We have designed a 32-bit RISC microprocessor with 16-/32-bit fixed-point DSP functionality. This processor, called YD-RISC, combines both general-purpose microprocessor and digital signal processor (DSP) functionality using the reduced instruction set computer (RISC) design principles. It has functional units for arithmetic operation, digital signal processing (DSP) and memory access. They operate in parallel in order to remove stall cycles after DSP or load/store instructions, which usually need one or more issue latency cycles in addition to the first issue cycle. High performance was achieved with these parallel functional units while adopting a sophisticated five-stage pipeline structure. The pipelined DSP unit can execute one 32-bit multiply-accumulate (MAC) or 16-bit complex multiply instruction every one or two cycles through two 17-b 17-b multipliers and an operand examination logic circuit. Power-saving techniques such as power-down mode and disabling execution blocks allow low power consumption. In the design of this processor, we use logic synthesis and automatic place-and-route. This top-down approach shortens design time, while a high clock frequency is achieved by refining the processor architecture.

The present paper discusses several promising application areas for optical data links based on high-performance vertical-cavity surface-emitting laser diodes (VCSELs). Both 850 and 980 nm emission wavelength devices realized in the GaAs-AlGaAs or InGaAs-AlGaAs material systems are considered. We show data transmission results of 10 Gb/s signals at 830 nm wavelength over a new high-bandwidth multimode silica fiber of up to 1.6 km length. The same fiber type is employed to demonstrate the first 40 Gb/s transport over 300 m distance by means of a 4-channel coarse wavelength-division multiplexing approach. A first 1 10 linear VCSEL array capable of 10 Gb/s per channel operation is presented for use in next generation parallel optical modules. To improve the singlemode emission characteristics for output power in the 5 mW range we introduce a new device concept incorporating a long monolithic cavity. For low-cost short-distance data links we investigate graded-index polymer optical fibers and report on up to 9 Gb/s transmission over a length of 100 m. Polymer waveguides are also used in an optical layer of a hybrid electrical-optical printed circuit board. Transmitted 10 Gb/s optical data over a prototype board show the potential of this new technology. Finally we present two-dimensional VCSEL arrays for highly parallel data transport on a CMOS chip level. Both 980 and 850 nm bottom emitting devices with modulation capabilities up to 12.5 Gb/s are discussed.

In an ideal fiber grating having a uniform refractive index modulation, the reflection or the transmission spectrum is symmetric with equal amount of side lobes on both sides of the resonant wavelength of the fiber grating. It is observed that a long-period fiber grating made by a non-uniform UV laser beam through a uniform amplitude mask has an asymmetric transmission spectrum. The asymmetric characteristic is explained with Mach-Zehnder effect in the long-period fiber grating. The non-uniform UV laser beam makes also a non-uniform index modulation along the fiber core. Therefore, a beam coupled to a cladding mode at a section of the grating can be re-coupled to the core mode after passing a certain distance. The re-coupled beam makes Mach-Zehnder-like interference with the un-coupled core mode. However, it is presented that the asymmetric phenomenon can be overcome by scanning the UV laser beam along the fiber over the mask. The beam scanning method is able to suffer the same fluence of the UV laser beam on the fiber. Finally, a linearly chirped long-period fiber grating was made using the non-uniform UV laser beam. Due to the asymmetricity the chirping effect was not clearly observed. It is also presented that the beam scanning method could remove the asymmetric problem and recover the typical spectrum of the linearly chirped fiber grating.

Many methods have been proposed to detect intrusions; for example, the pattern matching method on known intrusion patterns and the statistical approach to detecting deviation from normal activities. We investigated a new method for detecting intrusions based on the number of system calls during a user's network activity on a host machine. This method attempts to separate intrusions from normal activities by using discriminant analysis, a kind of multivariate analysis. We can detect intrusions by analyzing only 11 system calls occurring on a host machine by discriminant analysis with the Mahalanobis' distance, and can also tell whether an unknown sample is an intrusion. Our approach is a lightweight intrusion detection method, given that it requires only 11 system calls for analysis. Moreover, our approach does not require user profiles or a user activity database in order to detect intrusions. This paper explains our new method for the separation of intrusions and normal behavior by discriminant analysis, and describes the classification method by which to identify an unknown behavior.

Voice over IP (VoIP) is a generic name for services, systems and technology for telephony over an IP network. It is also referred to as Internet telephony and IP (Internet Protocol) telephony. Internet telephone client software attracted attention when it first appeared in 1995. Since that, VoIP has rapidly matured into a practical technology, propelled by the popularization and rapid development of the Internet. IP network traffic already exceeds telephone network traffic and is expected to further increase several-fold in the next few years. In future, the telephone network will be integrated into the IP network and telephony will become entirely VoIP. There are three expectations for VoIP. The first is inexpensive telephone service. The second expectation is for integrated telephony and IP network services such as a CTI (Computer Telephony Integration) system in which there is interworking with various Internet applications, such as e-mail and Web call-back for communication services of greater convenience rather than simple replacement of the telephone. The third expectation is for a platform for providing high-quality voice communication, multicast communication, and other such enhanced voice services that have a high degree of freedom. However, many problems remain to be overcome before the VoIP System is realized. The main problems are real-time transmission of voice that allows a smooth conversation, session control for providing a variety of services, and the proposal of new services. In this paper, we give an overview of VoIP and the problems that must be solved in order to realize it and propose some solutions regarding stream control and applications. We also describe session control and other topics that are being discussed in standardization forums.

In this paper, we investigate the performance characteristics of parallel switching architectures constructed by a stack of multistage switching networks. We first find that the performances of the previously proposed parallel switching architectures are much worse than the expected ones from analytic models which are based on the assumption that traffic is uniformly distributed at each stage of a switching network. We show that this phenomenon is closely related to a traffic-distribution capability of a parallel switching system and has a large influence on the performance. From these results, we then propose an architectural solution based on the Generalized Shuffle Network (GSN) and analyze its performance by proposing a new iterative analysis method. The proposed architecture uses self-routing and deflection routing, and inherently has a traffic-distribution capability to improve switch performances such as cell loss and delay in a cost-effective manner. From the comparison of simulation and analysis results, it is shown that the developed models are quite accurate in predicting the performance of a new parallel switching system.

We have built an active omni-directional range range sensor that can obtain an omni-directional depth data through the use of a laser conic plane and a conic mirror. In the navigation of the mobile robot, the proposed sensor system makes a laser conic plane by rotating the laser point source at high speed which creates a two-dimensional depth map, in real time, once an image is captured. Also, since the proposed sensor system measures the actual distance of the target objects, it is able to apply the proposed sensor system to other measurement tasks.

The present paper discusses several promising application areas for optical data links based on high-performance vertical-cavity surface-emitting laser diodes (VCSELs). Both 850 and 980 nm emission wavelength devices realized in the GaAs-AlGaAs or InGaAs-AlGaAs material systems are considered. We show data transmission results of 10 Gb/s signals at 830 nm wavelength over a new high-bandwidth multimode silica fiber of up to 1.6 km length. The same fiber type is employed to demonstrate the first 40 Gb/s transport over 300 m distance by means of a 4-channel coarse wavelength-division multiplexing approach. A first 1 10 linear VCSEL array capable of 10 Gb/s per channel operation is presented for use in next generation parallel optical modules. To improve the singlemode emission characteristics for output power in the 5 mW range we introduce a new device concept incorporating a long monolithic cavity. For low-cost short-distance data links we investigate graded-index polymer optical fibers and report on up to 9 Gb/s transmission over a length of 100 m. Polymer waveguides are also used in an optical layer of a hybrid electrical-optical printed circuit board. Transmitted 10 Gb/s optical data over a prototype board show the potential of this new technology. Finally we present two-dimensional VCSEL arrays for highly parallel data transport on a CMOS chip level. Both 980 and 850 nm bottom emitting devices with modulation capabilities up to 12.5 Gb/s are discussed.

We propose a new intelligent blind source separation algorithm for the mixture of sub-Gaussian and super-Gaussian sources. The algorithm consists of an update equation of the separating matrix and an adjustment equation of nonlinear functions. To verify the validity of the proposed algorithm, we compare the proposed algorithm with extant methods.

In the wireless asynchronous transfer mode (ATM) networks, a custom data link control (DLC) layer protocol with stronger error correction ability is needed for mitigating the affect of radio channel errors. This paper applies punctured turbo code schemes to the protection of the header and various payloads in wireless ATM cell, which are realized by the combination of programmable interleaving and puncturing. Their performance is analyzed for Rayleigh fading channel, which shows more significant reduction in cell loss rate (CLR) than the previous systems. Our proposal also provides good balance designs for CLR and the payload bit error rate (BER), and offers potential for future evolutionary improvement of the wireless ATM coding scheme.

This paper describes the evaluation of a fiber-optic reflective displacement sensor that is compensated for variations in light source intensity, pressure, temperature and opacity of ambient medium. Additionally, the distance information is averaged over several points on the target surface, which reduces signal fluctuations due to inhomogeneities. Furthermore, a practical optical fiber reflective sensor model of measuring oil film thickness for thrust bearing is set up in this paper. Actual measurements were made with HEC 3000 tons' thrust bearing and the results were in good agreement with theoretical calculations.

Orthogonal Frequency Division Multiplexing-Direct Sequence/Code Division Multiple Access (OFDM-DS/CDMA) systems provide frequency diversity gain avoiding inter symbol interference (ISI) in a frequency selective fading channel. However, path diversity gain can not be obtained by using conventional OFDM-DS/CDMA schemes. This paper proposes a new multiple antenna transmission system with combined path diversity and frequency diversity. Signal of each antenna is delayed by several chips to create artificial path diversity as well as frequency diversity of multi-carrier transmission in which can then be combined by using a RAKE receiver. Therefore multiple antenna transmission scheme creates a path diversity effect on uncorrelated signals in multi-carriers from each antenna. The received uncorrelated signals can be processed by Maximum Ratio Combining (MRC) diversity without ISI at a RAKE receiver even when we use FFT modulation. As a result, we can obtain combined path diversity and frequency diversity gain effectively by the RAKE system with the combination of multiple antennas.

A new method for active suppression of reflected sound waves is proposed in this paper. The proposed control system is based on the state feedback control. FEM (Finite Element Method) was applied to represent the sound field under the system equations as proposed by Samejima et al. A new performance index was derived so as to minimize the sound intensity leaving a control region, which was set around the control source on a wall. On the basis of the system equations and the new performance index, an optimal feedback law governing suppression of waves reflected from the wall was derived. In order to evaluate the validity of the proposed method, computer simulations in one- and two-dimensional sound fields were executed. In a one-dimensional sound field, the time response was examined, and the distribution of the instantaneous sound intensity was evaluated in a two-dimensional sound field. The results showed that the reflected sound waves can be suppressed quite well in one-dimensional sound fields by using this method and that the proposed method can potentially suppress the reflected sound waves in the two-dimensional sound fields as well.

The normalized least mean square (NLMS) algorithm has the drawback that the convergence speed of adaptive filter coefficients decreases when the reference signal has high auto-correlation. A technique to improve the convergence speed is to apply the decorrelated reference signal to the calculation of the gradient defined in the NLMS algorithm. So far, only the effect of the improvement is experimentally examined. The convergence property of the adaptive algorithm to which the technique is applied is not analized yet enough. This paper first defines a cost function properly representing the criterion to estimate the coefficients of adaptive filter. The name given in this paper to the adaptive algorithm exploiting the decorrelated reference signal, 'normalized least mean EE' algorithm, exactly expresses the criterion. This adaptive algorithm estimates the coefficients so as to minimize the product of E and E' that are the differences between the responses of the unknown system and the adaptive filter to the original and the decorrelated reference signals, respectively. By using the cost function, this paper second specifies the convergence condition of the normalized least mean EE' algorithm and finally presents computer simulations, which are calculated using real speech signal, to demonstrate the validity of the convergence condition.

The hardware implementation of a proposed high dimensional discrete torus knot code was successfully realized on an ASIC chip. The code has been worked on for more than a decade since then at Aichi Prefectural University and Nagoya Institutes of Technology, both in Nagoya, Japan. The hardware operation showed the ability to correct the errors about five to ten times the burst length, compared to the conventional codes, as expected from the code configuration and theory. The result in random error correction was also excellent, especially at a severely degraded error rate range of one hundredth to one tenth, and also for high grade characteristic exceeding 10-6. The operation was quite stable at the worst bit error rate and realized a high speed up to 50 Mbps, since the coder-decoder configuration consisted merely of an assemblage of parity check code and hardware circuitry with no critical loop path. The hardware architecture has a unique configuration and is suitable for large scale ASIC design. The developed code can be utilized for wider applications such as mobile computing and qualified digital communications, since the code will be expected to work well in both degraded and high grade channel situations.

In quantum communication theory, a realization of the optimum quantum receiver that minimizes the error probability is one of fundamental problems. A quantum receiver is described by detection operators. Therefore, it is very important to derive the optimum detection operators for a realization of the optimum quantum receiver. In general, it is difficult to derive the optimum detection operators, except for some simple cases. In addition, even if we could derive the optimum detection operators, it is not trivial what device corresponds to the operators. In this paper, we show a realization method of a quantum receiver which is described by a projection-valued measure (PVM) and apply the method to 3-ary phase-shift-keyed (3PSK) coherent-state signals.

To achieve a sound field reproduction system, it is important to design multichannel inverse filters which cancel the effects of room transfer functions. The design method in the frequency domain based on the least-norm solution (LNS) requires less memory and less calculation than the design method in the time domain. However, the LNS method cannot guarantee the causality or stability of the filters. In this paper, a design method of a time-domain inverse filter using iterative processing in the frequency domain for multichannel sound field reproduction is proposed, and the result of numerical analysis is described. The proposed method can decrease the squared error of every control point by 3-12 dB. Furthermore, the sound reproduced by this method attains over 13 dB improvement in the segmental signal-noise ratio (SNR) compared with that designed by the LNS method for real environment impulse responses.

One of the disadvantages of compressed data is their vulnerability, that is, even a single corrupted bit in compressed data may destroy the decompressed data completely. Therefore, Variable-to-Fixed length Arithmetic Coding, or VFAC, with error detecting capability is discussed. However, implementable error recovery method for compressed data has never been proposed. This paper proposes Burst Error Recovery Variable-to-Fixed length Arithmetic Coding, or BERVFAC, as well as Error Detecting Variable-to-Fixed length Arithmetic Coding, or EDVFAC. Both VFAC schemes achieve VF coding by inserting the internal states of the decompressor into compressed data. The internal states consist of width and offset of the sub-interval corresponding to the decompressed symbol and are also used for error detection. Convolutional operations are applied to encoding and decoding in order to propagate errors and improve error control capability. The proposed EDVFAC and BERVFAC are evaluated by theoretical analysis and computer simulations. The simulation results show that more than 99.99% of errors can be detected by EDVFAC. For BERVFAC, over 99.95% of l-burst errors can be corrected for l 32 and greater than 99.99% of other errors can be detected. The simulation results also show that the time-overhead necessary to decode the BERVFAC is about 12% when 10% of the received words are erroneous.

In order that speech recognition system may have a high recognition rate in a noisy environment, a wide-band sharp directional microphone system is required at the input for securing a high S/N ratio. The authors have already reported the realization of a wide-band uni-directional microphone system by three-microphone integration method. In this paper, we intend to describe the derivation of a sharp directivity function and the realization of its microphone system. First, setting the shape of the characteristic function to bring a sharp directional pattern and then expanding it into the Fourier series, we derive a new directivity function. Next, on the basis of this directivity function, we will present a sharp directional microphone system with only three non-directional microphones and the subsequent analog signal processing. And also, the directional pattern acquired by the proposed method and the effect of the dispersion in the sensitivity of the constituent microphones on the directivity are discussed in detail.

In mobile broadcasting environments, an information server periodically broadcasts a set of data items to a large mobile client population at every broadcast cycle and mobile clients retrieve the data items they need upon arrival at the broadcast channel. In such environments, the cost of data delivery is independent of the number of clients. Many applications such as auctions and stock quotes perform read-only transactions that require the clients to read consistent and current data for accurate result. Previous concurrency control mechanisms designed for such environments ensure that the clients read consistent data, but they cannot ensure latest-bound currency which allows the clients to read the latest or most up-to-date data. In this paper, we propose an efficient concurrency control mechanism that ensures latest-bound currency as well as update consistency, which is appropriate for the mobile broadcasting environments. To ensure latest-bound currency, the server computes control information using "virtual" broadcast cycles. This control information is also used for checking update consistency. Thus, the proposed mechanism allows all data committed in current broadcast cycle to be broadcast. We have performed simulation experiments to measure transaction aborts in order to evaluate the performance of the proposed mechanism. The result confirms that the proposed mechanism produces no stale reads and also shows that the proposed mechanism generates less transaction aborts than previous mechanisms, which implies that we can get higher data currency without increasing data inconsistency.