A novel closed-form algorithm is presented for estimating the 3-D location (azimuth angle, elevation angle, and range) of a single source in a uniform circular array (UCA) with a center sensor. Based on the centrosymmetry of the UCA and noncircularity of the source, the proposed algorithm decouples and estimates the 2-D direction of arrival (DOA), i.e. azimuth and elevation angles, and then estimates the range of the source. Notwithstanding a low computational complexity, the proposed algorithm provides an estimation performance close to that of the benchmark estimator 3-D MUSIC.

A sound field reproduction method is proposed that uses blind source separation and a head-related transfer function. In the proposed system, multichannel acoustic signals captured at distant microphones are decomposed to a set of location/signal pairs of virtual sound sources based on frequency-domain independent component analysis. After estimating the locations and the signals of the virtual sources by convolving the controlled acoustic transfer functions with each signal, the spatial sound is constructed at the selected point. In experiments, a sound field made by six sound sources is captured using 48 distant microphones and decomposed into sets of virtual sound sources. Since subjective evaluation shows no significant difference between natural and reconstructed sound when six virtual sources and are used, the effectiveness of the decomposing algorithm as well as the virtual source representation are confirmed.

In this paper, we propose a subcarrier resource allocation algorithm for managing the video quality degradation for multiuser orthogonal frequency division multiplex (OFDM) systems. The proposed algorithm exploits the unequal importance existing in different picture types for video coding and the diversity of subcarriers for multiuser systems. A model-based performance metric is first derived considering the error concealment and error propagation properties of the H.264 video coding structure. Based on the information on video quality enhancement existing in a packet to be transmitted, we propose the distortion management algorithm for balancing the subcarriers and power usages for each user and minimizing the overall video quality degradation. In the simulation results, the proposed algorithm demonstrates a more gradual video quality degradation for different numbers of users compared with other resource allocation schemes.

In this letter, we present an exact analytic expression for the maximum signal-to-interference ratio (SIR) for receivers communicating with multiple transmitting nodes over a general time-varying channel, where one of the nodes is chosen as a desired signal source based on the instantaneous channel condition and the other nodes act as interference sources. As an illustrative example, the maximum SIR distribution of a mobile receiver surrounded by three base stations (BS) is determined in a closed-form formula for Rayleigh fading channels, and its accuracy is confirmed using simulation results.

We present a novel frequency partitioning technique of fractional frequency reuse (FFR) that reduces the effect of co-channel interference and increases the capacity of OFDM systems. The usable sub-channel sets are classified into the common sub-channel sets for all cells and the dedicated sub-channel sets for specific cell types in FFR. The proposed fractional frequency reuse with ordering scheme (FFRO) can decrease the amount of interference in the common sub-channel sets by specially designing the sub-channel sets and the order of sub-channel assignment for specific cell types. Simulation results show that the proposed FFRO yields enhanced performance for both uniform and non-uniform distributions of traffic load.

We propose a velocity-based bicasting handover scheme for the efficient utilization of backhaul network resources in fourth-generation mobile systems. The original bicasting handover scheme adopts the mechanism of holding the data of a mobile station (MS) in all potential target base stations in advance, before the actual handover execution of the MS. The scheme minimizes the packet transmission delay caused by handover and achieves the goal of seamless connectivity, however, it results in an aggressive consumption of the backhaul network resources. Moreover, as the scheme gets widely adopted for high data rate real-time services and the demand for these services grows, the amount of the resources consumed due to the bicasting will increase tremendously. In this paper, we present a new bicasting handover scheme that reduces the data bicasting time, thereby improving the backhaul network resource utilization. Our scheme exploits the velocity parameter of MS and introduces a novel concept of bicasting threshold determined for the specific mobile speed groups. Simulations prove the efficiency of our scheme over the original one in overcoming the aggressive resource consumption at the backhaul network.

This paper presents the design of a CMOS RF Power Detector (PD) using 0.18 µm standard CMOS technology. The PD is an improved unbalanced source coupled pair incorporating an output differential amplifier and sink current steering. It realizes an input detectable power range of -30 to -20 dBm over 0.1-1 GHz. Also it shows a maximum data rate of 30 Mbps with 2 pF output loading under OOK modulation. The overall current consumption is 1.9 mA under a 1.5 V supply.

In digital speech communication over noisy high packet loss rate wireless channels, improving the overall performance of the realtime speech coding and transmission system is of great importance. A novel joint speech coding and transmission algorithm is proposed by fully exploiting the correlation between speech coding, channel coding and the transmission process. The proposed algorithm requires no algorithm delay and less bandwidth expansion while greatly enhancing the error correcting performance and the reconstructed speech quality compared with conventional algorithms. Simulations show that the residual error rate is reduced by 84.36% and the MOS (Mean Opinion Score) is improved over 38.86%.

We propose Adaptive Resource Allocation for the Partial Block MC-CDMA (ARA-PB/MC-CDMA) system. The ARA-PB/MC-CDMA system aims to improve total throughput performance and frequency efficiency across various channel conditions. It adaptively changes the number of blocks to improve the throughput performance and frequency efficiency according to the Signal to Interference Ratio (SIR). Therefore, the proposed system supports various Quality of Service (QoS) requirements for various SIR values.

Due to the dynamic nature and uncertainty of grid computing, system reliability can become very unpredictable. Thus, a well-defined scheduling mechanism that provides high system availability for grid applications is required. In this letter, we propose a SLA-constrained policy-based scheduling mechanism to enhance system performance in grid. Also, we implement the proposed model and show that our policy-based scheduling mechanism can guarantee high system availability as well as support load balancing on an experimental basis.

In this paper, we propose a computationally efficient method to solve the large dimension Adaptive Subcarrier Assignment and Bit Allocation (ASABA) problem of multiuser orthogonal frequency division multiplexing system. Our algorithm consists of three Ordinal Optimization (OO) stages to find a good enough solution to the considered problem. First of all, we reformulate the considered problem to separate it into subcarrier assignment and bit allocation problem such that the objective function of a feasible subcarrier assignment pattern is the corresponding optimal bit allocation for minimizing the total consumed power. Then in the first stage, we develop an approximate objective function to evaluate the performance of a subcarrier assignment pattern and use a genetic algorithm to search through the huge solution space and select s best subcarrier assignment patterns based on the approximate objective values. In the second stage, we employ an off-line trained artificial neural network to estimate the objective values of the s subcarrier assignment patterns obtained in stage 1 and select the l best patterns. In the third stage, we use the exact objective function to evaluate the l subcarrier assignment patterns obtained in stage 2, and the best one associated with the corresponding optimal bit allocation is the good enough solution that we seek. We apply our algorithm to numerous cases of large-dimension ASABA problems and compare the results with those obtained by four existing algorithms. The test results show that our algorithm is the best in both aspects of solution quality and computational efficiency.

Improving the overall performance of reliable speech communication in ultrashort wave radios over very noisy channels is of great importance and practical use. An iterative joint source-channel (de-)coding and (de-)modulation (JSCCM) algorithm is proposed for ITU-T Rec.G.729EV by both exploiting the residual redundancy and passing soft information throughout the receiver while introducing a systematic global iteration process. Being fully compatible with existing transmitter structure, the proposed algorithm does not introduce additional bandwidth expansion and transmission delay. Simulations show substantial error correcting performance and synthesized speech quality improvement over conventional separate designed systems in delay and bandwidth constraint channels by using the JSCCM algorithm.

To guarantee the timely provisioning of QoS to real-time oriented multiparty and distributed computing applications (e.g., video conferencing and grid computing) that require the utilization of time and quantity-related resources, the resource should be reserved in advance. However, this new type of reservation (i.e., advance reservation) may collide with legacy (i.e., immediate) reservations that do not specify their session duration. In this paper, to suggest a balanced solution between sharing pools of resources and the managing of collisions between reservation calls, a revenue-based resource sharing scheme (focusing on the bandwidth) is proposed. The proposed scheme attempts to avoid possible collisions by employing a virtual profile of resources reserved in advance in an effort to achieve enhanced resource utilization. It is shown through NS-2 [22] -based network simulations that the proposed scheme can achieve balanced performance when compared with other schemes, including static resource partitioning and complete resource sharing.

In real-time embedded systems, there is requirement for adapting both energy consumption and Quality of Services (QoS) of tasks according to their importance. This paper proposes an adaptive power-aware resource allocation method to resolve a trade-off between the energy consumption and QoS levels according to their importance with guaranteeing fairness. The proposed resource allocator consists of two components: the total resource optimizer to search for the optimal total resource and QoS-fairness-based allocator to allocate resource to tasks guaranteeing the fairness. These components adaptively achieve the optimal resource allocation formulated by a nonlinear optimization problem with the time complexity O(n) for the number of tasks n even if tasks' characteristics cannot be identified precisely. The simulation result shows that the rapidness of the convergence of the resource allocation to the optimal one is suitable for real-time systems with large number of tasks.

The frequency domain binaural model (FDBM) has been previously proposed to localize multiple sound sources. Since the method requires only two input signals and uses interaural phase and level differences caused by the diffraction generated by the head, flexibility in application is very high when the head is considered as an object. When an object is symmetric with respect to the two microphones, the performance of sound source localization is degraded, as a human being has front-back confusion due to the symmetry in a median plane. This paper proposes to reduce the degradation of performance on sound source localization by a combination of the microphone pair outputs using the FDBM. The proposed method is evaluated by applying to a security camera system, and the results showed performance improvement in sound source localization because of reducing the number of cones of confusion.

This letter reveals that linear lossy codes cannot attain the rate-distortion function in general, even if the source is binary i.i.d. and the distortion is measured by the Hamming distortion measure.

This letter proposes an efficient uplink scheduling algorithm for the voice over Internet protocol (VoIP) service with variable frame-duration according to the voice activity in IEEE 802.16e/m systems. The proposed algorithm dynamically changes the grant-interval to save the uplink bandwidth, and it uses the random access scheme when the voice activity changes from silent-period to talk-spurt. Numerical results show that the proposed algorithm can increase the VoIP capacity by 26 percent compared to the conventional extended real-time polling service (ertPS).

When the number of users in a service area increases in mobile multimedia services, no individual user can obtain satisfactory radio resources such as bandwidth and signal power because the resources are limited and shared. A solution for such a problem is user-position control. In the user-position control, the operator informs users of better communication areas (or spots) and navigates them to these positions. However, because of subjective costs caused by subjects moving from their original to a new position, they do not always attempt to move. To motivate users to contribute their resources in network services that require resource contributions for users, incentive-rewarding mechanisms have been proposed. However, there are no mechanisms that distribute rewards appropriately according to various subjective factors involving users. Furthermore, since the conventional mechanisms limit how rewards are paid, they are applicable only for the network service they targeted. In this paper, we propose a novel incentive-rewarding mechanism to solve these problems, using an external evaluator and interactive learning agents. We also investigated ways of appropriately controlling rewards based on user contributions and system service quality. We applied the proposed mechanism and reward control to the user-position control, and demonstrated its validity.

In this paper we show some new look at large deviation theorems from the viewpoint of the information-spectrum (IS) methods, which has been first exploited in information theory, and also demonstrate a new basic formula for the large deviation rate function in general, which is expressed as a pair of the lower and upper IS rate functions. In particular, we are interested in establishing the general large deviation rate functions that are derivable as the Fenchel-Legendre transform of the cumulant generating function. The final goal is to show, under some mild condition, a necessary and sufficient condition for the IS rate function to be derivable as the Fenchel-Legendre transform of the cumulant generating function, i.e., to be a rate function of Gartner-Ellis type.

A consensus problem has been studied in many fundamental and application fields to analyze coordinated behavior in multi-agent systems. In a consensus problem, it is usually assumed that a state of each agent is scalar and all agents have an identical linear consensus protocol. We present a consensus problem of multi-agent systems where each agent has multiple state variables and a performance value evaluated by a nonlinear performance function according to its current state. We derive sufficient conditions for agents to achieve consensus on the performance value using an algebraic graph theory and the mean value theorem. We also consider an application of a performance consensus problem to resource allocation in soft real-time systems so as to achieve a fair QoS (Quality of Service) level.