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Prior investigations have generally failed to confirm or deny the perceptual influence of highfrequency harmonics contained in musical sounds. Embedded within this experiment were two listening tests: one investigating threshold for differences in timbre, and thus, participant ability, and another seeking to find an influence of high-frequency harmonic content on timbre perception. Based on the premise that harmonics out of the range of auditory detection influence the resultant waveform and therefore may alter the percept of a sound's tonal character, this study found no evidence that capable listeners noticed an effect of high-frequency harmonics within a brief complex tone. [ABSTRACT FROM AUTHOR]

The just noticeable differences (JNDs) of room acoustics metrics are necessary for research and design of performing arts venues. The goal of this work was to evaluate the effects of different testing methods on the measured JND of clarity index for music (C80). An initial study was conducted to verify the findings of other published works that the C80 JND is approximately 1 dB, as currently listed in ISO 3382:2009 (International Organization for Standardization, Switzerland, 2009), however, the results suggested a higher value. In the second study, the effects of using two variations of the method of constant stimuli were examined, where one variation required the subjects to evaluate the pair of signals by listening to each of them in their entirety, while the second approach allowed the participants to switch back and forth in real-time. More consistent results were obtained with the latter variation and the results indicated a C80 JND greater than 1 dB. In the final study, an extensive training period using the first variation was required, based on the second study, and the data were collected using the second variation. The analysis revealed that for the conditions used in this study (concert hall and chamber music hall) that the C80 JND is approximately 3 dB. [ABSTRACT FROM AUTHOR]

The integration of basic geometrical acoustics into architectural design is not new. The arrival of 3D modelling packages such as Grasshopper has certainly increased the consultants' efficiency and dialogues within the design team. It has also forced the designers to go back to the basics. Efficient algorithms are complex and their development time is consuming. At first, Grasshopper's apparent limitation in complex calculus seems to confine its use to basic trigonometry, but this is not the case anymore. Much has been said about geometry analysis. But few acoustic designs are driven by actual parametric design. Recent collaborations with architects well versed in parametric design presented opportunities for the further development of acoustic concepts into real input of parametric design. Grasshopper can indeed be used to resolve inverse problems for which the output is assumed. The unknown to the problem is the 'precise geometry of the hall'. This paper describes the collaborative approach to develop a single algorithm for both acoustic and architectural form generation. [ABSTRACT FROM AUTHOR]

This paper considers the noise performance of microphone preamplifiers at low gain setting. A detailed noise analysis of the classic current-feedback instrumentation amplifier topology often used for transformerless microphone preamplifiers is presented. The noise contributors that are significant at low gain settings are identified. Several techniques to reduce their magnitude are discussed. One novel technique is introduced that is easily implemented at very low cost and requires a modest increase in power consumption only. Furthermore, it offers a benefit with respect to the frequency response and distortion performance of the circuit. The theoretical results are verified with experimental measurement data. [ABSTRACT FROM AUTHOR]

When we assess the quality of a voice recording two different aspects play a role--the voice characteristics (voice quality) and the audio chain characteristics (audio quality). Subjective experiments where no clear ideal reference is provided, so called absolute category rating experiments, assess the speech quality, i.e., the combined effect of voice and audio quality. This paper investigates whether voice postprocessing such as timbre optimization, loudness optimization, de-essing, room reverberation optimization, and (background) noise suppression can improve the quality of a high quality voice recording. It turned out that none of the processing provides a significant improvement in perceived quality. The best postprocessing is noise reduction to absolute silence, delivering only a non-significant improvement when the voice recording is of high quality. The subjective quality evaluations show a significant preference of male over female voice and a significant effect of speaker/sentence dependency on the perceived quality of certain types of degradation. The subjective results are compared with predictions made with the ITU-T standard for the objective assessment of speech quality POLQA (ITU-T Recommendation R863 versions 1.1 and 2.4) and shows that many speech quality effects are predicted correctly, on condition level as well as individual sentence level. [ABSTRACT FROM AUTHOR]

A new control technique for electrodynamic transducers is presented that stabilizes the voice coil position, compensates for nonlinear distortion, and generates a desired transfer response by preprocessing the electrical input signal. The control law is derived from transducer modeling using lumped elements and identifies all free parameters of the model by monitoring the electrical signals at the transducer terminals. The control system stays operative for any stimulus including music and other audio signals. The active stabilization is important for small loudspeakers generating the acoustical output at maximum efficiency. [ABSTRACT FROM AUTHOR]

Sound is a prominent element in the communication of humans with their environment. It can be either in an organized form, e.g., speech or music, or in a non-organized form, i.e., Sound Events (SEs). There are many studies focusing on emotion recognition from music content, but the research on emotion recognition from SEs is a rather new field. This works falls within this field and concludes that the rhythmic characteristics of an SE have a major impact on the listener's arousal. This result verifies the empirical knowledge that the rhythm of sound affects the listener's activation state. Moreover, we also investigate whether the above characteristics also affect the pleasure of the listener. Toward this aim, we have utilized a well known data set of emotionally annotated SEs, extracted various rhythm-related technical cues, and conducted a series of machine learning experiments. The overall results indicate a relation of the rhythm and listener's valence with accuracy results reaching up to 63%. [ABSTRACT FROM AUTHOR]

The acoustic behavior of ceiling absorbers can be predicted under different surface reaction assumptions: Local and extended reaction. This study aims to experimentally validate acoustic transfer functions near a ceiling absorber in an anechoic chamber based on the two surface reaction models. First, a ceiling absorber with two mounting conditions is modeled by equivalent fluid models, such as Delany-Bazley's, Miki's, and Komatsu's model, in various ways: (1) Local vs extended reaction and (2) plane-wave vs spherical-wave incidence. For a single absorber under anechoic conditions, the acoustic transfer functions for four source-receiver pairs are simulated using a pressure-based image source model, and then compared with measurements. For a rigid backing condition, both the local and extended reaction models agree well with the measurement. For an absorber backed by an air cavity, the extended reaction model agrees better at larger incidence angles at lower frequencies than the local reaction model. [ABSTRACT FROM AUTHOR]

This paper presents measurements and visualization of sound intensity around a human head simulator in a free field. A Cartesian robot, applied for precise positioning of the acoustic vector sensor, was used to measure sound intensity. Measurements were performed in a free field using a head and torso simulator and a setup consisting of four different loudspeaker configurations. The acoustic vector sensor was positioned around the head with a 5-cm step. Sound intensity was measured in 277 points. For every step three orthogonal sound intensity components were calculated. Pure tones of frequencies 250, 1000, and 4000 Hz were applied to analyze the acoustic field. Obtained results were used to provide visualizations of sound intensity distribution around the human head. The tool developed for this purpose utilized three-dimensional sound intensity measurements and visualization techniques. [ABSTRACT FROM AUTHOR]

Stereophonic reproduction is widely used to present audio information to listeners and to control sound image positions using amplitude panning in audio-visual systems and audio systems. However, with stereophonic reproduction, sound images are localized correctly only when a listener is located in a small listening position, a so-called sweet spot. This study was undertaken to develop a stereophonic reproduction system that provides precise sound image localization even to a listener who is not at the sweet spot. We propose a stereophonic reproduction system that detects a listener's position in a contactless manner and that controls the amplitude ratio and delay between left and right loudspeaker signals adaptively with respect to the listener's position, thereby engendering accurate sound image localization irrespective of the listener's position. Results of subjective experiments demonstrate that sound images presented using the proposed method are localized more accurately than with the conventional method, even when the listener is not located at the sweet spot. [ABSTRACT FROM AUTHOR]

Sound zone systems aim to produce regions within a room where listeners may consume separate audio programs with minimal acoustical interference. Often, there is a trade-off between the acoustic contrast achieved between the zones and the fidelity of the reproduced audio program (the target quality). An open question is whether reducing contrast (i.e., allowing greater interference) can improve target quality. The planarity control sound zoning method can be used to improve spatial reproduction, though at the expense of decreased contrast. Hence, this can be used to investigate the relationship between target quality (which is affected by the spatial presentation) and distraction (which is related to the perceived effect of interference). An experiment was conducted investigating target quality and distraction and examining their relationship with overall quality within sound zones. Sound zones were reproduced using acoustic contrast control, planarity control, and pressure matching applied to a circular loudspeaker array. Overall quality was related to target quality and distraction, each having a similar magnitude of effect; however, the result was dependent upon program combination. The highest mean overall quality was a compromise between distraction and target quality, with energy arriving from up to 15 degrees either side of the target direction. [ABSTRACT FROM AUTHOR]

Audio-on-audio interference situations are a common occurrence in everyday life; they may be naturally occurring or be a side-effect of a non-ideal personal sound zone system. In order to evaluate and optimize such situations in a perceptually relevant manner, it is desirable to develop a model of listener experience. Distraction ratings were collected for 100 randomly created audio-on-audio interference situations with music target and interferer programs. A large set of features was also extracted from the audio; the feature extraction was motivated by a qualitative analysis of subject responses. An iterative linear regression procedure was used to develop a predictive model. The selected features were related to the overall loudness, loudness ratio, perceptual evaluation of audio source separation (PEASS) toolbox interference-related perceptual score, and frequency content of the interferer. The model was found to predict accurately for the training and validation data sets (RMSE of approximately 10%), with the exception of a small number of outlying stimuli. [ABSTRACT FROM AUTHOR]

Recent high-resolution video technologies have enhanced the immersive experience of reproduced video content. To generate auditory images that can integrate with such high-resolution video, many spatial sound-rendering methods have been proposed. However, with such advanced methods, creating a convincing near-field sound image remains a challenge. In order to render such a near-field sound image in a relatively simple yet effective way, the authors have proposed a new method using an overhead planar loudspeaker. The subjective evaluation results showed that planar waves radiating from an overhead position generated a sound image very near to the listener when coupled with an additional filter that removes spectral cues associated with an overhead sound source. The results also showed that the proposed method could continuously control the distance of sound image between a screen and a listener position. Additional physical measurements revealed that the planar loudspeaker generated a smaller variance of group delay function at the listener's ears than conventional loudspeakers. This smaller variance of group delay function was the physical parameter associated with the near-field images. [ABSTRACT FROM AUTHOR]

Physical synthesis of the sound field that is generated by a moving virtual sound source has been the subject of extensive research in the recent decade. The article presents a new analytical modeling approach based on the wavenumber domain representation of the field emitted by a virtual monopole under uniform motion. Explicit and implicit frequency domain driving function expressions are derived for planar and linear secondary sources using the Spectral Division Method (SDM) and Wave Field Synthesis (WFS). The proposed method results in analytically correct driving functions and thereby correct synthesis of the Doppler effect. [ABSTRACT FROM AUTHOR]

A prototype 2D surround sound system was developed to investigate the use of higher order loudspeakers and active compensation to both eliminate unwanted reflections and to improve spatial accuracy in semi-reverberant rooms. The loudspeakers, and the calibration microphone, were cylindrical designs as these are lower cost than spherical designs and better suited to 2D reproduction. This paper presents the cylindrical coordinate-based theory for the calibration of the system, a description of the transducers and processing system, and experimental results obtained using a sound reproduction system consisting of five third-order loudspeakers. Approximate anechoic reproduction is achieved over a radius exceeding 90 mm and over a frequency range up to 8 kHz. [ABSTRACT FROM AUTHOR]

Digital equalization of audio systems is mostly performed on a channel-by-channel basis, i.e., loudspeakers are equalized separately and independently of each other. In reverberant rooms the spatial variability of the loudspeaker-room transfer function puts a limit on what can be achieved with such single-channel approaches, and considerably more far-reaching results can be obtained with multichannel methods. In this paper we present a multichannel equalizer design method where the loudspeaker to be equalized is assisted by several support loudspeakers that act in combination to improve the response of the main loudspeaker while suppressing the reverberation of the listening room. However, since a full dereverberation may not always be desirable, we shall here study the proposed method from the perspective of how to control the amount of room correction for a given number of contributing support loudspeakers. The amount of correction is controlled by a scalar parameter that we interpret as a "focus control," enabling the user to slide seamlessly between a single-channel equalizer and a full multichannel inversion of the room. [ABSTRACT FROM AUTHOR]

This letter investigates the impact of spectral compression on the vector Taylor series-based model adaptation algorithm. Unlike mel-frequency cepstral coefficients obtained by the logarithmic compression, the fractional power compression is used for extracting features. Since the relationship between acoustic models for clean and noisy speech depends on nonlinearity of the spectrum, it is important to select an appropriate compressive operator in the model adaptation. In this letter, the dependency of spectral nonlinearity on the speech recognition system is analyzed in various noisy environments. Experimental results confirm that the replacement of the compressive operator improves the performance of the model adaptation. [ABSTRACT FROM AUTHOR]

The covariance matrix of a multichannel audio signal is a measure that contains the channel energies and the inter-channel dependencies. This measure is perceptually relevant in frequency bands, since with the effect of the acoustic transfer path, it forms the inter-aural cues based on which the human spatial hearing system decodes the spatial sound. This is the foundation to several state of the art perceptual spatial processing techniques. In this paper we propose a generalized framework for spatial sound processing that operates directly in the covariance matrix domain. The technique optimizes the sound quality by a least mean square solution with respect to a defined prototype signal. Furthermore, the technique uses all available independent signal components in the input channels in order to minimize the undesired effects due to the decorrelators that are often employed in the concerned use cases. A wide variety of applications of the proposed method has been identified. [ABSTRACT FROM AUTHOR]

This paper presents an acoustic-to-articulatory inversion method based on an episodic memory. An episodic memory is an interesting model for two reasons. First, it does not rely on any assumptions about the mapping function but rather it relies on real synchronized acoustic and articulatory data streams. Second, the memory inherently represents the real articulatory dynamics as observed. It is argued that the computational models of episodic memory, as they are usually designed, cannot provide a satisfying solution for the acoustic-to-articulatory inversion problem due to the insufficient quantity of training data. Therefore, an episodic memory is proposed, called generative episodic memory (G-Mem), which is able to produce articulatory trajectories that do not belong to the set of episodes the memory is based on. The generative episodic memory is evaluated using two electromagnetic articulography corpora: one for English and one for French. Comparisons with a codebook-based method and with a classical episodic memory (which is termed concatenative episodic memory) are presented in order to evaluate the proposed generative episodic memory in terms of both its modeling of articulatory dynamics and its generalization capabilities. The results show the effectiveness of the method where an overall root-mean-square error of 1.65 mm and a correlation of 0.71 are obtained for the G-Mem method. They are comparable to those of methods recently proposed. [ABSTRACT FROM AUTHOR]

Limited information exists on the practical effects of lateral reflections in small rooms designed for high-quality sound reproduction and critical listening. A pilot study is undertaken to determine what effect specular and diffuse lateral reflections have on a trained listener. A task-based methodology is employed in which a highly trained subject is asked to perform a task commonly encountered in their daily work. The physical conditions of the listening environment are altered to minimize, maximize, and diffuse side-wall reflections. Results correlate the presence of strong lateral energy with an initial reduction of subjects ability to complete the task within normal tolerances, but adaptation soon occurs, restoring the subjects to practically normal pace and accuracy. [ABSTRACT FROM AUTHOR]

American National Standards (ANSI Standards) developed by Accredited Standards Committees S1, S2, S3, and S12 in the areas of acoustics, mechanical vibration and shock, bioacoustics, and noise, respectively, are published by the Acoustical Society of America (ASA). In addition to these standards, ASA publishes catalogs of Acoustical Standards, both National and International. To receive copies of the latest Standards catalogs, please contact Susan B. Blaeser. Comments are welcomed on all material in Acoustical Standards News. This Acoustical Standards News section in JASA, as well as the National and International Catalogs of Acoustical Standards, and other information on the Standards Program of the Acoustical Society of America, are available via the ASA home page: http://acousticalsociety.org. [ABSTRACT FROM AUTHOR]

In this paper, sound insulation performance of a prefabricated cabin mock-up is studied. The cabin mock-up consists of two identical rooms whose partition, ceiling, and side walls are sandwich panels composed of thin metal sheets and mineral wools. The floor is a steel plate with stiffeners, and plate thickness is 6 mm. It is shown that sealing gap between the floor and the partition improves STL (Sound Transmission Loss) significantly above 2000 Hz. Even after all gaps are sealed, the weighted sound reduction index Rw of the partition measured in the mock-up is lower than that from laboratory measurement by 9 dB. To find out how flanking transmission of noise degrades Rw of the partition, we measure structure-borne noise (SBN) of room surfaces. It is found that SBN of the steel floor is the most important factor in flanking transmission of SBN. Although SBNs of sandwich panels in the source room are larger than that of the floor, they decrease rapidly as they propagate to the receiving room, because sandwich panels provide considerably more damping than the steel floor. Consequently, SBN of the floor becomes dominant in the receiving room. Statistical energy analysis is performed, where SBN transmission along the floor is investigated. It is concluded that that the most effective way to reduce flanking transmission of SBN is to increase damping of the floor. [ABSTRACT FROM AUTHOR]

The acoustical performances of regular arrays of cylindrical elements, with their axes aligned and parallel to a ground plane, have been investigated through predictions and laboratory experiments. Semi-analytical predictions based on multiple scattering theory and numerical simulations based on a boundary element formulation have been made. Measurements have been made in an anechoic chamber using arrays of (a) cylindrical acoustically-rigid scatterers (PVC pipes) and (b) thin elastic shells. Insertion loss (IL) spectra due to the arrays have been measured without and with ground planes for several receiver heights. Data and predictions have been compared. The minima in the excess attenuation spectrum i.e., attenuation maxima due to the ground alone resulting from destructive interference between direct and ground-reflected sound waves, tend to have an adverse influence on the band gaps (BG) related to a periodic array in the free field when these two effects coincide. On the other hand, the presence of rigid ground may result in an IL for an array near the ground similar to or, in the case of the first BG, greater than that resulting from a double array, equivalent to the original array plus its ground plane mirror image, in the free field. [ABSTRACT FROM AUTHOR]

To reduce vehicle inner noise caused by road excitation, the road surface is built, Acoustic Transfer Vector (ATV) method and 1/3 octave filter method are used to get A-level sound pressure of the center frequency and total sound pressure felt by drivers for the sound between 20 and 250 Hz beside driver's ears, and so the sound response of human ears to sound loudness can be reflected truly. The excitation forces between suspension and body under asphalt road excitation are simulated by using dynamics software; the body vibration velocity response is calculated by response analysis function of finite element method; taken body vibration velocity response as boundary constrains, the acoustic response in body cabin is calculated by acoustic transfer vector method which is based on boundary element. The obtained relationship between road excitation and vehicle inner noise can provide references for vehicle inner noise control and suspension system optimization. [ABSTRACT FROM AUTHOR]

To study the properties of sound absorption characteristics of porous metallic materials at high temperature conditions, the sound absorption properties of porous materials with temperature changes was studied firstly in theory. And then, a temperature-controlled high-temperature impedance tube measurement device was designed. Different results of different acoustic properties porous metal material were measured at different high temperatures, and the relationships between absorption properties and temperatures were studied. The predicted and measured results are in good agreement basically, and the theoretical analysis and design of measuring device are reasonable. [ABSTRACT FROM AUTHOR]

The article presents several studies related to sound or environmental noise. It states the pilot study on the relationship between room acoustic parameters and sound power level produced by speakers. It states the study on the effects of noise to noise sensitive and insensitive persons while doing mental tasks. Moreover, it presents a study on the use of noise maps to determine the spatial properties of urban environmental noise generated at 345 noise monitoring stations in Tainan, Taiwan.

Despite the current expansion of interest in soundscapes, the roles and application of the concept, vis-à-vis that of environmental noise management, have not been adequately articulated. Soundscapes and environmental noise management are complementary, not competing, approaches with respect to the study and management of the outdoor acoustic environment. Their complementarity is best demonstrated by focusing on fundamental distinctions between them. This paper introduces the notion of the acoustic environment as a resource to be managed for different beneficial uses and for diversity, and with a focus on sounds of human preference. It provides the opportunity for a more nuanced understanding of human response to the acoustic environment. By contrast, environmental noise management focuses on sound as a waste, and on reducing the extent and intensity of its adverse effects on people. Soundscape design, planning and management are a useful augmentation to environmental noise management approaches, potentially expanding the scope of application of the tools of acoustic specialists. The two fields require quite different approaches to measurement. [ABSTRACT FROM AUTHOR]

This article takes critical issue with the well-circulated but often misapplied term "soundscape." Coined by Canadian composer Murray Schafer in his book "The Soundscape," the term has become one of the keywords of sound studies, but in its wide circulation it has become disconnected from its original scholarly concept and used broadly to apply to nearly any sonic phenomenon. Scholars either misapply it or redefine it to suit their needs. This article is an attempt to trace an intellectual history or genealogy of the term, and to open a conversation about the term's use, application, and utility for scholars of sound. This article draws on Schafer's work in an attempt to ground the term in its own intellectual history, and then traces the use of the term in a variety of sound studies works. The term "soundscape" emerges as at once indispensable and elusive, provocative and limited. By calling attention to background sound, Schafer shaped the field in ways that exceeded his own contribution. [ABSTRACT FROM AUTHOR]

Analysis and synthesis processes for the dan tranh, a Vietnamese plucked string instrument, are described. In the analysis part, parameters are extracted from the decay region of recorded tones by using partial decay estimation and short-time Fourier transforms. The parameters of the string model are estimated, and the excitation signals for the model are extracted from recorded tones. The quality of synthesized tones is also studied based on different recorded data, numerous analyzed partials, and amplitude normalization. The results are compared with the Korean gayageum with silk strings and the acoustic guitar with steel strings. The quality of the synthesized sounds is verified by a multiple stimulus with hidden reference and anchor (MUSHRA) test. The average score was about 87, based on 100. [ABSTRACT FROM AUTHOR]

Newly developed software, NoteView, is used to analyse the fundamental frequency (F0) and categorical as well as microtonal pitch from audio recordings of music performances from a single line (monophonic) instrument. The code is based on the SWIPE algorithm developed by Camacho and Harris. The features of the interface are described and a comparison of two performances of a familiar piece played by a professional French horn player is used as an example for the purpose of investigating the pitch stability of the two renditions. Results produced by NoteView indicate that pitch pairs across the two performances differed by a mean of 7 cents, and that within-note standard deviation was typically 6 cents. These results are examined using the various customisable views and statistics returned by the software. Some of the features and limitations of NoteView are discussed. The software is currently implemented in Matlab and is freely available from the UNSW Empirical Musicology web site http://empa.arts.unsw.edu.au/research-and-creative-practice/research-projects/ empirical-musicology/. [ABSTRACT FROM AUTHOR]

Abstract: The propagation of sound in long enclosures with branches has been studied theoretically and experimentally, and an efficient combined method is proposed to predict the sound field in long enclosures with branches. Based on the wave-acoustics theory, the theoretical analysis of the sound field of the long enclosures with branches is performed. This paper also investigated the sound field prediction of long enclosures with branches, by using the acoustic modeling program, ODEON. The results obtained by the theoretical analysis and the numerical simulation ODEON are compared with the experimental measurements, and the characteristics of the two methods for predicting the sound field of long enclosures with branches are analyzed. Compared with the experimental results, it is found that: (1) the results predicted by the theoretical analysis fluctuate relatively large with respect to the source–receiver distance, and the sound pressure level (SPL) attenuation obtained is smaller than that measured; and (2) the results predicted by the numerical simulation is smoother, and the calculated SPL attenuation is larger than that measured. To effectively predict the sound field of long enclosures with branches, a combined numerical method is thus proposed. The effectiveness of the proposed combined method is demonstrated by the scale-model experiments. [Copyright &y& Elsevier]

When a “massless” one-dimensional sound pulse (mass of a sound pulse is defined as an integral of the perturbation of density over the pulse length) reflects from a pressure-release boundary, its momentum changes sign. This obviously violates momentum conservation. However, in contrast to the case of an unbounded medium, calculation of the momentum in a bounded region includes a second-order term as well. Apparently, the second-order correction to the linear solution ensures momentum conservation in this case. The purpose of this Letter is to find a concrete form of this second-order correction. It appears that, as a result of the nonlinear interaction of the pulse with a pressure-release boundary, the latter experiences second-order net shift. This leads to the generation of a massive second-order rarefaction pulse whose momentum is directed opposite to the direction of propagation of the pulse itself. Appearance of this pulse ensures total momentum conservation. [ABSTRACT FROM AUTHOR]

Successful processing of materials by powder sintering relies on the creation of strong interparticle bonds. During certain critical stages of the sintering process, the medium may be modeled as two phases consisting of the particles and a surrounding matrix. Ultrasonic methods have been proposed as a potential tool for monitoring such sintering processes. Thus, an understanding of the propagation and scattering of elastic waves in two-phase solids is of fundamental importance to these monitoring techniques. In this article, expressions for the ultrasonic attenuations are developed based on the spatial statistics of the density and Lamé parameters of the material constituents under assumptions of statistical homogeneity and statistical isotropy. The formulation is based on the solution of the elastodynamic Dyson equation within the limits of the first-order smoothing approximation. Since the geometric two-point correlation function plays a central role in the model, numerical models are developed using Voronoi polycrystals surrounded by a matrix of different material properties. The spatial statistics of the medium are extracted from these models. The results presented suggest new ultrasonic techniques may be developed to extract multiple correlation lengths for such two-phase microstructures. [ABSTRACT FROM AUTHOR]

This paper proposes a computationally efficient method of determining eigenfunctions and eigenvalues of acoustic modes propagating in circular lined ducts with zero or uniform flow. Linings with circumferentially nonuniform impedance, as found in nacelle acoustics, are the focus. The method of solution adapts in two important respects—the presence of flow and the imposition of impedance boundary conditions—the series expansion method first proposed by Pagneux et al. [J. Acoust. Soc. Am. 110, 1307–1314 (2001)] to calculate the eigenvalues of Lamb modes. The inclusion of flow, and a corresponding different method of solution leading to an improved convergence of eigenvalues (O(N-5), N is the truncation of radial basis of expansion), is the important new feature as compared to the previous adaptation by Bi et al. [J. Acoust. Soc. Am. 122, 280–291 (2007)]. As a result, it becomes possible to safely determine and in a simple manner the eigenvalues and eigenfunctions of circumferentially nonuniform lined ducts in the presence of uniform flow, up to relatively high frequencies (e.g., 30

For an arbitrary anisotropic half-space with continuous vertical variation of material properties, an explicit closed-form expression for the coefficient B of high-frequency dispersion of the Rayleigh velocity vR(ω)≈vR(0)(1+B/ω) is derived. The result involves two matrices, one consisting of the surface-traction derivatives in velocity and the other of its Wentzel–Kramers–Brillouin coefficients, which are contracted with an amplitude vector of the Rayleigh wave in the reference homogeneous half-space. The “ingredients” are routinely defined through the fundamental elasticity matrix and its first derivative, both taken at v=vR(0) and referred to the surface. [ABSTRACT FROM AUTHOR]

A coupled pair of nonlinear parabolic equations is derived for the two components of the particle motion perpendicular to the axis of a shear wave beam in an isotropic elastic medium. The equations account for both quadratic and cubic nonlinearity. The present paper investigates, analytically and numerically, effects of cubic nonlinearity in shear wave beams for several polarizations: linear, elliptical, circular, and azimuthal. Comparisons are made with effects of quadratic nonlinearity in compressional wave beams. [ABSTRACT FROM AUTHOR]

The multilevel fast multipole algorithm (MLFMA) is extended to solve for acoustic wave scattering by very large objects with three-dimensional arbitrary shapes. Although the fast multipole method as the prototype of MLFMA was introduced to acoustics early, it has not been used to study acoustic problems with millions of unknowns. In this work, the MLFMA is applied to analyze the acoustic behavior for very large truncated ground with many trenches in order to investigate the approach for mitigating gun blast noise at proving grounds. The implementation of the MLFMA is based on the Nyström method to create matrix equations for the acoustic boundary integral equation. As the Nyström method has a simpler mechanism in the generation of far-interaction terms, which MLFMA acts on, the resulting scheme is more efficient than those based on the method of moments and the boundary element method (BEM). For near-interaction terms, the singular or near-singular integrals are evaluated using a robust technique, which differs from that in BEM. Due to the enhanced efficiency, the MLFMA can rapidly solve acoustic wave scattering problems with more than two million unknowns on workstations without involving parallel algorithms. Numerical examples are used to demonstrate the performance of the MLFMA with report of consumed CPU time and memory usage. [ABSTRACT FROM AUTHOR]

Despite significant recent advances in numerical methodologies for simulating rough-surface acoustic scattering, their applicability has been constrained by the limitations of state-of-the-art computational resources. This has been particularly true in high-frequency applications where the sheer size of the full-wave simulations render them impractical, and engineering processes must therefore rely on asymptotic models [e.g., Kirchhoff approximation (KA)]. However, the demands for high precision can make the latter inappropriate, thus efficient, error-controllable methodologies must be devised. This paper presents a computational strategy that combines the virtues of rigorous solvers (error control) with those of high-frequency asymptotic models (frequency-independent computational costs). These methods are based on high-order “boundary perturbations,” which display high precision and unparalleled efficiency. This is accomplished by incorporating asymptotic phase information to effect a significant decrease in computational effort, simultaneously retaining the full-wave nature of the approach. The developments of this contribution are constrained to configurations that preclude multiple scattering; it is further explained how the schemes can be made applicable to general scattering scenarios, though implementation details are left for future work. Even for single-scattering configurations, the approach presented here gives significant gains in accuracy when compared to asymptotic theories (e.g., KA) with modest additional computational cost. [ABSTRACT FROM AUTHOR]

Most investigators emphasize the importance of detecting the reflected signal from the defect to determine if the pipe wall has any damage and to predict the damage location. However, often the small signal from the defect is hidden behind the other arriving wave modes and signal noise. To overcome the difficulties associated with the identification of the small defect signal in the time history plots, in this paper the time history is analyzed well after the arrival of the first defect signal, and after different wave modes have propagated multiple times through the pipe. It is shown that the defective pipe can be clearly identified by analyzing these late arriving diffuse ultrasonic signals. Multiple reflections and scattering of the propagating wave modes by the defect and pipe ends do not hamper the defect detection capability; on the contrary, it apparently stabilizes the signal and makes it easier to distinguish the defective pipe from the defect-free pipe. This paper also highlights difficulties associated with the interpretation of the recorded time histories due to mode conversion by the defect. The design of electro-magnetic acoustic transducers used to generate and receive the guided waves in the pipe is briefly described in the paper. [ABSTRACT FROM AUTHOR]

Detection of echolocation calls is fundamental to quantitative analysis of bat acoustic signals. Automated methods of detection reduce the subjectivity of hand labeling of calls and speed up the detection process in an accurate and repeatable manner. A model-based detector was initialized using a baseline energy threshold detector, removing the need for hand labels to train the model, and shown to be superior to the baseline detector using synthetic calls in two experiments: (1) an artificial environment and (2) a field playback setting. Synthetic calls using a piecewise exponential frequency modulation function from five hypothetical species were employed to control the signal-to-noise ratio (SNR) in each experiment and to provide an absolute ground truth to judge detector performance. The model-based detector outperformed the baseline detector by 2.5 dB SNR in the artificial environment and 1.5 dB SNR in the field playback setting. Atmospheric absorption was measured for the synthetic calls, and 1.5 dB increased the effective detection radius by between 1 and 7 m depending on species. The results demonstrate that hand labels are not necessary for training detection models and that model-based detectors significantly increase the range of detection for a recording system. [ABSTRACT FROM AUTHOR]

Narrowband noise stimuli were used to derive relative weights for detecting frequency glides in a yes/no procedure. One stimulus set was restricted to the duration of the glide. For the second stimulus set, there were fringe noise bands preceding and following the glide. For both sets, the center frequency of the linear glide was either fixed at 1000 Hz or randomly chosen on each trial from the range of 800–1200 Hz. Relative weights as a function of time were derived using a linear model and the linear classification method [A. Ahumada, J. Vis., 2, 121–131 (2002)]. Sensitivity was better for the fixed- than random-frequency conditions, and weight patterns from the random-frequency conditions were less reliable than those obtained from the fixed-frequency conditions. The magnitudes of the relative weights tended to be larger for the second half of the stimulus, suggesting that subjects paid more attention to the later than the earlier parts of stimuli. In the random-frequency conditions, the linear model failed to account for subjects’ performance unless the stimuli were expressed in terms of relative changes in frequency rather than absolute frequency. [ABSTRACT FROM AUTHOR]

Pathologic changes within the organic constitution of vocal folds or a functional impairment of the larynx may result in disturbed or even irregular vocal fold vibrations. The consequences are perturbations of the acoustic speech signal which are perceived as a hoarse voice. By means of appropriate image processing techniques, the vocal fold dynamics are extracted from digital high-speed videos. This study addresses the approach to obtain a parametric description of the spatio-temporal characteristics of the vocal fold oscillations for the aim of classification. For this purpose a biomechanical vocal fold model is introduced. An automatic optimization procedure is developed for fitting the model dynamics to the observed vocal fold oscillations. Thus, the resulting parameter values represent a specific vibration pattern and serve as an objective quantification measure. Performance and reliability of the optimization procedure are validated with synthetically generated data sets. The high-speed videos of two normal voice subjects and six patients suffering from different voice disorders are processed. The resulting model parameters represent a rough approximation of physiological parameters along the entire vocal folds. [ABSTRACT FROM AUTHOR]

Using microphone arrays for estimating source locations and strengths has become common practice in aeroacoustic applications. The classical delay-and-sum approach suffers from low resolution and high sidelobes and the resulting beamforming maps are difficult to interpret. The deconvolution approach for the mapping of acoustic sources (DAMAS) deconvolution algorithm recovers the actual source levels from the contaminated delay-and-sum results by defining an inverse problem that can be represented as a linear system of equations. In this paper, the deconvolution problem is carried onto the sparse signal representation area and a sparsity constrained deconvolution approach (SC-DAMAS) is presented for solving the DAMAS inverse problem. A sparsity preserving covariance matrix fitting approach (CMF) is also presented to overcome the drawbacks of the DAMAS inverse problem. The proposed algorithms are convex optimization problems. Our simulations show that CMF and SC-DAMAS outperform DAMAS and as the noise in the measurements increases, CMF works better than both DAMAS and SC-DAMAS. It is observed that the proposed algorithms converge faster than DAMAS. A modification to SC-DAMAS is also provided which makes it significantly faster than DAMAS and CMF. For the correlated source case, the CMF-C algorithm is proposed and compared with DAMAS-C. Improvements in performance are obtained similar to the uncorrelated case. [ABSTRACT FROM AUTHOR]

Information transfer analysis [G. A. Miller and P. E. Nicely, J. Acoust. Soc. Am. 27, 338–352 (1955)] is a tool used to measure the extent to which speech features are transmitted to a listener, e.g., duration or formant frequencies for vowels; voicing, place and manner of articulation for consonants. An information transfer of 100% occurs when no confusions arise between phonemes belonging to different feature categories, e.g., between voiced and voiceless consonants. Conversely, an information transfer of 0% occurs when performance is purely random. As asserted by Miller and Nicely, the maximum-likelihood estimate for information transfer is biased to overestimate its true value when the number of stimulus presentations is small. This small-sample bias is examined here for three cases: a model of random performance with pseudorandom data, a data set drawn from Miller and Nicely, and reported data from three studies of speech perception by hearing impaired listeners. The amount of overestimation can be substantial, depending on the number of samples, the size of the confusion matrix analyzed, as well as the manner in which data are partitioned therein. [ABSTRACT FROM AUTHOR]

Which acoustic properties of the speech signal differ between rhythmically prominent syllables and non-prominent ones? A production experiment was conducted to identify these acoustic properties. Subjects read out repetitive text to a metronome, trying to match stressed syllables to its beat. The analysis searched for the function of the speech signal that best predicts the timing of the metronome ticks. The most important factor in this function is found to be the contrast in loudness between a syllable and its neighbors. The prominence of a syllable can be deduced from the specific loudness in an (approximately) 360 ms window centered on the syllable in question relative to an (approximately) 800-ms-wide symmetric window. [ABSTRACT FROM AUTHOR]

This study focuses on correlating speech confusion patterns, defined as consonant-vowel confusion as a function of the speech-to-noise ratio, and a model acoustic feature (AF) representation called the AI gram, defined as the articulation index density in the spectrotemporal domain. By collecting many responses from many talkers and listeners, the AF and psychophysical feature (event) is shown to be correlated via the AI-gram model and the confusion matrices at the utterance level, thereby explaining the listener confusion. Consonant /t/ is used as an example to identify its primary robust-to-noise feature, and a precise correlation of the acoustic information with the listeners’ confusions is used to label the event. The main spectrotemporal cue defining the /t/ event is an across-frequency temporal coincidence, wherein frequency spread and robustness vary across utterances, while the event remains invariant. The cross-frequency timing event is shown to be the key perceptual feature for consonants in a vowel following context. Coincidences are found to form the basic element of the auditory object. Neural circuits used for coincidence in binaural processing for localization across ears are proposed to be used within one ear across channels. It is further concluded that the event is based on the audibility of the /t/ burst rather than on any superthreshold property. [ABSTRACT FROM AUTHOR]

Broadband simulated dolphin echolocation signals were used to measure the ex situ backscatter properties of mesopelagic boundary community (MBC) in order to gain a better understanding of the echolocation process of spinner dolphins foraging on the MBC. Subjects were captured by trawling with a 2-m-opening Isaacs-Kidd Midwater Trawl. Backscatter measurements were conducted on the ship in a 2000 L seawater tank with the transducer placed on the bottom pointed upwards. Backscatter measurements were obtained in both the dorsal and lateral aspects for seven myctophids and only in the dorsal aspect for 16 more myctophids, six shrimps, and three squids. The echoes from the myctophids and shrimps usually had two highlights, one from the surface of the animal nearest the transducer and a second probably from the signal propagating through body of the subject and reflecting off the opposite surface of the animal. The squid echoes consisted mainly of a single highlight but sometimes had a low amplitude secondary highlight. The backscatter results were used to estimate the echolocation detection range for spinner dolphins foraging on the mesopelagic boundary community. The results were also compared with multi-frequency volume backscatter of the mesopelagic boundary community sound scattering layer. [ABSTRACT FROM AUTHOR]

The aim of the study was to establish whether /u/-fronting, a sound change in progress in standard southern British, could be linked synchronically to the fronting effects of a preceding anterior consonant both in speech production and speech perception. For the production study, which consisted of acoustic analyses of isolated monosyllables produced by two different age groups, it was shown for younger speakers that /u/ was phonetically fronted and that the coarticulatory influence of consonants on /u/ was less than in older speakers. For the perception study, responses were elicited from the same subjects to two minimal word-pair continua that differed in the direction of the consonants’ coarticulatory fronting effects on /u/. Consistent with their speech production, young listeners’ /u/ category boundary was shifted toward /i/ and they compensated perceptually less for the fronting effects of the consonants on /u/ than older listeners. The findings support Ohala’s model in which certain sound changes can be linked to the listener’s failure to compensate for coarticulation. The results are also shown to be consistent with episodic models of speech perception in which phonological frequency effects bring about a realignment of the variants of a phonological category in speech production and perception. [ABSTRACT FROM AUTHOR]

The present study systematically manipulated three acoustic cues—fundamental frequency (f0), amplitude envelope, and duration—to investigate their contributions to tonal contrasts in Mandarin. Simplified stimuli with all possible combinations of these three cues were presented for identification to eight normal-hearing listeners, all native speakers of Mandarin from Taiwan. The f0 information was conveyed either by an f0-controlled sawtooth carrier or a modulated noise so as to compare the performance achievable by a clear indication of voice f0 and what is possible with purely temporal coding of f0. Tone recognition performance with explicit f0 was much better than that with any combination of other acoustic cues (consistently greater than 90% correct compared to 33%–65%; chance is 25%). In the absence of explicit f0, the temporal coding of f0 and amplitude envelope both contributed somewhat to tone recognition, while duration had only a marginal effect. Performance based on these secondary cues varied greatly across listeners. These results explain the relatively poor perception of tone in cochlear implant users, given that cochlear implants currently provide only weak cues to f0, so that users must rely upon the purely temporal (and secondary) features for the perception of tone. [ABSTRACT FROM AUTHOR]

In cochlear implants (CIs), different talkers often produce different levels of speech understanding because of the spectrally distorted speech patterns provided by the implant device. A spectral normalization approach was used to transform the spectral characteristics of one talker to those of another talker. In Experiment 1, speech recognition with two talkers was measured in CI users, with and without spectral normalization. Results showed that the spectral normalization algorithm had small but significant effect on performance. In Experiment 2, the effects of spectral normalization were measured in CI users and normal-hearing (NH) subjects; a pitch-stretching technique was used to simulate six talkers with different fundamental frequencies and vocal tract configurations. NH baseline performance was nearly perfect with these pitch-shift transformations. For CI subjects, while there was considerable intersubject variability in performance with the different pitch-shift transformations, spectral normalization significantly improved the intelligibility of these simulated talkers. The results from Experiments 1 and 2 demonstrate that spectral normalization toward more-intelligible talkers significantly improved CI users’ speech understanding with less-intelligible talkers. The results suggest that spectral normalization using optimal reference patterns for individual CI patients may compensate for some of the acoustic variability across talkers. [ABSTRACT FROM AUTHOR]