RFID event filtering is an important issue of RFID data management. Tag read events from readers have some problems like unreliability, redundancy, and disordering of tag readings. Duplicated events lead to performance degradation of RFID systems with a flood of similar tag information. Therefore, this paper proposes a fuzzy logic-based quantized event filter. In order to reduce duplicated tag readings and solve disordering of tag readings, the filter applies a fuzzy logic system to control a filtering threshold by the change in circumstances of readers. Continuous tag readings are converted into discrete values for event generation by the filtering threshold. And, the filter generates as many events as the discrete values at a point of event generation time. Experimental results comparing the proposed filter with existing RFID event filters, such as the primitive event filter and the smoothing event filter, verify effectiveness and efficiency of the fuzzy logic-based quantized event filter.

A simple transmit diversity scheme based on cyclic-shift pseudo noise (PN) sequences for the time domain synchronous orthogonal frequency division multiplexing (TDS-OFDM) system is proposed and the corresponding channel estimation algorithms are also investigated. Simulation results are presented to verify the effectiveness of the proposed method.

To guarantee the timely provisioning of QoS to real-time oriented multiparty and distributed computing applications (e.g., video conferencing and grid computing) that require the utilization of time and quantity-related resources, the resource should be reserved in advance. However, this new type of reservation (i.e., advance reservation) may collide with legacy (i.e., immediate) reservations that do not specify their session duration. In this paper, to suggest a balanced solution between sharing pools of resources and the managing of collisions between reservation calls, a revenue-based resource sharing scheme (focusing on the bandwidth) is proposed. The proposed scheme attempts to avoid possible collisions by employing a virtual profile of resources reserved in advance in an effort to achieve enhanced resource utilization. It is shown through NS-2 [22] -based network simulations that the proposed scheme can achieve balanced performance when compared with other schemes, including static resource partitioning and complete resource sharing.

A highly integrated frequency quadrupler MMIC that uses three-dimensional MMIC (3D-MMIC) technology is presented. It consists of four driver amplifiers, two doublers, and a 2-band elimination filter. These seven circuits are integrated in only a 2.36 mm2 area. The filter sufficiently suppresses spurious output components. The third and fifth harmonic components, which are the spurious components nearest to the desired component, are well suppressed. The desired/undesired ratio is about 40 dB. The driver amplifiers make the quadrupler output a constant power of the desired multiplied signal under low input power. The MMIC supplies +5 dBm of the fourth harmonic component in the input power range from -10 dBm to +5 dBm. The power dissipation of the MMIC is only 160 mW.

To realize low-power wireless transceivers, it is necessary to improve the performance of frequency synthesizers, which are typically frequency multipliers composed of a phase-locked loop (PLL). However, PLLs generally consume a large amount of power and occupy a large area. To improve the frequency multiplier, we propose a pulse-injection-locked frequency multiplier (PILFM), where a spurious signal is suppressed using a pulse input signal. An injection-locked oscillator (ILO) in a PILFM was fabricated by a 0.18 µm 1P5M CMOS process. The core size is 10.8 µm10.5 µm. The power consumption of the ILO is 9.6 µW at 250 MHz, 255 µW at 2.4 GHz and 1.47 mW at 4.8 GHz. The phase noise is -105 dBc/Hz at a 1 MHz offset.

Since an FFT-based speech encryption system retains a considerable residual intelligibility, such as talk spurts and the original intonation in the encrypted speech, this makes it easy for eavesdroppers to deduce the information contents from the encrypted speech. In this letter, we propose a new technique based on the combination of an orthogonal frequency division multiplexing (OFDM) scheme and an appropriate QAM mapping method to remove the residual intelligibility from the encrypted speech by permuting several frequency components. In addition, the proposed OFDM-based speech encryption system needs only two FFT operations instead of the four required by the FFT-based speech encryption system. Simulation results are presented to show the effectiveness of this proposed technique.

In block transmission systems, blind channel shortening methods are known to be effective to reduce the influence of interblock interference which degrades the performance when the length of a channel impulse response is extremely long. Conventional methods assume that the transmitted signal is uncorrelated; however, this assumption is invalid in practical systems such as OFDM with null carriers and MC-CDMA. In this paper, we consider blind channel shortening methods for block transmissions when the transmitted samples within a block are correlated. First, the channel shortening ability of a conventional method is clarified. Next, a new method which exploits the fact that the transmitted samples in different blocks are uncorrelated is introduced. It is shown that the proposed method can shorten the channel properly under certain conditions. Finally, simulation results of OFDM and MC-CDMA systems are shown to verify the effectiveness of the proposed method compared with a conventional one.

Power-line communication (PLC) technique is one method to realize high-speed communications in a home network. In PLC channels, the transmission signal quality is degraded by colored non-Gaussian noise as well as frequency-selectivity of the channels. In this paper, we describe our investigation of the performance of a OQAM-MCT system in which a noise canceller is used jointly with a time-domain per-subcarrier adaptive equalizer. Furthermore, we propose a noise cancellation method designed for the OQAM-MCT system. The performance of the OQAM-MCT system is evaluated in PLC channels with measured impulse responses in the presence of measured noise. Computer simulation results show that the bit rate of the OQAM-MCT system is improved using both an adaptive equalizer and noise canceller, and that the OQAM-MCT system achieves better performance than an OFDM system with an insufficient length of the guard interval.

In real-time embedded systems, there is requirement for adapting both energy consumption and Quality of Services (QoS) of tasks according to their importance. This paper proposes an adaptive power-aware resource allocation method to resolve a trade-off between the energy consumption and QoS levels according to their importance with guaranteeing fairness. The proposed resource allocator consists of two components: the total resource optimizer to search for the optimal total resource and QoS-fairness-based allocator to allocate resource to tasks guaranteeing the fairness. These components adaptively achieve the optimal resource allocation formulated by a nonlinear optimization problem with the time complexity O(n) for the number of tasks n even if tasks' characteristics cannot be identified precisely. The simulation result shows that the rapidness of the convergence of the resource allocation to the optimal one is suitable for real-time systems with large number of tasks.

In this letter we propose a robust detection algorithm for audio watermarking for copyright protection. The watermark is embedded in the time domain of an audio signal by the normally used spread spectrum technique. The scheme of detection is an improvement of the conventional correlation detector. A high-pass filter is applied along with the linear prediction error filter for whitening the audio signal and an adaptive threshold is chosen for decision comparing. Experimental results show that our detection algorithm outperforms the conventional one not only because it improves the robustness to normal attacks but also because it can provide the robustness to time-invariant pitch-scale modification.

This letter presents an efficient adaptive beamformer to deal with the multipath environments created by signal source scatterings. To improve the performance possible with the fixed forgetting factor, the regular adaptive forgetting factor algorithm is derived and applied to the subarray recursive least squares (RLS) beamforming. Simulations confirm that the proposed scheme has better performance than not only the conventional RLS algorithm but also the subarray RLS and adaptive forgetting factor RLS algorithms.

Recent rapid development of high-speed wireless access technologies has created mixed WLAN (Wireless LAN) environments where QoS capable APs coexist with legacy APs. To provide QoS guarantee in this mixed WLAN environment, this paper proposes a new AP selection algorithm. The proposed algorithm assigns an STA (Station) to an AP in the overall WLAN service area. Simulation results show improvement in the VoIP performance in terms of an eMOS (estimated Mean Opinion Score) value and the FTP throughput compared to conventional algorithms.

In next generation mobile multimedia communications, different wireless access networks are expected to cooperate. However, it is a challenging task to choose an optimal transmission path in this scenario. This paper focuses on the problem of selecting the optimal access network for multicast services in the cooperative mobile and broadcasting networks. An algorithm is proposed, which considers multiple decision factors and multiple optimization objectives. An analytic hierarchy process (AHP) method is applied to schedule the service queue and an artificial neural network (ANN) is used to improve the flexibility of the algorithm. Simulation results show that by applying the AHP method, a group of weight ratios can be obtained to improve the performance of multiple objectives. And ANN method is effective to adaptively adjust weight ratios when users' new waiting threshold is generated.

The authors have demonstrated the local alignment of nematic liquid crystal with local micro-grating structure fabricated by the curing of an ultraviolet curable material via a three dimensional micro-fabrication technique known as two photon excitation direct laser writing [1]. The molecular alignment of the nematic liquid crystals on the fabricated micro-grating structures was firstly investigated by the observations of a local twisted nematic region in a liquid crystal cell made of a substrate with locally fabricated micro-grating structure and a counter substrate with rubbed polyimide. The optical polarizing microscope observation of the micro-grating structures indicated that liquid crystals molecules have aligned parallel to the grooves of the micro-grating structure and that local alignment was successfully achieved. The alignment characteristics of the liquid crystals on these micro-gratings was also investigated and discussed quantitatively in details through the measurement of anchoring energy by the conventional torque balance method and the Berreman method. The azimuthal anchoring energy for the micro-grating was found to be in the order of 10-6 J/m2 and inversely proportional to the grating period.

Focusing on time correlation of real communication channels, a channel quantization algorithm based on finite state vector quantization (FSVQ) is proposed. Firstly channels are partitioned into finite states, then codebooks corresponding to each state are constructed, which are used to quantize channels transferred from corresponding states. Further, the state transition function is designed to ensure the synchronization between transmitter and receiver. The proposed algorithm can achieve improved performance with the same feedback load compared with classical memoryless channel quantizer without consideration of the influence of time correlation. Simulation results verify the effectiveness of the proposed algorithm.

Transmit diversity systems based on orthogonal space-time block coding (OSTBC) usually suffer from rate loss and power spreading. Proper antenna selection scheme can help to more effectively utilize the transmit antennas and transmission power in such systems. In this paper, we propose a new antenna selection scheme for such systems based on the idea of antenna switching. In particular, targeting at reducing the number of pilot channels and RF chains, the transmitter now replaces the antennas with the lowest received SNR with unused ones if the output SNR of space time decoder at the receiver is below a certain threshold. With this new scheme, not only the number of pilot channels and RF chains to be implemented is decreased, the average amount of feedback information is also reduced. To analyze the performance of this scheme, we derive the exact integral closed form for the probability density function (PDF) of the received SNR. We show through numerical examples that the proposed scheme offers better performance than traditional OSTBC systems using all available transmitting antennas, with a small amount of feedback information. We also examine the effect of different antenna configuration and feedback delay.

This paper proposes the Gramian-preserving frequency transformation for linear discrete-time state-space systems. In this frequency transformation, we replace each delay element of a discrete-time system with an allpass system that has a balanced realization. This approach can generate transformed systems that have the same controllability/observability Gramians as those of the original system. From this result, we show that the Gramian-preserving frequency transformation gives us transformed systems with different magnitude characteristics, but with the same structural property with respect to the Gramians as that of the original system. This paper also presents a simple method for realization of the Gramian-preserving frequency transformation. This method makes use of the cascaded normalized lattice structure of allpass systems.

This paper performs application-level QoS and user-level QoS assessment of audio-video streaming in cross-layer designed wireless ad hoc networks. In order to achieve high QoS at the user-level, we employ link quality-based routing in the network layer and media synchronization control in the application layer. We adopt three link quality-based routing protocols: OLSR-SS (Signal Strength), AODV-SS, and LQHR (Link Quality-Based Hybrid Routing). OLSR-SS is a proactive routing protocol, while AODV-SS is a reactive one. LQHR is a hybrid protocol, which is a combination of proactive and reactive routing protocols. For application-level QoS assessment, we performed computer simulation with ns-2 where an IEEE 802.11b mesh topology network with 24 nodes was assumed. We also assessed user-level QoS by a subjective experiment with 30 assessors. From the assessment results, we find AODV-SS the best for networks with long inter-node distances, while LQHR outperforms AODV-SS for short inter-node distances. In addition, we also examine characteristics of the three schemes with respect to the application-level QoS in random topology networks.

This paper proposes an improved dynamic bandwidth allocation algorithm for dual Quality of Service (QoS) classes to maximize the utilization rate of the Resilient Packet Ring (RPR). To achieve dynamic bandwidth allocation for the two QoS classes in the RPR, each node measures the high priority traffic flow and assigns the appropriate bandwidth; the remaining bandwidth is used for low priority traffic. It passes a control frame containing the measured bandwidth of the high priority traffic to the other nodes. Based on the advertised high priority traffic bandwidth, any node that is congested transmits, to the other nodes, a fairness message to fairly allocate the remaining low priority bandwidth. Simulations demonstrate that the proposed algorithm enhances the utilization rate and reduces the delay of high priority frames.

Frequency offset estimation is an important technique in receiver design of wireless communications. In many applications, sampled single frequency tone is selected as training symbol/sequence for frequency synchronization. Under this assumption, frequency offset estimation can be regarded as the problem of single carrier frequency offset estimation. In this Letter, an approximate maximum likelihood frequency estimator is proposed. This estimator is efficient at moderate and high SNR's. Compared with other estimators, the proposed estimator is less sensitive to the variance threshold and offers feasible levels of computation complexity. The proposed estimator is suitable for high frequency offset cases and coarse/fine frequency synchronization applications.