Video Frame Loss Research Articles

Cooperative flow regulation protocol for real-time and non-real-time applications over satellite network

In current times, the intensity of Internet usage has been on a relentless rise, in pace with the release of numerous real-time applications like video transmission and non-real-time applications. Due to high error channels, the end-users will observe poor quality video over satellite network. The applications are competing themselves to inject their traffic into the network and hence end users get unfair shares of the bandwidth for the video transmissions. Also, the high transmission rate of video applications affects traditional Internet applications like e-mail. TCP-friendly rate control (TFRC), TFRC-satellite and multiple TFRC streams (MulTFRC), which are intended to provide congestion control, error control and increase bandwidth (BW) utilization respectively, may fail over the satellite networks with high error rate and high congestion. Besides, these connection-oriented protocols increase the applications overhead. In this paper, we present a cooperative flow regulation protocol (CFR), which provides fair access to both real-time and non-real-time applications, over networks with high congestion and high error. Verification of the proffered approach, by simulations, show that the CFR protocol reduces the loss of video frames by 30–40%, compared to DCCP and MulTFRC, and 50–60% compared to user datagram protocol (UDP) traffic, without compromise of the video quality.

Evaluation of Impairment Caused by MPEG Video Frame Loss

This article presents a study on the impact of video frame losses on the quality perceived by users. Video compression standards, such as MPEG, use a sequence of frames called Group of Pictures (GOP), which is a video compression method which a frame is expressed in terms of one or more neighboring frames. This dependence between frames impacts directly in the quality because a loss of a reference frame prevents the decoding of other frames in GOP, thereby reducing the user-perceived quality. The assessment of quality in this article is estimated by Peak Signal Noise Ratio (PSNR), which compares the original and the received images. Computer simulations were used to show that the degradation on the quality may vary for different patterns of GOPs and type of lost frames.

Impact of Playout Buffering on Mobile TV Performance

We study in this work the quality of experience of live mobile TV users in the presence of a playout buffer at the receiver side. We specifically consider an LTE network delivering both unicast and broadcast services, and study two cases: i. lossless channels where interruptions in mobile TV delivery are due to the absence of resources at the base station because they are totally consumed by higher priority real-time voice services, and ii. lossy channels, where we propose to exploit the existence of feedback channels to request, if possible, retransmission of the lost video frame. We derive in both cases several QoS metrics pertaining to mobile TV performance, such as interruption frequency and duration as well as video frame loss probability. Our results show the trends of these metrics as a function of several system parameters and yield design rules for network planning as well as for the playout buffer so as to enhance efficiently the live TV watching experience.

선박 네트워크에서 신뢰성있는 멀티미디어 데이터 전송을 위한 자원 관리 기법

In this paper, a resource management scheme for preventing the video frame loss in the presence of network congestion is proposed. WiMedia protocol is suitable for the application that supports the real-time multimedia service in the ship area network since it supports high speed data transfer. However, network congestion causes the degradation of video quality, since WiMedia standard discards video frames regardless of importance of video frame. Therefore, a resource management scheme for WiMedia network is proposed in this paper. The proposed technique can intelligently treat the network congestion, and solve the degradation of video quality.

A Distortion-Resistant Routing Framework for Video Traffic in Wireless Multihop Networks

Traditional routing metrics designed for wireless networks are application-agnostic. In this paper, we consider a wireless network where the application flows consist of video traffic. From a user perspective, reducing the level of video distortion is critical. We ask the question “Should the routing policies change if the end-to-end video distortion is to be minimized?” Popular link-quality-based routing metrics (such as ETX) do not account for dependence (in terms of congestion) across the links of a path; as a result, they can cause video flows to converge onto a few paths and, thus, cause high video distortion. To account for the evolution of the video frame loss process, we construct an analytical framework to, first, understand and, second, assess the impact of the wireless network on video distortion. The framework allows us to formulate a routing policy for minimizing distortion, based on which we design a protocol for routing video traffic. We find via simulations and testbed experiments that our protocol is efficient in reducing video distortion and minimizing the user experience degradation.

A Burst Effort Broadcasting Approach of MPEG-4 Video Transmission for Intervehicle Communication

Efficient and real-time video broadcasting helps to improve driving safety and to make traveling fun for drivers. However, it is hard to structure vehicles into permanent network topologies and schedules to broadcast video streaming real-time in intervehicle communication (IVC) in a high-mobility vehicular environment. In order to achieve this, we have designed a burst effort broadcasting (BEB) approach for IVC, which considers the challenges of high mobility and multihop broadcast, as well as the features of MPEG-4 video streams to adapt to highway scenarios. The BEB approach is distributed in real time without protocol overhead and comprises a queuing procedure and a scheduling scheme. The queuing procedure consists of the preprocess of video transmission, including the video shaping of groups of pictures and sequential reordering video frames. Based on the queuing procedure, a mobility-adaptive scheduler is applied to handle the broadcast and rebroadcast of the video stream. The concept of macroscopic broadcast is utilized to increase the broadcast performance and the video perceived quality of service (PQoS), and to reduce the number of unnecessary redundant broadcasts. As an evaluation, the real MPEG-4 video was conducted in simulation and the broadcast performance was compared with another protocol by the metrics of peak-signal-to-noise ratio (PSNR) and loss of video frames in different broadcasting scenarios. Then, the results were analyzed. The simulation proved that this approach is an efficient multihop broadcast solution that does indeed provide a realistic solution to promote a higher degree of video PQoS on highways.

Analytical model for MPEG video frame loss rates and playback interruptions on packet networks

This paper presents and studies objective video quality evaluation techniques for a network where frame losses can be considered independent, for example a best effort not heavy loaded packet switching network. The total or partial loss of a frame's information affects the quality of video playback, as the frame cannot be decoded and other frames that depend on it cannot be correctly decoded too. Therefore, during some time the video playback has errors in the image and the user will perceive them as interruptions. In this paper, the total number of decoded frames and the video playback interruptions duration will be considered important parameters to quantify the video quality. The analytical formulation for them will be presented and the importance of considering them together will be highlighted.

Synchronization-Sensitive Frame Estimation: Video Quality Enhancement

Network transmission is liable to errors and data loss. In movie transmission, packets of video frames are subject to loss or even explicit elimination for many reasons including congestion handling and the achievement of higher compression. Not only does the loss of video frames cause significant reduction in video quality, but it could also cause a loss of synchronization between the audio and video streams. If not corrected, this cumulative loss can seriously degrade the motion picture's quality beyond viewers' tolerance. In this paper, we study and classify the effect of audio-video de-synchronization. Afterwards, we develop and examine the performance and appropriateness of the application of many client-based techniques in the estimation of lost frames using the existing received frames, without the need for retransmissions or error control information. The estimated frames are injected at their appropriate locations in the movie stream to restore the loss. The objective is to enhance video quality by finding a very close estimate to the original frames at a suitable computation cost, and to contribute to the restoration of synchronization within the tolerance level of viewers.

An improved UDP protocol for video transmission over Internet-to-wireless networks

Packet video will become a significant portion of emerging and future wireless/Internet traffic. However, network congestion and wireless channel error yields tremendous packet loss and degraded video quality. In this paper, we propose a new complete user datagram protocol (CUDP), which utilizes channel error information obtained from the physical and link layers to assist error recovery at the packet level. We propose several maximal distance separable (MDS) code-based packet level error control coding schemes and derive analytical formulas to estimate the equivalent video frame loss for different versions of user datagram protocol (UDP). We validate the proposed packet coding and CUDP protocol using MPEG-coded video under various Internet packet loss and wireless channel profiles. Theoretic and simulation results show that the video quality can be substantially improved by utilizing the frame error information at UDP and application layer.