• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.13-20
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• 2001

Adaptive digital filters with long impulse response such as acoustic echo canceller and active noise controller suffer from slow convergence and computational burden. Subband techniques and multirate signal processing have been recently developed to improve the problem of computational complexity and slow convergence in conventional adaptive filter. Any FIR transfer function can be realized as a serial connection of interpolators followed by subfilters with a sparse impulse response. In this case, each interpolator which is related to the column vector of Hadamard matrix has band-pass magnitude response characteristics shifted uniformly. Subband technique using Hadamard transform and decimation of subband signal to reduce sampling rate are adapted to system modeling and acoustic noise cancellation In this paper, delayless subband structure with nonuniform bandwidth has been proposed to improve the performance of the convergence speed without aliasing due to decimation, where input signal is split into subband one using tree-structured filter bank, and the subband signal is decimated by a decimator to reduce the sampling rate in each channel, then subfilter with sparse impulse response is transformed to full band adaptive filter coefficient using Hadamard transform. It is shown by computer simulations that the proposed method can be adapted to general adaptive filtering.

• 전기의세계
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• v.45 no.6
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• pp.9-14
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• 1996

웨이브렛 변환은 1910년의 Haar웨이브렛[2]으로 거슬러 올라가는 수학적 배경을 가짐과 동시에, multirate-filter bank라는 신호처리(signal processing)를 이용한다. 웨이브렛 변환은 많은 신호처리에 사용될 수 있는 매우 일반적인 방법을 제공한다. 1절의 나머지 부분에서는 웨이브렛 변환이 나타나게 된 배경, 2절은 웨이브렛 변환은 수식적 정의, 3절은 웨이브렛의 응용분야, 4절은 발전된 웨이브렛의 모습에 대해서 기술한다.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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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 Transactions of the Korean Institute of Electrical Engineers D
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• v.54 no.10
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• pp.630-638
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• 2005

In FIR (Finite Impulse Response) filter applications, Nth-band F]U digital filters are known to be important due to their reduced computational requirements. The conventional methods for designing F]U filters use iterative approaches such as the well-known Parks-Mcclellan algorithm. The Parks-Mcclellan algorithm is also used to design Nth-band FIR digital filters. But a disadvantage of the Parks-Mcclellan algorithm is that it needs a good amount of design time. This paper describes a direct design method for Nth-band FIR Filters using Chebyshev polynomials, which provides a reduced design time over indirect methods such as the Parks-Mcclellan algorithm. The response of the resulting filter is equiripple in passband. Our proposed method produces a passband response that is equripple to within a minuscule error, comparable to that of the Parks-Mcclellan algorithm.

• Journal of Communications and Networks
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• v.12 no.6
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• pp.544-557
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• 2010

This paper addresses the problem of controlling the envelope's power peak of single carrier modulated signals, band limited by root-raised cosine (RRC) pulse shaping filters, in order to reduce power amplifier back-off for very small aperture terminals ground stations. Magnitude modulation (MM) is presented as a very efficient solution to the peak-to-average power ratio problem. This paper gives a detailed description of the MM concept and its recent evolutions. It starts by extending the look-up-table (LUT) based approach of the MM concept to M-ary constellations with M ${\leq}$ 16. The constellation and RRC symmetries are explored, allowing considerable reduction on LUT computation complexity and storage requirements. An effective multistage polyphase (MPMM) approach for the MM concept is then proposed. As opposed to traditional LUT-MM solutions, MM coefficients are computed in real-time by a low complexity multirate filter system. The back-off from high-power amplifier saturation is almost eliminated (reduction is greater than 95%) with just a 2-stage MPMM system even for very demanding roll-off cases (e.g., ${\alpha}$ = 0,1). Also, the MPMM is independent of modulation in use, allowing its easy application to constellations with M > 16.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.4
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• pp.639-644
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• 2001

Sequence Code Division Multiple Access) and TDMA(Time Division Multiple Access) system. In this system, the narrowband TDMA signals can cause intolerant interferences to CDMA signals. In this paper, DS-CDMA/TDMA frequency sharing system have been analyzed in AWGN(Additive white Gaussian Noise), MAI (Multi Access Interference) and NI(Narrowband Interference) environment. Also, performance improvement has been obtained by adopting an adaptive notch filtering scheme using complex filter bank and CCI canceller.

• Nuclear Engineering and Technology
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• v.50 no.5
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• pp.698-708
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• 2018

This article introduces a methodology of designing a wavelet operator suitable for multiscale modeling. The operator matrix transforms states of a multivariable system onto projection space. In addition, it imposes a specific structure on the system matrix in a multiscale environment. To be specific, the article deals with a diagonalizing transform that is useful for decoupled control of a system. It establishes that there exists a definite relationship between the model in the measurement space and that in the projection space. Methodology for deriving the multirate perfect reconstruction filter bank, associated with the wavelet operator, is presented. The efficacy of the proposed technique is demonstrated by modeling the point kinetics nuclear reactor. The outcome of the multiscale modeling approach is compared with that in the single-scale approach to bring out the advantage of the proposed method.

• Journal of Institute of Control, Robotics and Systems
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• v.19 no.2
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• pp.87-94
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• 2013

UKF (Unscented Kalman Filter) has been used in the nonlinear systems without initial accurate state estimates instead of EKF (Extended Kalman Filter) of the last decade because the UKF has robustness to the large initial estimation error. In the multirate integrated system such as INS (Inertial Navigation System)/GPS (Global Positioning System) integrated navigation system, however, it is difficult to implement the UKF based navigation algorithm in the mid-grade micro-processor due to the large computational burden. To overcome this problem, this paper proposes a MUKF (Modified UKF) that has a reduced computation burden using the basic idea that the change of the provability distribution for the state variables between measurement updates is small in the multi-rate INS/GPS integrated navigation filter. The performance of the proposed MUKF is verified by numerical simulations.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.3
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• pp.11-17
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• 2010

Multi-standard, multi-band, multi-service system to ensure a flexible interface between the SDR (Software Defined Radio) technology for the implementation of the Stability and Low-Power, Low-Calcualrion DDC (Digital Down Conversion) technology is essential. DDC technology consists of a digital channel filter. This is a typical digital filter because of the limited fisheries are vulnerable to overflow and rounding errors are drawbacks. In this paper, we overcome this disadvantage, we propose the structure of the DDC. The way WDF (Wave Digital Filter) Structural rounding error due to the structural resistance to noise. Therefore, This is the useful structure when the filter coefficients's word length is short. In addition, since IIR filters based on FIR filters based on the amount of computation is reduced because fewer than filter's tap. The proposed structure is used in DDC that CIC (Cascaded Integrator Comb) filter, WDF, IFOP (Interpolated Fourth-Order Polynomials) were analyzed with respect to, the results were confirmed by computer simulation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.8C
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• pp.806-815
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• 2006

It is studied to design a low pass filter of the SRC(sample rate converter), which is used to change the sampling rate of digital signals such as in digital modulation and demodulation systems. The larger the conversion ratio of the sample rate becomes, the more signal processing is needed for the filter, which corresponds to the more complexity in circuit realization. Thus it is important to reduce the amount of signal processing for the case of high conversion ratio. In this paper it is presented a design method of a two-stage cascaded FIR filter, which proved to have reduced amount of signal processing in comparison with a conventional single-stage one. The reduction effect of signal processing turned out to be more noticeable for larger value of conversion ratio, for instance, giving down to 72% in complexity for the conversion ratio of 32. It has been shown that the reduction effect is dependent to specific combination of conversion ratios of the cascaded filters. So an exhaustive search has been performed in order to obtain the optimal combination for various values of the total conversion ratio. In this paper every filter is considered to be implemented in the form of a polyphase FIR filter, and its coefficients are determined by use of the Parks-McCllelan algorithm.