We consider a new post-filtering algorithm for residual acoustic echo cancellation in hands-free application. The new post-filtering algorithm is composed of AR analysis, pitch prediction, and noise reduction algorithm. The residual acoustic echo is whitened via AR analysis and pitch prediction during no near-end talker period and then is cancelled by noise reduction algorithm. By removing speech characteristics of the residual acoustic echo, noise reduction algorithm reduces the power of the residual acoustic echo as well as the ambient noise. For the hands-free application in the moving car, the proposed system attenuated the interferences more than 15 dB at a constant speed of 80 km/h.

In this paper, we formally define and analyze the security notions of authenticated encryption in unconditional security setting. For confidentiality, we define the notions, APS (almost perfect secrecy) and NM (non-malleability), in terms of an information-theoretic viewpoint along with our model where multiple senders and receivers exist. For authenticity, we define the notions, IntC (integrity of ciphertexts) and IntP (integrity of plaintexts), from a view point of information theory. And then we combine the above notions to define the security notions of unconditionally secure authenticated encryption. Then, we analyze relations among the security notions. In particular, it is shown that the strongest security notion is the combined notion of APS and IntC. Finally, we formally define and analyze the following generic composition methods in the unconditional security setting along with our model: Encrypt-and-Sign, Sign-then-Encrypt and Encrypt-then-Sign. Consequently, it is shown that: the Encrypt-and-Sign composition method is not always secure; the Sign-then-Encrypt composition method is not always secure; and the Encrypt-then-Sign composition method is always secure, if a given encryption meets APS and a given signature is secure.

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.

An important barrier to the application of high-temperature superconducting microwave filters is their power-handling capability. To clarify the key parameters for improving the power-handling capability of rf filters based on high-temperature superconductors with microstrip structures, we synthesize bandpass filters with different layouts using several kinds of thin film high-temperature superconductors, and subject them to third-order intermodulation measurements. By improving the sensitivity of the measurement set-up through the selective reduction of the fundamental output signals, we succeed in measuring the intermodulation signals of the superconducting filters. The experimental results indicate that increasing the film thickness and utilizing MBE-grown films of NdBa2Cu3O7 films are effective in obtaining high-power handling microstrip filters.

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.

As various mobile communication systems have developed, dramatically integrated wireless network, where users can communicate seamlessly via several wireless access systems, have become expected. At present, there are many studies of integrated wireless network, but no study of a network design method. Therefore, in this paper, we discuss a network design method for integrated wireless networks. Because of the handover procedure, the network design where adjacent base stations are connected to the same router, regardless of radio system type, is simply considered. However, in such a design, where mobile users crowd into a particular area and users' access to the base stations located there increases, the load of these accesses is centralized to the single router. To overcome this problem, we propose a new network design wherein the base stations of heterogeneous wireless communication systems, the service areas of which overlap, are connected to a different router. In the proposed network design, although users' accesses are concentrated on the base stations located in a particular area, users in that area can be assigned bandwidth of several upper links according to the access conditions of the base stations in neighboring areas. Finally, we show the excellent performance of the proposed design by simulation experiments.

A CMOS DFE (decision feedback equalization) receiver with a clock-data skew compensation was implemented for the SSTL (stub-series terminated logic) SDRAM interface. The receiver consists of a 2 way interleaving DFE input buffer for ISI reduction and a X2 over-sampling phase detector for finding the optimum sampling clock position. The measurement results at 1.2 Gbps operation showed the increase of voltage margin by about 20% and the decrease of time jitter in the recovered sampling clock by about 40% by equalization in an SSTL channel with 2 pF 4 stub load. Active chip area and power consumption are 3001000 µm2 and 142 mW, respectively, with a 2.5 V, 0.25 µm CMOS process.

The rapid spread of cellular phones in recent years has facilitated not only voice communication but also Internet access via the cellular phone system, and in addition, subscriber demand has led to a diversification in the services provided. One service in high demand is the seamless use of cellular phones in both public and private wireless network areas. In the data world, there is already such an application in the form of public and private use of wireless LAN. However, an increase in the number of users would require the realization of low-cost, easy-to-install very small base stations (VSBS) that use the frequency band efficiently in order to allow private use of ordinary cellular phones. To bring such VSBS into effect, a technology that autonomously selects frequencies which do not interfere with the public communication system from out of the publicly used frequency band is essential for turning such VSBS into reality. This paper proposes a frequency selection algorithm that actively uses cellular phone features such as frequency selection and received signal level measurement, and discusses the results of verification experiments.

We propose a new authentication and key establishment (AKE) protocol that can be applied to low-power PDAs in Public Wireless LANs (PWLANs), using two factor authentication and precomputation. This protocol provides mutual authentication, identity privacy, and half forward-secrecy. The computational complexity that the client must perform is just one symmetric key encryption and five hash functions during the runtime of the protocol.

This paper proposes "Handover with Proactive Anchor Router Relocation and Data Buffering" to suppress packet loss and packet miss-ordering during handover. To prevent packet miss-ordering, anchor router is proactively relocated to the optimal position before the mobile terminal performs handover. And, to eliminate packet loss during handover, anchor router buffers the packet only during handover. Moreover, anchor router assigns sequential numbers to the buffered packets to eliminate duplicate packet reception. Simulation results show that our proposal eliminates packet miss-ordering and duplicate reception while preventing packet loss.

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.

The concepts of M-convexity and L-convexity, introduced by Murota (1996, 1998) for functions on the integer lattice, extract combinatorial structures in well-solved nonlinear combinatorial optimization problems. These concepts are extended to polyhedral convex functions and quadratic functions on the real space by Murota-Shioura (2000, 2001). In this paper, we consider a further extension to general convex functions. The main aim of this paper is to provide rigorous proofs for fundamental properties of general M-convex and L-convex functions.

The model for the process of F0 contour generation, first proposed by Fujisaki and his coworkers, has been successfully applied to Standard Chinese, which is a typical tone language with a distinct feature that both positive and negative tone commands are required. However, the inverse problem, viz., automatic derivation of the model parameters from an observed F0 contour of speech, cannot be solved analytically. Moreover, the extraction of model parameters for Standard Chinese is more difficult than for Japanese and English, because the polarity of tone commands cannot be inferred directly from the F0 contour itself. In this paper, an efficient method is proposed to solve the problem by using information on syllable timing and tone labels. With the same framework as for the successive approximation method proposed for Japanese and English, the method presented here for Standard Chinese is focused on the first-order estimation of tone command parameters. A set of intra-syllable and inter-syllable rules are constructed to recognize the tone command patterns within each syllable. The experiment shows that the method works effectively and gives results comparable to those obtained by manual analysis.

Currently, the IP infrastructure is widely available, which reduces the deployment and implementation cost of IP-based wireless networks. But, the corresponding researches on Quality of Service (QoS) in wireless mobile networks have not been done sufficiently. Although All-IP network has gotten the spotlight for both wired and wireless integrated networks, it has some unavoidable defects as it is based on best-effort IP transmission technologies. By focusing our attention on this point, we propose an Advanced QoS Provisioning Scheme (AQPS) that can provide satisfied QoS for multimedia applications based on accurate traffic measurements. With the AQPS, we consider 1xEV-DV system as a target system, extract QoS requirements for this system, and develop a corresponding QoS provisioning method using CDMA traffic measurement. The AQPS is designed to estimate delay parameters between a mobile station and a BTS (or repeater), verify whether the mobile station can support RSVP or not, operate a corresponding scheduler, and assign appropriate time slots based on traffic classes. One of the interesting results of this research is that the capability of distinguishing BTS' traffic from repeater's traffic helps save implementation cost of CDMA systems. The AQPS has been verified to outperform the existing system used in the current CDMA network and to efficiently manage network resources by computer simulations.

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.

Slant correction is a preprocessing technique to improve segmentation and recognition accuracy for handwritten word recognition. All conventional slant correction techniques were performed by the estimation of the average slant angle and the shear transformation. In this paper, a nonuniform slant correction technique for handwritten word recognition is proposed where the slant correction problem is formulated as a global optimal estimation problem of the sequence of local slant angles. The optimal estimation is performed by a dynamic programming based algorithm. From experimental results it was shown that the present technique outperforms conventional uniform slant correction techniques.

Cryptography and Coding Theory are closely related in many respects. Recently, the problem of "decoding Reed Solomon codes" (also known as "polynomial reconstruction") was suggested as an intractability assumption to base the security of protocols on. This has initiated a line of cryptographic research exploiting the rich algebraic structure of the problem and its variants. In this paper we give a short overview of the recent works in this area as well as list directions and open problems in Polynomial Reconstruction Based Cryptography.

This paper provides methods for construction of pairing-based cryptosystems based on non-supersingular elliptic curves.

We proposed a one-way trapdoor permutation f based multi-signature scheme which can keep tighter reduction rate. Assuming the underlying hash functions are ideal, our proposed scheme is not only provably secure, but are so in a tight. An ability to forge multi-signatures with a certain amount of computational resources implies the ability to invert a one-way trapdoor permutation f (on the same size modulus) with about the same computational effort. The proposed scheme provides the exact security against Adaptive-Chosen-Message-Attack and Adaptive-Insider-Attack by . can also attack in key generation phase, and act in collusion with corrupted signers.

A random number generator composed of single electron devices is presented. Due to stochastic behavior of electron tunneling process, single electron devices have intrinsic randomness. Using its randomness, a true random number generator can be implemented. Although fluctuation of device parameters degrades the performance of the proposed circuit, we show that the adjustment of the bias voltages can compensate the fluctuation.