bluuebunny/crossref_metadata_2025 huggingface dataset:
Experiments and computations of noise reduction by barriers
Authors: Salomons Erik M. , Beeks A. A. F. M.
Published: July 2002

Experimental and theoretical investigations are presented of the performance of noise barriers in the open air. The experimental work includes full-scale measurements on barriers and scale-model measurements on a barrier in a wind tunnel. The wind tunnel experiments indicate that meteorology has large effects on barrier attenuation. The results of the wind tunnel experiments are compared with results of numerical computations with the parabolic equation method. Good agreement with the experimental data is obtained only if wind speed profiles as measured in the wind tunnel are used in the computations. It is concluded that wind speed gradients induced by the barrier itself reduce the barrier attenuation considerably: reductions as large as 10 dB are found. The use of aerodynamically optimized barriers is proposed in order to minimize this reduction of barrier attenuation.
Vibration assessment in buildings from the underground trains
Authors: Tsukernikov Illya E. , Rybak Samuel A. , Kostarev Stanislav A. , Nekrasov Igor A.
Published: July 2002

The method approved by the Moscow government in 1997, which has been implemented by the regulation ‘‘Guideline for assessment of vibration from underground train operation in dwellings and administrative buildings’’ [1998, Moscow, TIMR], is considered. The method is based on the control of nonsteady vibration by two normative parameters: maximum and equivalent values of rms acceleration, and assessment is made as for frequency-weighted quantities, and so in octave frequency bands at the range 22, 5–90 Hz. The method makes it possible to determine the values of maximum effective rms acceleration during the rating time stated. The procedure of equivalent rms acceleration calculation is analogous to one of DIN 4150, part 2, and the train operation schedules during the rating times are used. Results of testing the method at one of the Moscow Metro lines are discussed. Recommendation on the enlargement of the frequency range controlled and on the correction of the criterion of the frequency-weighted rms acceleration assessments are proposed.
The last 100 years of magazine articles on noise
Authors: Bruce Robert D.
Published: July 2002

Articles on ‘‘noise’’ that were referenced in the Reader’s Guide to Literature (and similar guides) from 1890 to 1990 were reviewed recently. These articles present an interesting history of the spread of knowledge about noise-induced hearing loss. By determining the readership of each magazine and assuming that the information in that magazine was representative of the knowledge of that readership, it is possible to attribute certain levels of knowledge chronologically about noise-induced hearing loss to different segments of society. For example, the first group to learn about noise-induced hearing loss consisted of doctors who directly diagnosed hearing losses. Next to learn were the doctors who read journal articles written by the first doctors. Soon after the medical articles were published, the information propagated to the scientific/engineering and financial communities, and they added their own particular insights on the subject. Considerably later, magazines read by workers and their families published information on noise-induced hearing loss, but seldom with sufficient information to enable workers to protect their own hearing.
Hearing disorders from combined exposures to noise and solvents: A review
Authors: Morata Thais C.
Published: July 2002

Not until recently was solvent exposure found to have a significantly adverse effect on hearing. In some cases the risk can be comparable to the most potent and best-known occupational factor, excessive noise. In animals, combined exposure to solvents and noise can result in an increased loss of auditory function beyond the simple addition of the effects of each agent. An increased prevalence of hearing loss has been reported in humans as well. The balance function can also be affected by certain industrial chemicals. The importance of these observations becomes clear when the magnitude of exposed populations and the number of potentially hazardous chemicals found in work environments are taken into account. The need for research in this area is further heightened by the fact that there are no guidelines or standards for combined exposures to chemical and physical agents. The present paper reviews the effects of combined exposures to solvents and noise on hearing function and examines study designs, hearing assessment alternatives, and strategies for the analysis of these effects.
Measurements of acoustical emissions from large-calibre weapons
Authors: Korff Christa
Published: July 2002

Extreme pressure waves and a wide frequency band characterize the environment of a firing large-calibre weapon. Here the usual way to gain acoustical source data therefore fails. For this reason an international working group has developed a standard procedure to obtain the sound emission from such weapons. This will be described. The German Ministry of Defense adopted the method to collect the basic source data for a computer model to calculate the emissions of noise caused by the shooting artillery and explosions in the vicinity of military training areas. It supported the experiments that have been carried out in the last years at the Meppen proving ground in northern Germany. There the environmental conditions defined by the working group are met rather well. First experiences with the application of the standard procedure and some results of the model calculations are reported.
Results from Norwegian sound propagation experiments
Authors: Hole Lars R.
Published: July 2002

This paper describes four sound propagation experiments that were undertaken in Norway from 1994 to 1996. Trials were carried out at Finnskogen; a long-range site, and at Haslemoen; a short-range site. Summer and winter conditions were investigated at both sites. Two similar experiments were carried out at each site with and without snow cover. At Finnskogen, an experimental array of 24 by 24 km was set up in a terrain with smooth hills, primarily covered by forest. At Haslemoen, propagation was investigated out to 1.5 km above a uniform forest and a uniform, flat open field of pasture land. One, 8, and 64 kg cylindrical charges of C-4 explosives were used as sound sources. Measurements of acoustics, seismic ground response, complex ground impedance, and meteorology were carried out simultaneously. This paper gives an overview of the experiments and presents examples of data. An overview of theoretical research based on the collected dataset is also given. Literature for further details is referred. An easily accessible database with all data is available. Detailed information is also given at www.bpof.inter.net.
On the applicability of models for outdoor sound
Authors: Rasmussen Karsten B.
Published: July 2002

The suitable prediction model for outdoor sound fields depends on the situation and the application. Computationally intensive methods such as parabolic equation methods, FFP methods, and boundary element methods all have advantages in certain situations. These approaches are accurate and predict not only sound pressure levels but also phase information. Such methods are, however, not always able to predict the sound field for more complicated scenarios involving terrain features, atmospheric wind and temperature gradients, and turbulence. Another class of methods is based upon approximate theory—such methods are inherently less accurate but generally faster in computation and may also provide results for complicated scenarios. The accuracy of different approaches in realistic situations is discussed based upon measured and simulated data.
Low dispersion schemes. Mathematical model and demonstrative cases
Authors: Baysal Oktay
Published: July 2002

Considering the computational efficiency along with the high-fidelity to the underlying physical principles, significant advances have been achieved in aeroacoustics simulations. This, however, appears to have remained somewhat limited to the modeled periodic noise generation and propagation by assuming linear waves in uniform flow. Toward this goal, some useful applications have been computed with low-dispersion schemes, by solving the linearized Euler and Navier–Stokes equations. In addition to the benchmark cases reported in the CAA Workshops, these included a supersonic jet noise simulation. On the other hand, there are numerous aeroacoustics applications, such as subsonic jet noise and cavity noise, where the linear wave and uniform flow assumptions would be too compromising. Consequently, the linear dispersion-relation-preserving scheme and its boundary conditions have been extended to the nonlinear equations. It has been tested for a number of simple initial-value and periodic-source problems. Presently, a cavity noise problem and its suppression are being computed with this computational model. [Work partially supported by NASA Langley Research Center.]
Time-of-day penalty for aircraft noise
Authors: Miedema H. M. E. , de Jong R. G. , Vos H.
Published: July 2002

Noise metrics differ in the penalty they apply to noise in the evening and the night, and in the definition of these periods. In LAeq (24 h) no penalties are applied. DNL uses a penalty of 10 dB for the night (22–7 h), and DENL uses a penalty of 5 dB for the evening (19–23 h) and 10 dB for the night (23–7 h). An earlier meta-analysis [J. M. Fields, ‘‘The relative effect of noise at different times of the day—an analysis of existing survey data,’’ Hampton, VA, NASA, Report CR-3965 (1986)] did not reveal consistent evidence on the basis of which penalties could be chosen. Data from the 1997 Amsterdan airport study [De Jong et al., ‘‘Hinder, Gezondheid- en Belevingsaspecten in de regio Schiphol,’’ TNO-PG, Leiden, Netherlands, report 98.039 (1998); N=11,812] are used to estimate the time-of-day penalties that give an optimal relation with noise annoyance and self-reported sleep disturbance, respectively. LAeq’s are available for the following periods: 7–19, 19–22, 22–23, 23–6, and 6–7 h. Penalties are estimated for the whole sample, and for subgroups (e.g., age classes, different levels of noise sensitivity) separately.
Acoustic emission (AE) generated by thermomechanical stress in ceramics
Authors: Ranachowski Przemyslaw
Published: July 2002

The thermomechanical long-term toughness of electrotechnical porcelain of long-rod insulators was investigated. The objects were stressed in thermostat to 63% of nominal strength. The cyclic temperature changes from −25 deg to 40 deg were applied. This treatment corresponds to real environmental conditions on insulators of high-voltage lines. The AE method was used for monitoring the gradual destruction process during the test. The initiation and propagation of microcracks correlate with AE count rate. It was stated that the magnitudes of consecutive maxims of count rate during the cooling part of the cycle depend on the advancement of material destruction. The increase of these maxims means reduction of ceramics toughness related with multiplication of microcracks number and larger probability of critical cracks appearance. When the ceramics retains its strength, the maxims have approximately the same value in consecutive test cycles. The AE method was successfully applied in product certification procedure and control of electric power system. The degree and parameters of porosity of ceramics have been defined on the basis of velocity and attenuation of ultrasound. The good agreement of theoretical and experimental data was obtained. [Work supported by KBN Grant No. T 07B 03413.]
Modeling the effects of peripheral compression on temporal processing
Authors: Oxenham Andrew J.
Published: July 2002

Most psychoacoustic models to date have assumed that the peripheral auditory system can be treated as a bank of quasilinear bandpass filters. Physiological studies have shown that this approximation holds only for damaged cochleae: the response of the basilar membrane in a healthy cochlea to sounds around the characteristic frequency is highly nonlinear and compressive over a wide range of sound-pressure levels. Many aspects of psychoacoustic performance can be explained by taking these nonlinear characteristics into account. Short-term temporal integration in simultaneous and forward masking (where signal detectability improves with increasing signal duration), the critical masking interval (where the detectability of a brief signal temporally centered in a masker decreases with increasing masker duration), and the nonlinear growth of forward masking can be modeled with relatively simple assumptions if a physiologically reasonable nonlinearity is included. Similarly, some aspects of temporal processing in listeners with cochlear hearing impairment can be well described by making the reasonable assumption that cochlear compression is reduced or absent. Thus the altered temporal processing shown in some tasks by hearing-impaired listeners may in many cases be due to reduced cochlear compression and not to more central processing deficits. [Work supported by the NIH/NIDCD.]
Cochlear wavelength representation of acoustic frequency content
Authors: Wilson Timothy A.
Published: July 2002

Peak-to-peak wavelengths of the sinusoidal steady-state response of a simple cochlear model (i.e., incompressible, inviscid, linear fluid in a rigid box-shaped chamber with a linear stiffness-damping-mass cochlear-partition model having exponential parameter variation with place) were examined. For a wide range of mechanical parameter values and for a wide range of cochlear places (at least those basalward of the best place, though sometimes extending through the best place toward the cutoff regime), the wavelengths were found to be (1) a decreasing, approximately linear function of the logarithm of the stimulus frequency, and (2) a decreasing, approximately linear function of cochlear place. A simple algorithm for estimating the instantaneous frequency of sinusoidal signal components of complex signals (both constant frequency and chirps), as well as that of the center frequency of pulse- and noise-excited resonances, was developed. Estimator performance for the cases of signals plus noise and of noisy wavelength measurements will be presented.
A network model for the functional architecture of the mammalian medial superior olivary nucleus
Authors: Weiland Claus
Published: July 2002

The mammalian MSO is part of the superior olivary complex, the lowest level in the auditory pathway, which receives binaural input. The microarchitecture of the anatomical convergence of both paths is considered as fundamental for the nucleus’ function in time-domain processing. This architecture is mostly, according to a proposal of Jefress [L. Jefress, J. Comp. Physiol. Psychol., 35–39 (1948)], interpreted as a network of coincidence detectors. However, the topographic organization of the afferent innervation of the MSO shows much more complexity as is assumed in this model. The connection scheme of the afferent fibers indicates even aspects of terminal divergence, which should result in broadly tuned neurons instead of sharply tuned coincidence detectors. This study proposes a network model for the processing of interaural time delays (ITD) in the MSO based on connection patterns and response properties of the MSO fusiform cells. Instead of local coincidence detection, coding of ITD’s is distributed over a cluster of fusiform cells, where each individual cell’s degree of stimulus resolution is considered as low. While in the Jefress model each delay line is processed as a single channel, cluster formation based on MSO’s multipolar cells could be a mechanism for sampling over these different channels.
Modeling the precedence effect: Mechanisms of onset enhancement in binaural lateralization models
Authors: Dizon Roberto M. , Colburn H. Steven
Published: July 2002

The general behaviors of several cross-correlation-based lateralization models in response to dynamic impulsive stimuli, specifically those that elicit the precedence effect, are presented as a means of gaining insight into the contribution of individual stages of the models to precedencelike performance. In general, an enhancement of activity at onset relative to later activity in the model outputs (or equivalently, a temporary post-onset suppression in activity) is associated with the relative perceptual emphasis of early portions of stimuli as described by the precedence effect. Onset enhancement in cross-correlation models has thus far only been exhibited using dynamic mechanisms such as that used in a model by Lindemann [J. Acoust. Soc. Am. (1986)]. This work evaluates the dynamic properties of several cross-correlation models that include mechanisms of onset enhancement. These mechanisms include that of Lindemann’s model, peripheral mechanisms such as adaptation, and a physiologically based inhibition mechanism adapted from the IC model of Cai, Carney, and Colburn [J. Acoust. Soc. Am. (1998)]. In this work, binaural models are created from various permutations of the mechanisms described above, and the final and intermediate model outputs in response to a few sample stimuli are shown and compared. [Work supported by ONR and NIH (NIDCD R01 DC00100).]
Amplitude modulation discrimination interference (MDI) and speech intelligibility for normal-hearing and hearing-impaired subjects
Authors: Koopman Jan , van der Horst Rene , Dreschler Wouter A.
Published: July 2002

Modulation detection interference is the decrease of sensitivity for amplitude modulation when the target is presented in the presence of modulating maskers [W. A. Yost and S. J. Sheft, J. Acoust. Soc. Am. 85(2), 848–857 (1989)]. A typical MDI experiment was carried out in which the maskers were located at 500 and 4000 Hz and the target at 1400 Hz (covering the range of speech frequencies). The modulation depths of the maskers were held steady at 0, 0.18, or 0.30. The modulation rates were 4, 8, or 16 Hz. Also, speech intelligibility was measured by a sentence test in continuous noise. Both experiments were carried out for six normal-hearing and six hearing-impaired subjects. All signals were presented at MCL level. Both groups of subjects were most sensitive for amplitude modulations when they were presented in the presence of the nonmodulating maskers or without any masker at all; the sensitivity decreased by 8 dB for normal-hearing subjects and 6 dB for hearing-impaired subjects for the modulated masker condition. The modulation threshold also increased for increasing reference modulation depths. The amplitude modulation discrimination sensitivity showed a significant correlation with the critical signal-to-noise ratios for speech intelligibility.
Some inverse problems in structural acoustics
Authors: Ghosh Roy D. N. , Warner J. A. , Couchman L. S.
Published: July 2002

Fundamental problems of direct and inverse solutions of the multidimensional Helmholtz equation in structural acoustics are discussed. Inversion of an obstacle from the far-field pattern is compared to that of a medium and/or an inhomogeneity. An overview of various methods of solving these problems is presented. The linear techniques for a medium and/or inhomogeneity problem and some intrinsically nonlinear methods of inversion for an obstacle are described. Four nonlinear methods are compared, namely, the ansatz of potential over an internal curve, the dual space method using the Herglotz wave functions, methods based on the Rayleigh hypothesis, and the most recent method of shape differentiation combined with the Padé approximation. The efficacy and advantages of the last technique are illustrated by numerically reconstructing both penetrable and impenetrable obstacles of various shapes and orientations. The robustness of the method in regard to the initial conditions and regularization are demonstrated and its extension to elasticity and an acoustical shell structure are presented. Finally, some open questions and future directions are discussed.
Experimental reconstruction of the vibration field in a cylindrical shell
Authors: Kochkin Alexander A.
Published: July 2002

The problem of reconstruction of a vibration field in an unmeasured region of a thin cylindrical shell is considered. The forced vibrating shell of finite dimension is divided into two parts: the first part (the measurement region) is accessible for measuring any vibration characteristics while the second part (the reconstruction region) is considered as ‘‘inaccessible.’’ The problem is solved using the general approach–see the presentation by Yu. I. Bobrovnitskii at this Special Session. The vibration field of the shell was modeled by a finite sum of normal waves, amplitudes of which were determined by equating the modeling values to the data. The SVD technique was used in the inverse procedure for parameter identification. Most attention was paid to finding an appropriate method for choosing the best model of the shell which minimized the reconstruction error. For the measurement error 0.08 of the experiment conducted, the reconstruction error varied from 0.2 to 0.7 as the measurement region diminished from 0.7 to 0.3.
Constrained beamformer optimization for antenna arrays
Authors: Bronzel Marcus , Boche Holger
Published: July 2002

A common problem for hands-free voice terminals and for smart antennas in future mobile communication systems is to optimize the directional response of an array of microphones or antennas in order to enhance the signal-to-noise-and-interference ratio (SINR). Digital beamforming (DBF) algorithms are commonly used for spatial filtering, which effectively results in more benign channel characteristics with reduced interference and delay spread. A new constrained beamforming (CBF) algorithm is presented which allows optimum beam steering and interference cancellation for arbitrarily spaced directions for signals-of-interest (SOI) as well as for interfering waveforms. The proposed CBF combines the properties of the Bartlett and the Capon beamformer [L. C. Godara, ‘‘Application of antenna arrays to mobile communications, Part II: Beam-forming and direction of arrival considerations, Proc. of the IEEE 85, 1193–1245 (1997)] while maintaining minimum power requirements. It is shown that the result can be interpreted as the beam vector representation using special interpolating fundamental functions. [Work supported by BMBF.]
Real-time identification of multiple modes using IIR filters
Authors: Lim Tae W.
Published: July 2002

Infinite impulse response (IIR) filter design is investigated in this work for the real-time identification of modal parameters such as natural frequencies, damping ratios, and mode shapes of vibrating structures or acoustic medium. This investigation is mainly aimed at identifying multiple modes simultaneously using a new structure of IIR filters. The previous algorithm [Lim, Cabell, and Silcox, J. Vib. Acoust. 118, 649–656 (1996)] was limited to the identification of one or two modes due to its complexity of converting identified IIR filter coefficients to modal parameters. The new IIR filter structures are designed to allow direct conversion of filter coefficients to modal parameters and to facilitate the identification of multiple modes. The new approach provides promise for a realistic application of the algorithm for narrow-band signals, which typically contain multiple frequencies. The new IIR filter structure is described in detail in this paper. Results of the multiple mode identification are presented using the new approach. [This research was supported by Structural Acoustics Branch, NASA Langley Research Center.]
Predictive control for suppression of vibration induced by flow noise
Authors: Phan Minh Q. , Eure Kenneth , Juang Jer-Nan
Published: July 2002

Predictive control is based on an intuitively appealing concept where the current control action is based on a prediction of the system controlled output at some time step in the future. Originated from chemical process engineering, several predictive control methods have emerged, including model algorithmic control (MAC), dynamic matrix control (DMC), extended prediction self-adaptive control (EPSAC), extended horizon adaptive control, multistep multivariable adaptive regulator (MUSMAR), and generalized predictive control (GPC). Sharing a similar philosophy, the details of these controllers are different from each other due to different choices of cost functions, constraints, and dynamic models. Almost all of them are model-based methods. A data-based treatment of predictive control will be presented, and it will be shown how it is an effective tool to suppress flow-induced vibrations. In this approach, the predictive controller gains are synthesized directly from input–output data instead of working through an intermediate identification model. Experimental results will be used to illustrate the benefit of this data-based treatment of predictive control over conventional model-based approach. A technique will be developed to avoid drifting of the controller gains identified with closed-loop data, and comparison with other conventional methods such as conditional updating, dither, and leakage will be made.
Multichannel signal decorrelation using a new generalization of the prediction error filter
Authors: Cook Jonathan G. , Elliott Steve J.
Published: July 2002

In many multichannel adaptive active control systems the convergence speed is limited by the correlations between the reference signal channels. This paper describes a causal digital filter capable of reducing multiple channel correlated colored input data to an underlying set of uncorrelated unit variance white signals. This filter is the inverse of the spectral factorization matrix for this data set. The problem is approached in two steps. In the first step it is shown analytically that a single-point delay prediction error filter (PEF) will produce white output error data from multiple nonwhite inputs. The flaw in this process is that there is an inherent matrix of gains which scales and correlates the output channels. In the second step, a technique is presented to derive this matrix of gains from the spectral density matrix of the filter output and thus calculate the appropriate inverse filter. The output from this two-step process is uncorrelated, unit variance and white. The complete signal reduction algorithm is then demonstrated for multiple-channel simulated signals and is finally calculated and tested on correlated road noise data.
Nonlinear adaptive controller for electrodynamic transducers without additional sensor
Authors: Irrgang Stefan , Klippel Wolfgang
Published: July 2002

Nonlinearities inherent in common transducers (woofers, shakers) produce new spectral components (distortion) in the reproduced sound which affect the perceived sound quality and impair active noise and vibration control. Recent research has shown that these nonlinear mechanisms are predictable and a nonlinear control can compensate for these distortions by inverse preprocessing of the electric input signal. Adaptive schemes have been developed to adjust the control parameters to the particular transducer and to cope with parameter uncertainties due to heating and aging. The adaptive controller requires an acoustic or mechanic output signal derived from the transducer. Using an additional sensor increases the costs and is impractical under harsh environment. However, a motional signal can also be derived from the back-induced EMF by monitoring the input current of the transducer only. Following this approach a nonlinear adaptive observer is implemented in a digital control system providing a robust and cost-effective solution for audio applications and active noise control.
Speech quality evaluation: Auditory and instrumental procedures
Authors: Kettler Frank , Gierlich Hans Wilhelm , Diedrich Ekkehard
Published: July 2002

The complexity of signal processing in modern communications requires advanced test procedures. This goes both for telecommunication networks and for terminal equipment. Hands-free telephones (HFT) are a typical example of telecommunication equipment showing a nonlinear, time-variant, and speech-controlled transmission characteristic. In addition, the performance depends heavily on the environmental conditions like test rooms or ambient noise. New test procedures, both auditory and instrumental, have been developed during the last few years to determine quality aspects of HFT’s. The principles of auditory tests as conversational tests, specific double-talk tests, and a new realization of listening-only tests are discussed. These procedures are used for both parameter identification and value selection. The parameters found to be auditory relevant have been identified, and advanced tests for quality parameters—determining the auditory quality—were developed.
Psychoacoustically motivated objective speech quality evaluation procedures, PSQM, and improvements
Authors: Bereends John G. , Hekstra Andries P.
Published: July 2002

PSQM (Perceptual Speech Quality Measure), measuring speech quality objectively, has been standardized by ITU-T as recommendation P.861. PSQM characterizes the perception of the (degraded) output speech signal of the system in comparison to the (ideal) input speech. A perceptual model is used that maps input and output signals onto psychophysical representations using psychophysical equivalents of frequency (Bark) and intensity (compressed Sone). The quality of the device under test is determined with a simple cognitive mapping from the differences in the psychophysical representation to the perceived speech quality in terms of Mean Opinion Scores (MOS). Within an ITU benchmark testing a limited set of unknown codec distortions the PSQM showed high correlations (around 0.97) between subjectively perceived and objectively measured speech quality. When applying PSQM to a wide variety of real world distortions two major limitations show up: First, dynamic time warping effects, as they will be found, e.g., in Internet telephony, cause a degradation in correlation. Second, the perceptual model that is used within the PSQM method is too simple to account for a wide variety of distortions. This presentation discusses extensions to the PSQM method that allows application to real world distortions.
Selection of cross-fade regions for concatenation of demisyllable units
Authors: Kibre Nicholas J.
Published: July 2002

This paper describes how overlap regions are identified in demisyllabic filter trajectory concatenation units used in a residually excited formant synthesis system, currently being developed at PTI/STL [Pearson, Kibre, and Niedzielski, ICSLP 1998]. This approach has been shown to produce clear and human-like synthetic speech, but as in other concatenative methods smooth transitions in cross-fade regions are essential to sound quality. This can best be obtained if a nucleus region is identified for each segment type which has consistent filter trajectories in all tokens. Database size precludes manual tuning and labeling, and this paper considers and compares two approaches to automating this task. The first of these is a rule-based approach, in which observation and phonological theory are used to formulate an ideal cross-fade region for each segment. Each token is searched for its best match to this definition, as determined according to penalty weights for different kinds of deviation from it. The second is a statistical approach utilizing HMMs to model the cross-fadable nuclear region of each segment type, as well as the adjacent transition regions. Rather than manually marking these regions in the training data, embedded training can be used to discover optimal definitions for the three phases.
A combined wideband speech and audio coder using human articulatory and auditory models
Authors: Wang Guangyu
Published: July 2002

The modern high-quality speech and audio coding algorithms are mostly based on human articulatory models. On the other hand, high-quality audio coders are mostly based on models of human auditory system. However, neither of these two methods can provide acceptable performance for both music and speech. In this paper a coder is proposed which operates for both wideband speech and audio signal. The structure of the coder is based on both human articulatory and auditory models. In the first part the LPC parameters are drawn using the conventional autocorrelation method in order to simulate the human articulatory system. With LPC parameters the redundancy of input signal in time domain is removed. In the second part of the coder, the LPC residual signal is further transformed into frequency domain through MLT (modulated lapped transform). The MLT coefficients are then quantized in frequency domain using the human auditory properties such as masking property. The informal listening test has shown that the proposed coder provides very good quality for wideband speech and for most music signals.
Perceptual criteria for eliminating reflectors and occluders for efficient rendering of environmental sound
Authors: Martens William L. , Herder Jens
Published: July 2002

Given limited computational resources available for rendering spatial sound imagery, it is important to determine effective means for choosing what components of the rendering will provide the most audible differences in the results. Rather than begin with an analytic approach that attempts to predict audible differences on the basis of objective parameters, subjective tests were executed to determine the audible difference made by two types of sound obstruction: reflectors and occluders. Single-channel recordings of 90 short speech sounds were made in an anechoic chamber in the presence and absence of these two types of obstructions, and as the angle of those obstructions varied over a 90-deg range. These recordings were reproduced over a single loudspeaker in that anechoic chamber, and listeners were asked to rate how confident they were that the recording of each of these 90 stimuli included an obstruction. The results revealed the conditions under which these obstructions have a significant impact on the perceived spatial image. These confidence ratings were incorporated into an evaluation function used in determining which reflectors and occluders are most important for rendering.
Spatial auditory displays for speech communications: A comparison of free-field and virtual acoustic environments
Authors: Nelson Todd , Ericson Mark , Bolia Robert , McKinley Richard
Published: July 2002

The ability of listeners to monitor the simultaneous presentation of multiple speech signals was evaluated in free-field and virtual acoustic environments. Two acoustic environment conditions (free-field and virtual) were combined factorially with two spatial conditions (spatially separated and nonspatially separated), eight simultaneous talker conditions (1, 2, 3, 4, 5, 6, 7, and 8), and two sex of critical speech signal (male and female) to provide 64 experimental conditions. A within-subjects design was used. Participants, four males and four females, were required to detect and identify the presentation of critical speech signals among a background of nonsignal speech events. Speech stimuli consisted of a call sign and a color–number combination contained within a carrier phrase (e.g., ‘‘Ready TIGER go to WHITE ONE now.’’). The USAF Air Force Research Laboratory’s (AFRL) Auditory Localization Facility—a 277-speaker geodesic sphere housed within an anechoic chamber—was used for free-field presentation, and AFRL’s 3-D auditory display generators were employed for virtual presentation. Results indicated that spatial separation of the speech stimuli enhanced performance efficiency of the free field and the virtual conditions. Implications for the design of spatial auditory displays to enhance communication effectiveness and situation awareness are discussed.
Design criteria for auditory virtual environments
Authors: Sahrhage Jörg
Published: July 2002

During the last decade a large variety of different systems and technologies for generating auditory virtual environments emerged, ranging from low-cost displays for computer games to high-end interactive and immersive systems comprising real-time head tracking and more or less authentic room simulation. Auditory virtual environment generators employing different technological approaches have been developed for various application areas like navigation aids, teleconferencing systems, virtual control rooms, integrated multi-modal VE generators, or even as tools for psychophysical research. Depending on the particular application, perceptually relevant parameters (e.g., responsiveness, smoothness, and authenticity) will be identified on the basis of psychophysical knowledge. Further, an attempt will be made to derive and quantify adequate physical design criteria. An overview of the available auditory VE generators will be followed by a careful analysis and comparison of the system’s alleged capabilities and technological constraints. These will be checked against the proposed application-dependent design criteria.
Applications of the waveguide invariant approach
Authors: Kuperman William A. , D’Spain Gerald L. , Song Hee Chun , Thode Aaron M.
Published: July 2002

The waveguide invariant approach [S. D. Chuprov, Acoustics of the Ocean: Current Status, edited by Brekhovskikh and Andreevoi (Nauka, Moscow, 1982), pp. 71–91; L. M. Brekhovskikh and Y. P. Lysanov, Fundamentals of Ocean Acoustics, 2nd ed. (Springer-Verlag, Berlin, 1991), pp. 139–145] summarizes the dependence of robust interference phenomenon on the ocean environment by a parameter denoted ‘‘β.’’ This elegant approach is based on the dependence of group speed on phase speed for general classes of environments. It has been used to describe interference patterns in the range-frequency plane for range-independent and two-dimensional, mildly range-dependent environments. The approach is extended to include the third dimension, azimuth dependence. Furthermore, an analogous derivation for describing the ambiguity structure of a broadband linear matched-field processor reveals that the trajectories of sidelobes of such a processor in the range-frequency plane are also subject to the same invariant description. A connected result is that the focus of a phase-conjugate array can be shifted away from or toward the original probe source by a process directly related to the waveguide invariant formalism. These new applications have all been experimentally verified.
My life, my teachers, pupils, and friends
Authors: Brekhovskikh Leonid M.
Published: July 2002

The presentation will cover the following topics: Childhood dreams: Have they become a reality? Why and how did I become an acoustician? Sixty years of creativity in the golden times of Russian science, including oceanography and ocean acoustics. Contact with which outstanding persons did influence my world outlook? What could I pass on to my pupils? What does the ‘‘iron mode of life’’ mean? What is the short- and long-term future of Russian science from my point of view?
Structure-borne sound transmission at floor/joist connections in wood frame buildings
Authors: Bosmans Ivan , Nightingale Trevor R. T.
Published: July 2002

The overall sound transmission in wood frame buildings is largely affected by flanking transmission involving the structural members of the building. Due to the inhomogeneous and anisotropic nature of the building components, as well as the difficulty of modeling the structural connections, theoretical predictions of flanking transmission are characterized by a considerable uncertainty. This paper investigates structure-borne sound transmission at plate/beam junctions which are typical of floors in wood frame buildings. The aim of this study is to gain a better insight into the dynamic behavior of the floor assembly and to improve the accuracy of the theoretical predictions of vibrational energy flow through floors. For this purpose, measurements are carried out on a small floor section for various coupling conditions of the joists. By varying the number of fasteners used to attach the joist to the plate, frequency ranges are identified where the junction behaves as a line connection or as a series of local connections. The experimental results are compared to theoretical predictions using statistical energy analysis, where coupling loss factors are calculated based on the wave approach for semi-infinite plates.
A computer-based lung sound simulator
Authors: Kompis Martin , Schwarz Oliver , Becher Mathias , Burri Ulrich
Published: July 2002

A novel lung sound simulator for teaching and research purposes is presented. While heart sound simulators have been used for many years in medical schools and schools of nursing, first algorithms for the simulation of lung sounds have become available only recently [M. Kompis and E. Russi, Med. Biol. Eng. Comput. 35, 231–238 (1997)], and no practically useful device has been presented up to now. The newly developed lung sound simulator requires no specialized hardware except for a personal computer, a standard sound card, and loudspeakers which are able to reproduce the predominately low-frequency sounds. Normal, bronchial, and tracheal respiratory sounds can be simulated, and continuous and discontinuous adventitious lung sounds, as well as heart sounds, can be added. The attributes of these sounds, such as, e.g., loudness, duration, and pitch, are controlled by the user. Changes in the simulation parameters can be made effective almost immediately. Simulated sounds can be saved for later use, and recordings of real lung sounds can be reproduced from within the lung sound simulator program, allowing direct comparison between real and simulated sounds. Simulation parameters can be stored and libraries of simulated lung sounds can be created.
Measurement and visualization of vibroacoustic fields caused by impact on solid surfaces
Authors: Augutis Vygantas , Gailius Darius
Published: July 2002

The system for measuring and visualization of vibroacoustic fields, and observation of propagation of acoustic waves of various types on solids surfaces was developed. It is possible to reconstruct the measured vibration or compute modes of vibration (movie, animation) and determine types of acoustic waves and their speeds from the measured data arrays. The vibroacoustic fields of some objects were measured (glass block, steel block, rod, pipe, etc.) in a frequency range from 1 kHz to 10 MHz. The amplitudes of vibrations range from 10 to 2000 nm. Different devices for reception and actuation of vibrations were developed and used: piezoelectric, electromagnetic, microphones, etc. The results of the investigation are presented in CD-ROM (conditions of measurement, animated vibrations, modes).
Fan noise—In-duct sound power—What are we measuring and why?
Authors: Cory W. T. W.
Published: July 2002

The paper will review the history of the measurement of fan noise. Early standards will be described and related to the state of knowledge existing at the time. The need for reliable methods of measuring the noise in-duct was foreseen at an early stage. This required the development of microphone shields able to ensure accurate readings of noise in a moving airstream. The relationship of in-duct levels to those measured around the fan in reverberent or free-field conditions will be discussed. Finally, the forthcoming ISO standards will be introduced and the recognized need for a number of different levels to be recorded.
In-duct fan sound power measurement. Part I. Review of general problem
Authors: Neise Wolfgang , Arnold Frank
Published: July 2002

Many technical fan installations involve a duct on the inlet side and/or outlet side of a fan. Therefore the need arises for a measurement method for determination of the sound power radiated by fans or other fluid handling sources into a duct. In the paper the problems associated with such a method are discussed: Axial standing waves due to sound reflection from the duct end, acoustic loading of the source, turbulent flow pressures superimposed on the sound field, flow noise generated by flow/microphone probe interaction (self noise), discrimination between sound pressures and turbulent flow pressures, measurement position in the duct in view of higher-order mode sound propagation and directional characteristic of the microphone probe used, and modal distribution of sound power. Early and recent work on the above topics is reviewed. A brief description of the standarized in-duct method ISO 5136 (1990) is given.
Simulation of acoustic scattering from a trailing edge
Authors: Singer Bart A. , Brentner Kenneth S. , Lockard David P. , Lilley Geoffrey M.
Published: July 2002

Unsteady two-dimensional flow calculations were performed about a thin NACA airfoil with a bluff-body vortex generator positioned at 98% chord. The bluff body produced unsteady vortex shedding, which simulated large coherent eddies in a boundary layer. The shed vortices interacted and occasionally paired as they convected past the sharp trailing edge of the airfoil. The CFD calculations clearly showed acoustic waves emanating from the airfoil trailing edge. The Ffowcs Williams and Hawkings equation [J. E. Ffowcs Williams and D. L. Hawkings, Philos. Trans. R. Soc. A 264, 321–342 (1969)] was used to compute the acoustic field generated by the unsteady aerodynamic field. Directivity maps and Mach-number scalings have been obtained and compared with the theoretical predictions for trailing-edge noise. The noise below the airfoil displays characteristics typical of a sound field scattered from a sharp edge. The noise above the airfoil is more complicated; it contains significant contributions from both the scattered acoustic field and the bluff-body vortex generator. The effects of spanwise correlation length and the choice of integration surface will also be discussed.
Jet noise of subsonic aircraft: An aeroengine manufacturer’s perspective
Authors: Kempton A. J.
Published: July 2002

These papers trace the history of jet noise reduction from early pure jets to modern engines with high bypass ratios. Choice of engine cycle is important in reducing jet noise, but there is also the need to balance the requirements of cost, weight, and specific fuel consumption, in addition to minimizing other engine noise sources. Some techniques for reducing jet noise are illustrated. Forced mixers on long-cowl engines are effective in reducing low-frequency jet noise, but can cause an increase in high frequencies (especially in flight). For larger aircraft with more powerful engines, where the noise problem is most severe, installation issues dictate somewhat against long-cowl configurations. The current challenge is therefore to achieve similar noise reductions to forced mixers but on short-cowl engines. In the future, the requirement for further noise reductions might result in much more novel aeroplane and engine configurations. Jet noise experiments can easily give misleading results unless the facilities and rigs are designed to avoid contamination by excess noise and to provide correct simulation of boundary layers and other aerodynamic parameters (including flight simulation). Fortunately, testing at reduced scale is acceptable for noise. Advanced measurement techniques (like source location) can be very beneficial in aiding understanding.