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Journal of the Acoustical Society of America

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Jun 1994

Volume 95, Issue 6, pp. 3039-3689

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Free vibration of a thin spherical shell containing a compressible fluid

Mingsian R. Bai and Kuorung Wu

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3300-3310 (1994); (11 pages) | Cited 3 times

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Free vibration of a thin spherical shell filled with a compressible fluid is investigated. The interactions at the interface between the elastic structure and the compressible fluid are taken into account. The objective of this study is to develop a hybrid numerical technique for the free vibration analysis of sound–structure interaction problems. The boundary element method is employed for modeling the acoustic disturbances in the cavity, while the finite element method is used for modeling the structural dynamics of the shell. The formulations are then combined into a coupled numerical scheme for the total pressure‐displacement field. Natural frequencies and mode shapes are calculated by using the singular value decomposition algorithm. Physical insights into the resonance phenomena associated with sound–structure interactions are derived from the comparison between the results of the thin spherical shell, with and without the fluid loading effect.
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43.40.Ey Vibrations of shells

Analytical formulations for annular disk sound radiation using structural modes

Ming‐ran Lee and Rajendra Singh

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3311-3323 (1994); (13 pages) | Cited 1 time

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Sound radiation characteristics of an annular disk with application to the computer disk are examined analytically. The far‐field and the radiation impedance approaches are employed to calculate radiated sound of a disk vibrating at its elastic or rigid body modes. Modal sound power is formulated by approximating the structural modal functions and is expressed in terms of a power series of the wave number. Predictions show that axisymmetric vibration modes are more efficient radiators compared to those asymmetric disk modes that have the same number of nodal circles. Numerical results obtained by a boundary element program are used to support analytical predictions. Formulations are also extended to include the modal coupling and source rotation effects. The effect of coupling between disk vibration modes on the radiated sound is found to be significant if multimodes are excited. A simple empirical equation has been developed to predict the modal sound radiation efficiency of a rotating disk for selected vibration modes.
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43.40.Dx Vibrations of membranes and plates
43.40.Yq Instrumentation and techniques for tests and measurement relating to shock and vibration, including vibration pickups, indicators, and generators, mechanical impedance

Surface and body waves generated by a point traction applied on transversely isotropic solids

W. Lin, L. M. Keer, and Y. Xu

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3324-3331 (1994); (8 pages)

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This paper derives analytical expressions for the far‐field surface waves and body waves generated by a point traction on a semi‐infinite, transversely isotropic solid. The closed‐form expressions are also analytically and numerically checked with the related problem for isotropic media. Slowness and wave‐front curves and decay behavior of the surface waves are plotted for magnesium, cobalt, and a composite.
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43.40.At Experimental and theoretical studies of vibrating systems

Numerical model for acoustic radiation of an immersed elastic stucture and application to a thin cylindrical shell

Denise Félix

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3332-3338 (1994); (7 pages) | Cited 2 times

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This paper is concerned with a problem of scattering from an elastic target. It presents a three‐dimensional numerical model for determining the dynamic response and acoustic radiation of an elastic structure in the medium frequency range. This model is applied to a thin cylindrical shell immersed in a fluid. Comparison with experimental results shows the efficiency of this numerical model.
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43.40.Ey Vibrations of shells
43.40.Yq Instrumentation and techniques for tests and measurement relating to shock and vibration, including vibration pickups, indicators, and generators, mechanical impedance

A numerically stable global matrix method for cylindrically layered shells excited by ring forces

David C. Ricks and Henrik Schmidt

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3339-3349 (1994); (11 pages) | Cited 4 times

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This paper defines a numerically stable method for modeling cylindrical shells that can have multiple viscoelastic layers and external compliant coatings. The method is numerically stable over a wide range of axial wavenumbers and circumferential orders because it uses a ‘‘global matrix’’ approach with appropriately defined coefficients. The excitations used here are time‐harmonic ring forces that can push on the shell in the radial, circumferential, and axial directions. The ring forces can have a linear phase shift around the circumference of the shell, so helical waves can be excited and studied for any circumferential order. Analytically, the order can be complex, although the software is presently implemented for real orders. The method is verified by modeling a thin cylindrical shell surrounded by fluid. To provide a very demanding test of the numerical stability, the drive point response of the shell is computed. This computation demonstrates the numerical stability of the method over a wide range of axial wavenumbers and circumferential orders.
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43.40.Ey Vibrations of shells

The effects of elastic edge constraints and fluid loading on the resonant response of thick rectangular plates

Jamel Hammouda and Courtney B. Burroughs

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3350-3359 (1994); (10 pages)

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An analytic model of the response of rectangular plates simply supported on two opposing edges and elastically supported on the other edges is derived. Transverse shear and rotary inertia in the plate and adjustable stiffness constants in the elastic supports are included in the model. The elastic stiffness constants are adjusted to simulate classical boundary conditions and results compared to published results. The variation of frequencies of resonance with the stiffness of the elastic supports is shown. The effect of fluid on the resonant responses of simply supported plates is presented.
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43.40.Dx Vibrations of membranes and plates

Analytical and experimental determination of the vibration and pressure radiation from a submerged, stiffened cylindrical shell with two end plates

A. Harari, B. E. Sandman, and J. A. Zaldonis

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3360-3368 (1994); (9 pages) | Cited 2 times

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An analytical model of a finite, stiffened cylindrical shell with two end plates submerged in fluid was developed. Results obtained for the pressure at various points in the fluid as well as the drive point and transfer point accelerance are compared with experimental results. Good agreement between the analytical and the experimental results are reported.
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43.40.Ey Vibrations of shells
43.40.Yq Instrumentation and techniques for tests and measurement relating to shock and vibration, including vibration pickups, indicators, and generators, mechanical impedance

A new tool for predicting rapidly and rigorously the radiation efficiency of plate‐like structures

Noureddine Atalla and Jean Nicolas

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3369-3378 (1994); (10 pages) | Cited 2 times

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The objectives of this paper are twofold. Firstly, to present a comparison of several models used to evaluate the radiation efficiency of plate‐like radiators. The models compared ranged from very simple ones based on modal average expressions, to refined calculations of the radiation impedance matrix with cross modal coupling. It will be seen in particular that simple models, such as Ver and Holmer’s model, give useful guidelines especially for broadband excitations, spatially distributed forces, and thick plates. Expensive methods, such as the far‐field approach and BEM are normally used when the hypothesis made in simple models are violated (such as with thin plates and situations where cross modal coupling is important). Secondly, to present a new alternative approach allowing a quick and rigorous evaluation of the radiation efficiency of plate like radiators. This new approach, based on a Taylor’s expansion of the Green’s function, has several advantages over the classical approaches which will be discussed in the paper. This approach represents a step towards answering the accuracy versus computation time problematic currently brought up by some engineers.
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43.50.Gf Noise control at source: redesign, application of absorptive materials and reactive elements, mufflers, noise silencers, noise barriers, and attenuators, etc.
43.40.Yq Instrumentation and techniques for tests and measurement relating to shock and vibration, including vibration pickups, indicators, and generators, mechanical impedance
43.40.Dx Vibrations of membranes and plates

Constraint filtered‐x and filtered‐u least‐mean‐square algorithms for the active control of noise in ducts

In‐Soo Kim, Hee‐Seung Na, Kwang‐Joon Kim, and Youngjin Park

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3379-3389 (1994); (11 pages) | Cited 2 times

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In the active control of noise in ducts, it is common practice to locate an error microphone far from the control source to avoid the near‐field effects by evanescent waves. Such a distance between the control source and the error microphone makes a certain level of time delay inevitable and, hence, may yield undesirable effects on the convergence properties of control algorithms such as filtered‐x and filtered‐u least‐mean‐square (LMS). This paper discusses the dependence of the convergence rate on the acoustic error path in these popular algorithms and introduces new algorithms ‘‘constraint filtered‐x and constraint filtered‐u LMS,’’ which increase the convergence region regardless of the time delay in the acoustic error path. In the algorithms, coefficients of the controller are adapted using modified residuals that are defined in such a way that the control process become stationary. Performances of the new LMS algorithms are presented in comparison with those by the conventional algorithms based on computer simulations.
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43.50.Ki Active noise control
43.50.Gf Noise control at source: redesign, application of absorptive materials and reactive elements, mufflers, noise silencers, noise barriers, and attenuators, etc.
43.20.Mv Waveguides, wave propagation in tubes and ducts

Constrained optimization of active noise control systems in enclosures

T. C. Yang, C. H. Tseng, and S. F. Ling

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3390-3399 (1994); (10 pages) | Cited 1 time

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An effective software design tool is proposed for solving active noise control problems associated with constraints in which the complex strength and location of the secondary source of an active noise control system in an enclosed space are simultaneously optimized. The boundary element method is adopted to evaluate the sound field in enclosures. Furthermore, the boundary used could be of pressure, velocity, or impedance; in addition, the primary source may be at an arbitrary position. An optimizer based on sequential quadratic programming is selected for its accuracy, efficiency, and reliability. Bounds for design variables and proper constraints on the sound field and secondary source can be specified as required. The powerfulness of the proposed tool is demonstrated by optimizing an active control system for an enclosure. For a rectangular cavity, the optimal location of the secondary source is confirmed by observed simulations as always forming a dipole with the primary source situated at off‐resonance excitations and subsequently approaching a mirror image position of the primary source at resonance excitations. The optimal location of the controller is found to change with varied upper bounds of the strength of the secondary source. These findings show a discrepancy from those reported in previous researches based on an unconstrained formulation. Sensitivity analysis at the optimum is also included to provide information of practical concern for implementing optimized active noise control systems.
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43.50.Ki Active noise control
43.50.Jh Noise in buildings and general machinery noise
43.55.Ka Computer simulation of acoustics in enclosures, modeling

Inverse filtering of a loudspeaker and room acoustics using time‐delay neural networks

Po‐Rong Chang, C. G. Lin, and Bao‐Fuh Yeh

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3400-3408 (1994); (9 pages) | Cited 1 time

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This paper presents the design of a neural‐based acoustic control used for the equalization of the response of a sound reproduction system. The system usually can be modeled as a composite system of a loudspeaker and an acoustic signal‐transmission channel. Generally, an acoustic signal radiated inside a room is linearly distorted by wall reflections. However, in a loudspeaker, the nonlinearity in the suspension system produces a significant distortion at low frequencies and the inhomogeneity in the flux density causes a nonlinear distortion at large output signals. Both the linear and nonlinear distortions should be reduced so that high fidelity sound can be reproduced. However, the traditional adaptive equalizer which is only capable of dealing with linear systems or specific nonlinear systems cannot compensate these nonlinear distortions. The time‐delay feedforward neural network (TDNN) which has the capability to learn arbitrary nonlinearity and process the temporal audio patterns are particularly recognized as the best nonlinear inverse filter of the composite system. The performance of a TDNN‐based acoustic controller is verified by some simulation results.
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43.60.Gk Space-time signal processing, other than matched field processing
43.38.Ar Transducing principles, materials, and structures: general
43.55.Mc Room acoustics measuring instruments, computer measurement of room properties

Robust sequential MIPA array processor with dependent sampling

Mohammed Ketel and Ludwik Kurz

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3409-3418 (1994); (10 pages)

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The problem of detecting small stochastic signals in poorly specified noise environments, often encountered in underwater acoustics, is considered. The array processing detectors are designed to exhibit a near‐optimum performance for nominal distributions and maintain high efficiency to changing noise environments by adapting their optimum nonlinearities using an m‐interval polynomial approximation (MIPA) of them. Sequential operation of the detectors is used to reduce the average time to decision. Furthermore, the assumption of independent samples is relaxed while temporal and spatial noise dependences are considered separately. A formulation is given that predicts the thresholds under the time‐dependent sampling in order to maintain the same error probabilities as in the independent sampling case. For the space‐dependent noise process, the thresholds remain constant for fixed error probabilities. Instead, the cross‐scores need to be adjusted to compensate for space dependence. The performance of these detectors is compared to that of the corresponding independent noise. It is concluded that improved performance is achieved with a slight increase in structural complexity.
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43.60.Gk Space-time signal processing, other than matched field processing
43.60.Cg Statistical properties of signals and noise

Robust Fourier descriptors for characterizing amplitude‐modulated waveform shapes

Ben Pinkowski

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3419-3423 (1994); (5 pages) | Cited 3 times

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Fourier descriptors (FD’s) have traditionally been used to characterize the boundary shape of closed curves in computer‐based images. This study examines the usefulness of FD’s for characterizing the shape of amplitude‐modulated waveform boundaries. Twenty‐eight bioacoustic waveforms representing 7 species‐specific sound groups were examined. Nearest‐neighbor classification using 32‐point waveform envelopes yielded 85.7% recognition rates when Euclidean distances with 8 or 12 FD’s were employed. With 4 or 16 FD’s, recognition was 82.1% but dropped to 71.4% when the number of FD’s was reduced to 3. Evidently only the low‐order 4 or 5 FD’s are required to characterize most signals examined. Four FD’s were adequate representation for linear discriminant analysis, wherein 27 (96.1%) of the samples were classified correctly, and for hierarchical cluster analysis, wherein 26 (92.9%) of the samples clustered into dendrogram subtrees corresponding to the known sound groups. Two sounds accounted for many errors in the classification experiments, and these differed from typical sounds for their respective species groups. In such cases, more than a single FD reference pattern may be required to adequately represent the sounds of a given species.
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43.60.Gk Space-time signal processing, other than matched field processing

Effects of refraction on synthetic aperture sonar imaging

Kenneth D. Rolt and Henrik Schmidt

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3424-3429 (1994); (6 pages)

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Synthetic aperture sonar (SAS) imaging in a refracting environment requires that refraction be accounted for in the image processing to avoid blurring of the images. In deep water, e.g., this requires some knowledge of the refraction sound‐speed profile c(z). Ideally a ‘‘local’’ c(z) profile could be measured, but the measurement may only be valid for one platform position (not necessarily valid across the full synthetic aperture length) and at that one particular time. Historical data for c(z) may also be used for the region being imaged, but these data will not necessarily match the actual c(z) present during sonar data collection. This paper shows that SAS imagery in a refracting environment requires knowledge of the refraction for successful imaging, and that the refraction profile c(z) is a necessary part of matched‐filter image processing. For a deep water example where the exact c(z) profile is unknown, the use of historical c(z) data and reduced Doppler bandwidth, reduced‐resolution processing leads to useful sharp images that suffer only from incorrect bottom registration.
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43.60.Gk Space-time signal processing, other than matched field processing
43.60.Sx Acoustic holography

Transfer characteristic of the inner hair cell synapse: Steady‐state analysis

Mark Zagaeski, Alan R. Cody, Ian J. Russell, and David C. Mountain

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3430-3434 (1994); (5 pages) | Cited 5 times

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Inner hair cells (IHC) transduce mechanical to electrical energy in the mammalian cochlea producing a receptor potential which is a rectified, filtered representation of the mechanical input to the hair cell. The IHC synapse transfers the information in the receptor potential to the fibers of the auditory nerve (whose cell bodies form the spiral ganglion) where it is encoded as a pattern of action potentials. That transfer was investigated by comparing the steady‐state responses in pre‐ and post‐synaptic cells. A nonlinear transfer characteristic describing the synapse was generated by plotting the spiral ganglion cell firing rate as a function of the IHC receptor potential. For each spiral ganglion unit, the operating range maps onto a different portion of the nonlinear inner hair cell operating range, dependent on the neural unit’s threshold. Units whose rate‐level functions exhibit similar slopes but different thresholds can have dramatically differing sensitivities to changes in the IHC potential. This threshold‐dependent mapping supports the concept that information may be distributed amongst nerve fibers according to their threshold.
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43.64.Ld Physiology of hair cells
43.64.Pg Electrophysiology of the auditory nerve
43.64.Bt Models and theories of the auditory system

Measuring the human head‐related transfer functions: A novel method for the construction and calibration of a miniature ‘‘in‐ear’’ recording system

Danièle Pralong and Simon Carlile

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3435-3444 (1994); (10 pages) | Cited 6 times

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A method for the construction of a small ‘‘in‐ear’’ system for recording the human‐free field‐to‐eardrum and headphone‐to‐eardrum transfer functions is described. Customized in‐ear inserts were obtained by a simple ear printing and electroplating method, resulting in a thin (<0.25 mm) outer shell that minimized obstruction of the entrance at the ear canal. The insert can be used to position a microphone probe tube deep within the auditory canal. The effects of this recording system on the sound field in the ear canal were calibrated using a model head equipped with a second internal microphone close to the eardrum. Transfer functions were recorded for 343 different stimulus locations in free space and for a headphone sound source. For the free‐field stimuli the presence of the recording system resulted in a small attenuation with maximum effects around 3.5 and 12.5 kHz (−1.5 and −2.0 dB, respectively). Passing the data through an auditory filter model reduced the averaged attenuation to less than −1.4 dB. Phase was undistorted up to 2.5 kHz. These results suggests that the perturbations produced by the insert are unlikely to be perceptually relevant.
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43.64.Ha Acoustical properties of the outer ear; middle-ear mechanics and reflex
43.64.Yp Instruments and methods
43.66.Yw Instruments and methods related to hearing and its measurement

The location‐dependent nature of perceptually salient features of the human head‐related transfer functions

Simon Carlile and Danièle Pralong

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3445-3459 (1994); (15 pages) | Cited 8 times

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The human head‐related transfer function (HRTF) has been recorded binaurally from eight subjects using an ‘‘in‐ear’’ recording system, for 343 stimulus locations around the head. There are a number of systematic changes in the HRTF as a function of horizontal location and elevation, on and off the median plane, that could be used as cues to sound location. To identify which components of the HRTF might provide perceptually salient cues to sound location, the HRTFs were transformed using an auditory filter model which accounts for the frequency dependence of auditory sensitivity and the frequency and level‐dependent characteristics of the auditory filters. These transformations indicated a systematic variation in the frequency of the peak excitation as a function of the horizontal location of a broad band stimulus. Furthermore, there were differences in the frequency range over which elevation‐dependent changes in the excitation patterns varied as a function of the vertical meridian. Interaural level differences were also estimated using the excitation patterns. The across frequency pattern of ILDs were roughly symmetrical about the interaural axis, although there were substantial differences between each ear in the magnitude of the ILDs generated for ipsilateral sounds locations.
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43.64.Ha Acoustical properties of the outer ear; middle-ear mechanics and reflex
43.66.Qp Localization of sound sources

Otoacoustic emissions and quinine sulfate

Dennis McFadden and Edward G. Pasanen

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3460-3474 (1994); (15 pages) | Cited 9 times

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A moderate dose of quinine sulfate, administered to three young adult males, reduced or eliminated various forms of otoacoustic emission (OAE). The individual differences in response to the drug were substantial, but a number of generalizations did emerge. The time courses of onset and recovery were considerably more rapid than for the parallel effects produced by aspirin. Most spontaneous otoacoustic emissions (SOAEs) were eliminated within 7 h of the first 325‐mg dose (about 3 h after the second dose). Most SOAEs showed partial or complete recovery about 24 h after the last dose, although considerable instability often remained. The functions relating the magnitude of a distortion‐product OAE (DPOAE) to the sound‐pressure level (SPL) of the primary tones producing it were displaced toward higher primary levels and became lower sloped following quinine administration. The magnitudes of SOAEs, DPOAEs, and nonlinear peaks in the click‐evoked spectra declined and recovered with grossly similar time courses, but there were some partial dissociations. The ability of a DPOAE to suppress an SOAE lying about 50 Hz below it either increased slightly or remained about constant through the drug episode, even though the magnitudes of both DPOAE and SOAE were changing. On several occasions, increases in SOAE magnitude of as much as 10–20 dB were observed during the first 15–30 min of an SOAE measurement period (an initializing effect). Psychophysical measures revealed hearing losses of as much as 20 dB at some frequencies in some subjects. Several short‐lived ‘‘enhancements’’ of OAEs are discussed relative to similar quinine‐induced effects reported in an animal model.
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43.64.Jb Otoacoustic emissions
43.64.Kc Cochlear mechanics
43.64.Gz Biochemistry and pharmacology of the auditory system

Reducing informational masking by sound segregation

Gerald Kidd, Jr., Christine R. Mason, Phalguni S. Deliwala, William S. Woods, and H. Steven Colburn

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3475-3480 (1994); (6 pages) | Cited 52 times

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Informational masking was reduced using three stimulus presentation schemes that were intended to perceptually segregate the signal from the masker. The maskers were sets of sinusoids chosen randomly in frequency and intensity on each stimulus interval or, in some conditions, on every masker burst in a series of bursts within intervals. Masker components were excluded from the frequency region surrounding the 1000‐Hz signal to minimize the energetic masking. Masked thresholds as great as 60–70 dB above quiet threshold were observed for some subjects in some conditions. It was shown that this informational masking could be reduced as much as 40 dB by: (1) presenting the masker to both ears and signal to one ear; (2) playing different masker samples sequentially in each interval of every trial; or (3) presenting the signal in alternate bursts of multiple, identical masker samples. For the binaural manipulation, informational masking was reduced because the masker and signal were perceived as originating from different interaural locations. In the latter two manipulations, a difference in the spectral or temporal pattern of the signal and masker provided the detection cue. These effects were interpreted as evidence of the importance of perceptual segregation of sounds in noisy listening environments where signal reception is not limited by energetic masking.
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43.66.Dc Masking
43.66.Lj Perceptual effects of sound
43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music
43.66.Pn Binaural hearing

Experiments related to the detection of a tone masked by another tone

Virginia M. Richards

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3481-3498 (1994); (18 pages) | Cited 1 time

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Five experiments were completed in an effort to determine which cues allow the detection of a tone in the presence of another tone of a different frequency. Four cues associated with the added tone were examined (a) changes in overall level, (b) changes in envelope modulation (beats), (c) changes in pitch, and (d) changes in phase (frequency) modulation. Detection thresholds were measured using a two‐interval, forced choice paradigm, and for maskers of 400 and 3000 Hz. The data suggest that the presence of beats acts as a cue for detection when the signal and masker frequencies are within ten percent of one another. For larger frequency separations the data did not support any of the detection models examined [(a)–(d) above]. Based on limited direct observations and on the failure of the single‐channel models tested, it appears that the detection of a tone in the presence of another tone depends on the detection of changes in the pattern of intensity and/or the pattern of temporal synchrony across frequency.
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43.66.Dc Masking

Detection and discrimination of amplitude‐modulated signals by macaque monkeys

David B. Moody

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3499-3510 (1994); (12 pages) | Cited 3 times

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Behavioral thresholds were determined from macaque monkeys for detection of amplitude modulation (AM) and for discrimination of increases in AM frequency. A positive‐reinforcement, go/no‐go behavioral paradigm was used with a two‐down/one‐up psychophysical procedure for all determinations. In the first set of experiments, temporal modulation transfer functions (TMTFs) were determined at two different stimulus levels, for both gated and continuous noise carriers. In the second set of experiments, difference limens for AM frequency were determined using modulated noise and pure‐tone carrier signals. TMTFs for gated carriers exhibited a bandpasslike characteristic as has been previously shown. The high‐frequency cutoff determined from the average of the gated‐carrier TMTFs obtained at 58 dB SPL was 198 Hz, higher than that shown with wideband carriers for other species. With a continuous carrier, there was less of a low‐frequency cutoff in the TMTF, again corresponding to previous results. Unlike previous results, however, the present TMTFs showed an effect of stimulus level. Difference limens for AM frequency increased as a function of standard modulation frequency and then leveled off or decreased slightly with further increases in AM frequency. Taken together, the AM discrimination data, coupled with the high cutoff frequency of the TMTF, suggest that detection and discrimination of rapid temporal events may play an important role in the acoustic world of primates.
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43.66.Fe Discrimination: intensity and frequency
43.66.Gf Detection and discrimination of sound by animals
43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music
43.80.Lb Sound reception by animals: anatomy, physiology, auditory capacities, processing

Detection of mixed modulation using correlated and uncorrelated noise modulators

Aleksander Sek and Brian C. J. Moore

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3511-3517 (1994); (7 pages) | Cited 1 time

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This article is concerned with the mechanisms underlying the detection of amplitude modulation (AM), frequency modulation (FM), and mixed modulation (MM), i.e., simultaneously occurring AM and FM. In a previous study [B. C. J. Moore and A. Sek, J. Acoust. Soc. Am. 92, 3119–3131 (1992)], psychometric functions were measured for the detection of AM alone and FM alone, using a 10‐Hz sinusoidal modulator and a 1‐kHz carrier frequency. Detectability was then measured for combined AM and FM, with modulation depths selected so that each type of modulation would be equally detectable if presented alone. The detectability of the MM was better than would be predicted if the two types of modulation were coded completely independently. This study examined the possibility that the good detectability of MM was caused by the fact that the AM and the FM were correlated, so that each was predictable from the other. The design was similar to that of our earlier study, but the 10‐Hz sinusoidal modulator was replaced by a narrow‐band noise modulator. In the MM conditions, the modulators for AM and FM were either strongly positively correlated or essentially uncorrelated. In experiment 1, the waveforms of the noise modulators were fixed throughout the experiment (frozen noise). In experiment 2, the waveforms of the noise modulators were chosen independently for each trial. In both experiments, for both correlated and uncorrelated modulators, the detectability of the MM was better than would be predicted if the two types of modulation were coded completely independently. Performance was better for the correlated modulators than for the uncorrelated modulators. The results indicate that the high detectability of MM cannot be attributed solely to one type of modulation (e.g., FM) being predictable from the other type of modulation (e.g., AM).
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43.66.Fe Discrimination: intensity and frequency
43.66.Ba Models and theories of auditory processes
43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music

The internal representation of spectral contrast in hearing‐impaired listeners

Van Summers and Marjorie R. Leek

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3518-3528 (1994); (11 pages) | Cited 18 times

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Abnormal frequency resolution associated with sensorineural hearing impairment produces a smearing of spectral detail in the internal representation of complex acoustic stimuli. As a result, listeners with hearing loss may have difficulty locating spectral peaks (e.g., vowel formants) within stimuli which cue their identity. This study examined the relationship between frequency separation of peaks in a complex spectrum and the degree of spectral contrast preserved in the internal representations in normal and impaired auditory systems. Hearing‐impaired and normal‐hearing subjects discriminated a flat‐spectrum bandpass stimulus from a stimulus containing a sinusoidal ripple across its frequency range. The peak‐to‐valley amplitude (in dB) necessary for detection of the ripple was measured for ripple frequencies ranging from 1 to 9 cycles/oct. Auditory filter characteristics were also measured at 1 and 3 kHz in order to examine the internal representations of the stimuli after cochlear processing. There were clear differences between groups in both auditory filter characteristics and spectral contrast detection. However, the amount of contrast in the internal representations predicted from these measurements was nearly the same for all subjects, suggesting that the reduced frequency resolution of the hearing‐impaired group was largely responsible for differences in required peak‐to‐valley amplitude in the input spectra. Further, for all subjects, there was a trade‐off between the absolute level of internal contrast necessary for ripple detection and the number of samples of this contrast available to the listener.
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43.66.Fe Discrimination: intensity and frequency
43.71.Ky Speech perception by the hearing impaired
43.66.Jh Timbre, timbre in musical acoustics

The role of resolved and unresolved harmonics in pitch perception and frequency modulation discrimination

Trevor M. Shackleton and Robert P. Carlyon

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3529-3540 (1994); (12 pages) | Cited 85 times

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A series of experiments investigated the influence of harmonic resolvability on the pitch of, and the discriminability of differences in fundamental frequency (F0) between, frequency‐modulated (FM) harmonic complexes. Both F0 (62.5 to 250 Hz) and spectral region (LOW: 125–625 Hz, MID: 1375–1875 Hz, and HIGH: 3900–5400 Hz) were varied orthogonally. The harmonics that comprised each complex could be summed in either sine (0°) phase (SINE) or alternating sine‐cosine (0°–90°) phase (ALT). Stimuli were presented in a continuous pink‐noise background. Pitch‐matching experiments revealed that the pitch of ALT‐phase stimuli, relative to SINE‐phase stimuli, was increased by an octave in the HIGH region, for all F0’s, but was the same as that of SINE‐phase stimuli when presented in the LOW region. In the MID region, the pitch of ALT‐phase relative to SINE‐phase stimuli depended on F0, being an octave higher at low F0’s, equal at high F0’s, and unclear at intermediate F0’s. The same stimuli were then used in three measures of discriminability: FM detection thresholds (FMTs), frequency difference limens (FDLs), and FM direction discrimination thresholds (FMDDTs, defined as the minimum FM depth necessary for listeners to discriminate between two complexes modulated 180° out of phase with each other). For all three measures, at all F0’s, thresholds were low (<4% for FMTs, <5% for FMDDTs, and <1.5% for FDLs) when stimuli were presented in the LOW region, and high (≳10% for FMTs, ≳7% for FMDDTs, and ≳2.5% for FDLs) when presented in the HIGH region. When stimuli were presented in the MID region, thresholds were low for low F0’s, and high for high F0’s. Performance was not markedly affected by the phase relationship between the components of a complex, except for stimuli with intermediate F0’s in the MID spectral region, where FDLs and FMDDTs were much higher for ALT‐phase stimuli than for SINE‐phase stimuli, consistent with their unclear pitch. This difference was much smaller when FMTs were measured. The interaction between F0 and spectral region for both sets of experiments can be accounted for by a single definition of resolvability.
Show PACS
43.66.Hg Pitch
43.66.Nm Phase effects
43.66.Ba Models and theories of auditory processes

Comparing the fundamental frequencies of resolved and unresolved harmonics: Evidence for two pitch mechanisms?

Robert P. Carlyon and Trevor M. Shackleton

J. Acoust. Soc. Am. Volume 95, Issue 6, pp. 3541-3554 (1994); (14 pages) | Cited 50 times

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Four experiments measured sensitivity (d′) to differences in fundamental frequency (F0) between two simultaneously presented groups of frequency‐modulated harmonics. Each group was passed through a bandpass filter in either a LOW (125–625 Hz), MID (1375–1875 Hz), or HIGH (3900–5400 Hz) frequency region. In the first two experiments, a dynamic F0 difference (ΔF0) was created by introducing a 180° disparity between the frequency modulations imposed on the two groups. Experiment 1 measured sensitivity to such ΔF0’s between a MID group with a baseline F0 of 125 Hz and all components summed in sine phase, and a HIGH group, in four conditions. When the baseline F0 of the HIGH group was also 125 Hz, performance was good when its components were summed in sine phase and bad when they were in alternating phase. Conversely, when the HIGH F0 was 62.5 Hz, performance was better for alternating phase than for sine phase, consistent with alternating phase doubling the internal representation of HIGH group’s F0. Similar results were obtained for a comparison between the LOW and MID groups. Experiment 2 measured sensitivity to ΔF0’s between either the LOW and MID groups or between the MID and HIGH groups, for baseline F0s of 88 and 250 Hz. Sensitivity was best when the combination of frequency region and F0 was such that both groups were resolved or both unresolved by the peripheral auditory system, and worst when the groups differed in ‘‘resolvability.’’
Experiment 3 replicated experiment 2 using a different paradigm, in which the two groups were always modulated coherently, and in which the ΔF0 was constant throughout the signal. Experiments 2 and 3 also measured sensitivity to differences in F0 between successively presented tokens of the same group. Experiment 4 showed that the high sensitivity to (simultaneous) ΔF0’s when both groups were unresolved could be attributed to listeners detecting the pitch pulse asynchronies that inevitably arise from ΔF0’s. Finally, a method for predicting sensitivity to simultaneous, across‐frequency ΔF0’s from that to successive within‐channel differences, on the basis of optimum combination of information, was applied to the results of experiment 3. The method succeeded in predicting sensitivity to ΔF0’s between two groups of resolved harmonics, but over‐estimated performance when one group was resolved and the other unresolved. The results suggest that the F0’s of resolved and unresolved harmonics are processed by separate mechanisms, in contrast to the predictions of ‘‘autocorrelation’’ models of F0 encoding.
Show PACS
43.66.Fe Discrimination: intensity and frequency
43.66.Hg Pitch
43.66.Nm Phase effects
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