This paper describes a new optical label switching technique; wavelength and pilot tone frequency are combined to form labels that are used to control transport network routing. This technique is very attractive for achieving simple nodes that offer extremely rapid forwarding. Experimental results on the discrimination of optical labels and all-optical label conversion are also presented.

The effectiveness and possible applications of all-optical signal processing using highly-nonlinear dispersion-shifted fibers (HNL-DSFs) are described. Transparent and simultaneous processings of multi-channels WDM signal are key features of optical fiber processors. Simultaneous wavelength conversion of 3210 Gb/s WDM signal by four-wave mixing, all-optical 3R regeneration of 220 Gb/s WDM signal using nonlinear loop mirrors, and simultaneous recovery of 2020 GHz WDM optical clocks by supercontinuum were successfully demonstrated using HNL-DSFs, and possible applications of ultra-fast and multi-channel processing in future photonic networks are discussed.

This paper addresses the problem of detecting digital line components in a given binary image consisting of n black dots arranged over N N integer grids. The most popular method in computer vision for this purpose is the one called Hough Transform which transforms each black point to a sinusoidal curve to detect digital line components by voting on the dual plane. We start with a definition of a line component to be detected and present several different algorithms based on the definition. The one extreme is the conventional algorithm based on voting on the subdivided dual plane while the other is the one based on topological walk on an arrangement of sinusoidal curves defined by the Hough transform. Some intermediate algorithm based on half-planar range counting is also presented. Finally, we discuss how to incorporate several practical conditions associated with minimum density and restricted maximality.

We propose a dynamic assignment method of radio channels using their qualities and data packet sizes. In this method, a high quality channel is assigned for long packets and a low quality channel is assigned for short packets. Assuming a packet transfer method based on PHS (Personal Handy System) procedure, the transfer speed and delay characteristics are investigated by computer simulation and compared with conventional method (i.e. random assignment). Both the transfer speed characteristics and delay performance are improved under light traffic load.

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 this paper, we propose a Call Admission Control (CAC) scheme for the Direct Sequence-Code Division Multiple Access (DS-CDMA) systems supporting voice and data services and analyze the Erlang capacity under the proposed CAC scheme. Service groups are classified by Quality of Service (QoS) requirements such as the required Bit Error Rate (BER) and information bit rate, and Grade of Service (GoS) requirement such as required call blocking probability. Different traffics require different system resources based on their QoS requirements. In the proposed CAC scheme, some system resources are reserved exclusively for handoff calls to have high priority over new calls. Additionally, the queueing is allowed for both new and handoff data traffics that are not sensitive to delay. As a performance measure of the suggested CAC scheme, Erlang capacity is introduced. For the performance analysis, a four-dimensional Markov chain model is developed. As a numerical example, Erlang capacity of an IS-95B type system is depicted, and optimum values of system parameters such as the number of reservation channels and queue lengths are found. Finally, it is observed that Erlang capacity is improved more than 2 times by properly selecting the system parameters with the proposed CAC scheme. Also, the effect of handoff parameters on the Erlang capacity is observed.

An appropriate call admission control in the next generation wireless networks is expected to make efficient use of scarce wireless resource and improve quality-of-service, while supporting multimedia services. On one hand, blocking handoff calls is normally more annoying than blocking new calls. On the other hand, blocking new calls reduces resource utilization. More importantly, handoff call arrival rate depends strongly on call holding time. A novel Call-Holding-Time-Based Random Early Blocking (CHTREB) scheme is proposed to achieve the aforesaid goals in a two-tier cellular voice/data network. With CHTREB, new calls are accepted according to some acceptance probability taking into account the call hloding time difference between voice and data calls. An iterative algorithm is developed to calculate performance measures of interest, i.e., new call blocking probability and forced termination probability. First, simulation results are shown to verify analytical results. Then, numerical results are presented to show the robustness of CHTREB. It is found that CHTREB outperforms TR and CHTREB-FAP under both stationary and nonstationary traffic scenarios. Last but not least, the studied 2-tier system is compared with 1-tier counterpart. It is shown that 2-tier system performs better in terms of average number of handoffs per data call.

High-speed digital data transmission services with mobile equipment are becoming available. Though the visual signal is one of the expected media to be used with such transmission capabilities, the bandwidth of visual signal is much broader than the provided transmission bandwidth in general. Therefore efficient video encoding algorithms have to be introduced. The ITU-T Recommendation H.263 and ISO/IEC MPEG-4 are very powerful encoding algorithms for a wide range of video sequences. But a large amount of bits are generated in highly active scenes to encode them using such conventional methods. This results in frame skipping and degradation of decoded picture quality. In order to keep these degradations as low as possible, we proposed a Dynamic Resolution Conversion (DRC) method of the prediction error. In the method, a reduced resolution encoding is carried out when the input scene is highly active. Simulation results show that the proposed scheme can improve both coding frame rate and picture quality in a highly active scene. We also present in this paper that some analysis for the performance of the DRC method under the error prone environment that is inevitable with mobile communications.

This paper proposes a unified coding algorithm for the lossless and near-lossless compression of still color images. The proposed unified color image coding scheme can control the Peak Signal-to-Noise Ratio (PSNR) of the reconstructed image while the level of distortion on the RGB plane is suppressed to within a preset magnitude. In order to control the PSNR, the distortion level is adaptively changed at each pixel. An adaptive quantizer to control the distortion is designed on the basis of psychovisual criteria. Finally, experiments on Super High Definition (SHD) images show the effectiveness of the proposed algorithm.

We propose a mechanism to facilitate the development of component-based mobile applications with adaptive behaviors. The design principles and communication patterns of legacy software systems will greatly change in a forthcoming environment, where a variety of computing devices become embedded in home and office environments, users move around with/without portable computing devices, and all the devices are interconnected through wired/wireless networks. In the proposed mechanism, Improvised Assembly Mechanism (IAM), we realize functionality to compose an application in an ad hoc manner and to achieve the adaptation of applications by adding, replacing, supplementing, and relocating components at system runtime according to various environmental changes such as the locational changes of computing devices and users. The mechanism is implemented as a built-in functionality of the Soul component, which is one of the fundamental elements in the Possession model.

To improve the reliability and efficiency of multicast transmissions in wireless systems, a novel retransmission procedure is desired. In this paper, the representative acknowledgment scheme for reliable wireless multicast communications is proposed that offers quite a low packet loss rate. The proposed protocol carries out retransmissions in the datalink layer within the wireless region, and retransmissions do not affect the traffic in the wired region. The representative acknowledgment scheme employs both positive acknowledgment (ACK) and negative acknowledgment (NACK) to achieve reliable multicast transmissions and reduces the number of responses to be returned by forming groups of stations in the cell. One of the members in a group, called a representative station, returns a response for a received data frame while the others return a NACK if necessary. With this scheme, reliable multicast transmissions are enabled in wireless communications without spending much time as in conventional reliable multicast protocols. The performance of the proposed protocol is evaluated by numerical analyses and by computer simulation. The results show that 30% or more decrease in transmission time is achieved in a typical wireless environment.

We propose a p-persistent protocol for slow-frequency-hopped slotted random access networks, assuming that all the users know the number of backlog users in a slot. The proposed protocol delays new packet transmission until the number of users with a packet to be retransmitted decreases to the threshold or less. Performance of the proposed protocol is evaluated with a two-dimensional Markov chain for systems with finite population in terms of throughput efficiency and the average transmission delay. Numerical results show that the proposed protocol can achieve better performance by suppressing new packet transmission, compared to the conventional frequency-hopped slotted ALOHA. The optimum value of the threshold is also numerically derived.

Call admission control (CAC) is becoming vital for multimedia services in the ability of wireless/mobile networks to guarantee Quality of Service (QoS) partially due to the network's limited capacity. In this paper, we propose an optimal call admission control scheme with bandwidth reallocation algorithm (multi-class-CAC-BRA) for multi-classes of adaptive multimedia services in wireless/mobile networks. The multi-class-CAC-BRA approach optimizes revenue for service providers and satisfies QoS requirements for service users. The proposed approach adopts semi-Markov Decision Process to model both call admission control and bandwidth reallocation algorithm. In other words, whenever decisions are made, decisions are made for both call admission control and bandwidth reallocation. Since the non-adaptive multimedia traffic is a special case of the adaptive multimedia traffic, the non-adaptive optimal CAC scheme is a special case of our optimal multi-class-CAC-BRA scheme. Furthermore, the Interior-point Method in linear programming is used to solve the optimal decision problem. Simulation results reveal that the proposed multi-class-CAC-BRA scheme adapts itself well to adaptive multi-class multimedia traffic, achieves optimal revenue, and satisfies QoS requirements that are the upper bounds of handoff dropping probabilities. Our approach solves the optimal adaptive multimedia CAC problem. We believe that this work has both theoretical and practical significance.

A combination of a software and a systolic hardware implementation for the Quasi Arithmetic compression algorithm is presented. The hardware is implemented as a pipeline hardware implementation. The implementation doesn't change the the algorithm. It just split it into two parts. The combination of parallel software and pipeline hardware can give very fast compression without decline of the compression efficiency.

In this letter, we present a method to generate sets of sequences suitable for multicode transmission in quasi-synchronous (QS) CDMA systems. We focus on Gold code but extension to orthogonal Gold code is straightforward. We show that by appropriate classification of sequences, it is possible to have sets whose cross correlation is small in QS situations.

Error-free transmission of image fiber-optic two-dimensional (2-D) parallel interconnection using vertical-cavity surface-emitting laser (VCSEL)/photodiode (PD) arrays is demonstrated. Simple constructions of transmitter/receiver modules are proposed. Optical alignment is achieved without power-monitoring. Crosstalk from an adjacent channel was -34 dB. Misalignment tolerance for a BER of less than 10-9 was 85 µm. The results clearly indicate that the interconnection system built around an image fiber and 2-D VCSEL/PD arrays has promise for use in the highly parallel high-density optical interconnects of the future.

Most natural images are well modeled as smoothed areas segmented by edges. The smooth areas can be well represented by a wavelet transform with high regularity and with fewer coefficients which requires highpass filters with some vanishing moments. However for the regions around edges, short highpass filters are preferable. In one recently proposed approach, this problem was solved by switching filter banks using longer filters for smoothed areas of the images and shorter filters for areas with edges. This approach was applied to lossy image coding resulting in a reduction of ringing artifacts. As edges were predicted using neighboring pixels, the nonlinear transforms made the decorrelation more flexible. In this paper we propose a time-varying filterbank and apply it to lossless image coding. In this scheme, we estimate the standard deviation of the neighboring pixels of the current pixel by solving the maximum likelihood problem. The filterbank is switched between three filter banks, depending on the estimated standard deviation.

By combining the technology of all-optical saturable absorbers and the diffractive optics, a scheme of all-optical time division demultiplexing module is investigated. Following authors' proposal, design, test fabrication of the optical platform in the previous paper, this paper focuses on the characterization of switching performance. Using a multiple quantum well saturable absorber of InGaAs/InAlAs composition, and gain switched semiconductor laser pulses of 25 ps pulse width, the switching function was demonstrated experimentally at wavelength of 1.55 µm. The switching on-off ratio was compared among 4 lens configuration, 2 lens configuration (2L) and free space, collinear geometry. No degradation was observed in the case of 2 lens configuration in comparison to collinear illumination. Thus the feasibility of all-optical switch module with power efficiency and high speed is predicted, under the assumption of the progress in sub-micron lithography.

We propose a method to represent non-binary error patterns for Reed-Solomon codes using a trellis. All error patterns are sorted according to their Euclidean distances from the received vector. The decoder searches through the trellis until it finds a codeword. This results in a soft-decision maximum likelihood algorithm with lower complexity compared to other known MLD methods. The proposed MLD algorithm is subsequently modified to further simplify complexity, reflecting in a slight reduction in the error performance.

12-channel DC to 622-Mbit/s/ch optical transmitter and receiver have been developed for high-capacity and rather long (about 100 m) bit-parallel raw data transmission in intra- and inter-cabinet interconnection of large-scale switching, routing and computing system. Bit-parallel raw data transmission is done by using a bit-by-bit operational automatic decision threshold control receiver circuit with a DC-coupled configuration, the pin-PDs with their anodes and cathodes separated in a channel-by-channel manner, and a receiver preamplifier with a low-pass filter. The transmitter consists of a 12-channel LD sub-assembly unit and a LD driver LSI. The LD sub-assembly unit consists of a 12-channel array of high temperature characteristic 1.3-µm planar buried hetero-structure (PBH) LDs and 62.5/125 graded-index multi-mode fibers (GI62.5 MMFs). The 1.3-µm PBH LDs and the GI62.5 MMFs are optically coupled by passively visual alignment technology on the Si V-groove. The receiver consists of a 12-channel pin-PD sub-assembly unit and a receiver LSI. The pin-PD sub-assembly unit consist of a 12-channel array of pin-PDs and GI62.5 MMFs. They are optically coupled by using a flip-chip bonding on the Si V-groove. The transmitter and receiver each have eleven data channels and one clock channel. The size is as small as 3.6 cc for each modules, and the power consumptions are 1.7 W (transmitter) and 1.35 W (receiver). They transmitted a bit-parallel raw data through a 100-meter ribbon of GI62.5 MMFs in an ambient temperature range of 0-70C. They provide a synchronous PECL interface parallel link for with a 3.3-V single power supply.