• 제어로봇시스템학회:학술대회논문집
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• 2003.10a
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• pp.413-418
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• 2003

In this paper, a new multirate optimal finite impulse response (FIR) filter is proposed for the signal reconstruction in the nosy filter bank systems. The multirate optimal FIR filter replaces the conventional synthesis filters and the Kalman synthesis filter. First, the generic linear model is derived from the multirate state space model for an autoregressive (AR)input signal. Second, the multirate optimal FIR filter is derived from the multirate generic linear model using the minimum variance criterion. This paper also provides numerical examples and results. The simulation results illustrate that the performance is improved compared with conventional synthesis filters and the proposed filter has advantages over the Kalman synthesis filter.

• 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.

• 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.

• 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.

• 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.

• 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.

• 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.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.38 no.7
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• pp.455-463
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• 2001

In this Paper, an approximate processing method is proposed and tested. The proposed method uses variable CSD (VCSD) coefficients which approximate filter stopband attenuation by controlling the precision of the CSD coefficient sets. A decimation filter for Audio Codec '97 specifications has been designed having processor architecture that consists of program/data memory, arithmetic unit, energy/level decision, and sinc filter blocks, and fabricated with 0.6${\mu}{\textrm}{m}$ CMOS sea-of-gate technology. For the combined two halfband FIR filters in decimation filter, the number of addition operations were reduced to 63.5%, 35.7%, and 13.9%, compared to worst-case which is not an adaptive one. Experimental results show that the total power reduction rate of the filter is varying from 3.8 % to 9.0 % with respect to worst-case. The proposed approximate processing method using variable CSD coefficients is readily applicable to various kinds of filters and suitable, especially, for the speech and audio applications, like oversampling ADCs and DACs, filter banks, voice/audio codecs, etc.