Commercial markets employ digital right management (DRM) systems to protect valuable high-definition (HD) quality videos. DRM system uses watermarking to provide copyright protection and ownership authentication of multimedia contents. We propose a real-time video watermarking scheme for HD video in the uncompressed domain. Especially, our approach is in aspect of practical perspectives to satisfy perceptual quality, real-time processing, and robustness requirements. We simplify and optimize human visual system mask for real-time performance and also apply dithering technique for invisibility. Extensive experiments are performed to prove that the proposed scheme satisfies the invisibility, real-time processing, and robustness requirements against video processing attacks. We concentrate upon video processing attacks that commonly occur in HD quality videos to display on portable devices. These attacks include not only scaling and low bit-rate encoding, but also malicious attacks such as format conversion and frame rate change.

Intrusion detection system (IDS) has played a central role as an appliance to effectively defend our crucial computer systems or networks against attackers on the Internet. The most widely deployed and commercially available methods for intrusion detection employ signature-based detection. However, they cannot detect unknown intrusions intrinsically which are not matched to the signatures, and their methods consume huge amounts of cost and time to acquire the signatures. In order to cope with the problems, many researchers have proposed various kinds of methods that are based on unsupervised learning techniques. Although they enable one to construct intrusion detection model with low cost and effort, and have capability to detect unforeseen attacks, they still have mainly two problems in intrusion detection: a low detection rate and a high false positive rate. In this paper, we present a new clustering method to improve the detection rate while maintaining a low false positive rate. We evaluated our method using KDD Cup 1999 data set. Evaluation results show that superiority of our approach to other existing algorithms reported in the literature.

As the gate area decreases to the order of a square micron, individual trapping events can be detected as fluctuations between discrete levels of the drain current, known as random telegraph signal (RTS) noise. Many circuit application areas such as CMOS Image sensor and flash memory are already suffering from RTS noise. Especially, in case of flash memory, FN stress causes threshold voltage shift problems due to generation of additional oxide traps, which degrades circuit performance. In this paper, we investigated how FN stress effects on RTS noise behavior in MOSFET and monitored it in both the time domain and frequency domain.

A scalable DC model of lateral double diffused MOSFETs (LDMOSFETs) is presented in this paper. This model is based on physical analysis considering device geometry, carrier distributions, mobility degradation effect, and the effect of impact ionization. In this model, we divide the LDMOSFET into two regions to obtain the physical conduction model: one is channel region and the other is drift region. The channel region model is based on the BSIM3v3 model and the drift region employs voltage dependent resistance model considering the length of depleted region in the drift region. The modeling results are compared with measured I-V characteristics and the results show good agreements with the maximum error of 10% compared to the measured results of devices.

Cognitive Radio is an advanced enabling technology for efficient utilization of vacant spectrum due to its ability to sense the spectrum environment. Various detection methods have been proposed for spectrum sensing, which is the key function in implementing cognitive radio. However most of the existing methods put their interests in detecting TV signal and wireless microphone signals. In this paper, we explore the periodicity of the equally spaced pilot subcarriers in OFDM signal. Simulations in various fading environments show that the proposed cyclostationarity based detection method works well for OFDM signal.

This paper presents a coexistence model of IEEE 802.15.4 with IEEE 802.11b interference in fading channels and proposes two adaptive channel allocation schemes. The first avoids the IEEE 802.15.4 interference only and the second avoids both of the IEEE 802.15.4 and IEEE 802.11b interferences. Numerical results show that the proposed algorithms are effective for avoiding interferences and for maximizing network capacity since they select a channel which gives the maximum signal to noise ratio to the system.

This paper considers the problem of contention resolution algorithm for multi-class with quality of service (QoS) constrained for wireless communication. Five different channel reservation schemes are proposed, namely, UNI+MLA, UNI+DS, UNI+DS+MLA, Partial UNI and Partial UNI+MLA schemes for multimedia traffic, all are extensions of our recently proposed UNI scheme for single-class traffic. The goal is to achieve the highest system performance and enable each traffic type to meet its QoS requirements. We evaluate the performance of each scheme by mathematical analysis. The numerical results show that our proposed schemes are effective in enabling each traffic type to achieve the best successful rate possible in this kind of environment. Finally when comparing between our proposed schemes and conventional technique in terms of both throughput performance and QoS requirements it is found that the UNI+MLA, UNI+DS+MLA and Partial UNI+MLA schemes are relatively efficient and suitable for practical applications.

In heterogeneous network environments, it is mandatory to measure the grade of the video quality in order to guarantee the optimal quality of the video streaming service. Quality of Service (QoS) has become a key issue for service acceptability and user satisfaction. Although there have been many recent works regarding video quality, most of them have been limited to measuring quality within temporal and Signal-to-Noise Ratio (SNR) scalability. H.264/AVC Scalable Video Coding (SVC) has emerged and has been developed to support full scalability. This includes spatial, temporal, and SNR scalability, each of which shows different visual effects. The aim of this paper is to define and develop a novel video quality metric allowing full scalability. It focuses on the effect of frame rate, SNR, the change of spatial resolution, and motion characteristics using subjective quality assessment. Experimental results show the proposed quality metric has a high correlation to subjective quality and that it is useful in determining the video quality of SVC.

In this paper, we design a new coded cooperation protocol utilizing superposition modulation together with iterative decoding/detection algorithms. The aim of the proposed system is to apply "dirty paper coding" theory in the context of half-duplex relay systems. In the proposed system, the node transmits a superposed signal which consists of its own coded information and other node's re-coded information. The destination node detects and decodes the signal using the received signals at two continuous time-slots with iterative decoding algorithm. Moreover, the destination node detects the received signal using the results of decoding, iteratively. This paper provides the outage probability of the proposed system under the assumption that the proposed system can ideally perform dirty paper coding, and it is shown from the comparison between outage probabilities and simulated results that the proposed system can get close to the dirty paper coding theory.

This paper proposes a closed form solution to L2-sensitivity minimization of second-order state-space digital filters. Restricting ourselves to the second-order case of state-space digital filters, we can express the L2-sensitivity by a simple linear combination of exponential functions and formulate the L2-sensitivity minimization problem by a simple polynomial equation. As a result, the L2-sensitivity minimization problem can be converted into a problem to find the solution to a fourth-degree polynomial equation of constant coefficients, which can be algebraically solved in closed form without iterative calculations.

Today, TLS is widely used for achieving a secure communication system. And TLS is used PKI for server authentication and/or client authentication. However, its PKI environment, which is called as "multiple trust anchors environment," causes the problem that the verifier has to maintain huge number of CA certificates in the ubiquitous network because the increase of terminals connected to the network brings the increase of CAs. However, most of terminals in the ubiquitous network will not have enough memory to hold such huge number of CA certificates. Therefore, another PKI environment, "cross certification environment", is useful for the ubiquitous network. But, because current TLS is designed for the multiple trust anchors model, TLS cannot work efficiently on the cross-certification model. This paper proposes a TLS implementation method to support the cross certification model efficiently. Our proposal reduces the size of exchanged messages between the TLS client and the TLS server during the handshake process. Therefore, our proposal is suitable for implementing TLS in the terminals that do not have enough computing power and memory in ubiquitous network.

A new method of multi-bit embedding based on a protocol of secure asymmetric digital watermarking detection is proposed. Secure watermark detection has been achieved by means of allowing watermark verifier to detect a message without any secret information exposed in extraction process. Our methodology is based on an asymmetric property of a watermark algorithm which hybridizes a statistical watermark algorithm and a public-key algorithm. In 2004, Furukawa proposed a secure watermark detection scheme using patchwork watermarking and Paillier encryption, but the feasibility had not tested in his work. We have examined it and have shown that it has a drawback in heavy overhead in processing time. We overcome the issue by replacing the cryptosystem with the modified El Gamal encryption and improve performance in processing time. We have developed software implementation for both methods and have measured effective performance. The obtained result shows that the performance of our method is better than Frukawa's method under most of practical conditions. In our method, multiple bits can be embedded by assigning distinct generators in each bit, while the embedding algorithm of Frukawa's method assumes a single-bit message. This strongly enhances capability of multi-bit information embedding, and also improves communication and computation cost.

In this letter, we propose a shared adaptive packet loss concealment scheme for the high quality guaranteed Internet telephony service which connects multiple users. In order to recover packet loss efficiently in the all-IP based convergence environment, we provide a robust signal recovery scheme which is based on the shared adaptive both-side information utilization. This scheme is provided according to the average magnitude variation across the frames and the pitch period replication on the 1-port gateway (G/W) system. The simulated performance demonstrates that the proposed scheme has the advantages of low processing times and high recovery rates in the all-IP based ubiquitous environment.

This paper presents a new quarter-wavelength microstrip coupler compensated with a periodic sequence of floating metallic strips in the slots on the inner edges. After describing the characteristics of the coupled-line, as an example, a 15-dB coupler is designed and a high directivity of 30 dB or more in theory is obtained over a full band of a single-section coupler. Next, couplers with various coupling factors are designed, and the usefulness for very loose coupling is demonstrated. Furthermore, a three-section coupler is designed to show the effectiveness in a wide frequency range. The validity of the design concept and procedure is confirmed by electromagnetic simulations and experiments.

We propose an Adjustable Parallel TCP (AP-TCP) which is a new scheme to control the aggregate throughput of parallel TCP flows. The AP-TCP can adjust the aggregate throughput to be any desired level irrespective of the parallel size (the number of parallel TCP flows). To adjust the aggregate throughput, we modify the increment factor of each parallel TCP flow to K2/N2 where N is the number of parallel TCP flows and K is a value equivalent to any desired level for the aggregate throughput. Once K is given, the AP-TCP attempts to have K times more bandwidth than a single TCP flow when they are competing on the same network path. Another feature of the AP-TCP is its self-adjustment scheme. There is no central coordination or control overhead for parallel TCP flows. We analyze the model of the AP-TCP theoretically and evaluate it by using NS-2 simulation.

In virtual cellular network (VCN), proposed for high-speed mobile communications, the signal transmitted from a mobile terminal is received by some wireless ports distributed in each virtual cell and relayed to the central port that acts as a gateway to the core network. In this paper, we apply the multi-route MHMRC diversity in order to decrease the transmit power and increase the multi-hop link capacity. The transmit power, the interference power and the link capacity are evaluated for DS-CDMA multi-hop VCN by computer simulation. The multi-route MHMRC diversity can be applied to not only DS-CDMA but also other access schemes (i.e. MC-CDMA, OFDM, etc.).

Vertical handoff is a new type of handoff that is triggered when a mobile node moves over heterogeneous wireless networks with each proving different access bandwidth, transmission latency, and coverage. A mobile node can achieve higher throughput by accessing a higher bandwidth providing wireless network. However, TCP has to experience drastic changes of the bandwidth and the latency due to the vertical handoff which must be recognized as a network congestion, and this degrades end-to-end performance. In this paper, we propose a TCP context switching scheme, named Context-Switching TCP, that maintains TCP variables separately for different types of wireless networks. Through simulations, Context-Switching TCP shows higher performance than TCP SACK for vertical handoff. Especially, it shows much higher performance gain when vertical handoff occurs frequently.

Previously proposed batch signature schemes do not allow a signer to generate a signature immediately for sequentially asked signing queries. In this letter, we propose flexible batch signatures which do not need any waiting period and have very light computational overhead. Therefore our schemes are well suited for low power devices.

We present a method for the objective quality evaluation of noise-reduced speech in wideband speech communication services, which utilize speech with a wider bandwidth (e.g., 7 kHz) than the usual telephone bandwidth. Experiments indicate that the amount of residual noise and the distortion of speech and noise, which are quality factors, influence the perceived quality degradation of noise-reduced speech. From the results, we observe the principal relationships between these quality factors and perceived speech quality. On the basis of these relationships, we propose a method that quantifies each quality factor in noise-reduced speech by analyzing signals that can be measured and assesses the overall perceived quality of noise-reduced speech using values of these quality factors. To verify the validity of the method, we perform a subjective listening test and compare subjective quality of noise-reduced speech with its estimation. In the test, we use various types of background noise and noise-reduction algorithms. The verification results indicate that the correlation between subjective quality and its objective estimation is sufficiently high regardless of the type of background noise and noise-reduction algorithm.

This paper proposes a method to generate exceptional scenarios from a normal scenario written with a scenario language. This method includes (1) generation of exceptional plans and (2) generation of exceptional scenario by a user's selection of these plans. The proposed method enables users to decrease the omission of the possible exceptional scenarios in the early stages of development. The method will be illustrated with some examples.