A band-pass delta-sigma modulator (BPDSM) is a key building block to implement a digital intermediate frequency (IF) receiver in a wireless communication system. This paper proposes a time-interleaved (TI) switched-capacitor (SC) BPDSM architecture that consists of 5-stage TI blocks with recursive loop. The proposed TI BPDSM provides reduction in the clock frequency requirement by a factor of 5 and relaxes the settling time requirement to one-fourth of conventional approach. The test chip was designed and fabricated for a 30-MHz IF system with a 0.35-µm CMOS process. The measured peak SNR for a 200-kHz bandwidth is 63 dB while dissipating 75 mW from a 3.3-V supply and occupying 1.3 mm2.

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.

A simple method has been proposed for the measurement of the output power and phase characteristics of the 3rd-order inter-modulation distortion (IM3) components appearing in multistage power amplifiers. By adopting a unique definition of the phase for the IM3 components that is independent of the delay time caused by transmission lines and other instrument devices, it is possible to measure the phase, merely by using a vector signal analyzer. It is demonstrated that an accurate estimation of the IM3 characteristics of two-stage cascaded power amplifiers for cellular radio handheld terminals can be made by using the IM3 characteristics of the 1st and 2nd-stage amplifiers as measured by the proposed method. The results indicate that it is possible to reduce the dissipation power by 18% at 28 dBm RF output power with respect to conventional measurement methods. Further studies show that the error in the resultant vector of the estimated IM3 is less than 1 dB, when the asymmetry characteristics of the IM3 sidebands in the 2nd-stage amplifier are less than 7.3%.

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.

This paper presents some effective methods for improving the performance of a speaker identification system. Based on the multiresolution property of the wavelet transform, the input speech signal is decomposed into various frequency subbands in order not to spread noise distortions over the entire feature space. For capturing the characteristics of the vocal tract, the linear predictive cepstral coefficients (LPCC) of the lower frequency subband for each decomposition process are calculated. In addition, a hard threshold technique for the lower frequency subband in each decomposition process is also applied to eliminate the effect of noise interference. Furthermore, cepstral domain feature vector normalization is applied to all computed features in order to provide similar parameter statistics in all acoustic environments. In order to effectively utilize all these multiband speech features, we propose a modified vector quantization as the identifier. This model uses the multilayer concept to eliminate the interference among the multiband speech features and then uses the principal component analysis (PCA) method to evaluate the codebooks for capturing a more detailed distribution of the speaker's phoneme characteristics. The proposed method is evaluated using the KING speech database for text-independent speaker identification. Experimental results show that the recognition performance of the proposed method is better than those of the vector quantization (VQ) and the Gaussian mixture model (GMM) using full-band LPCC and mel-frequency cepstral coefficients (MFCC) features in both clean and noisy environments. Also, a satisfactory performance can be achieved in low SNR environments.

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 %.

Transitional fundamental frequency (F0) characteristics comprise a crucial part of F0 dynamics in singing. This paper examines the F0 characteristics during the note transition period. An analysis of the singing voice of a professional baritone strongly suggests that asymmetries exist in the mechanisms used for controlling rising and falling. Specifically, the F0 contour in rising transitions can be modeled as a step response from a critically-damped second-order linear system with fixed average/maximum speed of change, whereas that in falling transitions can be modeled as a step response from an underdamped second-order linear system with fixed transition time. The validity of the model is examined through auditory experiments using synthesized singing voice.

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.

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.

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.

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.

In mobile multimedia environment, it is very important to minimize handoff latency due to mobility. In terms of reducing handoff latency, Hierarchical Mobile IPv6 (HMIPv6) can be an efficient approach, which uses a mobility agent called Mobility Anchor Point (MAP) in order to localize registration process. However, MAP can be a single point of failure or performance bottleneck. In order to provide mobile users with satisfactory quality of service and fault-tolerant service, it is required to cope with the failure of mobility agents. In, we proposed Robust Hierarchical Mobile IPv6 (RH-MIPv6), which is an enhanced HMIPv6 for fault-tolerant mobile services. In RH-MIPv6, an MN configures two regional CoA and registers them to two MAPs during binding update procedures. When a MAP fails, MNs serviced by the faulty MAP (i.e., primary MAP) can be served by a failure-free MAP (i.e., secondary MAP) by failure detection/recovery schemes in the case of the RH-MIPv6. In this paper, we investigate the comparative study of RH-MIPv6 and HMIPv6 under several performance factors such as MAP unavailability, MAP reliability, packet loss rate, and MAP blocking probability. To do this, we utilize a semi-Markov chain and a M/G/C/C queuing model. Numerical results indicate that RH-MIPv6 outperforms HMIPv6 for all performance factors, especially when failure rate is high.

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.

We present a new space-time successive interference cancellation (ST-SIC) scheme with multiple transceiver antennas for direct-sequence code division multiple access (DS-CDMA) systems. The proposed scheme is computationally very efficient, while maintains the performance close to the previous space-time multiuser detection (ST-MUD) scheme. The bit error rate (BER) performance of the ST-SIC scheme for coherent phase shift keying (PSK) modulation is analytically examined in Rayleigh fading channels, and its validity and usefulness are demonstrated by computer simulations.

The objective of this research is to compare the performance of the Matrix Pencil Method (MPM) and well known root-MUSIC algorithm for high resolution DOA estimation. Performance of each technique in terms of the probability of resolution and SNR in the presence of noise is investigated. Simulation results show that the MPM has a superior resolution to the root-MUSIC algorithm.

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.

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.

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 propose a fast signature scheme which realizes short transmissions and minimal on-line computation. Our scheme requires a modular exponentiation as preprocessing (i.e., off-line computation). However, we need to acknowledge the existance of the following remarkable properties: neither multiplication nor modular reduction is used in the actual signature generation (i.e., on-line computation). Our scheme requires only two operations: hashing and addition. Although some fast signature schemes with small on-line computation have been proposed so far, those schemes require multiplication or modular reduction in the on-line phase. This leads to a large amount of work compared to that of addition. As far as we know, this is the first approach to obtain the fast signature without those two calculus methods.