiodic component in speech sounds
Authors: Kawahara Hideki
Published: October 2013

An effective method to extract and represent aperiodicity in speech sounds is proposed for making refinements on a high-quality speech analysis, modification and synthesis system straight [Kawahara et al., Speech Commun. 27, 187–207 (1999)], based on time-frequency analyses by the use of an instantaneous frequency and group delay. The proposed signal representation consists of a frequency domain aperiodicity measure and a time domain energy concentration measure to represent source attributes. The frequency domain aperiodicity measure is defined as a ratio between the lower and upper smoothed spectral envelopes to represent the relative energy distribution of aperiodic components. The time domain measure is defined as an effective duration of the aperiodic component and event distribution pattern. These aperiodicity parameters and F0 as time functions are used to generate the source signal for synthetic speech by controlling relative noise levels and the temporal envelope of the noise component of the mixed mode excitation signal, including fine timing and amplitude fluctuations. Relations between these parameters and temporal masking effects also will be discussed. [Work partly supported by mext Grant (C) 11650425 and CREST Japan.]
Complete optimization of excitation and model parameters in parametric speech coders
Authors: Lashkari Khosrow
Published: October 2013

In a speech coding system, synthesis error (the difference between the original speech at the encoder input and reproduced speech at the decoder output) is a more relevant measure of signal distortion than linear prediction (LP) error. By minimizing the synthesis error instead of the linear prediction error, the analysis and synthesis stages become more compatible. In a previous paper (J. Acoust. Soc. Am. 109, 2491), an Analysis-by-Synthesis (AbS) technique for joint optimization of the excitation and filter parameters in parametric speech coders were presented. Using a gradient search in the root domain, synthesis error is minimized by reoptimizing the filter parameters for a given excitation. This paper provides a major improvement over the previous work, resulting in more than 3 dB of gain in the segmental signal-to-noise ratio. The improvement is due to a new algorithm for computing the gradient vector in the gradient search procedure. The paper describes the new algorithm and report on the improved results. By adding an extra minimization step, this technique can be incorporated into the existing LPC, multipulse LPC, and CELP-type speech coding standards.