Ce-Kuen Shieh

An Evaluation Framework for More Realistic Simulations of MPEG Video Transmission

We present a novel and complete tool-set for evaluating the delivery quality of MPEG video transmissions in simulations of a network environment. This tool-set is based on the EvalVid framework. We extend the connecting interfaces of EvalVid to replace its simple error simulation model by a more general network simulator like NS2. With this combination, researchers and practitioners in general can analyze through simulation the performance of real video streams, i.e. taking into account the video semantics, under a large range of network scenarios. To demonstrate the usefulness of our new tool-set, we point out that it enables the investigation of the relationship between two popular objective metrics for Quality of Service (QoS) assessment of video quality delivery: the PSNR (Peak Signal to Noise Ratio) and the fraction of decodable frames. The results show that the fraction of decodable frames reflects well the behavior of the PSNR metric, while being less time-consuming. Therefore, the fraction of decodable frames can be an alternative metric to objectively assess through simulations the delivery quality of transmission in a network of publicly available video trace files.

A novel realistic simulation tool for video transmission over wireless network

The objective of this paper is to present a novel realistic simulation tool-set for evaluating video delivered quality over wireless network. This tool-set integrates EvalVid and NS-2. With the integration, researchers can easily analyze their designed mechanisms, such as network protocols or QoS control schemes in a realistic simulation environment. We used a case study about video transmission over wireless network to demonstrate the simulation with the tool-set. From the results, we found that the quality of video transmission with burst packet errors is superior to the random distributed packet errors in the same packet error rate. In addition, in the same packet error rate, unicast transmission leads to better video delivered quality than multicast transmission because of retransmission. Also, using the tool-set researchers can assess video quality not only with the evaluation metrics, but also with real video sequences. In brief, researchers who utilize our proposed QoS assessment framework will be benefited in verifying their designs regarding video transmission over wireless network

An adaptive cross-layer mapping algorithm for MPEG-4 video transmission over IEEE 802.11e WLAN

This paper proposes an adaptive cross-layer mapping algorithm to improve the transmission quality of MPEG-4 video stream over an IEEE 802.11e wireless network. Instead of classifying video data to a specific access category in an 802.11e network, we propose an algorithm that dynamically maps MPEG-4 video packets to appropriate access categories according to both the significance of the video data and the network traffic load. Our proposed cross-layer architecture passes information about the significance of video packets from the application layer to the media access control layer. The queue length of a specific access category is used to deduce the network traffic load. We conducted a performance evaluation of our proposed cross-layer approach under both light and heavily loaded network conditions. Our simulation results demonstrate: (a) superior performance of our proposed approach (under both light and heavy loads) over 802.11e (Enhanced Distributed Channel Access (EDCA) and static mapping schemes, (b) not only guarantees prioritized transmission of essential video data but also provides efficient queue length utilization.

The Packet Loss Effect on MPEG Video Transmission in Wireless Networks

The purpose of this paper is to study the packet loss effect on MPEG video transmission quality in wireless networks. First, we consider the distribution of packet losses in wireless network, including distributed and burst packet losses. Besides, we also discuss the additional packet drops by the play-out buffer at the received end, and the effect of the transmission packet size on the video delivered quality. From the results, we find that the effect of burst packet losses on the video delivered quality is less than distributed packet losses in the same packet loss rate. Moreover, the smaller size of the play-out buffer leads to more packet drops and worse video quality. Finally, if there is no video recovery for video transmission, the video delivered quality of the larger packet size will be better than smaller packet size

Multi-path transmission control scheme combining bandwidth aggregation and packet scheduling for real-time streaming in multi-path environment

These days, a wide variety of wireless interfaces are available to connect to Internet. When coverage area of these different technologies overlap, receiver equipped with multiple interfaces can use them simultaneously to improve the performance of its applications in terms of bandwidth rely on bandwidth aggregation. However, specific conditions such as bottleneck bandwidth and end-to-end delay need to be accounted for before using such techniques. If this problem of end-to-end delay and bottleneck bandwidth are not properly addressed, there may be many packets along multiple paths which can arrive late and can lead to a large number of out-of-order packets at the receiver, which can eventually cause serious degradation of video quality at the receiver. For this reason, in this study, the authors propose a multi-path transmission control scheme (MTCS) combining bandwidth aggregation and packet scheduling for real-time streaming in a multi-path environment. In a bandwidth aggregation scheme, the authors propose a mathematical model to find the transmission rate over each path in order to obtain the optimal total throughput. However, the end-to-end delay of each path is not the same. The out-of-order packets problem will become serious in a multi-path environment. Therefore the authors propose a packet scheduling scheme to arrange the transmission sequence in order to effectively minimise the impact of packet reordering at the receiver. Our proposed control scheme not only aggregates the available bandwidth of multiple paths, but also reduces the time of packet reordering at the receiver. Experimental results show with our proposed scheme, the authors not only obtain the optimal transmission throughput but also reduce packet reordering delays under varying drop and delay conditions caused by the underlying network.

A two markers system for improved MPEG video delivery in a DiffServ network

The failure of legacy packet markers such as the Single Rate Three Color Marker (SRTCM) and the Two Rate Three Color Marker (TRTCM) to distinguish important and less important data renders them unsuitable for MPEG video transmission in a DiffServ network. Therefore, we propose a Two Markers System (TMS) to improve the delivery quality of MPEG video streams. We use authentic MPEG4 video traffic traces to compare the performance of the proposed TMS system with those of legacy packet markers. Results show that TMS outperforms the legacy traffic markers in terms of the quality of the delivered MPEG video streams.

A RED-FEC Mechanism for Video Transmission Over WLANs

This paper proposes a Random Early Detection Forward Error Correction (RED-FEC) mechanism to improve the quality of video delivered over Wireless Local Area Networks (WLANs). In contrast to previous FEC schemes, in which the rate determination information is fed back from the receiver side, in our proposed method, the redundancy rate is calculated directly at the wireless Access Point (AP) in accordance with the network traffic load, as indicated by the AP queue length. An analytical model is developed to predict the effective packet loss rate of a video stream delivered over a WLAN with RED-FEC protection. The numerical results show that the proposed RED-FEC mechanism consistently achieves higher recovery efficiency than a conventional FEC scheme under high and low network loading conditions.

xAn Enhanced Adaptive FEC Mechanism for Video Delivery over Wireless Networks

This paper proposes an enhanced adaptive FEC (EAFEC) algorithm implemented in the access point (AP) to improve video delivery over wireless networks. Unlike traditional static FEC mechanisms, which add redundant data to transmission data in a fixed number, the number of redundant FEC data for the EAFEC mechanism is determined by the AP, which is based on both network traffic load and wireless channel state. The algorithm is based on two factors. One of them is the queue length in the access point, indicating network traffic load; the second factor is packet retransmission times, indicating wireless channel state. EAFEC usage would tune FEC data number generated in such a way that it would not unnecessarily add more traffic to the network in already congested scenario. Our work shows that the EAFEC algorithm improves system performance by dynamically tuning FEC strength to the current loss in the wireless channel. We use NS-2 simulation experiments to prove our algorithm

A Theoretical Framework for End-to-End Video Quality Prediction of MPEG-based Sequences

This paper presents a novel theoretical framework for end-to-end video quality prediction of MPEG-based video sequences. The proposed framework encloses two discrete models: (i) A model for predicting the video quality of an encoded signal at a pre-encoding stage and (ii) A model for mapping QoS-sensitive network parameters (i.e. packet loss) to video quality degradation. The efficiency of both the discrete models is experimentally validated, proving by this way the accuracy of the proposed framework.

Sub-packet forward error correction mechanism for video streaming over wireless networks

Traditional Forward Error Correction (FEC) mechanisms can be divided into Packet level FEC (PFEC) mechanisms and Byte level FEC (BFEC) mechanisms. The PFEC mechanism of recovering from errors in a source packet requires an entire FEC redundant packet even though the error involves a few bit errors. The recovery capability of the BFEC mechanism is only half of the FEC redundancy. Accordingly, an adaptive Sub-Packet FEC (SPFEC) mechanism is proposed in this paper to improve the quality of video streaming data over wireless networks, simultaneously enhancing the recovery performance and reducing the end-to-end delay jitter. The SPFEC mechanism divides a packet into n sub-packets by means of the concept of a virtual packet. The SPFEC mechanism uses a checksum in each sub-packet to identify the position of the error sub-packet. Simulation experiments show the adaptive SPFEC mechanism achieves high recovery performance and low end-to-end delay jitter. The SPFEC mechanism outperforms traditional FEC mechanism in terms of packet loss rate and video Peak Signal-to-Noise Ratio (PSNR). SPFEC offers an alternative for improved efficiency video streaming that will be of interest to the designers of the next generation environments.