• 한국소음진동공학회논문집
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• 제14권12호
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• pp.1338-1346
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• 2004

The positions of rotating sound sources have been localized by experiments with the Doppler effects removed. In order to de-Dopplerize the sound signals emitted from moving sources, two kinds of signal reconstruction methods were applied. One is the forward propagation method and the other is the backward propagation method. Forward propagation method analyze the source emission time based on the instantaneous distance between sensors and the assumed source position, then the signals are reconstructed with respect to the emission time. On the other hand, the backward method uses time delay to do-Dopplerize the acquired data for the received time of reference. In both techniques. the reconstructed signal data were processed using beamforming algorithm to produce power distributions at the frequencies of interest. Experiments have been carried out for varying frequencies, rotating speeds and the object distances. It is shown that the forward propagation method gives better performance in locating source position than the backward propagation method.

• 오토저널
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• 제14권5호
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• pp.95-106
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• 1992

Active control of acoustic noise is an application area of adaptive digital signal processing with increasingly interest along the last year. This work studies the implementation of the multichannel LMS filter and the application of this algorithm for the reduction of the noise inside a vechicle cabin using a number of 'secondary sources' drived by adaptive filtering of a reference noise source. Firstly, we propose the use of an adaptive method for the time-varient optimal convergence factor. Secondly, we propose the use of adaptive delayed inverse model to estimate the elastic-acoustic transfer function presented in vechicle cabin. The original, primary source is often periodic, with a known fundamental frequency. A suitably filtered reference signal can thus be used to drive the secondary sources. An algorithm is presented for adapting the coefficients of an FIR filter feeding such a secondary source in such a way as to minimize the output of a suitably placed microphone. In this algorithm, the coefficients of adaptive filter driving an array of secondary sources can be adapted to minimize the sum of the squares of the outputs of a number of error microphones. The multichannel LMS algorithm displays that such an algorithm is considered suitable to used for the global suppression of noise in vehicle cabin.

• IEIE Transactions on Smart Processing and Computing
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• 제2권1호
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• pp.11-19
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• 2013

A video-zoom driven audio-zoom algorithm is proposed to provide audio zooming effects according to the degree of video-zoom. The proposed algorithm is designed based on a super-directive beamformer operating with a 4-channel microphone array in conjunction with a soft masking process that uses the phase differences between microphones. The audio-zoom processed signal is obtained by multiplying the audio gain derived from the video-zoom level by the masked signal. The proposed algorithm is then implemented on a portable digital imaging device with a clock speed of 600 MHz after different levels of optimization, such as algorithmic level, C-code and memory optimization. As a result, the processing time of the proposed audio-zoom algorithm occupies 14.6% or less of the clock speed of the device. The performance evaluation conducted in a semi-anechoic chamber shows that the signals from the front direction can be amplified by approximately 10 dB compared to the other directions.

• 한국철도학회:학술대회논문집
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• 한국철도학회 2010년도 춘계학술대회 논문집
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• pp.6-12
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• 2010

철도소음은 공기역학적 소음, 차륜과 레일의 상호작용에 의한 전동소음과 고가구조음으로 나눌 수 있다. 열차의 속도한계에 의해 재래철도의 주소음원은 전동음이라고 알려져 있으며, 전동음에 대한 많은 연구가 이루어졌다. 또한 전동음을 주소음원으로 하여 철도소음에 대한 예측모델을 제시하고, 철도소음의 전파특징에 대한 연구가 다수 이루어졌다. 또한 근래에는 고가선로에 대한 고가진동음에 대한 연구가 진행되고 있다. 한편, 본 연구에서는 국내 고가철교에 거의 설치되어 있지 않은 방음터널에 대한 효과를 파악하고 앞으로 설치가 예상되는 고가철교 방음터널에 대한 설계 자료를 확보하기 위한 기초연구를 수행하였다.

• 대한기계학회논문집B
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• 제26권12호
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• pp.1674-1680
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• 2002

An experimental study was made of a large-scale vortical structure over a backward-facing step. The Reynolds number based on the step height was R $e_{H}$ =33,000. To recognize the large-scale vortex, three components of velocity were measured. The measurements were performed in the recirculation zone (x/H=4.0) and the reattachment zone(x/H=7.5). To measure the wall pressure fluctuations in a turbulent flow over a backward-facing step, a 32-channel microphone array was installed beneath the wall in the streamwise and spanwise directions. From the measured pressure field, the size of large-scale vortex was obtained. As a detailed study, a conditionally-averaging technique was employed to characterize the coherent structure of the large-scale vortex. To see the relationship between the flow field and the relevant spatial mode of the pressure field, the spatial box filtering (SBF) was examined. A cross-correlation between velocity and pressure fluctuations was performed to identify the structure and the length scale of the large-scale vortex.x.

• 제어로봇시스템학회논문지
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• 제17권3호
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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.

• 한국음향학회:학술대회논문집
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• 한국음향학회 2000년도 학술발표대회 논문집 제19권 2호
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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.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2005년도 추계학술대회논문집
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• pp.200-204
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• 2005

The use of beamforming method and de-Dopplerization technique was applied in studying the rotating sound sources. Acoustic analysis of a moving sound source required that the measured sound signals be do-Dopplerized and restored as of the original emission signals. Two main issues of the signal reconstruction in time domain are addressed herein: First, to remove Doppler effect from the measured data and to restore the original emission data of the moving source. The difference of the time domain beamforming from the frequency domain beamforming was mentioned. Also, the time domain beamforming method is deployed in the test and the comparisons were made to the frequency domain results. The time domain signal reconstruction was numerically simulated prior to the application. To validate the de-Dopplerization Performance, the rotating Point sources were examined and localized by the use of a phased array of microphone. The application of prop-rotor was conducted in a hovering condition. The results of reconstructing time signals of rotating sources and its locations were shown in the power distribution maps. In the prop-rotor measurements, the acoustic source locations were successfully verified in varying positions for different frequencies of interest.

• 한국소음진동공학회논문집
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• 제16권6호
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• pp.565-573
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• 2006

Acoustic analysis of a moving sound source required that the measured sound signals be do-Dopplerized and restored as of the original emission signals. The purpose of this research is development of beamforming technique can be applied to the rotor noise source identification. For the do-Dopplerization and reconstruction of emitted sound wave, Forward Propagation Method is applied to the time domain beamforming technique. And validation test were performed using rotating sound source constructed by bended pipe and horn driver. In the validation test using sinusoidal sound wave, sufficient performance of signal processing can be seen, and the effect of measuring duration for accuracy was compared. In the prop-rotor measurements, the acoustic source locations were successfully verified in varying positions for different frequencies and collective pitch angle, in hover condition.

• Journal of Ship and Ocean Technology
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• 제8권3호
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• pp.18-25
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• 2004

Sound reception system is required to detect the sound and the quadrantal direction of the other ship's horn sound, to overcome the effects of enclosed wall for navigation space, functioning as a sound barrier. However, the realized systems can only provide quadrantal information of the other ship. This paper presents a new arrangement of microphones, having geometrically symmetric deployment with the same distances between sensors and the same angles between adjacent sensors with respect to the geometrical center. The sound pressures received at microphones are transformed into the related envelope signals by applying Hilbert transform. The time delays between microphones are estimated by the correlation functions between the derived envelope signals. This envelope base processing mitigates the noises related to the reflection by ship and sea surface. Then, the directional information is easily defined by using the estimated time delays. The suggested method is verified by the generated signals using boundary element method for a small ship model with sea surface wave. The estimated direction is quite similar to the true one and therefore the proposed approach can be used as an efficient sound reception system.