bluuebunny/crossref_metadata_2025 huggingface dataset:
Across-speaker and within-speaker variability of British English sibilant spectral characteristics
Authors: Seitz P. F. D. , Bladon R. A. W. , Watson I. M. C.
Published: October 2005

Testing hypotheses regarding the potential as phoneme discriminators of certain properties of “physical” and “auditorily transformed” sibilant spectra [A. Bladon and F. Seitz, J. Acoust. Soc. Am. Suppl. 1 80, S19 (1986)], the present study draws on a database of 20 male and 20 female talkers, containing over 2000 tokens of /s/, /∫/, /z/, and /ȝ/ in a variety of phonetic contexts and speaking styles. Two automatic parameter extraction routines were used to obtain measurements, for each token, of physical and auditorily scaled values of the main spectral peak, gradients of the leading and trailing spectral edges, and bandwidth of the peak. Especially in females' productions of the apicals, large and characteristic variations in these parameters were observed as correlates of individual talker, phonetic context, and style. Overviews of the envelopes of physical and auditory variation are given; based upon this knowledge, a speaker-adaptive strategy for discrimination of British English sibilants is proposed. [Work supported by the Alvey Programme.]
Automatic detection for diphthongs
Authors: Chang Hisao M.
Published: October 2005

It is generally recognized that diphthongs do not have steady states and only display identities through a dynamic shift of their spectra. In Miller's auditory-perceptual theory, dynamic shifts of diphthong spectra are represented as paths on a “vowel map” [J. D. Miller, J. Acoust. Soc. Am. Suppl. 1 81, S16 (1987)]. From the utterances of English digits by two male and two female native speakers, an auditory-perceptual transformation system and a segmentation rule were used to transform the spectral prominences contours derived from those utterances containing diphthongs/aɪ/ and /eɪ/ in paths on the vowel map, and it was found that all paths from the same diphthong, independent on talker's sex and phonetic contexts, travel in the same route and form a “diphthong pip.” After the detailed analysis of the paths, an automatic detection algorithm was developed based on angles, durations, lengths, origins, and other attributes of the paths. When applying the algorithm to digit utterances from 21 new talkers, the recognition accuracies for /aɪ/ and /eɪ/ were 75% and 90% while the insertion rates were only 1.1% and 4.5%, respectively. [Work supported by NINCDS and done at Central Institute for the Deaf, St. Louis MO.]
Speaker-independent automatic vowel recognition based on overall spectral shape versus formants
Authors: Zahorian Stephen A. , Jagharghi Amir J.
Published: October 2005

Automatic recognition experiments were performed to compare overall spectral shape versus formants as speaker-independent acoustic parameters for vowel identity. Stimuli consisted of four repetitions of 11 vowels spoken by 17 female speakers and 12 male speakers (29*11*4 = 1276 total stimuli). Formants were computed automatically by peak picking of 12th-order LP model spectra. Spectral shape was represented using three methods: (1) by a cosine basis vector expansion of the power spectrum: (2) as the output of a 16-channel, 1/3-oct filter bank; and (3) as the output of a 16-channel mel-spaced filter bank. Automatic recognition was based on maximum likelihood estimation in a multidimensional space. For all cases considered, the representations based on spectal shape resulted in significantly higher recognition accuracy than for recognition based on only three formants. For example, using the entire database of all speakers and 11 vowels, recognition based on spectral shape was about 85% vs 69% for three formants. If the data were restricted to female speakers and the seven vowels /a,i,u,æ,ɝ,ɪ,ɛ/, recognition was about 97% based on spectral shape versus 84% for formants. These results indicate that, at least for automatic recognition of vowels, spectral peak detection is neither necessary nor sufficient. [Work supported by NSF.]
Formant tracking using statistical pattern recognition
Authors: Gayvert Robert T. , Hillenbrand James
Published: October 2005

Formant trackers that use peak-picking algorithms and ad hoc rules tend to suffer from occasional severe mistakes in labeling formants. Recently, Kopec [IEEE Trans. Acoust. Speech Signal Process. ASSP-34, 709–729 (1986)] used a statistical approach based on hidden Markov models to avoid these kinds of problems. This study uses another statistical technique based on discriminant analysis. A model is trained on handmarked formant trajectories and various features from a collection of utterances. The same features are then extracted from a test utterance and used to determine the most probable formant frequencies. Any combination of features, including peaks from LPC and DFT spectra, can be used with this technique. Preliminary results indicate that this method avoids serious mistakes. Results of evaluating the formant tracker on a large number of training and test sets will be presented. [Work supported by Rome Air Development Center under contract F3060285-C-0008.]
A unitary multielement ultrasonic array for focused imaging in air
Authors: Kay Leslie
Published: October 2005

A condenser microphone is described that incorporates 20 independent conductive elements coupled to a single laminar diaphragm. The elements are separately excited by an electrical signal, delayed in time relative to adjacent elements, to enable the formation of a focused beam within the close nearfield of the array. Using a radiation frequency covering an octave bandwidth from 100–200 kHz, a spatial resolution of 0.2 in. parallel to the aperture and 0.1 in. perpendicular to the apperture at the point of maximum focus has been achieved. No aperture shading has been incorporated at this stage so that the diffraction field can be explored. Since this is formed by a very wide bandwidth radiation, an unusual field plot was expected. The results are discussed and a video shows examples of the high resolution real time imaging in air that can now be achieved. This has many applications, particularly in automation as an alternative to optical vision when distance to an object is essential information.
Directional processing for interference reduction in hearing aids
Authors: Rabinowitz W. M. , Peterson P. M.
Published: October 2005

Listeners with hearing impairments often have great difficulty in interpreting acoustic environments having multiple sources and/or reverberation. Directional processing, obtained by weighting the outputs of multiple sensors, offers potential for reducing such interference, although cosmetic considerations impose fundamental limitations on performance. In fixed processing, weights are held constant to achieve a fixed antenna pattern (pointing straight ahead) and are chosen as a compromise between classical and supergain limits so as to maximize directivity with tolerable sensitivity to receiver noise. A particular second-order gradient microphone system, with a sensor span of 4.1 cm and formed simply by subtracting the outputs of two cardioid capsule hearing-aid microphones, exhibits a directivity of 8.5 dB, which is near optimum (9.0 dB) for its noise sensitivity. In adaptive processing, weights are continually adjusted according to properties of the environment. An adaptive algorithm using two omnidirectional microphones separated by 20 cm (≈ headwidth) has been evaluated with a target talker straight ahead and an interference source at 45° in three computer-simulated environments with zero, moderate, and severe reverberation. Both analytic measures and intelligibility tests indicate substantial reductions in interference in zero and moderate reverberation and no reduction, or degradation, in severe reverberation. Additional and more realistic evaluations of these fixed and adaptive systems are required to determine their suitability for practical hearing aids. [Work supported by NIH.]
Automatic speech-seeking microphone arrays for teleconferencing
Authors: Elko G. W.
Published: October 2005

The effects of reverberation on speech transduction in large rooms can severely limit the intelligibility of the desired speed signal. In isotropic acoustic fields, aside from locating the microphone in close proximity to the speech source, it is necessary to develop highly directional microphone systems. Concomitant with microphone array designs for teleconferencing is the need to steer the array at the desired speech source in order to maximize the direct-to-reverberant signal energy. The problem that will be discussed is that of designing broadband (speech bandwidth) microphone arrays that are capable of detecting a speech source and automatically steering the array to the direction of the source.
The evaluation of source strength using a microphone array over a hard reflecting surface
Authors: Glegg Stewart A. L. , Smart Richard C.
Published: October 2005

This study has been undertaken to measure the noise source height on highway vehicles. A vertical seven-element microphone array is used and the method of analysis is based on the source breakdown technique developed by Fisher and Tester. This is capable of giving narrow band or 13-oct source spectra for each individual source in a distribution, with resolution down to zero frequency, providing the approximate source positions are known, and there is a high signal-to-noise ratio. This method has been evaluated both in an anechoic chamber and in an outdoor environment over a hard surface. Good results have been obtained for sources which are 0.13 m apart down to 125 Hz. However, if the source positions are not exactly specified, the lower limiting frequency is increased to 500 Hz. [Work is sponsored by the Florida Department of Transportation.]
Characterization of the envelope protecting an acoustic detection system
Authors: De La Croix D. Vaucher , Lathuilière M. , Périer L.
Published: October 2005

In a previous paper, a scheme for computing the radiation perturbation due to a two-dimensional envelope of arbitrary shape enclosing an acoustical source was presented [D. Fichot and J. M. Parot, D.3-5, 12th ICA, Toronto (1986)]. The formulation associates structural finite elements and acoustic integral equations. Though originally aimed to solve fluid-structure interaction problems with strong coupling between transducer and envelope for ka values up to 30 (k = acoustic wavenumber, a = half-largest dimension of the envelope), the code named ABER2D was applied to determine the influence of the envelope shape and/or material on the directivity of the enclosed transducer with air as ambient fluid. Deviations from the free field reference configuration have been computed at low ka values (typically ka ≃ 1), starting from a thin circular envelope and progressively flattening it to elliptical shape. Such calculations have oriented choices concerning geometry and certain material characteristics and resulted in designing a flattened polypropylene protection envelope adapted to an acoustic detection system, which displays good transparency in the frequency band of interest. [Work supported by SEFT.]
Unidirectional, second-order gradient microphone
Authors: West J. E. , Sessler G. M. , Kubli R. A.
Published: October 2005

A second-order gradient microphone with unidirectional characteristics is described. The microphone consists of two commercially available, first-order gradient electret microphones that are properly baffled and arranged with their rotational axes in line. The output of one of these transducers is subtracted from the delayed output of the other transducer. The unidirectional microphone shows a directional characteristic that is relatively frequency independent, has a 3-dB beamwidth of the mainlobe of ± 40 degrees, and exhibits sidelobes 10–20 dB below the mainlobe. After equalization, the frequency response of the microphone in its direction of maximum sensitivity is within ± 3 dB between 0.3 and 4 kHz. The sensitivity of the microphone, including its equalization, is − 60 dBV/Pa at 1 kHz and the equivalent noise level for the above frequency range amounts to 28 dB linear or 20 dB(A).
Practical acoustic beams and synthesized source fields for turbulence detection at the lower atmosphere
Authors: Adekola S. Adeniyi
Published: October 2005

Computer simulations of new analytical results that are important extensions of an earlier work [S. A. Adekola, J. Acoust. Soc. Am. 76, 345–368 (1984)] are here presented for the echosonde (acoustic echosounding) system. It is shown that the source distributions synthesized from the preassigned directive pressure fields are not only confined within finite antenna aperture regions, but are also characterized by maximum intensities at the boresight region of the aperture and are considerably attenuated towards the rim of the antenna cuff. It is also shown that if the beamwidth of the pattern prescribed is extremely narrow, then the source distribution synthesized from it tends to produce an undesirable strong field or singularity at the rim of the antenna cuff. The paper then focuses attention on the factors governing the realizations of practical echosonde patterns; which are suitable for turbulence detection of atmospheric irregularities, such as thermal structures, dynamics, and turbulent velocity fields at the lower atmosphere; and from which physically realizable source fields exhibiting no abrupt discontinuities across the antenna aperture can be synthesized. Last, a comparative analysis shows that the approximate patterns generated from the synthesized source distributions manifest good mean-square fits to the idealized acoustic beams originally specified almost all over the entire visible ranges of the patterns prescribed. [Work supported by Lagos State University, Badagry Expressway, Ojo, Lagos, through a Visiting Professorial Appointment.]
The effect of varying the amplitude-frequency response on the masked SRT for sentences in hearing-impaired listeners
Authors: van Dijkhuizen Janette N. , Festen Joost M. , Plomp Reinier
Published: October 2005

A multichannel gain-control hearing aid, in which the frequency-dependent amplification is adapted automatically to the fluctuations of the incoming signal,. may optimally deliver speech to an impaired ear. Such a system requires that the speech-reception threshold (SRT) in noise is, within limits, unaffected by dynamic variations in the amplitude-frequency response of the hearing aid. For normal-hearing listeners, van Dijkhuizen et al. [J. Acoust. Soc. Am. 81, 465–496 (1987)] found that the masked SRT for sentences is remarkably resistant to dynamic variations in the slope of the amplitude-frequency response when it shaped both speech and noise. In this experiment, we studied corresponding conditions for hearing-impaired listeners. Again, the amplitude-frequency response shaped both speech and noise and thus left speech-to-noise ratios untouched. The results obtained so far also indicate that, for hearing-impaired listeners, dynamic variations of the amplitude-frequency response do not affect the SRT.
Discrimination of time-expanded tonal sequences
Authors: Sorkin R. D. , Snow M. P.
Published: October 2005

Subjects were presented with two sequences of tones, each consisting of eight 50-ms sinusoidal tones. Subjects had to discriminate whether the sequences were the same or different based solely on the frequency pattern defined by each sequence of tones. The correlation between the temporal envelopes of the two sequences, the variability of sequence timing, and the length of the intersequence interval were manipulated. In the “correlated” condition, the gaps between the tones in the sequences were randomly varied across trials (sequences within a trial had identical temporal envelopes); in the “uncorrelated” condition, the envelopes varied both across and within trials. The results replicated an earlier experiment [R. D. Sorkin, J. Acoust. Soc. Am. (in press)]. Performance in the correlated condition was independent of temporal variability but decreased with increases in temporal variability. This pattern of results is consistent with the assumed dominance of “trace” and “context” mode processing in the two conditions. In an additional manipulation, all tone and gap durations in the second sequence of each trial were expanded by 40%. Performance in the time-expanded condition resembled that found in the correlated condition. This result is interpreted as evidence for a duration scaling process in auditory memory. [Work supported by AFOSR.]
Proportional target-tone duration as a factor in the discriminability of tonal patterns
Authors: Watson Charles S. , Kidd Gary R.
Published: October 2005

A series of experiments on listeners' abilities to extract information from patterns with varying total durations and numbers of tonal components has previously been reported [J. Acoust. Soc. Am. Suppl. 1 73, S44 (1983); 77, S1 (1985)]. In those experiments listeners were tested in high-stimulus-uncertainty, same-different pattern discrimination tasks, in which the tonal patterns to be discriminated differed by changes in the frequency of one or more components. Discrimination performance in those tasks was consistent with previous measures of the frequency resolving power of the auditory system when the patterns contained one to three equal-duration components, for total pattern durations from 62.5–2000 ms. As the number of components was raised, discrimination thresholds increased by large amounts, often by factors of 10–100 for patterns with more than seven-eight components. While this result might imply an informational limit on pattern processing, it is also consistent with the hypothesis that target tones are equally well resolved if they occupy equal proportional durations of the patterns in which they occur. Results of a new experiment, in which proportional durations of target tones and the number of tones per pattern were independently varied, suggest that proportional duration of the target tones is in fact the primary determinant of pattern discriminability for tonal patterns ranging from 100–1500 ms in total duration. [Work supported by AFOSR and NIH/NINCDS.]
Critical bandwidth for modulation detection
Authors: Yost William A.
Published: October 2005

Two sinusoidal carriers separated by frequency F were each sinusoidally amplitude modulated (AM). Subjects were asked to discriminate between a complex in which the two AM signals had the same phase and a complex in which the phase spectra of the lower frequency AM components were delayed relative to the upper frequency AM components. The threshold value of the phase shift was measured from three-point psychometric functions obtained from a two-alternative, forced-choice task. A broadband-noise background and random variation in overall level were used to force the two normal-hearing listeners to attend to the temporal structure of the two-tone AM complexes. The frequency separation between the two carriers (F), the rates of AM modulation, and the form of the phase shift (entire waveform or envelope delay) were parameters in the study. The results indicate that the threshold for detecting the phase shift is small when the two carriers are within a critical band. However, listeners are still quite sensitive to the phase shifts for very large frequency separations between the carriers. The results will be discussed in terms of the auditory system's ability to use temporal modulation to identify common sources in a complex sound. [Work supported by AFOSR-Life Sciences and the NINCDS.]
Comodulation masking release for envelope decorrelation cue versus envelope level difference cue
Authors: Hall Joseph W. , Grose John H.
Published: October 2005

In past investigations of comodulation masking release (CMR) it was difficult to identify the detection cue used by the listener, as several different cues were available. The present study attempted to isolate two cues: envelope correlation and envelope level. The signal was either a 10-Hz-wide noise band centered on 500 Hz, or a 500-Hz pure tone. The maskers were 10-Hz-wide noise bands centered on 400 and 500 Hz (either comodulated or noncomodulated). In one experiment the same 10-Hz-wide noise band centered on 500 Hz was used as both the signal and the masker. Here, the signal resulted in no decorrelation of envelope, but did result in an envelope level difference. In the experiment using the 500-Hz pure tone as the signal, the signal resulted in a decorrelation cue; however, any envelope level cue was rendered useless in this experiment by randomly roving the level of the flanking band between each interval, and by adjusting the noise level of the on-signal band in the dummy intervals to the level of the noise-plus-signal in the signal interval. The major result was that substantial CMRs occurred both when the envelope level cue was available but the decorrelation cue was not, and when the decorrelation cue was available but the envelope level cue was not.
Comodulation detection differences with multiple signal bands
Authors: Wright Beverly A. , McFadden Dennis
Published: October 2005

Detection thresholds were determined for signals consisting of one, two, or five noise bands embedded in eight “cue” bands. The center frequencies of the signals ranged from 1250–3250 Hz in 500-Hz steps, and those of the cue bands ranged from 500–4000 Hz in 500-Hz steps. The temporal envelopes of the cue bands were either all correlated or all uncorrelated. The envelopes of the multiple signal bands (1) were correlated among themselves as well as with the cue bands, (2) were themselves correlated, but not with the cue bands, or (3) were themselves uncorrelated. As in previous comodulation experiments, thresholds were similarly poor when the signal and cue bands were either all correlated or all uncorrelated. Detectability was typically 7–10 dB better when the signal bands were uncorrelated with the all-correlated cue bands. These results were no different for single- and multiband signals. Thus the auditory system appears better able to detect both simple and complex signals in background having the same—rather than different—temporal envelopes, as long as the temporal structure of the signal(s) is different from that of the surrounding sounds. [Work supported by NINCDS Grand NS15895.]
Rate and number as limiting factors in the perception of sequences of syllables
Authors: Watson Charles S. , Espinoza-Varas Blas
Published: October 2005

Listeners' abilities to process the details of sequences of sounds have been shown to be limited by two major factors: the number of components in the sequence and the rate at which the components are presented. Individual differences in the ability to process large numbers of components at high presentation rates have been proposed to account for certain language- or learning-related disabilities. Little attention has been paid to the possible interactions between these two factors, nor to the distribution of these abilities in the population of normal-hearing listeners. A test was developed to measure the effects of the number of syllables and, independently, those of syllable rate, or listener's abilities to process syllable sequences. The sequences consist of two, four, or seven natural tokens of the spoken letters b, c, d, g, p, t, or v, produced as CVs ending in the vowel /i/. The duration of each token was edited to 75 ms. Syllable onset-to-onset times were 75, 150, and 300 ms. The listener's task was to transcribe the letters in each sequence, in a test format in which 180 sequences were presented, varying randomly in number and rate. Results for an initial test group show some individual's performance to be more limited by rate rather than by number, and vice versa. Implications for the interpretation of certain language-learning disorders are discussed. [Work supported by NIH/NINCDS and AFOSR.]
Informational limits in speech processing by normal and impaired listeners
Authors: Espinoza-Varas Blas , Watson Charles S. , Kyle Mary S.
Published: October 2005

It has been reported that some impaired listeners preceive speech as being “jumbled, or as going too fast,” which suggests that their informational capacity may be reduced. Recently, work with nonspeech stimuli [Watson and Foyle, J. Acoust. Soc. Am. 78, 375–380 (1985)] has shown that the amount of spectral-temporal information that can be extracted from sequential patterns can be limited very strongly by the number of freely varying components included in a pattern, and only weakly by the duration of the individual components. Using a variation of the speech test described in the previous paper, information capacity of normal and impaired listeners was compared for sequences of two, four, or seven syllables, presented at rates of 6.5 or 3.9 syllables/s. Percent information transmission (IT%) was estimated for each temporal position of the sequences, number of syllables, and syllable rate. The sensorineural impaired listeners had nearly uniform 45-to-55-dB losses, and ages similar to the normal listeners. Stimuli were presented at approximately constant sensation levels (30–40 dB) to minimize effects of differences in sensitivity. The maximum number of syllables yielding IT% ⩾ 50% was, on the average, 6–7 for normal listeners but 4–5 for the impaired listeners. [Work supported by NIH/NINCDS and AFOSR.]
Dynamical forecasting of mesoscale fronts and eddies for acoustical applications
Authors: Robinson Allan R.
Published: October 2005

Real-time nowcasts and forecasts of oceanic mesoscale fields (velocity, pressure, temperature, sound speed, etc.) are now feasible and have been initiated in selected regions. Predicting the “internal weather of the sea” is analogous to meterological weather forecasting. Our approach is based on recent progress in phenomenological knowledge and modeling capabilities and is systematic [A. R. Robinson, “Predicting open ocean currents, fronts, and eddies,” in Three-Dimensional Ocean Models of Marine and Estuarine Dynamics, edited by Nihoul and Jamart (Elsevier, Amsterdam, 1987)]; i.e., it combines dynamical model [A. R. Robinson and L. J. Walstad, “The Harvard open ocean model,” J. Appl. Numer. Math 3 (1987)] and observational estimates via data assimilation methods. The prediction problem will be overviewed, results from an ongoing forecast scheme for the Gulf Stream (gulfcasting) will be presented, the coupling of dynamical and acoustical models will be discussed, and preliminary results from a coupled system will be presented.
High-resolution ocean acoustic tomography
Authors: Worcester Peter F. , Cornuelle Bruce , Howe Bruce , Munk Walter
Published: October 2005

The ocean acoustic tomography experiments conducted to date can be divided into mapping experiments, in which many sources and/or receivers are used to generate sufficient crossing ray paths to resolve the mesoscale ocean structure, and overaging experiments, in which data are obtained in a vertical slice between a single pair of sources and/or receivers in order to construct horizontally averaged vertical profiles. While it is economically feasible to deploy enough instruments to construct an array capable of mapping an area a few hundred kilometers on a side, it becomes prohibitive for areas one megameter and larger in scale, even though many fewer tomography instruments are required than conventional point measurements. One possible approach to make high-resolution, three-dimensional maps of large areas of the ocean is to develop a hybrid system, in which a combination of moored instruments and precisely positioned sources and/or receivers suspended from ships are used to generate a dense set of crossing ray paths. Simulations indicate that four or five transceivers moored in a large array to make long-term measurements of the average ocean structure could be combined with periodic cruises with ship-suspended sources and/or receivers to give excellent maps of a square megameter of ocean. It will be essential in such experiments to use data assimilation techniques to combine the data obtained as a function of time by the ship into a numerical model of the ocean. Other data types (e.g., satellite altimeter data) can be assimilated at the same time to improve the resulting maps of the ocean.
Simulated reciprocal transmissions through a three-dimensional sound-speed and current structure
Authors: Mercer James A.
Published: October 2005

A computer code that integrates Hamilton's equations for a continuously defined medium was used to model a reciprocal transmission experiment along a 2000-km path. The description of the medium included the three-dimensional sound-speed and current structures of four mesoscale eddies and a western boundary jet. Eigenrays for the reciprocal transmissions are determined, displacements are compared, and the critical physical processes are identified. Travel time results are compared with those from less realistic two-dimensional and one-dimensional codes. [Work sponsored by ONT.]
Coherence loss modeling using the PE method
Authors: Tappert F. D.
Published: October 2005

Scattering by volume fluctuations of sound speed and by boundary roughness causes a loss of coherence of propagating cw signals in the ocean. A 3-D parabolic equation model for the coherent Green's function (p) is derived that describes angle-and depth-dependent scattering loss. By integrating over a narrow sector of azimuthal angles, it is shown that a 2-D vertical plane model gives an excellent approximation for forward scattering loss. The resulting 2-D cw model is efficiently solved numerically by generalizing the split-step Fourier algorithm to perform multiple local Fourier transforms by windowing in depth, transforming from depth to vertical angle, applying an angle-dependent filter, summing over layers, and finally inverse Fourier transforming. For rough surface scattering, this method gives a loss that is linear in the grazing angle for small angles, in agreement with Marsh-Mellon theory.
Error estimates for 3-D computational schemes of wave propagation
Authors: Knightly George H. , St. Mary Donald F.
Published: October 2005

For several computational schemes in underwater wave propagation, stability and consistency bounds are known that imply corresponding rates of convergence. It is shown, in some cases, how this information can be improved to yield explicit bounds for the truncation error between the exact solution of a continuous problem and the approximating solution generated by the computational scheme. In particular, those bounds are demonstrated for an implicit finite difference scheme in three dimensions.
Calculation of free-field deviation in an anechoic room
Authors: Wang Ji-qing , Biao Cai
Published: October 2005

After a review of various calculation methods currently available for free-field deviation (in Db) in an anechoic room, the authors present, in this paper, a more precise and rational calculation of the sound field by solving wave equations in consideration of the interference of first reflections only. In the case of pure tone, the phase factor plays an important role in the calculation; therefore, the free-field deviation (in Db) calculated also depends on the source position, measuring direction and distance from the source, and the frequency used in the test, besides other factors such as the dimensions of the room and sound absorption characteristics of all the boundary surfaces. For a broadband noise, the calculation becomes simpler since the phase factor can be ignored, and a general equation of such a calculation is given for any source position and different combinations of boundary absorption. The results of the calculation can be shown in a diagram with the aid of a microcomputer, and, hence, a three-dimensional free-field scope would be very useful as a guideline to design and to use an anechoic room.
Effect of fluidal—stractural coupling on sound waves in an enclosure in low-frequency range
Authors: Jie Pan , Bies D. A.
Published: October 2005

The acoustic properties of individual normal modes in a rectangular panel-cavity system were investigated experimentally. The system consisted a heavy five-sided concrete box closed on the sixth side with a test panel. Experimental evidence in this paper reveals the dependence of the modal decay time of sound waves in the enclosure upon the modal characteristics of the test panel, such as the modal coupling factors, resonance frequencies, and modal decay times of the test panel. The results of experimental measurements are used to verify some of the previous theoretical predictions. The agreement between the measured and calculated acoustical modal decay time shows that the modal coupling theory can be used to evaluate the sound wave behavior in those enclosures, in which the coupling between fluid borne and structure borne acoustical waves is modal and where the classical model based on the locally reactive boundary assumption does not apply. The phenomenon of the maximum sound energy absorption by the panel was also observed in the experiment.
Evaluation of siren alert notification systems
Authors: Lee Van M.
Published: October 2005

This paper describes the essential elements in evaluating the effectiveness of a large outdoor fixed siren warning system, the state-of-the-art of the component evaluation processes, and the potential improvements in the future. The essential components in evaluating the coverage—the percentage of population within the planning zone alerted—consist of: siren acoustic performance (dB versus frequency versus directivity), site terrain and meteorological effects on sound propagation, population statistics (activity patterns and age distribution by household size), building attenuation effects, and activity-sound level response relationships. The lack of standard test procedures, whether laboratory or field, and data reporting formats has led to many disputes concerning omnidirectional and directional sirens. Standard methods for site qualification, siren testing, and data reporting similar to ANSI S1.34 (1980) and S1.35 (1979) are discussed in relation to their effect on the design and evaluation of siren systems. Also discussed are the effects of air absorption, wind and temperature gradients, operating characteristics (omnidirectional or rotating), and terrain effects, including Rasmussen's application of barrier diffraction and impedance ground effects. By correlating the physical siren sound level coverage to the social response level of the alerted population, a systematic methodology is presented, which evaluates the overall effectiveness of a large outdoor siren alert system.
The influence of high-intensity sounds on the performance of air warning sirens
Authors: Pla Frederic G.
Published: October 2005

The use of sirens as air warning devices has been widespread for several hundred years. Due to the very high sound-pressure levels generated, finite-amplitude effects can result in large losses leading to poor sound generation efficiency. Although much work has been done on siren design, the importance of finite-amplitude effects has been neglected in the recent past, even though a review of earlier works shows that researchers such as Reyleigh, in 1903, King, in 1919, and later, Hart, in 1926, noticed the extra acoustic losses associated with high-intensity sound sources. Comparisons between linear and nonlinear models of wave propagation in a siren horn illustrate the importance of these types of losses, and show how they can be minimized. Acoustic saturation, which limits the sound-pressure level that can be perceived by a receiver at a given distance from a high-intensity sound source, and the implications of finite-amplitude effects on the rating of high-intensity sound sources in a laboratory environment are also discussed.
Monostatic angular distributions and their relation to resonances in acoustical scattering from fluid loaded solid elastic objects
Authors: Gaunaurd G. , Werby M. F.
Published: October 2005

It is possible to detect resonances from backscattered echoes (form functions) by varying the nondimensional frequency kL/2 in small increments. This gives rise to a familiar pattern of nulls followed by a sharp rise for the spherical case and similar results for the spheroidal case. This approach, however, limits one to one aspect angle for three-dimensional nonspherical problems and often entails tedious and time consuming calculations to find the often narrow resonance regions. Another approach suitable for three-dimensional problems is to examine closely spaced monostatic angular distributions enabling one to look for possible resonances over a continuous range of aspect angles. This method is used to examine scattering from solid steel spheroidal shells. The results indicate that Rayleigh type resonances are aspect angle dependent in contrast to our recent results on spheroidal shells. The results are consistent with the standing wave interpretation of resonances proposed by use earlier for shells and employed in work on material properties of solid elastic spheroids.
Another look at the Rayleigh angle backscattering
Authors: Nagy Peter B. , Adler Laszlo
Published: October 2005

The increased backscattering from a liquid-solid interface at the Rayleigh angle is a well-known indication of leaky surface wave generation (similar phenomena were observed from thin plates immersed in liquid due to leaky Lamb wave generation, too). This simple technique has proved to be very useful in material evaluation via measuring both surface and Lamb wave velocities. On the other hand, a comprehensive experimental study [M. de Billy, L. Adler, and G. Quentin, J. Acoust. Soc. Am. 72, 1018 (1982)] showed much stronger backscattered signals than predicted later by theoretical works based on back reflection. New experimental results are presented demonstrating that this back reflection is negligible in the backward signal, which is simply backscattering from inevitable material inhomogeneities greatly enhanced by surface or Lamb wave generation. Therefore, beside measuring macroscopic elastic properties in the previously suggested way, the backscattered signal can be used to characterize surface and near-surface material inhomogeneities as well. [This work was supported by the U.S. Department of Energy, Basic Energy Sciences, Grant DE-FG02-84ER45057A.004.]
Reduced auditory discrimination and compression in hearing aids
Authors: Dreschler Wouter A.
Published: October 2005

Two specific applications of single-channel compression are studied separately: whole-range compression as a compensation for recruitment, and compression limiting as an elegant method to limit excessive output levels. In cases of compression limiting, compression proved to be superior to peak clipping, especially for the perception of consonants: An average improvement in identification scores of 18% was found and less roll-over for the perception of final consonants at high presentation levels. In cases of whole-range compression, a small, but significant, increase in identification scores was found for settings with the compression knee-point at the lowest level. A detailed analysis of the patterns of confusions revealed clear qualitative differences in the perception of phonemes with and without compression. These differences can be brought into relation with an improved perception of temporal cues, like the preburst silent interval of plosives, and with spectral changes due to the activation of the compression circuit.
Frequency discrimination and consonant recognition in the hearing impaired
Authors: Ochs Marleen T.
Published: October 2005

The frequency difference limen (DL) for short-duration second-formant transitions (F2) presented alone and in speechlike environments was examined in normal hearing and hearing-impaired subjects. Four stimulus conditions were included: F2 transition alone; F2 transition in the presence of all formants (full formant); full-formant stimuli preceded by a burst and followed by a vowel; and full-formant stimuli preceded and followed by a vowel. For the simplest stimulus condition (F2 alone), all hearing-impaired subjects had DLs within the 95% confidence interval around the mean of the normals. For all other conditions, however, the hearing impaired demonstrated extreme intersubject variation. Some hearing-impaired subjects could not detect a formant transition of even 800 Hz, whereas others continued to perform like normal hearers. Consonant identification performance was also examined in CV and VCV environments. Consonants whose identification is believed to involve formant transitions were selected for study. In spite of modifications to the frequency discrimination task and use of relevant syllables, relatively weak correlations continue to be observed between F2 DL and consonant recognition ability, as well as between F2 DL and degree of hearing loss. [Work supported by NIH.]
Diffraction of pulses at a sharp edge
Authors: Schouten Gerrit
Published: October 2005

The closed expression for the diffracted field of a sound pulse is derived in a straightforward manner using Sommerfeld's method for the generation of many-valued solutions from known one-valued solutions of the wave equation. The spherical primary wave front carries a delta-function disturbance, whereas the spindle-shaped diffraction front carries a square-root singularity bounding the filled diffraction region where two dimensional aspects are dominating. Integration w.r.t. time of firing of the pulse yields the transient behavior of a hydrodynamic switch-on source with a unit-step wave front. The well-known result of Cagniard (1935) is thus reproduced in a direct way. Plots of the velocity field inside the diffraction region show the peculiarities of the transition from the wave solution to the steady Laplace solution. Applying the method of descent to the three-dimensional solutions yields the known two-dimensional equivalents, again in a more direct way than known from the literature [R. D. Turner (1956); H. A. Lauwerier (1962); E. M. de Jager (1964)].
Impact testing of light objects
Authors: Abdelhamid Mohamed K. , Pate Anna L.
Published: October 2005

This paper investigates the design of a measurement system that measures the impact signature of a light object. The design is based on a freely supported piezoelectric force transducer. Effects of top mass, base mass, and stiffness of the transducer were investigated. Two models were used in the analysis of the impact, a linear model with a 3 degree of freedom system and a Hertzian model. Using the Hertzian model, a criterion for choosing the optimum size of the base mass was developed. This criterion is based on the least-square error in the force spectrum. Similar results were obtained by the linear model and numerical simulations. Experimental confirmation of the results was conducted using small Teflon and nylon balls and a piezoelectric force transducer. The results agreed well with the model predictions.
Viscoelastic effects on solid amplitude and phase transformers
Authors: Campos L. M. B. C.
Published: October 2005

The longitudinal oscillations of nonuniform bars are used as displacement amplifiers in power tools and other devices. These often operate near resonance [E. Eisner, J. Acoust. Soc. Am. 35, 1367–1372 (1963)], so that the elastic model should be replaced by a viscoelastic one, which includes damping. In the present paper, the longitudinal vibrations of a tapered viscoelastic bar are discussed generally from the wave equations for the displacement and strain. Exact solutions are obtained for the exponential, catenoidal, sinusoidal, and inverse shapes, and also for the Gaussian and power-law shapes. These solutions generalize earlier results for elastic bars; e.g., the elastic Gaussian bar [D. A. Bies, J. Acoust. Soc. Am. 34, 1567–1569 (1962)] is generalized to a viscoelastic one. Diagrams of wavenumber and damping ratio versus frequency and viscous relaxation time are presented for several shapes of tapered bar; they describe the propagation and dissipation of oscillations and their effects on amplitude and phase.
Vibration analysis of mass-loaded beams
Authors: Manikanahally Dhanesh N. , Crocker Malcolm J.
Published: October 2005

A general procedure for determining the dynamic response of a mass-loaded free free beam subjected to a harmonic and transient force is given. Though free free beams are considered for analysis, the same procedure could be extended for other end conditions also. The beam is assumed to have structural damping for determining the steady-state response due to harmonic force excitation. The mode shapes for free vibration, dynamic response, and dynamic strain due to forced excitation are presented in a graphical form. The analysis is used to study a space structure, modeled as a mass-loaded free free beam, by making an exhaustive optimization search for minimum dynamic response due to harmonic and transient excitation forces. The computer program developed for the analysis is used to check some simple beam problems [G. B. Warburton, The Dynamical Behaviour of Structures (Pergamon, New York, 1976)]. [Work supported by SDIO/DNA, Contract No. DNA-001-85-C-0183.]
Procedure for the measurement of wavenumber/frequency admittance of structures
Authors: Grosh Karl , Hughes W. Jack , Burroughs Courthey B.
Published: October 2005

A procedure for measuring the wavenumber/frequency admittance of structures using an array of drives in a standing wave pattern was developed in a paper presented in the ASA meetings in Indianapolis, IN on 14 May 1987 [K. Grosh, J. H. Hughes, and C. B. Burroughs, J. Acoust. Soc. Am. Suppl. 1 81, S73 (1987)]. In this paper, data are presented to verify the validity of the measurement procedure. The wavenumber/frequency spectra are presented for arrays of point drives and the vibration responses of lightly damped beams and long beams with heavily damped ends. The effect of steering the array to different wavenumbers and varying the number of drives is examined.
Experimental measurement of power transmission in an L-shaped beam
Authors: McCollum M. , Gilbert T. , Rassineux J. , Cuschieri J.
Published: October 2005

The vibrational power transmitted across the joint between two beams forming an L-shaped structure, when one of the beams is excited by a point force at the free end, is measured experimentally. One method of calculating the power transmission is based on the assumption that the power transmitted to the receiver beam is equal to the power dissipated by that beam. The damping loss factor used in the calculation is obtained from measurements on the receiver beam. The results of the experimental analysis are compared to those obtained using a power flow technique, the results of which have been verified by the closed form solution for the global structure. The measurements show very good agreement with the analytical results in terms of the location of the resonant frequencies, except for low frequencies. The measured levels are, however, significantly lower than the predicted levels, especially at the resonant peaks, indicating that the measured damping levels are underestimated. The measured levels are generally very close to the predicted levels away from the natural frequencies. [Work supported by NASA and ONR.]
Project-type acoustics experiments in the USNA acoustics lab
Authors: Elder S. A.
Published: October 2005

For some years, the Naval Academy Physics Department has offered a senior-level course in physical acoustics, accompanied by a weekly laboratory. Since the classes are small, the laboratory offers a chance for tutoring the students in more advanced and creative projects than the usual “classic” lab experiments for courses at this level. Usually there is time to perform three or four projects over the semester, each one defined by a general area of interest (such as environmental acoustics, architectural acoustics, computational and data acquisition methods, etc.). Sometimes the project spills over into the next semester and becomes a credit earning research project beyond the course itself. Examples of successful student projects are: construction and testing of a loudspeaker (magneplanar, flame driven, etc.), acoustical investigation of a local concert hall or church, construction and testing of an acoustic guitar, development of computer-controlled data acquisition and analysis system. Specific examples, as well the general philosophy of the course, will be given.