The numerical properties of the recursive least squares (RLS) algorithm and its fast versions have been extensively studied. However, very few investigations are reported concerning the numerical behavior of the predictor based least squares (PLS) algorithms that provide the same least squares solutions as the RLS algorithm. This paper presents a comparative study on the numerical performances of the RLS and the backward PLS (BPLS) algorithms. Theoretical analysis of three main instability sources reported in the literature, including the overrange of the conversion factor, the loss of symmetry and the loss of positive definiteness of the inverse correlation matrix, has been done under a finite-precision arithmetic. Simulation results have confirmed the validity of our analysis. The results show that three main instability sources encountered in the RLS algorithm do not exist in the BPLS algorithm. Consequently, the BPLS algorithm provides a much more stable and robust numerical performance compared with the RLS algorithm.

It is shown that two-dimensional linear phase FIR digital filters with various shapes of frequency response can be designed and realized as modular array structures free of multiplier coefficients. The design can be performed by judicious selection of two low order linear phase transfer functions to be used at each module as kernel filters. Regular interconnection of the modules in L rows and K columns conditioned with boundary coefficients 1, 0 and 1/2 results in higher order digital filters. The kernels should be chosen appropriately to, first, generate the desired shape of frequency response characteristic and, second, lend themselves to multiplierless realization. When these two requirements are satisfied, the frequency response can be refined to possess narrower transition bands by adding additional rows and columns. General properties of the frequency response of the array are investigated resulting in Theorems that serve as valuable tools towards appropriate selection of the kernels. Several design examples are given. The array structures enjoy several favorable features. Specifically, regularity and lack of multiplier coefficients makes it suitable for high-speed systolic VLSI implementation. Computational complexity of the structure is also studied.

A novel technique is proposed to fabricate a chirped fiber Bragg grating utilizing thermal diffusion of core dopant. The chirped grating is written with a uniform period by using UV exposure technique in the fiber whose effective index of the guided mode varies along its length. Thermal diffusion of the core dopant it employed to realize this change of the effective index. Through the thermal diffusion process, the effective index of the fiber decreases from its initial value. When the grating is written in the diffused core region, its reflection wavelength becomes shorter than that in the non-diffused region. The continuous change of effective index is required for making a chirped grating. The fiber is heated by a non-uniform heat source. When the uniform grating is written in this region, the reflection wavelength smoothly changes along the fiber length although the grating period is constant. By optimizing the fiber parameters to realize a highly chirped grating, we have obtained a typical one whose bandwidth is 14.1 nm at half maximum and maximum rejection in transmission is 29 dB. Additionally, the proposed method has an advantage to control the chirp profile with high mechanical reliability.

Analog inverter is one of the most useful building blocks in analog circuits. This paper proposes an analog inverter consisting of a p-channel MOS (PMOS) and an n-channel MOS (NMOS) inverter and presents an application to all-pass filter realizations. The proposed circuit has a wide dynamic range by combining PMOS and NMOS inverters. When the proposed analog inverter is applied to an all-pass filter, the circuit configuration becomes simpler and occupies less chip area and power consumption.

In this paper, we present a novel way to design biorthogonal and paraunitary linear phase filter banks. The square error of the perfect reconstruction of the filter bank is expressed in quadratic form of filter coefficients and the cost function is minimized by solving linear equation iteratively without nonlinear optimization. With some modifications, this method is extended to the design of paraunitary filter banks. Furthermore, the lattice structure of odd-channel paraunitary filter banks is also derived. Design examples are given to validate the proposed method.

In the adaptive lattice estimation process, it is well known that the convergence speed of the successive stage is affected by the estimation errors of reflection coefficients in its preceding stages. In this paper, we propose block estimation methods of two-dimensional (2-D) adaptive lattice filter. The convergence speed of the proposed algorithm is significantly enhanced by improving the adaptive performance of preceding stages. Furthermore, this process can be simply realized. The modeling of 2-D AR field and texture image are demonstrated through computer simulations.

In this paper, a block implementation of high-speed IIR adaptive noise canceller is proposed. First, the block difference equation of an IIR filter is derived by the difference equation for high-speed signal processing. It is shown that the computational complexity for updating the coefficients of IIR adaptive filter can be reduced by using the relations between the elements of coefficient matrices of block difference equation. Secondly, the block implementation of IIR adaptive noise canceller is proposed in which the convergence rate is increased by successively adjusting filter Q-factors. Finally, the usefulness of proposed block implementation is verified by the computer simulations.

The asynchronous transfer mode (ATM) provides efficient switching capability for various kinds of communication services. To guarantee the minimum quality of services in the ATM networks, the bandwidth allocation setup procedure between the network nodes and users is very important. However, most of call admission control (CAC) methods which have been proposed so far are not fully appropriate to apply to real environments in terms of the complexity of the hardware implementation or the accuracy of assumptions about the cell-arrival processes. In addition, the success of broad bandwidth applications in the future multimedia environments will largely depend on the degree to which the efficiency in communication systems can be achieved, so that establishing high-speed CAC schemes in the ATM networks is an indispensable subject. This paper proposes a new cell-loss rate estimation method for the real time CAC in ATM networks. A neural network model using the Kalman filter algorithm was employed to improve the error minimizing process for the cell-loss estimation problem. In the process of optimizing the three-layer perceptron, the average, the variance, and the 3rd central moment of the number of cell arrivals were calculated, and cell-loss rate date based on the non-parametric method were adopted for outputs of the neural network. Evaluation results concerned with the convergence using the sum of square errors of outputs were also discussed in this paper. Using this algorithm, ATM cell-loss rates can be easily derived from the average and peak of cells rates coming from users. Results for the cell-loss estimation process suggest that the proposed method will be useful for high-speed ATM CAC in multimedia traffic environments.

This paper discusses the problems in designing virtual-path (VP) networks and underlying transmission-path (TP) networks using the "self-sizing" capability. Self-sizing implies an autonomous adjustment mechanism for VP bandwidths based on traffic conditions observed in real time. The notion of "bandwidth demand" has been introduced to overcome some of the problems with VP bandwidth sizing, e.g., complex traffic statistics and diverse quality of service requirements. Using the bandwidth demand concept, a VP-bandwidth-sizing procedure is proposed in which real-time estimates of VP bandwidth demand and successive VP bandwidth allocation are jointly utilized. Next, TP bandwidth demand, including extra capacity to cover single-link failures, is defined and used to measure the congestion level of the TP. Finally, a TP provisioning method is proposed that uses TP "lifetime" analysis.

This letter presents a realization of active current-mode resonator with complex coefficients using CCIIs. The resonator can be used for cascade or leapfrog configuration of high-order bandpass filters with complex coefficients. For realizing the resonators, only the grounded capacitors and the grounded resistors as passive elements are required, therfore the resonator is suitable for the integrated circuit realization. The letter shows that the response error of the proposed circuit caused by nonideality of active components is more easily compensated than that of voltage-mode counterpart. Experimental result is used for verifying the feasibility of the proposed resonator.

A new structure for adaptive AR spectral estimation based on multi-band decomposition of the linear prediction error is introduced and the mathematical background for the soulution of the related adaptive filtering problem is derived. The presented structure gives rise to AR spectral estimates that represent the true underlying spectrum with better fidelity than conventional LS methods by allowing an arbitrary trade-off between variance of spectral estimates and tracking ability of the estimator along the frequency spectrum. The linear prediction error is decomposed through a filter bank and components of each band are analyzed by different window lengths, allowing long windows to track slowly varying signals and short windows to observe fastly varying components. The correlation matrix of the input signal is shown to satisfy both time-update and order-update properties for rectangular windowing functions, and an RLS algorithm based on each property is presented. Adaptive forward and backward relations are used to derive a mathematical framework that serves as a basis for the design of fast RLS alogorithms. Also, computer experiments comparing the performance of conventional and the proposed multi-band methods are depicted and discussed.

To realize a low-cost WDM transceiver module based on a PLC-platform, simple, assembly techniques have been successfully developed. The formation of index marks with an accuracy of below 0.1 µm has made it possible to mount Opto-electronic devices on the silicon terrace of the PLC-platform by a passive alignment. A newly developed trench formation technique for inserting a 1.3/1.5 µm WDM dielectric filter enabled us not only to ensure a stable WDM function but also to prevent excess loss associated with the dielectric filter scheme. It is found that these two technologies are practically useful to achieve high-performance WDM transceiver module.

This paper is concened with the design and implementation of a 2-channel, 2-dimensional filter bank using rectangular (analog/digital) and quincunx (digital/digital) sampling. The associated analog low-pass filters are separable where as the digital low-pass filters are non-separable for a minimum sampling density requirement. The digital low-pass filters are Butterworth type filters, N = 9, realized as LWDFs. They, when itterated, approximate a valid scaling function (raised-consine scaling function). The obtained system can be used to compute a discrete wavelet transform.

The present paper examines a two-dimensional (2-D) joint-process lattice estimator and its implementation for image restoration. The gradient adaptive lattice (GAL) algorithm is used to update the filter coefficients. The proposed adaptive lattice estimator can represent a wider class of 2-D FIR systems than the conventional 2-D lattice models. Furthermore, its structure possesses orthogonality between the backward prediction errors. These results in superior convergence and tracking properties versus the transversal and other 2-D adaptive lattice estimators. The validity of the proposed model for image restoration is evaluated through computer simulations. In the examples, the implementation of the proposed lattice estimator as 2-D adaptive noise cancellator (ANC) and 2-D adaptive line enhancer (ALE) is considered.

For designing the underwater transmission system using directly projected audible sound by underwater loudspeaker to prevent a diving accident and/or to give a working instrucion, it is important to estimate the transmission loss for a wall not onl for pure tones but also for wideband signal such as voice and noise. In this paper, two practical methods of evaluating the underwater insulation effect for a single wall are discussed. One is a reconfirmation that the mass law which is frequently used in air still explains the transmission loss in water. Because parameters such as surface density and sound velocity in the mass law are widely changeable depending on the depth in water, much complexity is involved in preparing a theoretical curve for every parameter. So to avoid such complexity, a unified parameter Φ(=ωm cos θ/2ρc) is introduced to describe the mass law. This newly presented curve as a function of Φ is in good agreement with all rearranged experimental data for every kind of plates. The other is a proposition of new evaluating method of insulation effect of a wall for a wideband signal, using an idea of (100-α) percentage point of the noise level probability destribution, Lα. Firstly proposed method is confirmed experimentally and secondly proposed method is confirmed by a simulation experiment.

In this paper, new wavelet bases are presented. We address problems associated with the proposed matched filter in multirate systems, using an optimum receiver that maximises the SNR at the sampling instant. To satisfy the Nyquist (ISI-free transmission) and matched filter (maximum SNR at the sampling instant) criteria, the overall system filtering strategy requires to split the narrowest filter equally between transmitter and receiver. In data transmission systems a raised-cosine filter is therefore often used to bandlimit signals from which wavelet bases are derived. Sampling in multiresolution subspaces is also discussed.

This paper describes the design and performance of optical components for in-service fiber testing and monitoring in optical subscriber loops. As the number of test fibers increases, compact and cost-effective components are becoming more important. To meet this need, we have developed a highly-dense hybrid structure for optical couplers and filters, which both play key roles in testing systems. It was realized by utilizing a polyimide-base thin film filter and a waveguide-type wavelength insensitive coupler. This component operates by combining a signal and a test light with a ratio of 80/20% and isolating the test light with a value of 50 dB. The experimental samples were successfully fabricated with an excess loss of 1 dB, a return loss of 40 dB, a plolarization dependent loss (PDL) of 0.3 dB, and good environmental and mechanical stability. We successfully applied the samples to an optical branch module (OBM), and achieved a component density twice that of a conventional module. The optical characteristics of the OBM met our target values. The results we obtained for termination cords incorporating the polyimide-base filter were also satisfactory.

A digital filter is one of the fundamental elements in the digital video transmission, and a multiplier acts as the key factor that determines the operation speed and silicon area of the filter. In terms of the digital video transmission, the required performance of a multiplier is to operate at the speed of 20-100 MHz but with the precision of 8-10 bits. In the case of implementing such an FIR filter with more than a certain number of taps, the same number of multipliers are necessary to realize the speed. Moreover, even though the coefficients to the filter are desired to be programmable, it is possible to change coefficients in the vertical fly-back interval of television receivers. This allows the preloadability of coefficients to the filter such that each coefficient can be treated as a constant during the filtering operation. Motivated by these requirements and functionalities, a novel multiplier and FIR filter architecture is described, which is to be synthesized with the use of a high level synthesis tool of COMPASS Design Navigator, partly with the aid of the manual design by means of a 0.8µm CMOS library.

In this letter, we propose a blind adaptive receiver with nonlinear structure for DS/CDMA communication systems. The proposed receiver requires the signature waveform and timing for only the desired user. It is shown that the blind adaptation is equivalent to the adaptation with the training signal and the function to be minimized has no local minima.

A 3V-50 MHz analog CMOS current-mode continuous-time active filter with a negative resistance load (NRL) is proposed. In order to design a current-mode current integrator, a modified basic current mirror with a NRL to increase the output resistance is employed. The inherent circuit structure of the designed NRL current integrator, which minimizes the internal circuit nodes and enhances the gain bandwidth product, is capable of making the filter operate at the high frequency. The third order Butterworth low pass filter utilizing the designed NRL current integrator is synthesized and simulated with a 1.5 µm CMOS n-well process. Simulation result shows the cutoff frequency of 50 MHz and power consumption of 2.4mW/pole with a 3V power supply.