• 대한전기학회:학술대회논문집
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• 대한전기학회 2001년도 하계학술대회 논문집 D
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• pp.2555-2557
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• 2001

The adaptive IIR filters is more effective than adaptive FIR filters which have the same order of IIR filters. But the IIR filters may have stability problems especially when the adaptive algorithm is not converged. In this paper, a stabilizing procedure for adaptive IIR filters is proposed, and computer simulation is performed to show the effectiveness of proposed schemes.

• 음성과학
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• 제9권4호
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• pp.275-287
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• 2002

In man-machine communication by speech under the noisy environment, the quality of speech may be degraded severely for the machine to recognize correctly. Especially when the corrupting noise occupies the same band as the speech, the conventional fixed filters cannot filter out the noise effectively. In recent, to resolve such a problem adaptive noise canceller (ANC) is frequently used, which is based upon adaptive filters. The Adaptive recursive filters perform better than adaptive nonrecursive filters due to the added poles, but the stability may be severely threatened. In this paper an ANC system employing the adaptive recursive filter is proposed to enhance the speech corrupted by noise. And the stability of the adaptive recursive filter is guaranteed by employing the adaptive compensator.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2005년도 ICCAS
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• pp.1435-1438
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• 2005

In hands-free telephone systems, the received speech signal is fed back to the microphone and constitutes the so-called echo. To cancel the effect of this time-varying echo path, it is necessary to device an adaptive filter between the receiving and the transmitting ends. For a typical FIR realization, the length of the fullband adaptive filter results in high computational complexity and low convergence rate. Consequently, subband adaptive filtering schemes have been proposed to improve the performance. In this work, we use deterministic approach to analyze the relationship between fullband and subband adaptive filtering structures. With block adaptive filtering structure as an intermediate stage, the analysis is divided into two parts. First, to avoid aliasing, it is found that the matrix of block adaptive filters is in the form of pseudocirculant, and the elements of this matrix are the polyphase components of the fullband adaptive filter. Second, to transmit the near-end voice signal faithfully, the analysis and the synthesis filter banks in the subband adaptive filtering structure must form a perfect reconstruction pair. Using polyphase representation, the relationship between the block and the subband adaptive filters is derived.

• 대한음성학회:학술대회논문집
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• 대한음성학회 2002년도 11월 학술대회지
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• pp.179-182
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• 2002

The development of noise robust speech processing systems is becoming increasingly important as speech technology is currently widely applied in real world applications. Recently, to resolve such a noise problem, adaptive noise canceller(ANC) is frequently used, which is based upon adaptive filters. The adaptive recursive filters perform better than adaptive non-recursive filters due to the added poles, but the stability may be severely threatened. But these problems of adaptive recursive filters was solved by ACHARF algorithm. This paper presents a method which combines speaker verification system with ANC(Adaptive Noise Canceller) using the ACHARF algorithm. In the front-end stage, ANC is adopted to suppress the additive noise imposed on the speech signal. The results show that the performance of speaker verification system becomes better than before.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제11권1호
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• pp.552-569
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• 2017

The existing steganalysis method based on 2D Gabor filters can achieve a competitive detection performance for content-adaptive JPEG steganography. However, the feature dimensionality is still high and the time-consuming of feature extraction is relatively large because the optimal selection is not performed for 2D Gabor filters. To solve this problem, a new steganalysis method is proposed for content-adaptive JPEG steganography by selecting the optimal 2D Gabor filters. For the proposed method, the 2D Gabor filters with different parameter settings are generated first. Then, the feature is extracted by each 2D Gabor filter and the corresponding detection accuracy is used as the measure for filter selection. Next, some 2D Gabor filters are selected by a greedy strategy and the steganalysis feature is extracted by the selected filters. Last, the ensemble classifier is used to assemble the proposed steganalysis feature as well as the final steganalyzer. The experimental results show that the steganalysis feature extracted by the selected optimal 2D Gabor filters also can achieve a competitive detection performance while the feature dimensionality is reduced greatly.

• 대한전기학회논문지
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• 제36권9호
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• pp.685-690
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• 1987

Adaptive design of IIR digiral filters using equation error method has been studied. In this paper, a design technique of IIR digital filters using output error method with adaptive compensator is presented. In computer simulation results, it is shown that flat response characteristic in pass-band, below-40[dB] attenuation characteristic in stop-band, sharf cut-off characteristic in transition-band, and phase characteristic is linearin pass-band.

• 한국멀티미디어학회논문지
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• 제2권1호
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• pp.30-37
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• 1999

광대역 능동소음제거는 수 백개의 적응필터 텝 수를 갖는다. 탭 수가 긴 적응필터는 많은 계산량이 요구된다. 본 논문에서는 적응 계수벡터가 서브밴드로 계산되는 M-채널 QMF 펼터뱅크를 이용한 능동소음제거 시스템올 제안한다. 분해 필터뱅크와 합성 펼터뱅크는 cosine-modulated pseudo QMF 펼터를 사용한다. 오차경로의 전달특성을 온라인 인식하기 위한 기준신호는 적응필터의 출력신호와 저주파 대역의 서브 밴드 출력신호와의 차신호를 사용한다. 따라서 본 논문의 능동소음제거기는 계산량이 적고, 수렴속도가 빠른 견실한 시스템이 되도록 제안한다.

• The Journal of the Acoustical Society of Korea
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• 제17권2E호
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• pp.3-9
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• 1998

Since the introduction of hands-free phone set and teleconferencing system, acoustic echo cancellation has been a challenge for engineers. Recently many researches have shown that the best solution for the acoustic echo compensation problem is represented by an adaptive filter which iteratively tries to identify the unknown impulse response of the system from loudspeaker to microphone. In this paper, we apply the delayless subband adaptive filters and fast affine projection algorithm for the identification of room impulse response. Simulation results show 3∼8 dB more enhanced performance than conventional fullband adaptive filters or subband adaptive filters. In addition, fast affine projection algorithm shows better convergence speed at the expense of the low computational complexity than conventional LMS algorithm.

• 대한전기학회논문지:시스템및제어부문D
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• 제54권2호
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• pp.104-115
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• 2005

Median filters in the signal processing have been most widely used and have demonstrated the strongest effects. This paper proposes the adaptive median filters with noise detection. The proposed basic algorithm of the filters is to judge whether or not the noises exist on the ground of The Noise Judgment Standards. Just in case the existence of the noises is verified by the algorithm, it takes the median filter. In order to judge the existence of the noises by the algorithm, this paper introduced the noise detection method by local central variance. As a result of comparing and analyzing the features and performance of the proposed filters and the existing [5]-[10] filters on the same conditions, it was verified that the former proved to be better than the latter, Observed even by naked eyes, it was similar, too. Accordingly, it's proved that the adaptive median filters by local central variance are useful in removing the impulse noise of the median filter and reinforce the edge preservation ability.

• ETRI Journal
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• 제42권1호
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• pp.26-35
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• 2020

Herein, we consider uplink multiuser massive multiple-input multiple-output systems when multiple users transmit information symbols to a base station (BS) by applying simple space-time block coding (STBC). At the BS receiver, two detection filters for each user are used to detect the STBC information symbols. One of these filters is for odd-indexed symbols and the other for even-indexed symbols. Using constrained output variance metric minimization, we first derive a special relation between the closed-form optimal solutions for the two detection filters. Then, using the derived special relation, we propose a new blind adaptive algorithm for implementing the minimum output variance-based optimal filters. In the proposed adaptive algorithm, filter weight vectors are updated only in the region satisfying the special relation. Through a theoretical analysis of the convergence speed and a computer simulation, we demonstrate that the proposed scheme exhibits faster convergence speed and lower steady-state bit error rate than the conventional scheme.