We show a method to determine a Steiner Minimum Tree (SMT) and a necessary and sufficient condition that an SMT is a full Steiner tree for three given points in -geometry ( = 3m, m is a positive integer). The -geometry allows only orientations with angles i/ (i and ( 2) are integers), and fill up the gap between the rectilinear geometry ( = 2) and the Euclidean geometry ( = ). An SMT in -geometry ( = 3m) has a similar property to that in the Euclidean geometry. The method to determine an SMT in -geometry is an extension of the well-known method in the Euclidean geometry. The Steiner point in -geometry is any point in the intersection area with a parallelogram and a Steiner locus. Then there are infinite candidate locations of the Steiner point. The Steiner point in the Euclidean geometry is that in -geometry ( = 3m).

Caching of frequently accessed data has been shown to be a useful technique for reducing congestion on the narrow bandwidth of wireless channels. However, traditional client/server strategies for supporting transactional cache consistency, which require extensive communications between a client and a server, are not appropriate in a wireless mobile database. This paper proposes two, simple but effective, transactional cache consistency protocols for mobile read-only transactions by utilizing the broadcast-based solutions for the problem of invalidating caches. The novelty of our approach is that the consistency check on accessed data and the commitment protocol are implemented in a truly distributed fashion as an integral part of cache invalidation process. The applicability of proposed techniques is also examined by an analytical study.

For base station antenna array systems with time-division-duplex (TDD) mode, downlink channel responses are equal to uplink channel responses if the duplexing time is small, thus it is often believed that TDD mode simplies downlink beamforming problem as uplink weights can be applied for downlink directly. In this letter, we show that for TDD DS-CDMA systems, even though uplink and downlink channel responses are equal, optimal uplink weights are no longer equal to the optimal downlink ones due to asynchronous property in uplink and synchronous property in downlink, as well as different data rate traffic and QoS requirements. Computer simulations show that for asymmetric traffic, if uplink weights are used for downlink directly, downlink system capacity is less than 50% of that with optimal downlink weights.

This paper introduces an improved multistage parallel interference cancellation (PIC) technique that uses the reverse-link synchronous transmission technique (RLSTT) to improve the estimation of data at the initial stage. Because the subtraction of an interfering signal based on an incorrect bit decision quadruples the interference power for that signal, the relatively high decision bit error rate (BER) may lead to a poor cancellation or even a higher BER at the following stages. The RLSTT is a robust approach which takes into account the fact the tentative decision at the earlier stages is less reliable than the following stages and makes the earlier cancellation more reliable. The analysis demonstrates that a better transmission performance can be achieved by using the RLSTT at the initial stage of PIC.

In this paper, a new adaptive method is suggested using the complementally transformed minimum variance technique for the purpose of suppressing interference in additive white and colored Gaussian noise channels. The method is based on interference suppression by way of the resulting projection weight. The multiple access causes an interference problem in the code-division multiple access systems. An efficient adaptive algorithm should be used to suppress this interference for the improvement of system performance. Analytical and simulation results show that the new adaptive method has fast convergence rate and offers significant performance gain over the conventional detector, the MMSE detector, and the linear decorrelator. Finally, multipath fading induced performance loss, which leads to error probability floor, is established for the proposed method with combining schemes and shown by computer simulation.

In this paper, we investigate the optimal assignment problem of cells in PCS (Personal Communication Service) to switches on a ATM (Asynchronous Transfer Mode) network. Given cells and switches on an ATM network (whose locations are fixed and known), the problem is to group cells into clusters and assign these clusters to switches in an optimum manner. This problem is modeled as a complex integer programming problem. Since finding an optimal solution of this problem is NP-hard, a heuristic solution model consists of three phases (Cell Pre-Partitioning Phase, Cell Exchanging Phase, and Cell Migrating Phase) is proposed. Experimental results show that Cell Exchanging and Cell Migrating Phases can really reduce total cost near 44% on average.

An LCD Driver IC includes more than 300 buffer amplifiers on a single chip. The phase compensation capacitors (on-chip Miller capacitors) for the amplifiers are more than 1000 pF and occupy a large chip area. This paper describes a two-gain-stage amplifier in which an on-chip Miller capacitor is not used for phase compensation in an LCD Driver IC. In the proposed amplifier, phase compensation is achieved only by a newly introduced zero, which is formed by the load capacitance and a phase compensation resistor connected between the output of the amplifier and the capacitive load. Designs of the phase compensation resistor and the amplifier before compensation are discussed, considering a typical load capacitance range. The test chip was fabricated. The newly introduced zero successfully stabilized the amplifier. The chip area for the amplifier was reduced by 30-40%, compared with our previously reported one. The current consumption of the amplifier was only 5 µA. The experimental results of the fabricated test chip support that the proposed amplifier is suitable to an LCD driver IC with a smaller chip area.

A multi-range resolution radar using sideband spectrum energy is investigated. The basic system consists of a sharpening processor and least-error energy shaping filters. First, the sharpening processor makes long flat pulses sharpened. Next, the least-error shaping filters compress the input pulse into the desired pulse width. Then the output pulse width can become narrower than the reciprocal of the input bandwidth, because the least-error shaping filters make the equivalent bandwidth expanded by the enhancement of the sideband spectrum energy and the suppression of the main spectrum. The transmitted signals with simple phase modulation are studied to obtain the multi-range resolution where the pulse is compressed into a pulse with the same bandwidth and another pulse width equal to the reciprocal of the input bandwidth. The peak-to-sidelobe ratio after the pulse compression and the improvement factor of the output signal-to-noise ratio are measured. Further, the experimental results are shown to verify our proposed technique.

A hybrid data transmission technique for multimedia satellite broadcasting is proposed. The main-channel data and sub-channel data are simultaneously transmitted using QPSK modulation and 2ASK modulation, respectively, but the latter modulation timing is offset by half the main-channel QPSK symbol length in time. The BER performance in a Gaussian channel, the transmission bandwidth, and the transmit power peak factor are theoretically analyzed for various impulse responses of the sub-channel transmit filter. It is found that the use of the sub-channel transmit filter having a sine impulse response minimizes the sub-channel BER while keeping the transmission bandwidth and the transmit power peak factor lower than those of CAPSK transmission.

Lossless video coding is required in the fields of archiving and editing digital cinema or digital broadcasting contents. This paper proposes multiresolution lossless video coding using a discrete wavelet transform and adaptive inter/intra-frame prediction in the wavelet domain. The multiresolution structure based on the wavelet transform facilitates interchange among several video source formats such as Super High Definition (SHD) images, HDTV, SDTV, and mobile applications. In order to increase the compression ratio, and to keep the computational cost low, the adaptive inter/intra-frame prediction is performed in the lowest wavelet transform domain. The adaptive inter/intra-frame prediction can adapt to changes in the local inter/intra-frame statistics. Experiments on digital cinema test sequences confirm effectiveness of the proposed algorithm.

In SoC (system-on-a-chip) design, interfacing among IP (Intellectual Property) blocks is one of the most important issues. Since most IP's are provided by different vendors, they generally have different interface schemes and different operating frequencies. In this paper, we propose a new interface synthesis method with two features: 1) generation of the interface between IP's with different operating frequencies, and 2) minimization of the hardware resource required for the interface. We have demonstrated the proposed algorithm through its application to an MP3 decoder design example, where the IIS (Inter-IC Sound)-to-PCI (Peripheral Component Interconnect) protocol converter was successfully implemented using the proposed method.

Subband adaptive array (SBAA) has been realised as a promising method to perform space-time signal processing in mobile communications. Recently, several schemes of SBAA have been introduced. However, theoretical analysis of SBAA performance has been only limited to case of SBAA without using decimation. In this paper, we introduce a novel method to analyse the performance of SBAA using critical sampling with localised feedback scheme. We first provide a detailed analysis of the SBAA performance in the case of single path and multipath fading environment. The simulation results are then presented to verify the proposed method.

This paper presents the performance modeling, analysis, and simulation of SIP-T (Session Initiation Protocol for Telephones) signaling system in carrier class packet telephony network for NGN (Next Generation Networks). Until recently, fone of the greatest challenges in the migration from existing PSTN (Public Switched Telephone Network) toward NGN is to build a carrier class packet telephony network that preserves the ubiquity, quality, and reliability of PSTN services while allowing the greatest flexibility for use of new packet telephony technology. The SIP-T signaling system defined in IETF (Internet Engineering Task Force) draft is a mechanism that uses SIP (Session Initiation Protocol) to facilitate the interconnection of PSTN with carrier class packet telephony network. Based on IETF, the SIP-T signaling system not only promises scalability, flexibility, and interoperability with PSTN but also provides call control function of MGC (Media Gateway Controller) to set up, tear down, and manage VoIP (Voice over IP) calls in carrier class packet telephony network. In this paper, we derive the buffer size, the mean of queueing delay, and the variance of queueing delay of SIP-T signaling system that are the major performance evaluation parameters for improving QoS (Quality of Service) and system performance of MGC in carrier class packet telephony network focused on toll by-pass or tandem by-pass of PSTN. First, we assume a mathematical model of the M/G/1 queue with non-preemptive priority assignment to represent SIP-T signaling system. Second, we derive the formulas of buffer size, queueing delay, and delay variation for the non-preemptive priority queue by queueing theory respectively. Besides, some numerical examples of buffer size, queueing delay, and delay variation are presented as well. Finally, the theoretical estimates are shown to be in excellent consistence with simulation results.

This paper proposes a multi-hop wireless link system for radio access networks (RANs) of new generation mobile communication systems. The performance of the multi-hop wireless link system is evaluated from the viewpoints of total output power, co-frequency interference characteristics, and the system frequency bandwidth based on a comparison with that of the single-hop wireless link system, which is currently used as a RAN. The proposed system is effective in realizing an enormous approach link capacity from both the total output power and the co-frequency interference viewpoints. From the system frequency bandwidth viewpoint, the optimum number of relays in the multi-hop connection is determined to be three hops in a line-of-sight propagation environment in order to minimize the frequency bandwidth for transferring traffic. We conclude that the multi-hop wireless link system is suitable for new generation mobile communication systems.

This paper presents a novel low-energy memory design technique based on variable analysis for on-chip data memory (RAM) in application-specific systems, which called VAbM technique. It targets the exploitation of both data locality and effective data width of variables to reduce energy consumed by data transfer and storage. Variables with higher access frequency and smaller effective data width are assigned into a smaller low-energy memory with fewer bit lines and word lines, placed closer the processor. Under constraints of the number of memory banks, VAbM technique use variable analysis results to perform allocating and assigning on-chip RAM into multiple banks, which have different size with different number of word lines and different number of bit lines tailored to each application requirements. Experimental results with several real embedded applications demonstrate significant energy reduction up to 64.8% over monolithic memory, and 27.7% compared to memory designed by memory banking technique.

A quasi-periodic signal is a periodic signal with period and amplitude variations. Several physiological signals, including the electrocardiogram (ECG), can be treated as quasi-periodic. Vector quantization (VQ) is a valuable and universal tool for signal compression. However, compressing quasi-periodic signals using VQ presents several problems. First, a pre-trained codebook has little adaptation to signal variations, resulting in no quality control of reconstructed signals. Secondly, the periodicity of the signal causes data redundancy in the codebook, where many codevectors are highly correlated. These two problems are solved by the proposed codebook replenishment VQ (CRVQ) scheme based on a bar-shaped (BS) codebook structure. In the CRVQ, codevectors can be updated online according to signal variations, and the quality of reconstructed signals can be specified. With the BS codebook structure, the codebook redundancy is reduced significantly and great codebook storage space is saved; moreover variable-dimension (VD) codevectors can be used to minimize the coding bit rate subject to a distortion constraint. The theoretic rationale and implementation scheme of the VD-CRVQ is given. The ECG data from the MIT/BIH arrhythmic database are tested, and the result is substantially better than that of using other VQ compression methods.

Fully integrated VCO using the "turbo-charger" technique to improve phase noise characteristics is presented. The phase noise degradation of relatively lower oscillation frequency in tuning range was caused by oscillation amplitude lowering due to large total capacitance. On the other hand, the phase noise degradation of relatively higher frequency in tuning range was caused by excess current noise. A new "turbo-charger" circuit increased operation current to obtain sufficient transconductance of amplifier when oscillation frequency was lower to improve phase noise characteristics. The phase noise of VCO employing this technique was extremely low and stable, below -140-dBc/Hz at 3-MHz offset from oscillation frequency, in wide oscillation frequency range, approximately 200-MHz for 1200-MHz oscillation. This VCO was operated with 5.8-7.4-mA current consumption at 3-V supply voltage. The manufacturing process was 0.6-µm SiGe BiCMOS.

The potential network architecture of the emerging carrier class VoIP (Voice over IP) technology for NGN (Next Generation Networks) adopts distributed control architecture to take full advantage of scalability, reliability, flexibility, and interoperability. However, the design of distributed control architecture in the carrier class VoIP network is the state-of-the-art in decentralization and distribution of control. Different configurations of system elements, control scheme of inter system elements communications, signaling protocol, functional partitioning, and scheduling of jobs in call control processing may affect the system performance and QoS (Quality of Service) of MGC (Media Gateway Controller) in carrier class VoIP network. Hence, the modeling of distributed control architecture and its performance analysis are essential issues whenever optimum control architecture has to be determined to meet design requirements. Based on these reasons, this paper proposes several potential network architectures and focuses on the performance study of distributed control architecture in carrier class VoIP network. The SIGTRAN-based distributed control architecture model and the MGCP/MEGACO-based distributed control architecture model are presented. Then, we analyze the SIGTRAN-based distributed control architecture model between MGC and SG (Signaling Gateway) using WRR (Weighted Round Robin) and WF2Q (Worst-case Fair Weighted Fair Queueing) scheduling algorithms respectively. And, we analyze the MGCP/MEGACO-based distributed control architecture model between MGC and MG (Media Gateway) using M/G/1 gating service queueing model. Consequently, the results of performance analysis can be used to evaluate whether the performance of distributed control architecture model can meet the requirement of planning and design for carrier class VoIP network deployment.

This paper proposes an asymmetric bandwidth access network based on super-dense wavelength-division multiplexing (SD-WDM) technologies; the network guarantees 100 Mbps upstream and 1 Gbps downstream bandwidth to each user and supports wide-area transmission. The network minimizes operation and administration costs by consolidating switching equipment, as well as minimizing wavelength monitoring/stabilization functions by employing two technologies; the optical multi-carrier supply module (OCSM) for creating downstream signals and the directly modulated spectrum slicing scheme for creating upstream signals. After describing the configuration and features of the presented network, we demonstrate a bandwidth guaranteed network for each of 64 users with 100 Mbps upstream and 1 Gbps downstream bandwidth. The network provides 10-km access lines with under 7-dB loss from users to the access node and a 120-km metro-loop transmission line with under 25-dB loss from the access node to the center node.

In this paper, a novel continuous complementary sliding control was proposed to improve the tracking performance given the available control bandwidth and the extend of parameter uncertainty. With this control law, the ultimate bound of tracking error was shown to be reduced at least by half, as compared with the conventional continuous sliding control. More strikingly, the proposed control can effectively improve the error transient response during the reaching phase. We presented a composite complementary sliding control scheme for a class of uncertain nonlinear systems including the nonlinear electrohydraulic position servo control system, which will be used as an illustrated example. Simulation results indicated exceptional good tracking performance to step and sine wave reference inputs can be obtained. In addition, the disturbance rejection property of the controller to single-frequency sinusoidal disturbances is also outstanding.