bluuebunny/crossref_metadata_2025 huggingface dataset:
Physical modeling of Tibetan singing bowls using dynamic forcing
Authors: Etlinger David M.
Published: October 2013

The geometric simplicity of Tibetan singing bowls produces surprisingly complex vibrational behavior. The curvature of the bowls causes strong coupling between modes, and the physical parameters of actual bowls (namely their thickness and deep curvature, as well as their asymmetrical excitation) produce higher-order effects that cannot be neglected. In this paper we present a physical model for the bowl’s acoustical output. The model is based on analytical solutions for the natural modes, and predicts both transient and steady-state behavior. Using the method of dynamic forcing, we investigate bowl response for linear (striking) and nonlinear (rubbing) excitation methods, producing audio as well as numerical data. Finally, we use the model to gain insight into aesthetics and design: by comparing our simulations to actual antique or high-quality bowls, we can begin to determine what parameters contribute most to creating the desired sound.
Combined effects of occupational exposure to noise and organic solvents on the auditory system
Authors: Morioka Ikuharu , Minami Yoshihiro , Miyai Nobuyuki , Miyashita Kazuhisa , Yamamoto Hiroichi , Kaewboonchoo Orawan
Published: October 2013

Noise exposure is the most common cause of hearing loss. In many work environments where organic solvent exposure occurs, a high level of noise exposure is also common. However, as to the ototoxic effects of organic solvents, the earlier studies focused on the exposure to an organic solvent alone. It is necessary to clarify the combined effects of exposure to noise and organic solvents at the simulated situation of the work environments. The objective of this study was to review the literature on the ototoxic effects of exposure to organic solvents combined with noise. Two organic solvents were selected: methanol, a widely used organic solvents in the occupational field, and styrene, also widely used and a human neurotoxicant. Judging from the results of animal experiments and epidemiologic studies, a synergistical interaction of noise with such organic solvents’ ototoxicity was clarified even at the lower concentration where the organic solvent alone showed no ototoxicity in the experimental animals and at the concentration lower than the occupational exposure limits of organic solvents. Consequently, it is important to carefully access the occupational exposure limits when workers are exposed to organic solvents combined with noise, and to take measures against combined exposure.
Active noise control-like performance of a spherical reflector
Authors: Seto Kunisato , Khan Md. Tawhidul Islam , Teramoto Kenbu , Ohta Hiromitsu
Published: October 2013

The principle of cancellation of tonal noise by using a spherical reflector, the designing method of an appropriate reflector for canceling jet screech, the advantage of a spherical compared to a flat reflector for canceling a tonal noise, and a prospective application for an aircraft to protect the fuselage from acoustic fatigue are discussed. The thrust loss due to the new technique was also evaluated directly with a vertical wind tunnel. The change in the aerodynamic structure was optically observed by using a schlieren apparatus along with a high-speed video camera. [Work supported by grant of JSPS, Japan.]
A study of subjective effect on active noise control system
Authors: Utsumi Sayoko , Murai Eiji , Watanabe Yuko , Hamada Hareo
Published: October 2013

In recent years, the influence of various noises on human beings in our daily life has become a problem. Especially, the low-frequency noise has been considered to have a serious influence on work efficiency of people who are working under the noise exposure. On the other hand, active noise control (ANC) systems have achieved the efficient reduction noise, especially for the low-frequency noise. Actually, a number of successful ANC systems have been reported. Therefore, ANC systems may improve our work efficiency. In this paper, features of various noises are observed to get a representative value to be a sound source for subjective listening tests. Then, the effectiveness of the ANC system is evaluated to observe a physiological reaction of subjects. For that, response times to the short-term memory task will be measured in three different conditions, such as usual condition, under the noise exposure, and that with a running ANC system.
Noise levels with the potential to impede communication during surgery
Authors: Oudyk John , Stringer Bernadette
Published: October 2013

Surgical teams consist of personnel who may or may not regularly work together, among whom communication can be more difficult because of surgical masks, face shields, or goggles, factors believed to have contributed to a recent serious medication error that occurred when the anaesthetist misunderstood the surgeon’s request. In a multihospital, occupational health intervention study a sound level meter was used to measure ambient noise for 15 min or > during 256 surgeries included in the study. Results revealed an Leq of 70.7 dB(A) and peak of 115.8 dB(A) (at the 95th percentile). Since the clarity of words depends on maintaining a signal-to-noise ratio of at least 15 dB(A), personnel working with such noise would likely increase their speaking levels up to 75–85 dB(A), a level substantially higher than normal speech ranging from 55–65 dB(A). Although the effect of noise levels on patient safety, or the health of OR personnel who must raise their voices to be heard, as well as on the overall climate that results when noise is elevated, have not been thoroughly investigated, evidence of negative effects is growing. And, while further characterization of OR noise is justified, this should be carried out to develop tailored interventions that can then be evaluated. [Work was funded by Ontario’s Workplace Safety Insurance Board.]
Comparison of speech communication of renovated classroom by subjective method
Authors: Lam Coriolanus , Leung Chun Wah , Li Kai Ming
Published: October 2013

This paper presents measurement results on the acoustical characteristics of two standard and identical classrooms, one of which has been renovated. Several acoustical quantities that affect speech communication have been studied. These quantities were used to predict and quantify the characteristics and the qualities of measured classrooms that focused on speech communication. Subjective method by speech intelligibility tests was conducted in the measurements. Phonetically balanced monosyllabic word tests in both English and Chinese were carried out. In addition, public address messages, both in English and Chinese, from Hong Kong Mass Transit Railway were used as material for the tests. The results indicate that the renovation was beneficial on the quality of speech communication in general except for echo effects and increase in reverberation under several combinations of acoustical settings. These results were used to evaluate the relationship between speech perception quality and speech intelligibility. Besides, it will be used for new classroom designs in Hong Kong for better speech communication. [Work supported by Research Grants Council of The Hong Kong SAR Government, Research Committee of The Hong Kong Polytechnic University, and Architectural Services Department of The Hong Kong SAR Government.]
Speech privacy unmasked
Authors: Lubman David , Sutherland Louis C.
Published: October 2013

Background masking noise is widely used to achieve speech privacy between work stations in open offices and in nearby closed (executive) offices. Although speech privacy is often essential, high masking noise levels reduce the abilities of some workers to perform speaking and listening tasks effectively and without strain. Masking noise levels are usually set as high as office inhabitants will tolerate, typically about 45–48 dB (A-weighted). This is 10 to 20 dB higher than noise levels in typical home settings. Many who can communicate effectively in home settings are impaired in offices with background masking noise. People adversely impacted include those with mild or worse hearing disabilities of hearing, speech, and language. Many are older workers and non-native speakers. The current architectural trend to greater inclusiveness underscores the desirability of lower background masking noise levels. Designers can usually achieve speech privacy with lower masking noise levels by specifying partitions with higher noise isolation ratings. Higher sound isolation ratings cost more than higher masking noise levels. But higher costs may be balanced by the benefits of a more inclusive and humane workspace.
Objective and subjective evaluation of the acoustic comfort in public classrooms
Authors: Zannin Paulo H. , Diniz Fabiano B. , Krauser Daniele
Published: October 2013

The acoustic comfort of classrooms in a Brazilian public school has been evaluated through interviews with 62 teachers and 464 pupils and measurements of background noise, reverberation time, and sound insulation. Acoustic measurements have revealed the poor acoustic quality of the classrooms. Results have shown that teachers and pupils consider the noise generated and the voice of the teacher in neighboring classrooms as the main sources of annoyance inside the classroom. Acoustic simulations resulted in the suggestion of placement of perforated plywood on the ceiling, for reduction in reverberation time and increase in the acoustic comfort of the classrooms. [Work supported by DAAD, CNPq and FINEP.]
Classroom acoustics: Future criteria for the design and renovation of learning spaces
Authors: San Souci Sooch , Guerra Line , Teichner Nicolas , Campbell Dick
Published: October 2013

Recent literature suggests that some of today’s criteria for classroom acoustics might be in need of revision. This article explores the traditional acoustic classroom criteria as they are applied today, and why several of the design goals for construction or renovation of learning spaces might not be appropriate acoustic targets. The acknowledged quest for optimized acoustics for the enhancement of learning is discussed and related to a series of newer metrics for the measure and modeling of speech level, occupied and unoccupied noise, controlled reverberation, and SNR values presented on a seat-by-seat basis. Based on the measured results of classrooms designed and built in accordance to these newer metrics, our research indicates that these guidelines would incur an average increase of 2% to 8% to current budgets. Specifics involving building costs versus acoustic value are also discussed. Further, research involved in the evaluation of scholastic advantages is underway.
In-liquid plasma processing using high-frequency or microwave irradiation
Authors: Nomura Shinfuku , Toyota Hiromichi
Published: October 2013

In-liquid plasma is generated by applying high-frequency (27 MHz) or microwave (2.45 GHz) radiation to hydrocarbon liquids or water. In-liquid plasma is expected to replace gas-phase plasma because the reaction rates for chemical deposition of in-liquid plasma are much higher than those for conventional vapor deposition. Moreover, the cooling effect of the liquid mitigates thermal damage to the substrate. In-plasma generation requires the existence of bubbles because this plasma occurs in bubbles in liquid. Two methods exist for generating bubbles. One is to generate vapor bubbles by heating the liquid. The other is to generate acoustic cavitaion bubbles. Sonoplasma is created when acoustic cavitation is used to generate in liquid plasma. This plasma is not arc-discharged plasma, but rather high-frequency or microwave plasma. Therefore, it is useful in many industrial applications. Using this technology as a chemical reactor, synthesis of diamondlike carbon, polycrystalline silicon carbide, and carbon nanotubes, decomposition of waste oils, and the production of hydrogen can be achieved easily in hydrocarbon liquids. Temperatures were measured from the ratio of emission intensities of Hα and Hβ. Plasma temperatures in hydrocarbon liquids are higher than those in water.
Controlled cavitation in lab-on-a-chip systems
Authors: Ohl Claus-Dieter , Zwaan Ed , Le Gac Sèverine , Tsuji Kinko
Published: October 2013

Typically cavitation bubble dynamics is studied either in a free, liquid or close to rigid, free or flexible boundaries. Here, we report on a new experimental approach to study cavitation bubbles in confined geometries with sizes comparable to the cavitation bubble (20–100 μm). These fluid environments are fabricated in a polymeric material (PDMS, polydimethylsiloxane) using a conventional molding process from a silicon master (lab-on-a-chip systems). The bubbles are generated by focusing pulsed laser light into the microfluidic systems filled with a light-absorbing dye. The bubble dynamics is recorded stroboscopically and with high-speed photography. We report on bubble dynamics in various geometries such as in triangular, square, and disk-shaped structures. Additionally, we study a cavitation bubble-based micropump, the effect of the channel width on jetting dynamics, and the interaction of the cavitation bubble on suspension cells. The fluid flow is easily visualized by adding microparticles into the liquid. The authors propose that this new experimental approach opens an interesting toolbox in cavitation research.
Numerical investigation of interactions between ultrasound propagation and cavitation bubbles in shock wave therapy
Authors: Liebler Marko , Dreyer Thomas , Riedlinger Rainer E.
Published: October 2013

Acoustic cavitation in shock wave applications is responsible for several therapeutic effects as well as for undesired side effects. Using a two-phase continuum approach for bubbly liquids, a numerical model is presented to simulate the interactions between nonlinear ultrasound propagation and cavitation bubbles. Numerical results are presented, showing in what manner the presence of cavitation bubbles alters the pressure pulse propagation and how the bubble dynamics can be specifically affected by changing the shock pressure waveform. An increased bubble density leads to significant changes in the tensile part of the pressure waveform. This effect is caused by the changes in medium compressibility due to the oscillating cavitation bubbles. Calculations for the bubble cloud dynamics induced by three pressure waveforms with identical energy but different temporal forms of the tensile part are compared. Simulation results demonstrate that tensile parts with low amplitude but long duration increase the cavitation effect compared to high amplitude but short tensile phases.
Music sound restoration and the effect of presentation conditions
Authors: Sasaki Takayuki
Published: October 2013

When a music sound in a melody was deleted and replaced by a noise, a missing sound can be restored perceptually. This illusory phenomenon in audition, named music sound restoration, can be proved indirectly by inaccurate judgment of temporal position of the inserted noise. In the present study, music sound restoration was investigated in terms of the temporal position of replaced noise, headphone listening or free field listening, and the listeners’ knowledge of the phenomenon. The results showed that the accuracy of the noise location differed between the positions of replaced noise in the melody or within a beat. The perceptual restoration occurred not only in a headphone listening but also in a free-field listening. It was revealed that the accuracy of noise localization elevated when the listeners knew the stimulus construction, but it was still lower than that of a blank. It was also confirmed that a missing sound could be restored perceptually, not only in a melodic sequence but also in a simple sequence such as scale. [This work was supported by JSPS and 21st Century COE, Kyushu University.]
Japanese standards and situation of studies about heavy and soft impact sources
Authors: Hiramitsu Atsuo , Inoue Katsuo
Published: October 2013

This report describes standards of heavy and soft impact sources. They are used for measurement of floor impact sound insulation of buildings. In Japan, the method for field measurement of floor impact sound level (JIS A 1418) was revised in 2000 and separated into two parts. Part 1 uses a standard light impact source and corresponds to ISO 140-7. Part 2 uses heavy and soft impact sources. At that time, a rubber ball was standardized along with a car-tire source in JIS A 1418-2. Moreover, JIS A 1440-1 and JIS A 1440-2 for laboratory measurement of the reduction of transmitted floor impact sound have been prepared. JIS A 1440-1 corresponds to ISO 140-8, and JIS A 1440-2 uses heavy and soft impact sources. Floor impact sources characteristics are introduced. They include force characteristics, temperature dependence, stability, and others. In addition, among contents of JIS A 1440-2, the measurement method, calculation method, and a laboratory of a box-frame-type reinforced concrete construction are explained. In addition, study examples using a rubber ball are introduced. As examples, we present the floor impact sound level, the transmitted floor impact sound level, hearing tests on floor impact sound, and evaluation of walking sensations.
Heavy and light impact sources to rate impact sound transmission and changes due to applied floor toppings
Authors: Warnock Alf , Nightingale Trevor
Published: October 2013

The magnitude and spectrum of the power injected by an impact source depends on the impedance of the source and floor. If standardized impact tests are meant to give ratings that correlate well with subjective impressions of footstep noise, it follows that the impact source used should have the same impedance as an average human foot at least over the range of test frequencies. The ISO tapping machine, the Japanese tire machine, and an 18-cm-diam rubber ball do not satisfy this criterion. Consequently, their impact spectra differ from those from a live walker. Floor toppings, in particular, are ranked differently. Examples of discrepancies will be presented for direct transmission between vertically separated rooms. For horizontally and diagonally separated rooms, flanking transmission controls the impact sound pressure level. The sound pressure level depends not only on the power injected by the source, but also source location relative to the flanking junction because of propagation attenuation across the floor. Sensitivity to source location is similar for different sources (ISO tapping machine and Japanese ball), suggesting that the same source could be used for direct and flanking transmission measurements if the source adequately simulates the impedance of a human foot.
Timing variations of frequent utterances occurring in a very large corpus of spontaneous Japanese conversational speech
Authors: Campbell Nick
Published: October 2013

A corpus of spontaneous conversational Japanese speech was collected from volunteer subjects who wore high-quality head-mounted microphones and recorded their daily spoken interactions to minidisk over a period of 5 years. All recordings were transcribed and tagged according to interlocutor type, and a portion representing about 10% was further annotated for speech-act, speaker-state, emotion, and speaking style. This paper presents timing data from the corpus, showing how the same utterance can vary according to speaking style and other factors. It presents the hundred most common utterances in the corpus and relates their durations to spectral and prosodic characteristics that vary according to affect, attitude, intention, and relationship with the listener.
Temporal control in Japanese dialects
Authors: Kubozono Haruo
Published: October 2013

This paper discusses the extent to which temporal control varies among Japanese dialects. Most dialects of Japanese including Tokyo Japanese are so-called ‘‘mora dialects’’ in which the mora serves as the basic unit of timing. In the periphery of Japan, however, we find a couple of dialects called ‘‘syllable dialects’’ whose basic phonological unit is the syllable rather than the mora. Kagoshima Japanese (KJ) is one such dialect, but little is known about its temporal organization. The main purpose of this paper is to report on the results of an acoustic experiment on this syllable-based dialect and to compare them with those of the experiments about Tokyo Japanese (TJ). This comparison reveals the following four points: (i) geminate consonants are considerably longer than single consonants in both TJ and KJ, (ii) however, the phonetic duration of words in KJ is not determined by the number of moras involved but by the number of syllables, (iii) in KJ, vowels are consistently shorter when they follow geminate consonants than when following single consonants, and (iv) unlike English, Korean, and other languages, neither TJ nor KJ shows an effect of vowel shortening in closed syllables.
Effects of speaking style on the regularity of mora timing in Japanese
Authors: Brady Michael Connolly , Port Robert F. , Nagao Kyoko
Published: October 2013

The regularity of mora timing in Japanese has remained controversial over the years [Warner and Arai, Phonetica 58, 1–25 (2001)]. It is possible that the degree of regularity varies with speaking style. Four Japanese subjects spoke six-mora proper names with varying syllable structures where the second name had either three simple syllables (e.g., Tomiko) or two syllables with a long vowel (e.g., Tooko) or a long consonant (e.g., Tokko). These were read either in formal sentences, read in conversational style sentences, or used in spontaneous description of pictures of characters having these names. Moras were measured as the intervals between vowel onsets using an automatic vowel-onset detection algorithm. Although all styles suggested regular mora timing, the results show that the styles differ in the degree of temporal compensation for the moras constituted by long vowels and long consonants which are shorter than consonant-vowel syllables. The compensation was clearest in the most formal style of speech.
Representations of within-word structure and Japanese speech timing
Authors: Otake Takashi
Published: October 2013

One complication about the debate of mora timing is that durational units are closely related to syllable structure in which morae can be treated as subunits of syllables. As a consequence, durational properties of the components in three types of syllables, CVN, CVV, and CVQ, can equally be examined from the point of representations of within-word structure. Durational properties in Japanese speech timing undoubtedly play an important role, but knowledge of representations of within-word structure cannot be neglected because it is uniquely language specific. In this presentation it is discussed how knowledge of representations of within-word structure plays a significant role in recognition of morae from the point of psycholinguistic framework, presenting various types of data that were collected from (1) Japanese preschool children (before and after acquiring kana orthography), (2) Japanese monolinguals and Japanese English bilinguals, and (3) Japanese L2 learners (from beginning to advanced whose native language is English). Also reported are data from Telugu (a Dravidian language spoken in India), whose speech segmentation mechanism behaves like that of mora-timed language. It is hoped that knowledge of representations of within-word structure together with the durational properties will facilitate Japanese learners to acquire Japanese speech timing.
DiscourseWare, a blended learning courseware for understanding and producing Japanese audio-visual reactive tokens
Authors: Kawai Goh , Utashiro Takafumi
Published: October 2013

We developed blended learning courseware called DiscourseWare for learning Japanese audio-visual reactive tokens. Acquiring proper token choice, timing, frequency, duration, and intonation are emphasized. The courseware consists of interactive computer-based multimedia material and classroom instruction. Lessons start with instructor-led classroom instruction on the types and usage of various reactive tokens. Learners then independently use interactive computer-based courseware to learn how to choose appropriate reactive tokens. The courseware contains videos showing two native speakers conversing using a uncontrolled vocabulary and syntax and at a natural speed. The reactive token being taught is blanked out. Learners solve multiple-choice problems asking which reactive token would be appropriate in that context. Learners increase proficiency in interpreting various prosodic patterns of the same phonetic string. After computer-based training, learners regroup and practice exchanging reactive tokens. Our courseware is accompanied by a syllabus for reactive tokens that might interest researchers of comparative pragmatics as well as language instructors. The conference presentation will include DiscourseWare demonstrations, and before-and-after training comparisons of non-native learners’ reactive token productions as assessed through interviews with language teachers.
An issue of rate normalization for learners of Japanese
Authors: Hirata Yukari
Published: October 2013

This talk explores native and non-native speakers’ identification of Japanese phonemic length contrasts in sentences of varying speaking rates. Native Japanese speakers were found to use the rate of the sentence surrounding a target word as a perceptual cue (i.e., to normalize speech rate) to distinguish phonemic vowel and consonant length. This perceptual cue overrode more local cues such as physical duration of the target vowel or consonant. However, when beginning learners of Japanese participated in the same listening tasks, they were unable to normalize speech rate and identify length of vowels and consonants as native speakers did. Specifically, when asked to identify the length of vowels in /mV(V)mV(V)/ (with the first vowel accented) in sentences of slow and fast rates, the learners were least accurate in identifying the long vowel of /mVmVV/ in sentences spoken at a fast rate. The learners’ perception was biased by physical duration of vowels and by the presence or absence of pitch accent. Their identification accuracy for long vowels was more affected by speaking rate than that for short vowels. Taken together, these results suggest a need for a training program to help learners of Japanese to cope with speaking rate variations.
Effects of exposure and training on perception of Japanese length contrasts by English listeners
Authors: Tajima Keiichi , Kato Hiroaki , Rothwell Amanda , Akahane-Yamada Reiko , Munhall Kevin G.
Published: October 2013

Native English listeners are known to have difficulty distinguishing Japanese words that contrast in phonemic length, often realized as a contrast in vowel or consonant duration. The present study reports results from a series of experiments investigating the extent to which English listeners’ perception of such length contrasts can be modified with exposure to Japanese and with perceptual identification training. Listeners were trained in a minimal-pair identification paradigm with feedback. A pretest and posttest were also administered, using natural tokens of Japanese words containing various vowel and consonant length contrasts, produced in isolation and in a carrier sentence, at three speaking rates, and by multiple talkers. Results indicated that exposure and perceptual training substantially improved identification accuracy. Even though listeners were trained to identify words contrasting in vowel length only, performance also improved for other contrast types. Furthermore, speaking rate strongly affected performance, but training improved performance at all three rates, even though listeners were trained using only stimuli spoken at a normal rate. These results suggest that non-native listeners are highly susceptible to factors that affect the temporal characteristics of speech, but their perceptual strategies can be modified with exposure and training. [Work supported by NICT and JSPS.]
Categorization of phonemic length contrasts in Japanese by native and non-native listeners
Authors: Kato Hiroaki , Tajima Keiichi , Rothwell Amanda , Akahane-Yamada Reiko , Munhall Kevin
Published: October 2013

Previous studies reported that exposure to Japanese and identification training allowed English listeners to improve their identification accuracy of Japanese words with phonemic length contrasts. However, this does not necessarily mean their perceptual properties approach those of native listeners in all respects. Past perceptual studies have shown that native listeners show, for example, an extremely sharp short-to-long boundary and shift of the boundary to adapt to changes in the temporal context, i.e., typically speaking rate variations. To carefully investigate the effects of training on such properties, the present study analyzed the learners’ identification of stimulus continua between word pairs that minimally differed in the length of a phoneme. Overall results showed that English listeners’ boundaries tended to be sharpened by the identification training, but only to a limited extent. On the other hand, the results also showed that English listeners’ boundaries shifted across different speaking rates in ways that resemble native listeners’ adaptation tendencies. The latter result suggests that even non-native listeners can adjust their identification boundaries according to differences in temporal context. Discussion will include both prospects and limitations of listeners’ improvement by the perceptual training in this study. [Work supported by NICT and JSPS.]
Adaptive surround technology for stereo music signals
Authors: Murayama Yoshitaka , Gotoh Akira , Ota Jun , Hamada Hareo , Watanabe Yuko , Hamada Hareo
Published: October 2013

Computation of the left and right channel instantaneous difference (xL−xR) is mostly used to obtain the signals for surround sound. This signal is usually referred as the antiphase components. However, there are many problems to using the antiphase components in terms of its applicability in practice as well as the sound quality. For instance, when signals xL and xR are equal, or almost equal, this variation, that is to say, the correlation coefficient value, is high; there is no way to determine the direction vector uniquely, therefore an ambiguity may appear. Instead of the antiphase component’s computation, a new method is proposed by using an adaptive signal processing technique. In this paper, it is revealed that natural surround sound signals having no antiphase and uncomfortable feeling are adaptively generated by the proposed method. The other advantages from a signal processing point of view are also discussed. Finally, subjective evaluations on sound quality of a surround sound system are carried out.
A design of fast steering filters based on the adaptive fusion of predesigned finite impulse response filters for microphone array
Authors: Nakayama Masato , Nishiura Takanobu , Yamashita Yoichi
Published: October 2013

With a microphone array, a desired distant-talking speech signal can be acquired selectively by steering the directivity in noisy environments. To form directivity, adaptive beamformers have been proposed as conventional beamformers. They can form null directivity with a small number of transducers. However, adaptive beamformers have a serious drawback: it can’t realize the noise-robust adaptation in variable noisy environments, because it has necessary heavy computational costs to train multi-channel adaptive steering filters. To solve this problem, we propose to design the fast steering filters based on the adaptive fusion of predesigned FIR filters in variable noisy environments. The adaptive fusion of predesigned FIR filters is realized by adding (or subtracting) various predesigned FIR filters immediately. Therefore, the proposed method can quickly design the steering filters even in variable noisy environments. In addition, microphone calibration is indispensable for applying the proposed method. Therefore, we also propose the new microphone calibration method based on environmental noise without target signal. Evaluation experiments were conducted in a real acoustic room. As a result of evaluation experiments, we could confirm that the ASR (automatic speech recognition) performance of distant-talking speech was improved on the well-known condition of noise source directions. [Work supported by MEXT of Japan.]
Frequency domain simultaneous equations method for active noise control systems
Authors: Ujino Tadashi , Fujii Kensaku , Muneyasu Mitsuji , Fujita Yusuke
Published: October 2013

This study proposes a new method for feedforward-type active noise control systems. This method, named the frequency domain simultaneous equations method, is based on a different principle from the filtered-x algorithm requiring a filter modeled on a secondary path from a loudspeaker to an error microphone. Instead of the filter, this method uses an auxiliary filter identifying the overall path consisting of a primary path, a noise control filter, and the secondary path. As seen from the configuration of the overall path, the auxiliary filter can provide two independent equations when two different coefficient vectors are given to the noise control filter. In practical use, the two different coefficient vectors are provided by their own estimation errors. By solving the independent equations, the proposed method can estimate the coefficient vector of the noise control filter minimizing the output of the error microphone. This study first presents computer simulation results demonstrating that the proposed method can automatically recover the noise reduction that is in effect degraded by path changes. Finally, this study applies the proposed method to an experimental system and, by using the system, verifies that the proposed method is functional in practical systems.
A study of pinna effect on head-related transfer functions
Authors: Sekido Hiromi , Watanabe Yuko , Hamada Hareo
Published: October 2013

For the sound localization, an interaural difference is important. To obtain the desired localization, it is necessary to use a head-related transfer function (refer to HRTF), which is defined as the transfer function between a sound source and a surface of the ear drum. In our previous work, as far as in the case of reproducing the virtual sound source in the horizontal plane, it is confirmed to be capable of remarkable localization by using the numeric data calculated by a rigid sphere head model instead of a dummy head microphone. However, it is necessary to solve the problem of the front back confusion and the sound localization of the vertical plane. For that, it might be essential to add the pinna information to the rigid sphere model, to examine how each part of the pinna is related to the localization.
Sound reproduction system with switching control band range on every listening point
Authors: Tatekura Yosuke , Watanabe Takeshi
Published: October 2013

We propose a robust multichannel sound reproduction system that utilizes the relationship between the width of the actual listening area and the control frequency of the listening points. The conventional multichannel sound reproduction system uses inverse filters that are designed based on room transfer functions (RTFs) from loudspeakers to listening points in the listening room. However, the sound reproduction accuracy is deteriorated by room environment variations when fixed inverse filter coefficients are used because the condition number relevant to the instability of the matrix composed of the RTFs is large, especially at low frequencies; the inverse filters are unstable at those frequencies. This tendency is more remarkable when listening points are arranged more closely. To resolve the problem, we switch the control band at every listening point to avoid degrading the reproduced sound in low-band frequencies. The control band range of the listening points at both ears is only high band. That of the other listening points is the entire control band. Numerical simulation with real environmental data showed that improvement of the reproduction accuracy is about 6.4 dB on average, even with temperature fluctuations of 5°C as environmental variations of the listening room.
A method to make a moving virtual sound source
Authors: Chang Ji-Ho , Kim Yang-Hann
Published: October 2013

This study deals with a method that manipulates the sound field, which is excited by a moving source. That is, the moving source is virtually constructed by array speakers. It is realized by controlling pressure values on boundary of the field. This is, in fact, an attempt to implement the Kirchhoff-Helmholtz integral equation in practice. It is well known that we can generate a certain sound field in a space by controlling pressure values on boundary [S. Ise, Acustica, 85, 78–87 (1999)]. However, time varying sound field control has not been well addressed. Our aim is to manipulate sound field that is generated by a virtual source moving straightly outside the sound field or traveling around it. To express the method, the simple pure-tone case is considered and then expanded to more general cases. It also handled the problem related to eigenfrequencies at which boundary pressure control is not applicable. Computational and experimental results are introduced as well. [Work supported by BK21 & NRL.]
Partial-differential-equation-based sound source localization: Finite Fourier integral approach and its application to multiple source localization
Authors: Fujita Yuya , Ono Nobutaka , Ando Shigeru
Published: October 2013

In this paper, a method based on the sound source constraint PDE (partial differential equation) is developed for real complex environments with multiple sources. The sound source constraint PDE is integrated within a finite temporal interval with orthogonal Fourier bases to obtain simultaneous equations including source directions and distances as unknown variables. In them, the fast Fourier transform can be used rigorously and efficiently with least computation time. We show the equations enable us to obtain 3-D locations of up to four sources simultaneously from one spatial gradient observation of sound field. Numerical simulation comparison with the spatio-temporal-gradient-based method was carried out and showed the significant improvement of accuracy. Experimental results using a six proximity microphone array [N. Ono et al., in Proc. ASJ Autumn meetingCvol.IC, pp. 607–608, September, 2004 (in Japanese)] in the real environment will be reported.
Holographic diagnosis: Basic principles and applications
Authors: Jeon Jong-Hoon , Kim Yang-Hann
Published: October 2013

Acoustic signals achieved from a machine often have information on condition of the machine. This means that the location of faults can be found by using acoustic signal. Location of sound source can be found by using techniques such as MFAH (moving frame acoustic holography), which is applicable for pure tone noise and good signal-to-noise ratio, and beamforming, which basically assumes no reflected wave. Faults due to breakdown of rotating parts in a moving vehicle, however, make the above method hard to be directly applicable. The difficulty stems from three characteristics in the fault detection of the moving vehicle. First, the sound source radiates periodic impulsive noise. Second, there exist reflected waves from the ground since the array microphones should be settled between the vehicle and the ground to monitor the bottom side of the vehicle. Last, for the measurement problem, there exists noise due to other parts and wind for the moving vehicle, which degrades coherence among measured signals. In this research, holographic diagnosis, a machine diagnosis technique based on holographic approach, is introduced to find the position of faults effectively. The modified version of beamforming that models the moving periodic impulsive noise as a scan function is introduced.
Complex envelope implementation for fast and efficient acoustic holography
Authors: Park Choon-Su , Kim Yang-Hann
Published: October 2013

Acoustic holography allows us to predict the spatial pressure distribution on any surface of interest. To implement this method, we normally use Fourier transform in time and space domain. It is noteworthy that the data size is so huge that it takes a long time to calculate pressure field. Moreover, the reconstructed pressure field is frequently too complex to observe the characteristics of pressure field. One possible candidate is the complex envelope. The complex envelope in time domain is well known and widely used in various fields. We have attempted to extend this method in the space domain, so that we can have a rather compact sound pressure picture that provides the information we need, for example where sound sources are. This belongs to what we call the analysis problem of acoustic holography. We might want to draw some parts in detail but other parts in a rather coarse way. The complex envelope in space certainly meets this objective. First we start with the simplest case. We devise a monopole complex envelope. Then we extend it to finite size source case. Various holography examples are reprocessed according to what we propose and demonstrate how this method is practically fast and how it provides a better picture for analyzing the sound field.
Detection for the early reflection in the impulse response by the correlation function
Authors: Itakura Toru , Tang Yong , Tanaka Wakako , Shibayama Hideo
Published: October 2013

Detecting an early reflection sound contained in the impulse response measured in a room, we calculated the correlation factor between the early reflection and the impulse response. Calculating the correlation factor every sampling period, we make the correlation function. For the function, we set a threshold level. We detected the absolute correlation value larger than the threshold level. And, we set the zero value for subthreshold level. Following this algorithm, we generate a new signal with the high correlation factor for the early reflection. We compared the signal with the impulse response by using listening test.
Acoustic sound field simulator based on head-related transfer function and three-dimensional mirror-image
Authors: Kuroda Keitaro , Denda Yuki , Nakayama Masato , Nishiura Takanobu
Published: October 2013

The head-related transfer function (HRTF) is the most important factor to simulate an acoustic sound field for the virtual reality technology or the architecture field. However, the HRTF has individual differences on every listener, because it represents the real transfer function between the sound source point in the acoustic field and the listener’s ears. In addition, it is impossible to measure the HRTFs on every listener. To solve this problem, we focus on three-dimensional mirror-image. It can represent the virtual transfer function (especially, room reverberation) in an acoustic field by replacing all surface reflections with mirror-image sources. Therefore, we try to design the new acoustic sound field simulator by convoluting the real transfer function of the HRTF and the virtual transfer functions of the three-dimensional mirror-image. By this approach, the proposed simulator may realize the more accurate sound-image localization with room reverberation by three-dimensional mirror-image even if the HRTF of an other listener is employed. As a result of evaluation experiments in some conditions, the proposed simulator could improve the sound-image localization performance especially in an anteroposterior direction more than the conventional simulator based on HRTF. [Work supported by MEXT of Japan.]
Novel method for the efficient individualization of binaural audio signals
Authors: Shoji Seiichiro , Tew Anthony I.
Published: October 2013

A novel coding technique suitable for the transmission of binaural audio signals with concurrent multiple sound sources is proposed in this paper. Existing algorithms typically suffer from audible artefacts, poor localization, or excessive bandwidth. Especially with conventional intrasource coding methods, bandwidth is dependent on the number of sources and becomes greater as the number of sources increases. The proposed binaural transformation (BX) encoder synthesizes binaural signals using generic HRTFs and transmits them to the decoder along with side information. The bandwidth of the side information is independent of the number of sources. Using locally stored HRTFs matching those of the listener, the decoder turns the generic binaural signals into personalized ones. We present two types of BX codec: type A and type B. Psychoacoustic experiments based on Recommendation ITU-R BS.1116-1 were used to compare signals processed by these codecs with the corresponding unprocessed signals. For a variety of source materials (speech, vocals, guitars, and percussion) we show that high mean opinion scores are achievable using the type A codec when its parameters are set appropriately.
Objective and subjective evaluation of numerically estimated head-related transfer functions
Authors: Takane Shouichi , Abe Koji
Published: October 2013

In this study, head-related transfer functions (HRTFs) of a dummy head and human subjects are numerically estimated by using the boundary element method (BEM), and their availability for auralization is discussed from the subjective viewpoints. Geometries of their heads are obtained by a 3-D laser scanner. The effect of the torso for the estimation of the HRTFs is also examined and discussed via the 3-D measurement. The estimated HRTFs are compared for each subject with the measured HRTFs in various objective criteria, such as spectral distortion (SD), signal-to-distortion ratio (SDR), and interaural time/level differences (ITD/ILD). Availability of the estimated HRTFs for the synthesis of the 3-D sound image are then evaluated via hearing experiments. In our previous study [Proceedings of WESPAC IX (2006)], the results of the experiment is insufficient since the low-frequency range of the estimated HRTFs caused the ambiguity in sound localization. In this study, the frequency range of the estimation is extended to as high frequency as possible, and the availability of the estimated HRTFs for auralization is discussed from two viewpoints: the subjective difference between the estimated and measured HRTFs, and some subjective attributes obtained from the hearing of the stimuli convoluted with the estimated HRTFs.
The effect of head movement on sound localization using stereo microphones: Comparison with dummy heads
Authors: Endo Wataru , Kashino Makio , Hirahara Tatsuya
Published: October 2013

This study examined the effect of head movement on sound localization with a pair of microphones providing no head-related transfer function (HRTF) information, and with individualized, nonindividualized, and downsized dummy heads providing HRTF information with different degrees of distortion. In an anechoic room, white noise was presented for 5 s from one of 12 loudspeakers. A dummy head or a microphone pair was placed at the observation point. Listeners heard the sound via headphones outside the anechoic room and judged the direction of the source and whether the sound image was extracranial. Without head movement, localization accuracy with the microphone pair was significantly worse than that with dummy heads. When voluntary head movement was allowed and the dummy head was moved in accord with the listener’s movement, accuracy with nonindividualized and downsized dummy heads improved nearly to the individualized level. Accuracy with the microphone pair also improved with head movement, but did not reach the level of dummy heads. An extracranial image was perceived with dummy heads in most cases, but not often with the microphone pair even with head movement. Head movement cannot completely compensate for the lack of HRTFs.