This paper describes a prototype mechanical white noise generator has a source level of more than 170.0 dB (re 1µPa at 1 m) at the frequency range of 10 Hz to 100 kHz. The results of performance evaluation of the generator are as follows. The average source level of the generator measured by a step of 15°in horizontal (0 to 360。, 25 points) was 185.2 (SD (standard deviation): 2.3) dB (re 1µPa at 1 m). The maximum and minimum source levels were appeared at the frequency range of 2.5 to 5.0 kHz and around 100 kHz, respectively. The average source levels at 0°, 90。, 180。and 270° were 162.9 (SD: 10.6), 168.4 (SD: 10.0), 162.1 (SD: 9.1) and 166.5 (SD: 11.1) dB (re 1µPa at 1 m). The average source level measured by a step of 30° in vertical was 184.9 (SD: 2.2) dB (re 1µPa at 1 m). The relative maximum variation width of the source levels in horizontal and in vertical measurement were less than 7.0 dB and 1.0 dB, respectively.

In this study, we do research in a proper microphone arrangement method by way of evaluating and analyzing the relation between an open angle of the microphones and a time difference of an input sound signal for an exact definition of a sound source. Then, We can expect that existing robots or machines with an only vision system will have more excellent environmental adaptation ability by supports of hearing sense.

고압변압기의 1차측을 제어하고 2차측에 유도된 전압을 이용한 수중방전음원의 전기음향학적 제특성을 분석.검사한 결과는 다음과 같다. 1. 방전시 2차측 전류는 초기에는 Ohm 법칙을 따르다가 전류가 최고 6.3A 흘러 절연 파괴되었으며, 그 순간 방전음이 생성되었다. 2. 전류인가점과 방전음 생성문의 시간차는 약 3ms였으며, 전압이 최고일 때 절연파괴가 일어나 방전음이 생성되었다. 3. 전극의 끝이 뾰족할수록 2차측 전압이 높을수록 음압수준은 높았다. 4. 뾰족한 형태의 전극은 전극간격이 100cm일 때도 방전이 일어났으며 전극간격이 1cm이상부터 비교적 안정된 방전음이 생성되었다. 5. 방전음의 펄스폭은 약 0.15ms인 Shock Wave였으며, 10HKz 이하의 합성저주파 성분이었다.

In this paper, we present a Sound Source Localization (SSL) based GCC (Generalized Cross Correlation)–PHAT (Phase Transform) and new measurement method of angle with robot auditory system for a network-based intelligent service robot. The main goal of this paper is to analysis performance of TDOA and GCC-PHAT sound source localization method and new angle measurement method is compared. We use GCC-PHAT for measuring time delays between several microphones. And sound source location is calculated by using time delays and new measurement method of angle. The robot platform used in this work is wever-R2, which is a network-based intelligent service robot developed at Intelligent Robot Research Division in ETRI.

A purpose of developing a sound source tracking system in this paper is to reduce the noise efficiently from the received signal by microphone array and measure the signal's time delay between the microphones. I have applied the wavelet analysis algorithm to the system and calculated the sound source's relative position For the performance evaluation, I have compared with the results of utilizing the digital filtering methods based on the FIR LPF using Kaiser window function and the inverse Chebyshev IIR LPF. As a result, I have confirmed the fact that 'time-scale' filter using inverse discrete wavelet transform was suitable for this system.

We have reported the new measurement system which was substituted digital filter for the analog filter in order to develop the optimal system that could find the time delay between each sensors with high accuracy. And also we have confirmed through the experiments that the accuracy of measurements were differentiated by the methods what kind of digital filter had been adopted. This paper suggests two algorithms which approximate the sound source's bearing and distance. One is that sound source's relative bearing can be approximately regarded as the gradient of hyperbolic asymptote, the other is that the source's range can be approximated under the condition of a long range source relative to the sensor's interval. And a series of experiments were carried out with the source's distance 22.42meters and the random bearing interval within the limits of -90˚~+90˚. As a result, we have recognized that the approximation methods could measure the bearing and distance with higher accuracy than the method using trigonometric relation could do.

Before some experiments were carried out with analog bandpass filter which used for filtering the noise included in sound source signal. And this filter was constituted by condenser, register and operational amplifier. Hut these elements made the phase characteristics to differentiate in each sensing channel and cause a little of measurement error. We made new measurement system that was substituted digital filter for the analog filter in order to develop the optimal system which could find the time delay between each sensors with high accuracy. This paper describes the new system's constitution and the function of each parts. Specially three digital filters were designed and applied to the digital signal processing Part. And a series of experiments were carried out with the source＇s distance 9.53meters and the random bearing interval within the limits of 0˚ ~ 180˚. As a result, we have recognized that the accuracy of measurements were differentiated by the methods what kind of digital filter were adopted. And we have confirmed the facts that IIR LPF was suitable for sound source's bearing measurement and FIR LPF reduced the range measurement error.

The whistle is a very important information source for the safety navigation under foggy weather. But navigator has no concern about this, because it must be achived by human hearing sense and considered as a vague signal. If the range and relative bearing of signal source can be detected automatically, it would be very useful system for preventing marine casualties making a lot of economic loss and environment pollution. Before the algorithm of 2-dimensional sound source tracking system was reported. This paper describes the method that can obtain the time lag between three signals and the theory of cross-correlation analysis and subtraction method for cauculating the time lag by using the digital signal data sequences. And a series of experiments were carried out for various position of sound source in the range from 200cm to 530cm. As a result, we have recognized that sound source tracking system is possible to the sea field with improvement of position error.

When navigating in or near an area of restricted visibility, it is necessary to be heard the whistle, bell and/or the siren of lighthouses or ships at times. Even though we can get the brief informations about the property of sound, the direction and range of a sound radiator, it is not enough to get the accurate informations for decision making. Generally the audio frequency is known as 16~20, 000Hz, but the earshot is shorten and discrimination of sound is more difficult when there is some noise. The sound pressure is 60dB at the moment when human speaks 1 meter away. Usually the noise pressures are 40dB in a silent room and 60dB on the quiet street, respectively. It this study, the basic algorithm and a method of signal processing are suggested to trace the direction and range of the source radiator using the signals received through not a physical sense but the microphone sensors.