Future broadband mobile communication systems are necessary to achieve the bit rates of 100 Mbit/s. Orthogonal Frequency Division Multiplexing (OFDM) transmission is an attractive technology because it can remove the influence of frequency selective fading in broadband transmission by adding a suitable guard interval to each OFDM symbol. However, peak-to-average power ratio (PAPR) is very large in OFDM transmission. In this paper, we propose a new PAPR reduction method which can be applied even when unusable bands are inside the system band. In the proposed method, peak reduction signals are generated by iterative signal processing only in the usable frequency band, and filtering to remove out-of-band components of the peak reduction signals is incorporated into the iterative signal processing. The results of computer simulation show that the proposed method can effectively reduce peak power without expanding the spectrum both outside the system band and into unusable bands inside the system band. By using the proposed method, the broadband mobile communication system with low peak power and high flexibility of frequency band use can be realized.

In this paper, the 1-D real-valued discrete Gabor transform (RDGT) proposed in our previous work and its relationship with the complex-valued discrete Gabor transform (CDGT) are briefly reviewed. Block time-recursive RDGT algorithms for the efficient and fast computation of the 1-D RDGT coefficients and for the fast reconstruction of the original signal from the coefficients are then developed in both the critical sampling case and the oversampling case. Unified parallel lattice structures for the implementation of the algorithms are studied. And the computational complexity analysis and comparison show that the proposed algorithms provide a more efficient and faster approach for the computation of the discrete Gabor transforms.

This paper describes a 3.5 V operation InGaP HBT MMIC power amplifier module for use in GSM/EDGE dual-mode, 900/1800/1900 MHz triple band handset applications. Conventional GSM amplifiers have a high linear gain of 40 dB or more to realize efficiency operation in large gain compression state exceeding at least 5 dB. On the other hand, an EDGE amplifier needs a linear operation to prevent signal distortion. This means that a high linear gain amplifier cannot be applied to the EDGE amplifier, because the high gain leads to the high noise power in the receive band (Rx-noise). In order to solve this problem, we have changed the linear gain of the amplifier between GSM and EDGE mode. In EDGE mode, the stage number of the amplifier changes from three to two. To reduce a high gain, the first stage transistors in the amplifier is bypassed through the diode switches. This newly proposed bypass circuit enables a high gain in GSM mode and a low gain in EDGE, thus allowing the amplifier to operate with high efficiency in both modes while satisfying the Rx-noise specification. In conclusion, with diode switches and a band select switch built on the MMIC, the module delivers a Pout of 35.5 dBm and a PAE of about 50% for GSM900, a 33.4 dBm Pout and a 45% PAE for GSM1800/1900. While satisfying an error vector magnitude (EVM) of less than 4% and a receive-band noise power of less than -85 dBm/100 kHz, the module also delivers a 29.5 dBm Pout and a PAE of over 25% for EDGE900, a 28.5 dBm Pout and a PAE of over 25% for EDGE1800/1900.

This paper presents a framework for automatic video region-of-interest determination based on visual attention model. We view this work as a preliminary step towards the solution of high-level semantic video analysis. Facing such a challenging issue, in this work, a set of attempts on using video attention features and knowledge of computational media aesthetics are made. The three types of visual attention features we used are intensity, color, and motion. Referring to aesthetic principles, these features are combined according to camera motion types on the basis of a new proposed video analysis unit, frame-segment. We conduct subjective experiments on several kinds of video data and demonstrate the effectiveness of the proposed framework.

This paper presents a VLSI processor for high-speed and reliable stereo matching based on adaptive window-size control of SAD(Sum of Absolute Differences) computation. To reduce its computational complexity, SADs are computed using multi-resolution images. Parallel memory access is essential for highly parallel image processing. For parallel memory access, this paper also presents an optimal memory allocation that minimizes the hardware amount under the condition of parallel memory access at specified resolutions.

This paper describes two way satellite system environments on geostationary orbit (GEO) and performance enhancement mechanisms which reduces round trip time (RTT) and supports real-time services. We use performance enhancing proxy (PEP) for reducing round trip time and user-level real-time scheduler for reducing deadline violation tasks. The user-level real-time scheduling method classifies priority of user process into four types and those are reflected in kernel. With these dual performance enhancement mechanisms, we can improve quality of service (QoS) of end-user who connects to the DVB-RCS system.

While mobile networks have been enhanced to support a variety of mobile multimedia services such as video telephony and rich data content delivery, a new challenge is being created by the remarkable development of micro-device technologies such as micro processor-chips, sensors, and RF tags. These developments suggest the rapid emergence of the ubiquitous computing environment; computers supporting human life without imposing any stress on the users. The combination of broadband global networks and ubiquitous computing environment will lead to an entirely new class of services, which we call ubiquitous networking services. This paper discusses how to create ubiquitous service environments comparing global networking approaches which are based on fixed and mobile networks. It is shown that the mobile approach is better from service applicability and reliability viewpoints. Networking architecture is proposed which expand 4G mobile cellular networks to real space via gateways on the edges of the mobile network (i.e. mobile terminals). A new set of technical requirements will emerge via this approach, which may accelerate the paradigm shift from the current mobile network architecture and even from the Internet of today.

Since the inception of Globally Asynchronous Locally Synchronous (GALS) VLSI design, GALS has been considered a promising design technique for multi-clock-domain System-on-Chip (SoC). Among the handshake protocols available for SoC design, delay insensitive (DI) handshake protocol is becoming a core technology, since it facilitates robust data transfer regardless of wire delay variation. In this paper, a new data encoding scheme Differential Value Encoding (DVE) is proposed for two-phase 1-of-N DI handshake protocol. Compared with the conventional data encoding method, the proposed scheme effectively reduces the crosstalk effect on wires sending sequentially increasing data patterns, resulting in reduction of the data transfer time. Simulation results with SPEC CPU 2000 benchmarks and sequentially increasing data pattern reveal that the DVE scheme can reduce the crosstalk effect by tens of percentage and significantly decrease the data transfer time.

A new and simple image processing approach for the measurement of the length of pedestrian crossings with a view to develop a travel aid for the blind people is described. In a crossing, the usual black road surface is painted with constant width periodic white bands. The crossing length is estimated using vector geometry from the left- and the right-border lines, the first-, the second- and the end-edge lines of the crossing region. Image processing techniques are applied on the crossing image to find these lines. Experimental results using real road scenes with pedestrian crossing confirm the effectiveness of the proposed method.

This paper proposes "Multipath Control and Proactive Control" to realize a robust QoS control system for mobile multimedia communication in an IP-based cellular network. In this network, all kinds of traffic will share the same backbone network. This requires a QoS system that differentiates services according to the required quality. Though DiffServ is thought to be a promising technique for achieving QoS, an effective path control scheme and a technique that is suitable enough for rapid traffic changes are not yet available. Our solution is multipath control using linear optimization combined with proactive control using traffic anomaly detection. Simulation results show that multipath control and proactive control improve system performance in terms of throughput and packet loss when rapid traffic change takes place.

A robust microphone array for speech enhancement and noise suppression is studied in this paper. To overcome target signal cancellation problem of conventional beamformer caused by array imperfections or reverberation effects of acoustic enclosure, the proposed microphone array adopts an arbitrary model of channel transfer function (TF) relating microphone and speech source. Since the estimation of channel TF itself is often intractable, herein, transfer function ratio (TFR) is estimated instead and used to form a suboptimal beamformer. A robust TFR estimation method is proposed based on signal subspace analysis technique against stationary or slowly varying noise. Experiments using simulated signal and actual signal recorded in a real room illustrate that the proposed method has high performance in adverse environment.

This paper focuses on the underdetermined blind source separation (BSS) of three speech signals mixed in a real environment from measurements provided by two sensors. To date, solutions to the underdetermined BSS problem have mainly been based on the assumption that the speech signals are sufficiently sparse. They involve designing binary masks that extract signals at time-frequency points where only one signal was assumed to exist. The major issue encountered in previous work relates to the occurrence of distortion, which affects a separated signal with loud musical noise. To overcome this problem, we propose combining sparseness with the use of an estimated mixing matrix. First, we use a geometrical approach to detect when only one source is active and to perform a preliminary separation with a time-frequency mask. This information is then used to estimate the mixing matrix, which allows us to improve our separation. Experimental results show that this combination of time-frequency mask and mixing matrix estimation provides separated signals of better quality (less distortion, less musical noise) than those extracted without using the estimated mixing matrix in reverberant conditions where the reverberant time (TR) was 130 ms and 200 ms. Furthermore, informal listening tests clearly show that musical noise is deeply lowered by the proposed method comparatively to the classical approaches.

In this paper, a cell-driven multiplier generator is developed that can produce high-performance gate-level netlists for multiplier-related arithmetic functional units, including multipliers, multiplier and accumulators (MAC) and dot product calculator. The generator optimizes the speed/area performance both in the partial product compression and in the final addition stage for the specified process technology. In addition to the conventional CMOS full adder cells, we have also designed fast compression elements based on pass-transistor logic for further performance improvement of the generated multipliers. Simulation results show that our proposed generator could produce better multiplier-related functional units compared to those generated using Synopsys Designware library or other previously proposed approaches.

This paper introduces the multiple signal classification (MUSIC) method that utilizes the transfer characteristics of microphones located at the same place, namely aggregated microphones. The conventional microphone array realizes a sound localization system according to the differences in the arrival time, phase shift, and the level of the sound wave among each microphone. Therefore, it is difficult to miniaturize the microphone array. The objective of our research is to build a reliable miniaturized sound localization system using aggregated microphones. In this paper, we describe a sound system with N microphones. We then show that the microphone array system and the proposed aggregated microphone system can be described in the same framework. We apply the multiple signal classification to the method that utilizes the transfer characteristics of the microphones placed at a same location and compare the proposed method with the microphone array. In the proposed method, all microphones are placed at the same place. Hence, it is easy to miniaturize the system. This feature is considered to be useful for practical applications. The experimental results obtained in an ordinary room are shown to verify the validity of the measurement.

An efficient algorithm to reduce the noise from the Nuclear Magnetic Resonance Free Induction Decay (NMR FID) signals is presented, in this paper, via the oversampled real-valued discrete Gabor transform using the Gaussian synthesis window. An NMR FID signal in the Gabor transform domain (i.e., a joint time-frequency domain) is concentrated in a few number of Gabor transform coefficients while the noise is fairly distributed among all the coefficients. Therefore, the NMR FID signal can be significantly enhanced by performing a thresholding technique on the coefficients in the transform domain. Theoretical and simulation experimental analyses in this paper show that the oversampled Gabor transform using the Gaussian synthesis window is more suitable for the NMR FID signal enhancement than the critically-sampled one using the exponential synthesis window, because both the Gaussian synthesis window and its corresponding analysis window in the oversampling case can have better localization in the frequency domain than the exponential synthesis window and its corresponding analysis window in the critically-sampling case. Moreover, to speed up the transform, instead of the commonly-used complex-valued discrete Gabor transform, the real-valued discrete Gabor transform presented in our previous work is adopted in the proposed algorithm.

When video data are transmitted via the network, the quality of video data must be carefully chosen to be best under the condition that the transmission is not influenced by other internet services. They often use the simulcast type, which uses independent streams that are stored and transmitted for the quality, considering implementation, when they select the video quality. On the other hand, we had already proposed the scalable structure, which consists of base and enhancement data, but when they require the high quality video, these data are combined using the transcoding methods. In this paper, we propose the video contents delivery methods with scalable transcoding, in which users can update the quality of video data even after the transmission by base data and differential data. In order to reduce the total time of not only users' access time, but also watching time, we compare simulcast method with proposed methods in the total content utilization time using a video contents access model, and evaluate required transcoding time to reduce the waiting time of users.

It is well known that the trellises of lattices can be employed to decode efficiently. It was proved in [1] and [2] that if a lattice L has a finite trellis under the coordinate system , then there must exist a basis (b1,b2,,bn) of L such that Wi=span() for 1in. In this letter, we prove this important result in a completely different method, and give an efficient method to compute all bases of this type.

In this paper, we propose a new error feedback (EF) structure for 2-D separable-denominator digital filters described by a rational transfer function. In implementing two-dimensional separable-denominator digital filters, the minimum delay elements structures are common. In the proposed structure, the filter feedback-loop corresponding to denominator polynomial is placed at a different location compared to the commonly used structures. The proposed structure can minimize the roundoff noise more than the previous structure though the number of multipliers is less than that of previous one. Finally, we present a numerical example by designing the EF on the proposed structure and demonstrate the effectiveness of the proposed method.

The actual sound environment system exhibits various types of linear and non-linear characteristics, and it often contains an unknown structure. Furthermore, the observations in the sound environment are often in the level-quantized form. In this paper, a method for estimating the specific signal for stochastic systems with unknown structure and the quantized observation is proposed by introducing a system model of the conditional probability type. The effectiveness of the proposed theoretical method is confirmed by applying it to the actual problem of psychological evaluation for the sound environment.

The reduction of the signaling load associated with IP mobility management is one of the significant challenges to IP mobility support protocols. Hierarchical Mobile IPv6 (HMIPv6) aims to reduce the number of the signaling messages in the backbone networks, and improve handoff performance by reducing handoff latency. However, this does not imply any change to the periodic binding update (BU) to the home agent (HA) and the correspondent node (CN), and now a mobile node (MN) additionally should send it to the mobility anchor point (MAP). Moreover, the MAP should tunnel the received packets to be routed to the MN. These facts mean that the reduction of the BU messages in the backbone networks can be achieved at the expense of the increase in the signaling bandwidth consumption within a MAP domain. On the other hand, it is observed that an MN may habitually stay for a relatively long time or spend on using much Internet in a specific cell (hereafter, home cell) covering its home, office or laboratory, etc. Thus, considering the preceding facts and observation, HMIPv6 may not be favorable especially during a home cell residence time in terms of signaling bandwidth consumption. To overcome these drawbacks of HMIPv6, we propose a history-based auxiliary mobility management strategy (H-HMIPv6) to enable an MN to selectively switch its mobility management protocols according to whether it is currently in its home cell or not in HMIPv6 networks. The operation of H-HMIPv6 is almost the same as that of HMIPv6 except either when an MN enters/leaves its home cell or while it stays in its home cell. Once an MN knows using its history that it enters its home cell, it behaves as if it operates in Mobile IPv6 (MIPv6), not in HMIPv6, until it leaves its home cell; No periodic BU messages to the MAP and no packet tunneling occur during the MN's home cell residence time. The numerical results indicate that compared with HMIPv6, H-HMIPv6 has apparent potential to reduce the signaling bandwidth consumption and the MAP blocking probability.