Improve Video Quality Research Articles

Video transport over ad hoc networks: multistream coding with multipath transport

Enabling video transport over ad hoc networks is more challenging than over other wireless networks. The wireless links in an ad hoc network are highly error prone and can go down frequently because of node mobility, interference, channel fading, and the lack of infrastructure. However, the mesh topology of ad hoc networks implies that it is possible to establish multiple paths between a source and a destination. Indeed, multipath transport provides an extra degree of freedom in designing error resilient video coding and transport schemes. In this paper, we propose to combine multistream coding with multipath transport, to show that, in addition to traditional error control techniques, path diversity provides an effective means to combat transmission error in ad hoc networks. The schemes that we have examined are: 1) feedback based reference picture selection; 2) layered coding with selective automatic repeat request; and 3) multiple description motion compensation coding. All these techniques are based on the motion compensated prediction technique found in modern video coding standards. We studied the performance of these three schemes via extensive simulations using both Markov channel models and OPNET Modeler. To further validate the viability and performance advantages of these schemes, we implemented an ad hoc multiple path video streaming testbed using notebook computers and IEEE 802.11b cards. The results show that great improvement in video quality can be achieved over the standard schemes with limited additional cost. Each of these three video coding/transport techniques is best suited for a particular environment, depending on the availability of a feedback channel, the end-to-end delay constraint, and the error characteristics of the paths.

An accurate bit-rate control algorithm for video transcoding

In many video based applications, it is essential to precisely control the bit rate of video streams for transmission over different networks and channel bandwidth. One critical element in a bit rate control algorithm is the bit production model that predicts the number of produced bits when a certain quantization parameter is used. In this paper, we present a novel bit allocation and rate control algorithm for compressed domain video transcoding. The specific transcoding issue mentioned is referred to as bit rate adaptation or rate shaping. We first review the architectures of different bit rate adaptation transcoders and the generic rate control problem. Then, we propose and formulate an approximate linear bit allocation model, which is based on experimental results. Based on this model, we propose an adaptive bit estimation and allocation scheme for video transcoding. Implementation results show that the proposed algorithm can provide accurate bit allocation, stable buffer occupancy and improved video quality as compared to existing approaches. This rate control scheme can be used to provide flexible video bit rate adaptation and stable transmission of video streams over heterogeneous networks.

Unequal loss protection for robust transmission of motion compensated video over the internet

This paper presents an unequal loss protection (ULP) scheme for robust transmission of motion compensated video over the Internet. By exploiting the temporal dependency between frames, forward error correction (FEC) codes across packets are assigned to different frames in a group of pictures in the sense of minimizing the effect of error propagation, thus improving video quality significantly. To achieve optimal allocation of FEC codes, we formulate the effect of packet loss on video quality degradation as an expected length of error propagation (ELEP) model, which makes sense intuitively, as fewer frames corrupted implies better quality of reconstructed video. Experimental results show the validity of the proposed ELEP model and that the associated ULP scheme is robust to burst packet loss in the Internet. More importantly, graceful degradation of video quality is achieved by the proposed scheme as the packet loss rate of an Internet connection increases.

Videoconferenced continuing medical education in Nova Scotia.

Videoconferencing has been used for continuing medical education (CME) in Nova Scotia since a pilot project to four communities in 1995. The Nova Scotia Telehealth Network was developed after the pilot project. Using the network, the videoconferenced CME programme expanded over the next few years until in, 2000-1, 66 programmes were broadcast to 38 sites. During the expansion of the programme, we improved video quality and developed efficient methods of: scheduling and planning the content of the videoconferences; training faculty presenters in videoconferencing techniques; and evaluation. We consider this programme represents a success. However, several aspects could be improved. Faculty members still need encouragement to make visual aids legible by videoconference and to provide handouts. Also, there has been little upgrading of equipment over the past four years and some reduction in the reliability of connections.

Adaptive motion-compensation fine-granular-scalability (AMC-FGS) for wireless video

Transmission of video over wireless and mobile networks requires a scalable solution that is capable of adapting to the varying channel conditions in real-time (bit-rate scalability). Furthermore, video content needs to be coded in a scalable fashion to match the capabilities of a variety of devices (complexity scalability). These two scalability properties provide the flexibility that is necessary to satisfy the "anywhere, anytime and anyone" network paradigm of wireless systems. MPEG-4 fine-granular-scalability (FGS) is a flexible low-complexity solution for video streaming over heterogeneous networks (e.g., the Internet and wireless networks) and is highly resilient to packet losses. However, the flexibility and packet-loss resilience come at the expense of decreased coding efficiency compared with nonscalable coding. A novel scalable video-coding framework and corresponding compression methods for wireless video streaming is introduced. Building on the FGS approach, the proposed framework, which we refer to as adaptive motion-compensation FGS (AMC-FGS), provides improved video quality of up to 2 dB. Furthermore, the new scalability structures provide the FGS framework with the flexibility to provide tradeoffs between resilience, higher coding efficiency and terminal complexity for more efficient wireless transmission.

A rate-control scheme for video transport over wireless channels

We investigate the scenario of using the automatic repeat request (ARQ) retransmission scheme for two-way video communications over wireless Rayleigh fading channels. Video quality is the major concern of these applications. We show that, during the retransmissions of error packets, due to the reduced channel throughput, the video encoder buffer may fill-up quickly and cause the TMN8 rate-control algorithm to significantly reduce the bits allocated to each video frame. This results in PSNR degradation and many skipped frames. To minimize the number of frames skipped, we propose improved rate-control schemes that take into consideration the effects of the video buffer fill-up, an a priori channel model, and the channel feedback information. We also show that a discrete cosine transform (DCT) coefficient soft-thresholding scheme can be applied to further improve video quality. As a result, our proposed rate-control schemes encode the video sequences with less frame skipping and with higher PSNR compared to TMN8.

Picture quality optimization in ABR video services

In transmission of bursty traffic such as video and data, a network with a fixed capacity may have some unused bandwidth. The available bit-rate (ABR) scheme in ATM networks can make use of this spare channel capacity as a lower priority service. ABR has been developed initially for delay-tolerant services such as data. In this paper we demonstrate how with the help of a video transcoder delay-sensitive services such as video may be transmitted over ABR networks. The transcoder further compresses the video bit stream according to the rates dictated by the network with minimum degradation to the picture quality. We have optimised the video quality by combining the classical minimum cell rate (MCR) with the subjectively acceptable minimum quality (SAMQ). The performance of two different coding formats for ABR sources, namely the constant bit-rate (CBR) and variable bit-rate (VBR) video input to the transcoder are compared. We also present techniques for further reduction of congestion at the multiplexing switches to improve video quality and report on the impact of the forward and backward link delays on the ABR performance with a video transcoder.

Encoding stored video for streaming applications

In streaming video applications, video sequences are encoded off-line and stored in a server. Users may access the server over a constant bit rate channel. Examples of the streaming video applications are video-on-demand, archived video news, and noninteractive distance learning. Before the playback, part of the video bitstream is pre-loaded in the decoder buffer to ensure that every frame can be decoded at the scheduled time. For these streaming video applications, since the video is encoded off-line and the future video frames are available to the encoder, a more sophisticated bit-allocation scheme can be used to achieve better video quality. During the encoding process for streaming video, two requirements need to be considered: the pre-loading time that the video viewers have to wait and the physical buffer-size at the receiver (decoder) side. In this paper, we propose a sliding-window rate-control scheme that uses statistical information of the future video frames as a guidance to generate better video quality for video streaming involving constant bit rate channels. A quantized discrete cosine transform coefficient selection scheme based on the rate-distortion measurement is also used to improve the video quality. Simulation results show video quality improvements over the regular H.263 TMN8 encoder.

Error-resilient transcoding for video over wireless channels

We describe a method to maintain quality for video transported over wireless channels. The method is built on three fundamental blocks. First, we use a transcoder that injects spatial and temporal resilience into an encoded bitstream. The amount of resilience is tailored to the content of the video and the prevailing error conditions, as characterized by bit error rate. Second, we derive analytical models that characterize how corruption propagates in a video that is compressed using motion-compensated encoding and subjected to bit errors. Third, we use rate distortion theory to compute the optimal allocation of bit rate among spatial resilience, temporal resilience, and source rate. Furthermore, we use the analytical models to generate the resilience rate distortion functions that are used to compute the optimal resilience. The transcoder then injects this optimal resilience into the bitstream. Simulation results show that using a transcoder to optimally adjust the resilience improves video quality in the presence of errors while maintaining the same input bit rate.

Precise and fast error tracking for error-resilient transmission of H.263 video

A precise error tracking scheme for robust transmission of real-time H.263 video is presented. By utilizing a feedback channel, the decoder reports the addresses of corrupted blocks induced by transmission errors back to the encoder. With these negative acknowledgments, the encoder can precisely calculate and track the propagated errors by examining the backward motion dependency for each pixel in the current encoding frame. With this precise tracking, the error-propagation effects can be terminated completely by INTRA refreshing the affected macroblocks. In addition, by utilizing the four-corner tracking approximation and the linear motion model, a fast algorithm is also developed to further reduce the computation and memory requirements. The simulations show that both schemes yield significant video quality improvements in error-prone environments. The advantages of the low memory requirement and the low computation complexity are particularly suitable for real time implementation.

On spatial adaptation of motion-field smoothness in video coding

Most motion-compensation methods dealt with in the literature make strong assumptions about the smoothness of the underlying motion field. For instance, block-matching algorithms assume a blockwise-constant motion field and are adequate for translational motion models; control-grid interpolation assumes a blockwise bilinear motion field and captures zooming and warping fairly well. Time-varying imagery, however often contains both types of motion (as well as others), and hence exhibits a high degree of spatial variability of its motion-field smoothness properties. We develop a simple method to spatially adapt the smoothness of the motion field. The proposed method demonstrates substantial improvements in video quality over a wide range of bit rates. To this end, we introduce the notion of a motion field that is characterized by a set of labels. The labels provide the flexibility to adaptively switch between two different motion models locally. The individual motion models have very different smoothness properties. The switched framework for motion compensation performs significantly better than each of its constituent motion models, in terms of both visual quality and signal-to-noise ratio (0.3-0.7 dB on the average). Finally, we develop an extension of this method that enhances the overlapped block motion compensation scheme by allowing spatial adaptation of the window function.

Packet loss resilient transmission of MPEG video over the Internet

A method is proposed to protect MPEG video quality from packet loss for real-time transmission over the Internet. Because MPEG uses inter-frame coding, relatively small packet loss rates in IP transmission can dramatically reduce the quality of the received MPEG video. In the proposed high-priority protection (HiPP) method, the MPEG video stream is split into high- and low-priority partitions, using a technique similar to MPEG-2 data partitioning. Overhead resilient data for the MPEG video stream is created by applying forward error correction coding to only the high-priority portion of the video stream. The high- and low-priority data, and resilient data, are sent over a single channel, using a packetization method that maximizes resistance to burst losses, while minimizing delay and overhead. Because the proposed method has low delay and does not require re-transmission, it is well suited for interactive and multicast applications. Simulations were performed comparing the improvement in video quality using the HiPP method, using experimental Internet packet loss traces with loss rates in the range of 0–8.5%. Overhead resiliency data rates of 0%, 12.5%, 25%, and 37.5% were studied, with different compositions of the overhead data for the 25% and 37.5% overhead rates, in an attempt to find the “best” composition of the overhead data. In the presence of packet loss, the received video quality, as measured by PSNR and the Negsob measure, was significantly improved when the HiPP method was applied.

540 Mbit/sec Serial Video: The Evolution of Studio Interconnect

SMPTE 259M, with data rates up to 360 Mbits/sec, provides the robust high-speed infrastructure for television and production installations throughout the world. Today's broadcast and professional video market is undergoing tremendous change. Pressures to shorten time-to-air, improve video quality, reduce costs, and the advent of advanced television/high-definition television, are driving studios to adopt new technologies and business models. At the heart of this change is the need to further evolve the capability of serial digital interconnect. Operation at 540 Mbits/sec now enables a number of applications and capabilities that help studios address these changing requirements.

Feedback control mechanisms for real-time multipoint video services

While existing research shows that reactive congestion control mechanisms are capable of providing high video quality and channel utilization for point-to-point real-time video, there has been relatively little study of the reactive congestion control of point-to-multipoint video, especially in ATM networks. Problems complicating the provision of multipoint, feedback-based real-time video service include: (1) implosion of feedback returning to the source as the number of multicast destinations increases and (2) variance in the amount of available bandwidth on different branches in the multipoint connection. A new service architecture is proposed for real-time multicast video, and two multipoint feedback mechanisms to support this service are introduced and studied. The mechanisms support a minimum bandwidth guarantee and the best effort support of video traffic exceeding the minimum rate. They both rely on adaptive, multilayered coding at the video source and closed-loop feedback from the network in order to control both the high and low priority video generation rates of the video encoder. Simulation results show that the studied feedback mechanisms provide, at the minimum, a quality of video comparable to a constant bit rate (CBR) connection reserving the same amount of bandwidth. When unutilized network bandwidth becomes available, the mechanisms are capable of exploiting it to dynamically improve video quality beyond the minimum guaranteed level.

Basic characteristics of variable rate video coding in ATM environment

Basic characteristics of variable-rate video coders applied to asynchronous transfer mode (ATM) transmission are described. Burstiness of video information is evaluated for conference-type scenes using various coding algorithms. Three measures (distribution, autocorrelation, and coefficient of variation) are introduced to evaluate burstiness. Video sources are modeled and characterized by the autoregressive process and coefficient of variation. Video quality improvement achieved with variable rate transmission is evaluated using signal-to-noise ratio (SNR) and subjective ratings. An improvement of 5-10 dB in temporal SNR and 1 rank in mean opinion score are reported.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>