We propose a third-order low-pass notch filter realized by a single operational amplifier and a minimum number of equal-valued capacitors. As a design example we realize a Chebyshev filter with a ripple of 0.5 dB and it is shown that the experiment result is very good.

There have been several studies related to a reduction of the amount of computational resources used by Turing machines. As consequences, linear speed-up theorem" tape compression theorem", and reversal reduction theorem" have been obtained. In this paper, we consider reversal- and leaf-bounded alternating Turing machines, and then show that the number of leaves can be reduced by a constant factor without increasing the number of reversals. Thus our results say that a constant factor on the leaf complexity does not affect the power of reversal- and leaf-bounded alternating Turing machines

A systematic theory of the optimum sub-band interpolation using parallel wavelet filter banks presented with respect to a family of n-dimensional signals which are not necessarily band-limited. It is assumed that the Fourier spectrums of these signals have weighted L2 norms smaller than a given positive number. In this paper, we establish a theory that the presented optimum interpolation functions satisfy the generalized discrete orthogonality and minimize the wide variety of measures of error simultaneously. In the following discussion, we assume initially that the corresponding approximation formula uses the infinite number of interpolation functions having limited supports and functional forms different from each other. However, it should be noted that the resultant optimum interpolation functions can be realized as the parallel shift of the finite number of space-limited functions. Some remarks to the problem of distinction of images is presented relating to the generalized discrete orthogonality and the reciprocal property for the proposed approximation.

A method for detecting multiple rigid motions in images from an optical flow field obtained with multi-scale, multi-orientation filters is proposed. Convolving consecutive gray scale images with a set of eight orientation-selective spatial Gaussian filters yields eight gradient constraint equations for the two components of a flow vector at every location. The flow vector and an uncertainty measure are obtained from these equations. In the neighborhood of motion boundary, the uncertainty of the flow vectors increase. By using multiple sets of filters of different scales, multiple flow vectors are obtained at every location, from which the one with minimal uncertainty measure is selected. The obtained flow field is then segmented in order to solve the aperture problem and to remove noise without blurring discontinuity in the flow field. Discontinuities are first detected as those locations where flow vectors have relatively larger uncertainty measures. Then similar flow vectors are gouped into regions. By modeling flow vectors, regions are merged to form segments each of which belongs to a planar patch of a rigid object in the scene.

In practical applications of digital filters it is more realistic to treat multiplier coefficients as finite intervals than restricting them to infinite or very long word-length representations. However, this can not be done it the frequency response performance under interval assumption is difficult to analyze. In this paper, it is proved that stable lattice allpass filters possess bounded continuous phase response when lattice parameters vary in bounded intervals. It is shown that sharp bounds on the interval phase response can be computed easily at an arbitrary frequency using a simple recursive procedure. Application of this property to the problem of finite word-length lattice allpass filter design is also discussed. By formulating this problem as an interval design it is possible to solve it efficiently independent of the number system used to represent multiplier coefficients.

Most of actual environmental systems show a complicated fluctuation pattern of non-Gaussian type, owing to various kinds of factors. In the actual measurement, the fluctuation of random signal is usually contaminated by an external noise. Furthermore, it is very often that the reliable observation value can be obtained only within a definite fluctuating amplitude domain, because many of measuring equipments have their proper dynamic range and original random wave form is unreliable at the end of amplitude fluctuation. It becomes very important to establish a new signal detection method applicable to such an actual situation. This paper newly describes a dynamical state estimation algorithm for a successive observation contaminated by the external noise of an arbitrary distribution type, when the observation value is measured through a finite dynamic range of measurement. On the basis of the Bayes' theorem, this method is derived in the form of a wide sense digital filter, which is applicable to the non-Gaussian properties of the fluctuations, the actual observation in a finite amplitude domain and the existence of external noise. Differing from the well-known Kalman's filter and its improvement, the proposed state estimation method is newly derived especially by paying our attention to the statistical information on the observation value behind the saturation function instead of that on the resultant noisy observation. Finally, the proposed method is experimentally confirmed too by applying it to the actual problem for a reverberation time measurement from saturated noisy observations in room acoustics.

In multidimensional signal sampling, the orthogonal sampling scheme is the simplest one and is employed in various applications, while a non-orthogonal sampling scheme is its alternative candidate. The latter sampling scheme is used mainly in application where the reduction of the sampling rate is important. In three-dimensional (3-D) signal processing, there are two typical sampling schemes which belong to the non-orthogonal samplings; one is face-centered cubic sampling (FCCS) and the other is body-centered cubic sampling (BCCS). This paper proposes a new design method for 3-D band-limiting FIR filters required for such non-orthogonal sampling schemes. The proposed method employs the McClellan transformation technique. Unlike the usual 3-D McClellan transformation, however, the proposed design method uses 2-D prototype filters and 2-D transformation filters to obtain 3-D FIR filters. First, 3-D general sampling theory is discussed and the two types of typical non-orthogonal sampling schemes, FCCS and BCCS, are explained. Then, the proposed design method of 3-D bandlimiting filters for these sampling schemes is explained and an effective implementation of the designed filters is discussed briefly. Finally, design examples are given and the proposed method is compared with other method to show the effectiveness of our methos.

In this paper, we propose a spread spectrum pulse position modulation (SS-PPM) system, and describe its basic performances. In direct sequence spread spectrum (DS/SS) systems, pseudo-noise (PN) matched filters are often used as information demodulation devices. In the PN matched filter demodulation systems, for simple structure and low cost of each receiver, it is desired that each demodulator uses only one PN matched filter, and that signals transmitted from each transmitter are binary. In such systems, on-off keying (SS-OOK), binary-phase-shift keying (SS-BPSK) and differential phase-shift keying (SS-DPSK) have been conventionally used. As one of such systems, we propose the SS-PPM system; the SS-PPM system is divided into the following two systems: 1) the SS-PPM system without sequence inversion keying (SIK) of the spreading code (Without SIK for short); 2) the SS-PPM system with SIK of the spreading code (With SIK for short). As a result, we show that under the same bandwidth and the same code length, the data transmission rate of the SS-PPM system is superior to that of the other conventional SS systems, and that under the same band-width, the same code length and the same data transmission rate, the SS-PPM system is superior to the other conventional SS systems on the following points: 1) Single channel bit error rate (BER) (BER characteristics of the SS-PPM system improve with increasing the number of chip slots of the SS-PPM system, and as the number of chip slots increases, it approaches Shannon's limit); 2) Asynchronous CDMA BER; 3) Frequency utilization efficiency. In addition, we also show that With SIK is superior to Without SIK on these points.

A narrowband interference in direct sequence spread spectrum communication systems also affects the characteristics of a delay lock loop. In this paper, the delay errors of a baseband delay lock loop (DLL) in the presence of the interference which consists of a narrowband Gaussian noise and several tones are examined, and when a filter is used to reject the interference, the characteristics of the DLL are analyzed using the Fourier method. Furthermore, from the calculation results of the delay error in case where a prediction error filter with two-sided taps is used as the rejection filter, it is shown that the filter is necessary to keep the DLL in the lock-on state.

An adaptive signal processing using Acoustic Charge Transport device, which has great potential for processing very wide band signals in real time, is investigated. It shows that adaptive system for signals of bandwidth from dc up to 500 MHz can be implemented in real time.

An adaptive multichannel IIR lattice predictor for k-step ahead prediction is constructed and the effectiveness of the proposed predictor is evaluated using digital simulations.

In this paper, we attempt the comparison of the image/signal restoration between Projection Filter, which is regarded as one of the linear optimal filters, and the non-linear filter based on MEM. From the simulation, we show the advantage of MEM restoration filter in restoring noisy degraded images.

The spread spectrum system (abbreviated as SS system) is known to be an excellent communication system which resists jamming. Recently, its application to a simplified wireless communication system has been considered to be suited for consumer communication. In Japan, SS wireless LAN system has got the approval on 2.4GHz ISM band already. A compact SS transceiver for the SS wireless LAN is realized, whose data ratio is 230kbps. The SS transceiver is based on a direct sequence for the modulation, and the demodulation is carried out by a specially developed SAW device. In the first part of this paper, the technical conditions of the SS wireless LAN are mentioned. Then the SAW device and the principle of the demodulation are discussed. Finally, the configuration of the SS transceiver and the protocol of the SS wireless LAN are presented.

This letter presents a new algorithm for echo cancellers, which prevents the reduction of echo return loss due to a double-talk. The essence of the algorithm is to introduce signal delays to avoid the reduction. A convergence condition in the algorithm was examined by using the IIR filter expression of the NLMS algorithm, and it was concluded that the IIR filter should be a low pass filter with unity gain. The condition is accomplished by selecting a small step gain.

This review presents research topics and results on digital signal processing in the last twenty years in Japan. The main parts of the review consist of design and analysis of multidimensional digital filters, multiple-valued logic circuits and number systems for signal processing, and general purpose signal processors.

It is known that the resolution conversion based on orthogonal transform has a problem that is difference of luminance between the converted image and the original. In this paper, the scale factor of the system employing various orthogonal transforms is generally formulated by considering the DC gain, and the condition of alias free for DC component is indicated. If the condition is satisfied, then the scale factor is determined by only the basis functions.

An automatic tap assignment method in sub-band adaptive filter is proposed in this letter. The number of taps of the adaptive filter in each band is controlled by the mean-squared error. The numbers of taps increase in the bands which have large errors, while they decrease in the bands having small errors, until residual errors in all the bands become the same. In this way, the number of taps in a band is roughly proportional to the length of the impulse response of the unknown system in this band. The convergence rate and the residual error are improved, in comparison with existing uniform tap assignment. Effectiveness of the proposed method has been confirmed through computer simulation.

In this letter, a new structure of adaptive IIR notch filter is presented. The structure is based on direct form realization and uses the similar adaptation algorithm given in Ref. (4). A quantitative analysis for convergence properties is developed. It is shown that the proposed structure shows superior performance comparing with previously proposed designs. The results of computer simulations are presented to substantiate the analysis.

This letter presents a technique to cancel the parasitic effects of operational amplifier (op amp) in active filter design. To minimize the effects, an op amp model considering the parasitics (i.e. both parasitic poles and zeros) is utilized. It is shown that undesirable factors in the transfer function due to the parasitics can be canceled well by predistorting the passive element values of the circuit. As an example, an active-R highpass filter is evaluated both theoretically and numerically. In this way, the proposed technique can be effectively incorporated into the design of active filters.

We applied a filip-chip-bonding technique to GHz-band SAW filters. The SAW filters mounted by the stud-bump-bonding (SBB) technique which is a kind of flip-chip-bonding technique showed almost the same frequency characteristics as those mounted by the conventional wire-bonding technique at 1.5 GHz. The SAW filter configuration, fabrication process using the SBB, and its electrical characteristics are described and discussed. The SBB technique has a lot of potential to reduce the size and weight even above GHz frequencies.