bluuebunny/crossref_metadata_2025 huggingface dataset:
The furniture industry needs a new evaluation standard for evaluation of sound absorption
Authors: Hagberg Klas G. , Bard Delphine
Published: May 2013

Since decades the standard ISO 11654 are prevailing for evaluating sound absorption of products. The standard is developed and fully adapted to ceiling manufacturer, in particular mineral wool ceiling manufacturer. However, the standard is used independently of which interior product it is applied to, casing a lot of “misuse” and confusion amongst many manufacturer. In particular, when it comes to evaluation of various types of office screens, the ISO 11654 becomes a problem. There is no need, and probably not even possible, to calculate absorption factor for an office screen correctly. Therefore, Sweden decided to develop a new standard, SS 25269—“Acoustics—Evaluation of sound absorption of single objects.” Hence office screens should be treated as single objects to cover a wide range of variety. The standard specifies an evaluation method of the sound absorption using only sound absorption area for each object tested. It will simplify the evaluation and minimize risk for errors since there is no need to state the product area in order calculate absorption coefficient. Furthermore, the area does not have to be stated yet again when performing calculation of room characteristics when using the same products in the finished room.
The impact of acoustic oceanographic methods on estuarine dynamics research
Authors: Geyer W. Rockwell , Traykovski Peter , Lavery Andone
Published: May 2013

Estuaries present unique challenges for observational oceanographers, due to their intense spatial gradients and unrelenting temporal variability. The influence of spatial and temporal variation of estuarine structure and flow on the time-averaged regime is the most important research problem in estuarine physical oceanography. Acoustic methods have played an essential role in revealing this spatial and temporal variability, and new advances in acoustic methods are continuing to provide the most important advances in observations of estuarine processes. The measurement of acoustic backscatter has been a mainstay of estuarine physical oceanography, first for providing qualitative images of the density structure, then for quantifying suspended sediment distributions, and most recently for quantifying the intensity of stratified turbulence. Improved resolution of new systems is revealing the internal structure of shear instability and the mechanics of the transition to turbulence. Acoustic Doppler techniques are so routine now as to be taken for granted, but their impact on the field cannot be overstated, and the new advances in pulse-coherent velocity profiling are continuing this revolution in acoustical oceanography. Acoustic propagation in estuaries has not yet received much attention, but its importance to the operation of autonomous vehicles and long-term monitoring should bring this challenging acoustics problem to the forefront.
The spatial properties of breaking wave generated and bedload transport generated noise in the sediment layer of a shallow water wave guide
Authors: Barclay David R. , Zedel Len , Hay Alex E. , Hatcher Matthew G.
Published: May 2013

In May of 2012, three weeks of ambient noise measurements from a hydrophone buried 30 cm deep in the sediment were recorded at Advocate Beach, a 1:10 sloped beach at the head of the Bay of Fundy, Nova Scotia. While tides varied the mean water depth between 0 and 4 m, 0.8 m surface waves passed overhead, driving sediment bedload transport and creating an ambient noise field in the sediment consisting of two primary components: noise generated by bubbles formed in breaking waves at the surface and noise generated by the collisions of sand, gravel, and cobble in the bedload transport along the seabed. Both of these noise sources are stochastic and can be described by their second order statistics: power spectral density, spatial coherence, and directional density. In an effort to distinguish these two noise sources, the spatial properties of three full wave models of the noise field in the sediment are compared, using an infinite sheet of sources placed near the surface of a Pekeris waveguide to model breaking wave noise, near the fluid–fluid interface of a Pekeris waveguide to model bedload transport noise, or near the fluid–fluid interface of two infinite half-spaces to model bedload transport noise. Using integral transforms to solve the wave equation, each noise model is shown to be spatially inhomogeneous with a unique depth dependent intensity and coherence.
Acoustic measurements of the spatial distribution of suspended sediment at three sites on the Lower Mekong River
Authors: Moore Stephanie A. , Dramais Guillaume , Dussouillez Philippe , Le Coz Jerome , Rennie Colin , Camenen Benoit
Published: May 2013

The Mekong River spans thousands of kilometers, flows through six countries, and its basin is one of the world’s richest in terms of biodiversity. However, land-use changes, dredging of the river bed, and the construction of dams are changing its sediment dynamics and morphology. The resultant increases in bank erosion and reduction in sediment supply to floodplains may have adverse effects on the economical and biological productivity of the region. In order to monitor these changes, the current conditions must be well understood. Comprehensive measurements of the spatial distribution of sediment (both suspended and bed load) were made at three locations in different physiographic regions of the Lower Mekong at the end of the 2012 rainy season. Acoustic Doppler Current Profilers and a multifrequency acoustic backscatter system, the AQUAscat, were used in combination with water sampling to provide high resolution measurements of concentration and grain size. The AQUAscat consisted of four monostatic transducers operating at 0.5, 1, 2.5, and 4 MHz. At each site, it was deployed horizontally at five across-stream positions and 5–10 depths per vertical; a 10 m profile was recorded at each point. The spatial distribution of particle size and concentration were determined using multifrequency inversions of (1) backscattered intensity and (2) attenuation calculated from the intensity profiles. This data set provides a baseline to which to compare future measurements at these sites.
Acoustic propagation characteristics of the estuarine salt wedge
Authors: Reeder D. Benjamin
Published: May 2013

The estuarine environment often hosts a salt wedge—a layer of denser seawater advected by the rising tide under fresh water discharged by the river. The nature of the stratification is a function of the tide's range and speed of advance, river discharge volumetric flow rate, and river mouth morphology. The competing effects of temperature and salinity on sound speed present the question: Is the salt wedge acoustically observable? Using temperature and salinity profiles collected in situ, numerical results show that the salt wedge can impact acoustic propagation. Acoustically, this environment consists of two isospeed layers separated by a thin gradient. While this three-layer very shallow water acoustic waveguide is typically dominated by high angle multipath propagation, refraction occurring in the gradient layer allows some low-angle energy from near-surface sources to be trapped in the upper layer. Acoustic fluctuations observed at an upstream or downstream receiver depend upon the interaction between the advancing and receding tide and the river discharge, which can include the presence of internal waves propagating along the top of the salt wedge.
Quantification of stratified turbulence using acoustic propagation and broadband scattering techniques
Authors: Lavery Andone C. , Geyer Wayne R. , Scully Malcolm E.
Published: May 2013

Narrowband acoustical backscattering techniques have been used for decades as a tool for remote imaging of small-scale physical processes in energetic coastal environments, such as oceanic internal waves and microstructure, on spatial and temporal scales difficult to probe with in situ measurements. However, to date, it has been challenging to infer quantitative information about turbulent intensity from the measured backscatter, in part due to uncertainty in the sources of scattering. In contrast to narrowband techniques, emerging broadband techniques result in increased spectral classification and quantification capabilities. Broadband backscattering collected in the Connecticut River Estuary in 2009 in concert with in situ measurements of turbulence have illustrated the potential of these techniques for quantitative remote-sensing of microstructure intensity over relevant spatial and temporal scales. These measurements have resulted in remote quantification of finescale variability of turbulent mixing as well as examination of the mechanisms and structure of shear instability across a broad range of stratification and shear conditions. Upcoming acoustic propagation measurements directed at measuring acoustic scintillation in the Connecticut River Estuary in the presence of strongly stratified turbulence and shear instabilities, which allow remote-sensing of the path-averaged statistical structure and motion of the intervening flow, will also be discussed.
Investigation of low-frequency acoustic tissue properties of seagrass
Authors: Enenstein Gregory R. , Dolder Craig N. , Wilson Preston S. , Hermand Jean-Pierre
Published: May 2013

Understanding the acoustic properties of seagrass is important for applications in mine hunting, shallow water sonar performance, and acoustic remote sensing for ecological surveys. Previous laboratory and field investigations have shown that the plant biomass and tissue structure of seagrass, rather than just the overall gas content, play a determinant role in its acoustic behavior. Hence, effective medium models of propagation through seagrass meadows have been ruled out, and a complete description of both tissue structure and tissue elastic properties is required to describe the acoustic response of seagrass meadows. To begin to address these deficiencies, a resonance tube experiment was set up to determine the low-frequency acoustic response of multiple species of seagrass in relation to leaf biomass and tissue acoustic compliance independent of tissue structure. Responses to frequency-modulated signals in the range from 0.5 to 10 kHz were obtained for Thalassia testudinum (turtle grass) and Halodule wrightii (shoal grass), two species with well-differentiated morphological features. An elastic waveguide model was used to account for the minor effect of the tube walls on the resonance characteristics. Initial measurements of tissue compliance will be presented. [Work supported by ONR and ARL:UT.]
Three-dimensional nonlinear inverse scattering: Quantitative transmission algorithms, refraction corrected reflection, scanner design, and clinical results
Authors: Wiskin James , Borup David , Andre Michael , Johnson Steven , Greenleaf James , Parisky Yuri , Klock John
Published: May 2013

Research in quantitative whole breast ultrasound imaging has been developing rapidly. Recently, we published results from 2D transmission inverse scattering algorithms, based on optimization, incorporating diffraction, refraction, and limited multiple scattering effects, using data collected from an early prototype, which showed the feasibility of high resolution quantitative imaging of the breast tissue speed and attenuation, and concomitant refraction corrected reflection imaging. However, artifact problems in speed and attenuation result from the 2D algorithms and the data characteristics. The reflection algorithm uses the speed map to model refractive effects of rays, so these artifacts are unacceptable. The 3D inverse scattering algorithm presented here, using data from a new prototype, overcomes most of these artifacts. We then use a 3D refraction corrected 360° compounded reflection algorithm for high resolution speckle free reflection images. We discuss the transmission and reflection algorithms and the advanced scanner used to collect the data, as well as initial clinical results from the Mayo Clinic, Breast Cancer Imaging Center, Orange County, and the University California, San Diego. We show examples of cysts, fibroadenomas, calcifications, cancers, and DCIS, in dense, fatty, and average breast tissue, and compare these with hand-held ultrasound, MRI, and mammography, where available.
Approximation error method for full-wave tomography
Authors: Koponen Janne , Huttunen Tomi , Tarvainen Tanja , Kaipio Jari
Published: May 2013

In ultrasound tomography (UT), the speed of sound (SOS) is reconstructed based on ultrasound measurements made on the surface of the object. As a part of the reconstruction process, which includes the solution of the inverse problem, propagation of acoustic signals in the medium is simulated using a forward model. Consequently, modeling errors can generate artifacts into reconstructed SOS. Accurate full-wave models can be computationally heavy and thus impractical in many real applications. On the other hand, approximate models typically lead to less accurate reconstructions. In this study, measurement noise and modeling errors of UT are modeled in Bayesian framework, and a numerical method that takes both errors into account is developed. The performance of the method is investigated by numerical simulations in which artifacts generated by a fast but less accurate forward model and approximate boundary conditions are compensated.
Electromagnetic hydrophone for high-intensity focused ultrasound measurement
Authors: Grasland-Mongrain Pol , Mari Jean-Martial , Gilles Bruno , Lafon Cyril
Published: May 2013

An ultrasonic hydrophone based on the Lorentz force is introduced. When a metallic wire is moved by ultrasound while submitted to a magnetic field, the Lorentz force induces an electrical current proportional to the integral of pressure along the wire. 2D pressure field mapping is achieved by performing a tomography through wire translations and rotations in the imaging plane. Performances of this hydrophone are assessed in this study. Signal is linear over pressure from 10 kPa to at least 10 MPa with a determination coefficient R2 above 0.997. Excellent resistance to cavitation has been observed. Frequency bandwidth was measured against three different wire diameters: 70 μm, 100 μm, and 210 μm. Results showed that upper cut-off frequency decreases with increasing wire diameter. Additional measurements showed that wire tension has no visible effect on the signal. Such characteristics are potentially of great interest for high-intensity focused ultrasound and shockwave transducers calibration.
Computational fluid dynamics simulation of Rayleigh streaming in a vibrating resonator
Authors: Oosterhuis Joris P. , Bühler Simon , Wilcox Douglas , van der Meer Theo H.
Published: May 2013

Rayleigh streaming is a time-averaged flow that can exist in the thermal buffer tubes of thermoacoustic prime movers and refrigerators and is driven by the viscous stresses close to the solid boundaries. This mean flow leads to mean convective heat transport, which can have large impact on the performance of thermoacoustic devices. Rayleigh streaming in a standing wave resonator is simulated using a commercially available computational fluid dynamics (CFD) code and is compared to existing analytical models of Hamilton et al. (2003). A test case is developed, and a standing wave is generated by applying a harmonic volume force to the domain. Both the inner and outer streaming vortices are well described for a range of radii from R/δν = 3…20 and the magnitude of the streaming velocity matches analytical values. This paper shows the possibility of using available as-is CFD software for the simulation of streaming in a standing wave resonator. The presented results pave the way for the simulation of more complex geometries and studies to reduce the negative effects Rayleigh streaming can have on thermo-acoustic prime mover and refrigerator efficiency.
Properties of violin glides in the performance of cadential and noncadential sequences in solo works by Bach
Authors: Liu Jiaxi
Published: May 2013

This study examines the articulatory changes (“glides”) between the leading tone and tonic of cadential vs noncadential semitone sequences in solo violin performance. It was predicted that though these glides would have similar slopes, they would differ in duration and in semitone intonation, and that these latter properties could characterize the expression of cadential finality and the structural insignificance of noncadential sequences. Cadential (46) and noncadential (58) targets from 17 recordings by 13 professional violinists were analyzed using narrow-band spectrograms. Glide durations comprised 16% of the overall duration of semitone sequence irrespective of structure function. However, cadential glides comprised 28% of the duration of the leading tone compared with 11% for noncadential glides. As predicted, the leading tone tended to be sharp in both contexts, but the mean cadential interval was nonsignificantly larger by 18 cents, mainly because the tonic tended to be tuned more accurately in cadential sequences. Finally, the glide direction was linear and followed the natural vibrato trajectory in both contexts as expected. These data confirm that articulatory modifications play a prominent role in the performance of intended musical structure and suggest that such distinctions will influence structural expectancies.
Attenuation as a function of the canal length of custom-molded earplugs
Authors: Tufts Jennifer B. , Chen Siyuan , Marshall Lynne
Published: May 2013

The fit of a custom-molded earplug (CMEP) and the amount of attenuation it provides can be affected by variables related to the original earmold impression and the subsequent manufacturing process. One variable thought to affect the amount of attenuation is the length of the canal portion of the CMEP. In this pilot study, we systematically examined the relationship between CMEP canal length and attenuation in four human subjects. Two men and two women were fitted with CMEPs extending past the second bend of the ear canal. The attenuation provided by the CMEPs was measured over four visits to the laboratory. Prior to each visit, the canal portion of the subject’s test CMEP was shortened by 2 mm. As expected, attenuation decreased as canal length decreased for all subjects. However, the rate and pattern of decrease varied markedly. Anecdotal reports of comfort as a function of canal length also varied markedly. Results suggest that the critical region/s in the ear canal for maintaining a good acoustic seal may vary from person to person. Implications for future study will be discussed.
Implementation of a simplified, artificial external ear test fixture for measurement of the earplug induced auditory occlusion effect
Authors: Brummund Martin , Sgard Franck , Petit Yvan , Laville Frédéric , Boutin Jérôme
Published: May 2013

Earplugs remain a frequently used short-term solution for occupational hearing conservation. Due to comfort limitations, as induced by, e.g., the occlusion effect, workers often only wear earplugs for limited amounts of time and are likely to develop professional hearing loss. The occlusion effect expresses itself in the low frequencies through an altered perception of the wearer’s own voice and the amplification of physiological noises that occur upon earplug insertion. While many studies examined the occlusion effect experimentally, no study was found that attempted to implement an artificial external ear model dedicated to the measurement of the objective occlusion effect. A simplified external ear test fixture can help to better assess and design earplugs, because it allows standardized experimental testing. This work describes the implementation of a cylindrical artificial test fixture of the human outer ear that comprises the auditory canal as well as the bony, cartilaginous, and skin tissues that are made up of rigid polyurethane foam and two different types of silicone, respectively. Obtained measurement results are compared to literature findings.
Impulse noise attenuation by earplugs measured with the use of an acoustical test fixture and with the participation of subjects
Authors: Mlynski Rafal , Gorski Pawel
Published: May 2013

The effectiveness of impulse noise attenuation by hearing protector devices is most often determined collecting the data from measurements. In impulse noise conditions with high peak sound pressure level, it is necessary to replace a subject in a measurement with an acoustical test fixture. The use of the acoustical test fixture is important because of the potential risk of hearing damage occurring during impulse noise tests, performed with the participation of subjects. The impulse noise attenuation by earplugs determined from measurements carried out using acoustical test fixture was compared with attenuation determined with the participation of subjects (MIRE technique). The acoustical test fixture complied with the acoustic and mechanical requirements described in Standard No. ISO 4869-3 and was equipped with a chamber representing the external ear canal and a 2 cm3 chamber reflecting the acoustic properties of the middle ear. The results of measurements carried out with two different methods were comparable.
Acoustic standing wave based microsystem for low-concentration oil detection and separation
Authors: Wang Han , Liu Zhongzheng , Koo Chiwan , Kim Sungman , Cho Younghak , Kim Yong-Joe , Han Arum
Published: May 2013

Detection and quantification of extremely small amount of oil on site and at low cost has broad applications in environmental monitoring, both in oil spills as well as in routine marine/coastal ecosystem monitoring. For example, dispersed oil, generated through the use of chemical dispersants in oil spills to break up oil slick into small droplets so that they can be rapidly diluted in 3D space, are the greatest concern and poses the most challenges in detection. Fluorometry is the current standard method, however is bulky and expensive, limiting its wide deployment in the field. Here we demonstrate for the first time the development of an acoustic standing wave based microfluidic platform capable of processing large amount of liquid samples from which dispersed oil can be concentrated and separated to a detectable level by acoustophoretic force. The microfluidic platform consists of a recirculation channel structure into which dispersed oil droplets can be continuously separated from the main sample flow stream. A piezoelectric transducer attached at the bottom of the silicon-glass microfluidic channel creates the acoustic standing wave that exerts acoustophoretic force to oil droplets. An optical detector measures the presence of concentrated oil droplets by their distinct fluorescent signatures.
Bertram Scharf and his critical contributions to the field
Authors: Levitt Harry
Published: May 2013

Jerry Tobias introduced me to Bert Scharf via his book Foundations of Modern Auditory Theory. Our research in the 1970s was on the prediction of speech intelligibility from acoustic measurements; the similarity between Fletcher's 20 frequency bands of equal contributions to intelligibility and psychoacoustic measurements of critical bands was of great interest. Bert's chapter on critical bands provided an insightful, comprehensive, concise, and critical review of the state of the art in critical-band research. Our mutual research interests brought us together when he asked me to review a draft of a paper he was preparing. My review was highly critical and after submitting it to him I felt I had been too unforgiving in my review (reviewers can be wrong). I called him to explain that I had been overly critical of a fine paper. His response was the opposite of what I expected. He said it was the best review he had received and that he was extremely grateful for my input. I realized then that he—not only set high standards for others—but for himself as well. By adhering consistently to his high standards, his many contributions to the field have been long-lasting and critical.
Neural correlates of auditory attention
Authors: Shinn-Cunningham Barbara
Published: May 2013

Bert Scharf’s seminal studies on selective auditory attention were, in many ways, ahead of the times. Twenty years ago, many psychoacousticians viewed any consideration of “cognitive” factors or any effects driven by the intent of the listener, rather than the acoustics of the input, as outside of their realm of interest. However, today, a plethora of laboratories are exploring questions about what acoustic features enable listeners to focus attention, how bottom-up stimulus attributes interact with top-down control signals to determine what source a listener attends in a mixture of sources, and what neural mechanisms realize such selective auditory attention. This talk reviews recent work exploring selective auditory attention using a combination of behavioral studies and neuro-imaging techniques, all of which suggest that (1) listeners can focus attention on one, and only one, auditory object or stream at a time, and (2) executive control regions of the brain are engaged during attention to reduce across-object interference in the representation of whatever object is in the attentional foreground. These studies underscore the importance of auditory attention in allowing us to communicate in everyday settings containing multiple sound sources, and thus the foresight of Bert in tackling this problem when most others did not.
A hybrid modeling approach for vibroacoustic systems with attached sound packages
Authors: Alimonti Luca , Atalla Noureddine , Berry Alain , Sgard Franck
Published: May 2013

Modeling complex vibroacoustic systems including poroelastic materials using finite element (FE) based methods can be computationally expensive. Several attempts have been made to alleviate this drawback, such as high order hierarchical basis and substructuring approaches. Still, these methods remain computationally expensive or limited to simple configurations. On the other hand, analytical approaches, such as the Transfer Matrix Method (TMM), are often used, thanks to the lower computational burden. However, since the geometrical flexibility of the FE method is always needed in the low/mid-frequency range, attempts have been made to couple the FE model of the master system with a TM model of the sound package. Although these hybrid approaches seem promising, the open literature is not comprehensive. The aim of this work is to present a hybrid FE-TMM approach based on a Green's function formulation. The idea is to account for the sound package by approximating the effects over the treated surface using fundamental solutions (i.e., Green's functions) obtained by the TMM. A benchmark representative of typical applications is used to illustrate the capabilities of the presented methodology in terms of efficiency and accuracy in comparison to other classical methods.
Discontinuous Galerkin Methods for poroelastic materials
Authors: Dazel Olivier , Gabard Gwenael
Published: May 2013

In this work, we are interested in the development of a Discontinuous Galerkin Method (DGM) for sound absorbing materials. These materials are commonly used for noise and vibration control. The objective of this method is to discretize the structure and to represent in each element the field as a superposition of local solutions such as plane waves. This type of methods have shown their efficiency by requiring much smaller numbers of degrees of freedom compared to standard polynomial interpolations (i.e., FEM) especially when the frequency is increased. For poroelastic materials, the solutions are expressed in terms of Biot waves (two of them associated to compression waves and one corresponding to shear waves). The poroelastic problem is expressed as a first order model and the formulation of numerical flux at interfaces between elements is derived and implemented. Compared to classical DG methods for standard acoustics, this method is applied here to a dissipative medium and to a two-displacement field involving shear and compression waves. Several two-dimensional cases will be presented in order to validate the method and to compare against analytical and finite-element solutions (in displacement and mixed formulation). Results will be discussed in terms of accuracy of the method, errors, and conditioning of the linear systems.
Investigating the transmission loss effect via optimizing the insulator package on vehicle's firewall
Authors: Beigmoradi Sajjad , Jahani Kambiz , Hajabdollahi Hassan
Published: May 2013

Nowadays, noise and vibration of attributes of motor vehicles have a dominant effect on customers’ judgment about the cars, and hence, car manufacturers and OEM suppliers have dedicated remarkable time, budget, and concern to the investigations in this field. From NVH perspective, firewall is the foremost structural member in vehicle body design, since it is the main pass for transferring the engine induced noise to the passenger cabin. In this research, effect of the insulator package is studied through different configurations considering the radiated noise level and the optimized design is proposed using an optimization procedure. Indeed, it is concluded that adding the optimized insulator package can significantly refine the noise transmissibility while avoiding structural modifications in the firewall.
Voice stress analyses: Science and pseudoscience
Authors: Lacerda Francisco
Published: May 2013

Voice stress analyses could be relevant tools to detect deception in many forensic and security contexts. However, today’s commercial voice-based lie-detectors are not supported by convincing scientific evidence. In addition to the scientific implausibility of their working principles, the experimental evidence invoked by the sellers is either anecdotal or drawn from methodologically flawed experiments. Nevertheless, criminal investigators, authorities, and even some academics appear to be persuaded by the ungrounded claims of the aggressive propaganda from sellers of voice stress analysis gadgets, perhaps further enhanced by the portrays of “cutting-edge voice-analysis technology” in the entertainment industry. Clearly, because there is a serious threat to public justice and security if authorities adopt a naïve “open-minded” attitude toward sham lie-detection devices, this presentation will attempt to draw attention to plausibility and validity issues in connection with the claimed working principles of two commercial voice stress analyzers. The working principles will be discussed from a phonetics and speech analysis perspective and the processes that may lead naïve observers into interpreting as meaningful the spurious results generated by such commercial devices will be examined. Finally, the scope and limitations of using scientific phonetic analyses of voice to detect deception for forensic purposes will be discussed.
Precision and error of automatic speech recognition
Authors: VanDam Mark , Silbert Noah H.
Published: May 2013

Automatic speech recognition (ASR) software developed by the LENA Research Foundation (Boulder, CO) is an increasingly important tool in psycholinguistics. Naturalistic day-long recordings are segmented and assigned talker labels including those for KEY CHILD, ADULT MALE, and ADULT FEMALE. Performance of the system is a serious concern for ASR in general, not just the LENA system. Additional evidence of the software's performance is necessary to better interpret and understand accumulating research using this tool. Here we analyze the correspondence between computer and human segment labels corresponding to children, mothers, and fathers. Segments machine-labeled as ADULT MALE, ADULT FEMALE, and KEY CHILD were played to judges who identified each segment as Mother, Father, Child, or Other. Judges' responses were analyzed in terms of agreement, precision, and error. Overall agreement between machine and human coding was in the 70% range with Cohen's κ > .55. Machine coding appears to be better at coding KEY CHILD segments than the ADULT segments, and agreements for ADULT MALE labels were better than for ADULT FEMALE labels. The ASR system performed similarly when assigning segment labels for children and fathers, but less well for mothers. Overall error rates were generally very low. [Work support from NIH-NIDCD R01DC009560.]
Unsupervised machine learning for the accurate classification of the discourse maker like in code-switching utterances
Authors: Piccinini Page E. , Kramer Eric R.
Published: May 2013

Spanish-English bilinguals use the discourse marker like in English, Spanish, and code-switching utterances. An acoustic analysis found that the [l] and diphthong in like is produced differently depending on the type of utterance in which it occurs. To investigate the possible perceptual relevance of these differences, we built a logistic-polynomial regression model to classify like tokens based on acoustic data. The model first projects F1 and F2 values onto a space of time-dependent polynomials. We then apply multinomial logistic regression to classify these polynomials as English, Spanish, or code-switching. The area under the curve was 0.75, showing classification was significantly greater than random. This model outperforms models that rely on static values for F1 and F2, either at the midpoint of the token (DeLong’s test, p < 0.004), or two data points: the midpoint of the [a] of the diphthong, and the midpoint of the [ɪ] of the diphthong (DeLong’s test, p < 0.04). The superiority of the polynomial model suggests that the time-dependent progression of F1 and F2 values, rather than absolute formant values, is important for predicting an imminent code-switch. We hypothesize that listeners leverage these time-dependent changes in F1 and F2 to anticipate code-switches.
Experimental study of shout detection with the rahmonic structure
Authors: Kakino Naoto , Fukumori Takahiro , Nakayama Masato , Nishiura Takanobu
Published: May 2013

The surveillance systems with microphones have been developed to achieve a secure society. These systems can detect hazardous situations with observed speech but are generally very expensive. This is because the conventional systems manually detect hazardous situations by security officers. Thus, we focus on the automatic shout detection method, which can estimate hazardous situations. The acoustic model based on the Gaussian mixture model has been proposed as the conventional method to identify shouted and natural speeches. However, these methods have a problem that it is necessary to prepare huge training samples to accurately detect shouted speech. In the present paper, we focus on the rahmonic structure, which shows a subharmonic of fundamental frequency in the cepstrum domain because the rahmonic structure tends to arise in the shouted speech. In the present paper, we therefore propose the detection method of the shouted speech based on rahmonic structure. More specifically, we investigate rahmonic structure in the shouted speeches, and detect the shouted speech by utilizing the rahmonic structure model. We conducted evaluation experiments to confirm the effectiveness of the proposed method. As a result of evaluation experiments, we confirmed the effectiveness of the proposed method.
The detection of the sleepiness from the sounds obtained inside of the body
Authors: Akita Masanoti , Kamitabira Hiroyuki , Yoshida Tomohiko , Kanemitsu Syogo , Midorikawa Yoichi
Published: May 2013

This paper shows that the detecting method of the sleep-in sleep state or sleepiness from the sound signals in the human body. In former report, we showed that the detection of the sleepiness concerns with the piezoelectric sensors attached on car seat. Our preliminary examination showed that the spectrum of signals from the piezoelectric sensor have the tendency that the shapes of the spectral envelopes are flattened. And the sounds in human body are considered to have similar features. In this experiment, the signals of the piezoelectric sensor on the seat and the sounds in the human body are measured at the same time and the relation between the sounds and sleepiness are examined. The sounds inside the body are measured using NAM microphone system. The spectral envelopes of the signals from the left side and the right side of breath are calculated. The spectral envelopes from the seat are calculated at the same time. Twenty-two measurements by four examinees are done, and 8 sleeping data are measured. Flatness of the envelopes is defined using the lower order of cepstral coefficients, and the increase of the flattened spectrum is observed by the sounds from the sleeping data of three quarters.
Modeling occurrence tendency of adventitious sounds and noises for detection of abnormal lung sounds
Authors: Okubo Takanori , Yamashita Masaru , Yamauchi Katsuya , Matsunaga Shoichi
Published: May 2013

Diagnosis of pulmonary emphysema by using a stethoscope is based on the common knowledge that abnormal respiratory (adventitious) sounds usually appear in patients with pulmonary emphysema. However, the spectral similarity between adventitious sounds and noises at auscultation makes highly accurate automatic detection of adventitious sounds difficult. In this paper, we have proposed a novel method for distinguishing between normal lung sounds in healthy subjects and abnormal sounds, including adventitious sounds in patients, taking into account the occurrence tendency of adventitious sounds and noises. According to our investigation results, adventitious sounds occur repeatedly in successive inspiratory/expiratory phases of patients. On the other hand, noise sounds mix at random in lung sounds of both patients and healthy subjects. In our method, the occurrence tendency of these sounds is described using Gaussian distribution of a random variable obtained by subtracting the acoustic likelihood for abnormal respiration from the likelihood for normal respiration. The spectral likelihood calculated using hidden Markov models and the validity score of the occurrence tendency of the adventitious/noise sounds are combined to derive the classification result. Our method achieved a higher classification rate of 94.1% between normal and abnormal lung sounds than that achieved using the conventional method (87.4%).
A time-synchronous histogram equalization for noise robust speech recognition
Authors: Takahashi Fumiya , Kato Masaharu , Kosaka Tetsuo
Published: May 2013

The histogram equation (HEQ) technique is commonly adopted for feature space normalization in speech recognition systems. In this technique, a transform function is calculated directly from the histograms of both training and test data, and the nonlinear effects of additive noise are compensated. In order to estimate the transform function accurately, a certain amount of data are required. However, this is not suitable for real-time application because at least several seconds of evaluation data need to be accumulated before the transform function can be calculated. This means that the system cannot start the recognition process until the end of utterance. In this research, we aim to develop a new speech recognition method based on the HEQ technique for real-time processing. This method is called “time-synchronous frame-weighted HEQ (ts-FHEQ).” In the time-synchronous decoding, lack of data for estimating the histogram becomes a major problem. To resolve this problem, we introduce a frame weighting approach, where the degree of transform is controlled according to the number of data frames. Our speech recognition experiments verified that the proposed technique shows good performance and achieves substantial reduction of calculation time.
An overview of the development of resources, techniques, and, systems for Indian spoken languages
Authors: Agrawal Shyam S.
Published: May 2013

India possesses a large variety of languages and dialects spoken in different parts of the country. These languages possess some unique linguistic, phonological, and phonetic properties different from European languages. Research is being done in several of Indian languages—such as Hindi, Bangla, etc. to study the articulatory, acoustic—phonetic and prosodic nature for the purpose of creating standards of phonetic representation of phonemes and Pronunciation Lexicon in Indian Languages. Comprehensive and task specific language corpora, speech databases in laboratory as well as in mobile communication situation and the tools/technics required for processing of speech signals are being developed. The emphasis is on developing multilingual human-machine interaction systems. Some of the recently developed systems include multi-lingual speech recognition system for voice enabled services, multilingual text to speech synthesis system, speaker and language identification system for general purpose and forensic applications. Recognition of emotions in spoken speech, spoken language translation system, etc. The paper presents an overview of such studies conducted in various laboratories, academic institutions, and industries in India pertaining to these areas. The technologies used for data collection, processing, and recognition/ synthesis, etc., utilized and status of the development have been mentioned.
Control of frame loudspeaker array by minimizing fluctuations of frequency response and synthesized wave front
Authors: Ando Akio , Tokioka Aya
Published: May 2013

In sound reproduction with accompanying pictures, the localization of sound on the image display is problematic because a loudspeaker cannot be placed on the display. To gain a stable localization, the use of loudspeaker array set on the frame of the display may be a solution. In general, the loudspeaker array enables to control the perceptual depth of sound image by generating the appropriate curvature of the wave front corresponding to the source position. However, the frequency response and the shape of the wave front reproduced by such a frame loudspeaker array sometimes deteriorate, particularly when the virtual sound source has a certain distance back from the display. In this study, new parameters are introduced to scale the deterioration of the frequency response and the shape of the wave front. A new method is also introduced to generate the input signals to the loudspeakers on the basis of minimization of these parameters. The experimental result showed that the method generates the sound with small deterioration of the frequency response and the wave front regardless of the depth of the virtual source position, meaning that it can be used for the sound reproduction for 3D television.
Optimal beamformer designed for robustness against channel mismatch based on Monte Carlo Simulation
Authors: Bai Mingsian R. , Chen Ching-Cheng
Published: May 2013

Design of beamformers that withstand mismatch in channel characteristics of gain, phase, and position has been a key issue in array signal processing. These mismatch factors are random in nature and generally intractable by deterministic approaches. This paper examines the effects of channel mismatch on beamformer performance from a statistical perspective. The aim of this work is twofold: analysis and synthesis. In the analysis phase, the mismatch factors of microphone characteristics are assumed to be random variables following either uniform or Gaussian distribution. In the light of the Monte-Carlo Simulation (MCS), statistics including the mean, maximum, minimum, and the probability density function (pdf) of Directivity Factor (DF) can be efficiently obtained with random sampling. This provides useful information for choosing performance measures in the next synthesis phase. Optimal parameters of superdirective array designed using least squares (LS) and convex optimization (CVX) are determined based on the preceding performance measures. Simulation results have shown that the proposed statistical approach with different performance measures provided various performance-robustness tradeoffs in terms of parameter range for optimal beamformers.
Virtual sound source generation: Its various methods and performances
Authors: Kang Dong-Soo , Lee Jung-Min , Choi Jung-Woo , Song Min-Ho , Kim Yang-Hann
Published: May 2013

There are many means to generate a virtual sound source, or sources in the region of interest. For example, Wave Field Synthesis (WFS) or Higher Order Ambisonics (HOA) are good examples. These methods normally assume that loudspeakers are spatially distributed in the space. The region of interest where the desired sound is generated can have arbitrary shape; enclosed by surrounding loudspeakers or partially enclosed. Therefore, the performance of the method would be affected by the boundary conditions as well as the wave length of desired wave field. In other words, how the waves are distributed in the selected space. In recent work [Choi and Kim, IEEE Trans. Speech Audio Process. 20(7), 1976–1989 (2012)], a new approach was proposed to generate virtual sources in the space that is enclosed by an array of loudspeakers, which have been believed to be problematic with well-known methods. It is proved to be mathematically exact solution. However, “exact solution” does not necessarily mean that it is better than the others. In this paper, performances of these three methods are compared. Theoretical and experimental comparisons have been attempted and observed in this paper.
Aircraft sound environment reproduction: Sound field reproduction inside a cabin mock-up using microphone and actuator arrays
Authors: Gauthier Philippe-Aubert , Camier Cédric , Gauthier Olivier , Pasco Yann , Berry Alain
Published: May 2013

Sound environment reproduction of various ﬂight conditions in aircraft cabin mock-ups is useful for the design, demonstration, and jury testing of interior aircraft sound quality. To provide a faithfully perceived sound environment, time, frequency, and spatial characteristics should be preserved. Physical sound ﬁeld reproduction approaches for spatial sound reproduction are mandatory to immerse the listener in the proper sound ﬁeld so that localization cues are recreated. A 80-channel microphone array was built and used to capture a 2-h recording of in-flight sound environments within an actual Bombardier CRJ aircraft. An instrumented cabin mock-up was used to reproduce, in the least-mean-square sense, the recorded sound field using a 41-channel trim-panel actuator array. In this paper, experiments with multichannel equalization are reported. One of the practical difficulties was related to the use of the trim panels as sound sources. Windows and trim panels introduce audible squeaks and rattles if driven at low frequencies. Bass management was therefore implemented. Floor shakers and a subwoofer were used to recreate the low frequency content while the trim panels were only used for the high frequency range. The paper presents objective evaluations of reproduced sound ﬁelds. Results and practical compromises are reported.
Derivation of energy loss via coupled modes
Authors: Koch Robert A. , Stotts Steven A.
Published: May 2013

A coupled mode formalism can describe energy loss within an ocean waveguide due to all possible mechanisms, including bulk attenuation, scattering from rough surfaces, and volume inhomogeneities. First order corrections produced from mode-couplings can be incorporated into the modal loss. Accounting for these losses within an adiabatic mode approach would provide improvements over current standard modal propagation models. Highlights of the formalism for scattering from a rough surface will be provided.