This paper presents a new method of exciting the acoustic phase-conjugate wave and itsbasic experimental results. In recent years, the acoustic phase-conjugate wave hasattracted a special interest and several methods for it have been studied. Stimulated bysuch status, the authors have studied a method of exciting the acoustic phase-conjugatewave using nonlinear vibration in a piezoelectric resonator. That is, ultrasonic pulseswere radiated into the piezoelectric resonator driven in the resonance condition. In thiscase, the main frequency of the pulses was chosen as a half of the resonance frequency ofthe resonator. The experimental results made clear that the output level of the reflectedsignals from the resonator increased with the input voltage added to the resonator andthat such a characteristic was kept even if the ultrasonic pulses were radiated into theresonator at a certain angle of inclination. The characteristic is surely the same as that of the phase conjugate wave.

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The coding principle of the research is an application of recursive and adaptive identificationprocess of AR (auto-regressive) process. The system can reproduce speech waveonly by the transmission of the difference (error) signal between input speech and predictedone at sending end, with two identical linear predictors and Kalman Filters atsending and receiving end respectively. At sending and receiving end, predictors arecontrolled recursively and adaptively, in the same way, by the difference signal withKalman Filters. The final output is the sum of received error signal and predicted oneat the receiving end. Several techniques are developed to reduce a bit rate. A bit ratecan be reduced to 16 kbps with an average SNR better than 30dB, and a quality ofreproduced speech is good enough for an ordinary communication use. A bit rate canbe reduced to 8 kbps by an altanate use of control and free running samples, and voicequality is definitely improved by the addition of one bit transmission of the sign of difference signal at free running interval with a slight increase of bit rate to 12 kbps.

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The subjective speech-to-noise-ratio (SNR), derived from the forced-choice pair-comparisontest using the psychometric analysis procedure, has well represented overallspeech quality of speech coders in a single dimension. No significant speaker andlistener variation has been found for a wide range of waveform coders at the tests conductedin two separate sessions 14 months apart using different groups of Englishspeakers and listeners. The purpose of this study is to investigate reproducibility of themeasure conducting the same framework test using Japanese speakers and listeners.The test result shows the subjective SNR measure gives quite reliable scores evaluated in different laboratories with different language backgrounds.

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In many fields of digital signal processing such as acoustic echo path estimation, modelingof the unknown systems has been required. Since the selection of the model order isvery important in such modeling, various methods of selection have already been proposed.However, the order estimation of the ARMA model is very difficult due to thenoise of the observed data, even in the case of known input data. Since, in the presenceof additive noise in the output data, traditional methods fail to select a reasonable order, we propose a new algorithm to overcome this problem. Our proposed algorithm forARMA model order selection is based on singular value decomposition (SVD) and newcriteria of threshold values for the smallest singular value of the noisy data matrix. Inthe simulation, we show that our proposed algorithm is very effective for obtaining anappropriate rank of the noisy data matrix. We also show the good performance of our algorithm for practical estimation of the room acoustic transfer function.

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Modern echo integrators, as instruments to quantify biological scatterers in the water column, are capable of making absolute physical measurements. Only in recent years, however, have such measurements begun to be commonplace. It is the aim here to emphasize the absoluteness of echo integrator measurements and to encourage their performance. This is done by (1) defining the area backscattering coefficient, which is a most convenient and simple quantity, and (2) describing the essence of absolute calibration by means of a standard target, which follows an established, routine procedure. The impetus for this study is a recent work by Sawada and Furusawa [J. Acoust. Soc. Jpn.(E) 14, 243-249 (1993)], in which an operational expression is given for the volume backscattering coefficient to be used in a precision calibration.

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As is well-known, in the measurement of low frequency sound, a main spectrum componentof the wind noise coexists almost within the same frequency bands as the lowfrequency sound. Accordingly, it is essentially difficult to eliminate this effect of windnoise only by using the analog filter based on the usual electrical cricuit. From theabove point of view, in this paper, a new trial of removing the above effect of wind noiseare proposed, especially, by focusing the dynamically computer aided algorithm of stateestimation. Concretely, this method is based on the principle of Bayes' filter. Finally, the effectiveness of the proposed state estimation method is experimentally confirmedtoo by applying it to the actual data of low frequency sound signal observed in a free sound field.

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