The optimum level of music listened to in the presence of noise is studied and discussed from the point of masked loudness. By carrying out an experiment with five pieces of music, it is revealed that the increase in the optimum level of music is about 3dB against 10dBA noise level increment if the noise level lies between 55dBA and 75dBA. The increment of the optimum music level is less as compared with the above value for a lower noise level than 55dBA, and greater for a higher noise level than 75dBA. Even if the signal is radio news instead of music, this rate of increase seems to be unchanged. This tendency of the consistency of the optimum level may be attributable to the subjects' judgement that they intend to keep the masked loudness of music constant against the change of noise level. Selecting a standard signal, we carried out a simulation test to derive the standard equal loudness contour for the signal against various noise levels. Since these contours well simulate the experimental curves, they can be used as the standard for designing an automatic level controller such as that assembled in the playback system used in the presence of noise.

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In this paper, two kinds of parameter estimation methods for overcoming the obstruction of the background noise co-existing in many actual acoustical or vibrational measurements are studied. As it is very important in an estimation problem for the measurement with the background noise of non-Gaussian type to grasp first the statistical properties of noise from all viewpoints as precise as possible, two kinds of information quantities which enable us to evaluate the distance between two different types of distribution forms or the uncertainty between two random physical quantities (e. g., the input and the output of the acoustical system) are newly introduced as the estimation criteria, and two kinds of sequential estimation algorithms which can be applied to the estimation for the general non-linear acoustical system with the background noise of arbitrary distribution type are explicitly derived. Furthermore, the legitimacy and the effectiveness of these algorithms are confirmed by applying to the actual parameter estimation problem of the reverberation room under the background noise from an acoustical viewpoint.

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The mouth of a speaker was illuminated by a stroboscopic light source every 10ms, and pictures of both frontal and lateral views of the mouth were taken utilizing a special camera in which a long film was driven continuously. Changes in the dimensions of various parts of the mouth were measured in these pictures. The up-and-down movements of the centers of the upper and lower lips, and the lateral movements of the corners of the lips, were also recorded by attaching small metal pellets to the lips at these points and by illuminating them with the stroboscopic light source every 5ms. The frontal projections of the traces of these points were displayed three-dimensionally by adding the time axis.(This graph is called a “labiogram.”) The material used here comprised the 100 Japanese monosyllables and some words consisted of two, three or four moras. They were spoken by a female adult. On the basis of the stroboscopic observations, characteristics of the mouth shape for each kind of the speech sound and coarticulation effects were analyzed. The use of the visual information of the mouth shape to improve lipreading ability of Japanese was also discussed.

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An optical heterodyne method is described for studying the particle displacement of the pulsed acoustic wave in the megahertz frequency range. The acoustic wave falls on a thin plastic film and its motion is measured with the optical heterodyne technique, which allows to resolve local acoustic beam profiles and provides a means to measure the absolute displacement. The acoustic fields from a 2.25MHz flat transducer and a 5MHz focused one are investigated quantitatively. The measured beam patterns agree well with the theoretical ones. The absolute value of the displacement is also discussed. This method is proposed as a calibration procedure for the parameters related to the acoustic fields.

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Several investigators have discussed the problem whether median-plane localization needs binaural disparity cues or not. In order to make it clear, two localization tests are carried out with the localization simulation system using a digital computer. In each test, sound stimuli include either only spectral cues or only binaural disparity cues, and they are presented to subjects in binaural condition. The results show that the localization accuracies are almost the same regardless of the presence of binaural disparity cues, and that most sound images produced by only binaural disparity cues appear at a certain biased direction. Consequently, it is concluded that binaural disparity cues alone cannot accomplish median-plane localization.

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Signal compression method was applied for a measurement of wind effects to a sound attenuation by a barrier. This method uses special computer generated test signal, which can be compressed to a pulse like signal by a numerical phase shift filter. External noise can be reduced and effects of reflected sounds can be eliminated by this method, so it is suitable for a measurement of sound attenuation in noisy air flow conditions. Experiments were performed in rather small test room for short propagation path length. Obtained signals and computer processed waveforms are illustrated with the results of attenuation characteristics for five wind conditions.

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The loudness of impact sounds with wide-band spectrum was investigated, and compared with that of the impact sound with tonal character. The stimuli used in this experiment were synthesized by amplitude modulation of pink noise or white noise. As the results, the effectiveness of physical parameters on the loudness was found to be in the order of peak level, decay time and rise time, but the influence of rise time is smaller than that of difference among individual subjects. These tendencies are similar to the results of the experiment for impact sounds with tonal character. PSE for loudness of impact sounds with wide-band spectrum is influenced first of all by the peak sound pressure level and it does not always follow the energy principle. The mean sound pressure level observed through the circuit with time constant of 5ms and the one of 125ms, which was obtained from the experiment as to the impact sound with tonal character, was found to be a good measure for evaluating the loudness of impact sound with wide-band spectrum, too.

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