• Transactions of the Korean Society of Mechanical Engineers B
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• v.29 no.10 s.241
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• pp.1119-1129
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• 2005

A new measurement system of wall pressure fluctuations on a rotating machinery, composed of digital recording device, was developed and evaluated. The small-sized digital recording device was attached on the rotating machinery and then was detached for data reduction. In order to obtain the system transfer function of the digital recording system, a dynamic calibration was performed utilizing the signal from a 1/8 inch B&K microphone as input. The time history of the unsteady pressure was then reconstructed from the output of the sensor by using this transfer function. The reconstructed pressure signals showed good agreement with the reference signal in both temporal and spectral sense. This sensor was then used to measure the wall pressure fluctuations on a rotating blade. An array of microphones were installed on the blade in the circumferential and radial directions. Various statistical moments were obtained from the measurement data set. Comparison of these quantities with the existing studies demonstrated satisfactory agreement. These tests give credence to the relevance and reliability of this device for applications in more complicated turbulent rotating machineries.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.751-755
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• 2005

We propose a sound source localization method using the Head-Related-Transfer-Function (HRTF) to be implemented in a robot platform. In conventional localization methods, the location of a sound source is estimated from the time delays of wave fronts arriving in each microphone standing in an array formation in free-field. In case of a human head this corresponds to Interaural-Time-Delay (ITD) which is simply the time delay of incoming sound waves between the two ears. Although ITD is an excellent sound cue in stimulating a lateral perception on the horizontal plane, confusion is often raised when tracking the sound location from ITD alone because each sound source and its mirror image about the interaural axis share the same ITD. On the other hand, HRTFs associated with a dummy head microphone system or a robot platform with several microphones contain not only the information regarding proper time delays but also phase and magnitude distortions due to diffraction and scattering by the shading object such as the head and body of the platform. As a result, a set of HRTFs for any given platform provides a substantial amount of information as to the whereabouts of the source once proper analysis can be performed. In this study, we introduce new phase and magnitude criteria to be satisfied by a set of output signals from the microphones in order to find the sound source location in accordance with the HRTF database empirically obtained in an anechoic chamber with the given platform. The suggested method is verified through an experiment in a household environment and compared against the conventional method in performance.

• Journal of Institute of Control, Robotics and Systems
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• v.17 no.3
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• pp.211-217
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• 2011

This paper proposes a quasi-optimal linear DOA (Direction-of-Arrival) estimator which is necessary for the development of a real-time robot auditory system tracking moving acoustic source. It is well known that the use of conventional nonlinear filtering schemes may result in the severe performance degradation of DOA estimation and not be preferable for real-time implementation. These are mainly due to the inherent nonlinearity of the acoustic signal model used for DOA estimation. This motivates us to consider a new uncertain linear acoustic signal model based on the linear prediction relation of a noisy sinusoid. Using the suggested measurement model, it is shown that the resultant DOA estimation problem is cast into the NCRKF (Non-Conservative Robust Kalman Filtering) problem [12]. NCRKF-based DOA estimator provides reliable DOA estimates of a fast moving acoustic source in spite of using the noise-corrupted measurement matrix in the filter recursion and, as well, it is suitable for real-time implementation because of its linear recursive filter structure. The computational efficiency and DOA estimation performance of the proposed method are evaluated through the computer simulations.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.185-188
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• 2000

본 연구에서는 화자의 음성신호를 이용하여 추출된 공간정보를 통해 화자의 위치를 실시간으로 추적하는 시스템을 제안하고 실시간 구현하였다. 기존의 대표적인 화자 위치 추출 알고리듬인 CPSP(Cross Power Spectrum Phase)는 실내환경에서 심각하게 일어나는 반향신호에 취약한 단점을 갖고 있으므로 구현된 시스템에서는 위치 추적 성능 개선을 위하여 반향신호에 강인한 hybrid window 알고리듬을 제안하여 적용하였다. Hybrid window 알고리듬은 실내 환경에 적합한 hybrid window를 설계하여 수신된 음성신호에 적용함으로써 반향신호에 의한 상호 상관관계를 줄이고 직접 경로에 의한 신호들의 상관관계를 높임으로써 보다 정확한 시간 지연 추정을 가능하게 한다. 제안된 시스템의 성능분석을 위해 DSP를 이용해 실시간 구현된 하드웨어를 이용해 기존의 CPSP 알고리듬과 제안된 hybrid window를 적용한 시스템을 실제 환경에서의 실험하였고 제안한 알고리듬을 적용한 시스템이 $22\%$ 이상 성공적으로 화자의 위치를 추적하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.213-216
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• 2000

본 연구에서는 마이크로폰 어레이를 이용하여 화자의 음성신호로부터 화자의 위치를 추정하는 기존의 대표적인 알고리듬인 CPSP(Cross Power Spectrum Phase)로부터 보다 반향에 강인한 알고리듬인 저주파 위상 복원 알고리듬을 제안한다. CPSP 함수는 상호 상관관계(Cross Correlation)가 정규화 되어있는 형태를 갖는데, CPSP 함수의 최대 값 인덱스로부터 화자의 공간정보인 TDOA(Time Difference Of Arrival)를 추출한다. 그러나 CPSP 함수를 이용한 공간정보 추정 알고리듬은 실내환경에서 심각하게 일어나는 반향신호에 대해서 취약한 단점을 갖고 있다. 본 논문에서 제안하는 저주파 위상복원 알고리듬은 주파수 측면에서 반향신호가 CPSP 함수에 미치는 영향을 분석하여 반향으로 인하여 왜곡된 위상 성분을 복원함으로써 보다 신뢰도 있는 TDOA 추정을 가능하게 한다. 반향신호로 인한 CPSP의 위상은 저주파보다 고주파에서 심하게 왜곡되는데, 각각의 반향신호의 도달 시간을 기하학적 분포를 갖는 확률변수로 모델링하여 이를 수학적으로 증명하였다. 또한 실제 환경에서 채집한 음성신호를 이용한 모의 실험을 통해 개선된 알고리듬의 성능 개선을 확인하였다.

• Journal of the Korean Society for Railway
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• v.15 no.3
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• pp.231-236
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• 2012

In this paper, aerodynamic noise characteristics of high-speed trains were deduced from the beamforming method. First, pass-by noise of high-speed trains was measured by a microphone array system. This measurement suggested that the majority of the aerodynamic noise produced came from the bogie area, the pantograph and its cover, and inter-coach gaps. Then, beampower outputs of a position in high-speed trains were obtained from the beamforming method. By Fourier transform, sound characteristics of the position in the frequency domain were deduced from the beamforming power spectrum. From this study, aerodynamic noise characteristics from the major sources of high-speed trains were drawn.

• The Journal of the Acoustical Society of Korea
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• v.11 no.3
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• pp.53-60
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• 1992

Many adaptive beamforming methods have been studied for interference cancellation and speech signal enhancement in telephone conference and auditorium. Main aspect of adaptive beamforming methods for speech signal processing is different from radar, sonar and seismic signal processing because desire output signal should be apt to the human ear. Considering that phase of speech is quite insensible to the human ear, Sondhi proposed a nonlinear constrained optimization technique whose constraint was on the magnitude transfer function from the source to the output. In real environment the phase response of the speech signal affects the human auditorium system. So it is desirable to design linear phase system. In this paper, linear phase beamformer is proposed and sample processing algorithm is also proposed for real time consideration Simulation results show that the proposed algorithm yields more consistent beam patterns and deep nulls to the noise direction than Sondhi's.

• Journal of IKEEE
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• v.23 no.4
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• pp.1360-1365
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• 2019

In this paper, we describe a sound source localization(SSL) system which can be applied to mobile robot and automatic control systems. Usually the SSL method finds the Interaural Time Difference, the Interaural Level Difference, and uses the geometrical principle of microphone array. But here we proposed another approach based on the deep neural network to obtain the horizontal directional angle(azimuth) of the sound source. We pick up the sound source signals from the two microphones attached symmetrically on both sides of the robot to imitate the human ears. Here, we use difference of spectral distributions of sounds obtained from two microphones to train the network. We train the network with the data obtained at the multiples of 10 degrees and test with several data obtained at the random degrees. The result shows quite promising validity of our approach.

• The Journal of the Acoustical Society of Korea
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• v.35 no.2
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• pp.102-109
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• 2016

This paper proposes a noise suppression method through SNR (Singal-to Noise Ratio) estimation in the two microphone array environment of close spacing. The conventional method uses a noise suppression method for a gain function obtained through the SNR estimation based on coherence function from full band. However, this method cause performance decreased by the noise damage that affects all the feature vector component. So, we propose a noise suppression method that allocates a frequency domain signal into N constant multi frequency band and each frequency band gets a gain function through SNR estimation based on coherence function. Performance evaluation of the proposed method is shown by comparison with PESQ (Perceptual Evaluation of Speech Quality) value which is an objective quality evaluation method provided by the ITU-T (International Telecommunications Union Telecommunication).

• The Journal of the Acoustical Society of Korea
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• v.38 no.4
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• pp.406-413
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• 2019

This paper proposes a speech enhancement method through the process of combining the gain function with spectrum subtraction method in the two microphone array with close spacing. A speech enhancement method that uses a gain function estimated by the SNR (Signal-to Noise Ratio) based on the multi frequency band coherence function causes the performance degradation in high correlation between input noises of two channels. A new speech enhancement method is proposed where the weighted gain function is used by combining the gain function from the spectral subtraction. The performance evaluation of the proposed method was shown by comparison with PESQ (Perceptual Evaluation of Speech Quality) value which is an objective quality evaluation test provided by the ITU-T (International Telecommunications Union Telecommunication). In the PESQ tests, the maximum 0.217 of PESQ value is improved in the various background noise environments.