• 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.

• The Journal of the Acoustical Society of Korea
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• 제16권1E호
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• pp.41-48
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• 1997

The very purpose of subband codec is the attainment of data rate compression through the use of quantizer and optimum bit allocation for each decimated signal. Yet the question of floating-point quantization effects in subband codec has received scant attention. There has been no direct focus on the analysis of quantization errors, nor on design with quantization errors embedded explicitly in the criterion. This paper provides a rigorous theory for the modelling, analysis and optimum design of the general M-band subband codec in the presence of the floating-point quantization noise. The floating-point quantizers are embedded into the codec structure by its equivalent multiplicative noise model. We then decompose the analysis and synthesis subband filter banks of the codec into the polyphase form and construct an equivalent time-invariant structure to compute exact expression for the mean square quantization error in the reconstructed an equivalent time-invariant structure to compute exact expression for the mean square quantization error in the reconstructed output. The optimum design criteria of the subband codec is given to the design of the analysis/synthesis filter bank and the floating-point quantizer to minimize the output mean square error. Specific optimum design examples are developed with two types of filter of filter banks-orthonormal and biorthogonal filter bank, along with their perpormance analysis.

• 한국음향학회지
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• 제23권3호
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• pp.221-227
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• 2004

본 논문은 부밴드 인접투사 알고리즘 (SAPA)을 소개한다. SAPA의 개선된 성능은 인접투사 알고리즘에 부밴드 구조를 적용함으로써 가능하다. 본 알고리즘에서 적응 필터의 계수 갱신식은 부밴드 필터링을 위한 분해 필터뱅크로서 OQF (Orthogonal Quadrature Filter)를 사용함으로써 간단하게 유도된다. 유도된 SAPA는 빠른 수렴속도와 적은 계산량을 갖는다. 유색 입력신호에 대한 제안한 알고리즘의 효과를 실험을 통해 평가하였다.

• 대한의용생체공학회:학술대회논문집
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• 대한의용생체공학회 1995년도 춘계학술대회
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• pp.235-238
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• 1995

In this paper, we propose an effective coding method of the angiography by using the scalable structure in the frequency domain for MPACS(Medical Picture Archiving and Communication System). We employed the subband decomposition method and MPEG-2 system which is the international standard coding method of the general moving picture. After the subband decomposition is applied to split an input image into 4 bands in the spatial frequency domain, the motion compensated DPCM coding method of MPEG-2 is carried out for each subband. As a result, an easily controllable coding Structure is accomplished by composing the compound hit stream for each subband group. Follows are the simulation results of the proposed sheme for the angiography. A scalable structure which can be easily controlled for a loss of transmission or the band limit can be accomplisbed in the MPEG-2 stucture by the subband decomposition minimizing the side information. And by reducing the search area of the motion vector between -4 and 3, the processing speed of a codec is enhanced by more than two times without a loss of the picture quality compare with the conventional DCT coefficients decompositon method. And the processing speed is considerably improved in the case of the parallel construction of each subband in the hardware.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 1998년도 하계종합학술대회논문집
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• pp.677-680
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• 1998

In this paper, we proposed a lwo bit rate subband coding with adaptive vector quantization using the correlation between motion vector and block energy in subband. In this method, the difference between the input signal and the motion compensated interframe prediction signal is decomposed into several narrow bands using quadrature mirror filter (QMF) structure. The subband signals are then quantized by adaptive vector quantizers. In the codebook generating process, each classified region closer to the block value in the same region after the classification of region by the magnitude of motion vector and the variance values of subband block. Because codebook is genrated considering energy distribution of each region classified by motion vector and variance of subband block, this technique gives a very good visual quality at low bit rate coding.

• 한국통신학회논문지
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• 제21권5호
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• pp.1096-1108
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• 1996

In this paper, we propose a temporal adaptive tranform 3D SBC coder with motion compensation, exploiting redundancy in the temporal domain. We propose a temporal adaptivity measure, by which the R-D optimal temporal transform can be chaosen. The base temporal subband frame is coded using H.261-like MC-DCT coder, while the higher temporal subband frames are coded using the 2D adaptive wavelet packet bases, considering the various energy distribution which results from the temporal variation. In encoding the subbands, we employ adaptive scanning methods, uniform step-size quantization with VLC, and coded/not-coded flag reduction technique using the quadtree structure. From the simulation results, the proposed adaptive 3D subband coder shows about 0.29~3.14 dB gain over the H.261 and the fixed 3D subband coder techniques.

• The Journal of the Acoustical Society of Korea
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• 제15권2E호
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• pp.24-32
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• 1996

A rigorous theory for modeling, analysis, optimum nonseparable filter bank in multidimensional M-band quantized subband codec are developed in this paper. Each pdf-optimized quantizer is modeled by a nonlinear gain-plus-additive uncorrelated noise and embedded into the subband structure. We then decompose the analysis/synthesis filter banks into their polyphase components and shift the down-and up-samplers to the right and left of the analysis/synthesis polyphase matrices respectively. Focusing on the slow clock rate signal between the samplers, we derive the exact expression for the output mean square quantization error by using spatial-invariant analysis. We show that this error can be represented by two uncorrelated components : a distortion component due to the quantizer gain, and a random noise component due to fictitious uncorrelated noise at the uantizer. This mean square error is then minimized subject to perfect reconstruction (PR) constraints and the total bit allocation for the entire filter bank. The algorithm gives filter coefficients and subband bit allocations. Numerical design example for the optimum nonseparable orthonormal filter bank is given with a quincunx subsampling lattice.

• 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.

• 한국정보통신학회:학술대회논문집
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• 한국해양정보통신학회 2009년도 춘계학술대회
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• pp.883-887
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• 2009

본 논문은 AR(P) 입력에 대해 MPDSAP 알고리즘의 적응과정의 MSE의 통계적 분석을 제시한다. 부밴드 구조에서 인접투사 알고리즘은 적응필터에 다위상 분해와 노블아이덴티티를 적용함으로써 NLMS 알고리즘으로 변환된다. 또한, P차의 Autoregressive(AR) 입력은 정규직교 분해필터에 의해 사전에 백색화 될 수 있다. 부밴드 구조에서 AR(P) 입력의 사전-백색화(pre-whitening)는 SAP 적응필터의 MSE 행동의 통계적 해석을 위한 간단하고 유효한 근사화를 제공한다.

• 한국음향학회지
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• 제20권1호
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• pp.13-20
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• 2001

음향 에코우제거기나 소음제어와 같은 임펄스 응답이 긴 디지털 필터를 이용하여 필터링을 할 경우 수렴속도가 느리고 계산시간이 많이 걸린다. 이러한 기존의 필터링에서 생기는 계산시간이나 수렴속도 문제를 개선하기 위해서 서브밴드 필터링과 멀티레이트 신호처리 기술이 개발되었다. 모든 시스템의 전달함수는 interpolator와 임펄스 응답사이에 임의 수만큼의 0이 들어있는 sparse 임펄스 응답을 갖는 서브필터를 직렬로 연결한 구조로 표현할 수 있다. 이 경우에 interpolator는 Hadamard 행렬로 표현되고 저역통과필터 특성을 갖는 원형필터를 균일하게 이동시킨 것과 같다. 그래서 입력신호를 Hadamard 변환을 이용하여 각 서브대역으로 분할하고 decimation을 하여 샘플링 레이트를 줄이는 멀티레이트기술이 음향 함수 모델링이나 잡음제거에 응용할 수 있다. 본 논문에서는 decimation으로 생기는 에리어싱을 제거하고 수렴속도를 향상시키기 위해서 입력 신호를 트리구조를 갖는 필터뱅크를 이용하여 비균일한 서브대역으로 분할, 그리고 decimation을 하여 샘플링레이트를 변환하고 각 서브대역에서 계수를 갱신한 후 이 계수를 전대역으로 Hadamard 변환을 이용하여 변환하는 비균일한 대역폭을 갖는 delayless 필터 구조를 제안하고 이 구조를 컴퓨터 시뮬레이션을 통하여 성능을 검증한다.