Achieving optimal performance with minimal complexity are conflicting problems encountered in constructing receivers. In this paper, to solve the problem, we propose sector beamed space hopping which utilizes a Viterbi equalizing receiver. Reduction of the number of RF circuit sets, system complexity and decreasing the computational burden of the Viterbi equalizer through the use of sector beamed space hopping is presented. This is achieved using a sector beamed antenna which limits the number of paths in the multipath channel environment. This paper describes each key component which comprises the system and discusses the application of FH-SS communication. The channel is assumed to be an industrial indoor propagation channel, such as those found in a factory, where high reliability is required and many complex multipaths exist. We confirm through simulation that Viterbi equalization using less computational complexity can be obtained. It is found that there exists a trade off between system complexity and performance. Through the discussion of power consumption, cost and BER performance, we show that the proposed system achieves acceptable performance while having a low system complexity.

New bandpass filters (BPFs) with stub resonators are proposed for creating multiple attenuation poles. Firstly, the stub-dependent characteristics of the distributed-element stubs are examined theoretically. Secondly, the new BPFs with resonators of combined stubs are proposed. An advantage of these filters is the possibility of controlling the number of attenuation poles. The design of the proposed filter is carried out based on the general filter design with the narrow-band approximation technique. The transmission and reflection characteristics of the proposed BPFs are also examined theoretically and experimentally. The miniaturization of the filters is also carried out using the resonator with loaded-element stubs. The discussions lead us to the conclusion that the proposed design method of the filters are useful for controlling the number of attenuation poles of the BPF.

In this paper, we describe our SFQ circuit design and measurement carried out in SRL-ISTEC. We are studying an oversampling sigma-delta modulator and a counter-type decimation filter with multistage structure for developing AD converters for software-defined radio application. We are also developing a superconducting memory, whose peripheral circuits are constructed with SFQ circuits.

A new class of least-squares algorithms is presented for adaptive filtering. The idea is to use a fixed set of directions and perform line search with one direction at a time in a cyclic fashion. These algorithms are called Euclidean Direction Search (EDS) algorithms. The fast version of this class is called the Fast-EDS or FEDS algorithm. It is shown to have O(N) computational complexity and a convergence rate comparable to that of the RLS algorithm. Computer simulations are presented to illustrate the performance of the new algorithm.

We propose the new and highly accurate design theory of the high Tc superconducting (HTS) miniaturized coplanar waveguide (CPW) bandpass filters (BPFs) with highly packed meanderlines. BPFs are designed using the external quality factor (Qe) and coupling constant (k) (Q-k method). These parameters are estimated from the transmission coefficient obtained by the 2.5-dimensional electromagnetic field simulator. Moreover, the Q-k method is compared with the J-b method (designed using admittance inverter and susceptance slope parameter) presented previously; in this way we confirmed that the Q-k method has higher accuracy than the J-b method. We realized the design of a the highly packed meanderline CPW BPF (5 pole, center frequency = 2 GHz, fractional band width = 15 MHz, ripple = 0.1 dB) in a 3.5 mm 8 mm substrate.

An approach to the enhancement of speech signals corrupted by additive colored noise is proposed and the system architecture to implement the proposed idea in real-time communication is introduced in this paper. A combination of a bandpass FIR filtering technique with wiener filtering is used to improve the SNR for speech signals. The average SNR improvement (between input and output SNR) is 22.48 dB. The additive noises are the sound from a turbo prop aircraft. The system, which shows excellent performance, is designed based on a 16 bits fixed point DSP (ADSP-2181) from Analog Devices. Experiment results demonstrate that the FIR filter leads to a significant gain in SNR, thus visibly improvement for the quality and the intelligibility of the speech.

This paper proposes an extension of a conventional current conveyor (CC) concept and its applications. A relaxation of the definition of a conventional CC makes a CC simple. A novel current conveyor named extended current conveyor (ECC) is introduced. An ECC keeps the most significant feature of a CC, i.e., a CC is combination of a VCVS and a CCCS. On the other hand, the other conditions are relaxed. All terminals of the ECC are allowed to have offset voltage and offset current. An ECC usually has a simple structure with a small number of MOSFETs thanks to the relaxation of the conditions. Some circuit configurations for the ECC which have various characteristics are shown. An NIC and OTAs are realized using ECCs. Validity of an ECC is confirmed through HSPICE simulation.

In this paper, we propose a universal biquad filter that can realize all types of 2nd-order functions, such as Low-pass Filters (LPF), High-Pass Filters (HPF), Band-Pass Filters (BPF), Band-Elimination Filters (BEF), and All-Pass Filters (APF). Also, the filter types can be programmable digitally with built-in switches. The proposed circuit can be realized by using a CMOS technology that is suitable for a mixed digital-analog LSI. In addition, the circuit can operate in high frequencies with a low power supply voltage because it is based on a current-mode circuit. Finally, the proposed circuit is simulated by PSpice to confirm its characteristics.

In this paper, we discuss an IF image rejection system with variable bandwidth and center frequency. The system is consists of a pair of frequency mixers multiplied by the complex sinusoid and a complex analog filter. By employing the complex leapfrog structure using OTA-C configuration and the frequency transformation from the normalized LPF, the proposed system is capable of variable bandwidth and center frequency characteristics. SPICE simulations result more than 43 [dB] image rejection is achieved for 6 [kHz] and 12 [kHz] bandwidths at 50 [kHz] IF.

A simple method for improving out-of-band characteristics of a planar microwave filter is proposed. We clarify the close relationship among 'tap connection,' 'attenuation pole' and 'spurious responses' in filter design, theoretically and experimentally. Firstly, the basic characteristics of the resonator depending on the excitation method are examined. We show that skirt characteristics can be improved and spurious responses can be suppressed by using the tap connection technique. Secondly, the application examples of bandpass filters (BPFs) on the basis of the resonator with our principle are provided. It is confirmed that the resonator depending on the excitation method is useful for improving out-of-band characteristics of the planar microwave filter.

Image processing in transform domain has many advantages but it could be suffered from local effects such as a blocking artifact. In this paper, an image processing is performed by weighting coefficients in the compressed domain, i.e., filtering coefficients are appropriately selected according to the processing. Since we find the appropriate factors according to global image enhancement, blocking artifacts are reduced between inter-blocks. Experimental results show that the proposed technique has the advantages of simple computation and easy implementation.

An algorithm based on the wavelet transform (WT) was developed to analyze the QRS complex in a three-dimensional magnetocardiogram (3-D MCG) recorded from 3 normal subjects and 1 patient with anterior myocardial infarction (MI). By using a wavelet equivalent filter constructed with the WT algorithm, the high frequency components of the QRS complex related to the late fields (LF) were detected for the patient with anterior MI at different scale. We quantified the high frequency components of the QRS complex by calculating root-mean-square (RMS) value at different scale. The LF mainly existed in the frequency band of about 35.5 to 110.5 Hz with the amplitude of about 0.1 to 0.4 pT for Bx, By, and Bz components. In order to discuss the activities of the heart between the normal subject and the patient with anterior MI, we have also evaluated the spatial energy distribution (SED) of the QRS complex by displaying isoenergy contour maps at different scale. Being different from the normal subject, the patient with anterior MI represented different the pattern of the SED in various frequency band for the ST segment of the QRS complex of Bx, By, and Bz components. It is efficient to use the WT algorithm for analyzing the QRS complex in the 3-D MCG.

This letter proposes a new approach for feature extraction using steerable filters. This approach is based on the concept of orientation-energy histogram which yields the local direction of dominant orientation. The testing is carried out using a training set of 1000 and a set of 300 unknown 40 40 hand-written digits. As a result of the simulations, 92% correct recognition is provided.

It is shown that the effective secret-key size of TOYOCRYPT-HS1 stream cipher is only 96 bits, although the secret key consists of 128 bits. This characteristic opens a door for developing an algorithm for cryptanalysis based on the time-memory-data trade-off with the overall complexity significantly smaller than the exhaustive search over the effective key space.

In this paper, we propose a method to reconstruct current distributions in the human brain from neuromagnetic measurements. The proposed method is based on the weighted lead-field synthetic (WLFS) filtering technique with the weighting factors calculated from the results of previous source space scanning. In this method, in addition to the depth normalization technique, weighting factors of the WLFS are determined by the cost values previously calculated based on the multiple signal classification (MUSIC) scan. We performed computer simulations of this method under noisy measurement conditions and compared the results to those obtained with the conventional WLFS method. The results of the simulations indicate that the proposed method is effective for the reconstruction of the current distributions in the human brain using magnetoencephalographic (MEG) measurements, even if the signal-to-noise ratio of the measured data is relatively low. We applied the proposed method to the magnetoencephalographic data obtained during a mental image processing task that included object recognition and mental rotation operations. The results suggest that the proposed method can extract the neural activity in the extrastriate visual region and the parietal region. These results are in agreement with the results of previous positron emission tomography (PET) and functional magnetic resonance imaging (fMRI) studies.

Several methods have been developed for solving blind deconvolution problem. Recursive inverse filtering method is proposed recently and shown to have good convergence properties. This method requires accurate estimate of the region of support. In this paper, we propose to modify the original method by incorporating split, merge and grouping algorithm to find the region of support automatically.

We present a simple framework for multicasting video in an active network, in which we overcome heterogeneity in the quality requests by filtering the video stream at some properly located active nodes. The framework includes the requirements for the underlying active network and outlines the video multicast application. We then introduce a heuristic algorithm for electing the filtering nodes to conform a multicast distribution tree, in which we use an objective function to, for example, minimize the required bandwidth. We evaluate the performance of our algorithm comparing it with simulcast and layered encoded transmission through simulation experiments, showing some advantages of using active filtering.

The use of the homomorphic filter technique is described in order to enhance the contrast in the mammographic images, which is adopted to the dyadic wavelet transform. The proposed method has employed the nonlinear enhancement in homomorphic filtering as well as denoising method in the wavelet domains. Experimental results show that the homomorphic filtering method improves the contrast in breast tumor images such that the contrast improvement index is increased by two fold compared to the conventional wavelet-based enhancement technique.

This paper considers FIR filter design using linear predictive coding technique, for which the coefficients belong to a small set of integers, so that the coefficients have small wordlengths. Previously, integer programming was used to find the coefficients of such filters. However, the design method using integer programming suffers from high computational cost as the filter length increases. The computation can quickly become prohibition. In this paper, we propose two designs of predictive encoded FIR filters based on a modified Karmarkar's linear programming algorithm, which is known to be more suitable for solving large problems. First, we formulate the problem as a weighted minimax error problem and arrange it in a form that the modified Karmarkar algorithm can be applied. The design algorithm has the same (low) complexity as that of the weighted least-square method, but it can solve problems with some constraints, whereas the weighted least-square method cannot. However, the algorithm has a difficulty due to an ill condition caused by matrix inversion when the predictive filter order is high. To avoid this difficulty, we formulate the design as a weighted least absolute error problem. By using this second proposed algorithm, a filter with shorter coefficient wordlength can be found using a higher-order predictor filter at the expense of more computational cost. To further reduce the coefficient wordlength, the filter impulse response is separated into two sections having different ranges of coefficient values. Each section uses a different scaling factor to scale the coefficient values. With small coefficient wordlength, the filter can be realized without hardware multipliers using a low-radix signed-digit number representation. Each coefficient is distributed in space as 2-3 ternary {0,1} or quinary {0,1, 2} coefficients. Ternary coefficients require only add/subtract operation, while quinary coefficients require one-bit shift and add/subtract operations. The shift can be hardwired without any additional hardware.

In this paper, we propose a frequency domain active noise control (ANC) system without a secondary path model. The proposed system is based on the frequency domain simultaneous perturbation (FDSP) method we have proposed. In this system, the coefficients of the adaptive filter are updated only by error signals. The conventional ANC system using the filtered-x algorithm becomes unstable due to the error between the secondary path, from secondary source to error sensor, and its model. In contrast, the proposed ANC system has the advantage not to use the model. In this paper, we show the principle of the proposed ANC system, and examine its efficiency through computer simulations.