bluuebunny/crossref_metadata_2025 huggingface dataset:
Effect of target versus distracter-tone frequency region and variance on sample discrimination of frequency differences
Authors: Neff Donna L. , Wrage Rebecca L. , Jesteadt Walt
Published: July 2002

This experiment examined how performance in a complex frequency-discrimination task was influenced by the frequency region and relative frequency variability of target versus distracter tones. In a 2IFC sample-discrimination task, normal-hearing listeners were asked to select the interval containing target tones drawn from the higher of two overlapping Gaussian frequency distributions. The target distributions were placed in one of three frequency regions: ‘‘low’’ (500 Hz), ‘‘middle’’ (1414 Hz), or ‘‘high’’ (4000 Hz). For a particular target region (e.g., low), two distracter tones were then drawn at random and added, one from each of the Gaussian distributions at the two remaining frequency regions (e.g., middle and high). All distributions were equivariant on a logarithmic frequency scale. Based on the CoRE model [R. A. Lutfi and K. A. Doherty, J. Acoust. Soc. Am. 96, 3443–3450 (1994)], it was predicted that tones with greater variance would have a greater influence on performance independent of frequency region. The results indicate that the influence of variance depends on the frequency region of targets versus distractors, with notable individual differences and training effects. [Work supported by NIDCD.]
Comodulation masking release (CMR) for multiple temporal envelope maskers
Authors: Mendoza Lee
Published: July 2002

Thresholds for a pure-tone signal were measured in masking noise composed of multiple noise bands. In some conditions, one group of flanking bands (consisting of half of the flanking bands) was comodulated, and independent of the other group of comodulated flanking bands in terms of temporal envelope. The center frequencies of the flanking bands were arranged such that noise bands from the two groups were spectrally interleaved. In these conditions, the band of noise centered on the signal was a combination of two noise bands, which were each comodulated with one group of flanking bands. CMR was obtained but was not of the magnitude observed when all noise bands shared a common temporal envelope. Potential cues in the multiple envelope maskers included the information in the summed representation of the flanking bands, and the degree of temporal envelope correlation between the on-signal band and the flankers. Additional stimulus conditions examined the viability of these cues. The results of these experiments support an envelope correlation explanation for the CMR observed with the multiple temporal envelope maskers. [Work supported by the Louisiana Board of Regents.]
Auditory profile analysis: Effects of asynchrony and ear of presentation
Authors: Hill Nicholas I. , Bailey Peter J.
Published: July 2002

The ability of listeners to detect an increment in the level of a 1-kHz tone relative to four flanking tones (having frequencies of 0.16, 0.40, 2.50, and 6.25 kHz) was examined under four conditions of presentation. In two conditions all five tones were presented monaurally to the listener’s right ear, while in the other two conditions the 1-kHz tone was presented to the left ear with the flanking tones presented to the right. For each of the two spatial configurations, the tones were either gated on and off at the same time or else the 1-kHz tone was gated on 100 ms before, and off 100 ms after the flanking tones. The flanking tones were always gated on and off simultaneously and had a duration of 200 ms. The overall level of the stimuli was randomized on each presentation. Average thresholds in the two asynchronous conditions were approximately 10-dB higher (signal re: pedestal amplitude) than those in the synchronous, monotic condition. Ear differences alone resulted in an average elevation in threshold of 3 dB. These results suggest that asynchrony is more effective than spatial cues in preventing intensity information from being combined across frequency. [Work supported by UK MRC.]
An inversion approach based on multiple-aspect resonance analysis of finite cylindrical shells in water
Authors: Tesei Alessandra , Fox Warren L. J. , Maguer Alain
Published: July 2002

Inverse scattering of fluid-loaded, elastic, thin-walled cylindrical shells is addressed by multiple-aspect resonance analysis. A set of aspect-dependent acoustic phenomena is selected from membrane wave and resonance scattering theories, which are expected to be backscattered by fluid-filled or empty thin-walled shells and to give rise to resonance phenomena in the ka range (1,50). The features selected are from spatial axial modes, and Lamb-type, Scholte–Stoneley, and shear helical waves. The last three wave families are significant only over a small range of target aspects near broadside, with the width of this range depending on shell properties. Approximate equations are formulated relating resonance behavior in the aspect-frequency domain, to target parameters such as shell outer radius, thickness, length, and material. The inverse methodology is validated on experimental data from a steel cylindrical shell with flat end-caps, filled with air or water and suspended in the water column. The target was continuously rotated on the plane of its longitudinal axis while insonified by broadband pulses. Good agreement between theory and experiment encourages the extension of the approach to more complex scatterers for classification purposes. [Work partially supported by EC (MAST DEO project).]
A control friendly software to convert a low-cost DSP board into a powerful active noise controller
Authors: Minguez Antonio , Recuero Manuel
Published: July 2002

A windows environment software has been developed in order to convert the low cost EZ-KIT Lite DSP board of Analog Devices into a powerful monochannel active noise controller. The software, developed with LabVIEW, downloads the code to the EZ-KIT Lite and permits one to modify in real time any working parameter until obtaining the optimal control performance. Among the parameters to control are the gain inputs, outputs, low-pass filters, and dc removing. There are three types of adaptive algorithms that can be used: filtered-X/U LMS, multiple adaptive notch, and a genetic algorithm. All these algorithms are implemented in such a way that it is not necessary to do any previous cancellation path estimation. Also implemented is internal generator that produces up to four simultaneous tones added with a white noise. As it is said before, any parameter of the algorithms and of the generator can also be easily changed in real time, when the cancellation is in progress. This permits one to obtain the best parameters before implementing an active noise controller in any application.
Speech quality and the E-model
Authors: Jekosch Ute , Möller Sebastian
Published: July 2002

The E-model is a quality prediction model for network planning purposes. Its use is going to be recommended by the ITU-T. In comparison with both listening-only and conversation tests the model often shows high predictive power. Accordingly, network planners have high confidence in its applicability. However, until now it is not yet discretely understood which aspects of quality are actually covered by E-model predictions. To name only two, uncertainties do exist, e.g., for the mode of communication (listening only or conversation) and user-related factors. In the talk, a schematic is presented which covers, besides speech-communication-related elements (one-way voice transmission quality, communication effectiveness, ease of communication) also human-factor-related and service-related elements (comfort, costs). It is shown that matters are simplified by exclusively using the MOS as the basis for a thorough verification of the model. In the course of verification, measures of user acceptance, expectation, etc., are needed in order to understand how human- and service-related features influence the users’ perception of the quality of service. An approach is presented to obtain subjective judgments other than MOS on a scale which is closely related to the scale of the model, namely the transmission rating factor R.
Scenarios for economic conversation tests in telephone speech quality assessment
Authors: Wiegelmann Stephan , Möller Sebastian , Jekosch Ute
Published: July 2002

In telephony, there is a permanent need for auditory testing of the effect of different types of impairment on speech communication quality. Though quality perception is different in listening-only and conversation situations, for economic (time and money) reasons, in most cases only listening-only tests are carried out to assess telephone speech quality. For some parameters, however, conversation tests are mandatory, as the parameter under investigation affects only the conversation situation (e.g., echo, delay). In order to overcome the limitations of conversation tests, new scenarios have been developed which allow the testing of around three times as many circuit conditions within one test session compared to conventional conversation test scenarios. The scenarios cover everyday situations of information exchange, and meet most of the basic requirements of quality assessment scenarios (naturalness, balance, comparability, clarity). The scenarios are available in three languages and have been successfully used by different telecommunication test laboratories. Comparison with listening-only test results shows that differences exist between the two types of quality judgments, which underlines the necessity for conversation tests in order to get more valid quality judgments. Modifications of the scenarios for mobile and Internet telephony situations are presented.
Application of automatic speech recognition to evaluation of speech transmission quality in analog communication systems
Authors: Majewski Wojciech , Myslecki Wojciech
Published: July 2002

The preliminary results of application of automatic recognition of isolated words to objective evaluation of speech transmission quality in analog telephone channels are presented. A memoryless, finite state recognition system with LPC, FFT, and BF-FFT (where the speech signal was filtered in Bark bands) parametrization was applied. In classification stage a dynamic time warping and nearest-neighbor algorithm were utilized. Nonsense word lists consisting of 100 logotoms were recorded in a studio by a professional male speaker and utilized next as a test material. Speech transmission quality was examined in laboratory models of telephone channels with frequency bands of 300–3400, 400–2500, and 100–6000 Hz for speech-to-white-noise ratios in the range of +15 to −15 dB. The results of objective measurements expressed in percent of logotoms correctly recognized by the recognition system were compared under the same transmission conditions with subjectively measured logotom intelligibility. The best agreement between subjective and objective evaluation of speech transmission quality was obtained for automatic speech recognition utilizing BF–FFT parametrization. The results of objective evaluation of speech transmission quality by means of the presented method are encouraging and the experiments will be continued for other communication channels (e.g., digital) and different distortions and disturbances.
Automatic syllabic boundary extraction from connected speech
Authors: Harsin Andy
Published: July 2002

A computational algorithm is presented which locates syllabic boundaries in connected speech. Filtered, digitized speech is processed in two stages. The first stage is an auditory front end which filters the digitized speech into critical bandwidths. The critical bands are amplitude-scaled according to a standard frequency/amplitude function, then low-frequency amplitude modulations in the 2–30-Hz range are emphasized according to a perceptual modulation sensitivity function. Next, the critical bands are low-pass filtered at 100 Hz and decimated at a rate of 25:1. Then they are low-pass filtered at 100 Hz, again creating acoustic envelopes of the processed speech, one for each critical band. During the second stage of processing, an autocorrelation-based algorithm is run on the envelopes. The local minima of this algorithm are pooled across the critical-band envelopes, yielding syllabic boundaries. The utterances used to develop and test this algorithm were taken from the Harvard Phonetically Balanced Sentences. Currently, the algorithm places boundaries within a few tens of milliseconds of where a phonemic syllabification would place them. Work continues on fine-tuning the algorithm. Another goal is to compare the algorithm’s performance against human listeners’. [The author wishes to acknowledge the support of Lucent Technologies in conducting this project.]
System for wideband speech coding based on recursive filterbanks
Authors: Pietsch Ralf Th. , Lacroix Arild
Published: July 2002

Wideband speech coding becomes increasingly important because of the enhanced quality due to the extended frequency range. Possible applications are in various fields partly with fixed and partly with variable data rates: synchronous and asynchronous transmission over digital networks, mobile telephony, and multimedia systems. In this context a system is presented which is well suited for dialog communication. The signal analysis and synthesis is done by filterbanks based on recursive filters. The use of recursive filters implies short system delays in the filterbank. However, phase distortions are present and have to be considered carefully. Different methods were implemented to reduce phase distortions and to compensate group delay peaks. Coding is based on redundance and irrelevance reduction. The irrelevance reduction is implemented by the exploitation of psychoacoustical masking effects. Encoder and decoder are running on a single DSP32C signal processor in real time. The achieved data rate for good speech quality is 32 kbit/s and below.
Evaluation of monaural and binaural speech enhancement for robust auditory-based automatic speech recognition
Authors: Kleinschmidt Michael , Wittkop Thomas , Kollmeier Birger
Published: July 2002

A major deficiency in state-of-the-art automatic speech recognition systems is the lack of robustness in additive and convolutive noise. The model of auditory perception, as developed by Dau et al. [J. Acoust. Soc. Am. 99, 3615–3622 (1996)] for psychoacoustical purposes, partly overcomes these difficulties when used as a front-end for speech recognition. Especially in combination with locally-recurrent neural networks (LRNN) the model output, called ‘‘internal representation’’ had been shown to provide highly robust feature vectors [Tchorz and Kollmeier, J. Acoust. Soc. Am. (submitted)]. To further improve the performance of this auditory-based LRNN recognition system in background noise, different speech enhancement methods were examined. The minimum mean-square error (MMSE) short-term spectral amplitude estimator (STSA), as proposed by Ephraim and Malah [IEEE Trans. Acoust., Speech, Signal Process. 32, 1109–1121 (1984)], was compared to a binaural Wiener filter [Wittkop et al., this meeting], based on directional and coherence cues. Both noise reduction algorithms yield highly improved recognition rates in nonreverberant noisy conditions, while the performance in clean speech is not significantly affected. The algorithms were also evaluated in real-world reverberant conditions with speech-simulating noise and jammer speech.
A filtering model for efficient rendering of the spatial image of an occluded virtual sound source
Authors: Martens William L. , Herder Jens , Shiba Yoshiki
Published: July 2002

Rendering realistic spatial sound imagery for complex virtual environments must take into account the effects of obstructions such as reflectors and occluders. It is relatively well understood how to calculate the acoustical consequence that would be observed at a given observation point when an acoustically opaque object occludes a sound source. But the interference patterns generated by occluders of various geometries and orientations relative to the virtual source and receiver are computationally intense if accurate results are required. In many applications, however, it is sufficient to create a spatial image that is recognizable by the human listener as the sound of an occluded source. In the interest of improving audio rendering efficiency, a simplified filtering model was developed and its audio output submitted to psychophysical evaluation. Two perceptually salient components of occluder acoustics were identified that could be directly related to the geometry and orientation of a simple occluder. Actual occluder impulse responses measured in an anechoic chamber resembled the responses of a model incorporating only a variable duration delay line and a low-pass filter with variable cutoff frequency.
Theory of sound propagation in the ocean: A tribute to L. M. Brekhovskikh
Authors: Jensen Finn B.
Published: July 2002

This presentation will be an account (from a personal perspective) of the impact L. M. Brekhovskikh’s first (and now classic) book on Waves in Layered Media from 1960 has had on the development of modern-day numerical models for solving ocean acoustic problems. A whole generation of acoustic modelers, starting in the early 1970s, devoted a career to developing computer codes that would predict sound propagation through a realistic ocean environment, including refraction, scattering, and diffraction of sound within the water column itself, as well as reflection and scattering at ocean boundaries (sea surface and seabed). Not everybody realized that much of the underlying theory had been developed years before and presented in the monograph by Brekhovskikh in 1960 (English edition). In fact, much of the numerical work undertaken in the 1970s and 1980s was merely an automation—through the use of a computer—of solution procedures already outlined by Brekhovskikh. Few, if any, have contributed more to establishing a sound theoretical foundation for ocean acoustics, and L. M. Brekhovskikh’s influence on the field can hardly be overestimated.
The acoustics evolution of Chicago’s orchestra hall: Construction details
Authors: Schuette Dawn R. , Pfeiffer Scott D.
Published: July 2002

Chicago’s Orchestra Hall is a space that has undergone numerous modifications in its 94-year history. Recent efforts have been the most substantial, producing significant changes. This paper will supplement an invited paper by presenting details of design and construction methods that resulted in the acoustic improvements. Particular issues to be discussed will include construction methods of adding volume to an existing and operating building, details of massive construction used to increase bass response, and techniques of sealing fireproofing, plaster, and concrete within the acoustic volume.
Localization of sound in reverberant spaces
Authors: Hartmann William M. , Rakerd Brad
Published: July 2002

Sound sources in the horizontal plane can be localized on the basis of interaural time differences (ITD) or interaural level differences (ILD). It will be argued that without the benefit of an onset transient, listeners can employ only the ILD to localize steady-state broadband sources in a typical reverberant environment. The ITD provides a useful cue only if the cross-correlation (CC) function is adequately large. At low frequencies, a CC peak near 0.1 is barely adequate, but in a reverberant room the physical CC peaks are often less than that. The physical CC peak increases with increasing noise band frequency, but the value of the CC peak that is required for localization increases faster. Therefore, ITD is not useful at any frequency. By contrast, a comparison of physical and psychoacoustical measurements shows that ILD is a useful cue. An optimized ILD model is able to account for human sound localization performance in a reverberant space for broadband and low-pass filtered noises, for both normal listeners and listeners with a high-frequency hearing loss. [Work supported by the NIDCD, Grant DC00181.]
Sound localization with unilaterally degraded spectral cues
Authors: Wightman Frederic L. , Kistler Doris J.
Published: July 2002

Two cues are thought to determine the apparent position of a sound, interaural time differences and the spectral patterning produced by the pinnae and upper body. The role of the spectral cues is the least well understood. One area of uncertainty is the extent to which cues from both ears are used. This study of the issue asked listeners to indicate the apparent positions of virtual sources synthesized from each listener’s own head-related transfer functions (HRTFs). Four types of stimuli were intermingled during a run: (1) all cues preserved, (2) spectral cues degraded at the ear on the side contralateral to the source, (3) degraded cues on the same side as the source, and (4) cues degraded binaurally. Cues were degraded either by smoothing, which preserved the major interaural level cues, or by replacing the natural HRTF with a constant average spectrum, which perturbed the natural interaural level cues. The results suggest that degrading contralateral cues has only a minor effect, while degrading either ipsilateral or binaural spectral cues causes significant localization errors. Results from the two degradation conditions were the same, suggesting that interaural level differences are relatively unimportant localization cues. [Work supported by NIH, NASA, and ONR.]
Vibrations caused by construction work—Measurements and surveillance
Authors: Teuber Wolfgang , Voelker Ernst-Jo
Published: July 2002

Numerous construction principles applied especially in underground engineering are the sources of vibrations. The measurement of vibrations and evaluation of the results are made based upon different criteria. These criteria are formed taking into consideration the influences upon people in the surrounding area, the risk of damage to buildings and the possibility of the malfunctioning of technical systems such as computers or laser equipment. Measurements were taken while several different construction methods were being applied. Short-term measurements are, for the most part, insufficient and must be completed by long-term research. Maximum acceptable values are defined in existing standards based on the influences of vibrations upon persons and upon various kinds of buildings. For technical equipment, stricter limits had to be defined. When the type and location of construction work changes, the application of an automatic monitoring system is advantageous. Such a system is described which was installed to survey computers with special hard disc drives. Existing measurement results allow for predictions and calculations to be made in order to determine the best method of construction. Some principles of building methods, measurements, and vibration monitoring are mentioned.
Damping of string motion, a significant parameter for sound quality
Authors: Wogram Klaus
Published: July 2002

To improve the sound quality of pianos and upright pianos the piano maker mostly plans a change of the construction of the backs and the soundboard with its ribs, bridges, and back posts. Unfortunately he often disregards that all these efforts cannot be successful if the damping of the strings is insufficient. Its influence on the sound quality can be very much stronger than that of the soundboard. Guided by some examples, the importance of the dampers, their construction, their mass, and their position will be discussed. In addition the reduction of sound quality due to an insufficient damping of the not-speaking part of a string will be demonstrated.
A mixed use of finite element and geometrical techniques
Authors: Jean Philippe
Published: July 2002

The finite-element method is limited to either small geometries or to the low-frequency range, whereas geometrical approaches are best applied at medium and high frequencies, but do not take vibrating surfaces into account. When the vibrating boundaries of a volume are only a small part of the total boundary, it is worth combining the two methods. By using geometrical methods to compute a Green’s function that includes all wave reflections on the surfaces of the volume, it is possible to restrict the integral representation of the pressure field to the vibrating surfaces. The surfaces to be meshed will be reduced to surfaces with known velocity fields. This approach offers a means of reducing computation times. It has been applied to large volumes corresponding to a room with a vibrating concrete wall and to a train with a vibrating window.
How children recognize spoken words in speech recognition threshold assessment
Authors: Caroline Lebel , Michel Picard
Published: July 2002

The purpose of this study was to investigate the potential influence of language context effects on the speech recognition threshold (SRT) in children of different ages. To determine the respective contributions of audibility and linguistic factors to the SRT, the speech intelligibility index (SII), and indices k and j as proposed by Nittrouer and Boothroyd [J. Acoust. Soc. Am. 87, 2705–2715 (1990)] were calculated for the following conditions: (1) SRT using an open-set response format; (2) SRT using a closed-set response format (dissimilar foils to increase predictability); and (3) SRT using test words at the beginning of a short sentence. Listeners were three groups of 24 normally hearing children in the first, third, and fifth grades, respectively. Results indicated that SII and factor j decreased significantly and factor k increased proportionally as a function of both lexical and sentence-level context effects in all groups. This suggests that audibility factors are less critical in the facilitating presence of language contexts. The largest effect was shown for lexical context effects with predictable outcome of decreased SRTs. These findings are compelling evidence of language context effects playing a significant role in school-age children required to recognize speech at barely audible levels.
In-sediment coherence measurements in a very shallow-water area
Authors: Lopes Joseph L. , Paustian Iris C. , Stroud John S.
Published: July 2002

A measurement was conducted in a very shallow-water (VSW) region to investigate acoustic propagation into a sandy bottom. The measurement utilized several acoustic sources and two identical hydrophone arrays. The sources were mounted onto a stationary sonar tower complete with horizontal pan and vertical tilt motors. Each hydrophone array consisted of four transducers that were buried via water-jetting methods approximately 10 m from the sonar tower. One array was situated such that the projected acoustic beam was parallel to the rippled structure associated with the sandy bottom, while the other was located where the transmitted beam was perpendicular to the rippled structure. Results of the in-sediment coherence data will be presented as functions of hydrophone burial depth and range in the frequency range between 30 and 150 kHz for the two buried arrays.
Trapping modes of oscillations in an infinitely long waveguide with submerged object in the form of a massive die
Authors: Indeitsev Dmitri
Published: July 2002

It is shown for an infinitely long waveguide—first in the form of a two-dimensional and then a three-dimensional channel—filled with an ideal compressible liquid with a free surface, that there can exist standing waves as well as traveling waves localized in the region of a dynamic submerged object—a massive die. The appearance of such waves is due to the presence of a mixed spectrum of characteristic frequencies of oscillation. In addition to the continuous spectrum, a point (discrete) spectrum distributed along the axis of the continuous spectrum is also present. A relation is found between the geometric parameters of the channel and the massive die for which this spectrum exists.
Evaluation and modeling of small rooms
Authors: Arato-Borsi Eva , Poth Tamas , Furjes Andor
Published: July 2002

The objective characterization of the sound field in a small room with parameters relating to the subjective side needs great experimental efforts. Well-controlled sound is important and critical in reference listening rooms and recording rooms. These are small rooms in the acoustic sense. In order to find new objective parameters for room acoustical evaluation of these spaces, several subjective tests were carried out. In the paper, the results of the listening tests are given. The new objective parameters derived from measured impulse responses are presented. A series of modeling has been carried out by using various geometrical approaches, and the predicted room responses were compared to actual measurements. Because neither purely geometrical room acoustics algorithms nor traditional finite-element techniques can be used effectively, new investigations are needed. The goal of these investigations is to find a high- and medium-frequency modeling algorithm that predicts the most important sound field parameters accurately enough.
Intercomparison test of sound insulation measurement in test facilities. I
Authors: Schmitz A. , Meier A. , Raabe G.
Published: July 2002

The implementation of ISO 140 harmonized at the European level requires for the determination of the characteristic values of building elements that the airborne sound insulation of partitions be measured in test facilities with suppressed flanking transmission. When heavy separating elements are under test, it is also important to take the energy dissipation of the test facility itself into account, as these factors considerably influence the sound insulation measured. Additional measurements of the total loss factor inform about the energy dissipation and may be used to convert sound reduction indices obtained under different test facility conditions. However, in the past two years measuring techniques and experience of loss factor determination have developed to such an extent that reliable results can be obtained now. Moreover, in Germany all test facilities with flanking transmission are to be converted into facilities without flanking transmission. This is why a government-financed intercomparison was carried out, which was aimed at determining the new situation of the German test facilities by measuring the sound insulation of the same solid wall. In addition to the sound insulation, the total loss factors of the partition were measured. The first part of this paper reports on the background, the way of proceeding, and the methods of measurement used for the intercomparison, as well as on the results obtained in a preliminary study at PTB.
Predicting sound transmission through perforated brick walls using SEA
Authors: Fringuellino Marco , Smith Sean
Published: July 2002

The use of perforated bricks or blocks is the most common form of wall construction found in Central and Southern Europe. However, very few theoretical models have been developed for the purpose of predicting sound transmission through these walls or analyzing their behavior. In this paper statistical energy analysis (SEA) is adopted to predict the sound reduction index (R) for these type of walls for comparison with measured data. Due to the enormous variations in brick types used within these countries, measurements were carried out on walls with varying sizes of perforations, density, volume, and surface treatments. For the purpose of this paper special attention is given to the effect on the sound transmission due to the perforations and the effect of additional plastered layers. It was found that SEA can be used as an approximate prediction technique. However, due to the extreme cellular nature in the majority of these types of brick and lack of sufficient modal overlap, within the building acoustic frequencies under study, only very simple SEA models could be adopted. Further theoretical models must be developed outside of the SEA framework to provide sufficient analysis of the behavior for the majority of these multi-component structures.
Spotted Owl reactions to helicopter and chain saw noise
Authors: Delaney David , Pater Larry , Grubb Teryl , Reiser M. Hildegard , Beier Paul
Published: July 2002

Noise impacts on Mexican Spotted Owls (MSO) were documented during the 1995 non-nesting and the 1996 nesting seasons. Members of the MSO population in the Lincoln National Forest, NM, were exposed to typical helicopter and chain saw noise stimuli. The noise stimuli were presented at a variety of controlled levels and distances. Proximate behavioral responses, nesting success, and nesting productivity were observed and documented. The data were sufficient to establish response threshold distances and noise levels for both types of noise. No spotted owl flushes were recorded when the noise stimulus was >105 m distant, regardless of stimulus type. As stimulus distance decreased, spotted owl flush frequency was shown to increase. In terms of A-weighted noise level, the response threshold for helicopter noise was found to be 92 dB SEL and 46 dB LEQ for chain saws. [Work supported by U.S. Air Force.]
Auditory scene analysis for perceptually guided behavior in the echolocating bat
Authors: Moss Cynthia F.
Published: July 2002

The echolocating bat actively probes its environment with ultrasonic vocal signals and builds a three-dimensional (3-D) auditory scene from information carried by the returning echoes. The bat’s sonar receiver computes the azimuth and elevation of a sonar target from the intensity, timing, and spectrum of sonar echoes at the two ears. It estimates target distance from the time delay between each sound transmission and returning echo. Rapidly processing spatial-acoustic information, the bat’s perceptual system analyzes the 3-D auditory scene to support spatially guided behaviors in a dynamic environment. Psychophysical data from two-choice phantom target discrimination tasks suggest that the bat integrates and segregates echo streams along the delay axis. These findings, together with the results of laboratory insect capture experiments, show that echo delay information is used to guide and coordinate a complex set of adaptive motor behaviors. Measures of neutral spatial selectivity in the bat superior colliculus (SC) also suggest the importance of echo-delay processing for spatially guided behavior. Spatial tuning of SC neurons may be important to the bat’s acoustic orienting behavior, since changes in its head aim, pinna position, flight behavior, and vocal production patterns are closely coupled to the 3-D location of a target.
Variability in the acoustic response from natural shallow-water marine sediments
Authors: Lambert Douglas N. , Faas Richard W.
Published: July 2002

The large variation in sediment acoustic response observed between soft clays and sands provides a satisfactory means to estimate sediment properties remotely using acoustic techniques. However, the accuracy of these predictions is limited by variations in sediment properties, both horizontally and vertically on a scale of centimeters to meters, within a given sediment volume, and sets the accuracy limits or ‘‘error bars’’ for prediction of sediment properties. In order to quantify these limits, a series of measurements of acoustic response were made at six sites off Gulfport, MS in sediments ranging from sands to muds. A broadband transducer with an 8° constant beamwidth was mounted on a trolley suspended from an I-beam connected to a pair of legs at each end of the beam. This ‘‘swingset’’ was lowered to the seafloor with the transducer mounted at normal incidence and at an altitude of 3.0 m. Measurements were made at both 30 and 50 kHz at each of eight positions 0.3-m apart along the beam. Results suggest greater variability occurs in the vertical direction (6–24 dB) than in the horizontal direction (6–18 dB).
Optimized echo-envelope matching for acoustic sediment classification with high-frequency single-beam sonars
Authors: Sternlicht Daniel D. , de Moustier Christian P.
Published: July 2002

To extract geoacoustic parameters from the envelope shape of bottom backscattered echoes measured with single-beam high-frequency echosounders, it is advantageous to compare the measured average echo magnitude with an echo envelope model that incorporates bottom scattering theory and sediment properties measured at spatial scales commensurate with acoustic wavelength. With a fully calibrated sonar system, iterative comparisons between model and data yield sediment properties such as grain size, roughness, and volume scattering statistics. Results are presented for a two-phase optimization approach to echo-envelope matching: first the sediment type is established by iterating on the reflection coefficient to match the peak echo amplitude and to establish a general fit with generic values for the remaining geoacoustic parameters, then a three-parameter global optimization is performed using a combination of simulated annealing and downhill simplex searches over the allowable range of roughness spectral strength, volume scattering cross section, and a constrained range of reflection and bottom absorption coefficients which are correlated with mean grain size. Bottom echo envelopes collected at 33 and 93 kHz, over substrates ranging from clay to sand, yield solutions for grain size and geoacoustic properties which are consistent with ground-truth measurements. [Work supported by ONR N00014-94-1-0121.]
Inversion of short-range bistatic scatter for seabed properties
Authors: Pace Nicholas G.
Published: July 2002

Measurements of bistatic acoustic seabed scatter obtained from an AUV operated off the coast of Elba are reported and interpreted in terms of seabed properties. A directional transmitter operating alternatively at 8 and 16 kHz was located at the rear end of an AUV and a line array of receivers was located on an extension from the nose. The AUV was fully instrumented and was programed to follow routes consisting of 5-m radius circles with straight-line connections at a height above the seabed of approximately 3 m.
On the use of a modified optical flow algorithm for the correction of axial strain estimates in ultrasonic elastography in medical diagnosis
Authors: Lorenz Andreas , Ermert Helmut , Garcia-Schuermann Miguel , Sommerfeld Hans-Joerg , Senge Theodor , Philippou Stathis
Published: July 2002

Tumors of the prostate exhibit a significant mechanical hardness in comparison to the surrounding tissue. However, they often can not be reliably detected in the B-mode image. For this reason, digital palpation of the prostate is still a major tool in tumor diagnostics. Ultrasonic elastography estimates the tissue elasticity from an image series recorded under increasing axial tissue compression applied by the transrectal probe. As a new imaging modality, it promises to support conventional prostate screening methods. Usually, time delays from two corresponding A-lines are estimated by cross-correlation methods applied to the acquired rf data set assuming a uniform axial compression. However, in clinical in vivo data, especially in data which were acquired with a sector scanner, significant lateral motion of the tissue which deteriorates the elasticity estimate was observed. A method is presented which uses a modified ‘‘optical flow’’ algorithm to determine the global two-dimensional motion of the imaged object. This information is used to correct the corresponding A-lines according to the encountered lateral deformation. It is shown that this method is able to improve the strain estimates on data with low signal-to-noise conditions. Images of phantoms and clinical in vivo data are presented. [Work supported by DFG Grant ER 94/20-1.]
Ultrasound diagnostic of the blood vessels’ random neoplasm distributed in focus sphere limited environs
Authors: Oboznenko E. I.
Published: July 2002

The scattering of ultrasonic waves in the cylindrical volume, formed with a height of ct/2 (c—sound velocity, t—probe duration) and a focused beam of R radius in the vicinity of the focal zone is examined. There are N vessels of different diameter and random orientation of stream velocity relative to the beam axis. In the case of random blood vessels orientation, the average Doppler echo-signal frequency also will be random f. The estimation of coherent and noncoherent pressure in echo signal is reduced to statistical averaging for discrete scatters—blood vessels. The probability characteristics of the average effective surface of the vessels scattering with the different rules of wave’s incidence angle in the opposite direction (uniform laws, Gaussian laws, etc.) are determined. It was shown that with uniform law of the vessels’ orientation angle’s distribution, the Doppler frequency is approximately equal to: 〈f〉=f0(1−2〈M〉), but taking into account the random vessel orientation it will be: f=f0(1+2〈M〉cos〈Qinc〉), where 〈M〉 is the average Marh’s number, and a 〈Qinc〉 is the average wave’s angle of incidence. The proposed method serves for classification of the vessel’s neoplasm formation in cancer diagnostics.
Acoustic feedback reduction in hearing aids—A feasibility study
Authors: Roccasalva-Firenze Mariano , Rafaely Boaz
Published: July 2002

Acoustic feedback in hearing aids occurs when the receiver of the hearing aid produces an acoustic signal that leaks to the microphone and causes oscillations and instability, limiting the hearing aid gain. The problem is worse in vented hearing aids, which are commonly used to reduce the unnatural feeling caused by the ear occlusion. Common solutions to the feedback problem are low-pass filtering for analog hearing aids, and adaptive feedback cancellation for digital hearing aids. Hearing aids are commonly designed primarily to compensate for the hearing loss, and therefore often suffer from feedback instability. In this work it is shown that the hearing aid can perform better if both the feedback problem and the required compensation are considered in the hearing aid design. An experiment was performed to study the hearing aid feedback mechanism in both an ear simulator and in human ears under various conditions. Then hearing aid filters that minimize the feedback problem for the human-ear data, while providing maximum gain for hearing loss compensation, were computed. The filter design problem was formulated as a convex optimization problem and solved using sequential quadratic programming. Results of the feedback path measurements and the hearing aid filters will be presented.
A high performance wearable signal processor system for the evaluation of digital hearing aid algorithms
Authors: Rass Uwe , Steeger Gerhard H.
Published: July 2002

Today’s commercially available digital hearing aids contain specialized hardware to keep the power consumption low. These specialized processors are not able to run arbitrary signal processing strategies. However, new audiological algorithms have to be assessed in the everyday acoustical environment of the hearing impaired before developing specialized hardware. Therefore, flexible experimental hearing aids, which include general purpose signal processors, are needed for R&D purposes. Such a system has been developed, based on the high performance signal processor Motorola DSP56303. The wearable signal processor units also include high-quality stereo analog-to-digital (ADC) and digital-to-analog (DAC) converters with 20-bit word length each. The available dynamic range exceeds 90 dB. The input and output gain can be adjusted by digital potentiometers. The housing of the device is small enough to carry the experimental hearing aid in a pocket (dimensions 150×80×25 mm). Software tools have been developed to ease the development of new algorithms. A set of configurable modules implements all hardware-dependent software routines and gives easy access to the peripherals and interfaces. A comfortable fitting interface allows easy control of the signal processing unit from a PC, even by assistant personnel. Thus the unit is best suited for use in interdisciplinary research projects.
Development of a curve to predict community annoyance due to transportation noise exposure
Authors: Finegold Lawrence S.
Published: July 2002

This paper presents a brief review of the history of the development of a curve to predict community annoyance in response to transportation noise sources beginning with the original ‘‘Schultz curve’’ [J. Acoust. Soc. Am. 64, 377 (1978)]. This seminal article provided the results of a large-scale database meta-analysis (i.e., a secondary analysis of previously published data) using data from the 11 published transportation-elated community annoyance social surveys. The Schultz curve, which relates general transportation noise exposure (in terms of day–night average sound level, DNL) and the percent of a community ‘‘highly annoyed’’ (%HA) has been used for decades in environmental impact analyses in much of the world. The community annoyance database used by Schultz has been updated with additional social survey data, nearly tripling its original size in the initial Fidell et al. [J. Acoust. Soc. Am. 89, 221–233 (1991)] analysis. A more recent version of this meta-analysis by Finegold et al. [Noise Control Eng. 42(1), 25–30 (1994)] differs from the Fidell et al. analysis on several technical issues and includes 400 data points from 22 international social surveys. The results of the Finegold et al. analysis have recently been adopted as part of a U.S. standard on environmental noise (ANSI S12.9-1996/Part 4). This paper reviews the details of this meta-analysis.
Numerical method and software package for prediction of pressure pulsation in centrifugal ventilators
Authors: Timouchev S.
Published: July 2002

A numerical method and software package was developed for designers and researchers in the field of vibration and noise problems in centrifugal ventilators. The numerical procedure is based on a representation of unsteady compressible liquid flow as a form of vortex (pseudo-sound) and acoustic mode superposition. It gives a possibility to determine absolute amplitudes of pressure pulsation in the ventilator casing induced by unsteady interaction between nonuniform flow outgoing from a centrifugal impeller volute tongue or diffuser vanes, so-called blade passing tones. Inupt data includes 2-D ventilator geometry and operation mode. Acoustic impedance boundary conditions can be defined with a circuit specification or by direct inlet and outlet specific complex impedance definition. It is possible to take into account local wall-specific complex impedance to study a coating effect on reduction of pressure pulsation. Output data represents: unsteady pressure map in the ventilator casing; amplitude distribution map for any selected harmonic; pressure versus time curves with accompanied spectrum information at any point selected by user; total vibration load vectors acting on the ventilator volute casing; and static pressure, velocity, and vorticity distribution in the centrifugal impeller. Validation of the software was completed on the centrifugal pump air model with more than 400 measuring points for two different radial gaps. The mismatch with experimental data is mostly 1.5–2.0 dB of unsteady pressure amplitude. Effects of radial clearance, number and form of impeller blades, exit and wall impedance are considered as well. The successful validation of the numerical method shows a good prospect in optimizing centrifugal ventilator geometry with improving vibration and noise characteristics at the draft stage and reducing refinement costs.