Recently, voice over WLAN has become an attractive service, and it is expected to be the most popular application in the near future due to its low cost and easy deployment. It has been reported that there occurs unfairness between downlink and uplink in the 802.11 WLAN. This is mainly caused by CSMA/CA employed in DCF. All stations including an AP fairly compete for shared wireless medium. Thus, in particular, the unfairness has an adverse impact on bi-directional voice calls. Downlink voice connections become a primary factor to limit voice capacity. In this paper, we propose a novel medium access protocol, so called DCFmm, in order to improve QoS of downlink voice traffic as well as fairness between bi-directional voice connections. DCFmm is designed to enhance 802.11 DCF, and is fully compatible with the legacy DCF. In addition, it requires only protocol modifications of an AP. Thus, it can be easily implemented into existing 802.11 WLANs. DCFmm is compared with two conventional techniques through computer simulations. Extensive simulation results show that the proposed DCFmm can improve fairness between downlink and uplink, and consequently, support larger number of voice calls than DCF.

Recently, network coding (NC) has been popularly applied to wireless networks in order to improve scarce wireless capacity. In wireless LANs, NC can be applied to packet retransmission, and a base station can simultaneously retransmit multiple packets destined to different wireless stations by a single retransmission trial. On the other hand, NC creates additional packet delay at both base station and wireless stations, and hence, packet transfer delay may increase seriously. However, existing NC-based retransmission methods do not consider this additional delay explicitly. In addition, when the number of flows is small, NC exhibits less benefit because the chances of NC-based retransmission are highly reduced. Therefore, in this paper, we propose a novel NC-based retransmission method in order to improve packet transfer delay and jitter of received packets. Moreover, to achieve further improvement of delay, jitter and retransmission efficiency even when there exist a small number of traffic flows, we propose a retransmission method in which NC-based retransmission cooperates with the typical ARQ method. We overcome the disadvantage of NC-based retransmission by combining with ARQ cooperatively. Finally, we show the effectiveness of the proposed methods by extensive computer simulation.

To meet the bandwidth requirements of multimedia services, multipath transmission is a promising solution. In this paper, we consider multi-access networks, where WiMAX and WiFi links are set up at the same time. Multipath transmission suffers from the intrinsic problem of out-of-order packet delivery. This has an adverse impact on TCP and even UDP-based delay sensitive applications. However, multimedia streaming services allow some tolerance to transmission delay. Motivated by this observation, we investigate how to split multimedia flows over heterogeneous links. Wireless link capacity varies widely over time due to dynamic radio conditions. The capacity variations should be promptly reflected in traffic splitting in order to accomplish an equal load-balance. A practical prototype system has been implemented. We have performed extensive measurements from a prototype system. Through practical experimental results, we could verify two major research goals. One is that multimedia splitting can improve the overall network performance (e.g., the permitted multimedia sessions or the aggregated bandwidth) while still keeping an acceptable media quality. The other is an adaptation capability to varying link quality. It has been widely investigated under various radio conditions and different monitoring intervals. It is shown that the adaptive technique is effective under dynamic radio environments.

This letter proposes a novel post-processing method for self-similarity based super-resolution (SR). Existing back-projection (BP) methods enhance SR images by refining the reconstructed coarse high-frequency (HF) information. However, it causes artifacts due to interpolation and excessively smoothes small HF signals, particularly in texture regions. Motivated by these observations, we propose a novel post-processing method referred to as middle-frequency (MF) based refinement. The proposed method refines the reconstructed HF information in the MF domain rather than in the spatial domain, as in BP. In addition, it does not require an internal interpolation process, so it is free from the side-effects of interpolation. Experimental results show that the proposed algorithm provides superior performance in terms of both the quantity of reproduced HF information and the visual quality.

With the appearance of large OLED panels, the OLED TV industry has experienced significant growth. However, this technology is still in the early stages of commercialization, and some technical challenges remain to be overcome. During the development phase of a product, power consumption is one of the most important considerations. To reduce power consumption in OLED displays, we propose a method based on just-noticeable difference (JND). JND refers to the minimum visibility threshold when visual content is altered and results from physiological and psychophysical phenomena in the human visual system (HVS). A JND model suitable for OLED displays is derived from numerous experiments with OLED displays. With the use of JND, it is possible to reduce power consumption while minimizing perceptual image quality degradation.

Recently, network coding (NC) has been popularly applied to wireless networks in order to improve scarce wireless capacity. In wireless LANs, NC can be applied to packet retransmission in which multiple packets can be simultaneously transmitted by a single transmission trial at a base station (BS). In this paper, we assume wireless LANs with QoS functionality and propose adaptive cooperation between NC and IEEE 802.11e EDCA. In EDCA, when network load is high, QoS is significantly degraded even for high priority class. To solve this, existing methods improve backoff control, and decrease packet loss caused by collision. However, this cannot prevent packet loss caused by channel fading. In the proposed cooperation between NC and EDCA, QoS of all priority classes is improved from the aspect of efficient loss recovery. Unlike NC method with no QoS control, we encounter transmission scheduling problem among an NC packet, a single lost packet and a new packet. Moreover, in the constitution of packets encoded into NC packet, packet's intrinsic priority should be considered. Therefore, we propose how to schedule the packets to be transmitted in BS, and how to constitute NC packets to be encoded. Finally, we show the effectiveness of the proposed method by extensive computer simulations.

In wireless networks, the mechanism to adaptively select a link transmission rate based on channel variations is referred to as RA (rate adaptation). The operation may have a critical impact on the upper-layer application, specifically video streaming which has strict QoS requirements. Thus, RA should consider the QoS requirements and radio conditions at the same time. In this paper, we present a CV-RA (cross-layer video-oriented rate adaptation) scheme for video transmission over multi-rate wireless networks. The transmission rate is switched in a cross-layer optimized way, by simultaneously considering video R-D (rate-distortion) characteristics as well as wireless conditions. At the radio link layer, transmission rate selection is made using cross-layer optimization. As a result of RA, the effective link throughput dynamically changes. At the application layer, video source rate is adaptively controlled using cross-layer adaptation. CV-RA is compared to three traditional RA schemes. It can realize the highest possible visual communications for any channel condition. For the previous schemes, the variations of visual quality is high due to dynamic packet error rates. In contrast, for CV-RA, visual quality improves with the channel condition.

In packet networks including the Internet and commercial 3G wireless bearers, the network states that a streaming media application experiences are not known a priori and exhibit time-varying characteristics. For such dynamic environments, network-adaptive techniques are essential to efficiently deliver video data. In this paper, we propose a frame-based optimal scheduling algorithm which incorporates a MAP (Maximum A Posteriori) framework for adapting to varying network loss rate. The optimal transmission schedule is determined such that effective frame-rate is maximized at playback. Also, for multiple packets per frame, frame-based selection of delivery order greatly reduces computational complexity for a server scheduler when compared with packet-based scheduling techniques. In addition, by dynamically estimating instantaneous packet loss probability, the proposed scheduler performs network-adaptive transmission for streaming video over time-varying packet networks. Simulation results for test video sequence show that the proposed scheduling algorithm outperforms conventional ARQ-based schemes from a view point of reconstructed video quality as well as playable frame-rate. In particular, the proposed scheduling algorithm exhibits significant improvements of frame-rate over highly lossy channels.

In the self-similarity super resolution (SR) approach, similar examples are searched across down-scales in the image pyramid, and the computations of searching similar examples are very heavy. This makes it difficult to work in a real-time way under common software implementation. Therefore, the search process should be further accelerated at an algorithm level. Cauchy-Schwarz inequality has been used previously for fast vector quantization (VQ) encoding. The candidate patches in the search region of SR are analogous to the code-words in the VQ, and Cauchy-Schwarz inequality is exploited to exclude implausible candidate patches early. Consequently, significant acceleration of the similar patch search process is achieved. The proposed method can easily make an optimal trade-off between running speed and visual quality by appropriately configuring the bypass-threshold.

Recently, wireless LAN is achieving remarkable growth and maturity. On the other hand, by the advance of the Internet, the demand for multimedia communication services which include video and voice will be expected to grow. Therefore, in the future, the mechanism of QoS guarantee must be realized even in wireless LAN environment. So far, IEEE 802.11e EDCF has been proposed, which is a contention based channel access method to achieve the QoS guarantee in wireless LAN. However, this cannot realize the desired throughput ratio or deterministic target throughput in principle. In this paper, we expand the EDCF to solve such QoS issues and enable more flexible QoS control. Moreover, we show the effectiveness of our proposal by computer simulation.

In this paper, we address adaptive link switching over heterogeneous wireless access networks including IEEE 802.11. When an IEEE 802.11 link is congested, the transmission link of a terminal with multi-RATs (radio access technologies) is switched to another radio access systems. To this end, we propose link-level metrics of LC (link cost) and AC (access cost) for quantifying TCP congestion over IEEE 802.11 networks. The proposed metric can be easily measured at a local wireless terminal, and that enables each multi-RAT terminal to work in a distributed way. Through various indoor and outdoor experiments using a test-bed system, we verify that the proposed link level metrics are good indicators of TCP traffic congestion. Experimental results show that the proposed metrics can detect congestion occurrence quickly, and avoid the TCP throughput degradation of other neighboring terminals, when they are used for transmission link switching.

IEEE 802.11 DCF is a contention-based channel access protocol, and medium access delay greatly increases as the number of contending stations in a service area increases. This severely affects on delay-sensitive video applications. In this paper, we focus on MAC-layer solutions for realizing high quality video transmission in 802.11 DCF networks. A new channel access protocol based on the timestamp (TS) of video packets is proposed for real-time video. The TS information is carried by RTP header from the video application to 802.11 MAC. For video packets with the same RTP TS, they are simultaneously transmitted in a single channel access. Additional contention and back-off processes can be avoided because the whole packets of a video frame are completely delivered. The proposed TS-based access protocol can be easily implemented by the DCF with packet bursting. In addition, it is backward compatible to the legacy DCF. Extensive simulations show that the TS-based channel access achieves lower cumulative distributions of application-level video frame delay when compared to the DCF protocol.

Support Vector Machine (SVM) is one of the most widely used classifiers to categorize observations. This classifier deterministically selects a class that has the largest score for a classification output. In this letter, we propose a multiclass probabilistic classification method that reflects the degree of confidence. We apply the proposed method to age group classification and verify the performance.