The present paper proposes two novel and practical schemes for distributed and asynchronous power control in wireless ad hoc networks, in which users dynamically share several frequency bands as in "cognitive radio" networks. These schemes iteratively adjust transmit powers of individual network transmitters with respect to mutually caused interference in the shared bands. Their most attractive feature is that they find network-wide acceptable trade-offs to diverse signal-to-noise and interference (SINR) requirements and efficiently use techniques of stochastic approximation and time-averaging to guarantee a robust performance in random channels. Advantageously, both proposed algorithms do not assume any particular modulation, coding, QoS measure definition or network architecture, which assures their high applicability in the industry and research. Moreover, the broad definition and non-linear nature of these schemes mathematically generalize and thus encompass as a special case many widely deployed power control schemes such as e.g. those for achieving fixed SINR targets or using game-theoretic utility maximization. Simulations are provided to illustrate our approach and its better performance compared to standard algorithms.

The traffic with asymmetry between uplink and downlink has recently been getting remarkable on mobile communication systems providing multimedia communication services. In the future mobile communications, the accommodation of asymmetric traffic is essential to realize efficient multimedia mobile communication systems. This paper discusses asymmetric traffic accommodation in CDMA/FDD cellular packet communication systems and proposes its efficient scheme using an adaptive cell sizing technique. In the proposed scheme, each base station autonomously controls its coverage area so that almost the same communication quality can be achieved across the service area under the asymmetric traffic conditions. We present some numerical examples to demonstrate the effectiveness of the proposed scheme by using computer simulation. The simulation results show that, under asymmetric traffic conditions, the proposed scheme can provide fair communication quality across the service area in both links and can improve total transmission capacity in the uplink.

While efficient use of network resources is an important control objective of call admission control (CAC), the issue of fairness among services should also be taken into account. Game theory provides a suitable framework for formulating such fair and efficient CAC problem. Thus, in this paper, a game theoretic framework for selecting fair-efficient threshold parameters of CAC for the asymmetrical traffic case in CDMA mobile multimedia systems is proposed. For the cooperative game, the arbitration schemes for the interpersonal comparisons of utility and the bargaining problem, including the Nash, Raiffa, and modified Thomson solutions, are investigated. Furthermore, since CAC should be simple and flexible to provide a fast response to diverse QoS call requests during a connection setup, this paper also applies the concept of load factor to the previous Jeon and Jeong's CAC scheme and proposes an approximation approach to reduce the computational complexity (proposed throughput-based CAC scheme). From the numerical results, the proposed throughput-based CAC scheme shows a comparable performance to the previous Jeon and Jeong's CAC scheme while achieving lower computational complexity. All the solutions attain the fairness by satisfying their different fairness senses and efficiency by the Pareto optimality.

Multilink MIMO technique is a promising technology for cellular networks with a guaranteed quality-of-service. It will provide high capacity and wide coverage. We evaluated the downlink performance of the multilink MIMO system from the perspective of quality-of-service. The presence of Rayleigh fading, shadowing, and path loss was assumed. To evaluate the proposed system, we developed a performance measure for MIMO cellular system. The measure is ergodic capacity taking into account area coverage. Our numerical results show that the area coverage of proposed multilink MIMO system is greatly improved compared with that of the conventional singlelink MIMO system. Using the proposed measure, we also found that the multilink MIMO system could achieve high capacity with guaranteed QoS for a wide coverage.

The conventional successive adaptive loading algorithm, represented by Hughes-Hartogs algorithm, can be used to reduce the transmit power of orthogonal frequency division multiplexing (OFDM) system. However, two major disadvantages exist for this kind of algorithm: One is the long loading time delay caused by the bit-to-bit loading, i.e., only one bit is loaded in every iteration; the other is that there is no flexibility in freely pre-defining the candidate modulation set before the loading is finished. In order to solve these problems, we propose the low-loading-delay parallel adaptive loading algorithms aiming at reducing the transmit power under the condition that the data throughput and error rate are maintained to target values. Two improvements are achieved by the new algorithm. One is that it divides the successive adaptive loading into several independent small-scale loading (SS-Loading) procedures. "SS-Loading" can be performed in parallel mode. To support this, we propose two subband division methods (successive and sorted subband divisions). The simulation results show that for a large range of subband number (1-128), the loading time delay can be remarkably decreased (especially for the parallel adaptive loading based on sorted subband division, i.e., SRT parallel adaptive loading algorithm) with neglectable power efficiency loss, compared with Hughes-Hartogs algorithm. The second improvement is that the new algorithm allows us to pre-define the candidate modulation set, which provide flexibility for the system design, e.g. we can exclude those rarely used modulation modes. We also reveal that Hughes-Hartogs algorithm is actually a special case of the newly proposed algorithm.

We present an effective method of collision recovery for orthogonal frequency division multiplexing (OFDM)-based communications. For the OFDM system, the modulated message data can be demodulated using the partial time-domain OFDM signal. Therefore, the partial time-domain signal can be adopted to reconstruct the whole OFDM time-domain signal with estimated channel information. This property can be utilized to recover packets from the collisions. Since most collisions are cases in which a long packet collides with a short packet, the collided part is assumed to be short. The simulated results show that the method can recover the two collided packets with a certain probability and can be developed to solve the problem of hidden terminals. This method will dramatically benefit the protocol design of wireless networks, including ad hoc and sensor networks.

In this paper, we propose a novel highly reliable packet transmission protocol for ad-hoc wireless networks without deciding the route from a source node to a destination node in advance. The proposed protocol uses distributed surrounding nodes as transmitters of the retransmission packet when the transmitted packet contains an error. In this protocol, when a packet is not correctly received at the destination node, the source node and the surrounding nodes that have correctly received the packet simultaneously retransmit the same data packet to the destination node. The transmitting timing is triggered by the control packet transmitted from the source node. These operations are repeated until the packet reaches to the destination node like automatic repeat request (ARQ). Moreover, the retransmitted packet is encoded by one branch of the space time block code (STBC) for improving the performance with the network diversity effect. We call this method as STBC Distributed ARQ scheme. The performance of the proposed scheme is evaluated by the computer simulations.

Wireless mesh networks (WMNs) are gaining significant momentum as a promising technology for the next-coming state-of-the-art wireless networking. Among many factors, the performance of WMNs would be largely affected by the properness of the deployed routing protocols and the efficient usage of wireless resources. Routing protocols are required to well capture WMNs' features while wireless channels should be used efficiently in order to accommodate high amount of traffics over the mesh backbone. Recently, a Tree-based Routing (TBR) protocol become a popular state-of-the-art proactive routing protocol and its tree-based broadcasting become an often used technique. Though TBR protocol is well-suited for WMNs' architecture and the skewed nature of traffic toward the root, the protocol in its current form faces issues which has to be addressed. Specifically, when some or all nodes are equipped with multiple radios, to reduce collision and co-channel interference, not only the parent-child relationship but also the sibling relationship need to be constructed by the TBR protocol in the multi-channel WMNs. In this paper, we propose a hybrid tree-based protocol for concurrent routing and channel assignment over WMNs. The protocol makes use of sibling links to mitigate the aforementioned shortcomings of TBR protocol. Moreover, in order to address high backbone traffic, the protocol integrates a receiver-based channel assignment scheme. The protocol efficiently deploys the parent-child topological relationships of nodes to enhance efficiency of broadcast transmissions over receiver-based multi-channel WMNs. Simulation results over NS-2 network simulator reveal that our proposed hybrid tree-based protocol achieves much higher performance than the utilization of the original receiver-based CA and TBR protocol.

We propose a novel clustering scheme considering non-uniform correlation distribution derived by experimental environment property. Firstly, we investigate the entropy property of actual environment, and then show that its spatial correlation is not uniformly distributed. Based on this result, we present the clustering strategy which provides the efficient data aggregation. Through the simulation under the non-uniform correlation distribution, we show the advantage of the proposed scheme in terms of the energy consumption property per node and the network lifetime.

Recently, wireless sensor networks have been seen as a key technology for a ubiquitous computing society. In sensor networks, many network technologies have been developed, whose main concern is reduction of power consumption of sensor nodes. Moreover, these conventional approaches assume that a node in a sensor network operate in a finite quantity and initial battery of a node. However, if we use the sensor network in the natural environment, it means that the batteries of nodes must be exchanged to long term operation. From a viewpoint of the environmental sustainability it is also necessary for sensor nodes to be easily collected and replaced. This paper proposes a routing protocol for sensor networks with high node exchangeability in order to realize the continuous long-term operations of sensor networks. In the proposed routing protocol, power consumption of nodes is partially biased and the region is rotated in order to exchange a set of nodes easily. We evaluate the proposed routing protocol comparing with DSR, and a routing protocol where all nodes try to consume the battery equally. We use evaluation metrics biased toward transmitting data, the battery residue of nodes at the exchange time, the transition of operating nodes. The results show that the difference of the battery residue between the largest and the smallest nodes is 88% and node exchangeability improves by restricting the geographical area of exchanging nodes.

The large PAPR of orthogonal frequency division multiplexing (OFDM) transmission is one of the serious problems for mobile communications that require severe power saving. Iterative clipping and filtering is an effective method for the PAPR reduction of OFDM signals. This paper evaluates PAPR reduction effect with a graded band-limiting filter in the iterative clipping and filtering method. The evaluation result by computer simulation shows that the excellent peak reduction effect can be obtained in the fewer iteration numbers by using a roll-off filter instead of the conventional rectangular filter, and the iteration number with the roll-off filter achieving the same PAPR is fewer by twice. The result confirms that the clipping and filtering method by using a graded band-limiting filter can achieve low peak OFDM transmission with less computational complexity.

A new hybrid model to calculate echoes and ultrasonic fields scattered by flaws is presented. In this model, transmitted fields by a transducer and fields scattered by flaws are calculated using the Rayleigh integral and the FDTD method, respectively. Received echo signals by the transducer are obtained by the inner product of the transmitted fields and the scattered ones at each grid for the FDTD method, and integrating those in a calculation area. Since the calculation area for the FDTD method can be limited around flaws, the calculation time is short and computer memory usage can be reduced. Experiments are done for two types of test object to confirm the validity of the hybrid model. As a result, experimental results are in good agreement with calculated ones.

This paper presents the implementation of a 31-tap FIR Hilbert transform digital filter chip used in the digital-IF receivers, to confirm the effectiveness of our new design method. Our design method that we previously reported is based on a computation sharing multiplier using a new horizontal and vertical common subexpression techniques. A 31-tap FIR Hilbert transform digital filter was implemented and fabricated in 0.35 µm CMOS standard cell library. The chip's core contains approximately 33k transistors and occupies 0.86 mm². The chip also has an operating speed of 70 MHz over. The implementation results show that the proposed Hilbert transformer has a smallest cost factor and so that is a high performance filter.

Non-malleability is an important security property of commitment schemes. The property means security against the man-in-the-middle attack, and it is defined and proved in the simulation paradigm using the corresponding simulator. Many known non-malleable commitment schemes have the common drawback that their corresponding simulators do not work in a straight-line manner, requires rewinding of the adversary. Due to this fact, such schemes are proved non-malleable only in the stand-alone cases. In the multiple-instances setting, i.e., when the scheme is performed concurrently with many instances of itself, such schemes cannot be proved non-malleable. The paper shows an efficient commitment scheme proven to be non-malleable even in the multiple-instances setting, based on the KEA1 and DDH assumptions. Our scheme has a simulator that works in a straight-line manner by using the KEA1-extractor instead of the rewinding strategy.

The ESIGN signature scheme was initially proposed in 1985. Since then, several variants have been proposed, but only a few have been formally supported using the methodology of provable security. In addition, these schemes are different from the ESIGN-PSS signature scheme submitted to ISO/IEC-14888-2 for standardization. It is believed that ESIGN-PSS is secure against the chosen-message attack, however, there has not yet been any report verifying this belief. This paper presents the security proofs of ESIGN-PSS and a variant of this scheme, denoted ESIGN-PSS-R, which is a signature scheme comprising the ESIGN signature mechanism and the PSS-R mechanism.

This paper deals with the random number generation problem under the framework of a separate coding system for correlated memoryless sources posed and investigated by Slepian and Wolf. Two correlated data sequences with length n are separately encoded to nR₁, nR₂ bit messages at each location and those are sent to the information processing center where the encoder wish to generate an approximation of the sequence of independent uniformly distributed random variables with length nR₃ from two received random messages. The admissible rate region is defined by the set of all the triples (R₁,R₂,R₃) for which the approximation error goes to zero as n tends to infinity. In this paper we examine the asymptotic behavior of the approximation error inside and outside the admissible rate region. We derive an explicit lower bound of the optimal exponent for the approximation error to vanish and show that it can be attained by the universal codes. Furthermore, we derive an explicit lower bound of the optimal exponent for the approximation error to tend to 2 as n goes to infinity outside the admissible rate region.

Error control codes have extensively been applied to semiconductor memories using high density RAM chips with wide I/O data, e.g., with 8-bit or 16-bit I/O data. Recently, spotty byte errors called s-spotty byte errors are newly defined as t or fewer bits errors in a byte having length b bits, where 1 ≤ t ≤ b. This paper proposes another type of spotty byte errors, i.e., m-spotty byte errors, where more than t bits errors in a byte may occur due to hit by high energetic particles. For these errors, this paper presents generalized m-spotty byte error control codes with minimum m-spotty distance d.

The class of complete complementary sequences (of fixed length) have the ideal correlation properties and are good at increasing the channel usage efficiency but lacks in desirable sequence lengths. In spread spectrum communication systems, sequences having nice correlation properties are important in many ways such as in suppressing multi-user interference, for reliable initial synchronization and in separation of the multipath components. It would be even good if the sequences are easy to construct and have desirable lengths for the system under consideration. In this paper, M sets of sequences that constitute a complete complementary sequences with i^th set containing N sequences of length L_i each, i = 0, 1, ..., M - 1, is defined and a general method that constructs such a class of complete complementary sequences (of different lengths) is given. The proposed class of complete complementary sequences, constituted by sequence sets of different lengths, does not increase the data rates when short-length sequences are employed.

In this paper, we derive an upper bound for the average block error probability of a standard irregular low-density parity-check (LDPC) code ensemble under the maximum-likelihood (ML) decoding. Moreover, we show that the upper bound asymptotically decreases polynomially with the code length. Furthermore, when we consider several regular LDPC code ensembles as special cases of standard irregular ones over an additive white Gaussian noise channel, we numerically show that the signal-to-noise ratio (SNR) thresholds at which the proposed bound converges to zero as the code length tends to infinity are smaller than those for a bound provided by Miller et al.. We also give an example of a standard irregular LDPC code ensemble which has a lower SNR threshold than a given regular LDPC code ensemble.

The transmit power of Ultra Wideband (UWB) is limited in short range communications to avoid the interference with existing narrow-band communication systems. Since this limits UWB communication range, this paper proposes a novel relay scheme that uses shared frequency repeaters for impulse UWB signal relay to improve system range. After considering possible problems with the repeater, in particular the coupling interference between the input and output and relay-delay, a switching control method is proposed that offers short relay-delay and suppresses the coupling interference at the repeaters. With respect to the proposed relay scheme, Pulse-Position-Modulation (PPM) UWB-based signal relay is evaluated by analyzing its BER performance using the point-to-point transmission link model.

The performance of the blind multiuser detector for a DS-CDMA system with linearly constrained constant modulus (LCCM) criterion is known to highly depend on the exact knowledge of the desired user amplitude; it is usually not available at receiver end. In this paper, we propose a novel LC adaptive CM RLS (LC-ACM-RLS) algorithm to adaptively implement the optimal solution of the LCCM receiver, and to track the desired user's amplitude, simultaneously. From computer simulations, we verify the superiority of the new proposed algorithm over the conventional LCCM-RLS algorithm for multiple access interference (MAI) suppression. Also, for time-varying channel during the adaptation processes, if the amplitude of desired user is not available and varies with time, such as hand-off and Rayleigh fading environments, we show that the proposed LC-ACM-RLS algorithm has better tracking capability compared with the conventional approaches.

Yubazaki et al. have proposed "single input rule modules connected type fuzzy reasoning method" (SIRMs method, for short) whose final output is obtained by summarizing the product of the importance degrees and the inference results from single input fuzzy rule module. Another type of single input type fuzzy reasoning method proposed by Hayashi et al. (we call it "Single Input Connected fuzzy reasoning method" (SIC method, for short) in this paper) uses rule modules to each input item as well as SIRMs method. We expect that inference results of SIRMs method and SIC method have monotonicity if the antecedent parts and consequent parts of fuzzy rules in SIRMs rule modules have monotonicity. However, this paper points out that even if fuzzy rules in SIRMs rule modules have monotonicity, the inference results do not necessarily have monotonicity. Moreover, it clarifies the conditions for the monotonicity of inference results by SIRMs method and SIC method.

In this paper, we propose a new variable-bit-rate speech coder based on the waveform interpolation concept. After the coder extracts all parameters, the amounts of distortions between the current and the predicted parameters, which are estimated by extrapolation using the past two parameters, are measured for all parameters. A parameter is not transmitted unless the distortion exceeds the preset threshold. At the decoder side, the non-transmitted parameter is reconstructed by extrapolation with the past two parameters used to synthesize signals. In this way, we can reduce 26% of the total bit rate while maintaining the speech quality degradation below the 0.1 perceptual evaluation of speech quality (PESQ) score.

This letter proposes a low-complexity estimation method of integer frequency offset in orthogonal frequency division multiplexing (OFDM) systems. The performance and complexity of the proposed method are compared with that of Morelli and Mengali's method based on maximum likelihood (ML) technique. The results show that the performance of the proposed method is comparable to that of M&M method with reduced complexity.

The traditional feedback linearization method often requires the full system parameter and state information. In this paper, we consider an asymptotic stabilization problem of a class of feedback linearizable nonlinear systems by using less than the full parameter/state information. First, our approach is to classify system parameters into two categories--'directly used parameters' and 'indirectly used parameters.' Then, a feedback linearizing controller is designed by using only the 'directly used parameters' and the observer is utilized to estimate the transformed states (diffeomorphism) which includes 'indirectly used parameters.' Thus, in our control approach, we use only a partial set of system parameters and partial state information for asymptotic stability. The useful aspects of the proposed scheme are illustrated through an example.

This paper discusses the behavior of the second-order modes (Hankel singular values) of linear continuous-time systems under typical frequency transformations, such as lowpass-lowpass, lowpass-highpass, lowpass-bandpass, and lowpass-bandstop transformations. Our main result establishes the fact that the second-order modes are invariant under any of these typical frequency transformations. This means that any transformed system that is generated from a prototype system has the same second-order modes as those of the prototype system. We achieve the derivation of this result by describing the state-space equations and the controllability/observability Gramians of transformed systems.

This letter proposes a visible watermarking scheme for halftone images. It exploits HVS filtering to transform the image in binary domain into continuous-tone domain for watermark embedding. Then a codeword search operation converts the watermarked continuous-tone image into binary domain. The scheme is flexible for two weighting factors are involved to adjust the watermark embedding strength and the average intensity of the watermarked image. Moreover, it can be used in some applications where original continuous-tone images are not available and the halftoning method is unknown.

We propose the "Two-Step Search scheme with Linear search based Second step (TSS-LS)" by improving the conventional "Two-Step Search scheme with Bit reversal search based Second step (TSS-BS)" for reliable as well as rapid acquisition of Ultra Wide Band (UWB) signals in multipath channels. The proposed TSS-LS utilizes two different thresholds and search windows to achieve fast acquisition. Furthermore, unlike the TSS-BS in which the bit reversal algorithm is applied in both steps, the linear search is adopted for the second step in the proposed TSS-LS to correctly find the starting point in the range of effective delay spread of the multipath channels, and to obtain reliable bit error rate performance of the UWB systems.

Estimating the generalization error is one of the key ingredients of supervised learning since a good generalization error estimator can be used for model selection. An unbiased generalization error estimator called the subspace information criterion (SIC) is shown to be useful for model selection, but its range of application is limited to linear learning methods. In this paper, we extend SIC to be applicable to non-linear learning.