The aim of this investigation was to determine the detection thresholds of sinusoidal and random changes in envelope of amplitude modulated (AM) signals. The envelope changes were obtained as a result of modulation of a pure-tone carrier by chosen types of modulators, which gave either sinusoidal, pseudorandom, or random (Gaussian) amplitude modulation. For pseudorandom and random modulation, the detection thresholds were determined when only modulation depth (m) of AM signals was changed randomly (f_m const) or when both modulation depth and modulation frequency (f_m) were changed randomly (f_m random). The data showed that for low modulation frequency (f_m≤64Hz) the detection thresholds for sinusoidal amplitude modulation (SAM), pseudorandom amplitude modulation (PAM) and random amplitude modulation (RAM) overlapped one another in the limit of the standard deviation (SD). However, for higher modulation frequencies, the average threshold for the RAM (f_m, const) was about 3dB lower (around f_m=128Hz) than that for sinusoidal amplitude modulation. In this case the PAM threshold was similar, within the limit of SD, to the SAM threshold. However, when random changes in amplitude were combined with random changes in frequency, the PAM (f_m, random), and RAM (f_m random) thresholds decreased about 5 dB relative to the SAM threshold.

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This paper proposes a method that provides a new set of linear fixed coefficients filters to define and synthesize a new set of reference signals that are fewer in number than the original set of reference signals. It also can be used as the reference signals for an active noise cancellation system. To optimize the performance of the active noise cancellation system, it is very important to find proper reference signals that are coherent with the target noise. In a case of active road noise cancellation in a vehicle cabin, for satisfying this particular requirement, many accelerometers are installed to detect vibration on the vehicle's suspension mechanism. In most cases, these reference signals contain information about fewer number of noise sources compared to the number of accelerometers. Proposing the linear filtering method that is based on Singular Value Decomposition (SVD) of the power spectrum matrix of the reference signals enables the synthesis of the virtual reference signals. After defining the frequency response functions of the filters, we design the ARMA direct form digital filters. After designing the fixed coefficients filters for the virtual reference signals, we evaluate the performance of the ANC system through experimental tests and its results by comparing the original reference signals and the virtual reference signals to confirm the proposed method.

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Acoustic diffraction from a line source by a semi-infinite plane in the presence of a moving fluid is studied. A finite region in the vicinity of the edge has a soft (pressure release) boundary condition; the remaining part of the semi-infinite plane is rigid. The effects of convection are included at low Mach numbers. The problem which is solved is a mathematical model for a rigid barrier with a soft edge. The far field is calculated using integral transforms, the Wiener-Hopf technique and asymptotic approximations. The physical interpretation of the result is then discussed.

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In the actual situation, it is very difficult to evaluate the complicated sound environment only by using the lower order statistics of intensity such as Leq. In the actual acoustic measurement, a background noise always contaminates observation values and the statistics of the background noise are usually unknown. In this paper, the statistical independency between the acoustic signal and the background noise is introduced as an evaluation criterion to identify the sound insulation systems. That is, by considering the sound insulation wall as a linear system on an intensity scale, the system parameters are estimated on the basis of the statistical information from the lower to the higher orders, instead of employing the artificial evaluation criterion such as the least squares error method. Then, the output probability can be easily predicted by using the estimated parameters of the sound insulation system when the other input noise passes through the same insulation system. Finally, the validity and effectiveness of the proposed method are experimentally confirmed by applying it to the actually observed data.

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To clarify at what point a skilled pianist's ability surpasses a novice pianist's with regard to basic temporal controllability of maintaining a uniform tempo, skilled and novice pianists tapped a piano keyboard and an aluminum board in equal intervals. They used three types of finger motions for the tapping. The temporal fluctuation in the tapping was analyzed by Fourier analysis to estimate the temporal controllability. As a result, when tapping with a single finger, there was no difference in the controllability between skilled and novice pianists. Skilled pianists had excellent controllability when they coordinated multiple finger motions as well as when they tapped with a single finger. However, novice pianists showed poor control for multiple finger tapping in contrast to single finger tapping. These results suggest that years of piano lessons do not improve basic controllability of maintaining a uniform tempo. They only improve the ability to coordinate multiple finger motions. There were no significant differences between the piano keyboard and the aluminum board for either skilled or novice pianists. It rejected the assumption that novice pianists were unskilled at controlling a piano keyboard.

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The nonlinear distortion of the 20-MHz ultrasound focused by a solid acoustic lens is theoretically and experimentally examined. The nonlinearity which may exist in the solid lens is neglected in the present theory as well as the linear absorption within the lens. The focused beam is detected with a planar receiver as large as the beam width, after passing through the similar receiving lens. In another way, the beam is also detected with different planar receiver, whose aperture is larger than the beam width, without using a receiving lens. At lower source level, the experimental result agrees well with the successive approximation solution for the Khokhlov-Zabolotskaya-Kuznetsov equation both in the amplitudes and phase difference of the fundamental and second harmonic components. When the source is intense within the upper limit of 10-W peak input-power, the harmonic amplitudes of the fundamental and second harmonic components seriously decrease while passing through the focus, and tend to saturate due to nonlinear absorption.

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In many techniques of sound speed measurement, sound propagation is regarded as plane wave in the medium. However, because of the finite aperture size, the sound wave does not propagate as an ideal plane wave. This deviation can be explained as a phase difference and as an amplitude difference, compared to the phase and amplitude of the planar transmitted sound wave. This position dependent phase difference will cause an error in transmission time measurement and thus in sound speed measurement. In this paper this phase difference is calculated in a liquid medium for the sound field of planar circulartransducers and concave circular transducers. Also the sound pressure received due to reflection by planar reflectors normal to the axis of the transducers is calculated. It is foundthat for points on the axis of the transducers, the error in transmission time measurement aepenason tne aperture size and does not depend on the transducer shape. For planar reflectors, at the same distances from the transducers, the error on transmission time measurement isgreater for the concave transducer, than for the planar transducer with the same size and frequency. The experimental result is presented for the sound speed measurement in water medium.

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