bluuebunny/crossref_metadata_2025 huggingface dataset:
The 100- to 150-ms window of auditory temporal processing and its consequences
Authors: Divenyi Pierre
Published: October 2013

Robert Efron (Percept. Psychophys. 14, 518–530 [1973]) demonstrated that the minimum duration of a percept is between 100 and 130 ms for auditory, vibrotactile, and visual stimuli, suggesting the presence of a most likely centrally located temporal integrator with a time window of about that length. Single-unit studies in the cat auditory cortex also unveiled an integration time of 100-150 ms (C. E. Schreiner and R. V. Urbas, Hear. Res. 21, 227–241 [1986]). This window corresponds to the lower limit of the longest among Ira Hirsh’s three-tiered temporal processing ranges [in Sensation and Perception, edited by H. R. Moskovitz et al. (Riedel, Dordrecht, 1974), pp. 251–258]. The present study argues that a temporal window of this duration can account for a variety of auditory perceptual results requiring temporal processing, such as temporal order identification, segregation of dynamically changing streams, robust speech intelligibility, and backward masking. Results also show that this window routinely becomes longer with age, with the possibility that the increased length of the window may account for some well-known central auditory processing deficits in aging. [Work supported by NIH, the VA Medical Research, and the 21st Century COE Progam, Kyushu University.]
Types of illusory continuity of verbal and nonverbal sounds: Some observations leading to their discoveries
Authors: Warren Richard M.
Published: October 2013

When gaps of a few hundred ms in a signal (e.g., speech, music, steady-state tones, tone glides) are replaced by a louder sound capable of masking the missing portion, the signal appears intact; the absent fragment can be perceptually synthesized through reallocation of a portion of the louder sound’s auditory representation. General rules and the special rules governing various types of restoration will be discussed, as well as how they were found through a series of observations, inferences, and experiments. [Work supported by NIDCD Grant No. R01 DC000208, and travel expenses provided by the COE Program of Kyushu University.]
An analysis of timing microstructure in Mozart’s piano sonata K.331
Authors: Ohgushi Kengo
Published: October 2013

It is known that a phrase-final lengthening, including a pause, plays an important role both for the intelligibility and the naturalness of speech. This fact is likely to hold true for performance of music. The present study investigates (a) how the phrase-final lengthening appears in performances of Mozart’s piano sonata in A Major (K.331) and (b) what are common and individual features of the performances by five groups of pianists, European professional, Chinese student, Korean student, Japanese professional, and Japanese student pianists. One hundred and twelve pianists’ performances of the first eight measures are analyzed in terms of the inter-onset intervals (IOIs). The data are averaged for the respective groups. The IOI patterns indicate (a) the final notes of the first half measures as well as the final notes of the second half measures are lengthened, (b) final note lengthening is more prominent in the fourth measure where the first phrase ends than in the other measures, and (c) Japanese student pianists play the rhythm of dotted eighth note and the sixteenth note differently from the pianists of other countries. [Work supported by JSPS and 21st Century COE, Kyushu University.]
Lost silence as a consequence of stream segmentation
Authors: Mori Shuji
Published: October 2013

A sequence of successive tones can be segmented into perceptually distinctive streams by inserting a silent gap or by changing the frequency in mid-sequence. This study demonstrates that a silent gap is perceptually lost when the gap coincides with the frequency change that occurs between tones. Even a 40-ms gap is undetectable when the tonal frequency is shifted from 800 to 3200 Hz. One interpretation of this ‘‘lost silence’’ phenomenon is the prioritization of frequency change as a cue for auditory segmentation. A silent gap and a change in frequency both mark a stream boundary if they are salient. They will be redundant as boundary markers when they coincide with each other; the frequency change will preempt the priority in auditory processing, whereas gap information will receive little processing and eventually become indiscernible. This phenomenon might be related to perceptual ambiguity of the temporal relation of separate streams (Van Noorden, 1975). [Work supported by JSPS.]
Exclusive allocation of a gap that is shared by two separate sounds
Authors: Remijn Gerard B. , Nakajima Yoshitaka
Published: October 2013

When a short temporal gap (40 ms or less) is inserted at the crossing point of two separate sounds of unequal duration, the gap is often allocated only to the shorter sound, whereas the longer sound can be perceptually completed (modal completion). This study shows that the exclusive allocation of the gap to the shorter sound—and the perceptual completion of the longer sound—is most profound when the start of the short sound and the start of the gap are separated by 400 ms or less. A possible explanation for the exclusive allocation of the gap is that within this temporal window, attention is engaged in the formation of a new auditory stream, a process initiated by the shorter sound’s onset. When subsequent auditory elements—even a gap—arrive within the temporal window of attentional engagement, these elements will be automatically included in the new stream. [This study was supported by the COE program of Kyushu University, Japan, and the Japan Society for the Promotion of Science.]
Evaluation of mid- and high-frequency impact insulation methods
Authors: LoVerde John J. , Dong David Wayland
Published: October 2013

Impact insulation is currently measured using impact insulation class (IIC) and a standard tapping machine described in ASTM E492 and E1007 (and corresponding ISO metrics and standards). The resultant IIC value does not correlate well with subjective reaction. In previous papers, the authors have focused on the low-frequency thudding from footfalls in lightweight wood construction and shown that the impact sound pressure level (ISPL) in the 63-Hz octave band correlates better with subjective reaction than IIC or other metrics [LoVerde and Dong, J. Acoust. Soc. Am. 112, 2201 (2002); INCE Inter-Noise 2004 Proceedings, 167 (2004); J. Acoust. Soc. Am. 119, 3220 (2006)]. Further, the low-frequency ISPL is controlled by different variables than the mid/high-frequency ISPL. This suggests that a two-metric system would be valuable where the first metric describes the low-frequency performance and the second metric describes the mid/high-frequency performance. Field testing experience is presented showing that IIC does not adequately differentiate the mid/high-frequency performance of various resilient materials. Candidate mid/high-frequency metrics are evaluated to determine if any result in improved methods for evaluation and rank-ordering of floor/ceiling assemblies.
Flanking impact sound transmission in wood-framed multifamily construction
Authors: Quirt David , Nightingale Trevor
Published: October 2013

This paper reports findings from a recently completed study of flanking sound transmission involving the wall/floor junction in wood-framed multifamily buildings. Flanking transmission exists in all buildings, but it can be controlled through good design. The paper examines how common construction details affect flanking paths between horizontally, vertically, and diagonally separated rooms. Variables considered include mounting conditions and orientation of the floor joists, framing of the wall assembly and any associated fire blocking, and mounting and number of layers of gypsum board. Estimates of the apparent sound insulation were obtained by summing the energy transmitted by the direct path through the wall or floor assembly separating the pair of rooms, and all the flanking paths involving the relevant wall-floor junctions. Results indicate that if there is no floor topping (i.e., the subfloor is bare) the apparent sound insulation for both airborne and impact sources is typically limited by flanking transmission involving the floor. Since, the source room floor is an element common to all impact flanking paths, three different toppings were evaluated as treatment options, and additional layers of gypsum board and resilient mounting were considered as options for the walls. The effectiveness of each option is discussed
On the generation and perception characteristics of Japanese timing control
Authors: Sagisaka Yoshinori , Muto Makiko , Kato Hiroaki , Tsuzaki Minoru
Published: October 2013

We have been studying temporal characteristics of Japanese for decades. Not only acoustic measurements but also perceptual studies on temporal modification have revealed the control principles lying behind manifestation of segmental duration characteristics. This talk tries to introduce familiar generation principles such as ‘‘mora-timing’’ and phrasal tempo reset from the viewpoint of humans temporal organization through the understanding of duration statistics. Moreover, by showing perceptual characteristics including sentential factors, we would like to propose how linguistic units relate to temporal perception and what kind of information form can be considered in temporal cognition process. Some of our recent efforts will also be presented to investigate language-dependent temporal characteristics by observing non-native learners temporal control characteristics. We really expect that we will be able to have a better picture of generation and perception mechanism through the cross check of their characteristic differences between natives and non-natives. [Work supported by Waseda Univ. RISE research project of ‘‘Analysis and modeling of human mechanism in speech and language processing’’ and Grant-in-Aid for and Scientific Research A-2, No. 16200016 of JSPS.]
Japanese mora-timing and processing: The case of devoiced vowels
Authors: Warner Natasha , Ogasawara Naomi
Published: October 2013

An extensive literature has argued that Japanese speakers compensate for inherent durational differences to keep moras to approximately equal lengths (e.g., Han, 1962; Port et al., 1980; Minagawa-Kawai, 1999; reviewed in Warner and Arai, 2001a). However, near-regularity of moras may not be because of intentional compensation (Warner and Arai, 2001b). This paper examines Japanese listeners processing, rather than production, of one case of irregularly timed moras: vowel devoicing. When Japanese vowels are devoiced, the mora tends to be shorter than usual. If speakers aim for regular timing, one might predict that listeners would also expect moras to be regularly timed and would thus have difficulty processing sequences that deviate from regular mora timing. However, one could predict that listeners are aware of what deviations from regular timing to expect, and thus have no difficulty processing timing deviations. The results, from phoneme monitoring and lexical decision tasks, show that listeners only have difficulty processing devoiced vowels in environments where devoicing would not be expected. In fact, where devoicing would be typical, listeners find devoiced vowels easier to process. This suggests that in processing, knowledge of patterns in the language outweighs any bias toward regularly timed moras.
A frequency domain method to estimate the coefficients of feedback control filter
Authors: Iwamatsu Yusuke , Itou Masahiro , Fujii Kensaku
Published: October 2013

This study proposes a method for repeatedly updating coefficients of a feedback control filter used for canceling a feedback path from a loudspeaker to a noise detection microphone. The coefficients are typically estimated by feeding a sequence of extra noise to the loudspeaker. However, feeding such extra noise under active noise control is undesirable. The proposed method estimates the coefficients without feeding the extra noise. Using the estimation errors involved in the coefficients of the noise control filter instead of the extra noise, the proposed method provides two independent equations and estimates coefficients of the feedback control filter by solving those. The two independent equations are provided through linear prediction using an auxiliary filter. A solution of the equations with two unknowns (a primary noise and the feedback path) provides the coefficients of the feedback control filter, perfectly canceling the feedback path. A salient drawback is that the feedback component is extremely small in comparison to the primary noise that is incident on the noise detection microphone. For that reason, estimation of the coefficients is difficult. This method solves the problem by first estimating the primary noise and then the feedback path.
A new multichannel adaptive algorithm
Authors: Itoh Masahiro , Fujii Kensaku , Muneyasu Mitsuji
Published: October 2013

This study presents a new multichannel adaptive algorithm that identifies unknown systems without superposing independent components on reference signals. A problem in such multichannel systems is that the reference signals involve strong cross-correlative components between themselves. These components delay the convergence of the coefficients of adaptive filters. At worst, they can render identification impossible. Usually, this problem is solved by superposing slightly independent components on the reference signals. Various superposition methods are therefore proposed. However, this superposition is perceivable as discordant noise. This study hence examines a new algorithm that exploits the residual components obtained by applying a linear prediction technique to the reference signals. The cross-correlative components thereby decrease and the convergence is hastened. Using this algorithm, however, the residual components are not fed into the unknown systems and adaptive filters. This is a point of great difference from conventional methods. This study also analyzes the performance of the proposed algorithm and derives the convergence condition as a range of step size. Finally, computer simulations are used to verify the derived convergence condition and the increased convergence speed.
A design of virtual sound field simulator based on acoustic ray tracing and wall impedance
Authors: Yamada Kiichiro , Denda Yuki , Nakayama Masato , Nishiura Takanobu
Published: October 2013

The sound propagation analysis using an architectural scale model is an effective approach for architectural acoustic design of a concert hall, and so on. However, the scale model must be reconstructed together, changing the basic plan of the architectural design. To cope with this problem, an architectural sound field has been virtually designed on the computer by the sound field simulator in recent years. The acoustic ray tracing is one of the simulators based on geometric acoustics. It is an efficient solution from the viewpoints of simulation cost and time. However, it cannot simulate the wall materials, because it employs the delta function as the wall impedance. To overcome this problem, in this paper, we try to design the virtual sound field simulator based on acoustic ray tracing and wall impedance acquired by real measurements. We first measured the reflection characteristics of the wall with impulse response to acquire the wall impedance. Then, we designed the virtual sound field simulator by convoluting the acoustic rays with measured wall impedances. As a result of subjective evaluation experiments, the proposed simulator could realize a more realistic sound field than the conventional acoustic ray tracing. [Work supported by MEXT of Japan.]
Binaural transformation coding with simulated head tracking
Authors: Shoji Seiichiro , Tew Anthony I.
Published: October 2013

The binaural transformation codec synthesizes generic binaural audio signals and generates accompanying side information. In the decoder the side information is used to personalize the generic signal. The performance of the basic codec is described in another paper; this paper describes the estimated effect on quality of incorporating limited head tracking. Generic HRTFs were used to spatialize two concurrent sound sources spatialized at ±30 or ±80 deg. The generic binaural signal was personalized in the BX decoder and at the same time the individual sound sources were rotated. These induced rotations are equivalent to compensating for head yaw rotations of −80, −40, −20, −10, and −5 deg, and for head pitch rotations of 22.5, 45, and 90 deg. Listening tests based on Recommendation ITU-R BS.1116-1 were used to evaluate the processed sounds. The tests were conducted using speech, vocals, guitars, and percussion source materials. It was found that the quality of the processed sound tended to degrade as the head yaw rotation increased. However, except for the head yaw condition −80 deg, the quality of the processed sound remained greater than 4.0 on the mean opinion scale. No deterioration of the processed sound was observed for the applied pitch rotations.
Individuality of elevation-angle-sensitive notches in head-related transfer functions
Authors: Sekimoto Saho , Iwaya Yukio , Suzuki Yôiti
Published: October 2013

Perceived localization can be controlled with high accuracy in the rendered auditory space when listeners’ own head-related transfer functions (HRTFs) are used for rendering 3-D spatial audio. However, measurement of a listener’s own HRTFs in all directions requires a special measuring apparatus and a long measurement time, which entail a considerable physical load on subjects. Therefore, effective individualization methods of HRTFs must be very useful. In previous studies, a few notches around 6–10 kHz, whose frequencies change depending on the elevation, are inferred as important to determine the perceived elevation [e.g., E. A. G. Shaw and R. Teranishi, J. Acoust. Soc. Am. 44, 240–249 (1968)]. Therefore, this study examines the individuality of such notches, of which the frequency is elevation-angle sensitive. First, individual HRTFs from all directions are measured for 140 listeners. They are then analyzed using an autoregressive moving average (ARMA) model. Second, the frequencies of these notches are expressed smoothly as a function of the elevation angle using dynamic programming (DP). Results qualitatively clarified the dependency of each frequency on elevation. Moreover, distributions of the notch frequencies show systematic variation as the elevation angle changes.
Inverse filter analysis of common harmonic structure on Specmurt using Riemann’s ζ function
Authors: Ono Nobutaka , Saito Shoichiro , Kameoka Hirokazu , Sagayama Shigeki
Published: October 2013

In this paper, based on the interesting relationship between the log-spaced δ function sequence and Riemann’s ζ function, the analytical properties of the inverse filter of the common harmonic structure on specmurt analysis are discussed. Specmurt [S. Sagayama et. al., in Proc. SAPA (2004)] is a simple and efficient technique for the multi-pitch analysis of polyphonic music signals. If all tones have the same harmonic pattern, the power spectrum on the log-scaled frequency can be regarded as the convolution of the common harmonic structure and the distribution of fundamental frequencies. Based on the model, overtones are effectively suppressed by the inverse filtering of the common harmonic structure in specmurt. Thus, for the stable processing, analytic properties of the inverse filter are significant. Our new finding is that when the common harmonic structure is expressed as a log-spaced δ function sequence with a particular kind of decay, the Fourier transform is exactly equal to Riemann’s ζ function. Interpreting several properties of Riemann’s ζ function in a signal processing context gives us the new perspectives like an explicit representation or cascaded decomposition of the inverse filter.
Multipitch estimation based on correlation between a harmonic structure model and an observed spectrum for mixed audio signals
Authors: Uchida Yoichi , Osaka Naotoshi
Published: October 2013

In the research of sound source separation and resynthesis, multipitch estimation is an important topic. We propose a multipitch estimation technique for monaural audio signals. It introduces a method of calculating multiple pitches within a frame for a pitched sound signal. First, a harmonic structure is modeled for a pitch M (Hz), using the cases of odd integer multiples of M, even integer multiples of M, and odd integer multiples of M/2. Then correlations are derived between each of the harmonic structure models and an observed spectrum. Harmonic structure models are expressed as the summation of Gaussian functions in the frequency domain, where each harmonic has a peak at its center frequency. An evaluation function is dynamically defined to test how well arbitrary values of M fit the observed spectrum. The evaluation function is created by searching for weighted coefficients for the correlation values. Next, a number of maximal peaks equal to the number of sound sources are selected from the spectrum and defined as the estimation pitches. This method was applied to a mixed piano signal, and a mixed audio signal that includes flute, clarinet, and cello. The results are good except at octave unisons.
Directions of arrival (DOA) estimation by particle filtering with dominant frequency components in noisy environments
Authors: Mizumachi Mitsunori , Ikoma Norikazu , Niyada Katsuyuki
Published: October 2013

In noisy environments, DOA estimation is useful for achieving attractive human-machine interfaces in the real world. This article proposes a robust DOA finder based on cross correlations between observed signals by particle filtering with frequency selectivity. The particle filter is flexible for modeling the system because it requires no linearity or Gaussianity on the model. The temporal trajectory of DOAs is modeled using a Markovian state space model. It is estimated through the state estimation procedure using particle filters. Markov modeling is expected to achieve stable DOA estimation in time. It is also a key for the proposed method to selectively use the dominant frequency components, such as formant frequencies of speech, to improve noise robustness. Assuming the quasi-stationarity of noise signals in the presence of a time-varying target speech signal, the dominant frequencies are extracted using a deterministic approach similarly to spectral subtraction. A set of the subband cross-correlation functions at the dominant frequencies is regarded as likelihood and is used for updating particle weights to estimate posterior distribution. The feasibility of introducing both particle filtering and frequency selectivity into DOA estimation was confirmed in several noise conditions.
Sound source localization using a distributed microphone system in real environments
Authors: Cho Kook , Okumura Hajime , Nishiura Takanobu , Yamashita Yoichi
Published: October 2013

Accurate localization of sound sources is indispensable for high-quality sound capture. The difference of distance from a sound source to two microphones causes a time delay of arrival (TDOA). The cross-power spectrum phase analysis (CSP) method has been proposed to estimate the TDOA based on correlation in signals. The sound source can be localized theoretically as a crossing point of sound directions estimated using two pairs of microphones. However, in a real environment, microphones receive sounds that have originated through reverberation and reflection. This situation makes TDOA estimation difficult, and accurate sound localization is difficult when using only two pairs of microphones. The authors propose a new method of sound localization based on multiple microphones that are widely distributed and placed under the ceiling of a room. A number of pairs of microphones give a series of CSP correlation coefficients as a function of the time delay. The CSP coefficients for hypothetical sound sources are accumulated over all the microphone pairs. The number of microphones is 16. The sound source is localized as a hypothesis that maximizes the accumulated CSP coefficient. Experiment results obtained in a real room show that the proposed method improves the localization accuracy.
Signal processing for small-size microphone array system—Application to teleconferencing system
Authors: Gotoh Akira , Murayama Yoshitaka , Ota Jun , Hamada Hareo , Watanabe Yuko , Hamada Hareo
Published: October 2013

A small microphone array system, consisting of two or more closely arranged small microphones and a digital filter network, is newly proposed. A method based on sound field control using multiple-point least-squares filter design contributed to the configuration of a smaller size microphone array system. Several directivity patterns having unique focus directions can be configured by changing each set of optimum filter coefficients. In this paper, a new method of scanning speaker direction using the small microphone array is proposed and several performances used in teleconferencing systems are evaluated.
Novel interface using a microphone array for wheelchair control
Authors: Sasou Akira , Kojima Hiroaki
Published: October 2013

This paper describes a novel interface that enables the control of a wheelchair by disabled people without having to use their hands. Thus far, a voice-driven wheelchair has been developed for disabled people who find it difficult to move their hands but are able to speak. Such wheelchairs usually adopt a headset microphone in order to reduce noise interference. However, a headset microphone is not practical because disabled people cannot wear the headset independently. Another problem faced is that the microphone shifts away from the mouth due to involuntary movements. In the proposed system, however, we use a microphone array that is mounted on a wheelchair, so that the disabled people do not need to wear the microphone. The system is also robust to noises and is able to recognize a face’s direction by estimating the position of a sound source such as the mouth. For instance, when disabled people need to turn left, they face left and make a sound that is not restricted to voice commands. Thus, the proposed system does not require disabled people to utter clear voice commands. This feature enables the application of the system to more general cases.
Comparison of measured signal coherence time in shallow and deep water
Authors: Yang T. C.
Published: October 2013

Temporal coherence of acoustic signals determines the processing of a sonar system for various applications. This paper studies the frequency and range dependence of the temporal coherence time of signal propagation in deep water versus that in shallow water. The signal coherence time has been measured in deep water since the 1970s, in terms of a parameter called the phase rate (the rate of phase change of the signal). Signal coherence time in shallow water has received little attention until only recently. We compare measurement data from more than a dozen of experiments covering a wide range of frequencies at different source-receiver ranges in deep and shallow water. We find that the signal coherence time in both shallow and deep water can be fitted with a universal equation, which exhibits a 3/2 power frequency dependence and 1/2 power range dependence. The signal coherence time in shallow water is two to five times longer than the signal coherence time in deep water at the same frequency and range. The signal coherence times are comparable between shallow and deep water if measured in terms of the range-to-depth ratio. [This work is supported by the U.S. Office of Naval Research.]
PlaneRay: An underwater acoustic propagation model using ray tracing and plane wave reflection coefficients
Authors: Hovem Jens M. , Dong Hefeng
Published: October 2013

Recently there has been a renewed interest in using ray tracing techniques for propagation modeling in underwater acoustics and this paper describes a newly developed model. A special and essential feature of this model is a unique sorting and interpolation routine for efficient determination of a large number of eigenrays and also for range-dependent environments. No rays are traced into the bottom and the bottom interaction is modeled by plane wave reflection coefficients. In the current version of the model the bottom structure consists of a fluid sediment layer over a solid half-space. Important considerations are the accuracy of the ray model and, in particular, how accurate the plane wave reflection coefficients can represent the effects of a layered bottom. This problem is analyzed by comparing the time and frequency domains solution of the ray model with models based on the wave number integration technique. There are two conflicting requirements since ray tracing is valid for high frequencies and the used plane wave reflection coefficient is valid for low frequencies where the sediment layers are thin compared with the acoustic wavelength. A number of illustrative examples are presented.
Time-frequency analysis of mode coupling in shallow-water sound propagation
Authors: Badiey Mohsen , Grigorev Valery , Katsnelson Boris , Lynch James
Published: October 2013

Time-frequency (TF) analysis for propagation of broadband sound pulses in shallow water in presence of perturbations of waveguides parameters, such as internal waves, temperature fronts, or bottom variability is considered. Due to waveguide dispersion one can construct a TF diagram for individual normal modes using Gabor transform for each received pulse. If we have a temporal sequence of propagating pulses then we can find a temporal sequence of TF diagrams. It is shown that in the presence of spatial and temporal waveguide perturbations (such as the sound-speed profile in the water column or the sea bottom) structure of a TF diagram and its temporal behavior depends on waveguide parameters, on the source-receiver positions, and on the character of mechanisms influencing the sound propagation, such as mode coupling, horizontal refraction, or adiabatic regime. Dependence of the TF diagram on waveguide variability in geotime provides information about temporal behavior of the perturbations. Structure of TF diagram is analyzed for a shallow-water waveguide model as well as for the SWARM95 experimental data. [Work supported by ONR and RFBR.]
Warped-time-stretched pulse: An acoustic test signal robust against ambient noise
Authors: Morise Masanori , Irino Toshio , Banno Hideki , Kawahara Hideki
Published: October 2013

A new acoustic measurement signal that is a hybrid signal of time-stretched pulse (TSP), or lin-TSP, and logarithmic TSP (log-TSP) is proposed. The signal, referred to as warped-TSP [Morise et al., IEICE Trans. Fundamentals, A, J89-A(1), 7–14 (2006)], has a single parameter to adjust for better measurements in accordance with ambient noise conditions. It also provides a means to eliminate harmonic distortions produced mainly by loudspeaker systems. In this lecture, the definition and features of the warped-TSP in comparison with the lin-TSP and log-TSP are introduced. The following were shown: (1) the relationship between the parameters, the amplitude frequency characteristics, and the effect on the harmonic distortion components; (2) a method to select the optimal parameters of the warped-TSP for a specific measuring environment; and (3) the experimental results for a series of impulse response measurements under different ambient noise conditions. Those results show that the proposed method outperformed the lin-TSP and log-TSP under all conditions in terms of SNR of the measured impulse response. [This research was supported partly by grants-in-aid for scientific research (15300061 and 15650032) and a grant from the Faculty of Systems Engineering at Wakayama University.]
Dynamics of laser-trapped microbubbles
Authors: Takahira Hiroyuki
Published: October 2013

A laser-trapping method is utilized for microbubbles. Bubbles of the order of 10 microns in diameter are trapped and manipulated successfully using a dry objective lens with large working distance. The growth or shrinkage of a laser-trapped microbubble and the merger of microbubbles are observed with a high-speed camera to investigate the influence of gas diffusion on the stability of microbubbles. Two kinds of equilibrium radii are found for shrinking microbubbles. The first one is related to the equilibrium surface concentration of surfactant. The other is related to the decrease of the surface tension due to the compression of the surface area at the maximum surfactant concentration. The simulations in which the dynamic surface tension is considered are in good agreement with the experiments. The laser trapping technique is also applied to the motion of a microbubble in a shear flow. It is shown that the bubble escapes from the laser trap being repelled by the optical force in the shear flow. There is overall agreement between the experiments and the simulations in which the buoyancy force, the fluid dynamic forces, and the optical force are taken into account.
Novel methods of micro-object trapping by acoustic radiation force
Authors: Yamakoshi Yoshiki
Published: October 2013

It is expected that micro object trapping by acoustic radiation force is a useful method in future drug delivery systems in order to concentrate the payloads at desired position. In this paper, two novel methods of micro object trapping are presented. First method is micro object trapping by seed bubbles. This method uses seed bubbles, which have higher sensitivity to the ultrasonic wave, in order to trap micro objects, which are difficult to trap by conventional methods due to low volumetric oscillation under the ultrasonic wave. The Bjerkne’s force, which is produced by a secondary wave radiated from the seed bubbles, traps the target objects making bi-layer seed bubbletarget object mass. The Second method is micro bubble trapping by bubble nonlinear oscillation. Two ultrasonic waves with different frequencies (pumping and control waves) are introduced simultaneously. The frequency of the control wave is set to a harmonic frequency of the pumping wave. If the bubbles flow into the cross area of two waves, nonlinear oscillation by high intensity pumping wave generates the Bjerkne’s force, producing multiple traps with narrow separation along the control wave propagation direction. In order to demonstrate these methods, experiments using an ultrasonic wave contrast agent are shown.
Assessment of shock wave lithotripters via cavitation potential
Authors: Iloreta Jonathan I. , Szeri Andrew J. , Zhou Yufeng , Sankin Georgii , Zhong Pei
Published: October 2013

An analysis of bubbles in elastic media has been made in order to characterize shock wave lithotripters by gauging the potential for cavitation associated with the lithotripter shock wave (LSW). The method uses the maximum radius achieved by a bubble subjected to a LSW as the key parameter that defines the potential damage a lithotripter could cause at any point in the domain. The maximum radius is determined by an energy analysis. A new index—similar in spirit to the Mechanical Index of Holland and Apfel for diagnostic ultrasound—is proposed for use in gauging the likelihood of cavitation damage.
The correlation between cavitation noise power and bubble-induced heating in high-intensity focused ultrasound
Authors: Farny Caleb H. , Wu Tianming , Holt R. Glynn , Roy Ronald A.
Published: October 2013

It has been established that inertial cavitation is responsible for elevated heating during high-intensity focused ultrasound (HIFU) application for certain intensity regimes. The contribution of bubble-induced heating can be an important factor to consider, as it can be several times that expected from absorption of the primary ultrasound energy. Working in agar-graphite tissue phantoms with a 1.1-MHz HIFU transducer, an embedded type-E thermocouple, and a 15-MHz passive cavitation detector (PCD), the temperature and cavitation signal near the focus were measured for 5-s continuous wave HIFU insonations. The measured temperature was corrected for heating predicted from the primary ultrasound absorption and the transient thermocouple viscous heating artifact to isolate the temperature rise from the bubble activity. We have found that the temperature rise induced from the bubble activity correlates well with the instantaneous cavitation noise power as indicated by the mean square voltage output of the PCD. The results suggest that careful processing of the cavitation signals could serve as a proxy for measuring the heating contribution from inertial cavitation. [Work supported by the Dept. of the Army (Award No. DAMD17-02-2-0014) and the Center for Subsurface Sensing and Imaging Systems (NSF ERC Award No. EEC-9986821).]
Membrane permeabilization of adherent cells with laser-induced cavitation bubbles
Authors: Dijkink Rory , Ohl Claus-Dieter , Nijhuis Erwin , Le Gac Sèverine , Vermes Istvàn
Published: October 2013

Strongly oscillating bubbles close to cells can cause the opening of the cell’s membrane, thus to stimulate the uptake of molecules from the exterior. However, the volume oscillations of bubbles induce complex fluid flows, especially when bubble-bubble interaction takes place. Here, we report on an experiment where a single cavitation bubble is created close to a layer of adherent HeLa cells. The interaction distance between the bubble and the cell layer is controlled by adjusting the focus of the pulsed laser light, which creates the cavitation bubble. The dynamics of the bubble and the cells is recorded with high-speed photography. The poration of the cells is probed with different fluorescent stains to distinguish viable and permanent poration and programmed cell death (apopotosis). Quantitative data are presented as a function of the radial distance from the stagnation point. Our main finding is the importance of the asymmetrical collapse and the high-speed jet flow: After impact of the jet onto the substrate a strong boundary layer flow is responsible for shearing the cells.
Free-field calibration of 1/4 inch microphones for ultrasound by reciprocity technique
Authors: Takahashi Hironobu , Fujimori Takeshi , Horiuchi Ryuzo , Sato Sojun
Published: October 2013

Recently, equipment that radiates ultrasound radiation at frequencies far beyond the audible range is increasing in our environment. Such electronic equipment has switching regulators or inverter circuits, and many devices are unintended sources of ultrasound radiation. However, the effects of airborne ultrasound on human hearing and the human body have not been well investigated. To estimate the potential damage of airborne ultrasound radiation quantitatively, it is necessary to establish an acoustic standard for airborne ultrasound because the standard is a basis of acoustic measurement. With the intention of establishing a standard on airborne ultrasound, a free-field calibration system with an anechoic chamber was produced. The principle of free-field calibration techniques is introduced in this presentation. Type WS3 microphones (B&K 4939) were calibrated in the system to examine the calibration ability to be achieved. Results showed that it can calibrate a microphone from 10 to 100 kHz with dispersion of less than 1 dB. In addition, the effects that were dependent on the uncertainty of the calibration are discussed based on those results.
An environmentally robust silicon diaphragm microphone
Authors: Arimura Norihiro , Ohga Juro , Kimura Norio , Yasuno Yoshinobu
Published: October 2013

Recently, many small microphones installed in cellular phones are the electret condenser microphones (ECMs) that contain an organic film diaphragm. Although FEP of fluorocarbon polymer is generally used as the electret material, silicon dioxide is also used. Recently ECMs have been made small and thin while maintaining the basic sound performance according to the market demand. In addition, environment tests and the reflow soldering mounting process have been adjusted to meet market requirements. On the other hand, the examination satisfied the demand as the high temperature resistance was insufficient. This paper depicts an examination and a comparison of conventional ECM with the experimental model, a silicon diaphragm condenser microphone produced using the MEMS method. The silicon diaphragm satisfies high-temperature resistance and stable temperature characteristics because of its very small coefficient of linear expansion and it is measured total harmonic distortion (THD) on high pressure sound. Finally, it will be able to be used in high temperature and high pressure sound conditions in the future.
Automatic discrimination between singing and speaking voices for a flexible music retrieval system
Authors: Ohishi Yasunori , Goto Masataka , Itou Katunobu , Takeda Kazuya
Published: October 2013

This paper describes a music retrieval system that enables a user to retrieve a song by two different methods: by singing its melody or by saying its title. To allow the user to use those methods seamlessly without changing a voice input mode, a method of automatically discriminating between singing and speaking voices is indispensable. We therefore first investigated measures that characterize differences between singing and speaking voices. From subjective experiments, we found that human listeners discriminated between these two voices with 70% accuracy for 200-ms signals. These results showed that even short-term characteristics such as the spectral envelope represented as MFCC can be used as a discrimination cue, while the temporal structure is the most important cue when longer signals are given. According to these results, we then developed the automatic method of discriminating between singing and speaking voices by combining two measures: MFCC and an F0 (voice pitch) contour. Experimental results with our method showed that 68.1% accuracy was obtained for 200-ms signals and 87.3% accuracy was obtained for 2-s signals. Based on this method, we finally built a music retrieval system that can accept both singing voices for the melody and speaking voices for the title.
Soundscape, moderator effects, and economic implications
Authors: Hehner Cay , Schulte-Fortkamp Brigitte
Published: October 2013

Soundscape is considered with respect to moderator effects and the contribution of economics. It will be questioned whether soundscapes can work as a moderator concerning noise annoyance. As shown by the different investigations concerning soundscapes, a definition of the meaning of soundscapes is necessary. Evidently, the moderating effect of a given environment and its soundscape has to be discussed on three levels: (1) extension of factors that describe annoyance,(2) peculiar feature of burdensome noise contexts, and (3) discrepancies of the social and economic status of people living in areas where the rebuilding will change the quality of the area. It has to be determined and analyzed to what extent the Georgist method of resources taxation, as recently exemplified, e.g., in Alaska and in Wyoming, can be instrumental in funding soundscapes to moderate noise annoyance as it has been the case in funding free education and allowing the distribution of a citizen’s dividend.
Initial steps for the determination of environmental noise quality—The perception-related evaluation of traffic noise
Authors: Genuit Klaus , Guidati Sandro , Rossberg Sebastian , Fiebig Andr
Published: October 2013

Directives call for actions against noise pollution and noise annoyance. But, how do we eliminate harmful effects including annoyance due to the exposure of environmental noise without understanding the perception and evaluation of environmental noise? How do we preserve environmental noise quality where it is good (Directive 2002/49/EC) without identifying descriptors for noise quality? Various soundscape approaches based on different methodologies have been developed in the past. But, the measurement procedures must be realizable without much effort in order to achieve acceptance from legislation. Therefore, procedures have to be developed that capture the complexity of human hearing, on the one hand, and are feasible with respect to economy and time, on the other hand. The European project Quiet City (6FP PL516420) is dealing with, among other aspects, vehicle pass-by noise as a typical environmental noise source and its evaluation. Results of the analysis based on subjective assessments and psychoacoustic analyses carried out with respect to the development of an annoyance index will be presented and discussed. Such an index will provide valuable information for effective improvement of noise quality. The final aim is to develop a descriptor valid for complete traffic noise scenarios predicting environmental noise quality adequately.
When objective permissible noise limits of a municipal planning process and a subjective noise ordinance conflict
Authors: Hale Marlund
Published: October 2013

In most communities, proposed new building projects are required to conform with planning, community development, zoning, and/or building and safety specifications and standards. In cases of allowable noise exposure and noise limits, where certain of these requirements are quite specific while others are purposefully vague, conflicts between residential and commercial neighbors can lead to extreme disagreement and needless litigation. This paper describes a recent situation occurring in an upscale beach community, the resulting conflict over existing noise sources that comply with the limits of the city planning and permitting process, and the interesting findings of the court following the civil and criminal litigation that followed. Some suggestions are given to avoid these conflicting policy situations.
Is ultrasound appropriate to measure bone quality factors?
Authors: Laugier Pascal
Published: October 2013

Theoretical considerations support the concept that quantitative ultrasound variables measured in transmission are mainly determined by bone microstructure and material properties. All these properties are attributes of bone other than its bone mineral density (BMD) that may contribute to its quality and thus to strength or fragility. However, the limitations of this approach for a BMD-independent characterization of bone quality, long questioned, have become indisputable. Such considerations have prompted a research aiming at the development of new methods capable of measuring bone quality factors. New ultrasonic approaches are being investigated that use ultrasonic backscatter, guided waves, or nonlinear acoustics for studying bone microstructure or microdamage. These approaches, combined with sophisticated theoretical models or powerful computational tools, are advancing ideas regarding ultrasonic assessment of bone quality, which is not satisfactorily measured by x-ray techniques.
Scanning acoustic microscopy studies of cortical and trabecular bone in the femur and mandible
Authors: Katz J. Lawrence , Spence Paulette , Wang Yong , Misra Anil , Marangos Orestes , Hazony Dov , Nomura Tsutomu
Published: October 2013

Scanning acoustic microscopy (SAM) has been used to study the micromechanical properties of cortical and trabecular bone in both the human femur and mandible. SAM images vary in gray level reflecting the variations in reflectivity of the material under investigation. The reflection coefficient, r=(Z2−Z1)/(Z2+Z1), where the acoustic impedance (AI), Z=dv, d is the materials local density and v is the speed of sound at the focal point; Z2 represents the AI of the material, Z1 that of the fluid coupling the acoustic wave from the lens to the material. Femoral cortical bone consists of haversian systems (secondary osteons) and interstitial lamellae, both of which show systematic variations of high and low AI from lamella to lamella. The lamellar components defining the edges of trabecular cortical bone exhibit the same lamellar variations as seen in cortical bone. Mandibular bone, while oriented perpendicular to the direction of gravitational attraction, exhibits the same cortical and trabecular structural organizations as found in the femur. It also exhibits the same systematic alternations in lamellar AI as found in femoral bone. Both femoral and mandibular cortical bone have transverse isotropic symmetry. Thus, modeling elastic properties requires only five independent measurements.