bluuebunny/crossref_metadata_2025 huggingface dataset:
Acoustic (non)consequences of gestural variability in the production of accentual prominence
Authors: Harrington Jonathan , Fletcher Janet
Published: October 2005

In English, pitch accent placement makes a word prominent. Previous studies have documented that talkers can highlight accentual prominence by such strategies as lowering the jaw more in the stressed vowel of the word. This study examines the acoustic consequences of such supralaryngeal correlates of accenting. Speakers of Australian English produced multiple tokens of dialogues which elicited nuclear-accented and de-accented productions of the names Beeber, Deeder, and Geeger, which have a stressed high front vowel in different consonantal contexts. Recordings were made of tongue–dorsum and jaw movement using the Movetrack electromagnetometer system. The results showed multiple articulatory strategies that varied across talkers and consonantal context. For example, of two talkers that lowered the jaw more in accented syllables, one fronted the tongue to compensate for the jaw lowering in all contexts, whereas another did so only in the velar context. Nevertheless, accenting the word resulted in a consistent raising of the frequencies or amplitudes of spectral peaks in the region of the second and third formants. These results suggest a model of accent as ‘‘localized hyperarticulation’’ whereby the talker has multiple articulatory strategies for achieving a perceptual effect of ‘‘sharpening’’ the /i/ timbre to increase the word’s clarity.
Acoustic analysis and perceptual judgment of a child’s speech sound production during speech therapy
Authors: Siren Kathleen A. , Morris Sherrill A. , Wilcox Kim A.
Published: October 2005

Recent studies have attempted to evaluate the use of acoustic analysis in the assessment and treatment of phonological disorders [K. M. McGregor and R. G. Schwartz, J. Speech Hear. Res. 35, 596–603 (1992); Tyler et al., J. Speech Hear. Res. 36, 746–759 (1993)]. This investigation follows a child during therapy for multiple speech sound errors. Two speech sounds, a fricative /f/ and a stop consonant /t/, were chosen to represent two distinct consonant types in error which were being targeted in therapy for the period of this investigation (3 1/2 months, with a 1-year, 3-month followup). The child’s productions of these sounds were probed during a picture-naming activity which included the target sounds in both prevocalic and postvocalic positions. On-line clinical judgments (correct, incorrect, absent) were made during audiotaping of the child’s productions. Audiotape recordings were then analyzed spectrographically using a Kay Elemetrics CSL model 4300. Both temporal and spectral dimensions were documented to track changing sound productions across time and were compared to the perceptual judgments. Measurable acoustic features predicted perceptual sound changes to a limited degree. Results are compared to prior studies. Implications for the inclusion of acoustic analysis in the assessment and intervention of phonological disorders are discussed.
Objective and subjective measures of self-reported hearing deterioration in older hearing-impaired adults
Authors: Cheesman Margaret F. , Greenwood Kristina G.
Published: October 2005

A small proportion of hearing-impaired older adults reports hearing deterioration during their annual otologic examination without exhibiting any concomitant changes in audiometric or otologic results. This study examined the ability of a self-report questionnaire and a battery of speech perception tests to distinguish these subjects from a set of matched subjects who had no reported hearing deterioration. The Communication Profile for the Hearing Impaired (CPHI) and three performance-based speech perception measures [objective and subjective speech reception threshold tests (SRT) and a suprathreshold nonsense word task] were included in the test battery. Responses to some specific items of the CPHI and the difference between the subjective and objective SRTs in noise differed between listeners with reported hearing deterioration and those without. Those who reported deterioration underestimated their hearing ability to hear in noise more than the control group. They were also more aware of their communication difficulties in some communication settings. [Work supported by the Natural Sciences and Engineering Research Council of Canada, Health Canada, and Unitron Industries Ltd.]
Comparison of speech recognition threshold in noise using laboratory and clinical procedures
Authors: Thibodeau Linda M.
Published: October 2005

Determining threshold for speech in quiet and as a function of increasing noise levels has been used to quantify the effects of hearing loss in terms of attenuation and distortion [R. Plomp, J. Acoust. Soc. Am. 63, 533–549 (1978)]. The speech recognition threshold (SRT) in noise has also been found to be a sensitive measure of electroacoustic changes in amplification [D. Van Tasell and J. Yanz, J. Speech Hear. Res.30 , 377–386 (1987)]. The purpose of this study was to determine if there was close correspondence between the laboratory measure of SRT in noise and a shortened clinical version using the Van Tasell and Yanz monosyllabic, high-frequency words. The clinical procedure was based on the SRT method recommended by the American Speech, Language, and Hearing Association and involved two conditions, quiet and speech-shaped noise at a moderate intensity level. The derived values of attenuation and distortion were significantly correlated across the two procedures. [Research supported by NIDCD of NIH.]
Coarticulation in aphasia: Effects of utterance rate and complexity
Authors: Baum Shari R.
Published: October 2005

The magnitude and extent of anticipatory coarticulation were examined in groups of fluent and nonfluent aphasic patients, and normal control subjects. Previous investigations have shown inconsistent impairments in coarticulation in aphasic patients; studies have also demonstrated breakdowns in production as rate and complexity increase. One- and two-syllable target utterances were elicited at slow and fast rates of speech with or without a consonant intervening between the target consonant and vowel, and with or without a preceding schwa, to manipulate utterance complexity. Acoustic analyses (F2 and centroid frequencies) revealed that both groups of aphasic patients exhibited relatively normal patterns of anticipatory coarticulation. However, small but significant differences among the groups emerged in certain conditions. Surprisingly, increased utterance complexity was not found to reduce coarticulatory effects to a greater degree in the nonfluent relative to the fluent aphasic group. The ability of listeners to identify the upcoming excised vowel when presented with the target consonant alone was also examined to determine the perceptual salience of the coarticulatory cues. Results are discussed in relation to hypotheses concerning the underlying speech production deficits in fluent and nonfluent aphasic patients and the neural substrate for aspects of speech motor control. [Supported by MRC.]
Properties of naturally produced clear speech at normal speaking rates
Authors: Krause Jean C. , Braida Louis D.
Published: October 2005

In adverse listening conditions, talkers can increase their intelligibility by speaking clearly. While producing clear speech, however, talkers often significantly reduce their speaking rate. A recent study [J. C. Krause and L. D. Braida, J. Acoust. Soc. Am. 98, 2982(A) (1995)] showed that talkers can be trained to produce a form of clear speech at normal conversational rates. This finding suggests that acoustical factors other than reduced speaking rate are responsible for the high intelligibility of clear speech. To gain insight into these factors, the acoustical properties of conversational and clear speech were analyzed to determine phonological and phonetic differences between the two speaking modes. These differences were interpreted in terms of error patterns made by normal-hearing listeners identifying key words in the presence of wideband noise. Additional intelligibility tests investigated other degradations in order to explore the robustness of the high intelligibility of clear speech produced at conversational rates. Native and non-native listeners were employed for degradations consisting of additive noise, high-pass and low-pass filtering, and reverberation. [Work supported by NIH.]
Self-organizing map for the classification of normal and disordered voices
Authors: Callan Daniel , Kent Ray D. , Kent Jane F.
Published: October 2005

The classification of voice disorders has been hampered by the questionable reliability of perceptual scaling methods and by the generally poor correlations between perceptual and acoustic data. This report uses a self-organizing map to classify normal and disordered voices. The self-organizing feature map (SOFM) is essentially a two-dimensional representational mapping over an array of processing units of the multidimensional nonlinear regularities inherent in the input data space. The input data were five acoustic measures obtained from a MDVP [Multi-Dimensional Voice Protocol, Kay Elemetrics Corp.] analysis of sustained phonations included in the Voice Disorders Database version 1.03, compiled by the Voice and Speech Laboratory, Massachusetts Eye and Ear Infirmary. The measures included: jitter percent, fundamental frequency variation, shimmer percent, noise-to-harmonics ratio, and amplitude of the dominant cepstral rhamonic. The SOFM was trained with three voice samples: normal, hyperfunction, and anterior–posterior squeezing. The SOFM also was used to classify voices obtained from individuals with neurogenic speech disorders (dysarthrias). [Work supported by NIH.]
Evaluation method utilizing subjective MTF measurements for speech perception ability of hearing-impaired listeners
Authors: Miyata Hiroyuki
Published: October 2005

A new method for characterizing the speech perception performance of hearing-impaired listeners in both quiet and noisy environments is proposed. The concept of this method is that one’s spech perception ability depends on the quality of one’s perceived speech sound image. In this method, one’s perceived image quality is related to subjectively a measured modulation transfer function (MTF). The procedure for estimating the sound image quality from measured MTF values obeys that of the RASTI method [Houtgast et al., Acustica46 , 60 (1980)], which is a physical method for predicting speech intelligibility in rooms. The subjective MTFs are obtained from the signal detection experiments using an adaptive 2IFC procedure with a modulated masker noise [Miyata et al., Acustica 73, 200 (1991)]. The proposed method was used to evaluate the monosyllable recognition abilities of six hearing-impaired subjects. The experimental results provide good support to the method for characterizing the speech perception ability of hearing-impaired listeners.
A dynamic tracking model of spectral shape: The sequential grouping based on timbre
Authors: Masuda I. , Aikawa K.
Published: October 2005

This study proposes a new model for dynamically tracking spectral shape change. In previous studies on computational implementation of auditory scene analysis, the sequential grouping has not been fully realized. Since the new model can track the spectral shape, it is able to make the sequential grouping based on timbre feasible. First, a spectral envelope is converted into a series of frequencies by using the IFIS (inverse function of integrated spectrum [Ohmuro et al., Tech. Rep. Speech Acoust. Soc. Jpn. SP89-72 (1992) (in Japanese)]. IFIS has good interpolation and extrapolation characteristics. Spectral shape is represented as a set of frequencies on the IFIS axis. Next, the frequencies are tracked with the FM-tracking model [Aikawa etal ., J. Acoust. Soc. Am. 98, 2926(A) (1995)]. Aikawa et al.’s model represents the perceptual characteristic of sweep tones and is described by a second-order AR model. Finally, the frequencies are again converted into a spectrum. Furthermore, this new model has an additional function: spectral shape prediction. This is possible because the second-order AR model includes a prediction function, which can, for example, represent the recovered portion of the noise placed in a speech sound. The advantages of this model will be discussed in relation to speech signal separation techniques.
Correlation between acoustic pauses and perceptual pauses in speech
Authors: Minematsu Nobuaki , Nakagawa Seiichi
Published: October 2005

An analysis of spoken language in lecture style was carried out, where two kinds of pauses, acoustic pauses and perceptual pauses, were defined and detected from the same speech material. While the former were defined as speech segments with lower power than a given threshold and detected semi-automatically, the latter were defined as speech segments during which humans perceive a ‘‘break’’ in the utterances and obtained through perceptual experiments. After detecting both the pauses, the probability of an acoustic pause being also a perceptual pause was investigated separately for each of the speakers. As a result, it was found that the probability for the acoustic pauses of a given length was largely decreased when the speaking rate was lower. This implies that, especially in the case of pause-based speech processing, the pauses should be processed considering the speaking rate around them. Furthermore, the investigation of interjections and filled pauses around the perceptual pauses showed that ‘‘e[e],’’ ‘‘e[e]to,’’ and ‘‘de’’ caused the perceptual pauses with quite high probability.
Evidence for the predominance of vowel onsets to offsets in speaking-rate perception
Authors: Kato Hiroaki , Tsuzaki Minoru , Sagisaka Yoshinori
Published: October 2005

To investigate the functional differences between vowel (V) onsets and vowel offsets in the perception of temporal structures in speech stimuli, listeners were required to estimate the perceived difference in speaking rates from various four-mora Japanese words. In the V-onset condition, the inter-onset intervals of vowels were uniformly changed (either lengthened or shortened) while preserving their interoffset intervals, and vice versa, in the V-offset condition. These manipulations did not change the duration of the entire word. Each of the modified words was paired with its unmodified counterpart and was given to the listeners. Interestingly, the results showed that changing the V-onset intervals correlated with a change in the perceived speaking rate, despite the fact that the duration of the entire word was unchanged. However, the modifications of the V-offset intervals had no significant effect on the perceived speaking rate. Although there was no significant functional difference between V onsets and V offsets in the detection of local changes in the temporal structure [Kato et al., Proc. ICSLP-94, 1979–1982 (1994)], the current results successfully demonstrated that they do differ and suggest that V onsets are the dominant cue in determining the speaking rate, i.e., the tempo of events. The results will be discussed in relation to recent models predicting perceived beat locations.
The role of temporal and spectral cues in speech recognition
Authors: Loizou Philipos C. , Dorman Michael F.
Published: October 2005

It has been proposed [Shannon et al., Science 270, 303–304 (1995)] that the high levels of speech understanding achieved with a small number (three to four) of channels reflects the information content of ‘‘primarily temporal cues’’ for speech recognition. Here, this outcome is viewed differently. It suggests that only a small number of channels is necessary to allow reasonable resolution in the frequency domain. This interpretation is derived from the results of a set of experiments in which two conditions were created. In one, the speech materials were processed normally through a four-channel processor. In the other, the channel assignments were reversed; i.e., the envelope in channel 1 was directed to channel 4, etc. These stimuli had the same temporal information as the original stimuli but had abnormal information in the frequency domain. The identification of these stimuli was significantly worse than the identification of the normally processed stimuli. These results suggest that for a four-channel processor, speech recognition is determined principally by information in the frequency domain, and that temporal information plays only a supplemental role. Supporting evidence for this position comes from additional experiments in which speech was processed through a processor that outputs sine waves rather than bands of noise.
Analysis of interlabeler (dis)agreement in phonetic transcriptions of multiple languages
Authors: Lander T. , Oshika B. , Carlson J. , Durham T. , Bailey T.
Published: October 2005

This study extends previous evaluations of phonetic labeling agreement between multiple transcribers (ICSLP94, ICPhS95). In the current experiment five labelers transcribe a total of 2 min of conversational speech in English, French, German, Spanish, Vietnamese, and Japanese, with four female and four male talkers for each language. The paper presents three sets of results: labeler agreement, the effect on labeler agreement of using an orthographic reference as an aid, and analysis of the types of ‘‘errors’’ contributing to labeler (dis)agreement. Interlabeler agreement is measured at the phonetic, phonemic, and broad category levels of transcription. Broad category includes classes of sounds such as nasals, vowels, fricatives, and plosives. For half of the data in each language, transcribers could refer to an orthographic transcription; the paper describes the effect of this additional information on labeler agreement. The paper also presents detailed analyses of the kinds of insertions, deletions, and substitutions leading to labeler disagreement. Focus is on whether transcribers converge on a pattern of favored transcription conventions, such that what seems to be disagreement in identification of sound segments may be a matter of transcriber convention. [The authors would like to thank the National Science Foundation for their support of this research.]
Loudspeaker characteristics for active sound control
Authors: Laak Trevor , Allie Mark
Published: October 2005

One of the main issues limiting the performance of practical active sound control systems is acoustic output power limitations of real loud-speakers. Various loudspeaker characteristics and parameters are considered with respect to active sound control requirements. Issues such as loudspeaker resonant frequency, Q, nonlinearities, excursion limitations, thermal power, and group delay are discussed. Some discussion of required or resulting controller solutions is included to illustrate the effects of each parameter or characteristic on performance. Finally, a practical set of desirable loudspeaker characteristics is suggested to provide improved performance of active sound control systems.
Performance prediction for short plant length systems
Authors: Popovich Steven R.
Published: October 2005

Plant length dependent performance is commonly observed when performing feedforward active noise control for broadband disturbances. This is primarily due to high levels of low-frequency noise and limited coherence between input and error sensors. These types of systems present trade-offs in choosing between performance, output power requirements, and plant length. An additional design opportunity exists in terms of balancing the levels of cancellation achieved over frequency. The selection of an error signal emphasis filter determining the frequency dependence of the cost function being minimized by the controller adaptation can be of critical importance. While several explanations for this type of behavior have been offered, little work has been presented for quantifying performance limitations. A method based on calculation of Lagrange multipliers is presented for determining the optimal power limited solution of an FIR filter-based controller. This provides a useful tool for predicting the effects of emphasis filter design, actuator limitations and plant length on performance. This prediction method and the associated trade-offs are illustrated using data from an HVAC fan.
The behavior of an adaptive algorithm with moving primary source
Authors: Omoto Akira , Fujiwara Kyoji
Published: October 2005

Because of the rapid progress of DSP hardware performance, the application of active noise control to a moving noise source has become a much more realistic problem. This study describes the preliminary investigations concerning the behavior of a multichannel adaptive algorithm with a moving primary source. The steepest descent algorithms, such as multiple error filtered-x LMS, are often used in practical feedforward active control applications. A previous study showed that the stability and the performance of this algorithm strongly depend on the estimation errors of the transfer responses of the acoustic plant being controlled. The effect of the primary signal variations including the Doppler effect, however, had hardly been investigated. In this study, the convergence properties of the algorithms with the moving sound source were examined by numerical simulations. To simplify the problems and clarify the substantial limitations of the system, a pure tone and a simple free field was assumed as the primary signal and the sound field, respectively. Results showed that the system could converge, and the possible attenuation was varied corresponding to the positions of the primary source.
The use of the finite-amplitude method to measure the nonlinearity parameter B/A and the harmonic growth in some fluids
Authors: Moustafa Aly H. A.
Published: October 2005

The acoustic nonlinearity parameter B/A for water and for sugar solution, and the growth of the harmonic contents in a finite-amplitude signal propagating in these fluids are investigated experimentally. The apparatus used consists of a perspex cylinder containing the fluid and a 2.25-MHz transducer (source), separated by a variable distance from a 4.5-MHz transducer (receiver). As the separation distance is increased distortion builds up as expected and the second harmonic, in particular, could be clearly observed due to the high sensitivity of the receiver at this frequency. It was possible to catch and store the received pulse, after averaging it for a large number of transmissions, using a digital storage oscilloscope. The capture signal was then spectrally analyzed using MATLAB software. The results showed that the fundamental component was almost constant, while the second harmonic had a linearly increasing trend with the separation distance. The higher harmonics are negligibly small. The receiver was separately calibrated, and its measured sensitivity was used to convert the fundamental and the second harmonic voltages into pressures which were then used to calculate the value of B/A. The results agree relatively well with published values for water. For sugar solution, B/A was found to be less than that for water.
Application of complex MEM processing for measuring reflective architecture in acoustical archaeology
Authors: Aoki Yoshinao , Kudoh Yohji
Published: October 2005

A measurement of reflective characteristics of architecture is one of the fundamental studies in acoustical archaeology. In this paper an application of maximum entropy method (MEM) processing is discussed for measuring the characteristics of reflective architecture in acoustic archaeology by computer simulation. Chirp signals are supposed to be transmitted from a speaker and reflected acoustical signals of amplitude and phase are recorded using several receivers. Since the received signals are complex, MEM algorithms for processing complex signals are developed and the relative distances between reflective targets and receivers are calculated, resulting in the identification of the location of the targets in 2-D space. In a different view, the proposed mesurement technique is a reconstruction of a target from a one-dimensional complex acoustical hologram. Computer simulation is done in which chirp signals are supposed to be reflected from a target and complex acoustical signals are generated by superposing the signals received by several receivers. Pulse compression of signals of different receivers is conducted using the proposed MEM algorithm, resulting in detection of the distances between the target and receivers. Application of the proposed method to the measurement of the reflective characteristics of architecture in acoustic archaeology is discussed.
An ultrasound echo-signals tracking method for blood flow velocity estimation
Authors: Hongmin Wu , Ohtsuki Shigeo , Tanaka Motonao
Published: October 2005

Doppler techniques used in blood flow velocity estimation rely on the assumption that the phase of the envelope-detected echo signals changes linearly with depth. However, the random spatial distribution of scatterers makes it inadequate. Correlation searching techniques utilize time shifts to remedy this drawback, but the probability of the correct maximum detection is also low under certain measurement conditions. In this study, a more sensitive and simpler means of velocity estimation has been proposed. This method tracks a segment of the echo footprints by means of the least-sum-squared difference from the consecutive RF echo signals. Since a squared difference measure is used to extract the moving information, the detection results are sensitive to the locally large differences, no matter how large the amplitudes are here, and the probability of finding a correct match by this method is highly increased. A detailed computer simulation evaluated the interdependencies of the window size, the pulse duration, the velocity searching scope, and the aliasing problem on the estimation of the blood flow velocity. The results are proved to be more accurate and effective than those of the conventional techniques.
Noise removal from measurements while drilling an oil well
Authors: Hosono Kazuho , Moriyama Haruki
Published: October 2005

Systems to acquire borehole data during the drilling of oil and gas wells make use of measurement while drilling (MWD). One feature of this system is that it is able to do real-time measuring from a borehole; therefore there has been a lot of MWD use on drilling sites in recent years. There are a few types of MWD. Mud pulse-type MWD, which uses a drilling circuit fluid, is superior to the rest because of its reliability, accuracy of data, and less disturbance of the drilling schedule. The drilling circuit fluid is raised to a high pressure by a mud pump; borehole data which are recorded by the surface measuring system are contaminated by the pumping noise. Therefore, it is necessary to remove the pumping noise to get objective data. This report describes the pumping noise removal system and the method used for the telemetry system from 2000 m depth.
Optimization of the spectral signal processing for narrow-band stochastic acoustic signals
Authors: Gelman Leonid M. , Burau Yuri V. , Krytsyn Vladimir I.
Published: October 2005

The spectral digital signal processing methods for narrow-band acoustic signals are considered. One of the main parameters of spectral signal processing, spectral analyze resolution, is optimized earlier but only by using the energy criterion of optimality—the signal/noise ratio. In the present work, an optimization problem for the spectral analyze resolution was formulated and solved by using probability criterion of optimality, false-alarm probability for signal detection. It was considered a mixture of narrow-band signals and noisy interference and two important practical cases: (1) known center frequencies of signals; (2) unknown center frequencies of signals. A new nontraditional common results is obtained; it is shown that under final signal duration, optimum resolution mismatches substantially with a width of spectral signal density which allows considerable simplification hardware realization of spectral digital signal processing methods.
Complex rays and waveforms near caustics
Authors: Tindle Chris T. , Bold Gary E. J.
Published: October 2005

Ray theory can be carried into a shadow zone using rays of complex launch angle but the search for eigenrays is usually impractical. A new approach first finds the ‘‘measurement front,’’ i.e., the continuous curve of ray depth, as a function of arrival time at a fixed range. The measurement front folds back on itself when the rays touch a caustic. The measurement front can be continued beyond the caustic into the shadow zone using rays of complex launch angle. Points at a fixed depth on the measurement front represent ray arrivals at a hydrophone. Waveforms can be calculated directly from the ray arrivals. Airy functions are used for ray amplitudes near caustics.
Ray theory in mode models: Demonstration of high-frequency collective-mode interference effects
Authors: Manning Jason F. , Westwood Evan K. , Smith Eric
Published: October 2005

A number of acoustic reflection and refraction effects, normally associated with ray theory, are used here as tests of a normal mode model in the high-frequency limit, where the modes become dense in the complex wave-number plane. The ray effects, which are associated with saddle points or branch line integrals in the wave-number integration method, do not appear as characteristics of any single mode in the modal solution. Rather, they emerge as collective effects from the coherent addition of large numbers of modes. The effects demonstrated include beam splitting at the Brewster angle for a fluid–fluid interface and at the Rayleigh angle for a fluid–solid interface, beam displacement near the critical angle for total reflection, and beam spreading due to a fluid–solid lateral wave. Gaussian beams are modeled efficiently using the ORCA normal mode model, by assigning complex values to the source coordinates. This technique extends, in a full mode solution, methods that have previously been implemented for Gaussian beams propagating in free space [Deschamps, Electr. Lett. 7, 684 (1971)] and reflecting from single interfaces [Ra et al., J. Appl. Math.24 , 396 (1973)]. [Work supported by the U.S. Navy Office of Naval Research.]
Modal coupling in forward propagation through a layered dipping sediment with consolidation
Authors: Oba Roger M.
Published: October 2005

A parabolic equation with shear is used to compute the acoustic field in a shallow-water environment with a layered dipping sediment. Such sediments are typically formed by erosion of consolidated tectonically upturned layers. These are modeled in tilted sediment structure of a uniform vertical thickness over a range-independent hard sub-bottom. The water sound speed is range independent and depth dependent in a fixed depth environment. The PE propagated acoustic field is decomposed into local modes. In some instances the regularity of the layering permits a reasonable approximation by periodic layer spacing. Periodic layering gives significant structure to its modal coupling determined largely by the horizontal aspect of the layer spacing. Treating the interlayer spacing as a gamma-distributed random variable introduces phenomena not found in the periodic case. Even with very slight sediment tilt, transmission loss can be significantly different from the case with horizontal layering. The relative contribution of shear modulus compared to other parameters, especially density as it contributes to impedence mismatch, is explored.
Computations of rough surface reflection coefficients in a waveguide
Authors: Berman David H.
Published: October 2005

The effect of rough interface scattering in a waveguide is complicated by the possibility of scattered sound returning to the rough interface and then rescattering. A formalism for determining the effect of such rescattering on the mean field was developed in a previous paper [D. H. Berman, J. Acoust. Soc. Am. 96, 417–426 (1994)], and similar formalisms are described in A. G. Voronovich [Wave Scattering from Rough Surfaces (Springer-Verlag, Berlin, 1994)]. It has been argued that the effect of re- scattering will be small if the skip distance of rays associated with normal modes is larger than the correlation length of surface roughness. Using numerical evaluations of the Bourret approximation for the mean field, it is shown here that for correlation lengths comparable to the skip distance, rescattering can contribute significantly to the effective reflection coefficient for the mean field. However, it will also be shown that a good approximation to the effective reflection coefficient can be obtained by replacing the true surface roughness by a multiple of the surface roughness. The multiple is determined by the ratio of skip distance to correlation length. [Work supported by ONR.]
Construction of exact left symbols of the vertical slowness operator
Authors: de Hoop Maarten V. , van Stralen Mattheus J. N. , Blok Hans
Published: October 2005

The Bremmer series solution of the acoustical wave equation in generally inhomogeneous media, requires the introduction of pseudodifferential operators. Such operators describe the (de)composition, propagation and interaction of the up- and downgoing waves. The exact solutions of the corresponding symbols can be derived for several medium profiles. Of particular interest is the symbol associated with the vertical slowness or square root operator, involved in the propagation. This symbol constitutes the generalized slowness surface. These symbols serve as benchmarks for numerical one-way wave schemes involving waveguiding structures. Starting from the kernel representation of the transverse Helmholtz operator’s resolvent, a normal mode expansion is set up. The Schwartz kernels of negative fractional powers of the transverse Helmholtz operator and their left symbols can then be expressed in these normal modes as well. Using the composition relation for the original operator and its inverse square, a recursive relation for the left symbols is obtained. Results of the quadratic waveguiding (focusing) profile and a localized profile will be presented. The connection with the eigenfunctions and eigenvalues (normal mode expansion) will be highlighted.
Numerical implementation of the three-dimensional one-way wave propagator using the third-order Thiele approximation on a hexagonal grid
Authors: van Stralen Mattheus J. N. , de Hoop Maarten V. , Blok Hans
Published: October 2005

The Bremmer series solution of the 3-D acoustical wave equation in generally inhomogeneous media, requires the introduction of pseudodifferential operators. These operators describe the (de)composition, propagation and interaction of the up- and downgoing waves. In this paper, a sparse matrix representation of the propagator is derived. In a similar way the matrix representations of the (de)composition and interaction of the up- and downgoing waves can be derived. The focus is on designing sparse matrices, keeping the accuracy high at the cost of ignoring any critical scattering-angle phenomena. Such matrix representations follow from well-known rational approximations of the vertical slowness, the Laplace operator and the vertical derivative. To reduce artificial azimuthal anisotropy a hexagonal prism-type grid is employed for the 3-D configuration. An optimization procedure is followed to minimize the errors in phase and group slowness, in the high-frequency limit for a given discretization rate. Through the Bremmer series it also accounts for the backscattered field. The resulting algorithm provides an improvement of the parabolic equation method. Numerical results will be presented at the conference.
Doppler estimation over long ocean acoustic paths
Authors: Johns Murray D.
Published: October 2005

As part of the Acoustic Thermometry of Ocean Climate (ATOC) project, a moored autonomous system with a single hydrophone at the SOFAR axis depth was deployed in early 1996 off the New Zealand coast. The source was on the Pioneer Seamount off the California coast some 10 Mm across the Pacific Ocean from the receiver. Acoustic data have been recovered from the system for a period of 2 or 3 weeks in early April 1996. The received signals are affected by Doppler compression due to receiver motion and this must be accounted for in the data analysis. Problems in estimating and allowing for the effects of Doppler compression in these data will be discussed and results from the above data will be presented.
Sensitivity to acoustic environment of ensembles of acoustic transfer functions and received transients
Authors: Hayward Thomas J.
Published: October 2005

It is well known that the acoustic transfer function (Green’s function) of an ocean waveguide from a particular source position to a particular receiver position may depend sensitively on environmental parameters such as sound-speed profiles and bathymetry. On the other hand, it might reasonably be hypothesized that a large ensemble of acoustic transfer functions, indexed by source position, is in some statistical sense not highly sensitive to details of the acoustic environment. Such ensembles are important in predicting statistical variations of both signal and noise responses for broadband sonar systems. Since the acoustic environment is never known exactly, the sensitivity question needs to be addressed. This question may be investigated by estimating the probability density associated with an ensemble and the variation of this probability density with environmental parameters. In this work, an observation-space partitioning process [J. Acoust. Soc. Am. 96, 3312(A) (1994)] is applied to estimate the probability densities and investigate the sensitivity of ensembles of both acoustic transfer functions and received transients to environmental parameters. Implications for training of classifiers for automated recognition of acoustic transients are discussed.
Optimal source tracking in an uncertain shallow water environment
Authors: Tantum Stacy L. , Nolte Loren W.
Published: October 2005

The optimal approach to tracking a moving source in an uncertain environment is one which incorporates both the continuity of the environment in which the source is located and the nature of the source movement. This differs from the suboptimal approach which performs a series of optimal independent source localizations and then combines the results to estimate the path taken by the source. The optimum uncertain field processor (OUFP) [A. M. Richardson and L. W. Nolte, J. Acoust. Soc. Am. 89, 2280–2284 (1991)] is a class of algorithms which incorporates parameter estimation theory into source localization, making it a method of matched-field processing which is robust to uncertainties in the propagation environment and source parameters. However, previous applications of the OUFP have been for stationary sources. Incorporating the nature of the source motion into the OUFP results in a more accurate method of estimating the path taken by a moving source. The performance of this processor is evaluated by examining the probability of correctly estimating the source path and the mean absolute errors in range and depth of the estimated path. Specific results are presented for a shallow water environment. [Research supported by ONR (Ocean Acoustics).]
Broadband acoustic imaging of complex shaped targets
Authors: Pitt Pat , Chang Min-Fon
Published: October 2005

In some recent experiments conducted at the ARL:UT Lake Travis Testing Facility, acoustic images were created from scattering data of targets with complex geometries. The acoustic measurements were made in the volume in a monostatic scattering configuration at an approximate range of 40 m. The targets were constructed of either fiber-reinforced plastics and/or metals. A variety of large and small bandwidth product waveforms were chosen for the experiments including LFMs, simulated dolphin clicks, and short sinusoidal tone bursts. The frequencies of the waveforms span the range from 15 to 160 kHz. Data were taken by rotating the targets through 360 deg in finely spaced increments about several prechosen axes. The scattering data are processed in several ways. The processed data from different aspects are combined to reconstruct the shape of the object using image processing algorithms based on synthetic aperture and tomographic techniques. By varying the parameters of the incident waveforms, the resulting quality of the acoustic images is examined as a function of frequency and bandwidth. [Work supported by ONR.]
Theoretical investigation of the array bearing error caused by bathymetric refraction
Authors: Riley Joseph M. , Glegg Stewart A. L.
Published: October 2005

Experimental measurements over a laboratory scale model with a real three-dimensional bathymetry have identified the effects of bathymetric refraction on detecting a cw (continuous wave) source with a horizontal line array. The results of these experiments showed large bearing errors which vary rapidly and do not increase monotonically with range. The rapid variation of the bearing errors was unexpected, and a result which has not been previously identified. In order to verify the oscillation in bearing error seen in the experimental results, a simulation of bearing detection using a horizontal line array in a wedge shaped environment was evaluated. This approach uses the method of images to approximate the three-dimensional acoustic field. This theory uses a series of image sources which are centered around the apex of the wedge. The rapid variation in bearing error is shown to be caused by the interference of these image sources which represent eigenrays with different propagation paths. Numerical simulations also investigate the bearing ambiguity caused by bathymetric refraction at full ocean scale. These results identify the magnitude of the bearing error as a function of the acoustic similarity parameter kh (wave number multiplied by water depth). [Work supported by ONR.]
Acoustical measurements in occupied halls
Authors: Kirkegaard Dana L.
Published: October 2005

The presence of musicians and audience may cause significant changes in the reflection structure and the reverberation of a hall. This is particularly true in halls with sound reflective seating, e.g., Boston Symphony Hall. Occupied hall measurements require a minimally intrusive stimulus that is capable of achieving a high signal/noise ratio despite the background noise of the audience. A series of experiments were conducted to determine listener preferences and signal/noise ratios for various measurement stimuli, source levels, and number of averages. Stimuli included various swept sine, noise, maximum length sequence (MLS), and pulse signals. Background noise of the audience was simulated using loudspeakers. The level and spectrum were determined by measurements of background noise in occupied halls. Results indicate that the MLS stimulus, which is widely used for unoccupied hall measurements, is problematic for occupied hall measurements. This research has practical application in conducting occupied hall measurements.
Measurement of spatial impulse responses with a Soundfield Microphone
Authors: Essert Robert
Published: October 2005

Concert hall designers have a keen interest in measuring the spatial evolution of sound fields. So far, 3-D impulse responses have been measured with sound intensity methods, with tetrahedral microphone arrays, and with large microphone arrays. These methods are very sensitive to phase, and therefore to particulars of microphone construction and analysis bandwidth. In this paper a method is proposed for capturing the spatial impulse response of a room by means of orthogonal cosine directivity microphones. Tests have been done with a simple figure-eight/omni microphone pair oriented in three directions, one after the other, as well as with a purpose-built Soundfield microphone. Impulse responses are captured for each of the X, Y, Z, and omni (Soundfield B-format) channels. The directional impulse responses are then cross correlated with the omni impulse response to construct ‘‘direction cosines’’ of strength with respect to the listener position, over time. Data can be mapped graphically in 3-D space to visualize the evolution of the sound field; the degree of scatter in the data represents diffuseness. In addition to graphics, convincing three-dimensional auralizations have been performed with soundfield impulse responses.
Concert hall acoustics from a player’s standpoint
Authors: Senju Mariko
Published: October 2005

Music performance, a ‘‘momentary and spatial art,’’ is finally completed in a concert hall. Even with the same elements such as a musical instrument, a piece by a composer, imagination and technique of a soloist and their co-players, totally different music can be created according to the acoustics of a concert hall. Before a concert, players carefully grasp the acoustical nature of the hall and adapt their technique to harmonize with the hall. When players are successful in their adjustment, they, together with the audience in the hall, can truly create art. As is often said, a concert hall is the greatest musical instrument and, as such, its role in the performance of music cannot be underestimated. This fact demands attention and is worthy of far greater discussion by players, acousticians, and architects. In this paper, a consideration of concert hall acoustics is described from the standpoint of a violin soloist.
Toward an open room acoustics measurement system. I. Overview and hardware
Authors: Essert Robert , Kahle Eckhard
Published: October 2005

Recent development of PC-based acoustical measurement systems has allowed separation of hardware and software development. It is often no longer necessary to buy new hardware in order to add new features, but only to add or change the software algorithm. Having measured and stored room impulse responses (mono, directional, binaural, 3-D), one can use the same raw data later with new computer analysis routines. The CHRG room measurement project and the Alex round robin test have drawn attention to the differences between systems; different measurement teams have a variety of goals and priorities. The hope is to measure and analyze far more than reverberation times. On the one hand, specifications need to be provided that will guarantee comparable data for certain standard measurements, and on the other hand, future development needs to be accommodated. Starting with a superset of goals, an approach is outlined for defining which parts of the system should be subject to standards, while retaining the flexibility for unique or experimental measures.
Toward an open room acoustics measurement system. II. Software
Authors: Kahle Edward , Essert Robert
Published: October 2005

Measurements of objective room acoustical parameters via computer-aided acquisition of impulse responses have become standard practice in the acoustics community, and the number of dedicated computer programs for performing this task has multiplied over the past years. Notwithstanding this significant advance, two problematical areas remain, both of which can be linked to the closed and stand-alone nature of most applications. First, it is very difficult to track discrepancies between the results given by various programs down to individual elements or algorithms. Second, the closed nature of most of the programs inhibits the definition and refinement of new acoustical criteria which take account of our increasing understanding of human perception. To address these problems, and in order to foster more research into the correspondence between the objective description and the subjective evaluation of room acoustical quality, the authors feel that measurement programs will have to become more open, allowing the user to check individual algorithms and define new criteria in a straightforward manner. An outline of such an open, modular, measurement program, using the Matlab environment, will be given.