The tiled-display system has been used as a Computer Supported Cooperative Work (CSCW) environment, in which multiple local (and/or remote) participants cooperate using some shared applications whose outputs are displayed on a large-scale and high-resolution tiled-display, which is controlled by a cluster of PC's, one PC per display. In order to make the collaboration effective, each remote participant should be aware of all CSCW activities on the titled display system in real-time. This paper presents a capturing and delivering mechanism of all activities on titled-display system to remote participants in real-time. In the proposed mechanism, the screen images of all PC's are periodically captured and delivered to the Merging Server that maintains separate buffers to store the captured images from the PCs. The mechanism selects one tile image from each buffer, merges the images to make a screen shot of the whole tiled-display, clips a Region of Interest (ROI), compresses and streams it to remote participants in real-time. A technical challenge in the proposed mechanism is how to select a set of tile images, one from each buffer, for merging so that the tile images displayed at the same time on the tiled-display can be properly merged together. This paper presents three selection algorithms; a sequential selection algorithm, a capturing time based algorithm, and a capturing time and visual consistency based algorithm. It also proposes a mechanism of providing several virtual cameras on tiled-display system to remote participants by concurrently clipping several different ROI's from the same merged tiled-display images, and delivering them after compressing with video encoders requested by the remote participants. By interactively changing and resizing his/her own ROI, a remote participant can check the activities on the tiled-display effectively. Experiments on a 32 tiled-display system show that the proposed merging algorithm can build a tiled-display image stream synchronously, and the ROI-based clipping and delivering mechanism can provide individual views on the tiled-display system to multiple remote participants in real-time.

We propose a parametric packet-layer model for monitoring audio quality in multimedia streaming services such as Internet protocol television (IPTV). This model estimates audio quality of experience (QoE) on the basis of quality degradation due to coding and packet loss of an audio sequence. The input parameters of this model are audio bit rate, sampling rate, frame length, packet-loss frequency, and average burst length. Audio bit rate, packet-loss frequency, and average burst length are calculated from header information in received IP packets. For sampling rate, frame length, and audio codec type, the values or the names used in monitored services are input into this model directly. We performed a subjective listening test to examine the relationships between these input parameters and perceived audio quality. The codec used in this test was the Advanced Audio Codec-Low Complexity (AAC-LC), which is one of the international standards for audio coding. On the basis of the test results, we developed an audio quality evaluation model. The verification results indicate that audio quality estimated by the proposed model has a high correlation with perceived audio quality.

P2P (Peer-to-Peer) file sharing architectures have scalable and cost-effective features. Hence, the application of P2P architectures to media streaming is attractive and expected to be an alternative to the current video streaming using IP multicast or content delivery systems because the current systems require expensive network infrastructures and large scale centralized cache storage systems. In this paper, we investigate the P2P progressive download enabling Internet video streaming services. We demonstrated the capability of the P2P progressive download in both laboratory test network as well as in the Internet. Through the experiments, we clarified the contribution of the FTTH links to the P2P progressive download in the heterogeneous access networks consisting of FTTH and ADSL links. We analyzed the cause of some download performance degradation occurred in the experiment and discussed about the effective methods to provide the video streaming service using P2P progressive download in the current heterogeneous networks.

This paper describes a channel-adaptive packet scheduler for improved error control performance in a peer-cooperative distributed media streaming system. The proposed packet-scheduling algorithm was designed for the case in which streaming server peers rely on an error-recovery strategy using retransmission and application-layer automatic repeat request rather than error protection using forward error correction. The proposed scheduler can maximize retransmission opportunities and reduce the frame loss rate by using the observed channel status from each server peer. Simulation results show that the proposed algorithm enhances error-recovery performance in distributed multimedia streaming better than other schedulers.

Currently deployed mobile networks including High Speed Downlink Packet Access (HSDPA) offer only best-effort Quality of Service (QoS). In wireless best effort networks, the bandwidth variation is a critical problem, especially, for mobile devices with small buffers. This is because the bandwidth variation leads to packet losses caused by buffer overflow as well as picture freezing due to high transmission delay or buffer underflow. In this paper, in order to provide seamless video streaming over HSDPA, we propose an efficient real-time video streaming method that consists of the available bandwidth (AB) estimation for the HSDPA network and the transmission rate control to prevent buffer overflows/underflows. In the proposed method, the client estimates the AB and the estimated AB is fed back to the server through real-time transport control protocol (RTCP) packets. Then, the server adaptively adjusts the transmission rate according to the estimated AB and the buffer state obtained from the RTCP feedback information. Experimental results show that the proposed method achieves seamless video streaming over the HSDPA network providing higher video quality and lower transmission delay.

Bloom filters are widely used for various network applications. Because of the limited size of on-chip memory and the large volume of network traffic, Bloom filters are often required to update their contents incrementally. Two techniques have been used for this purpose: cold cache and double buffering. Cold cache outperforms double buffering in terms of the average cache ratio. However, double buffering works much better than cold cache for the worst-case cache hit ratio. In this paper, we propose a new updating scheme for Bloom filters, which updates the contents of Bloom filter incrementally while the assigned memory space is fully utilized. The proposed scheme works better than cold cache in terms of the average cache hit ratio. At the same time, it outperforms double buffering for the worst-case cache hit ratio.

In this letter, we propose a network-adaptive video streaming scheme based on cross-layered hop-by-hop video rate control in wireless multi-hop networks. We argue that existing end-to-end network-adaptive video rate control schemes, which utilize end-to-end statistics, exhibit serious performance degradation in severely interfered wireless network condition. To cope with this problem, in the proposed scheme, intermediate wireless nodes adjust video sending rate depending upon wireless channel condition measured at MAC (Medium Access Control) layer. Extensive experimental results from an IEEE 802.11a-based testbed show that the proposed scheme improves the perceptual video quality compared to an end-to-end scheme.

In this paper, we introduce layered low-density generator matrix (Layered-LDGM) codes for super high definition (SHD) scalable video systems. The layered-LDGM codes maintain the correspondence relationship of each layer from the encoder side to the decoder side. This resulting structure supports partial decoding. Furthermore, the proposed layered-LDGM codes create highly efficient forward error correcting (FEC) data by considering the relationship between each scalable component. Therefore, the proposed layered-LDGM codes raise the probability of restoring the important components. Simulations show that the proposed layered-LDGM codes offer better error resiliency than the existing method which creates FEC data for each scalable component independently. The proposed layered-LDGM codes support partial decoding and raise the probability of restoring the base component. These characteristics are very suitable for scalable video coding systems.

Delivering video streaming service over the Internet encounters some challenges. Two of them are heterogeneity of networks capacity and variability of video data rate. The capacity of network segments are constrained. Meanwhile, the rate of video data to be transmitted is highly variable in order to get near-constant images quality. Therefore, to send variable bit rate (VBR) video data over capacity-constrained network, its bit rate should be adjusted. In this paper a system to adjust the transmission bit rate of VBR MPEG video data called Transcoding-after-Smoothing (TaS), which is a combination of bit rate transcoding and bit rate smoothing algorithm, is proposed. The system smoothes out transmission rate of video data while at the same time also performs transcoding on some video frames when necessary in order to keep the transmission bit rate below a certain threshold value. Two kinds of TaS methods are proposed. One method does not have transcoding preference, while the other method uses frame type preference where an intra-coded frame is the last one that will be transcoded. These methods are implemented in our video server where a VBR video data is accessed by a client. Our experiment results show that the first TaS method yields significant reduction in the number of transcoded frames compared with the second TaS method and conventional frame-level transcoding. However, the second method performs better in minimizing the quality distortion.

In this paper, we propose a fault tolerant hybrid p2p-CDN video streaming arhitecture to overcome the problems caused by peer behavior in peer-to-peer (P2P) video streaming systems. Although there are several studies modeling and analytically investigating peer behaviors in P2P video streaming systems, they do not come up with a solution to guarantee the required Quality of the Services (QoS). Therefore, in this study a hybrid geographical location-time and interest based clustering algorithm is proposed to improve the success ratio and reduce the delivery time of required content. A Hybrid Fault Tolerant Video Streaming System (HFTS) over P2P networks conforming the required QoS and Fault Tolerance is also offered. The simulations indicate that the required QoS can be achieved in streaming video applications using the proposed hybrid approach.

In video streaming applications over the Internet, TCP-friendly rate control schemes are useful for improving network stability and inter-protocol fairness. However, they do not always guarantee a smooth video streaming. To simultaneously satisfy both the network and user requirements, video streaming applications should be quality-adaptive. In this paper, we propose a new quality adaptation mechanism to adjust the quality of congestion-controlled video stream by controlling the frame rate. Based on the current network condition, it controls the frame rate of video stream and the sending rate in a TCP-friendly manner. Through a simulation, we prove that our adaptation mechanism appropriately adjusts the quality of video stream while improving network stability.

This paper performs application-level QoS and user-level QoS assessment of audio-video streaming in cross-layer designed wireless ad hoc networks. In order to achieve high QoS at the user-level, we employ link quality-based routing in the network layer and media synchronization control in the application layer. We adopt three link quality-based routing protocols: OLSR-SS (Signal Strength), AODV-SS, and LQHR (Link Quality-Based Hybrid Routing). OLSR-SS is a proactive routing protocol, while AODV-SS is a reactive one. LQHR is a hybrid protocol, which is a combination of proactive and reactive routing protocols. For application-level QoS assessment, we performed computer simulation with ns-2 where an IEEE 802.11b mesh topology network with 24 nodes was assumed. We also assessed user-level QoS by a subjective experiment with 30 assessors. From the assessment results, we find AODV-SS the best for networks with long inter-node distances, while LQHR outperforms AODV-SS for short inter-node distances. In addition, we also examine characteristics of the three schemes with respect to the application-level QoS in random topology networks.

This letter proposes a QoS control method for video streaming service over wireless networks. Based on statistical analysis, the time-varying MAC parameters highly related to channel condition are selected to predict available bitrate. Adaptive bitrate control of scalably-encoded video guarantees continuity in streaming service even if the channel condition changes abruptly.

Delay jitter degrades the quality of delay-sensitive live media streaming. We investigate the use of multipath transmission with two paths to reduce delay jitter and, in this paper, propose a nearly equal delay path set configuration (NEED-PC) scheme that further improves the performance of the multipath delay jitter reduction method for delay-sensitive live media streaming. The NEED-PC scheme configures a pair of a maximally node-disjoint paths that have nearly equal path delays and satisfy a given delay constraint. The results of our simulation experiments show that path sets configured by the NEED-PC scheme exhibit better delay jitter reduction characteristics than a conventional scheme that chooses the shortest path as the primary path. We evaluate the performance of path sets configured by the NEED-PC scheme and find that the NEED-PC scheme reduces delay jitter when it is applied to a multipath delay jitter reduction method. We also investigate the trade-off between reduced delay jitter and the increased traffic load incurred by applying multipath transmission to more flows. The results show that the NEED-PC scheme is practically effective even if the amount of additional redundant traffic caused by using multipath transmission is taken into account.

This letter presents a novel frame splitting scheme for an error-robust audio streaming over packet-switching networks. In our approach to perceptual audio coding, an audio frame is split into several subframes based on the network configuration such that each packet can be decoded independently at the receiver. Through a subjective comparison category rating (CCR) test, it is discovered that our approach enhances the quality of the decoded audio signal under the lossy packet-switching networks environment.

With the rapid advances in wireless network communication, multimedia presentation has become more applicable. However, due to the limited wireless network resource and the mobility of Mobile Host (MH), QoS for wireless streaming is much more difficult to maintain. How to decrease Call Dropping Probability (CDP) in multimedia traffic while still keeping acceptable Call Block Probability (CBP) without sacrificing QoS has become an significant issue in providing wireless streaming services. In this paper, we propose a novel Dynamic Resources Adjustment (DRA) algorithm, which can dynamically borrow idle reserved resources in the serving cell or the target cell for handoffing MHs to compensate the shortage of bandwidth in media streaming. The experimental simulation results show that compared with traditional No Reservation (NR), and Resource Reservation in the six neighboring cells (RR-nb), and Resource Reservation in the target cell(RR-t), our proposed DRA algorithm can fully utilize unused reserved resources to effectively decrease the CDP while still keeping acceptable CBP with high bandwidth utilization.

In recent years, the inherent effectiveness of Peer-to-Peer (P2P) networks has been advocated to address scalability issues in large scale Internet-based on-Demand streaming services. Most of existing works adopt Cache-and-Relay (CR) scheme to exploit a cooperative paradigm among peers. In this paper, we mainly present our practical evaluation study of the scalability of the CR scheme by taking into account of more than 20,000,000 collected real traces. Based on trace-driven simulations, we conclude that the CR scheme is not as effective as previously reported in terms of saving server bandwidth.

This paper introduces an adaptive GOP structure (AGS), which adaptively defines the GOP structure according to the time-varying temporal properties of video sequences, and thus improves the coding efficiency of the MPEG & ITU-T's Joint Scalable Video Coding (JSVC) scheme, the method proposed in this paper, which adaptively modifies the size of GOP based on the image characteristics of video sequence, improves the coding efficiency up to 0.77 dB compared to the JSVC JSVM (Joint Scalable Video Model).

The purpose of this study is to maintain efficient backup routes for reconstructing overlay trees quickly. In most conventional methods, after a node leaves the trees, its child nodes start searching for the new parents. In this reactive approach, it takes a lot of time to find a new parent. In this paper, we propose a proactive approach to finding a next parent as the backup route node over the overlay tree before the current parent leaves. A proactive approach allows a node to find its new parent node immediately and switch to the backup route node smoothly. In our proposal, the structure of the overlay tree using a redundant degree can decide a backup route node without so much overhead. Simulations demonstrate our proactive approach can recover from node departures 2 times faster than reactive approaches, and can construct overlay trees with lower overheads than another proactive method. Additionally we carried out experiments over actual networks and their results support the effectiveness of our approach. We confirmed that our proposal achieved better streaming quality than conventional approaches.

Video transcoding technique is an efficient mechanism to deliver visual contents to a variety of users who have different network conditions or terminal devices with different display capabilities. In this paper, we propose two types of transcoding methods for adapting the bitrate of streaming video to the bandwidth of the transmission channel; spatial resolution reduction (SRR) transcoding and temporal resolution reduction (TRR) transcoding. The two transcoding methods are alternatively operated according to the requirements of users. Experimental results show that the proposed transcoding methods can preserve image quality while transcoding to the low bitrate.