Optical frequency division multiplexing (FDM) technique has the advantage of fully orthogonal transmissions. However, FDM system permits only a small number of FDM channels despite of a great effort, such as frequency stabilization. On the other hand, frequency-domain encoding code-division multiple-access (FE-CDMA) has been widely studied as a type of optical CDMA. In this system, encoding is done in the frequency domain of an ultrashort light pulse spread by optically Fourier transform. However, FE-CDMA accommodates very limited number of simultaneous users, though this scheme uses a vast optical bandwidth. It is attractive to consider the combination of both advantages of FDM and FE-CDMA. We propose FE-CDMA enhancement of FDM (FDM/FE-CDMA). Since in FDM/FE-CDMA the total bandwidth is partitioned into M optical bands and each band is encoded by the code with code length of Nc, we expect nearly perfect orthogonal transmissions. In addition, since the creation of FDM bands is realized by a passive filter, the optical frequency is precisely controlled and the optical frequency allocation is flexible. We derive the bit error rate (BER) as a function of the number of simultaneous users, bit rate, and the utilization efficiency of total bandwidth. We compare the performance of FDM/FE-CDMA with that of the conventional FE-CDMA in terms of the number of simultaneous users on condition that each chip width is constant. As a result, we show that FDM/FE-CDMA can support the larger number of simultaneous users than the conventional FE-CDMA at a given bit error rate under the same total bandwidth.

In this paper, we propose an adaptive lattice-transversal joint (LTJ) filter structure that is quite suitable for the practical implementation of the acoustic echo canceller. The structure maintains fast convergence of the lattice structure and low computational complexity of the transversal structure simultaneously. It is particularly more efficient in memory usage than any other existing fast-convergent algorithm for the acoustic echo cancellation.

Modeling error is the major concerning issue in the trajectory estimation. This paper formulates the dynamic model of a reentry vehicle in reentry phase for identification with an unmodeled acceleration input covering possible model errors. Moreover, this work presents a novel on-line estimation approach, adaptive filter, to identify the trajectory of a reentry vehicle from a single radar measured data. This proposed approach combines the extended Kalman filter and the recursive least-squares estimator of input with the hypothetical testing scheme. The recursive least-squares estimator is provided not only to extract the magnitude of the unmodeled input but to offer a testing criterion to detect the onset and presence of the input. Numerical simulation demonstrates the superior capabilities in accuracy and robustness of the proposed method. In real flight analysis, the adaptive filter also performs an excellent estimation and prediction performances. The recommended trajectory estimation method can support defense and tactical operations for anti-tactical ballistic missile warfare.

In this paper, we propose two new pipelined adaptive digital filter architectures. The architectures are based on an equivalent expression of the least mean square (LMS) algorithm. It is shown that one of the proposed architectures achieves the minimum output latency, or zero without affecting the convergence characteristics. We also show that, by increasing the output latency be one, the other architecture can be obtained which has a shorter critical path.

An optimum filter for extracting a time-varying harmonic signal from the noise-corrupted measurement is proposed. It is derived as a solution of the least mean square estimation with consideration of the pitch estimation error even without any assumption on the filter model. We obtain a comb-like impulse response which consists of homologous and dilated distribution of weights just located periodically with a pitch interval. This remarkable structure is well suited to the proportionally expanding error of pitch repetition times. Examples of the filter design are presented, and the performance of noise suppression is examined by comparison with conventional comb filters.

In this paper, a novel method is proposed for designing two channel biorthogonal filter banks with general IIR filters, which satisfy both the perfect reconstruction and causal stable conditions. Since the proposed filter banks are structurally perfect reconstruction implementation, the perfect reconstruction property is still preserved even when all filter coefficients are quantized. The proposed design method is based on the formulation of a generalized eigenvalue problem by using Remez multiple exchange algorithm. Then, the filter coefficients can be computed by solving the eigenvalue problem, and the optimal solution is easily obtained through a few iterations. One design example is presented to demonstrate the effectiveness of the proposed method.

The checkerboard effect is caused by the periodic time-variant property of multirate filters which consist of up-samplers and digital filters. Although the conditions for some one-dimensional (1D) multirate systems to avoid the checkerboard effect have been shown, the conditions for Multidimensional (MD) multirate systems have not been considered. In this paper, some theorems about the conditions for MD multirate filters without checkerboard effect are derived. In addition, we also consider MD multirate filter banks without checkerboard effect. Simulation examples show that the checkerboard effect can be avoided by using the proposed conditions.

This paper discusses a design technique for multidimensional (M-D) multirate filters which cause no checkerboard distortion. In the first part of this paper, a necessary and sufficient condition for M-D multirate filters to be checkerboard-distortion-free is derived in the frequency domain. Then, in the second part, this result is applied to a scanning line conversion system for television signals. To confirm the effectiveness of the derived condition, band-limiting filters with and without considering the condition are designed, and the results by these filters are compared. A reducibility of the number of delay elements in such a system is also considered to derive efficient implementation.

The authors have proposed a design method for two-dimensional (2-D) separable-denominator (SD) periodically time-variant digital filters (PTV DFs) and confirmed their superiority over 2-D time-invariant DFs. In that result, the periodicity matrix representing the periodicity of the varying filter coefficients is, however, restricted to two cases. This paper extends that idea so that the input-output relation of 2-D SD PTV DFs with an arbitrary periodicity matrix can be determined. This enables us to design wide range of 2-D PTV DFs.

2-pole band-pass filters (BPFs) with tap-excitation are prepared by using high temperature superconductors (HTS). The possibility of realizing superconducting coplanar filters with attenuation poles is revealed.

A constant modulus adaptive array algorithm is derived using analysis and synthesis filter banks to permit adaptive digital beamforming for wideband signals. The properties of the CMA adaptive array using the filter banks are investigated. This array would be used to realize adaptive digital beamforming when this is difficult by means of ordinary (that is, non-subband) processing due to the limited speed of signal processor operations. As an actual application, we present a beamspace adaptive array structure that combines the analysis and synthesis filter banks with RF-domain multibeam array antennas, such as those utilizing optical signal processing.

The temperature dependence of central wavelength of optical filters is a serious problem for the dense WDM systems. This dependence is owing to the temperature dependence of optical path-length of the waveguide. In this study, we realized a temperature independent silica-based optical filter at 1. 55 µm wavelength using an athermal waveguide, in which optical pathlength is independent of temperature. First, we designed a silica-based athermal waveguide, and next we designed and fabricated a ring resonator using the athermal waveguide. As a result, we successfully decreased the temperature dependence of central wavelength to less than 4 10 -4 nm/K, which is 3% and 0. 3% of corresponding values of conventional silica-based and semiconductor waveguide filters, respectively.

High- and low-reflection Bragg gratings with a flat-top spectral response free from ripples are proposed. Add/drop filters are created based on gratings photoinduced on planar waveguides by using the new design schemes. The measured spectral responses for the high and low reflection gratings are in good agreement with the calculated ones, and show the flat-top spectral responses.

Recently, a great deal of effort has been devoted to the design problem of "constrained least squares M-D FIR filter" because a significant improvement of the squared error is expected by a slight relaxation of the minimax error condition. Unfortunately, no design method has been reported, which has some theoretical guarantee of the convergence to the optimal solution. In this paper, we propose a class of novel design methods of "constrained least squares M-D FIR filter. " The most remarkable feature is that all of the proposed methods have theoretical guarantees of convergences to the unique optimal solution under any consistent set of prescribed maximal error conditions. The proposed methods are based on "convex projection techniques" that computes the metric projection onto the intersection of multiple closed convex sets in real Hilbert space. Moreover, some of the proposed methods can still be applied even for the problem with any inconsistent set of maximal error conditions. These lead to the unique optimal solution over the set of all filters that attain the least sum of squared distances to all constraint sets.

The Heart Rate Variability (HRV) analysis has become vigorous these days. One reason for this is that the HRV analysis investigates the dynamics of the autonomic nervous system activities which control the HRV. The Integral Pulse Frequency Modulation (IPFM) model is a pulse generating mechanism model in the nervous system, that is one of the models which connects the HRV to the autonomic nervous system activities. The IPFM model is a single frequency component model; however, the real HRV has multiple frequency components. Moreover, there are refractory periods after generating action potentials are initiated. Nevertheless, the IPFM model does not consider refractory periods. In order to make sure of the accuracy and the effectiveness of the integral function (IF) method applied to the real data, we consider the absolute refractory periods and two frequency components. In this investigation, the simulated HRV was made with a single and double frequency component using the IPFM model with and without absolute refractory periods. The original generating function of the IPFM model was demodulated by using the instantaneous heart rate tachogram. The power of the instantaneous pulse rate per minute was analyzed by the direct FFT method, the IF FFT method without the absolute refractory periods, and the IF FFT method with the absolute refractory periods. It was concluded that the IF FFT method can demodulate the original generating function accurately.

This paper presents a new algorithm that can solve the problem of selecting appropriate update step size in the LMS algorithm. The proposed algorithm, called a Complementary Pair LMS (CP-LMS) algorithm, consists of two adaptive filters with different update step sizes operating in parallel, one filter re-initializing the other with the better coefficient estimates whenever possible. This new algorithm provides the faster convergence speed and the smaller steady-state error than those of a single filter with a fixed or variable step size.

This letter proposes a method to transform normalized ARMA lattice filters, which are originally realized for signal analysis, into signal synthesis lattice filters. Although the transformation method has been proposed for normalized ARMA lattice filters with the MA order which is greater than or equal to the AR order, it has not been done when the AR order is greater than the MA order. With the proposed method, once an ARMA lattice filter with the AR order greater than the MA order is realized, then it can be transformed to the signal synthesis filter.

A RAKE combiner based on a matched filter (MF) can be relatively easily implemented since the despread signal components that have propagated along different paths appear sequentially at the MF output. An important design problem is how to accurately select the paths having sufficiently large signal-to-noise power ratios (SNRs). This paper proposes a simple path selection algorithm that uses two selection thresholds. The first threshold is to select the paths that provide largest SNRs. However, as the total received signal power (sum of the signal powers of all paths) decreases, some of the selected paths become noisy. Therefore, we introduce a second threshold that discards the noisy or noise-only paths from among those selected by the first threshold. We apply the proposed path selection algorithm to a pilot symbol-assisted coherent RAKE combiner and find by computer simulations a near optimum set of the two thresholds in frequency selective multipath Rayleigh fading channels. Several power delay profile shapes are considered. The simulation results demonstrate that the MF-based RAKE combiner with the two selection thresholds can achieve a bit-error-rate (BER) performance close to the ideal case (i. e. , the paths to be used for RAKE combining are selected for each power delay profile such that the required signal energy per information bit-to-noise spectrum density ratio (Eb/N0) is minimized).

In this paper, we consider the selection of analysis filters used in the delayless subband adaptive digital filter (SBADF) and propose to use simple analysis filters to reduce the computational complexity. The coefficients of filters are determined using the components of the first order Hadamard matrix. Because coefficients of Hadamard matrix are either 1 or -1, we can analyze signals without multiplication. Moreover, the conditions for convergence of the proposed method is considered. It is shown by computer simulations that the proposed method can converge to the Wiener filter.

In this paper a new general method for approximation of arbitrary multiband filter loss specifications, including all classical, maximally flat and equiripple approximations as special cases, is proposed. It is possible to specify different magnitude behavior (flat or equiripple of given degree) and different maximal losses in the different passbands and to optimize all transmission and attenuation zeroes positions or to have some of them fixed. The optimization procedures for adjustment of the filter response are based on modified Remez algorithm and are performed in s-domain what is regarded since recently as an advantage in the case of design of parallel allpass structures based IIR digital filters. A powerful algorithm and appropriate software are developed following the method and their efficiency is verified through design examples.