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

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

Complexity management for video encoders.

Software implementation of block-based video coding standards has been used in a wide range of applications. In many cases, such as real-time multimedia systems or power-constrained systems, the coding performance of software-only video encoders and decoders is limited by computational complexity. This thesis presents research work to develop techniques to manage computational complexity of video encoders. These techniques aim to provide significant complexity saving as well as adaptively controlling the computational complexity. This thesis first investigates experimentally the most computationally intensive functions in a video encoder. Based on the results of profile tests, several functions are selected as candidates, on which complexity reduction algorithms will be performed. These functions include discrete cosine transform and related functions as well as motion estimation. Adaptive complexity-reduction algorithms are proposed for computationally expensive functions: discrete cosine transform and motion estimation functions respectively. It is shown that these algorithms can flexibly control the computational complexity of each function with negligible loss of video quality. The inherent characteristics of coded macroblocks are investigated through experimental tests and they are categorized into "skipped" and" unskipped" macroblocks based on two parameters. An innovative algorithm is developed to reduce the computational complexity by predicting "skipped" macroblock prior to encoding and not carrying out the coding process on these macroblocks. The approaches described in this thesis can not only achieve adaptive control of the computational complexity of a video encoder, but also can manage the trade-off between complexity and distortion. These proposed algorithms are evaluated in terms of complexity reduction performance, rate-distortion performance and subjective and objective visual quality by experimental testing.The advantages and disadvantages of each algorithm are discussed