The adequate evaluation of sound quality is an important issue for the lossy compression codecs, such as MP3. ITU-R Rec BS. 1387-1 (PEAQ – Perceptual Evaluation of Audio Quality) is the most widely used method to evaluate sound quality objectively. However, PEAQ can only be used for mono signals or two channel stereo signals, because it considers only timbral factors when assessing sound quality. This paper introduces an improved objective quality assessment method that can be used for mono signals and multichannel audio signals that considers both “spatial” and “timbral” factors. The “spatial” factors, which measure perceptual distortions in spatial impression, are important to evaluate the quality of multichannel sounds.

We analyze subjective assessments by comparative evaluations of bulk data transmission by using two psychological methods (the method of successive categories and the constant method). From the results of the first experiment, the thresholds at which participants downloading a data file began to feel dissatisfaction with service degradation are 15.7 Mbps and 11.6 Mbps obtained by the two different methods when the throughput without the service degradation is approximately 22 Mbps. In the second experiment, we investigate the threshold of user satisfaction for various network environments. The threshold is 63% of the throughput of the user's usual network environment. Moreover, from the viewpoint of download time, users feel dissatisfaction with the quality when download time becomes 1.5 times longer. These values can be used to more effectively allocate network resources and thereby achieve higher service quality.

Most existing works on resource allocation in cooperative OFDMA systems have focused on homogeneous users with same service and demand. In this paper, we focus on resource allocation in a service differentiated cooperative OFDMA system where each user has a different QoS requirement. We investigate joint power allocation, relay selection and subcarrier assignment to maximize overall system rates with consideration of QoS guarantees and service support. By introducing QoS price, this combinatorial problem with exponential complexity is converted into a convex one, and a two-level dual-primal decomposition based QoS-aware resource allocation (QARA) algorithm is proposed to tackle the problem. Simulation results reveal that our proposed algorithm significantly outperforms previous works in terms of both services support and QoS satisfaction.

Video applications such as video conferencing among multiple users and video surveillance systems require multiple video connections and QoS guarantee. These days the video systems equipped with IEEE 802.11 LAN interfaces allows a broadband wireless access to the Internet at a reasonable price. However, according to the current IEEE 802.11e HCCA standard, if more than two video sessions are to be established simultaneously, some of them must share the TXOP because the available number of TSIDs for video transmission is restricted to two. In order to resolve this problem, we devise a scheme which can establish up to 13 video sessions by slightly modifying the frame structure while maintaining the compatibility with the current standard. Our scheme is implemented on the NCTUns 4.0 network simulator, and evaluated numerically in terms of throughput, delay, and PSNR. Also real video clips are used as input to our simulation. The results showed that our scheme guarantees the transmission bandwidth requested by each video session.

When a link or node fails in a network, the affected flows are automatically rerouted. This increases the hop counts of the flows, which can drastically degrade network performance. Keeping the hop lengths as stable as possible, i.e., minimizing the difference in hop length between the original flow and the rerouted flow is important for network reliability. Therefore, network service providers need a method for designing networks that stabilizes the flow hop length and maintains connectivity during a link or node failure with limited investment cost. First, we formulate the network design problem used for determining the set of links to be added that satisfies the required constraints on flow hop length stability, connectivity, and node degree. Next, we prove that this problem is NP-complete and present two approximation algorithms for the optimization problem so as to minimize the number of links added. Evaluation of the performance of these algorithms by using 39 backbone networks of commercial ISPs and networks generated by two well-known models showed that the proposed algorithms provide effective solutions in sufficiently short computation time.

We develop a distance function for finite Chu spaces based on their behavior. Typical examples are given to show the coincidence between the distance function and intuition. We show by example that the triangle inequality should not be satisfied when it comes to comparing two processes.

Admission control is a procedure to guarantee a given level of Quality of Service (QoS) by accepting or rejecting arrival connection requests. There are many studies on backlog or loss rate evaluation formulas for admission control at a single node. However, there are few studies on end-to-end evaluation formulas suitable for admission control. In a previous paper, the authors proposed a new stochastic network calculus for many flows using an approach taken from large deviations techniques and obtained asymptotic end-to-end evaluation formulas for output burstiness and backlog. In this paper, we apply this stochastic network calculus to a heterogeneous tandem network with many forwarding flows and cross traffic flows constrained by leaky buckets, and obtain a simple evaluation formula for the end-to-end backlog. In this formula, the end-to-end backlog can be evaluated by the traffic load at the bottle neck node. This result leads us to a natural extension of the evaluation formula for a single node.

Bandwidth is an extremely valuable and scarce resource in multimedia networks. Therefore, efficient bandwidth management is necessary in order to provide high Quality of Service (QoS) to users. In this paper, a new QoS-aware bandwidth allocation algorithm is proposed for the efficient use of available bandwidth. By using the multi-objective optimization technique and Talmud allocation rule, the bandwidth is adaptively controlled to maximize network efficiency while ensuring QoS provisioning. In addition, we adopt the online feedback strategy to dynamically respond to current network conditions. With a simulation study, we demonstrate that the proposed algorithm can adaptively approximate an optimized solution under widely diverse traffic load intensities.

An interactive support system for image quality enhancement to adjust display equipments according to the user's own subjectivity is developed. Interactive support system for image quality enhancement enable the parameters based on user's preference to be derived by only selecting user's preference images without adjusting image quality parameters directly. In the interactive support system for image quality enhancement, the more the number of parameters is, the more effective this system is. In this paper, lightness, color and sharpness are used as the image quality parameters and the images are enhanced by increasing the number of parameters. Shape of tone curve is controlled by two image quality adjustment parameters for lightness enhancement. Images are enhanced using two image quality adjustment parameters for color enhancement. The two parameters are controlled in L* a* b* color space. Degree and coarseness of image sharpness enhancement are adjusted by controlling a radius of mask of smoothing filter and weight of adding. To confirm the effectiveness of the proposed method, the image quality and derivation time of the proposed method are compared with a manual adjustment method.

Quantity-based event reliability protocols have been proposed for reliable event detection in wireless sensor networks. They support the event reliability by achieving the desired number of data packets successfully transmitted from sensor nodes sensing an event to a sink by controlling the transport process. However, since many data collisions and buffer overflows frequently happen due to data congestions on limited data delivery paths from an event to a sink, the quantity-based event reliability protocols are hard to achieve the desired number due to lost data packets. Thus, this letter proposes a Quality-based Event Reliability Protocol (QERP) utilizing a property that the data packets from sensor nodes have different Contribution Degree (CD) values for event detection according to their environmental conditions. QERP selects sensor nodes to forward their data packets according to CD, and differentially transports the data packets by CD-based buffer management and load balancing.

This paper addresses the coexistence issue of distributed heterogeneous networks where the network nodes are cognitive radio terminals. These nodes, operating as secondary users (SUs), might interfere with primary users (PUs) who are licensed to use a given frequency band. Further, due to the lack of coordination and the dissimilarity of the radio access technologies (RATs) among these wireless nodes, they might interfere with each other. To solve this coexistence problem, we propose an architecture that enables coordination among the distributed nodes. The architecture provides coexistence solutions and sends reconfiguration commands to SU networks. As an example, time sharing is considered as a solution. Further, the time slot allocation ratios and transmit powers are parameters encapsulated in the reconfiguration commands. The performance of the proposed scheme is evaluated in terms of the coexistence between PUs and SUs, as well as the coexistence among SUs. The former addresses the interference from SUs to PUs, whereas the latter addresses the sharing of an identified spectrum opportunity among heterogeneous SU networks for achieving an efficient spectrum usage. In this study, we first introduce a new parameter named as quality of coexistence (QoC), which is defined as the ratio between the quality of SU transmissions and the negative interference to PUs. In this study we assume that the SUs have multiple antennas and employ fixed transmit power control (fixed-TPC). By using the approximation to the distribution of a weighted sum of chi-square random variables (RVs), we develop an analytical model for the time slot allocation among SU networks. Using this analytical model, we obtain the optimal time slot allocation ratios as well as transmit powers of the SU networks by maximizing the QoC. This leads to an efficient spectrum usage among SUs and a minimized negative influence to the PUs. Results show that in a particular scenario the QoC can be increased by 30%.

Multimedia applications such as video and audio have recently come into much wider use. Because this heterogeneous traffic consumes most of the network's resources, call admission control (CAC) is required to maintain high-quality services. User satisfaction depends on CAC's success in accommodating application flows. Conventional CACs do not take into consideration user satisfaction because their main purpose is to improve the utilization of resources. Moreover, if we assume a service where an ISP provides a "flat-based charging," each user may receive same user satisfaction as a result of users being accommodated in a network, even if each has a different bandwidth. Therefore, we propose a novel CAC to maximize total user satisfaction based on a new philosophy where heterog eneous traffic is treated equally in networks. Theoretical analysis is used to derive optimal thresholds for various traffic configurations with a full search system. We also carried out theoretical numerical analysis to demonstrate the effectiveness of our new CAC. Moreover, we propose a sub-optimal threshold configuration obtained by using an approximation formula to develop practical CAC from these observations. We tested and confirmed that performance could be improved by using sub-optimal parameters.

Duty-cycle MAC protocols have been proposed for wireless sensor networks (WSNs) to reduce the energy consumed by idle listening, but they introduce significant end-to-end delivery latency. Several works have attempted to mitigate this latency, but they still have a problem on handling the packet loss. The quality of the wireless channel in WSNs is quite bad, so packets are frequently lost. In this letter, we present a novel duty-cycle MAC protocol, called REMAC (Retransmission-Enhanced duty-cycle MAC), which exploits both the network layer and the physical layer information. REMAC estimates the quality of the wireless channel and properly reserves the wireless channel to handle the packet loss. It can reduce the end-to-end packet delivery latency caused by the packet loss without sacrificing the energy efficiency. Simulation results show that REMAC outperforms RMAC in terms of the end-to-end packet delivery latency.

An adaptive dimming technique controls both LCD panel transmittance and its backlight luminance adequately and locally according to the input TV signal. The technique reduces the power consumption and also improves the picture quality. However, a steep change in backlight luminance distribution due to the application of the technique causes image degradation around the boundary of the segments when the LCD is viewed from an angle. The main factor of image degradation is the illumination of a pixel by neighboring pixel's corresponding backlight when the LCD is viewed from an angle rather than normal direction. From the subjective evaluation of image quality and computer simulation, it is found that the gradient of the backlight luminance variation to luminance at the border of the segment should be less than 0.022 per pixel in order to suppress the image degradation.

This paper proposes a cross layer wireless VoIP service which integrates an Adaptive QoS Playout (AQP) algorithm, E-model, Stream Control Transmission Protocol (SCTP), IEEE 802.21 Media Independent Handover (MIH) middleware and two user motion detection services. The proposed AQP algorithm integrates the effect of playout control and lost packet retransmission based on the E-model. Besides, by using the partial reliable transmission service from SCTP and the handoff notification from MIH services in a cross layer manner, AQP can reduce the lateness loss rate and improve speech quality under high frame error rates. In the simulations, the performance of AQP is compared with a fixed playout algorithm and four adaptive playout strategies. The simulation results show that the lateness loss rate of AQP is 2% lower than that of existing playout algorithms and the R-factor is 16% higher than the compared algorithms when a network has 50 ms wired propagation delay and 2.5% frame error rate.

The distribution of streaming multicast and real time audio/video applications in the Internet has been quickly increased in the Internet. Commonly, these applications rarely use congestion control and do not fairly share provided network capacity with TCP-based applications such as HTTP, FTP and emails. Therefore, Internet communities will be threatened by the increase of non-TCP-based applications that likely cause a significant increase of traffics congestion and starvation. This paper proposes a set of mechanisms, such as providing various data rates, background traffics, and various scenarios, to act friendly with TCP when sending multicast traffics. By using 8 scenarios of simulations, we use 6 layered multicast transmissions with background traffic Pareto with the shape factor 1.5 to evaluate performance metrics such as throughput, delay/latency, jitter, TCP friendliness, packet loss ratio, and convergence time. Our study shows that non TCP traffics behave fairly and respectful of the co-existent TCP-based applications that run on shared link transmissions even with background traffic. Another result shows that the simulation has low values on throughput, vary in jitter (0-10 ms), and packet loss ratio > 3%. It was also difficult to reach convergence time quickly when involving only non TCP traffics.

In this paper, we study an opportunistic scheduling scheme for the TDMA wireless network with relay stations. We model the time-varying channel condition of a wireless link as a stochastic process. Based on this model, we formulate an optimization problem for the opportunistic scheduling scheme that maximizes the expected system throughput while satisfying the QoS constraint of each user. In the opportunistic scheduling scheme for the system without relay stations, each user has only one communication path between the base station and itself, and thus only user selection is considered. However, in our opportunistic scheduling scheme for the system with relay stations, since there may exist multiple paths between the base station and a user, not only user selection but also path selection for the scheduled user is considered. In addition, we also propose an opportunistic time-sharing method for time-slot sharing between base station and relay stations. With the opportunistic time-sharing method, our opportunistic scheduling provides opportunistic resource sharing in three places in the system: user selection in a time-slot, path selection for the selected user, and time-slot sharing between base station and relay stations. Simulation results show that as the number of places that opportunistic resource sharing is applied to increases, the performance improvement also increases.

In this letter, delay-dependent stability criterion for linear time-delay systems with multiple time varying delays is proposed by employing the Lyapunov-Krasovskii functional approach and integral inequality. By the N-segmentation of delay length, we obtain less conservative results on the delay bounds which guarantee the asymptotic stability of the linear time-delay systems with multiple time varying delays. Simulation results show that the proposed stability criteria are less conservative than several other existing criteria.

The use of image contrast enhancement has become increasingly essential due to the need to better show the visual information contained within the image for all vision-based systems. This has lead to motivation for the design of a powerful and accurate automatic contrast enhancement for a digital image. Histogram equalization is the most commonly used contrast enhancement method. However, the conventional histogram equalization methods usually result in excessive contrast enhancement, which causes the unnatural look and visual artifacts of the processed image. In this paper, we propose a novel histogram equalization method using the automatic histogram separation along with the piecewise transformed function. The contrast enhancement results of the proposed method were not only analyzed through qualitative visual inspection and for quantitative accuracy, but are also compared to the results of other state-of-the-art methods.

In differentiated services, packet classification is used to categorize incoming packets into multiple forwarding classes based on pre-defined filters and make information accessible for quality of service. Although numerous algorithms have presented novel data structures to improve the search performance of packet classification, the performance of these algorithms are usually limited by the characteristics of filter databases. In this paper, we use a different approach of filter preprocessing to enhance the search performance of packet classification. Before generating the searchable data structures, we cluster filters in a bottom-up manner. The procedure of the filter clustering merges filters with high degrees of similarity. The experimental results show that the technique of filter clustering could significantly improve the search performance of Pruned Tuple Space Search, a notable hash-based algorithm. As compared to the prominent existing algorithms, our enhanced Pruned Tuple Space Search also has superior performance in terms of speed and space.