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.

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.

A major challenge in the design of multimedia networks is to meet the quality of service (QoS) requirements of all admitted users. Regulation and scheduling are key factors for fulfilling such requirements. We propose a rate-based regulation-scheduling scheme in which the regulation function is modulated by both the tagged stream's characteristics and the state information fed-back from the scheduler. The rate-jitter and bandwidth share of each tagged connection are controlled appropriately by considering the system time and the queue length of the scheduler. Simulation results have shown that the proposed scheme works better than other rate-based disciplines.

We present a novel adaptive method to improve the binarization quality of degraded word color images. The objective of this work is to solve a nonlinear problem concerning the binarization quality, that is, to achieve edge enhancement and noise reduction in images. The digitized data used in this work were extracted automatically from real world photos. The motion of objects with reference to static camera and bad environmental conditions provoked serious quality problems on those images. Conventional methods, such as the nonlinear adaptive filter method proposed by Mo, or Otsu's method cannot produce satisfactory binarization results for those types of degraded images. Among other problems, we note mainly that contrast (between shapes and backgrounds) varies greatly within every degraded image due to non-uniform illumination. The proposed method is based on the automatic extraction of background information, such as luminance distribution to adaptively control the intensity levels, that is, without the need for any manual fine-tuning of parameters. Consequently, the new method can avoid noise or inappropriate shapes in the output binary images. Otsu's method is also applied to automatic threshold selection for classifying the pixels into background and shape pixels. To demonstrate the efficiency and the feasibility of the new adaptive method, we present results obtained by the binarization system. The results were satisfactory as we expected, and we have concluded that they can be used successfully as data in further processing such as segmentation or extraction of characters. Furthermore, the method helps to increase the eventual efficiency of a recognition system for poor-quality word images, such as number plate photos with non-uniform illumination and low contrast.

This paper proposes a new steganographic method utilizing features of JPEG compression. The method embeds secret information using the number of zeroes in a block of quantized DCT coefficients in minimum coding units (MCU) of JPEG images. In the method, we can embed secret information into JPEG images with degradation like that by JPEG compression. Furthermore, the method causes little change of the histogram of quantized DCT coefficients, so it is hard to perceive secret information embedded by the method. The method mainly modifies boundaries between zero and non-zero DCT coefficients, so we can use the low frequency side of DCT coefficients for another steganographic method.

In order to benefit from the advantages of soft handoff (SHO), it is important that the SHO parameters (the SHO thresholds; T_ADD and T_DROP are well assigned. T_ADD is the threshold used for triggering a pilot with high strength to be added to the Active Set (AS) list. The AS means the pilots associated with the forward traffic channels assigned to mobile station. In contrast, T_DROP is the threshold used for triggering a pilot with low strength to be dropped from the AS list. This paper analyzes the effects of varying SHO thresholds in a cellular code division multiple access (CDMA) system on the blocking probability based on traffic load and geometrical distances in hexagonal layout of base stations (BSs). In addition, the previously proposed traffic load equation is applied to the proposed SHO model for balancing the numbers of new and handoff calls on the forward link capacity in case of uniform traffic load. The results show that the blocking probability is more sensitive to T_DROP than to T_ADD variations.

The paper discusses the spatio-temporal phenomena in autonomous two-layer Cellular Neural Networks (CNNs) with mutually coupled templates between two layers. By computer calculations, we show how pattern formations, autowaves and classical waves can be regenerated in the networks, and describe the properties of these phenomena in detail. In particular, we focus our discussion on the necessary conditions for generating these spatio-temporal phenomena. In addition, the influences of the template parameters and initial state conditions of CNNs on the spatio-temporal phenomena are investigated.

The issue of scalable Differentiated Services (DiffServ) admission control now is still an open research problem. We propose a new admission control model that can not only provide coarse grain Quality of Services (QoS), but also guarantee end-to-end QoS for assured service without per-flow state management at core routers within DiffServ domain. Associated with flow aggregation model, a hybrid signaling protocol is proposed to select the route satisfying the end-to-end QoS requirements. Simulation result shows that the proposed model can accurately manage resource, leading to much better performance when compared to other schemes.

This paper presents QoS control enhanced architecture for VoIP networks. In this architecture we use both the probe flow delay and average loss rate measurement systems. First we apply the probability-based EMBAC scheme on our delay system. Then we propose a new probability-based EMBAC with a severe congestion consideration scheme to improve the admission control scheme in both measurement systems. We compare the performance of the enhanced systems in terms of blocking probability under the same condition of achieving average packet loss rate no greater than the certain target by setting an appropriate admission threshold in each system under each scenario. In this study, it is shown through simulations that for the same target voice average loss rate, the enhanced systems proposed in this paper outperform the conventional schemes in handling the network resources. Then we will seek to prove that, for extra traffic loads within a busy period of time and with an optimal admission threshold chosen in advance, the enhanced systems can be a powerful and reliable EMBAC tool for VoIP networks in achieving high network performance with minimum blocking probability and minimum average loss rates. Finally it is shown that the enhanced systems have reasonable scalability.

Cohesion is an important software attribute, and it is one of significant criteria for assessing object-oriented software quality. Although several metrics for measuring cohesion have been proposed, there is an aspect which has not been supported by those existing metrics, that is "cohesive-part size." This paper proposes a new metric focusing on "cohesive-part size," and evaluates it in both of qualitative and quantitative ways, with a mathematical framework and an experiment measuring some Java classes, respectively. Through those evaluations, the proposed metric is showed to be a reasonable metric, and not redundant one. It can collaborate with other existing metrics in measuring class cohesion, and will contribute to more accurate measurement.

A VLSI-specific wavelet processing technique has been developed and implemented as a processor in accordance with the JPEG2000 specification. This proposed procedure of discrete wavelet transforms uses an altered calculation equations and makes use of intermediate results through wavelet calculation. The implementation of the proposed procedure is capable of realizing a highly efficient DWT for large size images in spite of using low hardware costs and a small size buffering memory. In order to obtain fast EBCOT processing, three types of parallel processing are introduced in the EBCOT architecture. The processor performs compression of 720480 pixels images with the speed of 30 frames per second (fps) at a required operating frequency as low as 32 MHz or lower. Furthermore, it need not divide an image into tiles so that the problem of deterioration of image quality due to tile division does not occur. A prototype of this processor has been fabricated in a 0.25-µm 5-layer CMOS process. The chip is 10.210.4 mm2 in size and consumes 2.0 W when supplied with 2.5 V and 32 MHz.

Software agents are knowledgeable, autonomous, situated and interactive software entities. Agents' interactions are of special importance when a group of agents interact with each other to solve a problem that is beyond the capability and knowledge of each individual. Efficiency, performance and overall quality of the multi-agent applications depend mainly on how the agents interact with each other effectively. In this paper, we suggest an agent model by which we can clearly distinguish different agent's interaction scenarios. The model has five attributes: goal, control, interface, identity and knowledge base. Using the model, we analyze and describe possible scenarios; devise the appropriate reasoning and decision making techniques for each scenario; and build a library of reasoning and decision making modules that can be used readily in the design and implementation of multiagent systems.

In this paper, we propose an extension to the image transport protocol (ITP). When images are transmitted through the Internet, TCP is generally used because it ensures the reliable transmission. However, interactivity will largely affected because of its acknowledgement scheme. This becomes remarkable in the network where packet-loss rate is relatively higher like wireless LANs. For more efficient image transmission, ITP was proposed. Like UDP, in ITP transmission, packets can be transmitted without acknowledgement of the reception. This contributes to improve the interactivity, on the other hand, some of packets may lost during transmission. ITP has a mechanism that the receiver-side can control the retransmission of the lost packets to maintain the quality of the received image. However, it is a hard task for the receiver to select which packets to be retransmitted. In this paper, we propose an extension to ITP by which the server can mark the importance of each packet. This helps the receivers to select important packets for requesting retransmission for server.

This paper presents a scalable and efficient quota-based admission control scheme for the 3rd generation (3G) operator's IP backbone network, where quota denotes a chunk of bandwidth. This research is motivated by the 3G operator's need for guaranteeing end-to-end IP QoS of mobile-to-mobile and mobile-to-server multimedia sessions. In the proposed scheme, the quota size of a path implies the proper amount of allocated and released resources on the path condition. Employing the quota size makes the job of allocating or releasing resources at nodes in a path simple so that it becomes scalable. Moreover, with this simple scheme, an edge node can be allowed not only to initiate the allocation/release request but also to perform admission control function. To maximize the efficiency, the path quota size varies depending on the bottleneck link condition in the path. In high offered load, the proposed scheme decreases the path quota size and retains higher utilization while it requires lower signaling cost than the fixed scheme using a fixed size aggregation. As the load lessens, it increases the path quota size and reduces the signaling cost significantly.

This paper investigates modified random timers based on uniform and exponentially distributed timers for feedback scalability for large groups. We observe the widely-used probability distribution functions and propose new ones that are aware of network delays. The awareness of network delays of our proposed modified p.d.fs proves to be able to achieve lower expected number of messages compared to the original ones given that the parameters are optimized for the network variables: the number of receivers, and the network delay. In our analysis we derive an equation to estimate the optimized parameter based on these network variables. We also simulate the p.d.fs for heterogenous network delays and find that each receiver only needs to be aware of its network delay.

The bi-level digital video, because of its simplicity and compactness, can be utilized to provide for a quick and faithful preview of its original content. The proposed bi-level digital video compression technique exploits the context-based probabilistic estimation model towards adaptive pixel prediction which can be used towards generating residual image frames which may then be Run-Length-Rice coded. Towards promoting error-resiliency and random-access, each bi-level digital video frame may be typed into either intra- or inter- picture format. The proposed technique can be seen, in comparison to existing JBIG compression technologies in simulation runs, to provide added temporal redundancy removal.

This paper describes a method of test data compression for a given test set using statistical encoding. In order to maximize the effectiveness of statistical encoding, the method first converts some specified input values in the test set to unspecified ones without losing fault coverage, and then reassigns appropriate logic values to the unspecified inputs. Experimental results for ISCAS-89 benchmark circuits show that the proposed method can on the average reduce the test data volume to less than 25% of that required for the original test set.

Interval arithmetic is able to be applied when we include the ranges of various functions. When we include them applying the interval arithmetic, the serious problem that the widths of the range inclusions increase extremely exists. In range inclusion of polynomials particularly, Horner's method and Alefeld's method are well known as the conventional methods which mitigates this problem. The purpose of this paper is to propose the new methods which are able to mitigate this problem more efficiently than the conventional methods. And in this paper, we show and compare the efficiencies of the new methods by some numerical examples.

We propose a new orthogonal frequency division multiplexing transmission scheme using orthogonal code multiplexing. This scheme makes all modulation symbols have the same reliability even in a frequency selective fading channel, through a distributed transmission of each symbol over the whole effective subcarriers using a distinct orthogonal code. As an appropriate set of orthogonal multiplexing codes, we use the set of discrete Fourier transform code sequences that hold the orthogonality irrespective of the length. Using this set, we also can greatly reduce the peak-to-average-power ratio (PAR) of the resulting signal. Simulation results show that the proposed scheme can significantly reduce the required signal-to-noise ratio at a given bit error rate over the existing schemes. The scheme can maintain the PAR within a reasonable range of about 6 dB even up to 512 subcarriers and works well even with PAR clipping of 1.5 dB.

We propose Optimal Temporal Decomposition (OTD) of speech for voice morphing preserving Δ cepstrum. OTD is an optimal modification of the original Temporal Decomposition (TD) by B. Atal. It is theoretically shown that OTD can achieve minimal spectral distortion for the TD-based approximation of time-varying LPC parameters. Moreover, by applying OTD to preserving Δ cepstrum, it is also theoretically shown that Δ cepstrum of a target speaker can be reflected to that of a source speaker. In frequency domain interpolation, the Laplacian Spectral Distortion (LSD) measure is introduced to improve the Inverse Function of Integrated Spectrum (IFIS) based non-uniform frequency warping. Experimental results indicate that Δ cepstrum of the OTD-based morphing spectra of a source speaker is mostly equal to that of a target speaker except for a piecewise constant factor and subjective listening tests show that the speech intelligibility of the proposed morphing method is superior to the conventional method.