• 전자공학회논문지A
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• 제32A권3호
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• pp.10-17
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• 1995

this thesis proposes a new echo canceller that can be used in a fulll-duplex digital subscriber loopmodem. The modem suffers from nonlinear distortion such as transmitted pulse asymmetry, saturation in transformers, and nonlinearity of data converters. The proposed nonlinear echo canceller can compensate the nolinear distortion by using a nonlinear digital filter based on canonical pieceewise-linear (CPWL) function. Numerical results based on computer simulation are given in this paper. It is shown that the convergence characteristics depend on the initial values of weights of linear filters with absoluters and that the nonlinearity in digital-to-analog(D/A) converter can be compensated by a relatively small number linear filters with absoluters. It is also shown that the proposed algorithm has a faster convergence rate in comparison with Voterra algorithm.

• 한국음향학회지
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• 제18권7호
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• pp.23-32
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• 1999

본 논문에서는 IIR 필터의 식 오차 방식 LMS 알고리듬과 VS-LMS 알고리듬의 수렴특성에 대한 통계적인 해석을 수행하였다. 사용된 입력신호가 백색 가우시안이라 가정하고 이들 알고리듬의 평균자승오차와 필터 계수의 평균 및 평균자승 특성에 대한 이론적인 관계식을 유도하였다. 컴퓨터 모의실험에 의하여 이론치와 실험치가 거의 일치함을 보임으로써 수렴 특성 해석 결과가 타당함을 보여주었다.

• 한국정보컨버전스학회논문지
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• 제5권2호
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• pp.101-108
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• 2012

In the present paper, it removed the low frequency noise under 1Hz which get up base wandering from the various noise which is included in ECG signals. It used wavelet filter, FIR filter and Adaptive FIR filter and compared the efficiency of the filter. The set condition of 3 kind filters which are the comparative object is the next contents. Used wavelet case, used generating functions db7 and after decomposition, the low frequency of 8 phases (cA8) replaced at 0. FIR filter case, filter coefficient set 1400, cutoff frequency (${\omega}$) set 0.002. Adaptive FIR filter case, collecting coefficients (${\mu}$) with 0.005. The comparative result from the output wave shape and FT spectrum, wavelet is excellent in base wandering removals compared FIR filter and Adaptive FIR filter. And SNR comparisons, wavelet filter(44.16) is high compare with other two filters(25.19 and 15.94).

• 전기학회논문지
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• 제59권2호
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• pp.476-483
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• 2010

If the data are pre-whitened, then gradient adaptive algorithms which are simpler than higher order statistics algorithms can be used in adaptive blind signal estimation. In this paper, we propose a blind subband affine projection algorithm for multiple-input multiple-output adaptive equalization in the blind environments. All of the adaptive filters in subband affine projection equalization are decomposed to polyphase components, and the coefficients of the decomposed adaptive sub-filters are updated by defining the multiple cost functions. An infinite impulse response filter bank is designed for the data pre-whitening. Pre-whitening procedure through subband filtering can speed up the convergence rate of the algorithm without additional computation. Simulation results are presented showing the proposed algorithm's convergence rate, blind equalization and blind signal separation performances.

• The Journal of the Acoustical Society of Korea
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• 제17권1E호
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• pp.20-29
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• 1998

This paper proposes a new stereophonic acoustic echo canceler with deploying delayless subband adpative filters. Due to the storong correlation between stereo signals, a stereophonic acoustic echo canceler is suffering from the slow convergence and the misalignment for estimating impulse responses corresponding to true echo paths at receiving room. Specially, dual adaptive filters for echo cancellation are significantly affected by the abrupt change of the transmission room environment so that the impariments on voice communication could be experienced. To prevent these performance degradations, this paper proposes a robust subband echo canceler employing pre-processing block so as to enhance the convergence speed and provide the insusceptibility to the environment change at transmission room.

• 한국음향학회지
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• 제12권1호
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• pp.37-46
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• 1993

The adaptive algorithm for the Volterra filter is considered. Owing to its simplicity, the LMS algorithm for adaptive Volterra filter(AVF) is widely used as in linear adaptive filters. However, the convergence speed is unsatisfactory. For improving the convergence speed, the frequency domain LMS second order adaptive Volterra filter(FLMS-AVF) is proposed and analyzed. We show that the time and frequency domain LMS AVF's have the same steady state performance under approprate conditons. Moreover, it can be shown that this algorithm can improve the convergence speed significantly by applying self-orthogonalizing method.

• 융합신호처리학회논문지
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• 제23권3호
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• pp.179-184
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• 2022

본 논문은 탐색 알고리즘 없이 자동으로 모델의 합성곱 필터의 개수를 최적으로 결정할 방법에 대해 연구하고자 한다. 본 논문은 SENet에서 제안한 SE Block을 합성곱 신경망에 연결하고 하단의 학습하지 않는 합성곱 신경망을 연결한 HSE Block을 제안한다. HSE Block 모델에 두 개의 데이터셋을 이용하여 필터의 개수를 3 epoch 당 1개씩 증가시키는 실험과 필터 내의값에 따라 필터의 개수를 증가시키는 실험을 수행하였다. 이 실험을 바탕으로 한 층의 HSE Block이 아닌 다층의 HSE Block으로 모델을 구성하고, 기존의 실험할 때 사용한 데이터셋에 비해 더욱 학습하기 어려운 데이터셋을 사용하여 실험을 진행하였다. 기존보다 학습하기 어려운 데이터셋에 대해 HSE Block의 개수를 2개, 3개, 4개, 5개로 두고 실험을 수행함으로써 HSE Block의 효과를 검증하였다.

• 융합신호처리학회논문지
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• 제4권1호
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• pp.49-55
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• 2003

현대산업사회의 발전에 따라 신호처리 분야 중 디지털필터의 사용은 급격히 증가하고 있으며, 특히 디지털 영상처리, 디지털 음성처리, CATV 및 각종 통신 분야 등에서 카메라의 Detail processor, Y/C separator, Ghost제거 필터, 표준변환기(NTSC-PAL), Noise reducer 등으로 많이 사용되고 있다. 이러한 디지털필터에는 일반적으로 IIR(infinite impulse response)과 FIR(finite impulse response) 필터가 있으며, 본 논문에서는 구현이 용이하고 선형위상특성을 갖는 FIR 디지털필터를 설계하였다. FIR 디지털필터 설계에 있어서 통과대역의 차단주파수 부근에서 깁스(gibbs) 현상에 의해 생긴 리플을 완화하기 위해 window함수를 사용한다. 그러나, 기존의 window는 고정된 값으로 되어 있으므로 설계목적에 적합한 window함수를 선택함에 있어 다소 문제점이 있다. 따라서, 본 논문에서는 설계목적에 따라 서 융통성있게 선택이 가능한 파라메터를 부가한 변형된 Hanning window를 설계하였으며, 타당성을 입증하기 위해 디지털필터를 설계하여 기존의 Hamming, Hanning, Blackman, Kaiser window와 비교하였으며, 판단기준으로 peak side-lobe와 천이특성 등을 사용하였다.

• 전기학회논문지
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• 제62권1호
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• pp.133-139
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• 2013

Convex combination of two adaptive filters is an efficient method to improve adaptive filter performances. In this paper, a subband convex combination method of two adaptive filters for fast convergence rate in the transient state and low steady state error is presented. The cost function of mixing parameter for a subband convex combination is defined, and from this, the coefficient update equation is derived. Steady state analysis is used to prove the stability of the subband convex combination. Some simulation examples in system identification scenario show the validity of the subband convex combination schemes.

• 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.