In this paper, we propose a new neural filter to which the features related to a given task are input, called a neural filter with features (NFF), to improve further the performance of the conventional neural filter. In order to handle the issue concerning the optimal selection of input features, we propose a framework composed of 1) manual selection of candidates for input features related to a given task and 2) training with automatically selection of the optimal input features required for achieving the given task. Experiments on the proposed framework with an application to improving the image quality of medical X-ray image sequences were performed. The experimental results demonstrated that the performance on edge-preserving smoothing of the NFF, obtained by the proposed framework, is superior to that of the conventional neural and dynamic filters.

An information retrieval model that intends to conceptually deal with the documents in the Internet is proposed. The first half of this model is a stage to select the documents, which may meet the user's long-term interests, by employing a filtering or retrieval system. The latter half is a stage for linear document arrangement and for adjacent cluster based browsing. For the document collection filtered out, similarity matrices are computed and then the documents are arranged such that the highly similar ones are adjacently placed. By this treatment the documents are considered to form the clusters, some of which are adjacently placed when they include similar documents. A user can satisfy her/his needs by first browsing in the clusters containing documents highly similar to a query, and next by extending the browsing process into the clusters adjacent to the ones just examined. In the adjacent clusters the documents having no keywords common to but conceptually related to the query can be found. Computational and statistical evaluations were done on two standard test collections. A virtual space navigator is also designed by using JAVA to assist a user in the browsing task.

This paper proposes a design method of 2-D variable IIR digital filters with high frequency tuning accuracy. In the proposed method, a parallel complex allpass structure is used as the prototype structure of the 2-D variable digital filters in order to obtain low sensitivity characteristic. Because the proposed 2-D variable digital filter is composed of first-order complex allpass sections connected in parallel, the proposed variable digital filter possesses several advantages such as low sensitivity characteristic in the passband, simple stability monitoring and high parallelism. In order to improve the frequency tuning accuracy of the proposed variable digital filter, each first-order complex allpass section is substituted by a new first-order complex allpass section with low sensitivity characteristic. Moreover, the coefficient sensitivity analysis of a 2-D parallel complex allpass structure is presented. Numerical examples show that the proposed 2-D variable IIR digital filter has high tuning accuracy under the finite coefficient wordlength.

This paper presents the transmission performance of a class-IV partial-response signaling SSB system and proposes a method that can improve its power efficiency. A line code that has no dc component has been used in the SSB transmission of digital signals. The type of line code, such as a partial-response signaling, increases the modulation states, and as a result, decreases the power efficiency. To overcome this obstacle, a new demodulation method called "re-filtering and combining" is proposed on the assumption of orthogonal phase detection. The demodulated quadrature channel is re-filtered by a Hilbert filter and is combined with the in-phase channel. It is confirmed by computer simulations that the new demodulation method improves the BER performance and a 3 dB improvement of the power efficiency is obtained.

This article presents a three-step method for designing linear phase FIR filters with signed-powers-of-two (SPT) coefficients. In Step one, a prototype optimal FIR filter is designed by the Remez exchange algorithm. In Step two, a scaling factor is selected by employing simple ad-hoc rules. In Step three, each coefficient of the prototype filter is scaled by the scaling factor and is quantized coarsely as the canonic-signed-digit (CSD) representation. Then, a mixed-integer-linear-programming (MILP) algorithm is applied to three least significant digits (LSDs) of the filter's coefficients to reduce the number of SPT terms. Design examples demonstrate that the proposed algorithm is able to produce linear phase fixed-point FIR filters using fewer SPT terms than the existing methods under the same normalized peak ripple magnitude (NPRM) specification.

The purpose of this paper is to improve the detection sensitivity of radio interferometry. The development of wideband signal processing techniques is necessary for high-sensitivity radio interferometry. The higher-order sampling technique can achieve wider bandwidth than expensive wide-baseband conversion system. The over sampling technique can improve the quantization loss from 11.89% to 3.99% at the 4-level quantized sampling. Moreover, the digital filter can reduce the folding noise incurred by the non-rectangular frequency response at a sampling process. It is confirmed that the wideband and low-loss A/D conversion system may be realized by implementing the higher-order sampling, over sampling and digital filtering techniques. In this paper, the key features focusing on these advanced techniques for radio astronomy are presented in detail.

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 space-time (ST) receiver is proposed for multiple access interference (MAI) and narrowband interference (NBI) suppression, and multipath diversity reception in wireless multi-carrier CDMA communications incorporating antenna arrays. The scheme involves three stages. First, an adaptive matched filter is attached to each finger at each antenna to combat the MAI. Second, an adaptive beamformer is constructed for each finger which provides effective reception of the signal of interest (SOI) and suppression of time-varying NBI. Finally, beamformer output data from different fingers are combined to capture the signal multipath components coherently. The proposed ST receiver is shown to perform reliably under strong interference, and outperform the ST MMSE receiver with pilot symbols aided channel estimation.

The cutoff filter and circular window are critically important elements of circular waveguide TE11 mode transmission systems, but no detailed analysis of their characteristics has yet been undertaken. In order to gain a detailed understanding of return loss and other frequency characteristics, one must analyze the electromagnetic (E-M) fields in the waveguide cavity to the cutoff domain and higher mode oscillations. In this work we present a theoretical analysis employing a method in which E-M fields are represented in the form of transmission equations, and show that the results are in remarkably close agreement with experimental results obtained from a prototype device. It is also demonstrated that the results obtained by the proposed method are far more accurate than those obtained using conventional approximation theory.

A multi-range resolution radar using sideband spectrum energy is investigated. The basic system consists of a sharpening processor and least-error energy shaping filters. First, the sharpening processor makes long flat pulses sharpened. Next, the least-error shaping filters compress the input pulse into the desired pulse width. Then the output pulse width can become narrower than the reciprocal of the input bandwidth, because the least-error shaping filters make the equivalent bandwidth expanded by the enhancement of the sideband spectrum energy and the suppression of the main spectrum. The transmitted signals with simple phase modulation are studied to obtain the multi-range resolution where the pulse is compressed into a pulse with the same bandwidth and another pulse width equal to the reciprocal of the input bandwidth. The peak-to-sidelobe ratio after the pulse compression and the improvement factor of the output signal-to-noise ratio are measured. Further, the experimental results are shown to verify our proposed technique.

This paper proposes a new method of retrieving images from large image databases. The method is based on VQ (Vector Quantization) of local texture features at interest points automatically detected in an image. The texture features are extracted by Gabor wavelet filter bank, and rearranged for rotation. These features are classified by VQ and then construct a pattern histogram. Retrievals are performed by just comparing pattern histograms between images.

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.

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.

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.

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.

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.

In this paper, a new maximum likelihood filter with finite impulse response (FIR) structures is proposed for state space signal models with both system and observation noises. This filter is called the maximum likelihood FIR (MLF) filter. The proposed MLF filter doesn't require a priori information of the window initial state and processes the finite observations on the most recent window linearly. The proposed MLF filter is first represented in a batch form, and then in an iterative form for computational advantage. The proposed MLF filter has good inherent properties such as time-invariance, unbiasedness, deadbeat, robustness. The validity of the proposed MLF filter is illustrated by a computer simulation on a sinusoidal signal.

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.

In general, the attenuation characteristics of band-pass filters can be improved by generating attenuation poles in the stop band. In this paper, a design method for a planar band-pass filter with attenuation poles based on a half-wavelength resonator is proposed. According to this design, the attenuation poles can be obtained at any desired frequency by means of coupling structures. Two kinds of filter with the characteristics of steep skirt and wide stop-band were designed and fabricated with the result that the validity of the design method was demonstrated. Therefore, a filter with excellent attenuation characteristics for various applications can be achieved.

In this paper, the co-existence of TDMA and W-CDMA spectrum sharing system (TDMA/W-CDMA overlaid system) with cellular architecture is discussed. In this system, both systems share the same frequency band to improve the spectrum efficiency. Overall rate, bit error ratio (BER) and spectrum efficiency of the system are calculated for the forward link (down-link) in the presence of AWGN channel. Taking into account the path loss and shadow fading loss in this system with cellular architecture, W-CDMA applying interference cancellation (IC) shows a substantial difference in spectrum efficiency, the overlaid system can provide a greater overall rate and higher spectrum efficiency than a single multiple access-based system such as TDMA system or W-CDMA system. The interference cancellation can significantly improve BER of the spectrum overlaid system.