• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.5B
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• pp.830-841
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• 2000

We extend the sue of the method of least square to develop a recursive algorithm for the design of adaptive transversal filters such that, given the least-square estimate of this vector of the filter at iteration n-1, we may compute the updated estimate of this vector at iteration a upon the arrival of new data. We begin the development of the RLS algorithm by reviewing some basic relations that pertain to the method of least squares. Then, by exploiting a relation in matrix algebra known as the matrix inversion lemma, we develop the RLS algorithm. An important feature of the RLS algorithm is that it utilizes information contained in the input data, extending back to the instant of time when the algorithm is initiated. In this paper, we propose new tap weight updated RLS algorithm in adaptive transversal filter with data-recycling buffer structure. We prove that convergence speed of learning curve of RLS algorithm with data-recycling buffer is faster than it of exiting RL algorithm to mean square error versus iteration number. Also the resulting rate of convergence is typically an order of magnitude faster than the simple LMS algorithm. We show that the number of desired sample is portion to increase to converge the specified value from the three dimension simulation result of mean square error according to the degree of channel amplitude distortion and data-recycle buffer number. This improvement of convergence character in performance, is achieved at the (B+1)times of convergence speed of mean square error increase in data recycle buffer number with new proposed RLS algorithm.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.4
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• pp.173-179
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• 2002

This paper describes an equalizer using the DFE (Decision Feedback Equalizer) structure, CMA (Constant Modulus Algorithm) and LMS (Least Mean Square) algorithms. The DFE structure has better channel adaptive performance and lower BER than the transversal structure. The proposed equalizer can be used for 16/64 QAM modems. We employ high speed multipliers, square logics and many CSAs (Carry Save Adder) for high speed operations. We have developed floating-point models and fixed-point models using the COSSAP$\^$TM/ CAD tool and developed VHDL filter. The proposed equalizer shows low BER in multipath fading channel. We have performed models. From the simulation results, we employ a 12 tap feedback filter and a 8 tap feedforward logic synthesis using the SYNOPSYS$\^$TM/ CAD tool and the SAMSUNG 0.5$\mu\textrm{m}$ standard cell library (STD80) and verified function and timing simulations. The total number of gates is about 130,000.

• The Journal of the Acoustical Society of Korea
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• v.15 no.2
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• pp.79-84
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• 1996

In this paper, adaptive correlator model using backpropagation neural network based on complex multilayer perceptron is presented for suppressing interference of narrow-band of direct sequence spread spectrum receiver in CDMA mobile communication systems. Recursive least square backpropagation algorithm with backpropagation error is used for fast convergence and better performance in adaptive correlator scheme. According to signal noise ratio and transmission power ratio, computer simulation results show that bit error ratio of adaptive correlator uswing backpropagation neural network improved than that of adaptive transversal filter of direct sequence spread spectrum considering of co-channel and narrow-band interference. Bit error ratio of adaptive correlator using backpropagation neural network is reduced about $10^{-1}$ than that of adaptive transversal filter where interference versus signal ratio is 5 dB.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.1C
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• pp.82-91
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• 2005

This paper introduces an efficient affine projection algorithm(APA) using iterative hyperplane projection. Among various fast converging adaptation algorithms, APA has been preferred to be employed for various applications due to its inherent effectiveness against the rank deficient problem. However, the amount of complexity of the conventional APA could not be negligible because of the accomplishment of sample matrix inversion(SMI). Moreover, the 'shifting invariance property' usually exploited in single channel case does not hold for the application of space-time decision-directed equalizer(STDE) deployed in single-input-multi-output(SIMO) systems. Thus, it is impossible to utilize the fast adaptation schemes such as fast transversal filter(FlF) having low-complexity. To accomplish such tasks, this paper introduces the low-complexity APA by employing hyperplane projection algorithm, which shows the excellent tracking capability as well as the fast convergence. In order to confirm th validity of the proposed method, its performance is evaluated under wireless SIMO channel in respect to bit error rate(BER) behavior and computational complexity.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.15 no.9
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• pp.764-774
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• 1990

In this paper, make a study on comparison of adaptive filters for predictor characteristics that transversal, lattice, and joint process lattice filter is using the LMS algorithm that is simple structure and pracotical application is easy. The theoical background and structure of each adaptive filters exhibit for practical design. Adaptive convergence condition for optimal weight vector and optimal reflection coefficient make clear, and it is also shown through computer simulation. The error signals and noise characteristics of these filters make a comparative study. In view of the results, joint process lattice filter is shown that most superior characteristic in these adaptie filters.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 1998.11a
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• pp.574-577
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• 1998

In this paper, a new simple and efficient technique to improve the convergence speed of LMS algorithm is proposed in an interference-limited multi-path fading environment as encountered in indoor wireless communications. The convergence characteristics of the proposed algorithm, whose coefficients are multiply adapted in a symbol time period by recycling the received data, are analyzed to prove theoretically the improvement of convergence speed. The theoretical analysis shows that the data-recycling in technique can increase convergence speed by (B+1) times without increasing the computational complexity substantially where B is the number of recycled data. The results of the computer simulation demonstrate that the simulation results are in accordance with the theoretical analysis and the superiority of the filter algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.15 no.2
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• pp.166-175
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• 1990

The conventional FIR filters can be very expensive to implement due to the complexity of multibit multipliers. This paper presents an new type of multiplierless structure which is particularly suited to the hardware implementation of small, low cost, low power, high speed digital filters. The filter structures consisting of a transversal filter with tap coefficiented to the combination of two elements of the set {0, $\pm$$2^n$;n = integer} and cascaded with a integrator are proposed. Performance has been tested via simulation on a digital computer, and the results show that the response characteristics of presented filters are as equally good as those of conventional finitewordlength filters.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.9 no.4
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• pp.11-22
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• 1999

Least-mean-square(LMS) adaptive filters have proven to be extremely useful in a number of signal processing tasks. However LMS adaptive filter suffer from a slow rate of convergence for a given steady-state mean square error as compared to the behavior of recursive least squares adaptive filter. In this paper an efficient signal interference control technique is introduced to improve the convergence speed of LMS algorithm with tap weighted vectors updating which were controled by reusing data which was abandoned data in the Adaptive transversal filter in the scheme with data recycling buffers. The computer simulation show that the character of convergence and the value of MSE of proposed algorithm are faster and lower than the existing LMS according to increasing the step-size parameter $\mu$ in the experimentally computed. learning curve. Also we find that convergence speed of proposed algorithm is increased by (B+1) time proportional to B which B is the number of recycled data buffer without complexity of computation. Adaptive transversal filter with proposed data recycling buffer algorithm could efficiently reject ISI of channel and increase speed of convergence in avoidance burden of computational complexity in reality when it was experimented having the same condition of LMS algorithm.

• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• v.2
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• pp.353-357
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• 2006

One-sided and two-sided ATF for GNSS receiver are deigned, implemented and evaluated in this paper. The difference f filter characteristics such as the location of zeros and the frequency response is reviewed and examined with experiments. NLMS adaptation algorithm is adopted for updating the weighting coefficients of the 12-tap FIR filter. he performance of ATF is evaluated using real signals consisting of the signals from GPS simulator and the signal generator. The output of ATF is fed into the SDR to evaluate SNR and the position accuracy. The complexity of implementation is also compared and the effects of the time delay and the phase delay are examined. The experimental results show that one-sided and two-sided ATF give similar performance against single tone CWI.

• The Journal of the Acoustical Society of Korea
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• v.18 no.4
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• pp.23-30
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• 1999

In this study, we propose a new optimization algorithm for a withdrawal weighted SAW transversal filter to satisfy given, specifications such as bandwidth, ripple, insertion loss, and sidelobe rejection level. An analysis tool for the withdrawal weighted filter has been produced by means of the delta function model, and has been applied to the design of a filter consisting of an uniform input IDT and a withdrawal weighted output IDT. This optimization algorithm consists of three routines, which eventually determines eight design parameters to satisfy the performance specifications. At the first step, the number of input and output IDT fingers and their geometrical size are determined by the insertion loss specification. At the next step, the bandwidth is controlled by the change of the IDT finger position. Finally, the sidelobe rejection level is modified through the add/skip technique of IDT fingers. The algorithm in this paper is distinct from conventional techniques in that it can simultaneously consider all the specifications such as bandwidth, ripple, sidelobe rejection level and insertion loss, and optimize the geometry of the withdrawal weighted SAW filter.