We have developed two modulator driver ICs that are based on the functional distributed circuit (FDC) topology for over 40-Gb/s optical transmission systems using InP HBT technology. The FDC topology enables both a wide bandwidth amplifier and high-speed digital functions. The none-return-to-zero (NRZ) driver IC, which is integrated with a D-type flip-flop, exhibits 2.6-Vp-p (differential output: 5.2 Vp-p) output-voltage swings with a high signal quality at 43 and 50 Gb/s. The return-to-zero (RZ) driver IC, which is integrated with a NRZ to RZ converter, produces 2.4-Vp-p (differential output: 4.8 Vp-p) output-voltage swings and excellent eye openings at 43 and 50 Gb/s. Furthermore, we conducted electro-optical modulation experiments using the developed modulator driver ICs and a dual drive LiNbO3 Mach-Zehnder modulator. We were able to obtain NRZ and RZ clear optical eye openings with low jitters and sufficient extinction ratios of more than 12 dB, at 43 and 50 Gb/s. These results indicate that the FDC has the potential to achieve a large output voltage and create high-speed functional ICs for over-40-Gb/s transmission systems.

Existing query tree protocols deal with RFID tags in a blind manner. They query tags in a fixed bit order based on the assumption that the tag ID numbers are uniformly distributed throughout the range of the entire ID space because readers have no prior knowledge of the tags. This paper attempts to distinguish RFID applications where readers are already aware of all tags used by the application. We propose a heuristic query tree (H-QT) protocol that uses heuristic to select effective bits from known tags for the best queries in a divide and conquer approach. The performance evaluation shows that the proposed protocol is superior to original query tree protocols because it significantly reduces the number of tag collisions and no tag response.

In this paper, we introduce a generalized minimum mean-square error short-time spectral amplitude estimator with a new prior estimation of the speech probability density function based on moment-cumulant transformation. From the objective and subjective evaluation experiments, we show the improved noise reduction performance of the proposed method.

We present area- and power-efficient pipeline 128- and 128/64-point fast Fourier transform (FFT) processors for 8x8 multiple-input multiple-output orthogonal frequency multiplexing (MIMO-OFDM) systems based on the specification framework of IEEE 802.11ac WLANs. Our new FFT processors use mixed-radix multipath delay commutator (MRMDC) architecture from the point of view of low complexity and high memory use. A conventional MRMDC architecture induces large circuits in delay commutators, which change the order of data sequences for the butterfly units. The proposed architecture replaces delay elements with new commutators that cooperate with other MIMO-OFDM processing blocks. These commutators are inserted in the front and rear of the input and output memory units. Our FFT processors exhibit a 50–51% reduction in logic gates and 70–72% reduction in power dissipation as compared with conventional ones.

The charge-redistribution low-swing differential logic (CLDL) circuits are presented in this work. It can implement a complex function in a single gate. The CLDL circuits utilizes the charge-redistribution and reduced-swing schemes to reduce the power dissipation and enhance the operation speed. In addition, a pipeline structure is formed by a series connection structure controlled by a true-single-phase clock, thereby achieving high-speed operation. The CLDL circuits perform more than 25% speedup and 31% in power-delay product compared to other differential circuits with true-single-phase clock. A pipelined multiplier-accumulator (MAC) using CLDL structure is fabricated in 0.35 µm single-poly four-metal CMOS process. The test chip is successfully verified to operate at 900-MHz.

This paper proposes a method of designing EM absorber panels under oblique incident waves. TM and TE wave reflection characteristics of the absorber panel show its anisotropy under oblique incidence. By using the wire array sheet proposed this paper, TM and TE reflection coefficients in oblique incidence can be matched at almost the same frequency range.

The peer-assisted streaming paradigm has been widely employed to distribute live video data on the internet recently. In general, the mesh-based pull approach is more robust and efficient than the tree-based push approach. However, pull protocol brings about longer streaming delay, which is caused by the handshaking process of advertising buffer map message, sending request message and scheduling of the data block. In this paper, we propose a new approach, mesh-push, to address this issue. Different from the traditional pull approach, mesh-push implements block scheduling algorithm at sender side, where the block transmission is initiated by the sender rather than by the receiver. We first formulate the optimal upload bandwidth utilization problem, then present the mesh-push approach, in which a token protocol is designed to avoid block redundancy; a min-cost flow model is employed to derive the optimal scheduling for the push peer; and a push peer selection algorithm is introduced to reduce control overhead. Finally, we evaluate mesh-push through simulation, the results of which show mesh-push outperforms the pull scheduling in streaming delay, and achieves comparable delivery ratio at the same time.

This paper demonstrates a pulse width controlled PLL without using an LPF. A pulse width controlled oscillator accepts the PFD output where its pulse width controls the oscillation frequency. In the pulse width controlled oscillator, the input pulse width is converted into soft thermometer code through a time to soft thermometer code converter and the code controls the ring oscillator frequency. By using this scheme, our PLL realizes LPF-less as well as quantization noise free operation. The prototype chip achieves 60 µm 20 µm layout area using 65 nm CMOS technology along with 1.73 ps rms jitter while consuming 2.81 mW under a 1.2 V supply with 3.125 GHz output frequency.

Multimedia applications such as music or video streaming, video teleconferencing and IP telephony are flourishing in packet-switched networks. Applications that generate such real-time data can have very diverse quality-of-service (QoS) requirements. In order to guarantee diverse QoS requirements, the combined use of a packet scheduling algorithm based on Generalized Processor Sharing (GPS) and leaky bucket traffic regulator is the most successful QoS mechanism. GPS can provide a minimum guaranteed service rate for each session and tight delay bounds for leaky bucket constrained sessions. However, the delay bounds for leaky bucket constrained sessions under GPS are unnecessarily large because each session is served according to its associated constant weight until the session buffer is empty. In order to solve this problem, a scheduling policy called Output Rate-Controlled Generalized Processor Sharing (ORC-GPS) was proposed in [17]. ORC-GPS is a rate-based scheduling like GPS, and controls the service rate in order to lower the delay bounds for leaky bucket constrained sessions. In this paper, we propose a call admission control (CAC) algorithm for ORC-GPS, for leaky-bucket constrained sessions with deterministic delay requirements. This CAC algorithm for ORC-GPS determines the optimal values of parameters of ORC-GPS from the deterministic delay requirements of the sessions. In numerical experiments, we compare the CAC algorithm for ORC-GPS with one for GPS in terms of schedulable region and computational complexity.

A method for efficiently representing the state equation in a class of max-plus linear systems is proposed. We introduce a construct referred to as 'cell' in which the list of possible longest paths is stored. By imposing interval constraints on the system parameters, we can reduce the complexity of the state equation. The proposed method would be useful in scheduling applications for systems with adjustable system parameters.

The IEEE 802.15.4 protocol is considered a promising technology for low-cost low-power wireless personal area networks. Researchers have discussed the feasibility of voice communications over IEEE 802.15.4 networks. To this end, the personal area network (PAN) coordinator allocates guaranteed time slots (GTSs) for voice communications in the beacon-enabled mode of IEEE 802.15.4. Although IEEE 802.15.4 is capable of supporting voice communications by GTS allocation, it is impossible to accommodate voice transmission beyond two hops due to the excessive transmission delay. In this paper, we propose a GTS allocation scheme for bidirectional voice traffic in IEEE 802.15.4 multihop networks. The goal of our proposed scheme is to achieve low end-to-end delay and packet drop ratio without a complex allocation algorithm. Thus, the proposed scheme allocates GTSs to devices for successful completion of voice transmission in a superframe duration. The proposed scheme also considers transceiver switching delay. This is relatively large compared to a time slot due to the low-cost and low-gain antenna designs. We analyze and validate the proposed scheme in terms of average end-to-end delay and packet drop ratio. Our scheme has lower end-to-end delay and packet drop ratio than the basic IEEE 802.15.4 GTS allocation scheme.

We propose a structure-generalized blind spatial subtraction array (BSSA), and the theoretical analysis of the amounts of musical noise and speech distortion. The structure of BSSA should be selected according to the application, i.e., a channelwise BSSA is recommended for listening but a conventional BSSA is suitable for speech recognition.

We propose a novel scheme to learn a language model (LM) for automatic speech recognition (ASR) directly from continuous speech. In the proposed method, we first generate phoneme lattices using an acoustic model with no linguistic constraints, then perform training over these phoneme lattices, simultaneously learning both lexical units and an LM. As a statistical framework for this learning problem, we use non-parametric Bayesian statistics, which make it possible to balance the learned model's complexity (such as the size of the learned vocabulary) and expressive power, and provide a principled learning algorithm through the use of Gibbs sampling. Implementation is performed using weighted finite state transducers (WFSTs), which allow for the simple handling of lattice input. Experimental results on natural, adult-directed speech demonstrate that LMs built using only continuous speech are able to significantly reduce ASR phoneme error rates. The proposed technique of joint Bayesian learning of lexical units and an LM over lattices is shown to significantly contribute to this improvement.

In next-generation networks, ultra-high speed transfer capability will become necessary to support a variety of advanced multimedia services. The Optical Wavelength Division Multiplexing (WDM) network is seen as one of promising networks. To deal with various multimedia services, the network should support not only unicast transmission but also multicast transmission. However, IP multicast has several problems, for example, all routers must have multicast functions. IP multicast needs routers with high overheads and excessive energy consumption. Hence, optical multicasting in optical WDM networks is a promising solution for future internet services. A tree-shared multicasting concept has been proposed to support multicast transmissions in optical WDM networks. This method assembles multiple multicast traffic streams into one burst and the burst is delivered using a shared-tree. However, this method can not be applied to dynamic multicasting. This paper proposes a novel WDM multicast design method for dynamic traffic demand using multiple shared-trees, which includes shared-tree generation/selection and wavelength assignment, for the purpose of simplifying the routing process and receiving multicast traffic efficiently. We evaluate its performance from the viewpoints of the burst loss probability and the number of redundant and useless transfers whose data is discarded at the egress edge nodes.

The energy consumption of the information and communication technology (ICT) industry, which has become a serious problem, is mostly due to the network infrastructure rather than the mobile terminals. In this paper, we focus on reducing the energy consumption of base stations (BSs) by adjusting their working modes (active or sleep). Specifically, the objective is to minimize the energy consumption while satisfying quality of service (QoS, e.g., blocking probability) requirement and, at the same time, avoiding frequent mode switching to reduce signaling and delay overhead. The problem is modeled as a dynamic programming (DP) problem, which is NP-hard in general. Based on cooperation among neighboring BSs, a low-complexity algorithm is proposed to reduce the size of state space as well as that of action space. Simulations demonstrate that, with the proposed algorithm, the active BS pattern well meets the time variation and the non-uniform spatial distribution of system traffic. Moreover, the tradeoff between the energy saving from BS sleeping and the cost of switching is well balanced by the proposed scheme.

P2P (peer-to-peer) file sharing systems have been in operation for years. However, recent studies show that many peers in P2P networks are free-riders, who download files but are unwilling to share. This paper proposes a randomized search algorithm that considers the potential of a peer for information exchange and controlled query forwarding in the search process. Based on churn situations, the simulation results in this study demonstrate that the proposed algorithm can reduce network traffic and search latency while searching for files in the system.

We propose a mobility enhancement method in which APs periodically change their RF channels in a predetermined order that prevents overlap of neighboring APs' channels. Improvement in the throughput is also achieved by manipulating the DCF uplink mode and the Downlink mode. A performance evaluation shows that the proposed scheme is superior to IEEE 802.11 WLAN in handoff delay and throughput.

Network structure has a great impact both on hazard spread and network immunization. The vulnerability of the network node is associated with each other, assortative or disassortative. Firstly, an algorithm for vulnerability relevance clustering is proposed to show that the vulnerability community phenomenon is obviously existent in complex networks. On this basis, next, a new indicator called network “hyper-betweenness” is given for evaluating the vulnerability of network node. Network hyper-betweenness can reflect the importance of network node in hazard spread better. Finally, the dynamic stochastic process of hazard spread is simulated based on Monte-Carlo sampling method and a two-player, non-cooperative, constant-sum game model is designed to obtain an equilibrated network immunization strategy.

Since the number of server providing the facilities for users is usually more than one, remote user authentication schemes used for multi-server architectures, rather than single server circumstance, is considered. As far as security is concerned, privacy is the most important requirements, though some other properties are also desirable in practice. Recently, a number of dynamic ID-based user authentication schemes have been proposed. However, most of those schemes have more or less weaknesses and/or security flaws. In the worst case, user privacy cannot be achieved since malicious servers or users can mount some attacks, i.e., server spoofing attack and impersonation attack, to identify the unique identifier of users and masquerade of one entity as some other. In this paper, we analyze two latest research works and demonstrate that they cannot achieve true anonymity and have some other weaknesses. We further propose the improvements to avoid those security problems. Besides user privacy, the key features of our scheme are including no verification table, freely chosen password, mutual authentication, low computation and communication cost, single registration, session key agreement, and being secure against the related attacks.

This paper proposes a low power artificial bandwidth extension (ABE) technique that reduces computational complexity by introducing a fast codebook mapping method. We also introduce a weighted classified codebook mapping method for constructing the spectral envelope of the wideband speech signal. Classified codebooks are used to reduce spectrum mapping errors caused by characteristic difference among voiced, unvoiced and onset sound. The weighted distortion measure is also used to handle the spectral sensibility. The performance of the proposed ABE system is evaluated by a spectral distortion (SD), a perceptual evaluation of speech quality (PESQ), informal listening tests and weighted million operations per second (WMOPS) calculations. With the use of fast codebook mapping, the WMOPS complexity of the codebook mapping module is reduced by 45.17%.