A new algorithm is proposed for improving the convergence of backpropagtion networks. This algorithm is obtained by combining the conjugate gradient method and the Kalman filter algorithm. Simulation results show that the proposed algorithm can perform satisfactorily in all cases considered.

This paper presents the results on IF/baseband up/down direct digital conversion and multiple channel analysis/synthesis software defined radio modules which are implemented using high speed ADC, DAC and FPGA, for IS-95 code division multiple access (CDMA) systems. The implemented system can directly down-convert multiple channel IS-95 CDMA IF signals to the baseband, and selectively analyze specific channel signals based on polyphase analysis filter bank techniques. Moreover, the analyzed baseband signals of multiple channels can be directly up-converted and synthesized in the same system. We have deployed the implemented system in IS-95 CDMA optical digital repeaters for PCS applications.

This paper proposes a private communication system with chaos using fixed-point digital computation. When fixed-point computation is adopted, chaotic properties of the modulated signal should be checked carefully as well as calculation error problems (especially, overflow problems). In this paper, we propose a novel chaos modem system for private communications including a chaotic neuron type nonlinearity, an unstable digital filter and an overflow function. We demonstrate that the modulated signal reveals hyperchaotic property within 10,000 data point fixed-point computation, and evaluate the security of this system in view of the sensitivity of coefficients for demodulation.

This paper reviews recent progresses in the research and development of planar photonic band-gap (PBG) structures, also called electromagnetic crystals, for microwave and millimeter-wave applications. Planar electromagnetic crystals are particularly attractive and intensively investigated because of their easy fabrication, low cost, and compatibility with standard planar circuit technology. Two configurations and their applications are described in this paper: a square lattice of holes etched in a ground plane and the recently developed Uniplanar Compact PBG (UC-PBG) structure. Basic properties as well as applications to microwave circuits are reported. These include harmonic tuning in power amplifier, leakage suppression in conductor-backed coplanar waveguide (CB-CPW), realization of planar slow-wave structure, and performance improvement in microstrip filters and patch antennas.

In this paper we present an analysis based on the indirect Lyapunov criteria, that is used to study the convergence of an infinite impulse response (IIR) adaptive digital filter (ADF) based on estimation of the allpass system. The analysis is then extended to investigate the necessity of directly estimating the transfer level of the unknown system. We consider two cases of modeling the ADF. In the first system, the allpass section of the ADF estimates only the real poles of the unknown system while in the second system, both real and complex poles the allpass section are estimated. From the analysis and computer simulation, we realize that the poles of the ADF converge selectively to the poles of the unknown system, depending on the sign of the step size of adaptation. Using these results we proposed a new method to control the convergence of the poles the IIR ADF based on estimation of the allpass system.

In this paper, we present a new approach for the design of partially adaptive broadband beamformers with the generalized sidelobe canceller (GSC) as an underlying structure. The approach designs the blocking matrix involved by utilizing a set of P-regular, M-band wavelet filters, whose vanishing moment property is shown to meet the requirement of a blocking matrix in the GSC structure. Furthermore, basing on the subband decomposition property of these wavelet filters, we introduce a new dynamic subband selection scheme succeeding the blocking matrix. The scheme only retains the principal subband components of the blocking matrix outputs based on a prescribed statistical hypothesis test and thus further reduces the dimension of weights in adaptive processing. As such, the overall computational complexity, which is mainly dictated by the dimension of adaptive weights, is substantially reduced. The furnished simulations show that this new approach offers comparable performance as the existing fully adaptive beamformers but with reduced computations.

This letter analyses the convergence behaviour of the transform domain least mean square (TDLMS) adaptive filtering algorithm which is based on a well known interpretation of the variable stepsize algorithm. With this interpretation, the analysis is considerably simplified. The time varying stepsize is implemented by the modified power estimator to redistribute the spread power after transformation. The main contribution of this letter is the statistical performance analysis in terms of mean and mean squared error of the weight error vector and the decorrelation property of the TDLMS is presented by the lower and upper bound of eigenvalue spread ratio. The theoretical analysis results are validated by Monte Carlo simulation.

This paper describes a jitter suppression technique for a clock multiplier IC that uses a phase-locked loop (PLL). It is shown that the jitter cutoff frequency of the jitter transfer function can be greatly improved by adding a surface acoustic wave (SAW) filter whose center frequency equals the input frequency. The jitter transfer function is mainly determined by the characteristics of the SAW filter. Therefore, the clock multiplier IC can be set at a high loop gain to minimize the jitter generation without increasing the jitter cutoff frequency. The use of a clock multiplier IC that was fabricated with Si bipolar technology and a SAW filter with the center frequency of 155.52 MHz and a quality (Q) factor of 1500 results in a very low jitter generation of 3.5 mUI rms and an extremely low jitter cutoff frequency of about 50 kHz when the clock multiplier converts a clock frequency of 155.52 MHz into a 2.48832-GHz signal.

This paper discusses a method of designing linear phase two-channel filter banks for integer wavelet transform. We show that the designed filter banks are easily structed as the lifting form by leading relationship between designed filters and lifting structure. The designed integer wavelets are applied to image compression to verify the efficiency of our method.

A nonlinear multiple complex sinusoidal estimator (NMSE) is proposed, as an extended and improved version with system noise of the single sinusoidal estimator previously presented by the author, for extracting multiple complex sinusoids in white noise. This estimator is derived by applying an extended complex Kalman filter (ECKF) to a noisy multiple complex sinusoidal model with state-representation, where the model becomes a nonlinear stochastic system. Proof of the stability is given by using a structure of the state-space signal model and Lyapunov techniques. Also, computer simulations demonstrate the effectiveness of the NMSE from various points of view.

The VSOP terminal is a new data-acquisition system for the Very-Long-Baseline Interferometry (VLBI). This terminal was primarily designed for ground telescopes in the VLBI Space Observatory Programme (VSOP). New technologies; higher-order sampling and digital filtering techniques, were introduced in the development. A cassette cart was also introduced, which supports 24-hour unattended operations at the maximum data rate of 256 Mbps. The higher-order sampling and digital filtering techniques achieve flat and constant phase response over bandwidth of 32 MHz without using expensive wide base-band converters. The digital filtering technique also enables a variety of observing modes defined on the VSOP terminal, even with a fixed sampling frequency in an A/D converter. The new terminals are installed at Nobeyama, Kashima, Usuda, Mizusawa, and Kagoshima radio observatories in Japan, and are being used in VSOP and other domestic VLBI observations. In this paper the key features of the VSOP terminal focusing on these advanced technologies are presented, and the results of performance tests are shown.

As the nonlinear adaptive filter, the neural filter is utilized to process the nonlinear signal and/or system. However, the neural filter requires large number of iterations for convergence. This letter presents a new structure of the multi-layer neural filter where the orthonormal transform is introduced into all inter-layers to accelerate the convergence speed. The proposed structure is called the transform domain neural filter (TDNF) for convenience. The weights are basically updated by the Back-Propagation (BP) algorithm but it must be modified since the error back-propagates through the orthogonal transform. Moreover, the variable step size which is normalized by the transformed signal power is introduced into the BP algorithm to realize the orthonormal transform. Through the computer simulation, it is confirmed that the introduction of the orthonormal transform is effective for speedup of convergence in the neural filter.

A 24-bit, 96 kHz stereo A/D converter (ADC) for DVD-audio has been developed. The single-chip integrates stereo delta-sigma modulators (Δ ΣM's), a voltage reference, and a decimation filter. A fourth-order cascaded Δ ΣM using a local feedback technique was employed to avoid overload without sacrificing noise performance. Low power switched-capacitor techniques were used for implementation. A two-stage decimation filter architecture that reduces digital switching noise was also developed. A merged multi-stage comb filter was used for the first stage, and a bit-serial finite-impulse-response (FIR) filter was used for the second stage. The 18.0 mm2 chip was fabricated in 0.6-µm CMOS with low threshold devices. Measured results show 117 dB A-weighted dynamic range in the 20 kHz passband, with 470 mW power dissipation at 5 V operation.

This paper presents a new data hiding scheme via steganographic image transformation, which is different from conventional data hiding techniques. The transformation is achieved in the frequency domain and the concept of Fourier filtering method is used. An input image is transformed into a fractal image, which can be used in Computer Graphic (CG) applications. One of the main advantages of this scheme is the amount of data to be hidden (embedded) is equal to that of the host signal (generated fractal image) while it is in general limited in the conventional data hiding schemes. Also both the opened fractal image and the hidden original one can be properly used depending on the situation. Unauthorized users will not notice the "secret" original image behind the fractal image, but even if they know that there is a hidden image it will be difficult for them to estimate the original image from the transformed image. Only authorized users who know the proper keys can regenerate the original image. The proposed method is applicable not only as a security tool for multimedia contents on web pages but also as a steganographic secret communication method through fractal images.

This paper presents a 7th-order channel-select filter for a spread-spectrum wireless receiver operating with a minimum power supply of 2.5 V. The channel-select filter implements a sharp transition from 2 MHz to 4 MHz and a stopband attenuation of 50 dB. The 7th-order filter is realized by a cascade of a passive RC integrator, a 3rd-order leapfrog filter, an operational amplifier based differentiator, a 2nd-order notch filter, and a 1st-order allpass filter. It is designed in a 0.35 µm single-poly BiCMOS process. Simulation results show feasibility of the proposed filter.

Visualization of 3-D ultrasound images is a challenging task due to the noisy and fuzzy nature of ultrasound imaging. This paper presents an efficient volume rendering technique for 3-D ultrasound image. A preprocessing technique of 2-D truncated-median filtering is proposed to reduce speckle noise of the ultrasound image. This paper also introduces an adaptive boundary detection method to reduce the computation time for volume rendering of ultrasound image. The proposed technique is compared to the conventional volume rendering methods with respect to the computation time and the subjective image quality. According to the comparison study, the proposed volume rendering method shows good performance for visualization of 3-D ultrasound image.

It is proposed a novel method that optimizes nonlinear filters by unsupervised learning using a novel definition of morphological pattern spectrum, called "morphological opening/closing spectrum (MOCS)." The MOCS can separate smaller portions of image objects from approximate shapes even if the shapes are degraded by noisy pixels. Our optimization method analogizes the linear low-pass filtering and Fourier spectrum: filter parameters are adjusted to reduce the portions of smaller sizes in MOCS, since they are regarded as the contributions of noises like high-frequency components. This method has an advantage that it uses only target noisy images and requires no example of ideal outputs. Experimental results of applications of this method to optimization of morphological open-closing filter for binary images are presented.

A nonlinear inverse filter is proposed for restoring signals degraded by a linear system and additive Gaussian noise. The proposed filter consists of combination of a linear high pass filter and an ε-filter, which is modified from the cascaded linear filter. The nonlinear property of the ε-filter is utilized to suppress pre-enhanced additive random noise and to restore sharp edges. It is demonstrated that the filter can be reduced to a multi-layered neural network model, and the optimal design is described by using the back propagation algorithm. The nonlinear function is approximated by a piecewise linear function, which results in simple and robust training algorithm. An application to image restoration is also presented, illustrating the effectiveness over the linear filter, especially when the amplitude of additive noise is small.

The purpose of this paper is to deal with the problem of recovering a signal from its noisy version. One example is to restore old images degraded by noise. The recovery solution is given within the framework of series expansion and we shall show that for the general case the recovery functions have to be elements of an extended pseudo biorthogonal basis (EPBOB) in order to suppress efficiently the corruption noise. After we discuss the different situations of noise, we provide some methods to construct the optimal EPBOB in order to deal with these situations.

A 5-pole lumped element bandpass filter (BPF) of center frequency 264.05 MHz and fractional bandwidth (FBW) 0.76% is designed and fabricated using YBa2Cu3O7-d (YBCO) thin films deposited on both sides of a MgO substrate(40 mm 40 mm 0.5 mm). The return loss, minimum insertion loss and ripple were measured to be 20.0 dB, less than 0.1 dB and less than 0.1 dB at 70 K, respectively. These results verify both the compactness and low loss characteristics in the VHF band. The simulated frequency response, where the frequency dependences of inductance (L) and capacitance (C) elements and housing effect are taken into account, is in good agreement with the measured frequency response.