• The Journal of the Acoustical Society of Korea
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• v.20 no.3
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• pp.8-14
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• 2001

In this paper, we present a new cost function for subband block adaptive algorithm and block conjugate gradient algorithm for noise cancellation of acoustic signal. For the cost function, we process the subband signals with data blocks for each subbands and recombine it a whole data block. After these process, the cost function has a quadratic form in adaptive filter coefficients, it guarantees the convergence of the suggested block conjugate gradient algorithm. And the block conjugate gradient algorithm which minimizes the suggested cost function has better performance than the case of full-band block conjugate gradient algorithm, the computer simulation results of noise cancellation show the efficiency of the suggested algorithm.

• Proceedings of the Optical Society of Korea Conference
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• 2001.02a
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• pp.152-153
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• 2001

파장 가변형 광 필터와 파장 선택형 광 스위치는 파장분할다중 광통신 및 광 교환 시스템을 구현하는데 매우 중요한 소자들 중의 하나이다. 특히 음향광학효과를 이용한 파장 가변형 광 필터(AOTF ： Acousto-Optic Tunable Filter)는 150nm 이상의 넓은 파장 가변범위, 1.5nm이하의 좁은 통과대역폭, 수 $\mu\textrm{s}$ 정도의 비교적 빠른 스위칭 속도 그리고 여러 개의 파장 채널을 동시에 선택할 수 있는 장점들을 가지고 있지만, 한편으로 표면 음향파(SAW： Surface Acoustic Wave) 구동에 필요한 RF 파워와 부 모드 (sidelobe)가 비교적 크다는 단점 때문에 실용화에 많은 제약을 받아왔다. (중략)

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.10
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• pp.1867-1872
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• 2016

In the area of digital signal processing, it is necessary to improve the quality of the speech signal after removing the background noise which exists in a various real environments. The important thing to consider when removing the background noise acoustically is that to solve the problem, depending on the information of the human auditory mechanism is mainly the amplitude spectrum of the speech signal. This paper introduces the characteristics of a frequency-weighted spectrum shaping filter for the extraction of the amplitude spectrum of the speech signal with the primary purpose. Therefore, this paper proposes an algorithm using the methods of a Wiener filter and the frequency-weighted spectrum shaping filter according to the acoustic model, after extracted the amplitude spectral information in the noisy speech signal. The spectral distortion (SD) output of the proposed algorithm is experimentally improved more than 5.28 dB compared to a conventional method.

• 전자공학회논문지 IE
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• v.44 no.4
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• pp.1-10
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• 2007

In this thesis, we study the implementation of hands-free phone in a car, taking acoustic echo canceller, in order to remove acoustic echo effectively. Conventional coustic echo canceller used for only adaptive filtering has much difficulty to solve both echo and double-talk problem. To tackle this problem, we propose acoustic echo canceller consisting of adaptive filter using a modified NLMS, VAD to catch exact voice activity duration using two independent forgetting factors, double-talk detector to detect fast and precise double talk duration using cross-correlation between microphone signal and residual echo, and output controller using VAD and double-talk detector. The proposed hands-free phone taking acoustic echo canceller shows the performance that has not acoustic echo and guarantees full duplex.

• The Journal of the Acoustical Society of Korea
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• v.29 no.6
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• pp.402-409
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• 2010

In this paper, we propose an efficient integrated suppression algorithm based on combined power of acoustic echo and background noise. The proposed method combines the acoustic echo and noise power by the weighting parameter derived from the decision rule based on the estimated echo to noise power ratio. Therefore, in the proposed approach, the acoustic echo and noise signal are able to be reduced through only one suppression filter based on the estimated combined power. The proposed unified structure improves the problems of the residual echo and noise resulted from the conventional unified structure where the noise suppression (NS) operation is placed after the acoustic echo suppression (AES) algorithm or vice versa. The performance of the proposed algorithm is evaluated by the objective test under various environments and yields better results compared with the conventional scheme.

• The Journal of the Acoustical Society of Korea
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• v.22 no.4
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• pp.261-268
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• 2003

Conventional acoustic echo cancellers using ES algorithm have simple structure and fast convergence speed compared with those using NLMS algorithm, but they are very weak to external noise because ES algorithm updates the adaptive filter taps based on average energy reduction rate of room impulse response in specific acoustical condition. To solve this problem, in this paper, a new update algorithm for acoustic echo canceller with stepsize matrix generator is proposed. A set of stepsizes is determined based on residual error energy which is estimated by two moving average operators, and applied to the echo canceller in matrix from, resulting in improved convergence speed. Simulations in various noise condition show that the proposed algorithm improves the robustness of acoustic echo canceller to external noise.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.7A
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• pp.520-528
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• 2003

This paper proposes a method of implementing synthetic stereo and an acoustic echo cancellation algorithm for multiple participant conference system. Synthetic stereo is generated by HRTF and two loudspeakers. A robust adaptive algorithm for synthetic stereo echo cancellation is proposed to reduce the weight misalignment due to near-end speech signals and ambient noises. The proposed adaptive algorithm is modified version of SMAP algorithm and the coefficients of adaptive filter is updated with cross correlation of input and estimation error signal normalized with sum of the autocorrelation of input signal and the power of the estimation error signal multiplied with projection order. This is more robust to projection order and ambient noise than conventional SMAP. Computer simulation show that the proposed algorithm effectively attenuates synthetic stereo acoustic echo.

• The Journal of the Acoustical Society of Korea
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• v.14 no.4
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• pp.64-73
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• 1995

We developed a surface acoustic wave (SAW) computer aided design (CAD) for mobile communication using Kaier window function. The systems are composed of modules for designing apodization weighted IDT-uniform IDT, withdrawal weighted IDT-withdrawal weighted IDT, and resonator type. The design of SAW bandpass with center frequencies from 222MHz to 343MHz were simulated by the developed CAD system. Although the method proposed in this paper is formulated primarily for SAW filters, it is equally applicable to finite impulse response (FIR) digital filter design.

• The Journal of the Acoustical Society of Korea
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• v.8 no.4
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• pp.29-38
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• 1989

Detection of an underwater target by acoustic wave raises various difficulties due to unpredictable noise interference which originates from clutter, reverberation, and variations of medium characteristics with time and location. The SNR and the range resolution of conventional SONAR systems using a matched filter are generally poor, since the latter is optimum only in the additive white noise case. Furthermore, it cannot compensate for variations of the detection level which are responsible for the resultant detection errors. In this paper, the unpredictable interferences are compensated for by using an adaptive filter. It recursively estimates the channel impulse response based on the received echo signal. In the low noise environments, the estimated impulse response is close to the true one, providing a good range resolution, and a matched filter is used subsequently for the purpose of detection. It is shown through computer simulation that good performance can be achieved via the two steps of filtering. Also, the detection level remains unchanged without any additional provisions. Finally, we present the characteristics of the employed adaptive filter parameters.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.11a
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• pp.1037-1041
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• 2007

With the advent of sound field simulator, many sound fields have been reproduced by obtaining the impulse responses of specific acoustic spaces like famous concert hall, opera house. This sound field reproduction has been done by the linear convolution operation between the sound input signal and the impulse response of certain acoustic space. However, the conventional finite impulse response based linear convolution operation always makes real-time implementation of sound field generator impossible due to the large amount of computational burden. This paper introduces the fast convolution method using perceptual redundancy in the processed signals, input audio signal and room impulse response. Temporal and spectral psycho-acoustic filters considering masking effects are implemented in the proposed convolution structure. It reduces the computational burden of convolution methods for realtime implementation of a sound field generator. The conventional convolutions are compared with the proposed one in views of computational burden and sound quality. In the proposed method, a considerable reduction in the computational burden was realized with acceptable changes in sound quality.