bluuebunny/crossref_metadata_2025 huggingface dataset:
Reducing the sound emission of TRANSRAPID by decreasing the aerodynamic sound sources
Authors: Schmitz Klaus-Peter , Runde Hans-Gerd
Published: July 2002

With the support of the BMBF and in commission of the MVP, the IABG pursues the TRANSRAPID test facility. The TRANSRAPID vehicle TR07 started in 1989 with a very low sound emission. A passby level of 94 dB was measured in a distance of 25 m (free propagation and without soundshields) and a speed of 400 km/h. First investigations with a microphone array pointed out a very calm upper part of the car body and some sound sources in the region of the bow as well as in the region of the levitation and guidance system (LGS). These results showed the low importance of the noise emissions caused by the turbulent boundary layer (TBL) as there was no sound emission found in the upper part of the vehicle. So the sound emission seemed to be caused by aerodynamic sound sources coming from discontinuities of the shape. To reduce the sound emission especially the shape of the bow was modified, the design of the covering plates of the LGS was optimized, and new inner coverings between the vehicle and the guideway were developed. All these changes have decreased the sound emission of the TR07; today a pass-by level of 88.5 dB is realized.
Free-field response of sound level meters
Authors: Banget-Mossaz Gaston , Durocher Jean-Noël
Published: July 2002

Within specified ranges, sound level meters should be able to measure a frequency-weighted sound pressure level at a given point, regardless of the direction of sound incidence and the frequency of the sound. Acoustical characteristics of sound level meters are specified in draft IEC 61672 as a function of frequency for plane progressive sound waves of steady sinusoidal signals under free-field conditions. The free-field frequency response of a microphone in a specified reference direction is modified by (a) sound reflected from the case of the sound level meter, (b) the apparatus that supports the instrument under test, and (c) the frequency response of the system that processes the signals from the microphone. From the simulated influence of sound waves reflected from an instrument equipped with a microphone having a FLAT frequency response, it is shown that linear frequency steps should be preferred to logarithmic frequency steps for evaluation of reflection effects. Test signal frequencies are calculated from the quotient of the speed of sound by the distances from the reflection point on the case to the microphone reference point. Measurement results are shown and comments noted regarding the procedures in the proposed standard for testing acoustical characteristics.
Frequency weightings and responses
Authors: Hedegaard Peter
Published: July 2002

The first frequency weightings for sound-level meters were described in the 1940s in American and German standards. The well-known A, B, and C frequency weightings were originally designed to match the equal loudness contours described by Fletcher and Munson. These frequency weightings were realizable by simple passive electronic circuits. Even though the frequency weightings in proposed IEC 61672 are unchanged, their use is changed. Frequency weightings are now defined by their poles and zeros, and may be calculated to any degree of precison. Since 1961, other frequency weightings have been standardized for special purposes but are largely obsolete, along with the B weighting. Tolerance limits are discussed for the different performance classes and their changes from 1961 to the present. Wide tolerance limits for the high- and low-frequency ends of the frequency range are necessary to allow for different microphone sizes, manufacturing tolerances, and unattended operation outdoors. The wide limits may cause differences in measurement of sound levels if the sound has strong high- or low-frequency components. Some unsuccessful attempts to solve the problem are discussed. The new Z-weighting, FLAT frequency response, and the use of C-weighting for measurements of peak sound-pressure level, are discussed.
Predicting the effects of blade geometry on rotor–stator interaction noise
Authors: Peake Nigel
Published: July 2002

One of the principal noise generation mechanisms in modern high-bypass turbofan engines corresponds to the interaction between convected vortical disturbances from a rotor and a downstream stator row. In this talk, a method for predicting this noise will be described, which takes full account of the effects of the stator blade geometry. These effects include the distortion of the incident disturbances by the nonuniform flow around the stator, noise generation by flow sources close to the blade leading edges, and refraction as the noise propagates back upstream. The approach adopted involves the use of asymptotic analysis based on the realistic limits of large reduced frequency and small, but nonzero, blade camber and thickness. A new feature of this work concerns the behavior of the noise near modal cutoff. Previous ray-theoretic solutions diverge at these points, but a uniformly valid solution will be presented here which overcomes this difficulty. This makes it possible to use these techniques to predict the response of the stator to a whole turbulence spectrum, and not just to a single gust.
A cut-off attenuator with perforated plate in a centrifugal fan and its aeroacoustic experiments
Authors: Lu Yadong , Tian Jing
Published: July 2002

A new type of low-noise centrifugal fan with a special cut-off attenuator with a perforated plate has been invented and designed. The principle of noise suppression with this cut-off attenuator and its structure characteristics is introduced in this paper. A series of experimental comparisons of this type of centrifugal fan with a prototype centrifugal fan have been carried out. The experimental results show that the A-weighted noise level of a prototype fan without a cut-off attenuator with a perforated plate is 93 dB. As a result of using this special attenuator, the noise level of a centrifugal fan is decreased to 88.3 dB. Thus, sound attenuation due to this attenuator is 4.7 dB. In addition, decreasing the airflow and keeping the rotating velocity of the fan unchangeable, noise reduction would be 3.5 to 5.0 dB. It can be shown from the spectrum measurements that by using this cut-off attenuator with a perforated plate, not only the discrete frequency noise components of blade passing frequency and its harmonics, which predominate the sound level of a centrigual fan, can be attenuated, but also the broadband noise can be suppressed. Meanwhile, the aerodynamic performance of a centrifugal fan has not been greatly changed. [Work suported by NSFC.]
Acoustic energy estimates in inhomogeneous moving media
Authors: Farassat F. , Farris Mark
Published: July 2002

In ducted fan engine noise research, there is a need for defining a simple and easy to use acoustic energy conservation law to help in quantification of noise-control techniques. There is a well-known conservation law relating acoustic energy and acoustic energy flux in the case of an isentropic irrotational flow. Several different approaches have been taken to generalize this conservation law. For example, Morfey [J. Sound Vib. (1971)] finds an identity by separating out the irrotational part of the perturbed flow. Myers [J. Fluid Mech. (1986)] is able to find a series of indentities by observing an algebraic relationship between the basic conservation of energy equation for a background flow and the underlying equations of motion. In an approximate sense, this algebraic relationship is preserved under perturbation. A third approach which seems to have not been pursued in the literature is a result known as Noether’s theorem. There is a Lagrangian formulation for the Euler equation of fluid mechanics [M. E. Taylor, PDE’s, Vol. 3, 1996]. Noether’s theorem says that any group action that leaves the Lagrangian action invariant leads to a conserved quantity. This presentation will include a survey of current results regarding acoustic energy and preliminary results on the symmetries of the Lagrangian.
A reverse flow theorem and acoustic reciprocity in compressible potential flows
Authors: Eversman Walter
Published: July 2002

A reverse flow reciprocity theorem for acoustic propagation in compressible potential flows in nonuniform ducts is derived in terms of steady flow and acoustic velocity potentials. A similar result for a quasi-one-dimensional formulation is also presented. The theorem is specialized to investigate the characteristics of the acoustic modal scattering matrices for propagation in transition sections for axially symmetric ducts in the presence of high-speed flow. It is found that reflection coefficients for modes incident at the source plane in direct and reverse flow are related, and in particular, that diagonal elements in the direct and reverse flow reflection coefficient matrices are invariant. The same result is true for modes incident at the termination plane. The transmission coefficient matrix for modes incident at the source plane in direct flow is related to the transmission coefficient matrix for modes incident at the termination plane in reverse flow. Simpler but consistent results are demonstrated in the quasi-one-dimensional case where it is shown that the acoustic power transmission coefficient is invariant to the direction of flow. Numerical results from finite-element solutions are used to support the theoretical conclusions.
Probing and stabilization of interfaces with acoustic radiation pressure: Application to bubbles and capillary bridges
Authors: Thiessen David B. , Marr-Lyon Mark J. , Marston Philip L. , Asaki Thomas J. , Morse Scot F.
Published: July 2002

Acoustic radiation pressure at interfaces can be used as a probe of dynamics and as a method of suppressing instability. Examples include the probing of the interfacial rheology of acoustically levitated bubbles in water in the presence of surfactants which are either insoluble [T. J. Asaki, D. B. Thiessen, and P. L. Marston, Phys. Rev. Lett. 75, 2686–2689 and 4336 (1995)] or soluble [T. J. Asaki and P. L. Marston, J. Acoust. Soc. Am. 102, 3372–3377 (1997)] in water. Modulated radiation pressure was also used to probe the dynamics of liquid bridges of oil surrounded by water [S. F. Morse, D. B. Thiessen, and P. L. Marston, Phys. Fluids 8, 3–5 (1996)] and to suppress the breakup of cylindrical bridges having lengths significantly beyond the natural Rayleigh–Plateau limiting length [M. J. Marr-Lyon, D. B. Thiessen, and P. L. Marston, J. Fluid Mech. 351, 345–357 (1997)]. In all of these examples the interfacial tension is significant for the normal modes of interest. Normal-mode frequencies can be significantly modified by using optical information for feedback control of the spatial distribution of the radiation pressure. [Work supported by the Office of Naval Research and by NASA.]
A theoretical investigation of sound transmission through a layered periodic structure in water
Authors: Moustafa Aly H. A.
Published: July 2002

This paper presents a theoretical study of sonic passbands and stopbands that occur with a periodic one-dimensional structure of plates in water. The study uses the characteristic matrix method and considers a periodic structure of some materials that are characterized by a constant density and phase speed: perspex, aluminum, and steel. The computed results show the variation of the pressure transmission coefficient with frequency for various systems of plate configurations in water, and takes into account the attenuation of the plate material. The pattern of the results indicates clearly that sound propagation can be prohibited at certain frequencies while free propagation is allowed at others. The results compare very well with published data that are obtained using other methods.
Physiological studies and neural mechanisms of echo suppression in the inferior colliculus of the cat
Authors: Litovsky Ruth Y. , Delgutte Bertrand , Yin Tom C. T.
Published: July 2002

Neurons in the inferior colliculus (IC) are sensitive to stimulus direction in free field and to individual directional cues, such as interaural disparities in time (ITD) and level (ILD). Our single-unit recordings in the IC of anesthetized cats explored neural mechanisms in the IC that might mediate echo suppression, using free-field and virtual space (VS) stimuli. Click pairs were used to simulate a sound source (lead) followed by a single reflection (lag) at delays of 1–100 ms. In both free field and VS, correlates of psychophysical echo suppression were found in the responses of most IC neurons. In many neurons, the suppression was directionally dependent in a way that could be linearly predicted from the response to the lead, with the most effective lead locations giving maximum suppression. Some neurons showed a decoupling between echo suppression and the directional response to the lead, suggesting that excitation and suppression are mediated by different neural mechanisms. Further decoupling was achieved by selectively holding constant some directional cues such as ITD and ILD in the VS stimuli. Results suggest that neural suppression in the IC depends on specific directional cues and is therefore a likely correlate of echo suppression in spatial hearing. [Work supported by NIH.]
Speech perception in telephone situations under ambient noise conditions
Authors: Hempel Thomas , Möller Sebastian , Jekosch Ute , Meunier Sabine , Regal Xavier
Published: July 2002

In a telephone situation, the partners of a conversation generally experience ambient noise because they are located in acoustically realistic environments. The impact of the ambient noise on the quality of a telecommunication service is a double one: on the one hand, the noise deteriorates the acoustical speech signal either directly or via the sidetone path of a telephone handset, thus leading to lower intelligibility and overall quality. On the other hand, it distracts the listener’s attention. Models which try to predict the effect of ambient noise for telephone network planning purposes in general only regard a diffuse room noise level. However, localization of possible ambient noise sources can help the listener to concentrate on the conversation rather than being distracted by the noise. A conversation test has been carried out in order to investigate the effect of ambient noise in both handset and hands-free telephone situations. Noise was either diffuse or localized at different positions relative to the listener. The test results are compared to intelligibility measures for masked speech given in literature. The investigation can help to quantify the impact of localized ambient noise and gives information to design realistic ambient noise environments for telephone or teleconferencing scenarios.
Notched-noise measures of frequency resolution in children: What acoustic cues are available at children’s threshold levels?
Authors: Allen Prudence , Korpela Louise
Published: July 2002

It has been suggested that young children’s masked detection thresholds are obtained at signal-to-noise ratios for which the signals produce changes in multiple features of the masker [Allen et al., J. Acoust. Soc. Am. (in press)]. In this study 500-, 2000-, and 4000-Hz tonal signals were added to flat- and notched-spectrum noise maskers at signal-to-noise ratios corresponding to children’s threshold levels measured in a previous study of frequency resolution [Allen et al., J. Speech Hear. Res. 32, 317–324 (1989)]. The magnitudes of level, frequency, and periodicity cues were determined, assuming several degrees of frequency selective filtering. Results suggested that for all signal plus masker combinations, independent of frequency and masker spectrum, children’s thresholds had been obtained at signal-to-noise ratios for which little or no frequency selectivity would have been required to produce potentially discriminable level differences [Jensen and Neff, Psychol. Sci. 4, 1-4-107] of approximately 6 dB, which were often accompanied by average frequency and/or periodicity differences. Age-related changes in frequency selectivity therefore may not be required to explain children’s higher masked detection thresholds as they are obtained at signal-to-noise ratios where multiple cues are present. [Work supported by NSERC.]
Effects of practice on performance for different masking tasks
Authors: Delahaye Roel , Fantini Deb A. , Meddis Ray
Published: July 2002

The present study investigates the effects of long-term practice on masked signal detection with experienced listeners. The on-frequency masker (OFM) consists of a 700-Hz pure tone, which is sinusoidally amplitude-modulated at a 10-Hz rate. Similar flanking components can be placed at 300, 400, 500, 900, 1000, and 1100 Hz. The signal is composed of three, 50-ms, 700-Hz tone bursts that are placed in the dips of the OFM envelope. The flanking component envelope phases were either in phase (comodulated condition) or 180° out of phase (uncorrelated condition) with the OFM. In the reference condition only the OFM was present. Two subjects gained experience (20 h) before the first data collection session took place. After extended practice (40 h) a second session took place. The results of the first session show a release from masking in the comodulated condition with regard to the reference condition (CMR R-C) and the uncorrelated condition (CMR U-C). Results from the second session show that reference thresholds decrease more than the comodulated and uncorrelated thresholds. This resulted in a reduction or disappearance of CMR(R-C) but a preservation of CMR(U-C). Thus the amount of practice is inversely related to the magnitude of CMR(R-C), while CMR(U-C) is independent of practice.
Acoustic weapons? Effects of very high sound levels
Authors: Altmann Jürgen
Published: July 2002

Mainly in the context of so-called nonlethal weapons, acoustic weapons are under research and development in a few countries. Reliable information is scarce; allegations, mainly in the military press, have attributed drastic effects in particular to infrasound. Humans would suffer from disorientation and nausea up to vomiting and uncontrolled defecation, however without lasting damage. For damage to material, embrittlement of metals and even localized earthquakes have been mentioned. A literature review was done to investigate such claims and find out other effects of strong sound in the infrasound and low-frequency audio, high-frequency audio, and ultrasound ranges, as well as of blast waves. Whereas the infrasound allegations cannot be confirmed, marked nonauditory effects exist in the low audio range at levels above 150 dB. At higher frequencies and with blast waves, mainly in the auditory system is affected. With increasing level, annoyance, pain, and permanent damage ensue; protection is possible, however. Very strong blast waves can cause lung rupture and death. Because application scenarios range from protection of fixed installations via riot control to armed conflict, potential regulation should not be restricted to the law of warfare, and preventive limits will need to be detailed.
Active control of harmonic and random sound transmission in a structural-acoustic system
Authors: Brennan M. J. , Kim S. M.
Published: July 2002

The active control of sound transmission in structural-acoustic coupled systems has been of interest in recent years. Two particular problems that have received attention are the active control of harmonic sound transmission into aircraft, and control of random noise transmission into the payload bay of space rockets. This paper describes an analytical and experimental investigation to compare the feedforward control of harmonic and random sound transmission into an acoustic cavity. A rectangular enclosure is considered that has five acoustically rigid walls and a flexible plate on the remaining side through which a plane acoustic wave is transmitted into the enclosure. The control systems are designed to reduce the acoustic potential energy inside the enclosure when the incident sound is either harmonic or random. Three control configurations, classified by the type of actuators, are investigated both theoretically and experimentally. They are the use of a structural actuator, the use of an acoustic source, and the simultaneous use of both structural and acoustic actuators. It is shown that the use of both actuators is desirable for the active control of both harmonic and random sound transmission into a coupled structural-acoustic system whose response is governed by plate and cavity controlled modes.
An experimental investigation of a novel active segmented partition for sound transmission control
Authors: Leishman Timothy W. , Tichy Jiri
Published: July 2002

A unique dual diaphragm configuration for individual elements of active segmented partitions has recently been advanced as a practical, efficient means of controlling sound transmission. The configuration incorporates a useful combination of electro-mechano-acoustical actuation and segmentation, enabling effective global control of transmitting surface vibrations. In this work, experimental efforts are reported that explore important active and passive characteristics of such elements while adjoined in a full partition array. Each element is independently and stably controlled using a single-channel filtered-x adaptive controller (single reference input, single error input, single output). Acoustic pressure scans of element cavities and laser vibrometer scans of their transmitting surfaces affirm important properties of this partition construction that have been previously theorized. Other measurements demonstrate that the controlled transmission loss of the lightweight array lies uniformly between 55 and 60 dB over nearly all of the evaluated spectrum (30 to 300 Hz). Such extensive control is generally very difficult to realize over the lower audible frequency spectrum using an extended lightweight structure. Further measurements illustrate that this is comparable to the passive transmission loss of a much more massive benchmark partition of the same cross-sectional area, but constructed of thick, spaced layers of concrete and sand. a)Currently at Crown International, 1718 W. Mishawaka Rd., Elkhart, IN 46515-1000.
A psychoacoustical model of the auditory periphery as the front end for ASR
Authors: Tchorz Jürgen , Kleinschmidt Michael , Kollmeier Birger
Published: July 2002

A psychoacoustical model of the auditory periphery was developed by Dau and others to predict human performance quantitatively in typical spectral and temporal masking experiments. Its main processing stages are basilar membrane filtering for spectral analysis, adaptive envelope compression in each frequency band which enhances changes in the input signal and suppresses steady-state portions, and low-pass filtering of envelope modulations. In the field of speech processing, the model was applied as the front end of an automatic speech recognition (ASR) system [J. Tchorz and B. Kollmeier, J. Acoust. Soc. Am. (submitted)]. Speaker-independent, isolated-digit recognition experiments in different types of noise were carried out to evaluate the robustness of the auditory-based ASR system in adverse conditions. Compared to a standard MFCC front end, the auditory-based preprocessing yielded significantly higher recognition rates in both additive and convolutive noise. Further experiments concentrated on the question, which of the processing stages of the psychoacoustically motivated model are essential for robust representation of speech? The results indicate that the adaptive compression stage and low-pass filtering of envelope modulations contribute most to robustness, whereas the spectral analysis part is less critical.
Some musings on systematic variability and speech recognition
Authors: Cohen Jordan
Published: July 2002

One of the major problems in speech recognition is the inability of trained models to generalize appropriately to channel variations, new speakers, or modified acoustics. The naive observer would believe that a multimillion-parameter system should be sufficient! The difficulty appears to be too many parameters rather than too few. For moderate-sized training corpora, systems learn all of the conditions in the training data rather than generalizing from the exemplars. (For instance, speech recognition algorithms will generally score the speech from a training speaker higher than that from a speaker who was excluded from the set.) One can force the issue by explicitly modeling systematic variation, and then ‘‘normalizing’’ at the front end or in the acoustic model. Two exemplars of this philosophy are Cepstral mean subtraction and vocal tract normalization [Frontiers in Speech Processing 94, LDC96s40, Linguistic Data Consortium (1995)]. In each case a single parameter from a very restrictive model is estimated, and accounting for the variability explicitly improves performance. Concrete examples of these situations are offered, and the impact of this work on future work in automatic speech recognition is discussed.
A study of the supraglottal flow behavior during phonation
Authors: Pelorson Xavier , Vilain Coriandre , Hirschberg Avraham
Published: July 2002

Most physical models of phonation assume the formation of a free jet at the end of the glottis. The precise behavior of this jet is generally overlooked, assuming a symmetrical flow and a total dissipation of the kinetic energy past the glottis. However, asymmetry of the glottal flow (Coanda effect) and false folds effects (reattachment of the jet) have been observed on the basis of quasi-steady flow numerical simulations. In this paper the relevance of these effects on the basis of unsteady in-vitro measurements using pulsating flows through symmetrical models of the glottis is discussed. It is shown that asymmetry of the flow due to the Coanda effect is typical of steady flows but does not have time to be established during unsteady conditions. An extension of this study concerning effects of an asymmetry of the glottis will also be discussed. Concerning the effects of the false folds, many parameters have been studied such as the width of the false folds, and the distance between the true and the false folds. It is shown that for typical phonation configurations the possible effects of the false folds remain very weak. This might not be the case for pathological configurations.
Synthesis of unvoiced speech sounds using an aeroacoustic source model
Authors: Krane Michael , Sinder Daniel , Flanagan James
Published: July 2002

An aeroacoustics-based method for synthesizing unvoiced speech sounds is presented. The source model is that of Howe (1975) for sound generated by flow of vorticity through a nonuniform duct. In this formulation three quantities must be known: the potential flow velocity, which changes according to the duct shape, the vorticity, and the velocity induced by vorticity in the flow and its images in the duct walls. The potential flow velocity magnitude is found using a transmission-line representation of the vocal tract, and the direction is found using a panel method. A model for the generation and evolution of vorticity has been developed in which the vortex shedding frequency is a random variable, and the vortex convects at a constant speed and direction. The computational model was used to simulate three cases: (1) a jet flowing through a pipe constriction, (2) a fricative sound (/s/), and (3) a VCV combination (/a s a/). Simulation results are presented and compared to experimental data, in terms of spectra and listening tests.
Bifurcations at voice onset–offset
Authors: Lucero Jorge C.
Published: July 2002

Oscillation hysteresis is a common phenomenon in cases of flow-induced oscillation. In phonation, it is usually observed as different biomechanical configurations of the vocal fold system between voice onset and offset. From the perspective of the theory of nonlinear dynamical systems, the phenomenon may be described as a combination of a subcritical Hopf bifurcation at which an unstable limit cycle is generated from an equilibrium position of the vocal folds, and a saddle-node bifurcation between limit cycles at which the unstable limit cycle and a stable limit cycle coalesce and cancel each other [J. C. Lucero, J. Sound Vib. (in press)]. This description is investigated using a two-mass model of the vocal folds coupled to a two-tube approximation of the vocal tract. The purpose is to simulate the hysteresis in terms of voice output parameters as airflow, intensity, and fundamental frequency. The simulations show good agreement with available experimental measurements at voice onset–offset [e.g., L. L. Koenig and R. S. McGowan, J. Acoust. Soc. Am. 100, 2689(A) (1996)]. Bifurcation diagrams are derived to illustrate and discuss the results. [Work supported by the CNPq (Brazil).]
A series of experiments exploring the spatial separation of multiple talkers with 3-D audio technology
Authors: McKinley Richard L. , Ericson Mark A. , Nelson W. Todd , Bolia Robert S. , Simpson Brian D.
Published: July 2002

Most auditory displays are monaural and do not take advantage of the capacity of the binaural system for spatial separation of multiple simultaneous talkers. A program of research has been developed at the Air Force Research Laboratory to investigate the enhanced communication effectiveness obtained with 3-D audio displays. Factors that are being manipulated include angular separation, the number of competing messages, and the sex of the target talker. Angular separation has been studied in both the horizontal and median planes, with between two and eight competing messages. The effect of duration of temporal overlap has also been explored. The goal of this research is to discover the maximum number of speech channels that a listener can effectively monitor, and whether this number is augmented by 3-D audio technology. Potential applications of 3-D audio technology for communications separation will be discussed.
A comparison of benchmark acoustic scattering solutions with tank experiments on buried and partially buried spheres
Authors: Lim Raymond , Simpson Harry J.
Published: July 2002

Predicting the scattering response of objects near boundaries is of growing interest in underwater acoustics, especially when these objects are buried. Standard propagation models suggest that burial in fast attenuating ocean bottoms can seriously retard efforts to find such objects with high-frequency imaging sonar because of the need to operate at close range to overcome loss in the backscattered signal strength, resulting in low search rates. However, recent observations of anomalous sound transmission at shallow grazing angles as well as anomalous long-range detections of buried objects call into question standard assumptions made in regard to the modeling of sound transmission through the water/sediment interface. The present work investigates the breakdown of these assumptions for the scattering problem by comparing laboratory measurements conducted in the NRL Shallow Water Laboratory with predictions of exact scattering solutions for simple targets at various degrees of burial. Scattering data collected recently for a 60-cm-diam. spherical steel shell is used in the investigation. A discussion of the agreement between theory and measurements, features of special interest, and the consequences to our understanding of the acoustics of buried objects is given. [Work supported by ONR.]
A review of ocean acoustics in the band 0.001–1.0 Hz
Authors: Stephen Ralph A.
Published: July 2002

The first ultra-low frequency (ULF) (0.001–1 Hz) ambient acoustic noise measurements were made in the 1960s using the Columbia-Point Arena ocean-bottom seismic station. Interest in ULF acoustics was spurred in the mid-1980s by the development of the differential pressure gauge. Also in the mid-1980s the development of high dynamic range, broadband seismometers revolutionized whole earth seismometry. The Global Seismic Network (GSN) set as its goal the uniform distribution of 128 seismic stations over the surface of the globe, to provide high-quality data sets for whole earth tomography and other earth scale problems. To accomplish this goal requires about 20 permanent stations on the deep ocean floor. In the future, the acquisition of long time series of ULF acoustic noise in the deep ocean will become much more convenient and the opportunity will be given to exploit these data sets to study oceanographic and earth processes. The sources of ambient noise in this band include infra-gravity waves, swell interaction at coastlines, currents and turbulence near the seafloor, earthquakes, tsunamis, nonlinear interaction between ocean-surface waves, and air turbulence. Water column reverberations, Stoneley modes in sediment layers, and the effects of propagation from distant sources also modify the pressure spectra. [Work supported by NSF.]
On comparative contributions of natural ultra-low-frequency hydrodynamic wave fields to sound signals’ variability on extended shallow-water traces
Authors: Semenov Andrew G.
Published: July 2002

The ocean’s ultra-low-frequency natural sound variability mechanism, related to sound interaction with extra slow hydrodynamic natural wave processes, gravitational surface, and internal wave, observed in the range of rad/s on long traces, is discussed. In the absence of natural internal waves, the ocean hydrodynamic field here is close to potential, so that its interaction with the sound field is minimal. Estimates are based on the model of ocean hydrodynamic waves representing the refraction index lattice for propagating sound modes. Small angle sound interaction with weak enough surface waves lattice leads to additional mode phase transformation increasing with trace length. Interaction is shown to be especially substantial for low-frequency sound (of frequency less than 100–200 Hz) and very long traces (of length not less than 80–100 km) in a shallow ocean region, where it could be comparable to interaction observed due to internal waves. While for a given sound frequency value and short trace length, sound-hydrodynamic wave fields’ interaction is determined exclusively by lattice refraction factor, for long enough traces, interaction is defined by refraction index and hydrodynamic wavelength product. This parameter variability for surface gravity waves in different ocean regions, based on generalized bottom pressure measurement data, is presented.
Results for tomographic geoacoustic inversion on simulated multiarray data
Authors: Tolstoy A.
Published: July 2002

Only recently have broadband multiarray data been collected which are suitable for tomographic inversion for the estimation of shallow-water geoacoustic parameters. In particular, this unique data set was collected in the Haro Strait region off the west coast of Canada by N. R. Chapman and colleagues. These data have now been simulated and studied in preparation for the tomographic inversion of the actual data, a very difficult problem involving hundreds of parameters. Results to be discussed here concern these simulated data and recent successes in their inversion for the estimation of such properties as sediment sound-speeds and densities over a region where such parameters vary as a function of range, azimuth, and depth. [Work supported by ONR.]
The experimental demonstration of selective mode excitation in a shallow-water environment
Authors: Preisig James C. , Buck John R. , Johnson Mark , Dou Hu
Published: July 2002

The ability to selectively excite single acoustic modes in a shallow-water environment is demonstrated and analyzed. Using vertical source and receiving arrays and the feedback control techniques based on those in Buck et al. [IEEE J. Oceanic Eng. 22, 281–291 (1997)], acoustic modes 1–3 were excited sequentially for periods up to 20 min with a fidelity of approximately 60 dB for mode 1 to 25 dB for mode 3. The filtering of the received signal used in the feedback loop by projection into the subspace spanned by the dominant system modes was found to significantly improve the system’s performance. This result will be explained and analyzed. The requirement for phase stabilization of the sampled received signal is also presented. The ability demonstrated here for high fidelity selectively controlled modal content of a transmitted acoustic signal offers potential for new methods for acoustic tomography and for studying underwater acoustic propagation. [Work supported by ONR Ocean Acoustics.]
AMOC: Preliminary results
Authors: Sagen Hanne , Johannessen Ola M. , Hobaek Halvor , Tjoetta Sigve
Published: July 2002

The scope of this presentation is to give preliminary results of the sensitivity study of sound propagation in the Arctic including the Fram Strait using propagation loss observations and existing acoustic propagation models. Special attention to the consequences for design of future acoustic monitoring concepts will be made. One of the important consequences of the preliminary results is the conflict between the optimum transmission frequency and the optimum frequency for retrieving information about the ice cover. If an experiment is designed to propagate over long distances at a relatively low frequency, then the reflection from a ‘‘smooth’’ ice cover is total and there will be no information about the ice thickness or internal properties of the ice cover. The attenuation due to the ice will be dominated by scattering from rough surface (large facets). So this type of experiment will give information about the averaged temperature through the travel times, while the ice information will be limited. Therefore two concepts have to be considered: one to measure ice parameters, which will need signals composed by appropriate frequencies, and another concept to measure averaged temperature, which needs low frequencies insensitive to ice cover.
Large aperture 1-3 composite longitudinal resonator (tonpilz) transducer
Authors: Allen Charles W. , Hughes W. Jack , Van Tol David J.
Published: July 2002

A large aperture underwater transducer utilizing 1-3 piezoelectric composite material that operates as a longitudinally resonant (tonpilz) device with a resonance frequency of 45 kHz has been fabricated and tested. The transducer consists of continuous plates of steel and aluminum that act as the rear and front masses, respectively, and a motor section of continuous 1-3 PZT-5H composite. The active radiating face (aluminum front mass) has dimensions of 6.3 in. (4.7 wavelengths) and 3.9 in. (2.9 wavelengths). Conventional transducers or arrays with radiating faces that have dimensions of over one wavelength require dicing or separate elements to reduce the effect of lateral modes in the components. The tonpilz transducer utilizing 1-3 composite material requires no dicing of any of its components, so fabrication of large aperture arrays that only require a single forward-looking narrow beam are greatly simplified. The measured acoustic responses versus frequency and angle demonstrate that there are no extraneous modes that affect the acoustic performance of the transducer and they compare well to theory for a one-dimensional longitudinal resonator. [Work sponsored by the Scripps Institution of Oceanography, University of California, San Diego.]
Evidence of nonlinear losses in electrostrictive materials
Authors: Robinson Harold C.
Published: July 2002

It has been well established that the equivalent piezoelectric properties used to model the large signal properties of electrostrictive materials such as lead magnesium niobate/lead titanate (PMN/PT) are functions of the operating conditions. Since these materials are nonlinear, DC bias field, AC drive field, prestress level, and temperature all affect the piezoelectric constants and the coupling factor. However, in transducer models based on equivalent circuits, the dielectric loss is represented as a constant under all of these conditions. Moreover, the dielectric loss tangent typically used in these models must either be inferred from transducer data or be measured under conditions (such as using an RLC meter) that do not represent the actual conditions of operation. This paper will present data, from both material and transducer measurements, suggesting that the losses in electrostrictive materials are strong functions of the temperature, drive field, and compressive prestress. Possible approaches to obtaining improved loss measures from material measurements will be discussed.
On the acoustic of inverse sinusoidal and catenoidal nozzles
Authors: Campos L. M. B. C. , Lau F. J. P.
Published: July 2002

The acoustic wave equation, for quasi-one-dimensional propagation, along a duct of varying cross section, containing a low Mach number mean flow, is obtained using as variables either the potential or the velocity; the ray approximation, which holds only for wavelength short compared with the lengthscales or variation of cross section and mean flow velocity, is used as a factor to reduce the wave equation to a Schrödinger form. It is shown that the latter, reduced from, for the potential, is the most convenient to study the acoustics of catenoidal and sinusoidal nozzles; it is found that these inherit respectively the filtering properties of catenoidal horns, and transparency properties of sinusoidal horns. This approach also applies to the exponential nozzle whereas, for the Gaussian nozzle, the sound field can be expressed in terms of Hermite functions, using a semi-reduced form of the wave equations. The exact solutions of the nozzle wave equation, for the four families of ducts, are plotted as amplitude and phase versus distance, for several combinations of frequency and low Mach number. The nozzle families considered include the catenoidal in the sinusoidal and exponential and the first six eigenfunctions of Gaussian nozzles. The acoustics of inverse sinusoidal and catenoidal nozzles can also be studied, without use of the preceding transformations, via the solution of modified Mathieu equation.
A spectral layer model in speech and music perception
Authors: Deutsch Werner A.
Published: July 2002

According to a model of the peripheral hearing system establishing a masking and overmasking paradigm, speech and music signals have been split into several spectral layers by means of spectral subtraction. Spectral components below the so-called psychoacoustical irrelevance threshold are eliminated in a first processing step. The resulting signal, which exhibits no noticeable difference to the unprocessed one, is fed into an overmasking procedure applying flattened hearing masking functions. This action implements a kind of a spectral peak detector device. Consequently, the discrimination between spectral peaks (auditory figure) and background components, especially in reverberant environments, is improved. As a result, different audible signal parts according to different spectral layers are obtained, separating spectral peaks and weaker spectral components. In a certain range, both parts of speech and music signals are equally intelligible. In music, leading voices can be extracted and separated from the orchestra sound. Recent research indicates that this type of figure-background discrimination based on the dynamic characteristics of the acoustic signal only can exhibit several perceptually relevant features in listeners with normal hearing as well as in subjects with sensineural hearing impairment.
Implementation issues in the development of a real-time, Windows-based system to study spatial hearing
Authors: Miller Joel D. , Abel Jonathan S. , Wenzel Elizabeth M.
Published: July 2002

As a tool to study spatial hearing, a customizable real-time 3-D-audio and virtual acoustic environment rendering system, Sound Laboratory (SLab), was developed. Design goals for this system included flexibility, extensibility, maintainability, and a fast development cycle per experiment. SLab is a Windows application composed of five distinct software layers: Application programming interface (API), 3D projection, signal flow translation, signal routing and processing, and digital signal processing library. The acoustic scene, including listener, source, and environment characteristics, is specified in the API layer and translated in the 3-D projection layer to geometric quantities, such as range and arrival angle, for each path rendered between the source and listener. The geometry and head-related transfer function coefficients are rendered using a set of parallel finite impulse response filters and delay lines built on Intel’s Signal Processing Library. This rendering architecture provides the flexibility and the extensibility required for psychoacoustic experimentation, while the modular nature of the programming interface ensures maintainability and quick development of new experiments. The psychoacoustic parameters to be manipulated, the signal flow diagram, and the five software layers are discussed in detail, and various implementation issues are examined.
Use of bidirectional reflectance distribution function in a particle tracing method
Authors: Dindart Jean-François , Embrechts Jean-Jacques , Sémidor Catherine
Published: July 2002

A previous article (DAFX98, Barcelone) described the particle tracing method, which can be viewed as an efficient alternative to distributed ray-tracing. This method is intended to offer a more accurate simulation of sound reflection than the classical ones. The reflection model in the particle tracing method is based on the bidirectional reflectance distribution function (brdf) which describes the directional distribution of reflected sound. As all mathematical framework, this function can be implemented in different ways. A model derived from physical acoustics and described in a recent paper by Embrechts (ASA/ICA Seattle 98) is used. Comparisons of computed and measured results are used to validate the method in a simple rectangular classroom of the Ecole d’architecture et de paysage de Bordeaux. The reflection parameters’ effects are made predominant in these experiments. The room acoustics parameters of this classroom are measured with different treatments of one wall (concerning its sound reflective properties). The reflective properties of the walls are also modelized with the Embrechts model and the room-acoustics criteria are computed in the same configurations. The validation on this simple room is finally discussed on the basis of the computed and measured results. a)National Fund for Scientific Research.
The harmonics-to-noise ratio applied to dog barks
Authors: Riede Tobias , Tembrock Guenter , Herzel Hanspeter , Brunnberg Leo
Published: July 2002

In this paper a parameter is introduced to animal bioacoustics which is commonly used in human phoniatrics—the harmonics-to-noise ratio (HNR). It is a ratio of the acoustic energy of the harmonic components to that of the noise. Barks of domestic dogs are short, loud, and broadband utterances. The rationale for this method is that the acoustic wave of a barking call is characterized by the ratio of harmonic and noise components. Fifty barks from each of ten adult dogs (dachshound) uttered during a mild threat were considered. The HNR is estimated as follows: a 50-ms time segment from the middle of the call is Fourier transformed (sampling frequency 30 kHz, 512-point Hanning window). A ten-point moving average curve was calculated from the spectrum. The maximum difference between the spectrum and the moving average is used as HNR. Two test persons were asked to arrange the barks of the ten dogs according to their personal perception of hoarseness. The measured HNR agrees perfectly with the perceived ordering. In one dog’s barks the HNR were measured twice to test the robustness of the procedure. A paired t-test showed no significant differences between the two samples. This exemplifies that the relatively simple HNR is valuable in animal bioacoustics.