Modern video compression usually consists of ME/MC (Motion Estimation/Motion Compensation), transform, and quantization of the transform coefficients. Efficient bit allocation technique to distribute available bits to motion parameters and quantized coefficients is an important part of the whole system. A method that is very complex and/or needs buffering of many future frames is not suitable for real time application. We develop an efficient bit allocation technique that utilizes the estimated effect of allocated bits to motion parameter and quantization on the overall quality. We also propose an hierarchical block based ME/MC technique that requires less computations than classical BMA (Block Matching Algorithm) while offering better motion estimation.

Multi-rate capabilities are supported by the physical layers of most 802.11 devices. To enhance the network throughput of MANETs, transfer rate adaptation schemes at MAC layer should employ the multi-rate capability at physical and the information of previous transmissions provided by MAC and physical layers. In this paper, we propose a transfer rate adaptation scheme plus back-to-back frame transmissions, and fragmentation at MAC layer, named TRAF. TRAF adopts a bi-direction-based approach with an extended option to select an appropriate rate for frame transmission under fast changing channel conditions. Consecutive back-to-back frame transmissions to fully utilize good channel quality during a coherent time interval and fragmentation algorithm to maintain high throughput under worse channel conditions are recommended in TRAF. Extensive simulation is experimented to evaluate the performance of TRAF. Regarding simulation results, frame delivery ratio, network throughput, and fairness of TRAF are significantly improved by comparing to that of fix rate, ARF, RBAR, OAR, and AAR protocols.

This paper considers Quasi-Reduced ordered Multi-valued Decision Diagrams with k bits (QRMDD(k)s) to represent binary logic functions. Experimental results show relations between the values of k and the numbers of nodes, the memory sizes, the numbers of memory accesses, and area-time complexity for QRMDD(k). For many benchmark functions, the numbers of nodes and memory accesses for QRMDD(k)s are nearly equal to of the corresponding Quasi-Reduced ordered Binary Decision Diagrams (QRBDDs), and the memory sizes and the area-time complexities for QRMDD(k)s are minimum when k = 2 and k = 3-6, respectively.

In communication systems such as mobile telecommunication systems and the Internet, resource sharing among coexisting real-time and non-real-time services is extremely important to provide multimedia services. This paper analytically investigates the performance of the packet data control algorithm proposed in. This algorithm efficiently uses radio resources by utilizing the remaining capacity that is not used by real-time services. The state probability vectors and transition probability matrices of both the real-time and non-real-time services are first derived and then the delay characteristics, the outage probability of voice users, and the outage probability of data users are evaluated. A performance analysis with high bit-rate non-real-time services is also presented.

A 2-dimensional cylindrical k-within-consecutive-(r, s)-out-of-(m, n):F system consists of m n components arranged on a cylindrical grid. Each of m circles has n components, and this system fails if and only if there exists a grid of size r s within which at least k components are failed. This system may be used into reliability models of "Feelers for measuring temperature on reaction chamber," "TFT Liquid Crystal Display system with 360 degree wide area" and others. In this paper, first, we propose an efficient algorithm for the reliability of a 2-dimensional cylindrical k-within-consecutive-(r, s)-out-of-(m, n):F system. The feature of this algorithm is calculating their system reliabilities with shorter computing time and smaller memory size than Akiba and Yamamoto. Next, we show some numerical examples so that our proposed algorithm is more effective than Akiba and Yamamoto for systems with large n.

In this paper, we derive expressions for the bit error probability of QPSK/OFDM on frequency-selective Rayleigh fading channels. In the OFDM system, ICI (interchannel interference) caused by Doppler spread of the channel degrades the error performance of the system and introduces the error floor even for coherent detection. Analysis results show that the error performance of QPSK/OFDM can be degraded as the normalized maximum Doppler frequency fD /Bsub is increased where fD is the maximum Doppler frequency and Bsub is the subchannel bandwidth. Computer simulations confirm the theoretical analysis results for BPSK and QPSK signals.

We propose a burst optical packet generator based on a novel photonic parallel-to-serial conversion scheme, and demonstrate 40-Gbit/s 16-bit optical packet generation from 16-ch parallel low-voltage TTL data streams. It consists of electrical 4:1 parallel-to-serial converters that employ InP metal-semiconductor-metal photodetectors, and an optical time-domain multiplexer with electroabsorption modulators. The proposed optical packet generator is suitable for burst optical packet generation and overcomes the electronic bandwidth limitation, which is prerequisite for achieving high-speed photonic packet switched networks. In addition, it can be driven by simple low-cost low-power CMOS logic circuits, and is compact and extensible in terms of the number of input channels due to the effective combination of electrical and optical multiplexing.

In this paper, general definitions of collusion secure codes are shown. Previously defined codes such as frameproof code, secure frameproof code, identifiable parent property code, totally c-secure code, traceability code, and (c,g/s)-secure code are redefined under various marking assumptions which are suitable for most of the fingerprinting systems. Then, new relationships among the combined notions of codes and the marking assumptions are revealed. Some (non)existence results are also shown.

This paper is focused on the measurement of the complex permittivity of a liquid phantom by the transmission line method using a coaxial line for measuring high-permittivity and high-loss materials. First, the complex permittivity of the liquid phantom material is measured under various physical lengths of the coaxial line for accurate measurement. Secondly, comparison between the measured result and the result obtained by the coaxial probe method is carried out in the frequency range from 0.5 to 3 GHz. Finally, the measurement error included in the complex permittivity is estimated quantitatively. The discussions lead to the conclusion that accurate measurement of the liquid material with high-permittivity and high-loss is possible by the presented method.

Service differentiation has been a subject of research for the past few years in the IETF; and in the current Internet, IP flows are mostly treated in a best-effort approach. However, for next generation networks it is expected that users would like to obtain service differentiation based on their preferences or profiles as well as the different types of multimedia they opt to receive or send. In addition, current Quality of Service (QoS) provisioning architectures have been designed mostly for the fixed networks without taking into consideration the wireless or radio links special requirements, such as low bandwidth availability, error prone communications, etc. In this paper we propose a QoS provisioning architecture for next generation networks that uses a hybrid approach to deal with both the wireless and wired (fixed) part of the network. For administering the scarce resource of the radio environment, we have developed a resource allocation algorithm based on micro-economic principles that uses associated piecewise linear utility functions which describe the benefit a user receives from the allocation of various amounts of resource. For the wired part of the network we have also developed a Core-Stateless Utility based Rate allocation Framework (SURF) for performing traffic policing where the flow's requirements are expressed using utility functions. The core routers maintain no per-flow state and implement a simple packet level admission control algorithm that is based on a threshold utility value that is computed dynamically. To tie in these two mechanisms, we developed a signaling mechanism that collect network statistics when a user starts a call and a QoS administrator entity (or Broker) perform the computations for allocating resources based on the information of available resources in the fixed and the wireless sections of the network. A comparison between the hybrid approach and the SURF approach to show the performance of the proposed architecture is presented later in the paper.

This paper describes an adaptive fingerprint-sensing scheme for a user authentication system with a fingerprint sensor LSI to obtain high-quality fingerprint images suitable for identification. The scheme is based on novel evaluation indexes of fingerprint-image quality and adjustable analog-to-digital (A/D) conversion. The scheme adjusts dynamically an A/D conversion range of the fingerprint sensor LSI while evaluating the image quality during real-time fingerprint-sensing operation. The evaluation indexes pertain to the contrast and the ridgelines of a fingerprint image. The A/D conversion range is adjusted by changing quantization resolution and offset. We developed a fingerprint sensor LSI and a user authentication system to evaluate the adaptive fingerprint-sensing scheme. The scheme obtained a fingerprint image suitable for identification and the system achieved an accurate identification rate with 0.36% of the false rejection rate (FRR) at 0.075% of the false acceptance rate (FAR). This confirms that the scheme is very effective in achieving accurate identification.

NMAC is a function for message authentication based on cryptographic hash functions such as SHA. It is shown to be a secure message authentication code if its compression function with fixed input length is a secure message authentication code and its iterated hash function with variable input length constructed with the compression function is weakly collision resistant. In this article, two results are shown on the strength of the weak collision resistance of the iterated hash function in NMAC. First, it is shown that the weak collision resistance of the iterated hash function in NMAC is not implied by the pseudorandomness of its compression function even if the MD-strengthening is assumed. Second, the weak collision resistance of the iterated hash function in NMAC implies the collision resistance of its compression function if the compression function is pseudorandom.

The high-end telecom terminal and PDAs, sometimes called Personal Trusted Devices (PTDs) are programmable, have tens of megabytes memory, and rather fast processors. In this paper we analyze, when it is energy-efficient to transfer application data compressed over the downlink and then decompress it at the terminal, or compress it first at the terminal and then send it compressed over up-link. These questions are meaningful in the context of usual application code or data and streams that are stored before presentation and require lossless compression methods to be used. We deduce an analytical model and assess the model parameters based on experiments in 2G (GSM) and 3G (FOMA) network. The results indicate that if the reduction through compression in size of the file to be downloaded is higher than ten per cent, energy is saved as compared to receiving the file uncompressed. For the upload case even two percent reduction in size is enough for energy savings at the terminal with the current transmission speeds and observed energy parameters. If time is saved using compressed files during transmission, then energy is certainly saved. From energy savings at the terminal we cannot deduce time savings, however. Energy and time consumed at the server for compression/decompression is considered negligible in this context and ignored. The same holds for the base stations and other fixed telecom infrastructure components.

Quality of Service (QoS) provisioning is a new but challenging research area in the field of Mobile Ad hoc Network (MANET) to support multimedia data communication. However, the existing QoS routing protocols in ad hoc network did not consider a major aspect of wireless environment, i.e., mutual interference. Interference between nodes belonging to two or more routes within the proximity of one another causes Route Coupling. This can be avoided by using zone-disjoint routes. Two routes are said to be zone disjoint if data communication over one path does not interfere with the data communication along the other path. In this paper, we have proposed a scheme for supporting priority-based QoS in MANET by classifying the traffic flows in the network into different priority classes and giving different treatment to the flows belonging to different classes during routing so that the high priority flows will achieve best possible throughput. Our objective is to reduce the effect of coupling between routes used by high and low priority traffic by reserving zone of communication. The part of the network, used for high priority data communication, i.e, high priority zone, will be avoided by low priority data through the selection of a different route that is maximally zone-disjoint with respect to high priority zones and which consequently allows contention-free transmission of high priority traffic. The suggested protocol in our paper selects shortest path for high priority traffic and diverse routes for low priority traffic that will minimally interfere with high priority flows, thus reducing the effect of coupling between high and low priority routes. This adaptive, priority-based routing protocol is implemented on Qualnet Simulator using directional antenna to prove the effectiveness of our proposal. The use of directional antenna in our protocol largely reduces the probability of radio interference between communicating hosts compared to omni-directional antenna and improves the overall utilization of the wireless medium in the context of ad hoc wireless network through Space Division Multiple Access (SDMA).

To examine the computational complexity of cryptographic primitives such as the discrete logarithm problem, the factoring problem and the Diffie-Hellman problem, we define a new problem called square-root exponent, which is a problem to compute a value whose discrete logarithm is a square root of the discrete logarithm of a given value. We analyze reduction between the discrete logarithm problem modulo a prime and the factoring problem through the square-root exponent. We also examine reductions among the computational version and the decisional version of the square-root exponent and the Diffie-Hellman problem and show that the gap between the computational square-root exponent and the decisional square-root exponent partially overlaps with the gap between the computational Diffie-Hellman and the decisional Diffie-Hellman under some condition.

We present a method for recognition of continuous Korean Sign Language (KSL). In the paper, we consider the segmentation problem of a continuous hand motion pattern in KSL. For this, we first extract sign sentences by removing linking gestures between sign sentences. We use a gesture tension model and fuzzy partitioning. Then, each sign sentence is disassembled into a set of elementary motions (EMs) according to its geometric pattern. The hidden Markov model is adopted to classify the segmented individual EMs.

We present a speaker adaptation method that makes it possible to determine articulatory parameters from an unknown speaker's speech spectrum using an HMM (Hidden Markov Model)-based speech production model. The model consists of HMMs of articulatory parameters for each phoneme and an articulatory-to-acoustic mapping that transforms the articulatory parameters into a speech spectrum for each HMM state. The model is statistically constructed by using actual articulatory-acoustic data. In the adaptation method, geometrical differences in the vocal tract as well as the articulatory behavior in the reference model are statistically adjusted to an unknown speaker. First, the articulatory parameters are estimated from an unknown speaker's speech spectrum using the reference model. Secondly, the articulatory-to-acoustic mapping is adjusted by maximizing the output probability of the acoustic parameters for the estimated articulatory parameters of the unknown speaker. With the adaptation method, the RMS error between the estimated articulatory parameters and the observed ones is 1.65 mm. The improvement rate over the speaker independent model is 56.1 %.

Efficient representations of a 3-D object shape and its texture data have attracted wide attention for the transmission of computer graphics data and for the development of multi-view real image rendering systems on computer networks. Polygonal mesh data, which consist of connectivity information, geometry data, and texture data, are often used for representing 3-D objects in many applications. This paper presents a wavelet coding technique for coding the geometry data structured on a triangular lattice plane obtained by structuring the connectivity of the polygonal mesh data. Since the structured geometry data have an arbitrarily-shaped support on the triangular lattice plane, a shape-adaptive wavelet transform was used to obtain the wavelet coefficients, whose number is identical to the number of original data, while preserving the self-similarity of the wavelet coefficients across subbands. In addition, the wavelet coding technique includes extensions of the zerotree entropy (ZTE) coding for taking into account the rate-distortion properties of the structured geometry data. The parent-children dependencies are defined as the set of wavelet coefficients from different bands that represent the same spatial region in the triangular lattice plane, and the wavelet coefficients in the spatial tree are optimally pruned based on the rate-distortion properties of the geometry data. Experiments in which proposed wavelet coding was applied to some sets of polygonal mesh data showed that the proposed wavelet coding achieved better coding efficiency than the Topologically Assisted Geometry Compression scheme adopted in the MPEG-4 standard.

This paper proposes two alternative approaches that do not make use of a dictionary but instead utilizes different types of learned features to segment words in a language that has no explicit word boundary. Both methods utilize decision trees as knowledge representation acquired from a training corpus in the segmentation process. The first method, a language-dependent technique, applies a set of constructed features patterns based on character types to generate a set of heuristic segmentation rules. It separates a running text into a sequence of small chunks based on the given patterns, and constructs a decision tree for word segmentation. The second method extracts statistics of character sequences from a training corpus and uses them as features for the process of constructing a set of rules by decision tree induction. The latter needs no linguistic knowledge. By experiments on Thai language, both methods achieve relatively high accuracy but the latter performs much better.

In this paper, an adaptive transmission scheme considering the stay time in a spot mobile access system is proposed. The proposed adaptive transmission scheme selects the modulation format according to the user's stay time in the spot communication zone and the types of data requested by each user. In the proposed system, when the stay time of a user is short, high-speed modulation is selected for this user. When the stay time of a user is long, a more reliable modulation format is selected. The computer simulation results show that the proposed transmission scheme without any channel estimation can achieve the same or better performance than when using the modulation format fixedly when the carrier-to-noise ratio changes rapidly.