In this paper, we consider Peer-to-Peer Video-on-Demand (P2P VoD) systems based on the BitTorrent file sharing protocol. Since the Rarest First policy adopted in the original BitTorrent protocol frequently fails to collect pieces corresponding to a video file by their playback time, we need to develop a new piece selection rule particularly designed for P2P VoDs. In the proposed scheme, we assume the existence of a media server which can upload any piece upon request, and try to bound the load of such media server with two techniques. The first technique is to estimate pieces which are not held by any peer and prefetch them from the media server. The second technique is to switch the mode of each peer according to the estimated size of the P2P network. The performance of the proposed scheme is evaluated by simulation.

Recently, the need for multimedia devices, such as mobile phones, digital TV, PMP, digital camcorders, digital cameras has increased. These devices provide various services for multimedia file manipulation, allowing multimedia contents playback, multimedia file editing, etc. Additionally, digital TV provides a recorded multimedia file copy to a portable USB disk. However, Linux Ext3 file system, as employed by these devices, has a lot of drawbacks, as it required a considerable amount of time and disk I/Os to store large-size edited multimedia files, and it is hard to access for typical PC users. Therefore, in this paper a design and implementation of an amortized Ext3 with FWAE (Fast Writing-After-Editing) for WinXP-based multimedia applications is described. The FWAE is a fast and efficient multimedia file editing/storing technique for the Ext3 that exploits inode block pointer re-setting and shared data blocks by simply modifying metadata information. Individual experiments in this research show that the amortized Ext3 with FWAE for WinXP not only dramatically improves written performance of the Ext3 by 16 times on average with various types of edited multimedia files but also notably reduces the amount of consumed disk space through data block sharing. Also, it provides ease and comfort to use for typical PC users unfamiliar with Linux OS.

In this letter, we propose an effective method to retrieve images from a 2D shape image database using discrete shock graphs combined with an adaptive selection algorithm. Experimental results show that our method is more accurate and fast than conventional approaches and reduces computational complexity.

Kernel biased discriminant analysis (KBDA), as a subspace learning algorithm, has been an attractive approach for the relevance feedback in content-based image retrieval. Its performance, however, still suffers from the “small sample learning” problem and “kernel learning” problem. Aiming to solve these problems, in this paper, we present a new semi-supervised scheme of KBDA (S-KBDA), in which the projection learning and the “kernel learning” are interweaved into a constrained optimization framework. Specifically, S-KBDA learns a subspace that preserves both the biased discriminant structure among the labeled samples, and the geometric structure among all training samples. In kernel optimization, we directly optimize the kernel matrix, rather than a kernel function, which makes the kernel learning more flexible and appropriate for the retrieval task. To solve the constrained optimization problem, a fast algorithm based on gradient ascent is developed. The image retrieval experiments are given to show the effectiveness of the S-KBDA scheme in comparison with the original KBDA, and the other two state-of-the-art algorithms.

In this letter, we propose an effective preamble based on constant amplitude and zero auto-correlation (CAZAC) sequence for multi-input multi-output (MIMO) and cooperative WiMedia ultra-wideband (UWB) systems. The proposed preamble even provides better single-channel estimation performance than the preamble specified in the standard in severe UWB channel model. The effectiveness of the proposed design is confirmed through the mean square error (MSE) performance.

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.

In our previous project, an XML-based authoring tool was provided for teachers to script multimedia teaching material with animated agents, and a stand-alone learning system was designed for students to display the material and interact with animated agents. We also provided evidence that the authoring tool and learning system in computer-assisted learning systems successfully enhances learning performance. The aim of this study is to continue the previous project, to develop a Web-based multimedia learning system that presents materials and an animated agent on a Web browser. The Web-based multimedia learning system can provide an opportunity for students to engage in independent learning or review of school course work. In order to demonstrate the efficiency of this learning system, it was applied to one elementary school. An experimental material, `Road Traffic Safety', was presented in two learning systems: a Web-based PowerPoint learning system and a Web-based multimedia learning system. The experiment was carried out in two classes that had a total of thirty-one 3rd-grade students. The results suggest that using our authoring tool in a Web-based learning system can improve learning, and in particular, enhance learners' problem-solving ability. Students with higher achievement on the post-test showed better comprehension in problem-solving questions. Furthermore, the feedback from the questionnaire surveys show students' learning interest can be fostered when an animated agent is integrated into multimedia teaching materials, and that students prefer to adopt the Web-based multimedia learning system for independent learning after school.

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.

This paper introduces a cycle-accurate Simulator for a dynamically REconfigurable MUlti-media System, called SimREMUS. SimREMUS can either be used at transaction-level, which allows the modeling and simulation of higher-level hardware and embedded software, or at register transfer level, if the dynamic system behavior is desired to be observed at signal level. Trade-offs among a set of criteria that are frequently used to characterize the design of a reconfigurable computing system, such as granularity, programmability, configurability as well as architecture of processing elements and route modules etc., can be quickly evaluated. Moreover, a complete tool chain for SimREMUS, including compiler and debugger, is developed. SimREMUS could simulate 270 k cycles per second for million gates SoC (System-on-a-Chip) and produced one H.264 1080p frame in 15 minutes, which might cost days on VCS (platform: CPU: E5200@ 2.5 Ghz, RAM: 2.0 GB). Simulation showed that 1080p@30 fps of H.264 High Profile@ Level 4 can be achieved when exploiting a 200 MHz working frequency on the VLSI architecture of REMUS.

In distributed spectrum sensing, spatially distributed sensors perform radio frequency (RF) sensing and forward the result to a fusion center (FC). Cognitive radio (CR) obtains spectral information from the FC. Distributed spectrum sensing facilitates reliable discovery of spectrum opportunity while providing enhanced protection to legacy systems. The overall performance of distributed spectrum sensing depends both on the quality of sensing at the individual sensors and the forwarding scheme from the individual sensors. In this aspect the choice of media access control (MAC) plays a significant role. We can improve the system performance by optimizing the MAC and the spectrum sensing parameters jointly. In this paper we propose an enhanced MAC scheme based on existing scheduled MAC protocols to yield a high performance distributed spectrum sensing. To demonstrate our idea, we provide computer simulation by considering energy detection based distributed spectrum sensors and IEEE 802.15.4 PHY and MAC parameters.

This paper proposes approaches to perform HW/SW (Hardware/Software) partition and parallelization of computing-intensive tasks of the H.264 HiP (High Profile) decoding algorithm on an embedded coarse-grained reconfigurable multimedia system, called REMUS (REconfigurable MUltimedia System). Several techniques, such as MB (Macro-Block) based parallelization, unfixed sub-block operation etc., are utilized to speed up the decoding process, satisfying the requirements of real-time and high quality H.264 applications. Tests show that the execution performance of MC (Motion Compensation), deblocking, and IDCT-IQ (Inverse Discrete Cosine Transform-Inverse Quantization) on REMUS is improved by 60%, 73%, 88.5% in the typical case and 60%, 69%, 88.5% in the worst case, respectively compared with that on XPP PACT (a commercial reconfigurable processor). Compared with ASIC solutions, the performance of MC is improved by 70%, 74% in the typical and in the worst case, respectively, while those of Deblocking remain the same. As for IDCT_IQ, the performance is improved by 17% no matter in the typical or worst case. Relying on the proposed techniques, 1080p@30 fps of H.264 HiP@ Level 4 decoding could be achieved on REMUS when utilizing a 200 MHz working frequency.

In this paper, we propose a new motion vector smoothing algorithm using weighted vector median filtering based on edge direction for frame interpolation. The proposed WVM (Weighted Vector Median) system adjusts the weighting values based on edge direction, which is derived from spatial coherence between the edge direction continuity of a moving object and motion vector (MV) reliability. The edge based weighting scheme removes the effect of outliers and irregular MVs from the MV smoothing process. Simulation results show that the proposed algorithm can correct wrong motion vectors and thus improve both the subjective and objective visual quality compared with conventional methods.

Motivated by the needs of modern agriculture, in this paper we present CropNET, a wireless multimedia sensor network system for agriculture monitoring. Both hardware and software designs of CropNET are tailored for sensing in wide farmland without human supervision. We have carried out multiple rounds of deployments. The evaluation results show that CropNET performs well and facilitates modern agriculture.

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.

The recent growth in available network bandwidth envisions the wide-spread use of broadband applications such as uncompressed HD-SDI (High-definition serial digital interface) over IP. These cutting-edge applications are also driving the development of a media-oriented infrastructure for networked collaboration. This paper introduces imCast, a high-quality digital media platform dealing with uncompressed HD-SDI over IP, and discusses its internal architecture in depth. imCast mainly provides cost-effective hardware-based approaches for high-quality media acquisition and presentation; flexible software-based approaches for presentation; and allows for economical network transmission. Experimental results (taken over best-effort IP networks) will demonstrate the functional feasibility and performance of imCast.

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.

This paper considers a new reactive fast handover MIPv6 (FMIPv6) mechanism to minimize packet loss of the existing mechanism. The primary idea of the proposed reactive FMIPv6 mechanism is that the serving access router buffers packets toward the mobile node (MN) as soon as the link layer between MN and serving base station is disconnected. To implement the proposed mechanism, the router discovery message exchanged between MN and serving access router is extended. In addition, the IEEE 802.21 Media Independent Handover Function event service message is defined newly. Through analytic performance evaluation and experiments, the proposed reactive FMIPv6 mechanism can be shown to minimize packet loss much than the existing mechanism.

UHF band passive RFID systems are being steadily adopted by industries because of their capability of long range automatic identification with passive tags. For an application which demands a large number of readers located in a limited geographical area, referred to as dense reader mode, interference rejection among readers is important. The coding method, baseband or subcarrier coding, in the tag-to-reader communication link results in a significant influence on the interference rejection performance. This paper examines the frequency sharing of baseband and subcarrier coding UHF RFID systems from the perspective of their transmission delay using a media access control (MAC) simulator. The validity of the numerical simulation was verified by an experiment. It is revealed that, in a mixed operation of baseband and subcarrier systems, assigning as many channels as possible to baseband system unless they do not exploit the subcarrier channels is the general principle for efficient frequency sharing. This frequency sharing principle is effective both to baseband and subcarrier coding systems. Otherwise, mixed operation fundamentally increases the transmission delay in subcarrier coding systems.

Proxy Mobile IPv6 (PMIPv6) has been proposed in order to overcome the limitations of host-based mobility management in IPv6 networks. However, packet losses during doing handover are still a problem. To solve this issue, several schemes have been developed, and can be classified into two approaches: predictive and reactive handover. Both approaches commonly use bi-directional tunnel between mobile access gateways (MAGs). In predictive schemes especially, mobility support for a mobile node (MN) is triggered by simplified link signal strength. Thereafter, the MN sends handover notification to its serving MAG, and is then able to initiate packet forwarding. Therefore, if the MN moves toward an unexpected MAG that does not have any pre-established tunnel with the serving MAG, it may lead to packet losses. In this paper, we define this problem as Early Packet Forwarding (EPF). As a solution, we propose an enhanced PMIPv6 scheme using two-phase tunnel control based on the IEEE 802.21 Media Independent Handover (MIH).

In the latest video coding frameworks, efficiency of motion vector (MV) coding is becoming increasingly important because of the growing bit rate portion of motion information. However, neither the conventional median predictor, nor the newer schemes such as the minimum bit rate prediction scheme and the hybrid scheme, can effectively eliminate the local redundancy of motion vectors. In this paper, we present the prioritized reference decision scheme for efficient motion vector coding, based on the H.264/AVC framework. This scheme makes use of a boolean indicator to specify whether the median predictor is to be used for the current MV or not. If not, the median prediction is considered not suitable for the current MV, and this information is used for refining the possible space of a group of reference MVs including 4 neighboring MVs and the zero MV. This group of MVs is organized to be a prioritized list so that the reference MV with highest priority is to be selected as the prediction value. Furthermore, the boolean indicators are coded into the modified code words of mb_type and sub_mb_type, so as to reduce the overhead. By applying the proposed scheme, the structure and the applicability problems with the state-of-the-art MBP scheme have been overcome. Experimental result shows that the proposed scheme achieves a considerable reduction of bits for MVDs, compared with the conventional median prediction algorithm. It also achieves a better and much stabler performance than MBP-based MV coding.