A four-quadrant-input linear transconductor generating a product or a product sum current is proposed. The proposed circuit eliminates the influence of channel length modulation and expands a dynamic input voltage range. As an application of the proposed circuit, the four-quadrant analog multiplier is designed. The four-quadrant analog multiplier consists of the proposed circuit, an input circuit and a class AB current buffer. HSPICE simulation results with 0.35 µm n-well single CMOS process parameter are shown in order to evaluate the proposed circuit.

Linearity of a transconductor with a theoretical linear characteristic is deteriorated by mobility degradation, in practice. In this paper, a technique to improve the linearity by combining a source-coupled pair with the transconductor is proposed. The proposed transconductor is the circuit that the deteriorated linearity of the conventional part is compensated by the transconductance characteristic of the source-coupled pair. In order to confirm the validity of the proposed technique, SPICE simulation is carried out. The transconductance change ratio of the proposed technique is about 1% and is 1/10 or less of the conventional circuit.

This paper proposes a new approach for analog portion testing, which can meet requirements for high-speed and high-accuracy testing simultaneously with reasonable cost. The key concept of the new method is cooperation of an LSI tester and some circuitry built in a target SoC device. We will explain the operation principle of the proposed method. The proposed method can be one of the methods to overcome today's expensive production test of analog portion on SoC (System on Chip) devices which heavily depends on LSI tester capability and will become harder in near future.

This paper discusses the least-squares linear filtering and smoothing (fixed-point and fixed-interval) problems of discrete-time signals from observations, perturbed by additive white noise, which can be randomly delayed by one sampling time. It is assumed that the Bernoulli random variables characterizing delay measurements are correlated in consecutive time instants. The marginal distribution of each of these variables, specified by the probability of a delay in the measurement, as well as their correlation function, are known. Using an innovation approach, the filtering, fixed-point and fixed-interval smoothing recursive algorithms are obtained without requiring the state-space model generating the signal; they use only the covariance functions of the signal and the noise, the delay probabilities and the correlation function of the Bernoulli variables. The algorithms are applied to a particular transmission model with stand-by sensors for the immediate replacement of a failed unit.

This paper presents a simple, yet effective hybrid of the minimum mean square error (MMSE) multi-user detection (MUD) and successive interference cancellation (SIC) for direct-sequence code division multiple access (DS-CDMA) systems. The proposed hybrid MUD first divides the users into groups, with each group consisting of users with a close power level. The SIC is then used to distinguish users among different groups, while the MMSE MUD is used to detect signals within each group. To further improve the performance impaired by the propagation errors, an information reuse scheme is also addressed, which can be used in conjunction with the hybrid MMSE/SIC MUD to adequately cancel the multiple access interferences (MAIs) so as to attain more accurate detections. Furthermore, the asymptotic multiuser efficiency (AME), a measure to characterize the near-far resistance capability, is also conducted to provide further insights into the new detectors. Furnished simulations, in both additive white Gaussian noise (AWGN) channels and slow flat Rayleigh fading channels, show that the performances of the proposed hybrid MMSE/SIC detectors, with or without the decision aided scheme, are superior to that of the SIC and, especially, the one with decision aided is close to that of the MMSE MUD but with substantially lower computational complexity.

This paper describes an objective measurement method designed to assess the perceived quality for digital videos. The proposed approach can be used either in the context of a reduced reference quality assessment or in the more challenging situation where no reference is available. In that way, it can be deployed in a QoS monitoring strategy in order to control the end-user perceived quality. The originality of the approach relies on the very limited computation resources which are involved, such a system could be integrated quite easily in a real time application. It uses a convolutional neural network (CNN) that allows a continuous time scoring of the video. Experiments conducted on different MPEG-2 videos, with bit rates ranging from 2 to 6 Mbits/s, show the effectiveness of the proposed approach. More specifically, a linear correlation criterion, between objective and subjective scoring, ranging from 0.90 up to 0.95 has been obtained on a set of typical TV videos in the case of a reduced reference assessment. Without any reference to the original video, the correlation criteria remains quite satisfying since it still lies between 0.85 and 0.90, which is quite high with respect to the difficulty of the task, and equivalent and more in some cases than the traditional PSNR, which is a full reference measurement.

Next generation codecs, benchmarked by the H.264/AVC standard, are providing substantial compression efficiency for the coding and transmission of video. Coupled with technologies offering larger transmission bandwidths over DSL, wireless and satellite networks, the capability of delivering high quality video services to the home is now a reality. The perceptual quality of the content delivered over communications networks will be crucial in ensuring a first-class customer experience. It is therefore important to assess the advantages and disadvantages of the optional features offered by next generation codecs. This paper describes a subjective assessment that was carried out to investigate the perceptual effects of switching the in loop de-blocking filter within the H.264/ AVC CODEC on or off. Although the filter is believed to substantially improve the perceptual quality of video, it has been suggested that in some cases negative perceptual effects can be produced. The H.264/AVC architecture allows de-blocking to be switched off in cases where there are limited processing resources or it is considered a negative perceptual effect may be introduced. This paper describes a study that examined the perceptual effects of de-blocking by employing a standardised subjective assessment methodology. The Absolute Category Rating (ACR) method was used to capture Difference Mean Opinion Scores (DMOS) for a range of video. Content was selected to span a wide and representative range of coding complexity. This content was then encoded at a variety of bit-rates to represent high, medium and low qualities. Results were used to examine the end-user perception of video quality when the de-blocking filter is switched on or off. The experimental design allowed the overall effects of the de-blocking filter to be examined and additionally the relationship between content and quality on the filter performance. The experiment found that the performance of the de-blocking filter was content-dependent. Results were used to discuss the advantages and disadvantages of in-loop de-blocking and there is an examination of content properties (e.g. spatial and temporal complexity) that influence the performance of de-blocking.

This paper proposes a parity placement scheme, Row-Oblique Parity (ROP), for protecting against double disk failure in disk array systems. It stores all data unencoded, and uses only exclusive-or (XOR) operations to compute parity. ROP is provably optimal in computational complexity, both during construction and reconstruction. It is optimal in the capacity of redundant information stored and accessed. The simplicity of ROP allowed us to implement it within the current available RAID framework.

This paper describes a numerical design procedure of element values of RC polyphase filters with equal minima in stopband and equal ripple in passband. Determination of element values of RC polyphase filters with equal-ripple characteristic have not been solved to the best knowledge of the authors. There found a paper tackling with the problem; however, it can only give sub-optimal solutions via numerical calculation [3]. We propose a numerical element value design procedure for RC polyphase filters with equi-ripple gain in both stopband and passband by using the coefficient matching method. Some design examples are given.

We propose a design theory of the miniaturized high temperature superconducting (HTS) coplanar waveguide (CPW) bandpass filter (BPF), which is composed of meanderline quarter-wavelength resonator, J- and K-inverters. The J- and K-inverters are realized by using interdigital gap and meander-shape inductor. To evaluate the kinetic inductance of the K-inverter, we fabricate the YBCO resonator connected with K-inverters and redesigned the YBCO filter parameters. Finally, we designed and fabricated the YBCO CPW quarter-wavelength resonator BPF by taking account of the kinetic inductance of the K-inverter. The experimental results are in agreement with the design parameters.

We present a quantitative evaluation of speech quality using the multiplexing scheme for the efficient transmission of voice signals in order to reduce the number of the IP packets carrying voice signals (called VoIP packets) transferred. The multiplexing scheme is applicable to a variety of media gateways controlling the bulk of voice streams over IP-based networks, based on VoIP technology. We speculated that the multiplexing scheme would reduce the degradation of speech quality due to packet loss since it also has a similar effect to interleaving the voice signal streams. However, the interleaving effect for maintaining speech quality in the scheme characterized by the feature of IP-based multiplication is not quantitatively clear. Through our end-to-end quality evaluation results of speech, as transmitted via the multiplexing scheme using dedicated hardware, we confirm the advantages of the multiplexing scheme from the perspective of achieving improved speech quality without increasing the processing delay when considering practical packet loss conditions within an IP-based network.

A parallel notch filter (PNF) for eliminating a sinusoidal signal whose frequency and phase are unknown, has been proposed previously. The PNF achieves both fast convergence and high estimation accuracy when the step-size for adaptation is appropriately determined. However, there has been no discussion of how to determine the appropriate step-size. In this paper, we derive the convergence condition on the step-size, and propose an adaptive algorithm with variable step-size so that convergence of the PNF is automatically satisfied. Moreover, we present a new filtering structure of the PNF that increases the convergence speed while keeping the estimation accuracy. We also derive a variable step-size scheme for the new PNF to guarantee the convergence. Simulation results show the effectiveness of the proposed method.

Content-Addressable Network (CAN) provides a mechanism that could retrieve objects in a P2P network by maintaining indices to those objects in a fully decentralized manner. In the CAN system, index caching is a useful technique for reducing the response time of retrieving objects. The key points of effective caching techniques are to improve cache hit ratio by actively sharing caches distributed over the P2P network with every node and to reduce a maintenance and/or routing overhead for locating the cache of a requested index. In this paper, we propose a new caching technique based on the notion of proxy-type caching techniques which have been widely used in WWW systems. It can achieve active cache sharing by incorporating the concept of proxy caching into the index access mechanism and locate a closer proxy cache of a requested index with a little routing overhead. By the result of simulations, we conclude that it can improve the response time of retrieving indices by 30% compared with conventional caching techniques.

We propose the concept of an opinion model for interactive multimodal services and apply it to an audiovisual communication service. First, psychological factors of an audiovisual communication service were extracted by using the semantic differential (SD) technique and factor analysis. Forty subjects participated in subjective tests and performed point-to-point conversational tasks on a PC-based video phone that exhibited various network qualities. The subjects assessed those qualities on the basis of 25 pairs of adjectives. Two psychological factors, i.e., an aesthetic feeling and a feeling of activity, were extracted from the results. Then, quality impairment factors affecting these two psychological factors were analyzed. We found that the aesthetic feeling was affected by IP packet loss and video coding bit rate, and the feeling of activity depended on delay time, video packet loss, video coding bit rate, and video frame rate. Using this result, we formulated an opinion model derived from the relationships among quality impairment factors, psychological factors, and overall quality. The validation test results indicated that the estimation error of our model was almost equivalent to the statistical reliability of the subjective score.

In this paper, we propose a novel dynamic bandwidth allocation scheme for the downlink real-time video streaming in the wireless cellular networks. Our scheme is able to maximize the bandwidth utilization, while satisfying the required packet loss probability, a QoS constraint, by dynamically determining the amount of bandwidth to be allocated at each unit time interval by measuring the queue length and the packet loss probability. The simulation results show that, without the need of a priori knowledge about the traffic traces, our scheme is able to achieve the same level of performance as what can be accomplished with the pre-calculated effective bandwidth in terms of the bandwidth utilization and the packet loss rate.

As an alternative solution to provide the quality of services (QoS) for broadband access over Ethernet Passive Optical Network (EPON), we present the usage of MAC control message for plural class queues and a traffic-class burst-polling based delta dynamic bandwidth allocation (DBA), referred to as TCBP-DDBA, scheme. For better QoS support, the TCBP-DDBA minimizes packet delays and delay variations for expedited forwarding packet and maximizes throughput for assured forwarding and best effort packets. The network resources are efficiently utilized and adaptively allocated to the three traffic classes for the given unbalanced traffic conditions by guaranteeing the requested QoS. Simulation results using OPNET show that the TCBP-DDBA scheme performs well in comparison to the conventional unit-based allocation scheme over the measurement parameters such as: packet delay, packet delay variation, and channel utilization.

The paper presents an adaptive algorithm named adaptive threshold nonlinear algorithm for use in adaptive filters in the complex-number domain (c-ATNA) in applications to digital QAM systems. Although the c-ATNA is very simple to implement, it makes adaptive filters highly robust against impulse noise and at the same time it ensures filter convergence as fast as that of the well-known LMS algorithm. Analysis is developed to derive a set of difference equations for calculating transient behavior as well as steady-state performance. Experiment with simulations and theoretical calculations for some examples of filter convergence in the presence of Contaminated Gaussian Noise demonstrates that the c-ATNA is effective in combating impulse noise. Good agreement between simulated and theoretical convergence proves the validity of the analysis.

We proposed and evaluated a speech packet loss concealment method which predicts lost segments from speech included in packets either before, or both before and after the lost packet. The lost segments are predicted recursively by using linear prediction both in the forward direction from the packet preceding the loss, and in the backward direction from the packet succeeding the lost segment. Predicted samples in each direction are smoothed by averaging using linear weights to obtain the final interpolated signal. The adjacent segments are also smoothed extensively to significantly reduce the speech quality discontinuity between the interpolated signal and the received speech signal. Subjective quality comparisons between the proposed method and the the packet loss concealment algorithm described in the ITU standard G.711 Appendix I showed similar scores up to about 10% packet loss. However, the proposed method showed higher scores above this loss rate, with Mean Opinion Score rating exceeding 2.4, even at an extremely high packet loss rate of 30%. Packet loss concealment of speech degraded with G.729 coding, and babble noise mixed speech showed similar trends, with the proposed method showing higher qualities at high loss rates. We plan to further improve the performance by using adaptive LPC prediction order depending on the estimated pitch, and adaptive LPC bandwidth expansion depending on the consecutive number of repetitive prediction, among many other improvements. We also plan to investigate complexity reduction using gradient LPC coefficient updates, and processing delay reduction using adaptive forward/bidirectional prediction modes depending on the measured packet loss ratio.

Recent improvements in the performance of end-computers and networks have made it feasible to construct a grid system over the Internet. A grid environment consists of many computers, each having a set of components and a distinct performance. These computers are shared among many users and managed in a distributed manner. Thus, it is important to focus on a situation in which the computers are used unevenly due to decentralized management by different task schedulers. In this study, which is a preliminary investigation of the performance of task allocation schemes employed in a decentralized environment, the average execution time of a long-lived task is analytically derived using the M/G/1-PS queue. Furthermore, assuming a more realistic condition, we evaluate the performance of some task allocation schemes adopted in the analysis, and clarify which scheme is applicable to a realistic grid environment.

In AMS/OFDM systems, a base station controls the modulation level of each subcarrier with feedback information (FBI), and then, adaptive modulated packets are transmitted from the base station to the mobile station. In this case, the mobile station requires modulation level information (MLI) to demodulate the received packet. The MLI is generally transmitted as a data symbol, so the throughput is degraded. To overcome this problem and increase the total throughput, in this paper, we propose superimposed frequency symbol based adaptive OFDM with frequency spreading and equalization. In the proposed system, each S/P transformed signal is spread by orthogonal spreading codes and combined. This means that each subcarrier holds several superimposed S/P transformed signals with the same power rate. In this case, the frequency-selective faded subcarriers obtain the same power rate for each S/P transformed signal. Therefore, the detected signals also obtain the same SINR, and as a result, we can assign the same modulation level for each frequency symbol spreading block. Hence, the proposed system requires only one piece of FBI and MLI for each frequency symbol spreading block, as compared with conventional adaptive OFDM.