Real-time Video Transmission Research Articles

A rate adaptation transcoding scheme for real-time video transmission over wireless channels

In this work, we investigate the problem of bit-rate adaptation transcoding for transmitting video over burst-error wireless channels, i.e., channels such that errors tend to occur in clusters during fading periods. In particular, we consider a scenario consisting of packet-based transmission with an Automatic Repeat reQuest (ARQ) error control and a feedback channel. With the acknowledgements received through the feedback channel and a statistical channel model, we have an estimate of the current channel state, and effective channel bandwidth. In this paper, we analyze the buffering implications of inserting a video transcoder at the wireless access point with the variable bit-rate channel as the target. We derive the conditions that both the source encoder and transcoder buffers have to meet for preventing the end decoder buffer from underflowing. Furthermore, we propose a bit-rate adaptation algorithm for VBR transcoders used in the wireless access point. Our experimental results demonstrate that the proposed algorithm can accurately control the bit-rate of the transcoded video stream and reduce the number of frames been skipped without violating the end-to-end delay requirement.

A new approach to optimize bandwidth reservation for real-time video transmission with deterministic guarantees

The transmission of video requires new service models for providing quality of service (QoS). Some of these models are based on resource reservation and admission control while others rely on resource provisioning. In both approaches, optimally estimating the resource requirements of a given video is a key issue, since they are always very demanding. This paper introduces a fast and bounded method to optimize the required network resources (bandwidth) to guarantee a maximum deterministic delay for a given workload and network state. The method is based on characterizing the workload of a stored video using a reduced set of points obtained from an off-line analysis of its empirical envelope. The paper also proposes the above workload characterization to perform the admission control test using a WFQ scheduler. The evaluation of this scheme shows that it can achieve an utilization higher than a 50% for a single flow with a 1-s deadline and reach up to an 80% with a 5-s delay. Results are also compared to tests based on EDF schemes, which have been proved to be optimal with one node. It is shown that the proposed scheme is a little less efficient than the optimal EDF scheduler with one node (as expected), but it is practically as efficient (or even better with a moderately high number of nodes) than the best known RC-EDF policies with several nodes. This is an interesting result because it shows that the WFQ schedulers can achieve a similar efficiency to EDF schedulers avoiding the complexity of their admission control tests.

Remote Video Monitoring Over the WWW

Remote video monitoring has become increasingly important for monitoring the security of destined locations. Traditional security monitoring systems using coaxial cables and VCR recording systems are expensive and ineffective. With the rapid growth of the Internet, it is now possible to use it as an intermediate transmission medium to support real-time video transmission. This paper proposes a web-based remote monitoring system known as iSecure. Apart from the essential live or stored video transmission, intelligent monitoring and web-based monitoring are also supported. As the transmission of live video data through the packet switched network environment of the Internet can result in packet loss and quality degradation, the iSecure system has implemented an adaptive transmission and recovery mechanism to enhance the quality of real-time video transmission. Intelligent monitoring for elevator security and face-based door access control applications has been incorporated. The iSecure system can be used as a framework for developing other intelligent remote monitoring applications.

Uncoordinated real-time video transmission in wireless multicode CDMA systems: An SMPT-based approach

We consider the real-time transmission of encoded video from distributed, uncoordinated wireless terminals to a central base station in a multicode CDMA system. Our approach is to employ the recently proposed simultaneous MAC packet transmission (SMPT) approach at the data link layer (in conjunction with UDP at the transport layer). We consider the real-time transmission of both video encoded in an open loop (i.e., without rate control) and video encoded in a closed loop (i.e., with rate control). We conduct extensive simulations and study quantitatively the trade-off between video quality, transmission delay (and jitter), and number of supported video streams (capacity). We find that the simple-to-deploy SMPT approach achieves significantly higher video quality and smaller delays than the conventional sequential transmission approach, while ensuring high capacity. In typical scenarios, with SMPT the probability of in-time video frame delivery is more than twice as large as with sequential transmission (for given delay bounds). Our results provide guidelines for the design and dimensioning of cellular wireless systems as well as ad hoc wireless systems.

Stochastic rate-control of video coders for wireless channels

We introduce a new approach to deal with the transmission of real-time video over wireless channels, based on a priori stochastic models for both source and channel. This new problem formulation captures in a natural way the stochastic nature of the channel as well as the uncertainty regarding the properties of the video sequence. Our formulation leads to an optimal control problem that can be solved off-line, employing standard stochastic dynamic programming techniques. The outcome of this optimization is an off-line control policy that is optimal in the sense of minimizing the average coding distortion. The on-line computational cost of the new approach is thus very low: all that is required during run-time is to identify the state of the system (source and channel). Unlike other optimization-based rate control techniques, which require a search for the optimal operating point, the operating points here for each allowable state of the system have been precalculated. We consider wireless packet-based transmission with Automatic Repeat reQuest (ARQ) error control. While a standard model has been adopted to characterize the channel behavior, a new model based on the concept of coding complexity has been devised in order to characterize the video source. Simulation results based on this new approach are provided and compared to other proposed rate-control strategies. They show how the use of model-based optimal policies has negligible on-line computational cost while providing a transmission quality comparable to that achieved with more costly deterministic dynamic programming techniques, and significantly better than for simpler algorithms that do not explicitly take into account the channel state.

Efficient traffic prediction scheme for real-time VBR MPEG video transmission over high-speed networks

The efficient transportation of real-time variable bit rate (VBR) video traffic in high-speed networks is currently an active area of research. The capability to predict VBR video traffic can significantly improve the effectiveness of numerous management tasks, including dynamic bandwidth allocation and congestion control. This paper proposes an adaptive traffic prediction method for VBR MPEG videos, a major multimedia application. Rapid traffic variations due to scene changes are analyzed, then a prediction scheme using the identification of scene changes related to I and P frames is presented. For predicting multiplexed MPEG traffic, a prediction interval is derived that represents a highly correlated traffic sequence. In addition, to reduce the prediction error, a less fluctuating signal instead of the original multiplexed traffic is used as the input for the predictor. Simulation results show that the proposed method is able to predict the original traffic more accurately than the conventional LMS method.

Transmission of real-time video over IP differentiated services

Multimedia applications require high bandwidth and guaranteed quality of service (QoS). The current Internet, which provides 'best effort' services, cannot meet the stringent QoS requirements for delivering MPEG videos. It is proposed that MPEG frames are transported through various service models of DiffServ. Performance analysis and simulation results show that the proposed approach can not only guarantee QoS but can also achieve high bandwidth utilisation.

Traffic specifications for the transmission of stored MPEG video on the Internet

Guarantees of quality-of-service (QoS) in the real-time transmission of stored video on the Internet is a challenging task for the success of many video on demand (VoD) applications. Two QoS classes have been specified by the IETF Integrated Services (intserv) Working Group: Guaranteed Services and Controlled-Load Services. For both of them, it is necessary to provide traffic sources with the capability of calculating the traffic characteristics to be declared to the network, Tspec, on the basis of a limited set of parameters statistically characterizing the traffic and the required level of QoS. The target of this paper is to develop an algorithm for the evaluation of the Tspec parameters which characterize the video stream when a given QoS is required. To this end an analytical framework modeling an MPEG stored-video server and the access network node is introduced. The video emission process is modeled with a switched batch Bernoulli process (SBBP), and performance in the video-server smoother is analytically evaluated. Then the token bucket at the network access point, loaded by the output traffic of the video-server smoother, is modeled to calculate the probability of marking nonconforming data packets.

Streaming video over the Internet: approaches and directions

Due to the explosive growth of the Internet and increasing demand for multimedia information on the Web, streaming video over the Internet has received tremendous attention from academia and industry. Transmission of real-time video typically has bandwidth, delay, and loss requirements. However, the current best-effort Internet does not offer any quality of service (QoS) guarantees to streaming video. Furthermore, for video multicast, it is difficult to achieve both efficiency and flexibility. Thus, Internet streaming video poses many challenges. In this article we cover six key areas of streaming video. Specifically, we cover video compression, application-layer QoS control, continuous media distribution services, streaming servers, media synchronization mechanisms, and protocols for streaming media. For each area, we address the particular issues and review major approaches and mechanisms. We also discuss the tradeoffs of the approaches and point out future research directions.

Transporting real-time video over the Internet: challenges and approaches

Delivering real-time video over the Internet is an important component of many Internet multimedia applications. Transmission of real-time video has bandwidth, delay, and loss requirements. However the current Internet does not offer any quality of service (QoS) guarantees to video transmission over the Internet. In addition, the heterogeneity of the networks and end systems makes it difficult to multicast Internet video in an efficient and flexible way. Thus, designing protocols and mechanisms for Internet video transmission poses many challenges. In this paper, we take a holistic approach to these challenges and present solutions from both transport and compression perspectives. With the holistic approach, we design a framework for transporting real-time Internet video, which includes two components, namely, congestion control and error control. Specifically congestion control consists of rate control, rate-adaptive encoding, and rate shaping; error control consists of forward error correction (FEC), retransmission error resilience, and error concealment. For the design of each component in the framework, we classify approaches and summarize representative research work. We point out there exists a design space which can be explored by video application designers and suggest that the synergy of both transport and compression could provide good solutions.

Scalable video coding and transport over broadband wireless networks

With the emergence of broadband wireless networks and increasing demand of multimedia information on the Internet, wireless multimedia services are foreseen to become widely deployed in the next decade. Real-time video transmission typically has requirements on quality of service (QoS). However, wireless channels are unreliable and the channel bandwidth varies with time, which may cause severe degradation in video quality. In addition, for video multicast, the heterogeneity of receivers makes it difficult to achieve efficiency and flexibility. To address these issues, three techniques, namely, scalable video coding, network-aware adaptation of end systems, and adaptive QoS support from networks, have been developed. This paper unifies the three techniques and presents an adaptive framework, which specifically addresses video transport over wireless networks. The adaptive framework consists of three basic components: (1) scalable video representations; (2) network-aware end systems; and (3) adaptive services. Under this framework, as wireless channel conditions change, mobile terminals and network elements can scale the video streams and transport the scaled video streams to receivers with a smooth change of perceptual quality. The key advantages of the adaptive framework are: (1) perceptual quality is changed gracefully during periods of QoS fluctuations and hand-offs; and (2) the resources are shared in a fair manner.

Error recovery for interactive video transmission over the Internet

Real-time interactive video transmission in the current Internet has mediocre quality because of high packet loss rates. Loss of packets in a video frame manifests itself not only in the reduced quality of that frame but also in the propagation of that distortion to successive frames. This error propagation problem is inherent in any motion compensation-based video codec. In this paper, we present a new error recovery scheme, called recovery from error spread using continuous updates (RESCU), that effectively alleviates error propagation in the transmission of interactive video. The main benefit of the RESCU scheme is that it allows more time for transport-level recovery such as retransmission and forward error correction to succeed while effectively masking out delays in recovering lost packets without introducing any playout delays, thus making it suitable for interactive video communication. Through simulation and real Internet experiments, we study the effectiveness and limitations of our proposed techniques and compare their performance to that of existing video error recovery techniques including H.263+ (NEWPRED). The study indicates that RESCU is effective in alleviating the error spread problem and can sustain much better video quality with less bit overhead than existing video error recovery techniques under various network environments.

Real-time transmission of multilayer video over ATM networks

With today's network technology, our expectations for high performance networked multimedia applications are increasing. However, no complete solution as yet exists for providing integrated services to networked multimedia systems. The difficulty lies in the nature of the multimedia data, basically video and audio, which is difficult to store and transmit. The coded video is sensitive to delay, delay variation, bit errors and losses. Our research focuses on the transmission of multilayer real-time video traffic over ATM (Asynchronous Transfer Mode) networks. This issue covers several aspects of ATM and multimedia networking such as video application layer, ATM adaptation layers, traffic and buffer management. The main assumption in this research is that the real-time video traffic is content based coded and consists of several video objects. A single video source with various video objects introduces the need for multiple priorities. We propose a multipriority traffic management model for ATM networks that includes a new priority cell mapping algorithm, a new priority based bandwidth allocation approach, a new adaptation layer definition, and a new multipriority buffer access control algorithm. Using multipriority traffic will complicate the traffic management, the buffer management and the work of the ATM adaptation layers. On the other hand, by allowing the network to have several priority layers during a connection, the overall quality for real-time video will be increased. To our knowledge, no method exists for managing a single connection for multipriority video traffic over ATM networks. Our model is new and addresses the essential transmission requirements of near future multipriority video coding systems over broadband networks.

A practical rate control algorithm for VBR MPEG-2 video transmission over ATM networks

A joint encoder and channel rate control system for real-time VBR video transmission over ATM networks is proposed. The proposed system consists of channel rate control and encoder rate control. The channel rate over unit GOP is allocated from the proposed 3-state Markov chain model, in which the states are transient according to video traffic. Given the channel rate, encoder rate control is performed by solving the minimum distortion equation using Lagrange multiplier. Experimental results show that our method maintains consistent quality, as compared with the conventional real-time implementations, especially when there exist scene changes in the image sequences.

Computer science using an interactive video classroom

Austin Community College is a publicly funded college that has the responsibility of serving the needs of students over a relatively large geographic area. Our college has six main campuses. At each campus we have an interactive video classroom that uses a network of compressed video units to allow real-time transmission of audio and video. While these rooms may be used for meetings and teleconferences, the primary purpose of these rooms is to simultaneously offer classes at multiple sites.

Home telecare system using cable television plants--an experimental field trial.

To solve the inconvenience of routine transportation of chronically ill and handicapped patients, this paper proposes a platform based on a hybrid fiber coaxial (HFC) network in Taiwan designed to make a home telecare system feasible. The aim of this home telecare system is to combine biomedical data, including three-channel electrocardiogram (ECG) and blood pressure (BP), video, and audio into a National Television Standard Committee (NTSC) channel for communication between the patient and healthcare provider. Digitized biomedical data and output from medical devices can be further modulated to a second audio program (SAP) subchannel which can be used for second-language audio in NTSC television signals. For long-distance transmission, we translate the digital biomedical data into the frequency domain using frequency shift key (FSK) technology and insert this signal into an SAP band. The whole system has been implemented and tested. The results obtained using this system clearly demonstrated that real-time video, audio, and biomedical data transmission are very clear with a carrier-to-noise ratio up to 43 dB.

Surveillance system based on data fusion from image and acoustic array sensors

The present publication shows a model of a monitoring system of high benefit, based on two complementary sensorial systems (image+acoustics), which allows the information received to fuse, and to prioritize the computational resources toward the zones with considerable risk. The application uses components such as: active acoustic radar based on a phase array of microphones, with techniques of beamforming, and multibeam detection schemes; efficient compression algorithms and video transmission in real time. The system is composed of an arbitrary number of sensors (monitoring probes), distributed by the monitoring space, a data network and a central server (monitoring manager). The sensors consist of a microprocessor, with its memory and three main modules: (1) acoustic radar; (2) video capture card; and (3) network card. The flexibility of the system allows new modules to be added such as: personal detector card; sound card; cards for telecontrol; etc. The sensor microprocessor preprocesses the information received by the modules, packs and transmits it to the central server through a data network. The system uses IP protocol, and SNMPv2 for management.

Robust image transmission over CDMA channels

The transmission of real-time images and video over wireless communication channels is a challenging problem. Digital compressed images are sensitive to bit errors which are typical in wireless communications. Moreover, the bandwidth at the air interface is currently a limiting factor because the first and second generation of mobile phone standards mainly support voice communications. We present a study of real-time image traffic over a radio link-the aim of this research is for videophone applications. We use the discrete wavelet transform (DWT) to compress images and a code division multiple access (CDMA) link to transfer images over wireless communications network. The experiment results indicate that it is possible to transfer 9-12 QCIF gray scale images per second over a CDMA link with graceful degradation in image quality.

Real-time Internet video using error resilient scalable compression and TCP-friendly transport protocol

We introduce a point to point real-time video transmission scheme over the Internet combining a low-delay TCP-friendly transport protocol in conjunction with a novel compression method that is error resilient and bandwidth-scalable. Compressed video is packetized into individually decodable packets of equal expected visual importance. Consequently, relatively constant video quality can be achieved at the receiver under lossy conditions. Furthermore, the packets can be truncated to instantaneously meet the time varying bandwidth imposed by a TCP-friendly transport protocol. As a result, adaptive flows that are friendly to other Internet traffic are produced. Actual Internet experiments together with simulations are used to evaluate the performance of the compression, transport, and the combined schemes.

Rate control for robust video transmission over burst-error wireless channels

We study the problem of rate control for transmission of video over burst-error wireless channels, i.e., channels such that errors tend to occur in clusters during fading periods. In particular we consider a scenario consisting of packet based transmission with automatic repeat request (ARQ) error control and a back channel. We start by showing how the delay constraints in real time video transmission can be translated into rate constraints at the encoder, where the applicable rate constraints at a given time depend on future channel rates. With the acknowledgments received through the back channel we have an estimate of the current channel state. This information, combined with an a priori model of the channel, allows us to statistically model the future channel rates. Thus the rate constraints at the encoder can be expressed in terms of the expected channel behavior. We can then formalize a rate distortion optimization problem, namely, that of assigning quantizers to each of the video blocks stored in the encoder buffer such that the quality of the received video is maximized. This requires that the rate constraints be included in the optimization, since violating a rate constraint is equivalent to violating a delay constraint and thus results in losing a video block. We formalize two possible approaches. The first one seeks to minimize the distortion for the expected rate constraints given the channel model and current observation. The second approach seeks to allocate bits so as to minimize the expected distortion for the given model. We use both dynamic programming and Lagrangian optimization approaches to solve these problems. Our simulation results demonstrate that both the video distortion at the decoder and packet loss rate can be significantly reduced when incorporating the channel information provided by the feedback channel and the a priori model into the rate control algorithm.