• The Journal of the Acoustical Society of Korea
• /
• v.24 no.1
• /
• pp.1-10
• /
• 2005

This paper presents a wideband LPA (local polynomial approximation) beamforming algorithm that is appropriate for wideband moving sources. The Proposed wideband LPA algorithm adopts STMV (steered minimum variance) method that utilizes a steered covariance matrix obtained from multiple frequency components in one data snapshot, instead of multiple data snapshots in one frequency bin. The wideband LPA cost function is formed using STMV weight vector. The Proposed algorithm searches for the instantaneous DOA and angular velocity that maximize the wideband LPA cost function. resulting in a higher resolution performance than that of a DS LPA beamforming algorithm. Several simulations using artificial data and sea trial data are used to demonstrate the performance of the Proposed algorithm.

• The Journal of the Acoustical Society of Korea
• /
• v.21 no.5
• /
• pp.446-454
• /
• 2002

This paper presents the Doppler Scanning technique which enables us to detect the relative positions of moving distributed sources using Doppler frequency shift estimate when the moving source consists of distributed sources with different signature frequencies. Doppler frequency shifts of characteristic frequencies of machinery noise sources such as ship's generator and propeller, with tine along CPA (Closest Point of Approach of moving source) are unique, and can be functioned with respect to each source position. Therefore, this technique can be applied to estimate the relative geometrical positions between machinery noise sources. The Extended Kalman Filter (EKF) which has a high frequency resolution with high time resolution, is adopted for improving accuracy of Doppler frequency shift estimate geometric resolution of machinery positions since machinery noise sources show in general low frequency band characteristics with limited spacial distance. The performance of the technique is examined by the numerical simulations and is verified by the experiment using loudspeaker sources on the roof of the car.

• The Journal of the Acoustical Society of Korea
• /
• v.35 no.2
• /
• pp.125-133
• /
• 2016

In order to obtain high-resolution seafloor images, research on SA (Synthetic Aperture) processing and the development of related underwater systems have been performed in many countries. Recently the SA processing is also recognized as an important technique in Korea and researchers started related basic study. However, most previous studies ignored the Doppler effect by a moving receiver array. In this paper reconstructed SAS (Synthetic Aperture Sonar) images and position errors are analyzed according to the speed of a moving array for understanding its moving effect on the SAS images. In the analysis the spatial frequency domain interpolation algorithm is used. The results show that as the moving speed of the array increases the estimated position error also increases and image distortion gets worse when we do not consider the array motion. However, if the compensated receiver signals considering the array motion are used the position error and image distortion can be eliminated. In conclusion a signal processing scheme which compensates the Doppler effect is necessary especially in the condition where the array speed is over 1 m/s.

• Transactions of the Korean Society of Mechanical Engineers A
• /
• v.29 no.12 s.243
• /
• pp.1597-1604
• /
• 2005

The identification of a moving noise source is important in reducing the source power of the transport systems such as airplanes or high speed trains. However, the direct measurement using a microphone running with noise source is usually difficult due to wind noise, white the source motion distorts the frequency characteristics of the pass-by sound measured at a fixed point. In this study the relationship between the spectra of the source and the pass-by sound signal is analyzed for an acoustic source moving at a constant velocity. Spectrum of the sound signal measured at a fixed point has an integral relationship with the source spectrum. Nevertheless direct conversion of the measured spectrum to the source spectrum is ill-posed due to the singularity of the integral kernel. Alternatively a differential equation approach is proposed, where the source characteristics can be recovered by solving a differential equation relating the source signal to the distorted measurement in time domain. The parameters such as the source speed and the time origin, required beforehand, are also determined only from the frequency-phase relationship using an auxiliary measurement. With the help of the regularization method, the source signal is successfully recovered. The effects of the parameter errors to the estimated frequency characteristics of the source are investigated through numerical simulations.

• The Journal of the Acoustical Society of Korea
• /
• v.36 no.6
• /
• pp.413-418
• /
• 2017

When measuring the radiated noise of an underwater vehicle, range information between acoustic source and receiver is an important evaluating factor, but it cannot use GPS. There is a method of using the cross correlation for finding the range of the acoustic source instead of the GPS. However, this method has heavy computational loads. This paper proposes a fast Fourier transform based method with a relatively small amount of computation to estimate the range of a source. The proposed method estimates Doppler frequencies of CW signals received at multiple receivers by fast Fourier transform and estimates the source range by comparing theoretical Doppler frequencies map previously calculated by a receiver position and source depth information. Simulation and lake trial were performed to verify the performance.

• The Journal of the Acoustical Society of Korea
• /
• v.27 no.1
• /
• pp.40-46
• /
• 2008

Weak target detection and localization in the presence of loud surface ship noise is a critical problem for matched field processing (MFP) in shallow water. For stationary sources, each signal component of received signal can be separated and interference can be suppressed using eigen space analysis schemes. However, source motion, in realistic cases, causes spreading of signal energies in their subspace. In this case, eigenvalues of target and interfere signal components are mixed and hard to be separated with usual phone space eigenvector decomposition (EVD) approaches. Our technique is based on mode space and utilizes the difference in their physical characteristics of surface and submerged sources. Performing EVD for modal cross spectral density matrix, interference components in the mode amplitude subspace can be classified and eliminated. This technique is demonstrated with synthetic data, and results are discussed.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.38C no.5
• /
• pp.434-439
• /
• 2013

It is very difficult to monitor the blind spots that are not recognized by traditional surveillance camera (CCTV) systems, and the surveillance efficiencies are very low though many accidents/events can be solved by the systems. In this paper, the fuzzy logic based sound source localization system using sound strength in the underground parking lot is suggested and the performance of the system is analyzed in order to enhance the stabilization and the accuracy of the localization algorithm in the suggested system. It is confirmed that the localization stabilization of the localization algorithm (SLA_fuzzy) using the fuzzy logic in the suggested system is 4 times higher than that of the conventional localization algorithm (SLA). In addition to this, the localization accuracy of the SLA_fuzzy in the suggested system is 29% higher than that of the SLA.

• Proceedings of the Acoustical Society of Korea Conference
• /
• spring
• /
• pp.53-56
• /
• 2002

본 논문은 2채널 스피커를 사용하여 청취자에게 보다 입체적인 음향 효과를 제시하기 위하여 크로스토크현상을 제거하고 청취자의 보다 자유로운 청취를 위해 최적 청취영역 확대를 위한 실험과 시스템 구현에 관한 것이다. 정면에 위치한 두 스피커로부터 교차경로인 크로스토크를 제거하기 위해 음질의 왜곡을 최소화하는 자유음장 모델을 이용하여 구현한 트랜스오럴 필터 사용하였고 최적 청취영역의 확대를 위해 스피커는 BPF(Band Pass Filter)를 이용하여 저주파와 고주파를 분리하여 각각 재생할 수 있는 스피커를 구성하였으며 저주파 영역은 제외하고 중고주파 영역을 이용하였으며 기존 크로스토크제거 시스템을 사용하여 고정된 한 점의 청취영역에서 좌${\cdot}$우로 5Cm씩 이동하au 100Cm까지 측정한 결과 30Cm, 55Cm, 75Cm, 90Cm, 100Cm에서 크로스토크제거됨을 알 수 있는 음의 분리도가 5dB이상 나타났다. 실험 결과 얻어진 각 지점들로부터 자유음장 모델을 이용하여 트랜스오럴 필터링 하였으며 각각의 간섭현상을 막기 위해 주파수 영역에서 심리음향에 기초한 1/3-Octave Band Pass Filter를 사용하여 음질 보상을 실시하였다. 음원을 제작하여 기존의 2채널 시스템에서 제시하는 음원을 각각의 위치의 음원과 비교하여 음질 평가를 실시하였으며 기존의 트랜스오럴 필터와 비교평가를 실시하였다.

• Journal of the Institute of Convergence Signal Processing
• /
• v.13 no.1
• /
• pp.1-5
• /
• 2012

In this study, we constructed a virtual auditory display(VAD) that enables listener to move in a room freely. The VAD system was installed in a soundproof room($4.7m(W){\times}2.8m(D){\times}3.0m(H)$). The system consisted of a personal computer, a sound presentation device, and a three-dimensional ultrasound sensor system. This system acquires listener's location and position from a three-dimension ultrasonic sensor system covering the entire room. Localization was realized by convolving the sound source with head related transfer functions(HRTFs) on personal computer(PC). The calculated result is generated through a LADOMi(Localization Auditory Display with Opened ear-canal for Mixed Reality). The HRTFs used in the experiment were measured for each listener with loudspeakers constantly 1.5m away from the center of the listener' s head in an anechoic room. To evaluate the system performance, we experimented a search task of a sound source position in the condition that the listener is able to move all around the room freely. As a result, the positioning error of presented sound source was within 30cm in average for all listeners.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 1999.06b
• /
• pp.75-78
• /
• 1999

본 논문에서는 멀티미디어 컨텐트 생성을 위한 입체음향 생성 저작도구 개발에 관하여 논의한다. Windows 95/98, Windows NT 환경의 PC 상에서 동작하는 입체음향 생성 저작도구는 일반적인 음향 편집기가 갖는 편집기능 및 음향 효과 기능 이외에 음상정위 기능, 음장제어 기능의 입체음향 기능을 가지고 있으며 스피커를 통하여 입체음향을 청취할 경우 발생하는 크로스톡크를 제거하는 기능도 가지고 있다. 개발된 저작도구를 이용하여 Mono, Stereo 형태로 저장된 음향파일을 순수한 소프트웨어 만으로 가공하여 바이노럴(Binaural) 형태의 입체음향을 생성하여 가상 음원의 위치 및 이동궤적을 정의할 수 있으며 가상공간이 갖는 공간감을 구현할 수 있다 또한 편리한 사용자 인터페이스 환경을 제공하여 GUI를 통하여 3차원 공간 상의 가상음원의 위치 및 이동 궤적과 가상공간을 사용자가 지정할 수 있도록 하였다 입체음향 생성저작도구는 일반 PC 환경에서 사용자가 가공하고자 하는 임의의 오디오 파일을 입체음향으로 생성할 수 있는 GUI 환경을 제공하며, 저비용으로 효과적인 입체음향 컨텐트를 제작할 수 있도록 함으로써, 게임 및 멀티미디어 컨텐트 제작의 고부가가치화와 입체음향 기술의 산업화에 기여할 것으로 기대된다.