This paper presents a planar monopole antenna with slits and a stub for multi-band operation in vehicles. The proposed antenna is at least 55% smaller than the regular rectangular monopole antenna and covers eight wireless application bands. Slits cut into the rectangular monopole alter the surface current paths so that the band coverage is expanded. A long bent-stub is also added to cover the lowest service band.

In this paper, we propose an unbounded model checking method for feature interaction verification for telecommunication systems. Unbounded model checking is a SAT-based verification method and has attracted recent attention as a powerful approach. The interpolation-based approach is one of the most promising unbounded model checking methods and has been proven to be effective for hardware verification. However, the application of unbounded model checking to asynchronous systems, such as telecommunication systems, has rarely been practiced. This is because, with the conventional encoding, the behavior of an asynchronous system can only be represented as a large propositional formula, thus resulting in large computational cost. To overcome this problem we propose to use a new scheme for encoding the behavior of the system and adapt the unbounded model checking algorithm to this encoding. By exploiting the concurrency of an asynchronous system, this encoding scheme allows a very concise formula to represent system's behavior. To demonstrate the effectiveness of our approach, we conduct experiments where 21 pairs of telecommunication services are verified using several methods including ours. The results show that our approach exhibits significant speed-up over unbounded model checking using the traditional encoding.

Internet routers need to classify incoming packets quickly into flows in order to support features such as Internet security, virtual private networks and Quality of Service (QoS). Packet classification uses information contained in the packet header, and a predefined rule table in the routers. Packet classification of multiple fields is generally a difficult problem. Hence, researchers have proposed various algorithms. This study proposes a multi-dimensional encoding method in which parameters such as the source IP address, destination IP address, source port, destination port and protocol type are placed in a multi-dimensional space. Similar to the previously best known algorithm, i.e., bitmap intersection, multi-dimensional encoding is based on the multi-dimensional range lookup approach, in which rules are divided into several multi-dimensional collision-free rule sets. These sets are then used to form the new coding vector to replace the bit vector of the bitmap intersection algorithm. The average memory storage of this encoding is θ (LNlog N) for each dimension, where L denotes the number of collision-free rule sets, and N represents the number of rules. The multi-dimensional encoding practically requires much less memory than bitmap intersection algorithm. Additionally, the computation needed for this encoding is as simple as bitmap intersection algorithm. The low memory requirement of the proposed scheme means that it not only decreases the cost of packet classification engine, but also increases the classification performance, since memory represents the performance bottleneck in the packet classification engine implementation using a network processor.

Several scheduling algorithms have been proposed for the downlink of a Code Division Multiple Access (CDMA) system with High Data Rate (HDR). Modified Largest Weighted Delay First (M-LWDF) scheduling algorithm selects a user according to the user current channel condition, user head-of-line packet delay and user Quality of Service (QoS) requirement. Proportional Fair (PF) scheduling algorithm has also been proposed for CDMA/HDR system and it selects a user according to the ratio of the user current channel rate and the user average channel rate, which provides good performance in terms of fairness. However, when variable bit rate (VBR) traffic is considered under different channel conditions for each user, both schedulers' performance decrease. M-LWDF scheduler can not guarantee the QoS requirement to be achieved and PF scheduler can not achieve a good fairness among the users. In this work, we propose a new scheduling algorithm to enhance M-LWDF and PF schedulers performance. Proposed scheduler selects a user according to the user input traffic characteristic, user current channel condition and user QoS requirement, which consists of a delay value with a maximum violation probability. We consider the well-known effective bandwidth expression, which takes into account the user QoS requirement and the user input traffic characteristics, to select a user to be scheduled. Properties of the proposed scheduling algorithm are investigated through simulations with constant bit rate (CBR) and VBR flows and performance comparisons with M-LWDF and PF schedulers. The results show a better performance of the proposed scheduler compared with M-LWDF and PF schedulers.

We describe the applicability of photonic crystal fiber (PCF) with an enlarged effective area Aeff to a distributed Raman amplification (DRA) transmission. We investigate the DRA transmission performance numerically over a large Aeff PCF taking account of the signal-to-noise ratio (SNR) improvement RSNR in the S, C, and L bands. We show that an RSNR of 3 dB can be expected by utilizing DRA with a maximum pump power of 500 mW when the Aeff of the PCF is 230 µm2.

This paper develops a complete architecture for constant false alarm rate (CFAR) detection based on a goodness-of-fit (GOF) test. This architecture begins with a logarithmic amplifier, which transforms the background distribution, whether Weibull or lognormal into a location-scale (LS) one, some relevant properties of which are exploited to ensure CFAR. A GOF test is adopted at last to decide whether the samples under test belong to the background or are abnormal given the background and so should be declared to be a target of interest. The performance of this new CFAR scheme is investigated both in homogeneous and multiple interfering targets environment.

In wireless cellular networks, streaming of continuous media (with strict QoS requirements) over wireless links is challenging due to their inherent unreliability characterized by location-dependent, bursty errors. To address this challenge, we present a two-step scheduling algorithm for a base station to provide streaming of continuous media to wireless clients over the error-prone wireless links. The proposed algorithm is capable of minimizing the packet loss rate of individual clients in the presence of error bursts, by transmitting packets in the round-robin manner and also adopting a mechanism for channel prediction and swapping.

We have introduced a new ultrasonic-based localization method that requires only one ultrasonic receiver to locate transmitters. In our previous reports [1],[2], we conducted several fundamental experiments, and proved the feasibility and accuracy of our system. However the performance in a more realistic environment has not yet been evaluated. In this paper, we have extended our localization system into a robot tracking system, and conducted experiments where the system tracked a moving robot. Localization was executed both by our proposed method and by the conventional TOA method. The experiment was repeated with different density of receivers. Thus we were able to compare the accuracy and the scalability between our proposed method and the conventional method. As a result 90-percentile of the position error was from 6.2 cm to 14.6 cm for the proposed method, from 4.0 cm to 6.1 cm for the conventional method. However our proposed method succeeded in calculating the position of the transmitter in 95% out of total attempts of localization with sparse receivers (4 receivers in about 5 m 5 m area), whereas the success rate was only 31% for the conventional method. From the result we concluded that although the proposed method is less accurate it can cover a wider area with sparse receivers than the conventional method. In addition to the dynamic tracking experiments, we also conducted some localization experiments where the robot stood still. This was because we wanted to investigate the reason why the localization accuracy degraded in the dynamic tracking. According to the result, the degradation of accuracy might be due to the systematic error in localization which is dependent on the geometric relationship between the transmitter and the receiver.

This paper presents the performance analysis of a De-correlated Modified Code Tracking Loop (D-MCTL) for synchronous direct-sequence code-division multiple-access (DS-CDMA) systems under multiuser environment. Previous studies have shown that the imbalance of multiple access interference (MAI) in the time lead and time lag portions of the signal causes tracking bias or instability problem in the traditional correlating tracking loop like delay lock loop (DLL) or modified code tracking loop (MCTL). In this paper, we exploit the de-correlating technique to combat the MAI at the on-time code position of the MCTL. Unlike applying the same technique to DLL which requires an extensive search algorithm to compensate the noise imbalance which may introduce small tracking bias under low signal-to-noise ratio (SNR), the proposed D-MCTL has much lower computational complexity and exhibits zero tracking bias for the whole range of SNR, regardless of the number of interfering users. Furthermore, performance analysis and simulations based on Gold codes show that the proposed scheme has better mean square tracking error, mean-time-to-lose-lock and near-far resistance than the other tracking schemes, including traditional DLL (T-DLL), traditional MCTL (T-MCTL) and modified de-correlated DLL (MD-DLL).

A novel DC-to-960 MHz impulse radio ultra-wideband (IR-UWB) transceiver based on threshold detection technique is developed. It features a digital pulse-shaping transmitter, a DC power-free pulse discriminator and an error-recovery phase-frequency detector. The developed transceiver in 90 nm CMOS achieves the lowest energy consumption of 2.2 pJ/bit transmitter and 1.9 pJ/bit receiver at 100 Mbps in the UWB transceivers.

A wide range, low jitter Duty Cycle Corrector (DCC) based on continuous-time integrator is proposed. It introduces little added jitter in the sampling edge, which make it good candidate for pipelined ADC application. The circuit is implemented in CMOS 0.35 µm 2P4M Mixed Signal process. The experimental results show the circuit can work for a wide frequency range from 500 kHz to 280 MHz, with a correction error within 50%1% under 200 MHz, and the acceptable duty cycle can be as wide as 1-99% for low frequency inputs.

Two-dimensional (2D) matrix symbols have higher storage capacity than conventional bar-codes, and hence have been used in various applications, including parts management in factories and Internet site addressing in camera-equipped mobile phones. These symbols generally utilize strong error control codes to protect data from errors caused by blots and scratches, and therefore require a large number of check bits. Because 2D matrix symbols are expressed in black and white dot patterns, blots and scratches often induce clusters of unidirectional errors (i.e., errors that affect black but not white dots, or vice versa). This paper proposes a new class of unidirectional lm ln-clustered error correcting codes capable of correcting unidirectional errors confined to a rectangle with lm rows and ln columns. The proposed code employs 2D interleaved parity-checks, as well as vertical and horizontal arithmetic residue checks. Clustered error pattern is derived using the 2D interleaved parity-checks, while vertical and horizontal positions of the error are calculated using the vertical and horizontal arithmetic residue checks. This paper also derives an upper bound on the number of codewords based on Hamming bound. Evaluation shows that the proposed code provides high code rate close to the bound. For example, for correcting a cluster of unidirectional 40 40 errors in 150 150 codeword, the code rate of the proposed code is 0.9272, while the upper bound is 0.9284.

To mitigate the asynchronous ICI (Inter-Cell Interference), SCM (Spatial Covariance Matrix) of the asynchronous ICI plus background noise should be accurately estimated for MIMO-OFDMA (Multiple-input Multiple-output-Orthogonal Frequency Division Multiple Access) system. Generally, it is assumed that the SCM of the asynchronous ICI plus background noise is estimated by using training symbols. However, it is difficult to measure the interference statistics for a long time and considering that training symbols are not appropriate for OFDMA system such as LTE (3GPP Long Term Evolution). Therefore, noise reduction method is required to improve the estimation accuracy. Although the conventional time-domain low-pass type weighting method can be effective for noise reduction, it causes significant estimation error due to the spectral leakage in practical OFDM system. Therefore, we propose a time-domain sinc type weighing method which can not only reduce noise effectively minimizing estimation error caused by the spectral leakage but also can be implemented using frequency-domain weighted moving average filter easily. We also consider the iterative CFR (Channel Frequency Response) and SCM estimation method which can effectively reduce the estimation error of both CFR and SCM, and improve the performance for LTE system. By using computer simulation, we show that the proposed method can provide up to 2.5 dB SIR (Signal to Interference Ratio) gain compared with the conventional method, and verify that the proposed method is attractive and suitable for implementation with stable operation.

This paper proposes a method to construct a basis conversion matrix between two given bases in Fpm. In the proposed method, Gauss period normal basis (GNB) works as a bridge between the two bases. The proposed method exploits this property and construct a basis conversion matrix mostly faster than EDF-based algorithm on average in polynomial time. Finally, simulation results are reported in which the proposed method compute a basis conversion matrix within 30 msec on average with Celeron (2.00 GHz) when mlog p≈160.

In an embedded system where a single application or a class of applications is repeatedly executed on a processor, its cache configuration can be customized such that an optimal one is achieved. We can have an optimal cache configuration which minimizes overall memory access time by varying the three cache parameters: the number of sets, a line size, and an associativity. In this paper, we first propose two cache simulation algorithms: CRCB1 and CRCB2, based on Cache Inclusion Property. They realize exact cache simulation but decrease the number of cache hit/miss judgments dramatically. We further propose three more cache design space exploration algorithms: CRMF1, CRMF2, and CRMF3, based on our experimental observations. They can find an almost optimal cache configuration from the viewpoint of access time. By using our approach, the number of cache hit/miss judgments required for optimizing cache configurations is reduced to 1/10-1/50 compared to conventional approaches. As a result, our proposed approach totally runs an average of 3.2 times faster and a maximum of 5.3 times faster compared to the fastest approach proposed so far. Our proposed cache simulation approach achieves the world fastest cache design space exploration when optimizing total memory access time.

In this paper, we analyze the extended real-time Polling Service (ertPS) algorithm in IEEE 802.16e systems, which is designed to support Voice-over-Internet-Protocol (VoIP) services with data packets of various sizes and silence suppression. The analysis uses a two-dimensional Markov Chain, where the grant size and the voice packet state are considered, and an approximation formula for the total throughput in the ertPS algorithm is derived. Next, to improve the performance of the ertPS algorithm, we propose an enhanced uplink resource allocation algorithm, called the e 2rtPS algorithm, for VoIP services in IEEE 802.16e systems. The e 2rtPS algorithm considers the queue status information and tries to alleviate the queue congestion as soon as possible by using remaining network resources. Numerical results are provided to show the accuracy of the approximation analysis for the ertPS algorithm and to verify the effectiveness of the e 2rtPS algorithm.

In this paper, we propose a three-phase text error correction model consisting of a word spacing error correction phase, a syllable-based spelling error correction phase, and a word-based spelling error correction phase. In order to reduce the text error correction complexity, the proposed model corrects text errors step by step. With the aim of correcting word spacing errors, spelling errors, and mixed errors in SMS messages, the proposed model tries to separately manage the word spacing error correction phase and the spelling error correction phase. For the purpose of utilizing both the syllable-based approach covering various errors and the word-based approach correcting some specific errors accurately, the proposed model subdivides the spelling error correction phase into the syllable-based phase and the word-based phase. Experimental results show that the proposed model can improve the performance by solving the text error correction problem based on the divide-and-conquer strategy.

Frequency-domain equalization (FDE) has been studied for suppressing inter-symbol interference (ISI) due to frequency selective fading in single carrier systems. When a high-mobility terminal is assumed in the system, channel transition within an FDE block cannot be ignored. The ISI reduction performance of FDE degrades since the cyclicity of the channel matrix is lost. To solve this problem, a method of dividing the received data block into multiple subblocks has been proposed, where pseudo cyclic prefix (CP) processing is introduced to realize periodicity in each subblock. In this method, the performance is degraded by the inherently-inaccurate pseudo CP. In this paper, we study the application of frequency-domain turbo equalization (FDTE) to subblock processing for improving the accuracy of pseudo CP. The simulation results show that FDTE with subblock processing yields remarkable performance improvements.

Codebook based multiple-input multiple-output (MIMO) precoding can significantly improve the system spectral efficiency with limited feedback and has been accepted as one of the most promising techniques for the Evolved UTRA (E-UTRA). Compared with single-user (SU) MIMO, multi-user (MU) MIMO can further improve the system spectral efficiency due to increased multi-user diversity gain. MU-MIMO is preferred for the case of a large number of users,when the total feedback overhead will become a problem. In order to reduce the feedback overhead, feedback of single channel quality indicator (CQI), e.g. rank 1 CQI, is required in E-UTRA currently. The main challenge is how to obtain CQIs of other ranks at Node B for rank adaptation with single CQI feedback. In this paper, an adaptive CQI update scheme at Node B based on statistical characteristics of CQI of various ranks is proposed. To further increase the accuracy of CQI at Node B for data transmission, an adaptive CQI feedback scheme is then proposed in which single CQI with the rank same as previously scheduled is fed back. Simulation results show that our proposed CQI update scheme can achieve 2.5-5% gain compared with the conventional method with fixed backoff. Moreover, with the proposed adaptive feedback scheme, 20-40% performance gain can be obtained and the performance can approach the upper bound.

This paper proposes block-wise resource block (RB)-level distributed OFDMA transmission with ND-block division in order to obtain the frequency diversity effect even for low-rate traffic (here ND indicates the number of virtual RBs within one physical RB) in Evolved UTRA downlink. More specifically, we propose a constraint rule such that distributed transmission is multiplexed into a different physical RB from that of localized transmission in order to achieve the same resource assignment and independent decoding between the distributed and localized transmissions. Based on the proposed rule, a virtual RB for distributed transmission is segmented into ND blocks with the size of 1/ND of the original virtual RB. Then, the ND virtual blocks with the size of 1/ND are mapped together into each ND physical RB in a distributed manner, resulting in a large frequency diversity effect. Numerical calculations show that the block-wise RB-level distributed transmission can reduce the number of control signaling bits required for resource assignment compared to the subcarrier-level distributed transmission scheme, which provides the best performance. Moreover, a system-level simulation shows that the loss in the cell throughput employing the block-wise RB-level distributed transmission compared to that using the subcarrier-level transmission is only within 3-4% when the channel load is 0.5 and 1.0, i.e., the maximum loss is 3-4% at approximately 90% in the cumulative distribution function (CDF).