• Journal of the korean Society of Automotive Engineers
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• v.14 no.5
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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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• v.2 no.1
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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.

• Proceedings of the KSR Conference
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• 2010.06a
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• pp.6-12
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• 2010

The source of wayside noise for the train are the aerodynamic noise, wheel/rail noise, and power unit noise. The major source of railway noise is the wheel/rail noise caused by the interaction between the wheels and rails. The Structure borne noise is mainly a low frequency problem. The train noise and vibration nearby the elevated railway make one specific issue. The microphone array method is used to search sound radiation characteristics of elevated structure to predict the noise propagation from an elevated railway. In this paper, the train noise and structure borne noise by train are measured. From the results, we investigated the effect on the sound absorption tunnel for elevated railway.

• Transactions of the Korean Society of Mechanical Engineers B
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• v.26 no.12
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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.

• 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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• 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.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.11a
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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.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.16 no.6 s.111
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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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• v.8 no.3
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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.

• 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.