bluuebunny/crossref_metadata_2025 huggingface dataset:
Multiscale elastic imaging & modeling of musculoskeletal tissues
Authors: Raum Kay , Schrof Susanne , Tiburtius Sara , Grimal Quentin , Gerisch Alf
Published: April 2012

Sophisticated technical materials that are used in everyday life are often inspired by nature. Hard biological tissues, e.g. mineralized tendons, bone and teeth are natural examples of achieving unique combinations and also great variability of stiffness and strength. In order to achieve these goals, bone uses various design concepts, e.g. reinforcing a soft and flexible collagen matrix by stiff, but brittle mineral particles, sandwich compounding of anisotropic (directional) films, weight reduction by directional pores and spongy networks. Although many details of the genetics, biology, pathology and mechanics of bone have been uncovered, we still lack of a detailed understanding of bone structure at the nano- and microscales. Towards this goal, both experimental data of heterogeneous elastic and structural parameters from all length scales (from the centimeter to the nanometer scale) and theoretical models that can simulate the deformation behavior based on these data are required. In this presentation the concept of multi-modal coupled multi-scale assessment of tissue properties (using quantitative ultrasound, synchrotron radiation μCT and vibrational microscopy) and modeling (using various homogenization techniques) will be presented with an emphasis of applications in musculoskeletal research, e.g. bone and cartilage healing.
Noise source identification in high speed jets based on virtual microphone arrays
Authors: Morris Philip , Du Yongle
Published: April 2012

Phased arrays have become a popular experimental technique for noise source identification. These techniques are generally limited in resolution due to the number of available microphones. In addition, noise source models are relatively simple, involving either uncorrelated point sources or simple coherent line sources. The former limitation is not present in numerical simulations, where the number and location of virtual microphones is effectively unlimited. The present paper describes numerical simulations of high speed jet noise using a Detached Eddy Simulation turbulence model. The nozzle, which is included in the simulations, is typical of the geometries found in high performance military aircraft. An Immersed Boundary Method is used simulate the effect of chevrons at the nozzle exit. The far field noise is predicted using solutions of the Ffowcs Williams – Hawkings equation. Phased array results are presented for both a baseline and chevron nozzle and the differences are discussed. In addition, near field virtual arrays are sampled and analyzed to include both radiating and non-radiating components. The implications of the results from the far and near field arrays in terms of noise source characteristics are presented. The results are compared with available experimental observations.
Effect of Riemann flux solver on the accuracy of spectral difference method for CAA problems
Authors: Gao Junhui , Li Xiaodong , Wang Qiqi
Published: April 2012

The spectral difference (SD) method is a new high-order method for unstructured grids proposed recently by Liu et al. (2006). In this paper, a two dimensional computation aeroacoustics (CAA) tool based on SD method is developed. Five Riemann solvers are implemented in the current code, including Roe scheme, advection upstream splitting method (AUSM), flux-vector-splitting scheme, Rusanov scheme, convective upwind and split pressure (CUSP) scheme. A comparison of these Riemann solvers is carried out with three CAA workshop benchmark problems. The relative error of each solver in simulating of entropy, vorticity and acoustic waves is presented. The accuracy of the SD method with each Riemann solver is obtained. It is found that the usually used Rusanov scheme is less accurate than other solvers. AUSM and CUSP schemes are more accurate than others in simulating acoustic waves. Meanwhile, the effect of mesh quality on the accuracy of SD method is investigated. Gaussian distributed random error is superimposed on a base mesh to change the mesh quality. The accuracy of each solver on the skewed mesh is presented and compared with the results on base mesh. It is shown that mesh quality has little effect on the accuracy of SD method if the mesh resolution is sufficient.
Control of edge-scattering noise via permeable surfaces
Authors: Moon Young J. , Bai Ikhyun , Hwang Seungtae
Published: April 2012

The edge-scattering noise generation mechanism is first studied, in line with the existing thoeries of Howe, Amiet, and others. Then the edge-scattering noise is controlled by attempting various permeable edges such as porous surfaces and slitted edges. The basic underlying mechanism of noise reduction is to be understood, examining the three-dimensional scattering of a line-vortex embedded in the laminar boundary layer over the flat plate with the porous and slitted trailing-edges. More realistic investigations will follow by the large-edge simulation (LES) of a turbulent boundary layer over the flat plate, solving the filtered, three-dimensional, compressible Navier-Stokes equations with the six-order compact finite-difference scheme and the four-stage Runge-Kutta method.
A robust selection method of time-delay difference for DOA estimation
Authors: Li Zhiyu , Hou Zhiguo , Li Zhaoli
Published: April 2012

DOA estimation method base on time-delay differences is used widely in the field of passive source location because its solidity in interference circumstance. To meet the real-time demands of some location systems, a time-delay differences robust selection method is presented in this paper by analysis of DOA estimation errors. In this method, only part of time-delay differences is estimated for the measuring of DOA to reduce the computational complexity. Computer simulation results indicate the new method has less computational cost by contrast with traditional method, and based on the reasonable time-delay differences the DOA estimation results are more accurate.
Publishing in the Journal of the Acoustical Society of America
Authors: Zhang Zhaoyan
Published: April 2012

Peer-reviewed journals such as the Journal of the Acoustical Society of America provide researchers the best venue to distribute research accomplishments to a broad readership. While this publishing process may be time-consuming and sometimes frustrating, authors can take an active control of this process by preparing a good-quality manuscript and acquiring a better understanding of the peer-review process. The purpose of this talk is to share the present author's experiences as an JASA author and a current JASA associate editor in preparing a publishable manuscript for JASA. Suggestions are given on how to prepare a scientifically significant manuscript and how to benefit from the peer-review process to further improve the manuscript.
Guidelines for prospective authors to submit acceptable manuscripts to the Journal of the Acoustical Society of America
Authors: Wu Sean F. , Xiang Ning
Published: April 2012

The Editor-In-Chief of the Journal of the Acoustical Society of America (JASA) has recently compiled a list of problems in the manuscripts submitted to JASA that often lead to an outright rejection by the Associate Editors handling their review. These problems often occur during submission and writing of a manuscript. They include selecting a title, listing the authors, composing an abstract, defining the scope of work, presenting the background and significance of the research, reporting and discussing the major discovery, ideas and results, showing the work and results that have been published by others in other journals already, drawing concise conclusions, citing references, displaying equations, figures, tables, etc. Last but not the least is the English writing that should be grammatically correct and easy to understand by someone with a similar background. This talk gives a quick overview of these potential problems, which are frequently shown in the manuscripts submitted by authors overseas. The goal of this talk is to provide helpful suggestions and guidelines to the prospective authors whose native language is not English to submit manuscripts that can pass the initial screening and ultimately get published in JASA. Disguised examples of some problematic manuscripts are discussed and analyzed.
Discussions on publishing excellence in the Journal of the Acoustical Society of America and JASA Express Letters
Authors: Xiang Ning , Cheng Li
Published: April 2012

This workshop involves a number of invited speakers from the editorial board members of the Journal of the Acoustical Society of America (JASA, including JASA Express Letters) and from representative JASA paper authors to discuss the JASA peer-review process, the criteria for accepting manuscripts, and successful preparation of manuscripts for JASA publications. Following the presentations by the invited speakers, a panel discussion offers a platform for those in the audience, particularly potential authors, to ask relevant questions and for the panelists to give multiple replies. The panelists also include a liaison from the American Institute of Physics to the Acoustical Society of America for all publishing affairs. [The panelists: Whitlow Au, Li Cheng, Robert Harington, Allan Pierce, S.K. Tang, Sean Wu Ning Xiang, Zhaoyan Zhang]
Real-time segmentation of the temporal evolution of musical sounds
Authors: Glover John , Lazzarini Victor , Timoney Joseph
Published: April 2012

Since the studies of Helmholtz, it has been known that the temporal evolution of musical sounds plays an important role in our perception of timbre. The accurate temporal segmentation of musical sounds into regions with distinct characteristics is therefore of interest to the study of timbre perception as well as to different forms of sound modelling and manipulation. Following on from recent work by Peeters and Caetano et al, this paper presents a new method for the automatic segmentation of the temporal evolution of isolated musical sounds in real-time. We define attack, sustain and release segments using cues from a combination of the amplitude envelope, the spectro-temporal evolution and a measurement of the stability of the sound that is derived from the onset detection function. We conclude with an evaluation and discussion of some potential applications of the method.
Wind instrument sound design with centroid-controlled spectral template synthesis
Authors: Wun Simon , Horner Andrew , Beauchamp James
Published: April 2012

Most previous sound synthesis research has been oriented to instrument imitation and data reduction, whereas perceptual control has received little attention. In other words, the parameters in traditional sound synthesis are not perceptually meaningful. Using common synthesis techniques such as multiple wavetable synthesis and additive synthesis, we have no obvious way to imitate acoustic tones while allowing perceptual control. An important perceptual parameter is spectral centroid, which strongly correlates with a sound's brightness. Spectral centroid and attack time are two universally-recognized timbral features that strongly influence discrimination and identification of musical instruments. This paper investigates the generality and effectiveness of spectral template synthesis, a perceptually-based technique for synthesizing wind instrument tones. Synthesis from spectral envelope templates is driven by spectral centroid or other control functions. Control by spectral centroid has the advantage of direct manipulation of a perceptually-salient feature. Unlike other synthesis techniques, spectral template synthesis is designed to track changes in spectral centroid and mimic acoustic tones at the same time. This work has application in the synthesis of natural realistic sounds that go beyond the normal timbral boundaries of acoustic instruments. [This work was supported by RGC grants 613510 and 613111.]
Relating timbre discrimination to perceptual distances between interpolated percussive timbres
Authors: Martens William L. , McKinnon-Bassett Mark
Published: April 2012

A set of percussive timbres was generated using a hybrid resynthesis that was based upon the analysis of recorded conga and bongo drums. Comprising the set were drum timbres resulting from parametric variation in both damping of the low-frequency resonance associated with pitch of the drum, and variation in a higher-frequency resonance associated with percussive attack transients. Listeners were presented with all pairwise comparisons of the synthetic drum sounds, and were asked first to perform timbral discriminations for each pair, and subsequently to produce pairwise dissimilarity judgments. Underlying perceptual scales values were derived for each timbre from discrimination performance along the two manipulated stimulus dimensions, and these values predicted well the perceptual distances that were fit to the stimulus space coordinates derived from the dissimilarity judgments. Taken together, the results provide a basis for developing a reliable control structure for the synthesis of such percussive timbres.
A new sinusoidal model for synthesis of musical instruments
Authors: Sharma Sudhendu Raj , Ge Zhenhao , Smith Mark J. T.
Published: April 2012

Many algorithms have been developed over the years to synthesize acoustic sounds and are now used commercially in acoustic synthesizers and digital keyboard products. The issue with these algorithms is the tradeoff among sound fidelity, algorithm complexity, and hardware/storage requirements-the last two of which are directly related to the cost of the system. In this reported work, we introduce a new sinusoidal model for digitally synthesizing musical instruments. The new algorithm has unusually high fidelity, minimal memory requirements, and high computational efficiency. The algorithm is based on the “analysis-by-synthesis overlap add” (ABS/OLA) sinusoidal model, which models musical sounds as a short-time weighted sum of constant frequencies, phases, and amplitudes. The new model we introduce incorporates a novel dynamic pitch and frequency control feature in synthesis that allows very high quality instrument sounds to be generated over a wide range of pitches from a very short sampled recording of the musical instrument. Sound modifications can be performed parametrically within the framework all using fast Fourier transforms (FFTs) for high efficiency. Examples of synthetically generated non-western musical instruments will be presented during the conference and contrasted with competing technologies to illustrate the advantages of the new method.
Performance of active noise barrier with a moving sound source
Authors: Tao Jiancheng , Deng Yiqing , Qiu Xiaojun
Published: April 2012

Active noise control is an effective technology to improve noise reduction performance of traditional passive noise barriers at low-frequencies. Previous researches on active noise barriers are mainly on the assumption that the primary sound source is stationary. Effects of source motion on the performance of an active noise barrier are investigated in this paper. First, an analytical model of a passive barrier with a moving sound source is introduced, which can be used to calculate the primary sound field distribution in time domain. Then, the performance of applying active noise control on such a barrier is investigated numerically. Finally, the experimental results with a practical prototype of active noise barrier will be reported and be compared with the numerical results.
Active control of exhaust noise using an air horn
Authors: Min Dongki , Kim Deokman , Park Junhong
Published: April 2012

The noise generated by internal combustion engine is reduced by passive muffling systems. For the passive muffling system, its performance significantly degrades especially when multiple low frequency tonal components exist in the flow. The tonal components occur from explosion process of the engine, and radiates as a monopole from the outlet. In this study, active noise cancelation using FxLMS algorithm is proposed to reduce the exhaust engine noise. Air-horn which is capable of being operated at high temperatures is proposed for cancelation of the radiated noise. The vibration input the diaphragm of the air-horn allowed the active control of the frequency and phase of the radiated sound. The FxLMS algorithm was used to actively control the sound radiation from the air-horn to achieve cancelation of the noise from the muffler. The sound radiation from the air-horn induces dipole-like noise radiation from the exhaust system, and significantly reduced the radiated sound power.
A broadband active control algorithm without cancellation path modeling
Authors: Gao Min , Qiu Xiaojun
Published: April 2012

Recently, an active control algorithm without cancellation path modeling has been investigated, which does not require identification of the cancellation path. The algorithm adopts the standard LMS to update the adaptive filter coefficients, but unlike the FXLMS algorithm, the reference signal does not need to pass through the cancellation path model and the proper update direction of the adaptive filter coefficients is chosen automatically by monitoring the excess noise power. Simulation and experimental results show that the algorithm works well to sinusoidal noise, multi-tune noise and narrowband noise within 40 Hz bandwidth, with similar noise reduction performance to that of the FXLMS algorithm. However, it is found that the algorithm does not work well for broadband noise and wider narrowband noise. Aiming at this, this paper investigates the mechanism of the algorithm for broadband noise and explores potential solution to the problem.
Why soundscape? The new approach to “measure” quality of life
Authors: Schulte-Fortkamp Brigitte , Voigt Kay
Published: April 2012

It is now about 15 years that Soundscape came into the field of community noise and sound quality. The Soundscape approach has provided essential knowledge for the demanding tasks which are required for the design and planning of sustainable environments to support to wellbeing, health, and quality of life, respectively. The multidimensional Soundscape approach puts emphasis on the way the acoustic environment is perceived, experienced and understood by the individual and by society (ISO/TC 43/SC 1/WG 54). Moreover, it accounts for people's concerns and integrates the exposed people as experts. The process of tuning of noise pollution or sound design with respect to the expertise of people's mind is related to the strategy of triangulation of interdisciplinary data. Moreover, the Soundscape approach provides the frame work to integrate contextual and subjective variables to improve the respective Soundscape with regard to people's expertise. This paper will highlight the process of Soundscape and its application with respect to ISO/TC43/SC 1/WG 54 and the COST network TD0804 on Soundscape and Landscape with regard to its implementation and dissemination in the diverse fields of acoustic environments and its definitive meaning concerning quality of life.
A case study of soundscape design based on acoustical investigation
Authors: Ma Hui , Zhang Sen
Published: April 2012

Jinwan square is an important part of Haihe river, mother river of Tianjin, a Chinese city. In order to create a lively and comfortable environment for Jinwan square, soundscape design is necessary because the acoustical situation in this square is evaluated to be noisy and boring. Based on the sound investigation held in four seasons including sound level, sound type, the relationship between different sounds and sound expectation, a soundscape design focusing on adding natural sounds and controlling both road traffic noise and construction noise was done. Through this study, the process and the method of how to do a soundscape design in certain area were tried to be concluded.
Human-machine interaction as influencing factor of indoor soundscape evaluation
Authors: Steffens Jochen , Schulte-Fortkamp Brigitte , Becker-Schweitzer Joerg
Published: April 2012

In many ways people interact with machines like vehicles or household appliances. Thereby they consciously and unconsciously perceive information about the machines' performance and operating state. Apart from visual or haptic feedback important information is transferred via the acoustical input. In order to design well-assessed indoor soundscapes like car interiors or kitchen it is essential to involve people in early product development processes. However, in order to evoke everyday perception processes in listening studies and to ensure ecological valid sound evaluations realistic human-machine interaction has to be reconstructed. Only under quasi-real conditions the user needs acoustical feedback from the used devices. This determines the user's attitude to the sound and thus to its evaluation to a large extent. Within this contribution case studies will be presented which expound the influence of interaction on cognitive, emotional, and motivational aspects within the sound evaluation in soundscape research.
Cell structure in waves diffracted by a wedge
Authors: Ueda Mitsuhiro
Published: April 2012

Waves diffracted by a wedge made of perfectly reflecting material exhibit characteristic spatial pattern depending on an aperture angle of the wedge. For examples, in the wedge of aperture angle π, that is, a perfectly reflecting plane and that of aperture angle π/2, that is, a corner cube, diffracted waves are identically zero. And in the wedge of aperture angle 2π, that is, a semi-infinite plane, diffracted waves are symmetric with respect to a central axis of the wedge. The relation between the pattern and the aperture angle, however, has not been studied in detail so far since there is no appreciate model for diffracted waves and the rigorous solution for diffracted waves is so complex that only the simplest case can be analyzed. We have proposed the new mathematical model for diffracted waves where they can be expressed as a sum of two more fundamental quantities called elementary diffracted waves. The new model reveals that cell structure exists in waves diffracted by the wedge of aperture angle π multiplied by a rational number less than 2. The cases mentioned above can be explained in terms of the cell structure.
Ultrasonic cleaning of the root canal
Authors: Verhaagen Bram , Boutsioukis Christos , Jiang Lei-Meng , Macedo Ricardo , Walmsley Damien , Van der Sluis Luc , Versluis Michel
Published: April 2012

A crucial step during a root canal treatment is the irrigation, where an antimicrobial fluid is injected into the root canal to eradicate all bacteria from the root canal system. Agitation of the fluid using a miniature file oscillating at 30 kHz has shown a significant improvement in the cleaning efficacy over conventional syringe irrigation. However, the exact cleaning mechanisms, being acoustic streaming, cavitation or an enhanced chemical effect, are not fully understood. Here we investigate ultrasonically activated irrigation through experiments and numerical simulations in order to understand the relative importance of each of the three cleaning mechanisms. We combine high-speed imaging and micro-Particle Imaging Velocimetry to visualize the flow pattern and cavitation in a root canal model (sub-millimeter dimensions), at timescales relevant to the cleaning processes (microseconds). Measurements of the acoustic streaming are coupled to the oscillation characteristics of the file as simulated numerically and measured with a laser vibrometer. Comparison between the streaming pattern inside the root canal and in the free field shows the importance of the confinement of the root canal on the acoustic streaming. The results give new insight into the role of acoustic streaming for the cleaning of root canals.
Acoustic motion processing in auditory cortex
Authors: Lomber Stephen
Published: April 2012

Within extrastriate visual cortex of humans, monkeys and cats, individual cortical areas are specialized for spatial or motion processing. In cat auditory cortex, four regions have been identified to be critical for accurately determining the spatial location of an acoustic stimulus. The purpose of the present investigation was to determine if there is an area in auditory cortex specialized for acoustic motion processing or if areas involved in spatial localization are also critical for acoustic motion processing. Or, is there an acoustic MT? To accomplish this, cats were trained to perform two tasks: a spatial localization task using a static stimulus and a task that required the animals to discriminate leftward from rightward apparent acoustic motion. Focal reversible cooling was used to bilaterally deactivate each of the thirteen areas of cat auditory cortex. Overall, the results show that areas involved in acoustic motion processing are also involved in static spatial localization. An area that is uniquely involved in acoustic motion processing was not identified. These results suggest that spatial localization functions may be a prerequisite for acoustic motion processing in auditory cortex. Supported by the Canadian Institutes of Health Research and the Natural Sciences and Engineering Research Council of Canada.
An analysis of influencing factors for structural damage imaging using Lamb waves
Authors: Zhang Haiyan
Published: April 2012

The use of Lamb waves for the inspection of large plate-like structures and their structural health monitoring (SHM) has been a topic of considerable interest in the development of advanced quantitative nondestructive evaluation techniques. Current interrogation algorithm using Lamb waves can roughly identify the location and severity of potential damages. It is well documented that Lamb waves may have multiple modes at a certain frequency, and accompanied by possible dispersion for each specific mode at different frequencies. In order to improve the accuracy of damage identification, some factors such as frequency, mode and cycle number of excitation signal must be taken into account. This paper presents a probability-based imaging approach for evaluating through-thickness hole damage in an aluminium plate. This method predicts the damage in terms of the probability of its occurrence at a certain spatial position of the structure. The influences of various factors on imaging results are analyzed. The results demonstrate that the imaging algorithm can be used to identify the location and severity of damage, but the identifying accuracy is highly related to frequency, mode and cycle number of exciting Lamb wave signal.
Reduction of computation time using Graphics Processing Unit for the detection of a crack in a large scale concrete structure
Authors: Katsurakawa Yuhei , Fujioka Toyota , Nagata Yoshifumi , Abe Masato
Published: April 2012

This paper describe a method for estimating a crack position in a concrete structure using several accelerometers. An array of accelerometers is installed to the concrete structure and a low frequency vibration is made with a small impulse hammer. A reflection wave is generated from the crack position if a crack exists. Because the concrete structure is elastic, it has three wave-propagation modes. It is difficult to estimate the position precisely because the power of necessary primary-wave mode is weaker than that of surface-wave mode. To estimate the crack position precisely, we have proposed a method for eliminating the unwanted surface-wave and side-wall reflections, in which five parameters are used to estimate an unwanted surface-wave or a side-wall reflection by least mean square technique. Since it takes, however, a very long time to estimate a single unwanted wave, the method did not work if two waves overlap with each other. Therefore, we propose the parallelization of genetic algorithm on GPU using CUDA. As a result, the processing time was shortened dramatically compared to conventional one, and we could distinguish two waves reflected from two close boundaries of a caisson, a huge concrete structure which is used as a breakwater.
Restoring clipped speech signal based on spectral transformation of each frequency band
Authors: Hayakawa Makoto , Morise Masanori , Nishiura Takanobu
Published: April 2012

In recent years, high-quality speech recording is requested for comfortably using a communication system. However, a clipping distortion caused by input exceeding the maximum range of amplifier is one of the problems with a sound quality degradation. Although linear prediction method has been conventionally proposed for restoring a clipped speech signal, it has a problem that the frequently clipped speech signal degrades the restoration performance by increasing the prediction errors. In this paper, we propose a method for restoring a clipped speech signal based on a spectral transformation of each frequency band. In this method, the spectral envelope of target speech signal in each frame is approximated to the spectral envelope of original speech signal to remove the influence of a clipping distortion. In particular, the spectral envelope in higher frequency domain including a static characteristic of the speaker is replaced with the spectral envelope of the unclipped speech signal prepared in advance. Then, the spectral envelope in lower frequency domain including a characteristic of phoneme is approximated with Gaussian Mixture Models. We carried out an evaluation experiment for sound quality of speech signal processed by the proposed method. As a result, we confirmed the effectiveness of the proposed method.
A study on the technique using fractal dimension in the selection of the kind of sound
Authors: Muto Kenji , Shibayama Hideo , Makabe Yoshiaki , Asai Kikuo , Kondo Kimio
Published: April 2012

There is a paper which described that the selection of the sound which has a different fractal dimension is possible by aural. The fractal dimension of each short time window of conversation sound was analyzed, referring to the paper. In this paper, we showed about the technique using fractal dimension in the selection of the kind of sound. The analyzed sound was a conversation sound with which the engine sound and the chirp sound mix. The value of fractal dimension in the case of the chirp sound of bird or the engine sound indicated a value different from the fractal dimension in the case of the conversation sound. We thought that our technique has shown the characteristic of sound source by one parameter. It is possible to use the fractal dimension to judge a mixing surrounding sound in the teleconference at the direction of the speaker where the voice was used. To transmit a clear voice in the teleconference, the voice is utilized to the estimation in the direction of the speaker. In the real system, the case that the voice mixes with surrounding sound, it is distinguishable by fractal dimension.
Reproduction of human-phonatory radiation characteristics with a polyhedron loudspeaker
Authors: Yoshimoto Naoki , Morise Masanori , Nishiura Takanobu
Published: April 2012

Spoken dialogue systems have been studied for car navigation systems and voice search systems. For evaluation, a loudspeaker is used instead of a human because these systems require various kinds of speech samples. However, the sounds radiated by loudspeaker can not reproduce human-phonatory radiation characteristics. Therefore, the mouth simulator is utilized to reproduce human-phonatory radiation characteristics. Although it is based on the average mouth shape, shapes of mouth are different among phonemes. Therefore, due to the hardware structure, it can not accurately reproduce various human-phonatory radiation characteristics affected by shapes of mouth. In this study, we developed a polyhedron loudspeaker to solve this problem. It consists of eleven loudspeakers which are independently controlled. Controlling eleven loudspeakers makes it possible to reproduce desired radiation characteristics. Besides, we try to reproduce human-phonatory radiation characteristics of each Japanese five vowel with an adaptive algorithm based on the MINT (Multi-input/output INverse Theorem). We carried out an experiment to verify the effectiveness of the proposed method. As a result, it was confirmed that phonatory radiation characteristics of Japanese five vowels could be accurately approximated compared with the mouth simulator.
Aircraft flight parameter estimation via multipath delays using ground-based microphone array
Authors: Xie Wei , Guan Luyang , Bao Ming , Li Xiaodong
Published: April 2012

The acoustic signal from a low flying aircraft received by ground-based microphone array is characterized by the interference fringe pattern which is caused by reflection of the ground. In this paper, a model is developed to estimate the parameters of an aircraft's motion with the assumption that the aircraft flies in a straight line at a constant height. The model estimate the speed and height of the aircraft based on the different time of arrival due to the multipath delay at different locations. And the short-time cepstrum method is adopted to estimate the multipath delay accurately. To evaluate the performance of the model, the experimental results and error analysis are presented.
Digital communication system using beamsteering for difference frequency in a parametric array
Authors: Lee Chong Hyun , Lee Jaeil , Bae Jinho , Paeng Dong-Guk , Lee Seung Wook , Shin Jungchae , Jung Jin Woo
Published: April 2012

Digital acoustic signal processing can be applied to sonar and acoustic communications. Especially, transmitting acoustic signal to the desired direction has many applications in military and industry fields. In this paper, we present a steerable digital communication system using parametric array transducers. To evaluate the proposed system, we build digital communication system by using transducer array, power amplifier and Labview software. The Labview software is composed of two parts. The first part is designed to generate beam to the desired direction by changing parameters such as number of sensors, complex weight of each sensor, type of transmit data and etc. The second part is to generate modulated signal of ASK, FSK and PSK, and to demodulate the received signals. With laboratory experiments, we verify the performance of the proposed communication system. Experimental results show that the system can be used to mitigate multipath effect in shallow water and can achieve high data-rate transmission
Loudspeaker compensation using mixed phase technology
Authors: Ye Chao , Wu Ming , Wu Shuaibing , Yang Jun
Published: April 2012

In a sound reproduction system loudspeakers always introduce distortions. The compensation of loudspeaker responses using digital signal technology is becoming an important part of improving sound reproduction quality. Several FIR and IIR filter design methods have been proposed to equalize the response of loudspeaker systems. However, high order filters are needed to obtain excellent resolution at low frequencies. From a psychoacoustic point of view, warped filters have been employed to improve the resolution at low frequencies, but at the expense of poor resolution at high frequencies. On the other hand, the use of warped filters increases the complexity of the filter structure. In this paper, a mixed phase technology is proposed to equalize the response of loudspeaker systems. Loudspeaker response can be decomposed into minimum and excess phase components, which are inverted respectively to construct the equalization filter. An all pass filter based on position-independent excess phase zeros is cascaded with a minimum phase filter to improve the phase response in the region. The experimental results demonstrate that the time-domain response of the loudspeaker systems is improved by using the presented mixed phase technology.
Estimation of demodulation ratio for the parametric loudspeaker based on spectral envelope
Authors: Ikefuji Daisuke , Morise Masanori , Nishiura Takanobu
Published: April 2012

A parametric loudspeaker with the higher directivity transmits the sound wave to only particular area. It emits an AM (Amplitude-Modulated) wave demodulating into an audible sound wave by nonlinear interaction in the air. Therefore, longer distance is required for fully demodulating the AM wave into the audible sound wave. On the other hand, a power of the audible sound wave decays depending on the distance. Thus, the parametric loudspeaker must be utilized on the suitable distance for reproducing the fully demodulated wave with an enough power. However, no criterion has been conventionally proposed to measure the demodulation ratio in the past. Accordingly, the criterion should be formulated to measure demodulation ratio for appropriately utilizing the parametric loudspeaker. Thus in this paper, we aim at formulating it for estimation of demodulation ratio. We therefore propose the criterion based on a spectral envelope of the reproduced TSP (Time-Stretched-Pulse) with parametric loudspeaker. The proposed criterion is defined based on the correlations between the spectral envelope in every distance and that in the target with the reproduced TSP. As a result of the objective experiment, we confirmed the availability of the proposed criterion.
The pattern of echoes in feature space
Authors: Li Xiukun , Xia Zhi
Published: April 2012

In order to increase the performance of recognition in underwater bottom object detection, the distribution pattern of different kinds of echoes is researched in this paper. The past literatures more focus on the characteristics of object echoes in feature space, but the research of joint distribution pattern of object echoes, reverberation and other interference is few. In this paper, reverberation and fake object echoes are taken as the mainly interference for recognizing object echoes, and they are assumed have steady characteristic in time-frequency feature space, respectively. There are two problems are discussed: the separability between different kinds of echoes in feature space, and the stability of them. To deal with these two problems, feature compression and cluster analysis are adopted. And the feature space is generated by FDWT. The data acquired from a lake experiment is processed in this paper, and the processing results prove that object echoes, fake object echoes and reverberation within limited time scale have steady distribution pattern in FDWT feature space.
Diversity combining for long-range acoustic communication in deep water using a towed array
Authors: Song Hee-Chun , Hodgkiss William
Published: April 2012

A recent experiment showed that coherent long-range acoustic communication is feasible in deep water over a ~550-km range between a source towed slowly at 75-m depth and a horizontal line array towed at 3.5 knots at 200-m depth. This paper further demonstrates that diversity combining mitigates channel fading and increases the output SNR. Using sparse channel-estimate-based equalization, three transmissions are combined successfully to decode a 40-Hz bandwidth (230-270 Hz) 8-PSK (phase-shift-keying) communication signal, achieving an effective data rate of 17 bits/s at ~550 km range.
Robust adaptive beamforming based on convex optimization
Authors: Guo Lianghao , Guo Xin , Ge Feng-Xiang
Published: April 2012

Adaptive beamforming is an efficient way of spatial filtering in the presence of interference and noise. However, the conventional adaptive beamforming, e.g., MVDR, may degrade significantly due to the poorly estimated covariance matrix or steering vector errors. Convex optimization has now emerged as a major signal processing tool and made a significant impact on numerous problems because of its foundational nature and potential ability in signal processing. Thus in this paper, several robust adaptive beamforming algorithms based on the convex optimization are presented and evaluated, where a novel mathematical tool called “Yalmip” is used to solve the second-order cone problem in these algorithms. Numerical simulations and comparison show that these robust adaptive beamforming algorithms still approach robust to the above-mentioned problems.
High-resolution direction finding without coherent signals number based on multibeam system
Authors: Zhuo Jie , Li Bing
Published: April 2012

Multibeam acoustic imaging systems are widely utilized for both large- and small-scale underwater investigations, especially for detecting underwater target. Processing backscattered echoes from the targets, the system can plot an acoustic image of target, and then estimate the parameters, such as direction and range. The backscattered echoes of target are coherent signals, and can be modeled as multiple highlights. Usually, the distance between the highlights is close, and the number of the highlights is unknown. Then, the directions cannot be directly estimated due to these highlights are not distinguished in the acoustic image. In this paper, an high-resolution method is proposed for estimating the directions of arrival (DOAs) of coherent highlight signals without signals number. The multibeam underwater acoustic imaging technology and the beamspace MUSIC algorithm are combined together. In this method, eigen-decomposition is skipped, so that coherent signals' DOA can be estimated by high-resolution MUSIC method under the situations with unknown signals number. Due to estimating the DOAs of coherent signals in beamspace, the precision of DOA estimation is enhanced, and the computation complexity is less than the element-space MUSIC method. The method's feasibility and robustness are analyzed and verified by the simulation data.
Generalized array invariant and its application on broadband source ranging
Authors: Shang Qi Chun , Zhang Shuang , Wang Ning
Published: April 2012

A passive source ranging method based on the array invariant in shallow water is discussed in this paper. The arriving time and elevation angle of sound are used to describe the multi-modal propagation and single-mode dispersion. Based on the two parameters, we rederive the array invariant in a different way, which allows simple physical interpretation. The array invariant in its original form is not exact invariant, but depends (weakly) on mode number and frequency. The accuracy of the method based on this notion is limited when the sound speed of seabed is different significantly from that in the water column. A modified technique ( generalized array invariant) is proposed in this talk to improve the problem provided when the sound speed of seabed is known. The proposed method is testified by simulation and experimental data.
Impulse signal reconstruction using bi-receiver data
Authors: Wang Haozhong , Wang Ning , Gao Dazhi
Published: April 2012

Signal reconstruction is used widely in target identification and communication in underwater. A novel method for impulse signal reconstruction using the observed data of two receivers that are arranged in the same depth with a certain horizontal interval, is proposed in this talk. This method needs no a prior environmental information but the ranges between the source and the receivers. Although the Green's function depends on the range, frequency and on the source/receiver depth, the spectrum of signal is only dependent on the frequency. The waveguide invariant notion of shallow water provides a compensation mechanism between the frequency and range shift. According to this mechanism, the amplitudes and phases of Green function spectral ingredients can be extracted respectively. The impulse signal is then obtained by employing the deconvolution. The method is applied to the signal reconstruction of a high S/N ratio real data, the correlation coefficient between the reconstructed and original signals is over 0.95.
Syllables intelligibility in relation to the autocorrelation and cepstrum model: the case of Chinese in Taiwan
Authors: Chen Chiung Yao
Published: April 2012

The articulation of some special pronunciations of vowel inconsistently raises with the rapid speech transmission index (RASTI) tested using monosyllables for Chinese phonics. In the researching of speech intelligibility in room, the factors being considered have to include not only the qualities of sound field but the pronunciation characteristics of syllables as well. Therefore, with regard to the defect of RASTI measurements, we utilized the autocorrelation and the cepstrum of monosyllables recorded in rooms to compare the articulation with the physical phenomena of intelligibilities. Thus, we found that the minimum effective duration of autocorrelation function (τ) of all testing syllables signals recorded in rooms, associated well with the articulation for each individual room. Specially, for individual syllable signals, they are further shown significant correlation between the cepstral energy of monosyllables and the articulation collected in all rooms. The cepstrum acts as a spatially intelligible detector for syllables, and the autocorrelation is a good response of the pronunciation characteristics of syllables.
Binaural room acoustics II: Distributions and consequences of interaural differences
Authors: Hartmann William M. , Rakerd Brad , Macaulay Eric J.
Published: April 2012

Binaural room acoustics attempts to generalize the acoustical properties of rooms as they appear at the two ears of a listener. Because of the importance of sound localization, interest has focused on interaural differences in intensity level and phase. Distributions of these interaural differences, as measured with an artificial head in different rooms, are in reasonable agreement with spherical-head computations having either the direct-to-reverberant ratio or the reverberation time as the main parameter. The psychological relevance of these distributions was tested in experiments using rooms with different reverberation times where listeners were required to report the source azimuth for steady-state pure tones having frequencies between 200 and 1200 Hz. Simultaneously probe microphones in the ear canals recorded interaural differences. Interest centered on the choices made by listeners between plausible and implausible interaural differences resulting from the sound fields in the room. Particular attention was given to measurements made near the binaural critical distance [Hartmann and Rakerd, J. Acoust. Soc. Am. 130 2352 (2011). [Work supported by the AFOSR, grant 11NO002]
Correlation of ultrasonic backscatter parameters with transmission parameters and BMD in cancellous bone
Authors: Han Haijie , Ta Dean
Published: April 2012

Quantitative ultrasound (QUS) has been suggested to have a performance equal to dual-energy X-ray absorptiometry (DXA) for the assessment of bone. In this paper, human cancellous bone is investigated in vivo using QUS and DXA. The ultrasonic backscatter method and its parameters (spectral centroid shift (SCS), apparent integrated backscatter (AIB)) are also introduced. The experimental ultrasonic backscatter signals are collected from 300 volunteers' calcanea in two hospitals, and the two parameters are calculated. Finally, correlation between ultrasonic backscatter parameters and transmission parameters (speed of sound (SOS), broadband ultrasonic attenuation (BUA) and stiffness index (SI)), as well as correlation between backscatter parameters and bone material density (BMD) are analyzed. The results showed that correlations between backscatter parameters and SI are better than that of SOS and BUA. SI correlates positively with SCS (r=0.613, p<0.05), as well with AIB (r=0.463, p<0.05). According to the size of SCS and AIB of ultrasonic backscattered signals, the status of cancellous bone may be assessed.