• 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.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 하계종합학술대회 논문집(4)
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• pp.182-185
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• 2000

In this paper, we present a new adaptive filter structure which is based on polyphase decomposition of the filter to be adapted. This structure uses wavelet transform to acquire transform-domain coefficients of the input signal. With this coefficients RLS algorithm is used for adaptation. Particularly, using the polyphase parallel structure, we can trace the system which has very long impulse response with only increasing the subband, and show that computational savings can be achieved. The proposed structure was applied to system identification for performance estimation and compared with fullband adaptive filter.

• 전기학회논문지
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• 제60권2호
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• pp.423-430
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• 2011

In adaptive filtering, the sparseness of impulse response and input signal characteristics are very important factors of it's performance. This paper presents a subband improved proportionate normalized least square (SIPNLMS) algorithm which combines IPNLMS for impulse response sparseness and subband filtering for prewhitening the input signal. As drawing and combining the advantage of conventional approaches, the proposed algorithm, for impulse responses exhibiting high sparseness, achieve improved convergence speed and tracking ability. Simulation results, using colored signal(AR(4)) and speech input signals, show improved performance compared to fullband structure of existing methods.

• 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.

• 융합신호처리학회논문지
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• 제12권1호
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• pp.20-25
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• 2011

부밴드 구조에서 투사차원이 P인 전밴드 인접투사 적응필터는 최대 군위상 분해와 노블아이덴티티를 적용함으로써 P개의 적응 부필터로 분해된다. 각각의 적용 부필터는 투사차원이 1인 간단한 계수 갱신식을 갖게 된다. 이러한 부밴드 분해기법은 구현관점에서 가장 실용적인 해법 중 하나이다. 많은 응용에서 활용을 위해 MPDSAP 적응 필터의 최적 스텝사이즈 해석이 필요하다. 본 논문은 MPDSAP 적응 필터의 MSE에 대한 개선된 해석 모델을 제안하고 초기 최적 스텝사이즈를 유도한다. 컴퓨터 시뮬레이션을 통해 MSE의 개선된 해석 모텔과 유도된 최적 초기 스텝사이즈에 대해 이론과 실험적 결과 사이의 일치함을 확인하였다.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2003년도 춘계학술대회논문집
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• pp.645-651
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• 2003

The system in which a microphone and a loudspeaker are simultaneously used can cause an echo. The echo is caused by feedback between the output of the loudspeaker and the input of the microphone. The acoustic echo canceller is a device to cancel the echo in a communication system. Its general procedure for cancellation is first estimating the plant response of the feedback path and then eliminating the feedback signal from the input signal. In this paper, full-band and sub-band approaches are compared by using some simulation examples.

• 대한전자공학회논문지SP
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• 제46권6호
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• pp.163-174
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• 2009

부밴드 구조에서 적응필터에 최대 다위상 분해와 노블아이덴티티를 적용함으로써 전밴드 인접투사 알고리즘은 부밴드 인접투사 알고리즘으로 변환된다. 최대 다위상 분해된 부밴드 인접투사 (Maximally Polyphase Decomposed Subband Affine Projection: MPDSAP) 알고리즘은 각 부밴드의 적응 부필터에서 사용되는 투사차원이 1인 부밴드 인접투사 알고리즘의 특별한 형태다. MPDSAP 알고리즘의 계수갱신식은 NLMS 알고리즘과 유사한 형식을 갖기 때문에 실제 구현관점에서 보다 좋은 알고리즘 선택이 될 수 있다. 본 논문은 MPDSAP 알고리즘의 새로운 통계적 해석을 제시한다. 해석적 모델은 정규직교 분해필터를 갖는 부밴드 구조에서 Autoregressive (AR) 입력과 임의의 적응이득에 대해 유도된다. 정규직교 분해필터에 의한 사전 백색화는 AR 입력과 임의의 적응이득에 대한 MPDSAP 알고리즘의 간단한 해석적 모델의 유도를 가능하게 한다.