An adder-based arithmetic VLSI processor using the SD number system is proposed for the applications of real-time computation such as intelligent robot system. Especially in the intelligent robot control system, not only high throughput but also small latency is a very important subject to make quick response for the sensor feedback situation, because the next input sample is obtained only after the robot actually moves. It is essential in the VLSI architecture for the intelligent robot system to make the latency as small as possible. The use of parallelism is an effective approach to reduce the latency. To meet the requirement, an architecture of a new multiple-valued arithmetic VLSI processor is developed. In the processor, addition and subtraction are performed by using the single adderbased processing element (PE). More complex basic arithmetic operations such as multiplication and division are performed by the appropriate data communications between the adder-based PEs with preserving their parallelism. In the proposed architecture, fine-grain parallel processing at the adder-based PE level is realized, and all the PEs can be fully utilized for any parallel arithmetic operations according to adder-based data dependency graph. As a result, the processing speed will be greatly increased in comparison with the conventional parallel processors having the different kinds of the arithmetic PEs such as an adder, a multiplier and a divider. To realize the arithmetic VLSI processor using the adder-based PEs, we introduce the signed-digit (SD) number system for the parallel arithmetic operations because the SD arithmetic has the advantage of modularity as well as parallelism. The multiple-valued bidirectional currentmode technology is also used for the implementation of the compact and high-speed adder-based PE, and the reduction of the number of the interconnections. It is demonstrated that these advantges of the multiple-valued technology are fully used for the implementation of the arithmetic VLSI processor. As a result, the latency of the proposed multiple-valued processor is reduced to 25% that of the binary processor integrated in the same chip size.

This paper studies multidimensional linear periodically shift-variant digital filters (LPSV filters). The notion of a generalized multidimensional transfer function is presented for LPSV filters. The frequency characteristic of the filters is discussed in terms of this transfer function. Since LPSV filters can decompose the spectrum of an input signal into some spectral partitions and rearrange the spectrum, LPSV filters can serve as a frequency scrambler. To show the effect of multidimensional frequency scramble, 2-D LPSV filters are designed based on the 1-D Parks-McClellan algorithm. The resultant LPSV filters divide the input spectrum into some components that are permuted and possibly inverted with keeping the symmetric of the spectrum. Experimental results are presented to illustrate the effectiveness of frequency scramble for real images.

This letter extends the overfitting lattice filter for ARMA parameter estimation with additive noise proposed by Sun and Yahagi. A new way of calculating the lattice parameters is proposed, making their computation truly recursive. This simplifies the method in Ref.(1), and makes it suitable to the parameter estimation of high-order systems.

The evolution of Multiple-Valued Logic (MVL) circuits has been inexorably tied to the rapid technological changes induced by evolving needs and emerging developments in computing methodologies. Unfortunately for MVL, the numbers of designers of technologies and circuits whose lives are dedicated to the improvement of binary techniques, are large and overwhelming. Correspondingly, technological developments in MVL typically await the appearance of a problem or technique in the larger binary world to motivate and/or make possible some new advance. Such opportunities are inevitably quite transient since each such problem is simultaneously attacked by many others of a more conventional bent, and, as well, each technological change begets yet another, quickly. It is in the sensing of this reality that the present paper is written. Correspondingly, its thrust is two-fold: One target is the possibility of encouraging a leap ahead through modest technological projection. The other is the possibility of identifying application areas that already exist in this unbalanced competition, but which are specially suited to multiple-valued solutions. For example, it has been clear for decades that one such area is that of arithmetic. Correspondingly, we in MVL must strive quickly to concentrate our efforts on applications that exploit such demonstrable strengths. Some such applications are includes here; others are visible historically, many probably remain to be found: Search on!

The main interest of this paper is the theoretical analysis of a recursive periodically time varying digital filter. The generalized transfer function of a recursive periodically time varying digital filter was obtained from its difference equation. It was proved that by making use of the generalized transfer function, we can not only derive the input and output relationship of a recursive periodically time varying digital filter easily but also obtain its equivalent structure effectively. An interesting property of a recursive periodically time varying digital filter was also derived by making use of its generalized transfer function. Moreover, it was completed in this paper the investigation of the generalized transfer functions and impulse responses of other periodically time varying models, including an input sampling polyphase model and an output sampling polyphase model. Meanwhile, the multirate Quadrature Mirror Filter bank system was proved by the authors to be a periodically time varying system. Several examples were also provided to illustrate the effectiveness of using the generalized transfer function to obtain the equivalent structure of a recursive periodically time varying digital filter.

A multicarrier modulation is considered as an effective technique in high speed digital transmission under the multipath fading. In this paper, we theoretically analyze the bit error rate (BER) performance of the multicarrier modulation/differential detection scheme, and show the trade-offs between the BERs and the number of carriers or the guard period to clarify the optimum values to minimize the BER in the number of carriers and the guard period.

This paper proposes and investigates an adaptive equalizer with diversity-combining over a multipath fading channel. It consists of two space-diversity antennas and a Ts/2-spaced decision-feedback-equalizer (DFE). Received signals from the two antennas are alternatively switched and fed into the feed forward-filter of DFE. We call this structure a Switched Input Combining Equalizer with diversity-combining (SICE). By using an SICE, the receiver structure for combining diversity equalization can be simplified, because it needs only two receiver sections up to IF BPF. The bit error rate (BER) performance of SICE was evaluated by both computer simulation and experiment over a multipath fading channel. We experimentally confirmed the excellent BER performance, around 1% of BER over a multipath fading channel at 160Hz of maximum doppler fading frequency. Therefore, the proposed SICE is applicable to highly reliable transmission in the 1.5-GHz-band mobile radio.

Theoretical prediction of propagation is required for the future urban mobile communications, in order to make possible precise and universal prediction for arbitrary conditions. The necessity and the fundamental concept of theoretical prediction are introduced, and the theoretical prediction of mean field strength in urban areas is reviewed and discussed. Theoretical method is important particularly in prediction of multipath delay characteristics, in relation to the prediction of error rates in digital mobile radio communications.

This paper presents an efficient multi-precision modular-multiplication algorithm which minimizes the calculation RAM space required when implementing public-key schemes with software on general-purpose computers including smart cards and personal computers. Many modular-multiplication algorithms cannot be efficiently realized on small systems due to their high RAM consumption. The Montgomery algorithm, which can rapidly perform modular multiplication, has received a lot of attention. Unfortunately, the Montgomery algorithm is difficult to implement, especially in smart cards which have extremely limited RAM space. Furthermore, when the modulus of modular multiplication is frequently changed, or when the number of permissible repeated modular multiplications is small, pre- and post-processing operations such as conversion from/to the Montgomery space become wasteful. The proposed algorithm avoids these problems because it requires only half the RAM space and no pre- and post-processing operations. The algorithm is a radical extension to the approximation methods that use the most significant bits and our newly proposed lookahead determination method. This paper gives a proof of the completeness of this method, describes implementation results using a smart card, introduces a theory supported by the results, and considers the optimal technique to enhance the speed of this method.

The design of N-dimensional (N-D) FIR filters requires in general an enormous computational effort. One of the most successful methods for design and implementation is the McClellan transformation. In this paper a numerically simple technique for determining the coefficients of the transformation is suggested. This appears to be the simplest available method for the design of N-D hyperspherically symmetric FIR filters with excellent symmetry.

Digital mobile communication systems have been developed to cope with remarkable growth of the existing analog cellular telephone market. The digital cellular system needs to meet the following requirements: higher frequency efficiency, increased system capacity, new services including ISDN services, and network reliability improvements. New techniques supporting digital mobile radio sevices are presented in this paper, with special focus on TDMA radio link control. Radio channel and signalling structures are designed to achieve spectrum efficiency and flexibility. A random access scheme giving excellent access capability is shown. This paper also presents the system design and configurations of a TDMA digital cellular system based on the Japan standard, which is being developed by NTT Mobile Communications Network Inc. The system is based on the OSI model to enhance system flexibility for future services. Various techniques and devices to achieve economical and compact base stations and portable telephones are developed: multiple-carrier amplifier with extremely low distortion, facsimile/data adaptors with error free and high throughput, coherent detection with adaptive carrier tracking, and so on.

Land mobile communications in Japan have shown remarkable progress in recent years. The total number of all types of radio stations has exceeded 750 million as of March, 1992 and more than 80% of them are used for land mobile communications. The more radio telecommunications becomes popular, the more demand for communicating at any time, at any place and with anyone, intensifies. Various new land mobile systems such as digital cellular telephones have been developed and to be introduced. These new systems are designed to promote effective frequency use in order to meet the exploding demand for it. The digitalization of land mobile communication systems will be the key technology which enable to bring the new possibility in the land mobile communications.

This paper describes voice communication connection controls in digital public land mobile networks (D-PLMNs). Voice communications in the D-PLMNs are carried at about 10 kbit/s over narrow-band TDMA channels with highly efficient cellular voice encoding schemes. Extensive research is being carried on half-rate voice encoding schemes that will effectively double radio resources. We first outline the configuration of voice communication connection between a cellular phone in the D-PLMN and a telephone in a fixed network, and we describe the optimum position for the CODECs that transform cellular voice codes to the conventional voice codes used in the fixed network, and vice versa. Then we propose a CODEC-bypassed communication control scheme that improves the quality of voice communication between cellular phones. And we propose a cellular voice code negotiation scheme in the D-PLMN which supports different cellular voice encoding schemes. We also propose an efficient channel reassignment scheme for effectively assigning TDMA channels to voice calls with two different bitrates (full-rate and half-rate), and we analyze this scheme's traffic capability. Finally, we describe a dual-tone multiple-frequency (DTMF) signal transmission scheme and estimate the number of DTMF signal senders required in the D-PLMN.

This paper describes a new trial of dynamical parameter estimation for the actual room acoustic system, in a practical case when the input excitation is polluted by a background noise in contrast with the usual case when the output observation is polluted. The room acoustic system is first formulated as a discrete time model, by taking into consideration the original standpoint defining the system parameter and the existence of the background noise polluting the input excitation. Then, the recurrence estimation algorithm on a reverberation time of room is dynamically derived from Bayesian viewpoint (based on the statistical information of background noise and instantaneously observed data), which is applicable to the actual situation with the non-Gaussian type sound fluctuation, the non-linear observation, and the input background noise. Finally, the theoretical result is experimentally confirmed by applying it to the actual estimation problem of a reverberation time.

This paper describes a nonstationary spectral analysis method and its application to prognosis and diagnosis of automobiles. An instantaneous frequency spectrum is considered first at a single point of time based on the instantaneous representation of autocorrelation. The spectral distortion is then considered on two-dimensional spectrum, and the filtering is introduced into the instantaneous autocorrelations. By the above procedure, the Instantaneous Covariance method (ICOV), the Instantaneous Maximum Entropy Method (IMEM), and the Wigner method are shown and they are unified. The IMEM is used for the time-dependent spectral estimation of vibration and acoustic sound signals of automobiles. A multi-dimensional (M-D) space is composed based on the variables which are obtained by the IMEM. The M-D space is transformed into a simple two-dimensional (2-D) plane by a projection matrix chosen by the experiments. The proposed method is confirmed useful to analyze nonstationary signals, and it is expected to implement automatic supervising, prognosis and diagnosis for a traffic system.

This paper presents a new method for estimating both the parameters of a nonminimum phase system and its unknown input signal. An approximate inverse system method is used to estimate the unknown input signal, and then, by using a Kalman filter, approximately consistent parameter estimates of the nonminimum phase system can be obtained effectively. This method can be used to estimate the parameters of a nonminimum phase system and a minimum phase one in the case when the input signal is a white noise or an impulse sequence.

This paper proposes a new adaptive algorithm for acoustic echo cancellers with four times the convergence speed for a speech input, at almost the same computational load, of the normalized LMS (NLMS). This algorithm reflects both the statistics of the variation of a room impulse response and the whitening of the received input signal. This algorithm, called the ESP (exponentially weighted step-size projection) algorithm, uses a different step size for each coefficient of an adaptive transversal filter. These step sizes are time-invariant and weighted proportional to the expected variation of a room impulse response. As a result, the algorithm adjusts coefficients with large errors in large steps, and coefficients with small errors in small steps. The algorithm is based on the fact that the expected variation of a room impulse response becomes progressively smaller along the series by the same exponential ratio as the impulse response energy decay. This algorithm also reflects the whitening of the received input signal, i.e., it removes the correlation between consecutive received input vectors. This process is effective for speech, which has a highly non-white spectrum. A geometric interpretation of the proposed algorithm is derived and the convergence condition is proved. A fast profection algorithm is introduced to reduce the computational complexity and modified for a practical multiple DSP structure so that it requires almost the same computational load, 2L multiply-add operations, as the conventional NLMS. The algorithm is implemented in an acoustic echo canceller constructed with multiple DSP chips, and its fast convergence is demonstrated.

In most of the methods proposed so far to design approximately linear phase IIR digital filters (IIR DFs), the design takes place only in the time or in the frequency domain. However, when both magnitude and phase responses are considered, IIR DFs with better frequency responses can be obtained if their characteristics in both domains are taken into account. This paper proposes a design method for approximately linear phase IIR DFs, which is based on parameter estimation techniques in the time domain followed by a nonlinear optimization algorithm in the frequency domain. Several examples are presented, illustrating the proposed method.

A film carrier with 48 peripheral-contacts, which is applicable to ultra-high speed GaAs digital integrated circuits (ICs) with a more than 10 Gbps operation, has been developed. The film carrier has been realized using the following newly developed techniques; (1) wave guides with a well-controlled characteristic impedance of 50 Ω, (2) precise vias of as small as 50 µm diameter conducting both sides of grounded metal planes on a polyimide film, and (3) a feed-through structure for high speed input signals with good impedance matching. The film carrier was molded by resin after ILB (inner lead bonding) to a chip with a copper plate heat spreader. As an application, the film carrier has been applied to a 3 Gbps operational 4-bit GaAs multiplexer IC, and has been proved to have excellent high-frequency characteristics.

In case of the system identification problem, such as an echo canceller, estimated impulse response obtained by the frequency-domain adaptive filter based on the circular convolution has estimation error because the unknown system is based on the linear convolution in the time domain. In this correspondence, we consider a sufficient condition to reduce the estimation error.