We describe three types of software-defined-radio (SDR) receivers based on superconducting technologies. The superconducting analog bandpass filters are essential for all types of the receivers. Another key component is an analog-to-digital converters (ADCs), which are required to have high resolution with a broad band width. The complementary Δ ADC based on the single-flux-quantum circuit is a promising candidate for the SDR receivers because it has a practical nature together with above-mentioned requirements. The experimentally obtained signal-to-noise ratio (SNR) and sensitivity, which are closely related to the resolution, are 34 dB and 20 µA for a quarter of the full-scale input with a band width of about 20 MHz. If we use the optimum decimation filter, the ADC is expected to have the SNR of 82 dB and the sensitivity of 300 nA. These values meet the requirements of the easiest type of the SDR receiver. After new fabrication process has been introduced and the architecture of the ADC has been improved, all types of recievers could be realized based on superconductors.

This paper proposes a new approach for analog portion testing, which can meet requirements for high-speed and high-accuracy testing simultaneously with reasonable cost. The key concept of the new method is cooperation of an LSI tester and some circuitry built in a target SoC device. We will explain the operation principle of the proposed method. The proposed method can be one of the methods to overcome today's expensive production test of analog portion on SoC (System on Chip) devices which heavily depends on LSI tester capability and will become harder in near future.

We fabricated Josephson vortex flow transistors (JVFTs) with a parallel array of Josephson junctions that were prepared using c-axis-oriented 100-nm-thick YBa2Cu3O7-δ (YBCO) thin films grown on 24bicrystal MgO (100) substrates. We observed clear modulations of the critical current and the flow voltage with DC current input to the control line that was inductively coupled to the array of junctions. From the results, we estimated the parameters of the device, e.g., the mutual inductance and the self-inductance, and calculated the operation frequency at which the device potentially exhibited these parameters. Moreover, the current gain and the transresistance were evaluated and found to be 0.5 and 0.15 Ω, respectively. In addition, we observed the high-frequency responses of the JVFT to the input AC current of the sine wave or the square pulse wave. A clear oscillation of the output voltage could be observed with a 1 MHz sine wave and 250 kHz square pulse wave. We also discussed the feasibility of higher frequency operation by using it as an input interface for a single flux quantum (SFQ) logic circuit.

A four-quadrant-input linear transconductor generating a product or a product sum current is proposed. The proposed circuit eliminates the influence of channel length modulation and expands a dynamic input voltage range. As an application of the proposed circuit, the four-quadrant analog multiplier is designed. The four-quadrant analog multiplier consists of the proposed circuit, an input circuit and a class AB current buffer. HSPICE simulation results with 0.35 µm n-well single CMOS process parameter are shown in order to evaluate the proposed circuit.

The 2-terminal shortest path problem is to find a shortest path between two specified vertices in a given graph G. In this paper, we consider this problem in the following situation: G is given before the two vertices are specified so that some preprocessing is allowed to reduce the response time. We present a method for calculating lower bounds of the length of the shortest path for any pair of vertices. Experimental results show that the A* algorithm with our method performs much better than previous methods.

This paper presents a method of predicting future human driving behavior under the condition that its resultant behavior and past observations are given. The proposed method makes use of a dynamic Bayesian network and the junction tree algorithm for probabilistic inference. The method is applied to behavior prediction for a vehicle assumed to stop at an intersection. Such a predictive system would facilitate warning and assistance to prevent dangerous activities, such as red-light violations, by allowing detection of a deviation from normal behavior.

In recent works [1],[4], it has been shown that the damping of a linear time invariant system relates to the so-called characteristic ratios (αk, k=1,…, n-1) which are defined by coefficients of the denominator of the transfer function. However, the exact relations are not yet fully understood. For the purpose of exploring the issue, this paper presents the analysis of time response sensitivity to the characteristic ratio change. We begin with the sensitivity of output to the perturbations of coefficients of the system denominator and then the first order approximation of the αk perturbation effect is computed by an explicit transfer function. The results are extended to all-pole systems in order to investigate which characteristic ratios act dominantly on step response. The same analysis is also performed to a special class of systems whose denominator is composed of so called K-polynomial. Finally, some illustrative examples are given.

A novel error detection method based on coded block pattern (CBP) information verification is proposed for error concealment of inter-coded video frames transmitted in wireless channel. This method firstly modifies the original video stream structure by the aggregation of certain important information, and then inserts some error verification bits into the video stream for each encoded macro block (MB), these bits can be used as reference information to determine whether each encoded MB is corrupted. Experimental results on additive Gauss white noise simulation wireless channel and H.263+ baseline codec show that the proposed method can outperform other reference approaches on error detection performance. In addition, it can preserve the original video quality with a small coding overhead increase.

A simple self-biased receiver system with a dual branch architecture consisting of a low-power consumption receiver and a rectenna is introduced. The system is efficiently integrated with a dual-fed circular sector antenna with harmonic rejection characteristics without a BPF. The receiver portion is designed by utilizing a low-noise amplifier (LNA) with low power consumption and a self-heterodyne mixer, while the rectenna achieves high conversion efficiency up to 80%, thanks to the harmonic rejection of the circular sector antenna. The rectified DC power from the rectenna is applied for a bias of the receiver without any external bias. Simultaneously, an ASK digital signal demodulation without an extra power supply are implemented successfully.

Wireless sensor networks present a promising opportunity for realizing many practical applications. Tracking is one of the important applications of these networks. Many approaches have been proposed in the literature to deal with the tracking problem. Recently, a particular type of tracking problem called on-site tracking has been introduced [15],[16]. On-site tracking has been characterized as the tracking in which the sink is eventually required to be present in the vicinity of the target, possibly to perform further actions. In this paper, first we propose two efficient on-site tracking algorithms. Then, we derive theoretical upper bounds for the tracking time and the number of messages generated by the sensor nodes during the tracking for our algorithms. Finally, we present a simulation study that we conducted to evaluate the performance of our algorithms. The results show that our algorithms are efficient as compared to the other existing methods that can solve the on-site tracking problem. In particular, the path adaptive nature of the sink in our algorithms allows the network to conserve the energy and the sink to reduce the tracking time.

For alleviating the high peak-to-average power ratio (PAPR) problem of orthogonal frequency division multiplexing (OFDM), the OFDM combined with time division multiplexing (TDM) using frequency-domain equalization (FDE) was proposed. In this paper, the theoretical bit error rate (BER) analysis of the OFDM/TDM in a frequency-selective fading channel is presented. The conditional BER expression is derived, based on a Gaussian approximation of the inter-symbol interference (ISI) arising from channel frequency-selectivity, for the given set of channel gains. Various FDE techniques as in multi-carrier code division multiple access (MC-CDMA), i.e., zero forcing (ZF), maximum ratio combining (MRC) and minimum mean square error (MMSE) criteria are considered. The average BER performance is evaluated by Monte-Carlo numerical computation method using the derived conditional BER expression.

In this letter, we study the capacity of fading channels with perfect channel side information (CSI) at the receiver and quantized CSI at the transmitter. We present a general algorithm for the joint design of optimal quantization and power control for maximizing the forward link capacity over flat fading channels. Numerical results for Rayleigh fading are given.

Admission control is becoming an essential technique for IP networks to provide full-fledged multimedia streaming services. Although signaling-based schemes are utilized to achieve this, these are difficult to deploy and can hardly achieve strict admission control taking the properties of packet arrival into consideration. In this paper, we propose a novel admission control strategy called the Tentative Accommodating and Congestion Confirming Strategy (TACCS). The main idea is to accommodate incoming flows tentatively and confirm congestion after a certain period. Since tentative accommodating enables us to generate the same situation as where incoming flows have been accommodated, TACCS makes it possible to control admission considering the properties of packet arrival after they have been accommodated, without collecting resource information in advance. From the results of mathematical analysis, we confirmed that TACCS enabled a domain to control admission without a centralized management agent and we provided guidelines for configuring parameters of TACCS.

In this paper, routing properties of cube-based optoelectronic OTIS networks are explored. We show emulations of various cubical network topologies on their OTIS augmented variants, including the n-D grid networks, shuffle-exchange, and de Brujin networks. An analytical performance model for OTIS-cube networks is proposed. The model is validated by means of comparison with rigorously obtained simulation results. Using this model, the performance characteristics of the OTIS-hypercube network are evaluated in view of a number of different constraints. Moreover, we compare the performance characteristics of the OTIS-hypercube with that of equivalent fully-electronic networks under various implementation constraints.

The performance of parallel computing applications is highly dependent on the efficiency of the underlying communication operations. While often characterized as dynamic, these communication operations frequently exhibit spatial and temporal locality as well as regularity in structure. These characteristics can be exploited to improve communication performance if the correct prediction model is selected to a suitable communication topology. In this paper we describe an entropy based methodology for quantifying and evaluating the success of different prediction models on actual workloads drawn from representative parallel benchmarks. We evaluate two different prediction criteria and combinations thereof: (1) Messages are partitioned by source node. (2) Use of a first order context model. We also describe the threshold for predication designed to largely avoid incorrect predication overheads. Our results show for simple predication models, even on highly dynamic benchmark applications, predictability can be improved by several orders of magnitude. In fact, using simple prediction techniques, over 75% of the communication volume is accurately predictable.

With the increasing demand for IP telephony services using Voice over IP (VoIP) technology, techniques for monitoring speech quality in actual networks are required to manage the quality of VoIP services constantly. Since the speech quality of VoIP is affected by IP network performance factors, non-intrusive methods of monitoring the quality of service (QoS) by passively measuring network performance are being watched with keen interest. VQmon technology is one of the non-intrusive quality monitoring methods. Although the monitoring functions of the VQmon for post-arrived packet behavior events at VoIP-gateways are effective, the estimating algorithm does not take differences in the implementations of VoIP-gateway products into account. We therefore propose a non-intrusive method of monitoring QoS that works in conjunction with ITU-T Recommendation P.862 "PESQ" that takes the characteristics of VoIP-gateway products into consideration. We compared the performance of non-intrusive quality monitoring technology such as VQmon and the proposed method in terms of estimating the accuracy of speech quality and mouth-to-ear delay. The experimental results revealed that the proposed method outperforms the conventional one, achieving sufficient accuracy for quality monitoring of VoIP services.

Vigna and Ghezzi showed that the language of grid graphs could not be constructed by their context-free graph grammars [1]. In this paper, we construct a context-sensitive graph grammar for the two-dimensional grid graphs.

In wavelength-routed optical networks, wavelength converters are considered as one of the most critical network resources because they can significantly reduce the blocking probability, but still remain quite expensive. Unfortunately, previous wavelength assignment algorithms have seldom considered their presence. Therefore, in this paper, we propose a novel dynamic algorithm that can minimize the number of wavelength translations. Our algorithm establishes lightpaths by connecting a minimum number of wavelength-continuous segments. We mathematically prove the correctness of our algorithm. Then, we carry out extensive performance evaluations over three typical topologies with full-range or limited-range converters to compare our proposed algorithm with first-fit and most-used algorithms. The simulations show that, to obtain similar blocking performance, our algorithm requires much fewer converters, or the same number of converters but with smaller conversion ranges. From another perspective, with the same conversion capacity, our algorithm can significantly improve the blocking performance. Our algorithm is also scalable due to its polynomial time complexity and insignificant local signaling overhead.

Multi-projector technology has been under consideration in recent years. This technology allows the generation of wide field of view and high-resolution images in a cost-effective manner. It is expected to be applied extensively to training simulators where vivid immersive sensations and precision are required. However, in many systems the viewing frustums cannot be automatically assigned for distributed rendering, and the required manual setup is complicated and difficult. This is because the camera should be coincide exactly with a desired eye point to avoid perspective distortions. For the actual applications, the camera is seldom able to be set up at the desired eye point because of physical constraints, e.g., a narrow cockpit with many instruments. To resolve this issue, we have developed a "virtual camera method" that yields high-precision calibration regardless of the camera position. This method takes advantage of the quadratic nature of the display surface. We developed a practical real-time multi-projector display system for applications such as training simulators, that require high-accuracy in geometry and rapid response time.

This paper describes the author's perspective on multimedia quality prediction methodologies for multimedia communications in advanced mobile and internet protocol (IP)-based telephony, and reports related experiments and trials. First, the paper describes the need for perceptual QoS (Quality of Service) assessment in which various quality factors in multimedia communications for advanced mobile and IP-based telephony are analyzed. Then an objective quality prediction scheme is proposed from the viewpoints of quality measurement tools for each quality factor and an opinion model for compound quality factors in mobile and IP-based communications networks. Finally, the author's current trials of measurement tools and opinion models are described.