This paper deals with the problem of autoregressive (AR) spectral estimation from a finite set of noisy observations without a priori knowledge of additive noise power. A joint technique is proposed based on the high-order and true-order AR model fitting to the observed noisy process. The first approach utilizes the uncompensated lattice filter algorithm to estimate the parameters of the over-fitted AR model and is one-pass. The latter uses the noise compensated low-order Yule-Walker (LOYW) equations to estimate the true-order AR model parameters and is iterative. The desired AR parameters, equivalently the roots, are extracted from the over-fitted model roots using a root matching technique that utilizes the results obtained from the second approach. This method is highly accurate and is particularly suitable for cases where the system of unknown equations are strongly nonlinear at low SNR and uniqueness of solution from the LOYW equations cannot be guaranteed. In addition, fuzzy logic is adopted for calculating the step size adaptively with the cost function to reduce the computational time of the iterative total search technique. Several numerical examples are presented to evaluate the performance of the proposed scheme in this paper.

Message Sequence Chart (MSC) standardized by International Telecommunication Union is a graphical and textual language for specification of concurrent systems. It has been used formally as well as informally to specify behavior of real-time systems, in particular telecommunication switching systems. Formal verification of a system specification is crucial to ensure that implementation of the system works correctly. In particular, verification methods based on finite states have been widely used in telecommunication systems design. The methods determine global system states and transitions between them (i. e. , build a global state transition graph (GSTG)), and verify the system's desired properties, such as safety and liveness, on the GSTG. In this paper, we focus on construction of GSTGs from MSC specifications. We propose action dependency graph as an intuitive description of semantics of MSC specifications and present an algorithm to translate MSC specifications to action dependency graphs as well as an algorithm to construct a global state transition graph from an action dependency graph.

2.4-GHz-band mid-speed (1- to 2-Mbit/sec) wireless LAN systems are being widely used in offices and factories. Electromagnetic interference can occur between these systems because they use the same frequency range. In this paper, we investigate the characteristics of the interference between wireless LAN systems that use direct-sequence (DS) systems and frequency-hopping (FH) systems. The interference characteristics were measured for three DS systems and one FH system that meet the IEEE 802.11 and RCR standards and that use different modulation methods. Our results indicate that throughput depends on the system and the modulation method. We have also developed a model that can be used to calculate the interference characteristics between DS and FH systems by considering the bandwidth of their transmission signals, the dwell time of the FH system, and the time that the DS system needs to transmit a data frame. We used this model to calculate the bit error rate (BER) characteristics of the systems used in our experiment, and the results indicate that BER characteristics depend on the modulation method. The throughput characteristics of the systems used in our experiment were also calculated, and agreed with the experiment results within +/- 5 dB. The throughput characteristics of wireless LAN systems based on IEEE 802.11 were also calculated when the signal level was higher than the receiver noise level. The results show that FH systems require a D/U ratio about 7 or 8 dB higher than the ratio required in DS systems because the parameters in the standard differ between FH and DS systems.

A novel body potential-controlling technique for floating SOI CMOS circuits is proposed and verified in this study. High-speed operation is realized with a small chip size by using body-floating SOI transistors. The use of this technique allows the threshold voltage of the body-floating transistors to be varied transitionally. Therefore, the standby current of SOI CMOS logic is reduced to less than 1/50th of that required by the non-controlled operation of the body potential, and the logic operates at a high speed during the active period. There is no speed penalty for the recovery operation from the standby mode. This technique supports sub-1 V operation, which will be required by future battery-operated devices with wide-range covering.

B-ISDN is expected to be applied in the near future to video delivery systems for the broadcast of news and television programs. The demand for such services is increasing, and in particular, on-demand services are becoming more desirable. On-demand services allow viewers to request their favorite programs at the time that is convenient, hence catering for the wide range of modern lifestyles. As for on-demand services, there already exist Video on Demand (VoD) systems such as the original VoD or Near VoD. However, such systems have not yet been widely implemented because of the inefficient cost of communication resources, and storage. The authors' research is aimed at producing an efficient VoD system based on a high speed network. We are focused in particular on the forms of data transmission, and in this paper, we propose a new VoD system called Burst VoD. Burst VoD aggressively utilizes the multicasting technique, and involves dividing the program resource data into block files and transmitting them to viewer terminals as burst traffic over a high speed network. Simulation results comparing Burst VoD with conventional VoD show that Burst VoD achieves lower request blocking rates, efficient utilization of networks with multicasting, and almost on-demand response time to requests.

This paper describes an application of a speech translation system to another task/domain in the real-world by using developmental data collected from real-world interactions. The total cost for this task-alteration was calculated to be 9 Person-Month. The newly applied system was also evaluated by using speech data collected from real-world interactions. For real-world speech having a machine-friendly speaking style, the newly applied system could recognize typical sentences with a word accuracy of 90% or better. We also found that, concerning the overall speech translation performance, the system could translate about 80% of the input Japanese speech into acceptable English sentences.

This paper proposes an algorithm that adaptively estimates time-varying noise variance used in Kalman filtering for real-time speech signal enhancement. In the speech signal contaminated by white noise, the spectral components except dominant ones in high frequency band are expected to reflect the noise energy. Our approach is first to find the dominant energy bands over speech spectrum using LPC. We then calculate the average value of the actual spectral components over the high frequency region excluding the dominant energy bands and use it as the noise variance. The resulting noise variance estimate is then applied to Kalman filtering to suppress the background noise. Experimental results indicate that the proposed approach achieves a significant improvement in terms of speech enhancement over those of the conventional Kalman filtering that uses the average noise power over silence interval only. As a refinement of our results, we employ multiple-Kalman filtering with multiple noise models and improve the intelligibility.

Within the next few decades, high-end telecommunication systems on the larger nationwide network will require a switching capacity of over 5 Tbps. Advanced optical transmission technologies, such as wavelength division multiplexing (WDM) will support optical-fiber data transmission at such speeds. However, semiconductors may not be capable of high-throughput data switching because of the limitations by power consumption and operating speed, and pin count. Superconducting single flux quantum (SFQ) technology is a promising approach for overcoming these problems. This paper proposed an optical-electrical-SFQ hybrid switching system and a novel switch architecture. This architecture uses time-shifted internal speedup, shuffle and grouping exchange and a Batcher-Banyan switch. Our proposed switch consists of an interface circuit with small buffers, a Batcher sorter, a time-shift-speedup buffer (TSSB), a Banyan switch, and a slowdown buffer. Simulations showed good scalability up to 100 Tbps, which no router could ever offer such features.

An automated method for cryptanalysis of DFT-based analog speech scramblers is presented through statistical estimation treatments. In the proposed system, the ciphertext only attack is formulated as a combinatorial optimization problem leading to a search for the most likely key estimate. For greater efficiency, we also explore the benefits of genetic algorithm to develop an estimation method which takes into account the doubly stochastic characteristics of the underlying keyspace. Simulation results indicate that the global explorative properties of genetic algorithms make them very effective at estimating the most likely permutation and by using this estimate significant amount of the intelligibility can be recovered from the ciphertext following the attack on DFT-based speech scramblers.

High-speed systems require a wide-frequency-range clock system for data processing. Phase-locked loop (PLL) is used for such a system that requires wide-range variable frequency clock. Frequency calibration method enables the voltage-controlled oscillator (VCO) in a PLL to cover the expected frequency range for high-speed applications that require a wide locking range. Frequency range adjustment is implemented by means of a current digital to analog converter (DAC), which controls the performance curves of a VCO and a bias circuit. This method adjusts the VCO's frequency-voltage performance curves before functional operation so that a PLL can cover requested frequency range with its best condition. Both the limit of control voltage and its target reference voltage are given with same voltage reference. This ensures correct performance after frequency adjustment even under the temperature fluctuation. It eliminates post-production physical adjustment such as fuse trimming which increases the cost and TAT in manufacturing and testing. A high-speed wide-locking range VCO with an automatic frequency performance calibration circuit is implemented within small space in a high-speed hard disk drive channel with 0.25-µm 2.5 V CMOS four-layer metal technology.

A known problem of the four-phase handshaking protocol is that a return-to-zero phase of the signals involved in the handshake is necessary before starting another cycle, in which no useful work is usually done. In this paper we first define an easy-to-write specification style to specify four-phase handshaking asynchronous controllers that can be translated to an STG to obtain a gate-level implementation using existing synthesis methods. Then, we propose an algorithm that takes the specification written using our specification style and finds an optimized timing in which the idle-phase overhead of its gate-level implementation is reduced.

In this paper, we define content-identifier (ContentID) to represent the characteristics of shot. The ContentID carries both positional and temporal color information. Based on the concept of ContentID, we propose a video retrieval method. The method is robust to compression, format conversion, frame dropping and noise such as watermark and so on. Furthermore, based on our retrieval method, we implemented a copyright protection system for digital video using spread-spectrum based watermarking technique.

We analyze a scheme that provides frequency hopping pattern for DS/FH spread spectrum. The proposed scheme, based on the theory of finite projective planes, intends to make the number of transmitting terminals uniform across all channels and distribute the interference to all the participant terminals equally. Thus, when a terminal is in a state of power surge, the probability of having the worst case of interference for terminals sharing the same channel is reduced. In the performance evaluation, we demonstrate that the bit error rate is reduced by an order of magnitude through the use of the proposed hopping pattern for both internal and external interference.

A signal processing approach is discussed which has the potential for imaging strongly scattering objects from a series of scattering experiments. The method is based on a linear spectral estimation technique to replace the filtered backpropagation for limited discrete data and a subsequent nonlinear signal processing step to remove the contribution of multiple scattering my means of homomorphic filtering. Details of this approach are discussed and illustrated by applying the imaging algorithm to both simulated and real data.

We propose a new CALL (Computer-Assisted Language Learning) system for non-native learners of Japanese using speech recognition methods. The aim of the system is to help them develop natural pronunciation by automatically detecting their pronunciation errors and then providing effective feedback instruction. An automatic scoring method based on HMM log-likelihood is used to assess their pronunciation. Native speakers' scores are normalized by the mean and standard deviation for each phoneme and are used as threshold values to detect pronunciation errors. Unlike previous CALL systems, we not only detect pronunciation errors but also generate appropriate feedback to improve them. Especially for the feedback of consonants, we propose a novel method based on the classification of the place and manner of articulation. The effectiveness of our system is demonstrated with preliminary trials by several non-native speakers.

In this paper, we present a low-power and area-efficient design method of embedded high-speed A/D converters for mixed analog-digital system LSI's. As the A/D converter topology, a 1.5 bit/stage interleaved pipeline A/D converter is employed, because the basic topology covers a wide range of specifications on the conversion frequency and the resolution. The design method determines the minimum DC supply current, the minimum device sizes and the minimum number of channels to meet the precision given by the specification. This paper also points out that the interleaved pipeline structure is very effective for low-power design of high-speed A/D converters whose sampling frequency is over 100 MHz.

In this paper, the authors derived the distributions of the probability of detection and of false alarm in function of the decision threshold. An Optimized Threshold Decision (OTD) algorithm was proposed to decide the optimal threshold for reaching the best system performance in a given known channel noise. By applying this OTD algorithm, the multiple access capacity can thus be maximized.

We introduce a high-level synthesis framework to automatically synthesize asynchronous circuits, especially speed-independent circuits, from a concurrent programming language called ALPEH. ALPEH is a high-level concurrent algorithmic specification that can model complex concurrent control flows, logical and arithmetic computations, and communications in easy way. This specification language has been developed to be translated into a Petri net. The major contribution of this paper is the generation of globally optimized control circuits during preserving neat formalism in the specification.

In this paper, we study macro/micro diversity techniques for code acquisition of a direct-sequence spread-spectrum signal in an indoor packet communication system. In the system discussed, the base station has several radio ports each with a cluster of antennas, and the terminal also has multiple antennas. The performance in the uplink of this system is analyzed under Lognormal shadowing and flat Rayleigh fading. The numerical results show great performance improvements by proposed diversity techniques. In addition, it is clarified that the mean acquisition time, which is often used as the measure of performance, is not suitable for packet radio systems as it underestimates the necessary preamble length for initial code acquisition.

This paper analyzes a generalized secret-key authentication system from a viewpoint of the information-spectrum methods. In the generalized secret-key authentication system, for each n 1 a legitimate sender transmits a cryptogram Wn to a legitimate receiver sharing a key En in the presence of an opponent who tries to cheat the legitimate receiver. A generalized version of the Simmons' bounds on the success probabilities of the impersonation attack and a certain kind of substitution attack are obtained.