In order to enable a truly pervasive computing environment, next generation networks (including B3G and 4G) will merge the broadband wireless and wireline networking infrastructure. However, due to the tremendous complexity in administration and the unreliability of the wireless channel, provision of hard-guarantees for services on such networks will not happen in the foreseeable future. This consequently makes it particularly challenging to offer viable AV conferencing services due to their stringent synchronization, delay and data fidelity requirements. We propose in this paper a robust application-level solution for wireless mobile AV conferencing on B3G/4G networks. Expecting no special treatment from the network, we apply a novel adaptive delay and synchronization control mechanism to maintain the synchronization and reduce the latency as much as possible. We also employ a robust video coding technique that has better error-resilience capability. We investigate the performance of the proposed solution through simulations using a three-state hidden Markov chain as the generic end-to-end transport channel model. The results show that our scheme yields tight synchronization performance, relatively low end-to-end latency and satisfactory presentation quality. The scheme successfully provides a fairly robust AV conferencing service.
Internet technologies are increasingly facilitating real-time monitoring of Bridges and Highways. The advances in wireless communications for instance, are allowing practical deployments for large extended systems. Sensor data, including video signals, can be used for long-term condition assessment, traffic-load regulation, emergency response, and seismic safety applications. Computer-based automated signal-analysis algorithms routinely process the incoming data and determine anomalies based on pre-defined response thresholds and more involved signal analysis techniques. Upon authentication, appropriate action may be authorized for maintenance, early warning, and/or emergency response. In such a strategy, data from thousands of sensors can be analyzed with near real-time and long-term assessment and decision-making implications. Addressing the above, a flexible and scalable (e.g., for an entire Highway system, or portfolio of Networked Civil Infrastructure) software architecture/framework is being developed and implemented. This framework will network and integrate real-time heterogeneous sensor data, database and archiving systems, computer vision, data analysis and interpretation, physics-based numerical simulation of complex structural systems, visualization, reliability & risk analysis, and rational statistical decision-making procedures. Thus, within this framework, data is converted into information, information into knowledge, and knowledge into decision at the end of the pipeline. Such a decision-support system contributes to the vitality of our economy, as rehabilitation, renewal, replacement, and/or maintenance of this infrastructure are estimated to require expenditures in the Trillion-dollar range nationwide, including issues of Homeland security and natural disaster mitigation. A pilot website (http://bridge.ucsd.edu/compositedeck.html) currently depicts some basic elements of the envisioned integrated health monitoring analysis framework.
When designing an encoder for a real-time video application over a wireless channel, we must take into consideration the unpredictable fluctuation of the quality of the channel and its impact on the transmitted video data. This uncertainty motivates the development of an adaptive video encoding mechanism that can compensate for the infidelity caused either by data loss and/or by the post-processing (error concealment) at the decoder. In this paper, we first explore the major factors that cause quality degradation. We then propose an adaptive progressive replenishment algorithm for a packet loss rate (PLR) feedback enabled system. Assuming the availability of a feedback channel, we discuss a video quality assessment method, which allows the encoder to be aware of the decoder-side perceptual quality. Finally, we present a novel dual-feedback mechanism that guarantees an acceptable level of quality at the receiver side with modest increase in the complexity of the encoder.
Supporting real-time multimedia applications over the packet wireless network poses a huge challenge due to the stringent QoS requirements and the time-varying and location-varying characteristics of the wireless channel. In this paper, we study a downlink packet scheduling algorithm designed to support real-time applications with hard deadlines over the wireless network. The scheduling depends on the channel characteristics as well as the deadline of each packet. A hybrid TDMA/CDMA physical layer is used to assess the channel condition and to support the target BER of different users. RTP packet headers are used to calculate the deadline of each packet. A modified form of EDF is used to schedule packets in order of urgency and channel condition. Our results indicate that by jointly considering the channel information and the application-level information, a significant performance gain can be achieved. In addition, a more fair service is achieved even though channel conditions of different users vary greatly due to propagation losses and fading effects of the wireless channel.
In this paper, we investigated the possibility and effectiveness of applying ARQ schemes to the transmission of MPEG-2 encoded video streams over fixed wireless links in B- FWANs in video-on-demand services. Simulations were conducted to evaluate the overall system performance, which is then compared to a scheme using FEC with cell interleaving. Important metrics relevant to the broadband video services are analyzed, including the quality of the reconstructed video sequences, the excess delay and delay variation introduced by the ARQ scheme, the throughput, and the required average channel bandwidth. A hidden Markov model was also introduced to model the variation of the wireless channel.
In this paper, we address the issue of error control in transmitting MPEG-2 encoded video streams over broadband fixed wireless access networks for broadcast or multicast services. Because of the error-prone nature of wireless channels, error control is mandatory when MPEG-2 video streams are transported over wireless access networks to end user. To prevent overloading the reliable wireline networks error control has to be applied locally. FEC is a must for broadcast or multicast services. Because of the important role of MPEG-2 control information in the decoding process, it must be given priority service in the form of excess error protection in order to achieve the desired QoS. In this paper, a header redundancy FEC (HRFEC) strategy is introduced and an implementation of it (type-I HRFEC scheme) is described. The overhead and delay jitter associated with the type-I HRFEC is also estimated. Simulation results on the performance of type-I HRFEC indicates that it improves the reception statistics of MPEG-2 control. As a direct, the quality, measured in terms of objective grade point and PSNR of the reconstructed video sequence, is improved.
Hybrid ARQ schemes can yield much better throughput and reliability than static FEC schemes for the transmission of data over time-varying wireless channels. However these schemes result in higher delay. They adapt to the varying channel conditions by retransmitting erroneous packets, this results in variable effective data rates for current PCS networks because the channel bandwidth is constant. Hybrid ARQ schemes are currently being proposed as the error control schemes for real-time video transmission. The standardization process is on-going in ITU, MPEG-4 and wireless ATM forum. The important issue is how to ensure low delay while taking advantage of the high throughput and reliability that these schemes provide for. In this paper we propose an adaptive source rate control (ASRC) protocol which can work together with the hybrid ARQ error control schemes to achieve efficient transmission of real-time video with low delay and high reliability. The ASRC scheme adjusts the source rate based on the channel conditions, the transport buffer occupancy and the delay constraints. It optimizes the video quality by dynamically changing both the number of the forced update (intracoded) macroblocks and the quantization scale used in a frame. The number of the forced update macroblocks used in a frame is first adjusted according to the allocated source rate. This reduces the fluctuation of the quantization scale with the change in the channel conditions during encoding so that the uniformity of the video quality is improved. The simulation results show that the proposed ASRC protocol performs very well for both slow fading and fast fading channels.
The currently specified variable bit-rate (VBR) service class for real-time traffic on broad-band networks severely sacrifices network utilization to provide QoS support for multimedia traffic. A new service class, called VBR+, has been proposed to balance the goals of providing acceptable QoS and achieving high network utilization. VBR+ is a flexible service class that extends the traditional VBR service with bandwidth renegotiation. Bandwidth renegotiation is well suited for the dynamic traffic profiles of multimedia applications. Renegotiation allows a more efficient network capacity allocation and potentially allows the network to operate at more aggressive statistical multiplexing regimes while maintaining acceptable QoS. Some quality degradation is caused by source rate control when the network is congested and renegotiation requests cannot be fully satisfied. This paper quantifies the trade-off between video quality and network utilization for VBR+ transport. The multiplexing performance of VBR+ traffic is obtained via simulation using MPEG-2 video traces obtained using NEC's VisuaLink codec. Results shows that VBR+ transport can maintain acceptable video quality at 70 - 80% link utilization. This represents a 20 - 30% improvement on utilization over the currently specified VBR service for comparable video quality.
Providing high bit rate real-time video services has been a major driving factor in the advancement of high speed networking technology such as ATM based BISDN. In this paper, we describe MPEG2Tool, an X-window-based software implementation of the MPEG-2 video compression algorithm with many additional useful functions. The ultimate goal of designing this toolkit was to facilitate the study of MPEG video transmission over ATM-based networks. The toolkit consists of four major modules, which appear as four push-buttons in the main Motif menu: (1) encoding, (2) statistical analysis, (3) transmission simulation, and (4) decoding.
The effects of digital transmission errors on H.263 codecs are analyzed and the transmission of H.263 coded video over a TDMA radio link is investigated. Numerical results for the channel SNR required for providing acceptable video quality under various channel coding and interleaving strategies are presented. Fading on radio channels causes significant transmission errors and H.263 coded bit streams are very vulnerable to errors. Therefore, powerful forward error correction (FEC) codes are necessary to protect the data so that it can be successfully transmitted at acceptable signal power levels. However, FEC imposes a high bandwidth overhead. In order to make best use of the available channel bandwidth and to alleviate the overall impact of errors on the video sequence, a two-layer source coding and unequal error protection scheme based on H.263 is also studied. The scheme can tolerate more transmission errors and leads to more graceful degradation in quality when the channel SNR decreases. In lossy environments, it yields better video quality at no extra bandwidth cost.
For high-end real-time video services, a prioritized transmission scheme (PT) should match the video information distribution of the high priority and low priority bitstreams generated by a layered source coder. This paper studies two PT schemes, which are respectively coupled- concatenated transmission (CCT) and coupled-interlaced transmission (CIT). Simulation results reveal that: given the same amount of network resource, the CIT scheme outperforms both the CCT scheme and the non-layered scheme in terms of the delivered video quality and the network resource utilization.
When cell loss occurs, error concealment can play a critical role in recovering the viewing quality of impaired video. In this paper, we first present a Slice Interleaving (SI) algorithm which has been demonstrated to be able to effectively prevent vertical adjacent slice loss, thereby enhancing the performance of error concealment techniques that rely heavily on interpolation. Experimentation on approximately 100 video clips using a time-variant Markov Chain (MC) cell loss model that emulates a fluctuating bursty cell loss environment, allowed us to conduct a comprehensive and comparative study of error concealment mechanisms in different coding domains. We then designed a hybrid concealment algorithm, such that, for an arbitrary video sequence, several concealment mechanisms can be adaptively merged together to achieve the best performance. Simulation results show that our hybrid algorithm can effectively detect frames or macroblocks with scene change and/or excessive irregular motion, and adaptively switch to an appropriate concealment module to achieve the optimal video quality.
The demand for multimedia applications has spurred significant interest in the area of video compression. Yet, as the complexity of compression algorithms increase, the design and optimization of video applications has become both formidable and time consuming. In this paper, we outline an object-oriented C++ video toolkit and illustrate its usage in an R&D setting. This toolkit enables the user to rapidly construct complex video algorithms using familiar objects and operations. A set of statistical gathering tools and MPEG extensions which perform MPEG decoding and encoding are also provided. After describing these components, we demonstrate the use of the toolkit to design a complex applications called the MPEGEditor which performs standard editing operations such as splicing and fading on MPEG sequences through an X Windows/Motif graphical user interface. As a research tool, we illustrate how to expand the toolkit to incorporate new compression techniques. In particular, we show how to extend the MPEG encoding algorithm to include a number of prioritization techniques which are not part of the MPEG standard.