Mixed streaming of video over wireless networks

In recent years, transmission of video over the Internet has become an important application. As wireless networks are becoming increasingly popular, it is expected that video will be an important application over wireless networks as well. Unlike wired networks, wireless networks have high data loss rates. Streaming video in the presence of high data loss can be a challenge because it results in errors in the video.Video applications produce large amounts of data that need to be compressed for efficient storage and transmission. Video encoders compress data into dependent frames and independent frames. During transmission, the compressed video may lose some data. Depending on where the packet loss occurs in the video, the error can propagate for a long time. If the error occurs on a reference frame at the beginning of the video, all the frames that depend on the reference frame will not be decoded successfully. This thesis presents the concept of mixed streaming, which reduces the impact of video propagation errors in error prone networks. Mixed streaming delivers a video file using two levels of reliability; reliable and unreliable. This allows sensitive parts of the video to be delivered reliably while less sensitive areas of the video are transmitted unreliably. Experiments are conducted that study the behavior of mixed streaming over error prone wireless networks. Results show that mixed streaming makes it possible to reduce the impact of errors by making sure that errors on reference frames are corrected. Correcting errors on reference frames limits the time for which errors can propagate, thereby improving the video quality. Results also show that the delay cost associated with the mixed streaming approach is reasonable for fairly high packet loss rates

End-to-end adaptation scheme for ubiquitous remote experimentation

Remote experimentation is an effective e-learning paradigm for supporting hands-on education using laboratory equipment at distance. The current trend is to enable remote experimentation in mobile and ubiquitous learning. In such a context, the remote experimentation software should enable effective telemonitoring and teleoperation, no matter the kind of device used to access the equipment. It should also be sufficiently lenient so as to handle the rapidly evolving wireless and mobile communication environment. While the current Internet bandwidth allows remote experimentation to work flawlessly on fixed connections such as LANs, mobile users suffer from both the versatile nature of wireless communications and the limitation of the mobile devices. These conditions impose that the remote experimentation software should integrate adaptation features. For effective ubiquitous remote experimentation, it should ideally be guaranteed that the information representing the state of the remote equipment is rendered (to the end user) at the same pace at which it has been acquired, yet possibly at the cost of a somewhat minimal time delay between the acquisition and rendering phases. In this respect, an end-to-end adaptation scheme is proposed that explicitly handles the inherent variability of the connection and the versatility of the mobile devices considered in ubiquitous remote experimentation. Instead of relying on a stochastic approach, the proposed adaptation scheme relies on a deterministic mass-balance equivalence model. The effectiveness of the proposed adaptation scheme is demonstrated in critical conditions corresponding to remote experimentation carried out using a PDA over a Bluetooth lin

Scalable Live Video in MAX/MSP/Jitter

This thesis describes the mcl.jit software library we developed to support scalable live video coding and transmission in Max/MSP/Jitter. Video codecs from this library have been successfully used in several telematic dance performances created by dancers and media artists from the School for the Contemporary Arts at Simon Fraser University during the last two years. The mcl.jit library also includes Region-Of-Interest (ROI) coding and motion detection objects, which can be used in a variety of interactive multimedia applications besides distributed dance performance. We also developed a combined bit rate and frame rate control method for live video for the mcl.jit library. This method differs from previously developed frame rate control approaches in that it does not assume that video is prerecorded before frame rate adjustment. The proposed method was compared to another state-of-the-art method through an extensive subjective evaluation study, the results of which indicate the superiority of the proposed approach

Portable Video Streaming Network

This dissertation addresses the challenge of developing a video call system capable of supporting both Android mobile devices and fixed computers. Addi tionally, it analyses the quality of video achieved and its variation in the presence of network bandwidth and packet loss constraints. A prototype of a video call system was implemented using a web application and the Web Real-Time Communication (WebRTC) library. Clients use WebRTC to stream video over a Traversal Using Relays around NAT (TURN) relay server, allowing them to send video to any terminal connected to the Internet. Signalling was implemented using WebSockets and a Node.js server. A quality testing prototype was also implemented, which supports sending pre-recorded videos and capturing and storing video recordings at the sender and receiver. The Video Multimethod Assessment Fusion (VMAF) metric was used as the main video quality metric, based on the comparison between the transmitted and received videos. The quality of a video encoded using the open source video encoder VP8 was analysed in constrained network setups. The results measured the video quality degradation and percentage of received frames, showing that the system is resilient to some bandwidth strangulation and packet loss, although with a noticeable video quality degradation.Esta dissertação aborda o desafio de desenvolver um sistema de videochamada capaz de suportar dispositivos móveis Android e computadores fixos. Além disso, analisa a qualidade do vídeo obtida e sua variação na presença de restrições de largura de banda da rede e perda de pacotes. Um protótipo de um sistema de videochamada foi implementado usando uma aplicação web e a biblioteca Web Real-Time Communication (WebRTC). Os clientes usam WebRTC para transmitir o vídeo através de um servidor de retransmissão Traversal Using Relays around NAT (TURN), permitindo que enviem vídeo a qualquer cliente ligado à Internet. A sinalização foi implementada usando WebSockets e um servidor Node.js. Também foi implementado um protótipo de teste de qualidade, que suporta o envio de vídeos pré-gravados e a captura e armazenamento de gravações de vídeo no emissor e no recetor. A métrica Video Multimethod Assessment Fusion (VMAF) foi utilizada como a principal métrica de qualidade de vídeo, com base na comparação entre os vídeos transmitidos e recebidos. A qualidade de um vídeo codificado usando VP8 foi analisada em configurações de rede com limitações. Os resultados mediram a degradação da qualidade do vídeo e a percentagem de tramas recebidas, mostrando que o sistema é resiliente a algum estrangulamento da largura de banda e perda de pacotes, embora com uma degradação percetível da qualidade do vídeo

Scalable reliable on-demand media streaming protocols

This thesis considers the problem of delivering streaming media, on-demand, to potentially large numbers of concurrent clients. The problem has motivated the development in prior work of scalable protocols based on multicast or broadcast. However, previous protocols do not allow clients to efficiently: 1) recover from packet loss; 2) share bandwidth fairly with competing flows; or 3) maximize the playback quality at the client for any given client reception rate characteristics. In this work, new protocols, namely Reliable Periodic Broadcast (RPB) and Reliable Bandwidth Skimming (RBS), are developed that efficiently recover from packet loss and achieve close to the best possible server bandwidth scalability for a given set of client characteristics. To share bandwidth fairly with competing traffic such as TCP, these protocols can employ the Vegas Multicast Rate Control (VMRC) protocol proposed in this work. The VMRC protocol exhibits TCP Vegas-like behavior. In comparison to prior rate control protocols, VMRC provides less oscillatory reception rates to clients, and operates without inducing packet loss when the bottleneck link is lightly loaded. The VMRC protocol incorporates a new technique for dynamically adjusting the TCP Vegas threshold parameters based on measured characteristics of the network. This technique implements fair sharing of network resources with other types of competing flows, including widely deployed versions of TCP such as TCP Reno. This fair sharing is not possible with the previously defined static Vegas threshold parameters. The RPB protocol is extended to efficiently support quality adaptation. The Optimized Heterogeneous Periodic Broadcast (HPB) is designed to support a range of client reception rates and efficiently support static quality adaptation by allowing clients to work-ahead before beginning playback to receive a media file of the desired quality. A dynamic quality adaptation technique is developed and evaluated which allows clients to achieve more uniform playback quality given time-varying client reception rates

Adaptive delivery of real-time streaming video

Thesis (M.Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2001.Includes bibliographical references (p. 87-92).While there is an increasing demand for streaming video applications on the Internet, various network characteristics make the deployment of these applications more challenging than traditional Internet applications like email and the Web. The applications that transmit data over the Internet must cope with the time-varying bandwidth and delay characteristics of the Internet and must be resilient to packet loss. This thesis examines these challenges and presents a system design and implementation that ameliorates some of the important problems with video streaming over the Internet. Video sequences are typically compressed in a format such as MPEG-4 to achieve bandwidth efficiency. Video compression exploits redundancy between frames to achieve higher compression. However, packet loss can be detrimental to compressed video with interdependent frames because errors potentially propagate across many frames. While the need for low latency prevents the retransmission of all lost data, we leverage the characteristics of MPEG-4 to selectively retransmit only the most important data in order to limit the propagation of errors. We quantify the effects of packet loss on the quality of MPEG-4 video, develop an analytical model to explain these effects, and present an RTP-compatible protocol-which we call SR-RTP--to adaptively deliver higher quality video in the face of packet loss. The Internet's variable bandwidth and delay make it difficult to achieve high utilization, Tcp friendliness, and a high-quality constant playout rate; a video streaming system should adapt to these changing conditions and tailor the quality of the transmitted bitstream to available bandwidth. Traditional congestion avoidance schemes such as TCP's additive-increase/multiplicative/decrease (AIMD) cause large oscillations in transmission rates that degrade the perceptual quality of the video stream. To combat bandwidth variation, we design a scheme for performing quality adaptation of layered video for a general family of congestion control algorithms called binomial congestion control and show that a combination of smooth congestion control and clever receiver-buffered quality adaptation can reduce oscillations, increase interactivity, and deliver higher quality video for a given amount of buffering. We have integrated this selective reliability and quality adaptation into a publicly available software library. Using this system as a testbed, we show that the use of selective reliability can greatly increase the quality of received video, and that the use of binomial congestion control and receiver quality adaptation allow for increased user interactivity and better video quality.by Nicholas G. Feamster.M.Eng