• 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.

• 한국통신학회논문지
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• 제8권2호
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• pp.43-47
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• 1983

FIR필터와 IIR필터를 각각 독립적으로 설계하여 결합한 혼성 FIR/IIR필터에 의하여 이산신호의 보간을 위한 다상회로를 설계하였다. 제안된 다상회로는 백색 가우시안 잡음을 이용한 평가방법을 통하여 이산신호의 보간 필터로서의 유용성이 확인되었다.

• 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.

• 융합신호처리학회논문지
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• 제8권3호
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• pp.163-170
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• 2007

본 논문에서는 bicubic 방법보다 하드웨어 구조는 간단하면서 확대 이미지는 더욱 선명한 polyphase scaler를 제안하였다. 제안된 시스템은 새로운 픽셀을 얻기 위해서 디지털 필터의 group delay를 이용하여 해상도를 확대하는 방식을 사용하였으며 bicubic 방법과 비교하여 더 좋은 성능을 가지면서 하드웨어 구조를 간단히 하여 모바일 기기 등에 쉽게 적용이 가능하도록 설계하였다. 기존 polyhpase 필터는 해상도 확대시 영상을 흐릿하게 하는 blurring 노이즈가 발생하는 문제점을 가지고 있었다. 그래서 입력 신호의 고주파 성분을 증폭시키도록 Polyphase 필터를 boost-up 필터로 수정하여 보다 선명한 결과를 얻을 수 있었다. 본 논문에서 제안하는 polyphase scaler는 Xilinx Virtex2 FPGA를 이용하여 하드웨어 검증을 수행하였다. 제안된 polyphase scaler는 핸드폰 카메라의 디지털 줌으로 사용되어 질 수 있다.

• ETRI Journal
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• 제25권5호
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• pp.345-355
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• 2003

In this paper, we propose a method for designing a class of M-channel, causal, stable, perfect reconstruction, infinite impulse response (IIR), and parallel uniform discrete Fourier transform (DFT) filter banks. It is based on a previously proposed structure by Martinez et al. [1] for IIR digital filter design for sampling rate reduction. The proposed filter bank has a modular structure and is therefore very well suited for VLSI implementation. Moreover, the current structure is more efficient in terms of computational complexity than the most general IIR DFT filter bank, and this results in a reduced computational complexity by more than 50% in both the critically sampled and oversampled cases. In the polyphase oversampled DFT filter bank case, we get flexible stop-band attenuation, which is also taken care of in the proposed algorithm.

• 융합신호처리학회 학술대회논문집
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• 한국신호처리시스템학회 2000년도 하계종합학술대회논문집
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• pp.165-168
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• 2000

In this paper, the polyphase filter which has good ripple characteristic in the passband is proposed. This filter consists of the digital all-pass filter of parallel structure and it is the half-band filter with all zeros in unit circle. To approach easily in designing hardware, we determined the coefficients to the 16bit 1.15 format. To evaluate the performance of this filter, we analyzed the phase characteristic in each branch and simulated each filter with small coefficients. From the result, we have got to good ripple characteristic and also analyzed the fifth and the seventh, and compared them with four part : ripple, group delay, transition bandwidth, stopband attenuation.

• 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.

• 한국통신학회논문지
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• 제25권6B호
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• pp.1106-1113
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• 2000

본 논문에서는 비정수배 과표본화(fractional ratio oversampling) 구조를 위한 DFT 기반의 폴리페이즈 필터뱅크를 제안하였다. 비정수배 과표본화는 임계표본화(critical sampling)보다 적은 에일리어징을 가지며 동시에 정수배과표본화(integer ratio oversampling)보다 적은 계산량을 갖는다. 또한 비정수배 과표본화의 에일리어징 감소효과로 인해 필터뱅크 설계에 자유도가 높아진다. 제안된 비정수배 과표본화 폴리페이즈 구조를 서브밴드 적응필터에 응용하여 긴 적응필터를 필요로 하는 음향반향 제거 실험을 수행하였다. 반향제거 실험결과 비정수배 과표본화는 정수배 과표본화보다 적은 계산량으로 유사한 성능을 보였다.

• 한국군사과학기술학회지
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• 제22권3호
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• pp.332-340
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• 2019

Electronic warfare systems are needed to be advantageous in the modern war. Many radar threat signals with various frequency spectrums and complicated techniques exist. For detecting the threats, a receiver with wide and narrow-band digital processing is needed. To process a wide-band searching mode, a polyphase filter bank has become the architecture of choice to efficiently detect threats. A polyphase N-path filter aligns the re-sampled time series in each path, and a discrete Fourier transform aligns phase and separates the sub-channel baseband aliases. Multiple threats and CW are detected or rejected when the signals are received in different sub-channels. And also, to process a narrow-band precision mode, a direct down converter is needed to reduce aliasing by using a decimation filter. These digital logics are designed in a FPGA. This paper shows how to design and develop a wide and narrow-band digital receiver that is capable to detect the threats.

• 융합신호처리학회논문지
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• 제7권2호
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• pp.65-72
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• 2006

디지털신호처리에 있어서 신호의 다상분해(polyphase decomposition)는 멀티레이트(multirate) 시스템 구현에 있어서 다양한 용도로 사용된다. 특히 디지털 필터설계 등에 있어서 다채널 신호를 제공함으로써 병열연산을 가능하게 하여 알고리즘의 성능을 개선하는데 특히 유용하다. 일반적으로 다상분해신호는 원신호를 다운샘플링을 위해 데시메이션(decimation)하기 때문에 원신호에 비해 많은 고주파 성분을 포함하여 주파수 대역이 확장되는 경향이 있다. 이러한 성질은 다상분해신호의 활용, 즉 디지털 신호처리 시스템의 구조나 성능에 상당한 제약을 초래한다. 따라서 본 연구에서는 다상분해신호에 대한 완전 대역압축 및 복원 방법을 이론적으로 제시하고 Matlab 시뮬레이션을 통하여 그 유효성을 확인한다.