In this paper, we propose a novel higher order time-frequency distribution (GDH) for a discrete time signal. This distribution is defined over the original discrete time-frequency grids through a delicate discretization of an equivalent expression of a higher order distribution, for a continuous time signal, in [4]. We also present a constructive design method, for the kernel of the GDH, by which the distribution satisfies (i) the alias free condition as well as (ii) the marginal conditions. Numerical examples show that the proposed distributions reasonably suppress the artifacts which are observed severely in the Wigner distribution and its simple higher order generalization.

This paper presents data coding techniques for a stable single-bit noise-shaping quantizer, which has a cascade structure of a multi-bit ΣΔ modulator and a binary interpolator. The binary interpolator chooses a pre-optimized binary vector for each input sample and successively generates the chosen binary vector as an output bit stream. The binary vectors can have different lengths. The paper also proposes two methods to evaluate and bound output errors of a binary interpolator. A multi-bit ΣΔ modulator is designed to cause no overload for all possible input signals whose amplitudes are bounded to a specified level, and thus the ΣΔ modulator rigorously guarantees the stability condition. In design examples, we have evaluated Signal-to-Noise and Distortion Ratios (SNDRs) and noise spectra and then confirmed that our stable quantizers can sharply shape output noise spectra.

Adaptive optimization of the notch bandwidth of a lattice-based adaptive infinite impulse response (IIR) notch filter is presented in this paper. The filter is used to improve the performance of a direct sequence spread spectrum (DSSS) binary phase shift keying (BPSK) communication system by suppressing a narrow-band interference at the receiver. A least mean square (LMS) algorithm used to adapt the notch bandwidth coefficient to its optimum value which corresponds to the maximum signal to noise ratio (SNR) improvement factor is derived. Bit error rate (BER) improvement gained by the DSSS communication system using the filter with the optimized notch bandwidth is also shown. Computer simulation results are compared with those obtained analytically to demonstrate the validity of theoretical predictions for various received signal parameters.

Subband adaptive array (SBAA) has been realised as a promising method to perform space-time signal processing in mobile communications. Recently, several schemes of SBAA have been introduced. However, theoretical analysis of SBAA performance has been only limited to case of SBAA without using decimation. In this paper, we introduce a novel method to analyse the performance of SBAA using critical sampling with localised feedback scheme. We first provide a detailed analysis of the SBAA performance in the case of single path and multipath fading environment. The simulation results are then presented to verify the proposed method.

The effect of subarray size (equal to the order of the prediction model plus one) on the estimation performance of a previously proposed forward-backward linear prediction (FBLP) based cyclic method is investigated. This method incorporates an overdetermined FBLP model with a subarray scheme and is used to estimate the directions-of-arrival (DOAs) of coherent cyclostationary signals impinging on a uniform linear array (ULA) from the corresponding polynomial or spectrum formed by the prediction coefficients. However, the decorrelation is obtained at the expense of a reduced working array aperture, as it is with the spatial smoothing (SS) technique. In this paper, an analytical expression of the mean-squared-error (MSE) of the spectral peak position is derived using the linear approximation for higher signal-to-noise ratio (SNR). Then the subarray size that minimizes this approximate MSE is identified. The effect of subarray size on the DOA estimation is demonstrated and the theoretical analysis is substantiated through numerical examples.

Lossless video coding is required in the fields of archiving and editing digital cinema or digital broadcasting contents. This paper proposes multiresolution lossless video coding using a discrete wavelet transform and adaptive inter/intra-frame prediction in the wavelet domain. The multiresolution structure based on the wavelet transform facilitates interchange among several video source formats such as Super High Definition (SHD) images, HDTV, SDTV, and mobile applications. In order to increase the compression ratio, and to keep the computational cost low, the adaptive inter/intra-frame prediction is performed in the lowest wavelet transform domain. The adaptive inter/intra-frame prediction can adapt to changes in the local inter/intra-frame statistics. Experiments on digital cinema test sequences confirm effectiveness of the proposed algorithm.

This paper presents a method of simple adaptive control (SAC) for nonlinear systems using Elman recurrent neural networks (ERNNs). The control input is given by the sum of the output of a simple adaptive controller and the output of the ERNN. The ERNN is used to compensate the nonlinearity of plant dynamics that is not taken into consideration in the usual SAC. The role of the ERNN is to construct a linearized model by minimizing the output error caused by nonlinearities in the control systems.

A sound localization method based on the adaptive estimation of inverse Ear Canal Transfer Functions (ECTFs) using a stereo earphone-microphone combination is proposed. This method can adaptively obtain the individual's transfer functions to fit the listener in real-time. We evaluate our sound localization method by studying the relationship between the estimation error of inverse ECTFs and the auditory sound localization score perceived by several listener. As a result, we clarified that the estimation error required of inverse ECTFs are less than -10 dB. In addition, we describe two adaptive inverse filtering methods in order to realize real-time signal processing implementation using affine projection algorithm and discusses the convergence time of an adaptive inverse filter to determine the initial value. It is clarified that method 2 based on copy weights with initial value is more effective than method 1 with filtered-x algorithm, in terms of convergence, if the initial value is the average of many listeners' impulse responses for our sound localization method.

We describe a method of compensating temperature fluctuation by a linear-time-warping processing in a sound reproduction system. This technique is applied to impulse responses of room transfer functions, to achieve a high-quality sound reproduction system, particularly one that treats high-frequency components. First, the impulse responses are measured before and after temperature fluctuation, and the former are converted to the latter by the proposed process. Next, we design inverse filters for the system, and evaluate the improvement of the reproduction accuracy and spectrum distortion. By the compensation method, we can improve the reproduction accuracy at any frequency. Moreover, we propose an adaptive algorithm for the estimation of a suitable warping ratio, using the observed signal of reproduced sound obtained at only one control point. Using the proposed algorithm, we can improve the reproduction accuracy at each control point by about 14 dB, in which a difference in temperature is 1.4.

In this paper, we present an efficient VLSI architecture for the stand-alone application of a speech recognition system based on discriminative Bayesian neural network (DBNN). Regarding the recognition phase, the architecture of the Bayesian distance unit (BDU) is constructed first. In association with the BDU, we propose a template-serial architecture for the path distance accumulation to perform the recognition procedure. A corresponding architecture is also developed to accelerate the discriminative training procedure. It contains the intelligent look-up table for the sigmoid function. In comparison to the traditional one-table method, the memory size reduces drastically with only slight loss of accuracy. Combining the proposed hardware accelerators with the cost efficient programmable core, we took the most out of both programmable and application-specific architectures, including performance, design complexity, and flexibility.

Hypercomplex coefficient digital filters provide several attractive advantages such as compact realization with reduced system order, inherent parallelism. However, they also possess a drawback in that a multiplier requires a large amount of computations. This paper proposes a computationally efficient implementation of digital filters whose coefficient is a type of hypercomplex number; a bicomplex number. By decomposing a bicomplex multiplier into two parallel complex multipliers, we show that hypercomplex digital filters can be implemented as two parallel complex digital filters. The proposed implementation offers more than a 60% reduction in the count of real multipliers.

This paper proposes a new method to reduce sinusoidal noise components whose frequencies are known. The new method is based on the simultaneous equations technique. The technique does not require the secondary path filter: thereby the automatic recovering of the noise reduction effect deteriorated by secondary path changes becomes possible. This paper also presents computer simulation results to examine the performance of the new method.

In this paper, we propose lossless/lossy coding criterion as a new objective criterion to theoretically evaluate coding performance of the lossless/lossy wavelet (LLW). The proposed lossless/lossy coding criterion consists of three parameters: "lossless coding criterion," "quantization-lossy coding gain" and "rounding errors. " The first parameter is a criterion to evaluate lossless coding performance of the LLW, whereas the second and the third parameters are criteria to evaluate lossy coding performance of the LLW at low bit rate and high bit rate, respectively. Relation among those three parameters is clearly illustrated in this paper. Performances of 15 kinds of the LLW are measured with two-dimensional (2D) octave-decomposition by applying some standard images and 2D AR(1) model as input signals.

An image halftoning technique that uses a simple GA has proven to be effective generating bi-level halftone images with quality higher than conventional techniques. Many devices are designed to handle more than two halftone levels and a GA based multi-level halftoning technique is desirable. In this paper we extend the bi-level halftoning technique to generate multi-level halftone images. Also we introduce an improved GA (GA-SRM) into the proposed multi-level halftoning technique. Experimental results show that the proposed technique can effectively generate high quality multi-level halftone images and that the inclusion of GA-SRM substantially contributes reducing memory usage and accelerating image generation.

This paper describes an effective technique for coding QCIF video sequences based on a JPEG2000 codec. In the proposed method, multiple frames are combined into one large picture. The larger picture enables images to be coded more efficiently. Image quality is further improved by combining the frames symmetrically. The video sequence is efficiently coded by adapting the time correlation of the video sequences to spatial correlation. We demonstrated the effectiveness of this method by encoding QCIF video sequences using JPEG2000.

This paper suggests that a watermarking technique based on the number theoretic transform (NTT) may effectively be employed for detecting alterations on lossless digital master images. Due to its fragility, the NTT-based technique is sensitive to detecting alterations, compared with that based on the discrete Fourier transform (DFT).

A novel formulation for the Scalar-field approach of Integral Equation formulation of the Measured Equation of Invariance (SIE-MEI) is derived from the scalar reciprocity relation to solve the scalar Helmholtz equation. The basics of this formulation are similar to IE-MEI method for the electromagnetic (EM) problem. The surface integral equation is derived from reciprocity relation and on-surface MEI postulates are used. As a result it generates a sparse linear system with the same number of unknowns as of Boundary Element Method (BEM) and keeps the merits in minimum storage memory requirements and CPU time consumption for computing the final matrix. IE-MEI method has been proposed for two-dimensional (2D) electromagnetic problem, but three-dimensional (3D) problem is very difficult to be extend. This scalar-field approach of IE-MEI method is identical to electromagnetic in 2D, but easily extended to the 3D scalar-field scattering problem contrary to EM problem. The numerical results of sphere and cube are verified with some rigorous or numerical solutions, which give excellent agreement.

An LCD Driver IC includes more than 300 buffer amplifiers on a single chip. The phase compensation capacitors (on-chip Miller capacitors) for the amplifiers are more than 1000 pF and occupy a large chip area. This paper describes a two-gain-stage amplifier in which an on-chip Miller capacitor is not used for phase compensation in an LCD Driver IC. In the proposed amplifier, phase compensation is achieved only by a newly introduced zero, which is formed by the load capacitance and a phase compensation resistor connected between the output of the amplifier and the capacitive load. Designs of the phase compensation resistor and the amplifier before compensation are discussed, considering a typical load capacitance range. The test chip was fabricated. The newly introduced zero successfully stabilized the amplifier. The chip area for the amplifier was reduced by 30-40%, compared with our previously reported one. The current consumption of the amplifier was only 5 µA. The experimental results of the fabricated test chip support that the proposed amplifier is suitable to an LCD driver IC with a smaller chip area.

In this paper, we present new stability conditions for a class of large-scale hybrid dynamical systems composed of a number of interconnected hybrid subsystems. The stability conditions are given in terms of discontinuous Lyapunov functions of the stable hybrid subsystems. Furthermore, the stability conditions are represented by LMIs (Linear Matrix Inequalities) which are computationally tractable.

In this paper, a novel continuous complementary sliding control was proposed to improve the tracking performance given the available control bandwidth and the extend of parameter uncertainty. With this control law, the ultimate bound of tracking error was shown to be reduced at least by half, as compared with the conventional continuous sliding control. More strikingly, the proposed control can effectively improve the error transient response during the reaching phase. We presented a composite complementary sliding control scheme for a class of uncertain nonlinear systems including the nonlinear electrohydraulic position servo control system, which will be used as an illustrated example. Simulation results indicated exceptional good tracking performance to step and sine wave reference inputs can be obtained. In addition, the disturbance rejection property of the controller to single-frequency sinusoidal disturbances is also outstanding.

In SoC (system-on-a-chip) design, interfacing among IP (Intellectual Property) blocks is one of the most important issues. Since most IP's are provided by different vendors, they generally have different interface schemes and different operating frequencies. In this paper, we propose a new interface synthesis method with two features: 1) generation of the interface between IP's with different operating frequencies, and 2) minimization of the hardware resource required for the interface. We have demonstrated the proposed algorithm through its application to an MP3 decoder design example, where the IIS (Inter-IC Sound)-to-PCI (Peripheral Component Interconnect) protocol converter was successfully implemented using the proposed method.

We show a method to determine a Steiner Minimum Tree (SMT) and a necessary and sufficient condition that an SMT is a full Steiner tree for three given points in -geometry ( = 3m, m is a positive integer). The -geometry allows only orientations with angles i/ (i and ( 2) are integers), and fill up the gap between the rectilinear geometry ( = 2) and the Euclidean geometry ( = ). An SMT in -geometry ( = 3m) has a similar property to that in the Euclidean geometry. The method to determine an SMT in -geometry is an extension of the well-known method in the Euclidean geometry. The Steiner point in -geometry is any point in the intersection area with a parallelogram and a Steiner locus. Then there are infinite candidate locations of the Steiner point. The Steiner point in the Euclidean geometry is that in -geometry ( = 3m).

Guideline effects on the movement accuracy in virtual torus spaces with various guidelines are investigated by means of moving deviations from an indicated path. Guidelines with various widths are set at the center or the sides (upper, lower, left and right) in virtual torus spaces. Participants travel at a constant velocity in full-sized virtual torus spaces for various guideline conditions. It is shown that there exists a tendency that moving deviations from the indicated path are more reduced as the guideline width becomes narrower and the movement accuracy is improved. For example, in the case of side guidelines with the width of 10 cm or less, the mean value of moving deviations is approximately reduced to half of that with no guidelines. By setting side guidelines, the value of the standard deviation of moving deviations is approximately reduced to 2/3 of that with no guidelines.

To classify the significant wavelet coefficients into edge area and noise area, a morphological clustering filter applied to wavelet shrinkage is introduced. New methods for wavelet shrinkage using morphological clustering filter are used in noise removal, and the performance is evaluated under various noise conditions.

It is widely known that the family of projection filters includes the generalized inverse filter, and that the family of parametric projection filters includes parametric generalized projection filters. However, relations between the family of parametric projection filters and constrained least squares filters are not sufficiently clarified. In this paper, we consider relations between parameter estimation and image restoration by these families. As a result, we show that the restored image by the family of parametric projection filters is a maximum penalized likelihood estimator, and that it agrees with the restored image by constrained least squares filter under some suitable conditions.

A voltage-mode universal biquadratic filter using single current-feedback amplifier (CFA), two capacitors and three resistors is presented. The new circuit has four inputs and one output and can realize all the standard filter functions, that is, lowpass, bandpass, highpass, notch and allpass filters, without changing the circuit topology. The use of only one current-feedback amplifier simplifiers the configuration.

In this letter we present a new way for computing generalized eigenvalue problems in engineering applications. To transform a generalized eigenvalue problem into an associated problem for solving nonlinear dynamic equations by using optimization techniques, we can determine all eigenvalues and their eigenvectors for general complex matrices. Numerical examples are given to verify the formula of dynamic equations.

In this paper, we show that a priori probabilities of information bits can be incorporated into metrics for syndrome decoding. Then it is confirmed that soft-in/soft-out decoding is also possible for syndrome decoding in the same way as for Viterbi decoding. The derived results again show that the two decoding algorithms are dual to each other.