bluuebunny/crossref_metadata_2025 huggingface dataset:
Development of compact fiber-optic microphones
Authors: Cho Young C. , Gary Charles K. , Ozcan Meric , George Thomas
Published: July 2002

Advanced pressure sensors have been developed at NASA Ames Research Center, using fiber-optic technology. The development includes: compact size microphone for aeroacoustic measurements in wind tunnels with minimized flow–sensor interaction, and compact size pressure sensors for real-time measurements of flow transition over an airfoil. A prototype pressure sensor was designed and fabricated with a silicon nitride diaphragm mounted on a miniaturized ferrule. The sensor has a 3-mm diameter. Preliminary tests demonstrated excellent performance. The frequency response is steady and uniform within the design frequency limit, 100 to 5000 Hz. Its sensitivity, measured in terms of signal-to-noise ratio, is almost 10 dB better than the best condenser microphone available in a similar size (0.125-in diameter), and is better than any existing fiber-optic pressure sensor by at least three orders of magnitude. The overall performance of this sensor exceeds the initial expectation. The compact size and light weight of these sensors provide several advantages. The small size could allow tens of hundreds of sensors to be used together for applications such as microphone arrays. The flow–sensor interaction is smaller, providing more accurate measurements of pressure fluctuation of air flows over flight surfaces.
Acoustic backscatter from a sandy seafloor with anisotropic roughness
Authors: Brothers Richard J. , Heald Gary J. , Thomson Patricia A. G.
Published: July 2002

A high-frequency array was used to gather backscatter data from a sandy seabed with an anisotropic roughness spectrum. The source and receiver were fixed to a pan and tilt head so that the orientation to the seabed could be varied. A significant reduction in the backscatter was observed as the transducer was trained from the across ripple direction to the along ripple direction. The variation in backscatter with respect to the grazing angle is compared with the composite roughness model and modifications have been incorporated to take account of the bedform roughness and local grazing angle. The wideband transducer employed allowed a range of bandwidths (up to 60 kHz) to be used, including short pulse and chirp waveforms. Variation in the number of elements in the beamformed data showed the variation in backscatter due to change in beamwidth. Supporting measurements of the experiment site were made by SACLANTCEN, including stereo photography and grab samples of the sediment.
Loudness and annoyance of steady-state and temporally varying noise
Authors: Hellman Rhona , Broner Norman
Published: July 2002

Loudness and annoyance judgments were obtained by magnitude estimation for low-frequency noises. In one experiment, listeners judged the loudness and annoyance of 10-s noise bursts; in a second experiment, loudness and annoyance judgments of a continuous noise were made at regular temporal intervals over an exposure period of 1 h. All listening was through loudspeakers in an ASHRAE test room. For noise stimuli containing dominant low-frequency tones, the annoyance-to-loudness ratios ranged from 2.0 to nearly 8.0 under steady-state conditions. This large ratio variation was found despite nearly equal A-weighted levels. Moreover, consistent with a recent study [Hellman et al., J. Acoust. Soc. Am. 101, 2176–2185 (1997)], loudness decreased with time, more at low than at moderate sensation levels (SL). Perceived annoyance exhibited a similar dependence on SL with time, suggesting that loudness and annoyance are related. However, for noise stimuli with spectral peaks at 31.5 and 25 Hz, annoyance declined by a smaller percentage than loudness. This effect produced an increase in the annoyance-to-loudness ratio over the 1-h listening period. [Work supported in part by ASHRAE.]
Application of loudness quantities in a field study
Authors: Widmann Ulrich , Lercher Peter
Published: July 2002

The evaluation of noise emissions in field studies is mainly based on the concept of energy-equivalent A-weighted sound level. This averaging method has the advantage of being physically well defined and easy to compute. However, several laboratory studies showed that this method is aurally not adequate and the specific acoustic characteristics of traffic sources cannot sufficiently be discriminated. In a pilot study a good correlation was found between annoyance reactions of residents near a highway after noise abatement measures and perceived loudness. Residents’ reactions near the highway could be well predicted using a percentile value of N5. To predict the reactions for subjects living at larger distances from the source the effect of masking by environmental background noise has to be considered additionally using the psychoacoustic concept of partially masked loudness. The pilot study was restricted to a small number of inhabitants in one village. In a new field study in Austria it was attempted to confirm the preliminary results of the pilot study within a larger population. The ratings of loudness quantities, i.e., different percentile values, and of physical quantities, i.e., energy-equivalent A-weighted level and level percentiles, are investigated.
Factors influencing equal-loudness-level contours
Authors: Reckhardt Carsten , Mellert Volker , Kollmeier Birger
Published: July 2002

Data on equal-loudness-level contours taken from the literature show considerable differences between the results of different laboratories, even when using the recommended alternative forced-choice (AFC) method. Some of these differences can be attributed to free parameters in the procedures that the experimentator could change. This work investigates the influence of the experimental setup on the resulting curves obtained with an adaptive two-alternative forced-choice procedure for the 30- and 50-phon curves in the range from 200 Hz to 1 kHz. For the adaptive procedure, the amount of change in the resulting equal-loudness-level contours is measured for different free parameters (e.g., initial step size, starting level). It is shown that for a simple adaptive procedure the influence of the starting level is smaller than the range effect when using the method of constant stimuli. The effect of the starting level is reduced when using large initial step size in the adaptive procedure. However, the effect can still amount to 6 dB. This effect diminishes when using an interleaved procedure. The maximum measurement error for the different procedures is estimated. In addition, it is shown that interindividual variance is reduced when the individual threshold is taken into account.
Estimation of the sound pressure at the eardrum
Authors: Hudde Herbert , Engel Andreas
Published: July 2002

In hearing experiments or in audiological investigations, the results usually have to be referred to an appropriate input signal. Typically, the input voltage of a headphone or the sound pressure generated in a free sound field or under a different reference condition are used. In many cases, however, it would be preferable to know the sound pressure at the eardrum, as it is more closely related to the auditory sensation. If headphones are used, the reference to the pressure at drum can be approximately achieved by means of an ear simulator, such as the well-known ear simulator, according to DIN IEC 711. But the ear simulator can only reproduce the acoustical properties of an average ear. The resulting errors depend on various parameters, including those of the headphone used. A considerable portion of the error is caused by the individual ear canals. These errors alone can exceed 10 dB in the audiologically used frequency range up to 8 kHz. Some methods of estimating the pressure at the drum are investigated. Optimum results are obtained if both the sound pressure and the volume velocity at the ear canal entrance are measured. But fairly good results can also be achieved without measuring the volume velocity.
Separation of aerosol fine particles by high-power ultrasound
Authors: Gallego-Juarez Juan A. , Gonzalez-Gomez Iziar , Riera F. de Sarabia Enrique
Published: July 2002

The application of high-intensity acoustic fields to an aerosol can produce the agglomeration of the suspended particles, shifting their size distribution to a large range. Acoustic agglomeration is a process in which acoustic forces cause particles to interact and eventually to collide. The complex mechanism of this process involves orthokinetic and hydrodynamic effects. The orthokinetic effect refers to direct collisions produced among particles of different size and/or density which are differently entrained by the acoustic field. The hydrodynamic effects refer to particle interactions generated through the surrounding fluid by hydrodynamic forces induced by the acoustic field. One important hydrodynamic effect is the acoustic wake effect which is related to the asymmetry of the flow field around the moving particles. Acoustic agglomeration has a potential use in the separation of fine particles (smaller than 2.5 μm) which are emitted from industrial and residential combustion and from vehicle exhaust. In this paper recent theoretical and experimental advances in the knowledge of the basic mechanisms of acoustic agglomeration are presented, together with some results about the improvement of fine particle separation efficiency of conventional filters when the aerosol is previously treated with high-power ultrasound. [The authors acknowledge the support of CICYT-AMB96-A211-C02.]
Ultrasonic micromanipulation of particles and effects of acoustic streaming
Authors: Mitome Hideto , Kozuka Teruyuki , Tuziuti Toru
Published: July 2002

Noncontact manipulation of particles suspended in water has been studied using action of acoustic radiation pressure. A standing wave field in an unconfined vessel was used but the sound field was not necessarily resonant. When alumina suspension was poured into the sound field, particles were trapped and formed agglomeration. Changing the frequency continuously, the agglomerated particles were transported along the sound beam axis corresponding to the changes of wavelength. Selecting an appropriate frequency increment, they were transported in the opposite direction at different positions. Using a line-focused transducer with a back electrode divided into strips, it was possible to transport the particles perpendicular to the sound beam axis. A combination of these techniques realized two-dimensional manipulation without contact. Generation of acoustic streaming from the transducer to the reflector was observed even in the standing wave field due to nonuniformity of the sound field. To suppress this phenomenon for the stable manipulation, use of a tone-burst wave was proposed, noting the differences in the characteristic time between the streaming and trapping; the former is much longer than the latter and it is possible to keep the effective trapping force unchanged while suppressing the generation of streaming.
Scattering enhancements and resonator frequency shifts resulting from the interaction between ultrasonic fields and suspended particles and bubbles
Authors: Marston Philip L. , Kwiatkowski Christopher S. , Simpson Harry J.
Published: July 2002

The radiation pressure of ultrasonic standing waves in water is known to cause suspended particles to migrate to energetically preferred locations. The spatial periodicity of the resulting bands of enhanced particle concentration makes possible significant acoustic signatures of this migration even if the volume fraction for the initially uniform spatial distribution is quite small. Our initial approach used acoustic Bragg scattering of a tilted high-frequency probe beam from the induced layers [H. J. Simpson and P. L. Marston, J. Acoust. Soc. Am. 98, 1731–1741 (1995); in Nonlinear Acoustics, edited by M. F. Hamilton and D. T. Blackstock (Academic, San Diego, 1998), Chap. 14]. The scattering is enhanced because the particles in the induced layers scatter sound with what amounts to the same phase. This effect was also observed with a probe wave collinear with the pump standing wave [C. S. Kwiatkowski, Ph.D. thesis, Washington State University, 1997]. The particle migration can also shift the frequency of the standing wave resonance [C. S. Kwiatkowski and P. L. Marston, J. Acoust. Soc. Am. 103, 3290–3300 (1998)] and a related frequency shift with dilute bubble clouds has also been observed. [Work supported by the Office of Naval Research.]
Optimal resonant stimulation of sonoluminescence
Authors: Rüggeberg Matthias , Holzfuss Joachim
Published: July 2002

In single-bubble sonoluminescence, a bubble is levitated and forced to oscillate by a standing sound wave. Once per cycle, at the maximum compression of the bubble volume, the emission of a pulse of visible light is observed; due to the strong bubble collapse, a shock wave is emitted in the liquid also. It was found that these shock waves can be visualized by the means of a schlieren method. Using a stroboscopic illumination of the bubble, its dynamical behavior and the shocks can be recorded in slow motion by a video camera. Experimentally measured velocities of the shock waves were compared with numerical calculations. Further, the recorded shock waves contain information about the bubble position and collapse time. In the regime of instable sonoluminescence, sudden changes in bubble position depending on bubble size were observed. The phenomenon in the context of interactions of the bubble with the complex sound field is discussed. [Work supported by the DFG (SFB 185).]
Sonoluminescing bubbles rectify argon: Comparing predictions with latest experiments
Authors: Hilgenfeldt Sascha , Lohse Detlef
Published: July 2002

For the ASA meeting in Hawaii in 1996, the Session Chair, Robert Apfel, posed a challenge to the theoreticians working on single-bubble sonoluminescence (SBSL): experiments should be suggested and predictions made. The crucial predictions following from the hydrodynamical/chemical approach to SBSL were phase diagrams for SBSL bubbles, the rectification of argon in SBSL air bubbles, the existence of stable nitrogen bubbles, and the ejection of chemical reaction products from the SBSL bubble [S. Hilgenfeldt, D. Lohse, and M. Brenner, Phys. Fluids 8, 2808 (1996); D. Lohse et al., Phys. Rev. Lett. 78, 1359 (1997); D. Lohse and S. Hilgenfeldt, J. Chem. Phys. 107, 6986 (1997)]. Meanwhile, many beautiful experiments have been done by Crum and Matula, by Apfel and Ketterling, by Gaitan and Holt, by Suslick et al., by Barber, Putterman et al., and by many others. In this talk a comparison between these experiments and above predictions is presented.
Numerical simulations of stationary single-bubble sonoluminescence
Authors: Nogin Vladimir N. , Karlykhanov Nikolay G. , Kovalenko Gennadiy V. , Simonenko Vadim A.
Published: July 2002

In one-dimensional approximation, the approach was developed to simulate the regime of stationary behavior (groves, collapse, and rebounc-ing) for a gas-filled single bubble located in the center of a spherical water-filled vessel under an external resonant periodic sound field. The simulations were performed for realistic equations of state for water and gas (air or argon), with identification of the shock wave in the bubble, with account of surface tension on the water–gas interface, and heat conduction for the matter in the bubble (molecular and radiative). The bubble radius dependence versus time was obtained for varying external conditions: amplitude and frequency of acoustic field, temperature of water, and magnitude of surface tension. Calculated parameters for a stationary bubble (pressure in bubble vicinity, minimal and stationary radii of bubble) were demonstrated to show a good agreement with experimental data. [Work supported by CRDF.]
Measurements of peripheral compression using forward masking and pulsation threshold
Authors: Plack Christopher J. , Oxenham Andrew J.
Published: July 2002

Development of a reliable psychophysical measure of the basilar-membrane (BM) response function is of great importance, not least as a diagnostic tool for sensorineural hearing loss. Described here are attempts to estimate the response function using forward masking and pulsation threshold. Both techniques measure, in effect, the masker level needed to mask a higher frequency signal as a function of signal level. In the forward-masking design, the signal is a 4-ms tone burst presented 2 ms afer a 100-ms masker. In the pulsation-threshold design, 200-ms signal bursts are alternated with 200-ms masker bursts and the listener determines the masker level at which the signal just begins to sound ‘‘continuous’’ as opposed to ‘‘pulsed.’’ This is assumed to occur when the masker and signal excitation levels are equal at the BM place tuned to the signal. The experiemnts are based on the finding that the BM response function is roughly linear for stimuli with frequencies well below characteristic frequency (CF). A plot of masker level at threshold against signal level provides, therefore, an estimate of the BM response to the (CF) signal. Results from normal and impaired listeners will be presented and the relative merits of the techniques discussed.
Loudness and ABR findings with short signals: More compression than expected?
Authors: Kollmeier Birger , Verhey Jesko , Wegner Oliver , Dau Torsten
Published: July 2002

Previous work on loudness perception was primarily focused on stationary stimuli. However, knowledge about the processing and perception of temporally varying sounds is of special interest for various applications, such as, e.g., speech perception and hearing aids. We therefore studied the loudness of brief sounds both with psychophysical and electrophysiological methods (i.e., auditory evoked potentials) as a function of the stimulus bandwidth. The loudness summation effect appeared to be larger for brief stimuli than for long stimuli. Also, the magnitude of wave V in ABR recordings for brief stimuli with equal subjective loudness was found to depend on the stimulus bandwidth. Both results indicate that the effective compression for short signals is larger than for long signals. This finding is in conflict with predictions derived with current loudness models or with a model of the effective processing in the auditory system. The consequences of these findings for the nonlinear compression stage in auditory models will be discussed.
The cocktail party effect with conversational speech
Authors: Kondash James W. , Gilkey Robert H. , Isabelle Scott K. , Simpson Brian D.
Published: July 2002

Most often the cocktail party effect has been evaluated for relatively simple stimuli (e.g., isolated words or simple sentences masked by noise) in anechoic environments. Here, a stimulus situation is used which is similar to that employed by Plomp [Acustica 34, 200–211 (1976)] (i.e., continuous discourse for both target and masker) in a reverberant room. However, instead of asking subjects to make a subjective judgment of intelligibility, the ‘‘tracking procedure’’ of DeFillipo and Scott is used [J. Acoust. Soc. Am. 63, 1186–1192 (1978)] to simulate a conversation. This procedure measures the number of words per minute (WPM) successfully transmitted between the target talker and the subject. Preliminary results show increases in WPM of about 20% when the masker is separated from the target by 90 deg in the horizontal plane relative to the case when both the target and masker are at the same location in front of the subject. (The change in S/N necessary to maintain constant WPM between these conditions indicates that the magnitude of the effect is similar to that observed by Plomp.) Separating the target and masker(s) in elevation has little effect on WPM. [Work supported by AFOSR and the Ohio Board of Regents.]
Control of vibration of flexible manipulators
Authors: Tokhi M. O.
Published: July 2002

Flexible manipulator systems are receiving increasing attention due to their advantages over conventional rigid manipulators. The advantages of flexible manipulators are faster response, lower energy consumption, relatively smaller actuators, higher payload to weight ratio and, in general, less overall cost. However, due to their flexible nature, induced vibrations appear in the system during and after a positioning motion. This implies that in developing a control strategy for such systems, both the rigid body as well as flexible motion (vibration) have to be taken into account. A considerable amount of research work has already been carried out on the vibration control of flexible manipulators. These range from open-loop control to closed-loop control strategies of fixed and adaptive nature. This paper presents an investigation into the development of several approaches for vibration control of flexible manipulators. A single-link flexible manipulator is considered. Open-loop control techniques, including Gaussian-shaped and filtered command methods, and closed-loop control approaches of fixed and adaptive nature, including joint-based collocated control and hybrid collocated and non-collocated control, are developed. These are verified within simulation and practical rig-based environments. Finally, a comparative assessment of the performance of these strategies in vibration suppression of flexible manipulators is provided.
Multichannel feedback control for the isolation of equipment from base vibration
Authors: Serrand Marc J. , Elliott Stephen J.
Published: July 2002

Passive systems for the isolation of equipment from base vibration involve an inherent compromise between good high-frequency isolation, which requires low values of isolator damping, and limited excitation of the rigid body modes, which requires high values of isolator damping. It is known that a single-channel active control system using ‘‘skyhook damping,’’ in which a secondary force acts in proportion to the absolute velocity of the equipment, can give good damping in a lumped mass-spring-damper system without compromising the high-frequency isolation. In this experimental study a multichannel skyhook damper was used to study the isolation of a rigid mass from the vibrations of a flexible base plate. With the actuators and sensors wired together and the system excited symmetrically, a single-channel controller was initially implemented which not only showed reductions in the equipment vibration of up to 40 dB at the heave resonance at 15 Hz, but also showed substantial reductions at modes of the flexible base plate below about 200 Hz. The multichannel controller was also able to control the pitching mode of the equipment which is apparent when the system is not excited symmetrically.
Phonetic transcriptions—Aligning canonical and pronunciation forms
Authors: Hirschfeld Diane
Published: July 2002

In database preparation for concatenative speech synthesis, the alignment of canonical and pronunciation form phone sequences plays a key role for further processing. Syllabification and word boundary detection are based on canonical transcriptions, but accent detection and phrasing is coupled to the speech signal and therefore to the pronunciation form. In text-to-speech processing, canonical transcriptions are available. Unit selection is done on the speech signal and has to take into account the realized phone sequence. If a good alignment of canonical and pronunciation form of the database is available, pronunciation variants are generated implicitly in unit selection. Finally, correct alignment of phone sequences is interesting for other disciplines, too (i.e., generation of pronunciation lexica for recognition). On the bases of manual alignment for our database of a male speaker of German, the most frequent deviations from the canonical form are categorized and assigned to syllable constituents and phone combinations. For automatic alignment, a system of articulatory/acoustic features of phone combinations is developed. Similarity measures are derived from this feature system for a given phone sequence. The presented algorithm is evaluated in relation to manual alignment results.
Spoken word recognition by non-native listeners
Authors: Bradlow Ann R.
Published: July 2002

While the ability to perceive novel phoneme contrasts is necessary for accurate spoken word recognition by non-native listeners, it is likely that novel phoneme perception functions in a linguistically meaningful manner only once the contrast in question signals a known lexical contrast. In other words, knowledge of the sound-based structure of the target language lexicon is just as important for non-native speech perception as experience with the structure of the target language phoneme inventory. In order to understand non-native speech perception fully, there is a need to investigate recognition of word-sized units by non-native listeners using stimulus materials that are well controlled in terms of the sound-based structure of the target language lexicon. Accordingly, how non-native listeners recognize English words under conditions known to affect spoken word recognition by native English listeners was investigated. Results showed that the ability to take advantage of consistent surface phonetic information, such as a consistent talker across items, is a perceptual skill that transfers easily from first to second language perception. However, non-native listeners had particular difficulty with words that are easily confused with other similar sounding words, suggesting that non-native word recognition may be compromised when fine phonetic discrimation at the segmental level is required.
Phonemic repertoire effects in lexical activation
Authors: Cutler Anne
Published: July 2002

Languages differ in the size and makeup of their phonemic repertoire. Phoneme-detection experiments have shown that speakers of languages with large, confusable vowel repertoires (English, Dutch) differ in their expectations of phonetic variability from speakers of languages with small distinct vowel repertoires (Spanish, Japanese). The present study asks whether these different expectations also constrain the activation of spoken words, via two word reconstruction experiments, in which listeners reconstruct from nonwords (e.g., eltimate) a real word by changing just one phoneme (ultimate, estimate). Dutch listeners responded significantly faster and made fewer errors when required to change vowels as opposed to consonants; when allowed to change any phoneme, they altered vowels significantly more often than consonants, and vowel responses were made more rapidly than consonant responses. This suggests that for these listeners, vowel information constrains lexical activation less tightly (allows more potential alternative candidates) than consonant information. In a language with few, distinct vowels, however, this weaker constraint of vowel information on lexical activation is not found: Spanish subjects, in the same task, responded equally rapidly and accurately whether required to alter vowels or consonants; when allowed to change any phoneme, they marginally preferred to alter consonants rather than vowels.
Explaining individual differences in head-related transfer functions
Authors: Wightman Frederic L. , Kistler Doris J.
Published: July 2002

The fidelity of virtual auditory images is degraded by differences between the listener’s head-related transfer functions (HRTFs) and the HRTFs used to synthesize the virtual images. Middlebrooks (this meeting) argues that simple frequency scaling accounts for many of these individual differences. In an attempt to understand more fully both the capabilities and limitations of frequency scaling, the HRTFs from 100 individuals were analyzed. The difference between each pair of HRTFs was expressed as a single number, the average of the squared dB differences at 40 log-spaced frequencies from 0.12 to 12 kHz. A multidimensional scaling analysis (ALSCAL) was performed on the resulting difference matrix. A two-dimensional solution accounted for over 80% of the total variance. The frequency scaling factor needed for the optimal fit of each individual’s HRTF set to the HRTF set at the center of the 2-D solution was then determined. The correlation between the optimal frequency scalar and the coordinate of each HRTF set on one dimension of the scaling solution was over 0.95. Attempts to reveal a physical basis for the second dimension have been less successful. [Work supported by NIH, NASA, and ONR.]
Characterization of reflected hydroacoustic signals for ctbt localization
Authors: Angell Jeffrey , Farrell Ted , Pulli Jay
Published: July 2002

The sparse nature of the proposed IMS hydroacoustic network implies that maximizing the information from each sensor is critical to overall network performance for nuclear event localization. Performance may be enhanced using information contained in late-time reflected arrivals from continental shelves, islands, and seamounts. The primary focus here is to characterize reflected signals and determine the conditions under which reflections are likely to occur. Several data sets, including a 1970 ship-scuttling explosion conducted in the North Atlantic (CHASE 21 event) are analyzed. Travel-time ellipses and bearing estimates generated from measured arrival times of waveform envelopes are used to determine possible geographic locations of acoustic reflections. Environmental characteristics at these locations are examined in order to understand the conditions under which reflections are likely to occur. Time- and frequency-domain characteristics of both the direct arrivals and reflections are measured and discussed. [Work sponsored by Defense Special Weapons Agency Contract Number DSWA01-97-C-0164.]
Rapid response, a demonstration of ocean assessment capabilities
Authors: Sellschopp Jürgen , Akal Tuncay
Published: July 2002

The suitability of sound waves as a carrier of information in the ocean is known to be highly variable. In the past, the assessment of the controlling environmental parameters has typically been subject to post-cruise analysis and often took years. SACLANT Undersea Research Centre and NATO’s organization for military oceanography agreed upon a three-year series of surveys in the Mediterranean Sea aimed at the development of a methodology for rapid environmental assessment (REA). A large number of research ships and patrol aircraft were provided by various nations. Researchers of numerous institutes were engaged in the REA trials. Time-invariant parameters such as bottom properties were measured directly or determined by inversion. The results were transferred for optional replacement in a standard database. Time-dependent parameters such as the sound velocity structure were predicted by means of real-time ocean models initialized and updated with survey data. The infrastructure for timely delivery of REA products and forecasts was developed, the main requirements being appropriate data channels and data fusion capacity. Rapid response was the first demonstration by the research community that the critical characteristics of the acoustic environment can be provided by the end of a mission.
Extracting sea-bottom information from reverberation data
Authors: Ellis Dale D. , Preston John R.
Published: July 2002

Reverberation measurements contain information about both the scattering and propagation effects. This paper will discuss our efforts to extract bottom loss and bottom scattering from reverberation data, including at-sea results obtained during three recent Rapid Environmental Assessment (REA) trials with the SACLANT Undersea Research Centre. There is often a strong interrelationship between the bottom loss and scattering, and attempts to deal with the uniqueness of the extracted parameters are discussed. Earlier attempts using another simplified, but automated, inverse scheme indicated very good agreement with independently measured bottom loss. In the recent REA trials the initial bottom loss was obtained using some a priori geologic information for the experiment area. Then the bottom parameters were manually adjusted to obtain a good fit to the data from several nearby locations and array headings. The best fits were obtained by first adjusting the bottom loss to fit the rate of decay, and then adjusting the strength of the Lambert coefficient to fit the level.
Room acoustical measurements and architectural documentation of auditoriums in Berlin considering the speech intelligibility in dependence of spatial distribution of sound fields
Authors: Krämer Peter S.
Published: July 2002

Measurements of spatial room acoustical impulse responses were performed to provide a database for the examination of speech intelligibility (SI) in real rooms. As is well known, the spatial distribution of reflections can influence the SI, especially in reverberant rooms. Nevertheless, the qualitative and quantitative dependences are mostly unknown. Room acoustical measurements thus were realized using a dodecahedron loudspeaker for measuring reverberation time, an interference-free loudspeaker with omnidirectional characteristic for recording impulse responses, and an artificial speaker and a dummy head microphone for measurements of SI criteria and recording of logatom syllables. A specially constructed spaced four-capsule microphone in tetrahedron arrangement was used for measuring also. The four-microphone analysis technique can recognize intensity and direction of the early reflections [Odjija and Krämer (1999)]. Results of the measurements will be documented and published on a CD-ROM, together with the architectural data of the auditoriums. First results of the room acoustical measurements are presented. Prior studies have shown that reproducing the measured binaural cues with headphones does not influence the SI significntly. Subjective measurements of SI and analysis of spatial dependences will be performed in a second stage of the project.
The interrelation between the music sound clarity index C80 and the subjective evaluation of the hall acoustics
Authors: Stauskis Vytautas J.
Published: July 2002

It has been established by acoustic investigations in actual halls that the clarity index C80 may have optimum recommended values, while subjectively the sounding of music in those halls may be evaluated as poor. Three halls were selected for investigations: (1) a church with the volume of 27000 m3, (2) Vilnius Opera Theatre with the volume of 8700 m3, and (c) a concert hall with the volume of 4900 m3. In hall (1), the values of C80 are from −1.4 to −3.9 dB. The reverberation time is 6.6 s. According to subjective evaluations, the acoustics of this hall is suitable neither for classical nor romantic music. In hall (2), the values of C80 vary from −1.1 to 2.4 dB in all rows in the stalls. The reverberation time is very short and varies between 1.2 s. The sound of an orchestra in the hall is evaluated as very poor. In hall (3), the values of C80 are within the range of −1.0 to 2.8 dB and the reverberation time is 1.55 s. The sound of chamber music is evaluated as good, while that of symphony music is evaluated as satisfactory. This shows that this subjective indicator may not provide full characterization of the hall acoustics.
Validation of a system for synthesizing distant signatures of underwater explosions for sea mammal hearing studies
Authors: Clark Joseph A. , Young Jane A. , Gaspin Joel B.
Published: July 2002

A system for synthesizing distant signatures of underwater explosions was described in an earlier talk [J. A. Clark and Jane A. Young, J. Acoust. Soc. Am. 102, 3177(A) (1997)]. The system is now being used to study expected temporary threshold shifts (TTS) of sea mammals near underwater explosive test ranges in the Baltic sea. The results of these studies will be used to define safe range criteria for mammal hearing which will then be incorporated into the methodology for assessing potential damage to marine mammal hearing. Typical underwater explosion signals (time histories) were selected from an extensive set of blast signals computed for a variety of charge weights, explosion depths, ranges from the blast, and water conditions specified by sound-speed profiles and water depths. The selected signals were synthesized and measured at the location of the test mammals (bottlenose dolphins). The fidelity of the synthesized signals to the computed waveforms was determined by comparing such characteristics as peak pressures, positions of major spikes in the time histories, and one-third octave band energy spectra. Examples of typical signals used for the TTS studies and their measured characteristics will be presented.
Biosonar in a textured world
Authors: Müller Rolf , Kuc Roman
Published: July 2002

Capabilities of bats to analyze spectral profiles have been demonstrated by several researchers. However, suggestions as to the informational content of the biologically accessible features in terms of the echo-generating process are not yet entirely convincing. Invoking a general analogy to visual color perception does not provide an operational definition of the hypothetical perceptual mechanism. The proposed mechanisms are mostly restricted to a scenario, where the spacing of two reflectors may be determined via the resulting comb-filter transfer function. The generality of this concept is curtailed by most natural surfaces constituting aggregates of multiple reflecting facets with a pronounced random component to their arrangement. Classification of natural textures is conceived to be a pivotal capability which may underly performance in several tasks probably performed by the animals, e.g., landmark identification and control of scanning movement. In order to understand how the demands of texture classification relate to the key features of biological sonar systems, the following steps are taken: Starting from a database of textures typical of bat’s habitats (i.e., foliage), the implications of two paramount attributes of biological sonar systems are studied: sensor mobility and a joint time-frequency representation encoded in spike patterns.
High-resolution target strength measurements in deep water
Authors: Ona Egil , Svellingen Ingvald
Published: July 2002

High-resolution target strength data can only be claimed valid in situations where it can be safely shown to be much less than one target per pulse volume. Many fish species occur in densities and at depths where this demand hardly can be met with standard, hull-mounted, survey transducers. This paper describes a new and simple method for obtaining such data with some examples of target strength and target tracking data obtained in deep water at three different cruises. The system used is the Simrad EK-500 split beam echo sounder, connected to an oil-filled ES38D, pressure-resistant transducer, on cable lengths from 400-800 m. The transducer was lowered as a probe the desired depth, often inside or close to the fish layer, with the vessel stationary, or slowly drifting. For maximizing the number of detections per fish, the echo sounder was operated at maximum pulse repetition frequency. Recordings shown are from three different species, small myctophid fishes at 200–400 m, hake at 200–400 m, and herring at 50–400 m. High-quality target strength distributions were frequently obtained in less than 1 h at a typical target strength station.
Design of a transducer-driving system scanning around an aquatic target in an indoor tank
Authors: Ishii Ken , Okumura Tsuyoshi , Sawada Kouichi , Miyanohana Yoichi
Published: July 2002

A squid has been specified as a kind of fish that has to been managed scientifically. Estimation of its resources via acoustic surveys is required. But the basic data of the target strength (TS) of a squid are limited. So far, in the measurements of the TS pattern of fish, echo levels are observed while changing the aspect of a fish. On the other hand, since a squid is more fragile than a fish, and unstable in shape, the rotating unit of fish is not available for a squid. So the positioning system that makes a transducer scan precisely around an aquatic target suspended in the water has been designed. This system is loaded in the electric flatcar, the probe of which can move three dimensionally, on the top of a tank which is 10 m in width, 15 m in length, and 10 m in depth. It is hoped that precise TS pattern measurements are possible so that the target is suspended without damage owing to this transducer-driving system. The detail of control software and an example of measurements and subjects are described in this paper.
Intensive training on acoustics of pipe organs for organ builders and organists
Authors: Angster Judit , Miklos Andras
Published: July 2002

A special course on acoustics is organized regularly in the Fraunhofer-Institute fr Bauphysik (IBP) for training pipe organ builders, official organ experts, and organists. The title of the 4-day intensive course is ‘‘Pipe Organ and Church Acoustics’’ and it is scheduled at the end of January every year, when most of the organ builders are less occupied than in other periods. The number of participants is limited to 20, but the course is repeated in February, if necessary. The working language of the course is German, but English and French are also possible. The curriculum of the course is based on the ongoing research on pipe organs in the IBP and it is divided to three parts: fundamentals of acoustics; recent results of research on organ and church acoustics; and laboratory demonstrations and practical applications. During the demonstrations there is a limited possibility to perform measurements suggested by the participants. The content of the lectures and demonstrations is upgraded every year with the newest results of the research. In this paper the experiences of the eight courses held up to now will be discussed. A short video clip showing some of the laboratory demonstrations will also be presented.
An artificial neural network model for analysis and synthesis of pianists’ performance styles
Authors: Bresin Roberto
Published: July 2002

A model for synthesis and analysis of professional pianists’ performance styles by means of artificial neural networks (ANNs) is presented. The style of pianists was synthesized using Bruno Repp’s measurements [J. Acoust. Soc. Am. 92, 2546–2568 (1992)] of Schumann’s Trumerei as performed by 24 famous pianists. It will be shown that, by using significant score parameters, ANN can learn and reproduce the style of a pianist at a local level. The behavior of the trained ANN and the performance characteristics of the pianists were analyzed by running the trained ANN in a special way. All ANN input neurones were locked to fixed values, except one or two neurones whose input values were allowed to change in time. For example, if one of these neurones is related to the duration of the current note, the output of the ANN gives a curve showing expressive deviations produced by the ANN, when the duration of the note is varying. In this way it is possible to extract behavior rules of performing action of the ANN, and compare them with the performance style of pianists as well as of rule-based performance systems. The model is developed in the Java language and is available on the Internet.
Empirical formulas for noise of high speed trains
Authors: van Haaren Rik , Looijmans Karel
Published: July 2002

Aeroacoustic sources at high-speed trains are dominant noise sources for trains running at high speeds. The pantograph is an important source located at a high position. Therefore, this source cannot be handled by applying noise barriers. Noise measurements are carried out in the DNW windtunnel for a ADtrans pantograph, using a 2-D microphone array. Measurement results show the noise sources at the pantograph to be located in three areas. The top area is the most important source. Extensive tests with several pantograph configurations indicate that the noise is not dominantly emitted by some specific parts, but more by a large number of parts as a whole. By using empirical formulas, models are formulated to estimate the broadband noise emission of the pantograph, as well in amplitude as in spectral distribution. Analysis of the experimental results indicate that the turbulence in the wake is the dominant noise generating mechanism. The noise levels scale with a power of six to the flow velocity. On the basis of the models it is expected that noise reductions in broadband noise up to 10 dB are possible. Inflow turbulence is an important noise-generating mechanism at the foot region; however, this can easily be handled by adapting the flow around the foot region with a spoiler. The numerical models can be used to design low-noise train parts. The detailed noise measurement results supply important information that can be used for the design of acoustically optimal constructions.
Aeroacoustic simulations and stochastic approach using linearized Euler’s equations
Authors: Bailly Christophe , Juvé Daniel
Published: July 2002

Sound generation and propagation in a turbulent flow is a very difficult numerical problem. Indeed, acoustic fluctuations are very small by comparison to the mean and turbulent fields. Linearized Euler’s equations provide accurate numerical solutions only in working on perturbations. In the stochastic noise generation and radiation model, the mean flow is calculated by solving the Reynolds-averaged Navier–Stokes equations. Then, Euler’s equations are linearized around this mean flow field. A turbulent source term is introduced in the equations, and the turbulent velocity field is modeled by a sum of random Fourier modes. The numerical solution is obtained by using a dispersion-relation-preserving scheme in space, combined with a fourth-order Runge–Kutta algorithm in time. In order to validate the numerical method, radiation of multipolar sources in a uniform and sheared mean flow has been investigated in a two-dimensional case. The calculated solutions favorably compare with analytical solutions or with ray tracing. Then, the SNGR model is applied to a subsonic jet noise in a 3-D geometry. Numerical results and future developments will be discussed.
Acoustic radiation due to gust–airfoil and blade–vortex interactions
Authors: Agarwal Ramesh K.
Published: July 2002

The problem of acoustic radiation due to gust–airfoil and blade–vortex interaction in compressible mean flow is considered numerically. The acoustic perturbation field is calculated by solving the acoustic equations derived from the unsteady Euler equations by linearizing about a steady mean flow and assuming a single frequency disturbance. A computational code is developed which is validated by computing the unsteady loads on a flat plate due to a gust in incompressible flow (Sears problem), and unsteady loads and acoustic radiation due to a compact and noncompact gust in compressible flow. For these three cases the numerical results are compared with the exact analytical solutions and asymptotic solutions. Excellent agreement is obtained. The validated code is then employed to perform parametric studies by varying the Mach number of the mean flow, the angle of attack, and the geometry (thickness and camber) of the airfoil. Conclusions are drawn about the magnitude and phase of acoustic radiation due to gust–airfoil interaction for various Mach numbers, angles of attack, and airfoil geometries. The validated code is also employed to compute the acoustic radiation due to blade–vortex interaction. Calculations are compared with those of other investigations for both the mean flow and the acoustic variables.