bluuebunny/crossref_metadata_2025 huggingface dataset:
Geoacoustic inversion of noise coherence in shallow water
Authors: Thomson David J. , Desharnais Francine , Drover Matthew L. , Gillard Chris A.
Published: October 2013

It is known that the geoacoustic properties of a shallow-water sea-bed can be inferred from relatively simple measurements of the ambient noise coherence between a pair of vertically separated hydrophones [D. M. F. Chapman, ‘‘Surface-generated noise in shallow water: A model,’’ Proc. Inst. Acoust. 9, 1–11 (1987)]. The design of an autonomous buoy package for acquiring geoacoustic information by this method is currently being considered by DRDC-Atlantic in support of matched-field localization efforts that are being developed for use with rapidly deployable arrays. Initially, vertical coherence estimates from a simple shallow water noise model were fit to measured coherences by adjusting geoacoustic parameters by a trial and error procedure. A more systematic approach involves combining noise coherence models with nonlinear global optimization methods based on matched-coherence processing concepts to search the space of possible sea-bed parameters more efficiently. In this paper, we report on recent efforts to use a hybrid simplex simulated annealing scheme [M. R. Fallat and S. E. Dosso, ‘‘Geoacoustic inversion via local, global, and hybrid algorithms,’’ J. Acoust. Soc. Am. 105, 3219–3230 (1999)] to match an increasingly realistic suite of candidate geoacoustic parametrizations to some acoustic noise coherent data measured with modified sonobuoys deployed at several shallow water locations on the Scotian Shelf.
Breaking waves and ambient sound in the ocean
Authors: Melville W. Kendall
Published: October 2013

It is now well accepted that breaking waves at the ocean surface are the primary source of sea-surface sound. However, the relationship between the kinematics and dynamics of breaking waves, their acoustic source strength and their statistical description has not been explored in any comprehensive way. In this paper, the components of such a description are reviewed and discussed in the context of laboratory and field studies from the literature, theoretical models of breaking statistics, and recent field measurements of the kinematics and statistics of wave breaking. The implications of the results for the use of ambient sound to quantify processes of air–sea interaction are discussed. [Work supported by ONR and NSF.]
Observation of the laryngeal movements for throat singing
Authors: Sakakibara Ken-Ichi , Konishi Tomoko , Murano Emi Z. , Imagawa Hiroshi , Kumada Masanobu , Kondo Kazumasa , Niimi Seiji
Published: October 2013

Throat singing is a traditional singing style of people who live around the Altai Mountains. Khöömei in Tyva and Khöömij in Mongolia are representative styles of throat singing. The laryngeal voices of throat singing is classified into (i) a drone voice which is the basic laryngeal voice in throat singing and used as drone and (ii) a kargyraa voice which is very low pitched with the range outside the modal register. In throat singing, the special features of the laryngeal movements are observed by using simultaneous recording of high-speed digital images, EGG, and sound wave forms. In the drone voice, the ventricular folds (VTFs) vibrate in the same frequency as the vocal folds (VFs) but in opposite phases. In the kargyraa voice, the VTFs can be assumed to close once for every two periods of closure of the VFs, and this closing blocks airflow and contributes to the generation of the subharmonic tone of kargyraa. Results show that in throat singing the VTFs vibrate and contribute to producing the laryngeal voice, which generates the special timbre and whistle-like overtone.
Some recent examples of fractal music
Authors: Guerra-Torres Cesar , Hinojosa-Rivera Moises , Garza-Garza Juan Angel , Elizondo-Garza Fernando J.
Published: October 2013

Since the first published studies of fractal music by Mandelbrot and Voss, the relationship between music, mathematics, and fractal geometry has been a very active field of research. It has been found that the music of classical composers can be characterized by fractal or self-affine parameters, which in turn serve as the basis for synthetic fractal music. In this work is presented a brief discussion of the state of the art as well as some recent examples of fractal music, including a live demonstration.
An acoustic study of the Brazilian cuica
Authors: Wheeler Paul A.
Published: October 2013

The cuica is a friction drum of African origin played in the batucada (an ensemble of instruments used for the samba) during the Brazilian carnival. It is played by rubbing a bamboo rod which is connected to the center of a drum head, giving a rhythmic grunting sound. Pitch is changed by applying pressure to the membrane. This paper discusses several acoustic aspects of a folk cuica (made of a gourd) including the waveforms, spectra, and time envelopes produced. Rubbing the bamboo rod gives a primitive saw-toothed excitation, similar to a bowed violin string. This is connected to the center of a membrane which modifies and radiates the sound. The body of the cuica contributes little to the sound.
Overview of predicting noise levels in indoor industrial spaces
Authors: Brittain Frank
Published: October 2013

Predicting indoor noise in industrial facilities is a vital part of designing industrial plants. The predicted levels are used in the design process to determine compliance with occupation noise exposure limits, and to estimate levels inside the walls as starting point for predicting community noise radiated by buildings. Once levels are predicted, the noise controls needed can be developed. Special methodologies are needed, because normal room acoustics found in architectural acoustics texts is valid only for nearly empty rooms with limited absorption and ranges of room dimensions. The fittings inside industrial spaces can profoundly affect the propagation of noise and the resulting noise levels. In an industrial space, such as a power plant, there is no such thing as a reverberant field, except in isolated areas. In industrial spaces, including factories, predicting noise levels by summing free and reverberant fields gives erroneous results that are usually overly conservative. This paper discusses normal empty room acoustics, and problems typically encountered when it is applied to industrial spaces, particularly those with a high density of fittings or very large spaces. Also, alternative methodologies for predicting indoor noise levels in industrial spaces, which are based on standards and software, are identified and discussed.
Acoustics of industrial buildings
Authors: Vorlaender Michael
Published: October 2013

Industrial halls have significantly different shapes as rooms for other purposes. Typically these enclosures are very large and/or flat, which means that the room height is usually very much smaller than width and length. Sound fields in these types of enclosures cannot be expected to have an approximate diffuse sound field. Instead, the decay curves are bent, while sound level versus distance (sound propagation curve) is declining. This contribution summarizes physical properties of sound fields in flat and long rooms, gives some examples of predictions based on image sources, on Kuttruffs integral equation, results of scale model experiments and ray-tracing algorithms with two different levels of complexity. It can be shown that in many cases predictions can yield sufficiently accurate results. However, problems will occur in cases where rooms are heavily filled with scattering objects (fittings). On the other hand, the exact spatial distribution of sound in these nondiffuse cases has shown to have less influence on the perceived annoyance than expected. Accordingly, predictions of sound levels with reasonable accuracy should be sufficient to describe the subjective effect of listening, unpleasantness and annoyance in industrial halls.
Computer-based method for the diagnostics and rehabilitation of tinnitus patients
Authors: Kostek Bozena , Czyzewski Andrzej , Skarzynski Henryk
Published: October 2013

The proposed method is an electronic diagnostic and rehabilitation system for people suffering from internal ear noise, namely tinnitus patients and people with an abnormally high hearing sensitivity—hyperacusis patients. Thanks to the method employing multimedia personal computers to the fitting masking sounds to the patient’s needs, tinnitus patients can be rehabilitated using the masking or habituation method. The subject is asked to answer detailed questions in the electronic questionnaire and next to identify those sounds which most strongly resemble those they perceive as ear noise. Following an algorithm analysis of the results and analysis of the selection of sounds made by the patient, the computer diagnoses the patient as free from ear noise and hyperacusis or classifies them into a risk group. Next, the patient is informed about the result and can then read about the causes of the ailment and the recommended treatment and also can download compressed sounds applicable in therapy to the rehabilitation tool coupled with the personal computer which is a programmable ear noise masker. In the paper the algorithm of analysis of patients answers and acoustic characteristics of tinnitus maskers are discussed. Hitherto obtained results of this method application are demonstrated.
Electroacoustic verification of FM benefits in advanced hearing aid circuitry
Authors: Schafer Erin C. , Thibodeau Linda M.
Published: October 2013

Miniature FM receivers attached to ear-level hearing aids can provide significant improvements in speech recognition in noisy environments. These receivers are designed to couple to the hearing aid via a boot connection with limited or no adjustment of hearing aid settings. Ideally, the circuitry allows the FM signal to be approximately 10 dB more intense than the typical signal from the hearing aid microphone, allowing for an FM advantage. Furthermore, for intense input levels, the output limiting should not differ for the hearing aid compared to the hearing aid and FM receiver combined, i.e., FM transparency. Following procedures recommended by the American Speech, Language, and Hearing Association, the electroacoustic responses of digital and conventional aids were measured with and without coupling to FM systems at typical conversational input levels and maximum input levels. The rms difference between the electroacoustic responses of the hearing aid alone and coupled to the FM was used to quantify the FM advantage at typical input levels and the FM transparency at high input levels. The finding of great variability in both FM advantage and transparency supports the need for additional fitting controls or design modifications to obtain the maximum FM benefit.
Effect of noise on social relationships
Authors: Panuszka Karla , Panuszka Ryszard
Published: October 2013

Sociology is the scientific study of human society and social behavior. The effect of noise and vibrations under unwanted vibroacoustical fields shows important changes in social interactions. A new social question arises from these studies: What effect does unwanted noise pollution play on human environments and how are relationships changed? It has been observed in workers that noise and vibrations play an important factor in machinery safety. Music shows a positive effect on relationships by providing a person’s mental attitude with a happy euphoria. Natural sounds of waterfalls, streams, and rivers also show an improved mood that enhances relationships in a positive way. Staszic University AGH has documented these interactions under the influence of acoustical fields. The human ear cannot hear sound waves below 20 Hz, but these waves have been shown to have effects on people’s interactions. The main focus of sociology is interactions of large populations; thus, noise interactions need to be studied. Three areas of investigations could arise from these studies: family relationship responses, occupational safety, and domestic and business violence. Broad investigation results will be displayed.
Melody recognition with temporal and spectral cues in normal-hearing and cochlear-implant listeners
Authors: Kong Ying-Yee , Zeng Fan-Gang
Published: October 2013

Cochlear-implant users can achieve a high level of speech recognition, but their ability to appreciate music is severely limited. This study investigates the relative contribution of the temporal and spectral cues to melody recognition. Two sets of 12 familiar songs were generated: one set contained both rhythm and melody information (the rhythm–melody condition), whereas the other set contained only melody information (the no-rhythm condition, where all notes had the same duration). Normal-hearing listeners achieved 95%–100% accuracy in both conditions. While cochlear-implant listeners achieved 42%–94% accuracy in the rhythm–melody condition, they performed essentially at chance level in the no-rhythm condition. To further identify the cues used in melody recognition, temporal envelopes from the original broadband signal were extracted and modulated with white noise. When the rhythm cue was available, the normal-hearing listeners achieved 60%–100% recognition, which was similar to the performance achieved by the cochlear-implant listeners. When the rhythm cue was not available, both normal-hearing and cochlear-implant listeners performed essentially at chance level. These results suggest that the present cochlear-implant listeners relied solely on temporal information to recognize familiar melodies. The fine structure information is certainly needed to allow true appreciation of music for cochlear-implant users.
Binaural phase masking experiments in stereo audio
Authors: Iliev Alexander I. , Scordilis Michael S.
Published: October 2013

Researchers have established that in binaural hearing the smallest detectable angular separation between two sources, commonly referred to as the minimal audible angle (MAA), for a pair of sources on the horizontal plane depends on the frequency of the emitted pure tone and the azimuth angular separation between the sources. One interesting way is to view the sources’ angular perturbation within the MAA limits as noise in the phase domain, and the listener inability to detect this perturbation as the result of a masking process. The present discussion focuses on experimental procedures for examining the perception of MAA and the corresponding interaural phase difference (IPD) when complex sound sources are located in the most sensitive region, which is directly in front of the observer (both azimuth and elevation angles at 0° degrees). Sound stimuli were viewed as the linear combination of pure tones, as provided by Fourier analysis. Results indicate that masking is achieved when the IPD is disturbed within a threshold limit corresponding to the MAA for pure tone sources. Listening tests using stereo audio further validated our observations. [Work supported by Watermark Technologies.]
Auditory performance in an open field
Authors: Fluitt Kim , Letowski Tomasz , Mermagen Timothy
Published: October 2013

Detection and recognition of acoustic sources in an open field are important elements of situational awareness on the battlefield. They are affected by many technical and environmental conditions such as the type of sound, the distance to a sound source, terrain configuration, meteorological conditions, hearing capabilities of the listener, the level of background noise, and the listeners’ familiarity with the sound source. A limited body of knowledge about auditory perception of sources located over long distances makes it difficult to develop models predicting auditory behavior on the battlefield. Our purpose in the present study was to determine the listeners’ abilities to detect, recognize, localize, and estimate distances to sound sources from 25 to 800 meters from the listening position. Data were also collected for meteorological conditions (wind direction and strength, temperature, atmospheric pressure, humidity) and background noise level for each experimental trial. Forty subjects (men and women, ages 18–25) participated in the study. Nine types of sounds were presented from six loudspeakers in random order; each series was presented four times. Partial results indicate that both detection and recognition declined at distances greater than approximately 200 m and distance estimation was grossly underestimated by listeners. Specific results will be presented.
Comparing some convolution-based methods for creation of surround sound
Authors: Czyzewski Andrzej , Kostek Bozena
Published: October 2013

Spatialization of the sound using the multichannel techniques is now getting widespread. One can derive many rules for surround sound recording and reproduction. However, there exists only few methods suitable for recording sound in large auditoria ensuring its proper subsequent reproduction in small reproduction rooms, preserving spatial properties of sound acquired in the original recording location. Some experiments presented in the paper were devoted to the simulation of acoustics of the recording hall using the convolution of a monophonic audio signal with the multichannel impulse response of the hall. A special microphone setup was created for this task and an original method of recording multichannel impulse response of auditory halls was conceived and implemented. In this method the acoustical signal recorded quasianechoically was convolved with five impulse responses of the simulated room measured in the room corners and on the stage. Another examined method which is more standard employed convolution of monophonic signals with long-term averaged HRTFs (Head-Related Transfer Funtions). Surround recordings made with both mentioned convolution techniques were then compared on the basis of subjective testing results. The details of the examined surround recording methods and results of their assessments will be discussed in the paper. [Work supported by KBN, Grant No. 8 T11D 00218.]
Vibration analysis and building vibration isolation design for an automated people mover system in airport terminal structures
Authors: Phillips James E.
Published: October 2013

Detailed Finite Element Analysis (FEA) models were developed for a proposed airport terminal expansion project. An Automated People Mover (APM) system is incorporated into the airport structures for shuttling passengers quickly between terminals. The dynamic forces imparted onto the structures by the moving APM vehicles and the analysis approach were based upon established techniques developed for addressing ground-borne and structure-borne vibrations from rail systems. Measurements were conducted at two other major airports with existing rubber tire APM systems on aerial structures. These measurements provided baseline vibration levels for the analysis as well as the forces imparted to the structure by the APM vehicles. The results of the analyses were utilized in the design of a vibration isolation system included in the structural design of a guideway within a terminal building in order to mitigate structure-borne vibration from APM operations.
Novel features for robust speech recognition
Authors: Potamianos Alexandros
Published: October 2013

Recently there has been much research in the area of robust front-ends and new features for automatic speech recongition (ASR). These efforts have had limited success for certain databases and recording conditions. In this work, we review some recent work on features for ASR: the articulatory front-end of Li et al. (2000), nonlinear modulation features [Quatieri (2002); Dimitriadis and Maragos (2002)], chaotic features [Pitsikalis and Maragos (2002)], short-time spectral moments [Paliwal et al. (2000)], etc. We extend the work of Potamianos and Maragos (2001) to show how some of these features relate to the standard front-end of short-time smooth spectral envelope. We also analyze some of these new features using classification and regression trees to show the relevance of the features for phone-classification task.
Fundamental frequency estimation using signal embedding in state space
Authors: Terez Dmitry
Published: October 2013

A new robust nonlinear method for determination of fundamental frequency (F0) was recently proposed with application to speech pitch detection [Terez, Proc. ICASSP 1, 345–348 (2002)]. The method uses state-space embedding technique originally introduced for analyzing chaotic signals. The new method has been generalized and tested on different types of speech signals, as well as on a variety of other acoustic signals. In addition, some artificially generated nonstationary and complex wave forms have been used to test the limits of the method in comparison with other known (short-term) F0-estimation techniques (e.g., correlation, spectrum or cepstrum-based methods). Evaluation results demonstrate a unique combination of properties distinguishing the new method from conventional techniques. In particular, reliable and accurate F0 estimates can be obtained for clean periodic signals using signal segments slightly longer than one complete fundamental period. Other properties include immunity to speech formants and robust performance on noisy and band-limited speech signals. Some improvements are introduced to reduce the number of required computations and to achieve higher (subsample) accuracy. The method waused to implement a robust pitch-tracking algorithm for speech processing applications. Further information and demo software can be found at http://www.soundmathtech.com/pitch.
The challenges of archiving networked-based multimedia performances (Performance cryogenics)
Authors: Cohen Elizabeth , Cooperstock Jeremy , Kyriakakis Chris
Published: October 2013

Music archives and libraries have cultural preservation at the core of their charters. New forms of art often race ahead of the preservation infrastructure. The ability to stream multiple synchronized ultra-low latency streams of audio and video across a continent for a distributed interactive performance such as music and dance with high-definition video and multichannel audio raises a series of challenges for the architects of digital libraries and those responsible for cultural preservation. The archiving of such performances presents numerous challenges that go beyond simply recording each stream. Case studies of storage and subsequent retrieval issues for Internet2 collaborative performances are discussed. The development of shared reality and immersive environments generate issues about, What constitutes an archived performance that occurs across a network (in multiple spaces over time)? What are the families of necessary metadata to reconstruct this virtual world in another venue or era? For example, if the network exhibited changes in latency the performers most likely adapted. In a future recreation, the latency will most likely be completely different. We discuss the parameters of immersive environment acquisition and rendering, network architectures, software architecture, musical/choreographic scores, and environmental acoustics that must be considered to address this problem.
Model for coherent scattering from a network of buried river channels in a stratified ocean waveguide
Authors: Lee Sunwoong , Ratilal Purnima , Makris Nicholas C.
Published: October 2013

A theoretical model of three-dimensional bistatic scattering from a network of buried river channels in a continental shelf waveguide is presented. The scattered field on the river channel walls is estimated to be locally specular using the Kirchhoff approximation. This field is then propagated out to a distant receiving array using Green’s theorem and the waveguide Green function, so that it is also valid for scattering in the near field of the river channel network. This model is applied to calculate the scattered and beamformed field from a network of buried river channels in the East Coast strataform area, whose geomorphology was mapped out by numerous geophysical surveys. The modeled scattered field is found to be dependent upon the bistatic orientation and projected area of the river channel walls relative to the source and receiver. To verify the modeling, the result of the simulation is compared with data from the Geoclutter 2001 acoustic experiment where images of buried river channel networks were acquired from long-range using a bistatic sonar system.
Range-dependent modal eigenvalue estimation in the SWAT experiment
Authors: Souza Luiz L. , Frisk George V. , Becker Kyle M.
Published: October 2013

Three modal mapping experiments (MOMAX) have been conducted in shallow water environments: two in the East coast STRATAFORM site off the New Jersey coast; and one in the Gulf of Mexico off the Florida coast. A low-frequency source emits a set of CW signals in the range 20–500 Hz, and the field is measured by drifting buoys. The ultimate goal of these experiments is to invert the acoustic field data in water for the sound speed profile in the seabed. The analysis has focused on measuring the modal characteristic wavenumbers (eigenvalues) as a function of position. In general, one has to resort to range-dependent spectral analysis. Techniques based on both the short-term Fourier transform and high-resolution parametric spectrum estimation have been used for measuring eigenvalues. New results from the year 2000 SWAT (MOMAX III) experiment at the STRATAFORM site will be discussed. These include a comparison of data obtained from tracks along and across the New Jersey shelf. [Work supported by ONR and WHOI Academic Programs Office.]
Geoacoustic inversion of broadband acoustic recordings of a surface ship on an horizontal line array in the Gulf of Mexico
Authors: Koch Robert A. , Knobles David P.
Published: October 2013

As part of a comprehensive experiment to support research on geoacoustic inversion techniques, acoustic signals from a surface ship were recorded on a bottom mounted, horizontal line array at a soft sediment, shallow-water site in the Gulf of Mexico off Port Aransas, TX. Along with the surface ship recordings, water sound-speed profiles, cw TL data at 53, 103, 153, 503, and 953 Hz, and recordings of light bulb implosions were obtained. In this paper a method for performing inversions from broadband beam data is demonstrated and compared with corresponding inversions from broadband element data and with previously reported inversions of the light bulb implosions. The performance in a simulated annealing algorithm of cost functions that involve coherent or incoherent sums over frequency and multiple time segments is presented, and the uncertainties in the geoacoustic parameter values are examined. Comparisons of measured TL with predicted TL for the bottom descriptions resulting from the inversions are used to judge validity of the inverted profiles and the efficacy of the various cost functions. The applications of this work to ocean environments for which the bottom parameter values are unknown or uncertain will be discussed. [Work supported by ONR.]
MegaSound: Sound in Irish megalithic buildings
Authors: Reijs Victor
Published: October 2013

Stimulated by the studies done by Paul Deveraux and Robert Jahn, research has been conducted on the sound properties of two megalithic chambers is Ireland: Dowth South and Fourknocks I. As reference measurements two normal rooms (bed- and bathroom) have been studied. The following aspects will be covered in the presentation: some theoretical background on acoustical modes (within a passage, a chamber, and a combination of them: Helmholtz resonator); tips for doing sound experiments inside megalithic chambers (like: equipment, measurement software, power provisioning and calibrating); frequency response measurements (between 20 and 200 Hz) for the surveyed chambers/rooms; comparison of the results with other researchers’ results; background on the pitch of the human (male, female, and child) voices in neolithic times and recommendations for future research. The presentation also provides insight in the aeralization (simulation) of sound in a megalithic chamber, covering: software that can do these simulations; issues in finding the basic information, e.g., acoustic absorption coefficients and provide examples of the results. I would like to thank all the people who have provided constructive feedback on my work (http://www.iol.ie/∼geniet/eng/megasound.htm).
Computer analysis of sound recordings from two Anasazi sites in northwestern New Mexico
Authors: Loose Richard
Published: October 2013

Sound recordings were made at a natural outdoor amphitheater in Chaco Canyon and in a reconstructed great kiva at Aztec Ruins. Recordings included computer-generated tones and swept sine waves, classical concert flute, Native American flute, conch shell trumpet, and prerecorded music. Recording equipment included analog tape deck, digital minidisk recorder, and direct digital recording to a laptop computer disk. Microphones and geophones were used as transducers. The natural amphitheater lies between the ruins of Pueblo Bonito and Chetro Ketl. It is a semicircular arc in a sandstone cliff measuring 500 ft. wide and 75 ft. high. The radius of the arc was verified with aerial photography, and an acoustic ray trace was generated using cad software. The arc is in an overhanging cliff face and brings distant sounds to a line focus. Along this line, there are unusual acoustic effects at conjugate foci. Time history analysis of recordings from both sites showed that a 60-dB reverb decay lasted from 1.8 to 2.0 s, nearly ideal for public performances of music. Echoes from the amphitheater were perceived to be upshifted in pitch, but this was not seen in FFT analysis. Geophones placed on the floor of the great kiva showed a resonance at 95 Hz.
Physiological measures of auditory temporal integration in chinchillas and humans
Authors: Clock Eddins Ann , Eddins David A.
Published: October 2013

Temporal integration is a phenomenon in which signal detection improves with increasing signal duration. The magnitude of threshold improvement has been shown to vary as a function of signal frequency and is often diminished in the presence of hearing loss. In studies of chinchilla auditory nerve fibers and cochlear nucleus (CN) neurons, we have shown that single neuron thresholds improve with increasing duration in a manner similar to data from psychophysical studies. Moreover, it appears that a critical feature of duration coding in the CN is the pattern of spike activity rather than the total number of spikes over time. Thresholds measured in evoked-potential studies with chinchillas at the level of the inferior colliculus and with humans at the level of the cortex also reflect physiological evidence of temporal integration, with comparable frequency effects and degradation with hearing loss like that observed psychophysically. Although frequency effects that show greater threshold improvement for low versus high frequencies have been reported in the literature for decades, the underlying mechanism remains unclear and is under further investigation psychophysically and physiologically in our laboratories. The results of these investigations will be discussed along with potential underlying mechanisms.
Modeling the pressure pulse shape of piezoelectric lithotripters
Authors: Dreyer Thomas , Riedlinger Rainer
Published: October 2013

Piezoelectric focusing transducers are widely used in extracorporeal lithotripsy. To optimize the therapeutically relevant focal pressure pulse it is necessary to affect the generated pulse shape at the transducer surface. Therefore a modeling approach is required containing the acousto-mechanical properties of the transducer structure as well as the influence of the electrical drive. The procedure presented here uses three dimensional transient finite element simulations to calculate an electro-acoustical impulse response of the transducer structure and linear systems theory to model the influence of the driving circuit on the emitted acoustical signal. Applying a short electrical pulse an acoustical impulse response can be simulated under plane wave conditions, which is valid at sufficiently large distances from the transducer. Focal pressures are estimated rapidly by linear calculations or accurately by a nonlinear propagation model. The influence of electrical drive conditions on the emitted acoustical signal is investigated very efficiently by a convolution with the desired electrical input, avoiding FEM simulations for each case. Reverting this process the required driving voltage course for a given pressure signal is determined. Alterations of the pressure signal in terms of pulse width and tensile components are demonstrated theoretically, varying the design parameters of the transducer.
Single number to represent hearing protector attenuation
Authors: Gerges Samir N. Y. , Gaste Eric N. , Steffen Vera M.
Published: October 2013

The existing single numbers for hearing protector noise attenuation are NRR, SNR, HML, and NRRsf. All of these numbers are calculated using a standard ambient noise called pink noise with a total level of 100 dBC. This pink noise does not represent real industrial noise. This pink noise has its peak values in dBA at 2 kHz, which is not usually the case for most real industrial environments. Also, this single number is not associated with statistical distribution and standard deviation. In this paper a trial is presented to calculate the NRRsf using measured industrial noise spectrum. The calculation is carried out for a typical foam-hearing protector. The REAT naive subject results obtained from laboratory measurements of 20 subjects using ANSI 12.6-1997 (B) gives 20 individual attenuations as function of frequency. For each one of the real noise spectra measured in industry, using one subject attenuation, an NRRsf individual number is obtained. The average NRRsf and standard deviation of these attenuation values are presented along with their statistical distribution. These results are discussed and compared to the single classical numbers.
Seismic coda
Authors: Fehler Michael
Published: October 2013

Observations and analysis of seismic scattering in the heterogeneous earth have grown from initial observations in the 1960s into a well-developed subfield of seismology. The field presents many challenging, interesting, and fruitful areas of research for seismologists. The term seismic coda has been used to describe the overall discipline but the term coda was originally used only for the tail portion of the seismograms recorded within 100 km of earthquakes. Several characteristics of seismic coda have been identified that provide important constraints on models to explain the observed waveforms. It is now generally accepted that coda is a superposition of waves that have been multiply scattered from heterogeneities in the Earth’s lithosphere. Coda waves, even if they are not completely understood, have practical use for seismologists because some of their characteristics lead to extremely reliable methods for quantifying relative sizes of earthquakes and relative local site amplification characteristics in seismic hazards investigations. Observations and theoretical modeling studies of coda waves have provided insight into the complexity of wave propagation in the Earth and have yielded new insights into the character of the Earth’s lithosphere.
Quasi-2D diffusion of elastic waves
Authors: Van Tiggelen Bart , Tregoures Nicolas , Campillo Michel , Margerin Ludovic , Lacombe Celine
Published: October 2013

Recent studies on seismic Coda and seismic equipartition have shown the possibility to model the propagation of seismic waves around 5 Hz in the crust using a heterogeneous layered model with a free surface overlying a homogeneous half-space. The depth of the layer is roughly 30 km. If the mean free path of the waves is in excess of this value, and this is probably the case in many places in the world (except perhaps Mexico), the wave diffusion should be described by mode diffusion in a 2D layer, rather than plane waves in a 3D medium. This notion affects many observed and potentially observable quantities, such as equipartition, leakage of energy, and coherent backscattering. [Work supported by the French Ministry of Research.]
Simulations of acoustic streaming inside and surrounding thermoacoustic stacks
Authors: Hamilton Mark F. , Ilinskii Yurii A. , Zabolotskaya Evgenia A.
Published: October 2013

Acoustic streaming generated inside and surrounding thermoacoustic engine stacks by standing waves in resonators is investigated numerically. The study is performed using the nonlinear 2-D code described recently by the authors [J. Acoust. Soc. Am. 111, 2076 (2002)]. The stack consists of parallel plates of negligible thickness. There is no restriction on plate separation, or on the length or position of the stack in the resonator. Temperature dependence of the fluid viscosity is taken into account. A temperature gradient imposed on the stack drives the sound field. Acoustic streaming patterns, investigated in terms of average mass flow velocity, are calculated by time averaging the momentum density. For typical plate separations on the order of the thermoviscous penetration depth, only the inner, boundary-layer streaming vortices exist between the plates. These inner vortices rotate in directions opposite those of the outer, more familiar Rayleigh streaming vortices encountered in wide channels (wider than about 10 penetration depths). The inner vortices extend beyond the ends of the stack out to distances on the order of the acoustic particle displacement. Flow patterns near the entrances to the stack are investigated as functions of channel width and acoustic amplitude [Work supported by ONR.]
Extraction and visualization acoustic contour of the voice signal, its use as a method for the implementation and rehabilitation of voice and speech
Authors: Suarez Guerra Sergio
Published: October 2013

The possibility to see what is said has resulted in something quite new, as a method for the implementation and rehabilitation of voice and speech. The immediate visualization of the principals acoustic parameters of the voice signal, in a graphic form, associating them with images that represent what was said, resulted in an additional alternative very stimulant in the field of Phoniatry, with applications in Special Schools. In researches done in the Special Schools during one year, results denoted a sustained advance in the learning of correct diction, from students, that in addition to the traditional method used an extraction and visualization acoustic parameters system. This system, Exparam V.2.0, has been proposed for its introduction at a national level into the Special School of the Republic of Cuba and the final statistics of its use will be ready at the end of the course 2002/2003. At the CIC/IPN, Mexico D.F., is at conclusion another system, which will support the Phoniatry clinic consulting room management using voice analysis in the same manner involving the acoustic profile representation and the inclusion of the patient’s Clinic data base. This system is being tested at the Human Communication Institute. (To be presented in Spanish.)
Quantifying nonverbal communicative behavior in face-to-face human dialogues
Authors: Skhiri Mustapha , Cerrato Loredana
Published: October 2013

The referred study is based on the assumption that understanding how humans use nonverbal behavior in dialogues can be very useful in the design of more natural-looking animated talking heads. The goal of the study is twofold: (1) to explore how people use specific facial expressions and head movements to serve important dialogue functions, and (2) to show evidence that it is possible to measure and quantify the entity of these movements with the Qualisys MacReflex motion tracking system. Naturally elicited dialogues between humans have been analyzed with focus on the attention on those nonverbal behaviors that serve the very relevant functions of regulating the conversational flux (i.e., turn taking) and producing information about the state of communication (i.e., feedback). The results show that eyebrow raising, head nods, and head shakes are typical signals involved during the exchange of speaking turns, as well as in the production and elicitation of feedback. These movements can be easily measured and quantified, and this measure can be implemented in animated talking heads.
Automatic target recognition in acoustics: An overview
Authors: Sacha John R.
Published: October 2013

Automatic target recognition (ATR) constitutes one of the major uses for acoustical signal processing. ATR is employed in manned systems for operator workload reduction and performance improvement, as well as in autonomous applications. An overview of some of the major components involved in the architecture of such systems is provided. Feature extraction is the most critical step of ATR and is necessarily application specific. Generic feature selection and ranking methods are presented, including heuristic search and information–theoretic measures. Basic pattern recognition definitions and techniques are reviewed. Commonly used classification paradigms include classical statistical formulations, both parametric and nonparametric, and neural nets; support vector machines and nonmetric methods such as decision forests are some alternative techniques that have received recent attention. A few practical issues often encountered when constructing recognition systems, including training data requirements, ground truth labeling, and performance evaluation methodologies and metrics, are also addressed.
Advances in algorithm fusion for automated sea mine detection and classification
Authors: Dobeck Gerald J. , Cobb J. Tory
Published: October 2013

Along with other sensors, the Navy uses high-resolution sonar to detect and classify sea mines in mine-hunting operations. Scientists and engineers have devoted substantial effort to the development of automated detection and classification (D/C) algorithms for these high-resolution systems. Several factors spurred these efforts, including: (1) aids for operators to reduce work overload; (2) more optimal use of all available data; and (3) the introduction of unmanned minehunting systems. The environments where sea mines are typically laid (harbor areas, shipping lanes, and the littorals) give rise to many false alarms caused by natural, biologic, and manmade clutter. The objective of the automated D/C algorithms is to eliminate most of these false alarms while maintaining a very high probability of mine detection and classification (PdPc). In recent years, the benefits of fusing the outputs of multiple D/C algorithms (Algorithm Fusion) have been studied. To date, the results have been remarkable, including reliable robustness to new environments. In this paper a brief history of existing Algorithm Fusion technology and some techniques recently used to improve performance are presented. An exploration of new developments is presented in conclusion.
Acoustic propagation in the coastal environment of the Florida Straits—recent experimental results
Authors: Deferrari Harry , Williams Neil , Nguyen Hien
Published: October 2013

An autonomous source was moored at ranges of 10 and then 20 km from a vertical receiver array with 32 elements in a depth of 145 m of water. M sequences were transmitted for 28 days at six center frequencies from 100 to 3200 in one octave increments. Arrivals and paths are identified with models and then fluctuation statistics, coherence, and predictability are examined in a parameter space of frequency, range, and receiver depth. A group of refracted-bottom-reflected (RBR) modes/rays has nearly equal group velocities and tends to focus in time and depth forming intense arrivals especially at the depth of the transmitted. A second group of surface reflected bottom reflected (SRBR) modes produce arrivals that fan out in time. Coastal areas inside western boundary currents have exceptionally variable sound speed fields owing to dynamical effects such as meanders, shelf waves, eddies, coastal upwelling and energetic internal waves and tides. Sound speed fluctuations are observed to be an order greater than the deep ocean. Very large changes in mean sound speed profiles and extreme gradients occur at subinertial periods. Also, potential energy of the internal wave field varies with the same longer periods as do statistical properties of observed acoustic signals.
Horizontal wavenumbers from a fluctuating waveguide
Authors: Field Robert , Newcomb Joal , Showalter James , George Jacob
Published: October 2013

From September 26 through November 2, 2000 the U.S. and Japan conducted a joint experiment in the New Jersey Bight off the U.S. East Coast. The experiment was conducted in three legs, the first of which investigated low frequency acoustic fluctuations in the presence of a shelf/slope front and internal waves. Legs II and III investigated acoustic scattering at 5.5 kHz and methods of acoustic inversion for sea floor properties, respectively. In this paper we will focus on the experimental results of leg I and discuss implications for doing acoustic inversions in a strongly fluctuating medium. Theoretical results based on the Pekeris waveguide show that a large-scale, slowly fluctuating sound speed can cause rapid fluctuations in the acoustic field in the form of acoustic harmonics. Pekeris waveguide calculations are compared with a subset of the acoustic data. Theoretical results and parabolic equation simulations are used to show the impact of the water column sound speed on the horizontal wavenumbers and hence on acoustic inversion methods that utilize them. [Work supported by ONR, Program Element PE62435N.]
A broadband ocean sediment acoustics model for signal processing applications
Authors: Chotiros Nicholas P. , Isakson Marcia J.
Published: October 2013

It has been shown that fluid and viscoelastic solid approximations cannot accommodate the observed sound speed dispersion and enhanced reflection loss over sandy shallow water sediments. A plausible poroelastic model has been developed for the high frequency band (>50 kHz) using measurements from several sources. This model, with constant coefficients, is unable to track the observed sound speed dispersion at lower frequencies. It is hypothesized that one parameter, the frame bulk modulus, varies with frequency in a relaxation process associated with squirt flow at the grain–grain contact. This hypothesis has the potential to be a critical component in broadband acoustic models of granular ocean sediments. It will link measurements at high frequencies to propagation modeling at low frequencies, provide accurate, physics based, models of propagation loss, and a means to invert for bottom properties over a broad range of frequencies. [Work supported by ONR, Undersea Signal Processing.]
Sparse acoustic gridding in an azimuthally complex environment
Authors: Rike Erik R. , DelBalzo Donald R.
Published: October 2013

A very sparse acoustic gridding method, the Objective Grid/Radials using Environmentally sensitive Selection (OGRES) algorithm, for sonar system performance predictions was reported for a realistic synthetic square annular ridge environment [E. R. Rike and D. R. DelBalzo, Proc. 6th ECUA (2002)] and a complex real-world environment in the Sea of Japan [E. R. Rike and D. R. DelBalzo, Proc. IEEE Oceans (2002)]. The sparse OGRES grids significantly outperformed uniform grids (matched for calculation time) in accuracy. Both cases were based on ground-truth uniform grids high in spatial density, but moderate-to-low in azimuthal density. The present study is a very sparse gridding of a realistic synthetic spiral ridge based on a ground-truth uniform grid high in both spatial and azimuthal density. This analysis extends the previous work by exploring the relationship between the spatial curvature of bathymetric features and optimal azimuthal grid density. The trade-offs between calculation time and accuracy of sonar system predictions for both very sparse and uniform grids are discussed. [Work supported by ONR/NRL under PE 62435N.]