MPEG Video Streams Research Articles

Adaptive video streaming: pre-encoded MPEG-4 with bandwidth scaling

The increasing popularity of streaming video is a cause for concern for the stability of the Internet because most streaming video content is currently delivered via UDP, without any end-to-end congestion control. Since the Internet relies on end systems implementing transmit rate regulation, there has recently been significant interest in congestion control mechanisms that are both fair to TCP and effective in delivering real-time streams. In this paper we design and implement a protocol that attempts to maximize the quality of real-time MPEG-4 video streams while simultaneously providing basic end-to-end congestion control. While several adaptive protocols have been proposed in the literature [An End-to-End rate-based congestion control mechanism for real-time streams in the Internet, in: Proceedings of INFOCOMM’99, March 1999; IEEE Trans. Multimedia 3 (2001) 339], the unique feature of our protocol, the Video Transport Protocol (VTP), is its use of receiver side bandwidth estimation. Such estimation is transmitted to the sender and enables it to adapt to network conditions by altering its sending rate and the bitrate of the transmitted video stream. We deploy our protocol in a real network testbed and extensively study its behavior under varying link speeds and background traffic profiles using the FreeBSD Dummynet link emulator [Dummynet and Forward Error Correction, in: Proceedings of Freenix’98. June 1998]. Our results show that VTP delivers consistent quality video in moderately congested networks and fairly shares bandwidth with TCP in all but a few extreme cases. We also describe some of the challenges in implementing an adaptive video streaming protocol.

A DCT-based MPEG-2 transparent scrambling algorithm

Recently, several encryption algorithms have been proposed, which are applied to MPEG-2 video streams. These algorithms try to optimize the encryption process and improve encryption speed by exploiting the spatial and temporal properties of video retrieval and display process. As a new branch of encryption, transparent scrambling has its own merits and special prospects. A new transparent scrambling algorithm proposed in this paper performs an arbitrarily degraded view of decoded picture by encrypting in DCT domain. This technique is applicable to the broadcast systems such as DVB/ATSC, which use MPEG-2 as compression standard.

Reduced energy decoding of MPEG streams

Long battery life and high performance multimedia decoding are competing design goals for portable appliances. For a target level of QoS, the achievable battery life can be increased by dynamically adjusting the supply voltage throughout execution. In this paper, an efficient offline scheduling algorithm is proposed for preprocessing stored MPEG audio and video streams. It computes the order and voltage settings at which the appliance's CPU decodes the frames, reducing energy consumption without violating timing or buffering constraints. Our experimental results elucidate the trade-off between QoS and energy consumption. They demonstrate that the scheduler reduces CPU energy consumption by 19%, without any sacrifice of quality, and by nearly 50%, with only slightly reduced quality. The results also evaluate how the QoS/energy trade-off is affected by buffering and processor speed.

Prediction of MPEG-coded video source traffic using recurrent neural networks

Predicting traffic generated by multimedia sources is needed for effective dynamic bandwidth allocation and for multimedia quality-of-service (QoS) control strategies implemented at the network edges. The time-series representing frame or visual object plane (VOP) sizes of an MPEG-coded stream is extremely noisy, and it has very long-range time dependencies. This paper provides an approach for developing MPEG-coded real-time video traffic predictors for use in single-step (SS) and multistep (MS) prediction horizons. The designed SS predictor consists of one recurrent network for I-VOPs and two feedforward networks for P- and B-VOPs, respectively. These are used for single-frame-ahead prediction. A moving average of the frame or VOP sizes time-series is generated from the individual frame sizes and used for both SS and MS prediction. The resulting MS predictor is based on recurrent networks, and it is used to perform two-step-ahead and four-step-ahead prediction, corresponding to multistep prediction horizons of 1 and 2 s, respectively. All of the predictors are designed using a segment of a single MPEG-4 video stream, and they are tested for accuracy on complete video streams with a variety of quantization levels, coded with both MPEG-1 and MPEG-4. Comparisons with SS prediction results of MPEG-1 coded video traces from the recent literature are presented. No similar results are available for prediction of MPEG-4 coded video traces and for MS prediction. These are considered unique contributions of this research.

QoS‐oriented traffic management in multimedia satellite systems

AbstractThe next generation of satellite systems will likely use some on‐board switching techniques enabling the direct transmission of data from any up‐link spot‐beam to any down‐link spot‐beam. If the switch used in the payload of the system is an ATM‐based shared buffer switch, some losses and delays can occur. Fortunately, these drawbacks can be fought thanks to some buffer management policies (HSPO, SMXQ, EPD) that either ensure low delay or low loss ratio.Real‐time (voice or MPEG video streams) and non‐real‐time applications (file transfer, email, web browsing using TCP/IP protocols) should be carried over these satellite systems with different quality of service (QoS) requirements. Different combinations of buffer management policies are presented to guaranty the QoS required by the user. Especially, special emphasis is put on the way to transmit efficiently layer 3 messages (IP datagrams) within a mix of traffic in order to avoid some high layer retransmission techniques (provided in TCP for instance) wasting both time and bandwidth. Copyright © 2003 John Wiley & Sons, Ltd.

Performance Characterization of Wireless Multimedia through Heterogeneous Terrestrial-Satellite Network

In this work the transmission of multimedia services through heterogeneous integrated terrestrial-satellite networks is analyzed by means of measurement on a real test-bed including a video on demand (VoD) server, an IEEE802.11 wireless local area network (WLAN) integrated with a wired LAN, and a geostationary Ka-band satellite link in service at the Telecommunication Laboratory at IEIIT-BO/CNR, University of Bologna, Bologna, Italy. At first the WLAN radio link is characterized in terms of the mean received power level for different terminal locations within the Lab. Secondary, MPEG-1 video streams are transmitted with different data-rate from the VoD server to many wireless stations and the traffic, for both the single source and the aggregated traffic, is characterized. An integrated terrestrial-satellite LAN is simulated as an example of utilization of source characterization. Moreover, the quality of service in terms of the average dropped MPEG frames-to-transmitted frames ratio is measured for different channel conditions, data-rate, and the number of wireless terminals requiring the service.

Objective quality assessment of MPEG-2 video streams by using CBP neural networks

The increasing use of compression standards in broadcasting digital TV has raised the need for established criteria to measure perceived quality. Novel methods must take into account the specific artifacts introduced by digital compression techniques. This paper presents a methodology using circular backpropagation (CBP) neural networks for the objective quality assessment of motion picture expert group (MPEG) video streams. Objective features are continuously extracted from compressed video streams on a frame-by-frame basis; they feed the CBP network estimating the corresponding perceived quality. The resulting adaptive modeling of subjective perception supports a real-time system for monitoring displayed video quality. The overall system mimics perception but does not require an analytical model of the underlying physical phenomenon. The ability to process compressed video streams represents a crucial advantage over existing approaches, as avoiding the decoding process greatly enhances the system's real-time performance. Experimental evidence confirmed the approach validity. The system was tested on real test videos; they included different contents ranging from fiction to sport. The neural model provided a satisfactory, continuous-time approximation for actual scoring curves, which was validated statistically in terms of confidence analysis. As expected, videos with slow-varying contents such as fiction featured the best performances.

A new paradigm for analysis of MPEG compressed videos

The increasing use of multimedia streams nowadays necessitates the development of efficient and effective methodologies for manipulating databases storing these streams. Moreover, content-based access to multimedia databases requires in its first stage to parse the video stream into separate shots then apply a method to summarize the huge amount of data involved in each shot. This work proposes a new paradigm capable of robustly and effectively analyzing the compressed MPEG video data. First, an abstract representation of the compressed MPEG video stream is extracted and used as input to a neural network module (NNM) that performs the shot detection task. Second, we propose two adaptive algorithms to effectively select key frames from segmented video shots produced by the segmentation stage. Both algorithms apply a two-level adaptation mechanism in which the first level is based on the dimension of the input video file while the second level is performed on a shot-by-shot basis in order to account for the fact that different shots have different levels of activity. Experimental results show the efficiency and robustness of the proposed system in detecting shot boundaries and flashlights occurring within shots and in selecting the near optimal set of key frames (KFs) required to represent each shot.

A new paradigm for analysis of MPEG compressed videos

The increasing use of multimedia streams nowadays necessitates the development of efficient and effective methodologies for manipulating databases storing these streams. Moreover, content-based access to multimedia databases requires in its first stage to parse the video stream into separate shots then apply a method to summarize the huge amount of data involved in each shot. This work proposes a new paradigm capable of robustly and effectively analyzing the compressed MPEG video data. First, an abstract representation of the compressed MPEG video stream is extracted and used as input to a neural network module (NNM) that performs the shot detection task. Second, we propose two adaptive algorithms to effectively select key frames from segmented video shots produced by the segmentation stage. Both algorithms apply a two-level adaptation mechanism in which the first level is based on the dimension of the input video file while the second level is performed on a shot-by-shot basis in order to account for the fact that different shots have different levels of activity. Experimental results show the efficiency and robustness of the proposed system in detecting shot boundaries and flashlights occurring within shots and in selecting the near optimal set of key frames (KFs) required to represent each shot.

Objective assessment of MPEG-2 video quality

The increasing use of video compression standards in broadcasting television systems has required, in recent years, the development of video quality measurements that take into account artifacts specifically caused by digital compression techniques. In this paper we present a methodology for the objective quality as- sessment of MPEG video streams by using circular back- propagation feedforward neural networks. Mapping neural networks can render nonlinear relationships between objective features and subjective judgments, thus avoiding any simplifying assumption on the complexity of the model. The neural network processes an in- stantaneous set of input values, and yields an associated estimate of perceived quality. Therefore, the neural-network approach turns objective quality assessment into adaptive modeling of subjective perception. The objective features used for the estimate are chosen according to the assessed relevance to perceived quality and are continuously extracted in real time from compressed video streams. The overall system mimics perception but does not require any ana- lytical model of the underlying physical phenomenon. The capability to process compressed video streams represents an important ad- vantage over existing approaches, like avoiding the stream- decoding process greatly enhances real-time performance. Experi- mental results confirm that the system provides satisfactory, continuous-time approximations for actual scoring curves concern- ing real test videos. © 2002 SPIE and IS&T.

ＶＬＩＷ型メディアプロセッサを用いたＭＰＥＧ‐２オールフォーマットビデオデコーダ

An all-format decoder (AFD), which is an MPEG-2 MP@HL video decoder, has been developed for digital television. It uses a media accelerated processor for consumer appliances 2000 (MAPCA2000), which is based on the very-long-instruction-word architecture. The AFD decodes an MPEG-2 video stream using some of the discrete cosine transform coefficients in the input stream. The algorithm it uses to decode the stream reduces the amount of memory and processor power required. By efficiently using single-instruction multiple-data instructions, a direct-memory-access engine, and a co-processor for variable-length decoding, the AFD can decode 1080i-format video streams in real time using a standard MAPCA2000 running on a 300-MHz CPU with a 100-MHz external-memory bus clock and only 3.5 MB of external memory.

A new transport protocol for broadcasting/multicasting MPEG-2 video over wireless ATM access networks

Because of the telecommunications de-regulation act and progress in wireless technologies, we will see the co-existence of heterogeneous broadband access infrastructures in the broadband video service industry in the near future. In this paper, we addressed the error control issue when transmitting MPEG-2 video streams over wireless access networks for broadband video broadcast or multicast services. An end-to-end transport protocol based on ATM and wireless ATM technologies is proposed. For video services, the underlying transport network should be transparent and quality should be maintained uniformly over all the segments whether wireline or wireless links. For network resources to be used efficiently, error control should be applied locally on the wireless segments so as to avoid the excessive overhead over the reliable wireline portions. Because a broadband video broadcast or multicast service is a one-to-multiple point service, FEC is the most prevalent error control mechanism. Due to the important role of MPEG-2 control information in the decoding process, priority MPEG-2 control information has to be differentiated from MPEG-2 data information, and excess error protection has to be allocated to it in order to achieve satisfactory QoS. Therefore, a header redundancy FEC (HRFEC) scheme for error control is applied at the local distribution centers before the MPEG-2 encoded video streams are transmitted over the wireless channels. HRFEC is an FEC-based selective protection scheme, which allocates extra error protection to important control information. Simulation results show that the quality of the reconstructed video sequence is vastly improved by using HRFEC, when the channel condition is poor.

A platform-neutral live IP/TV presentation system

We propose a novel object-oriented design approach for developing a platform-neutral IP/TV presentation system for playing live TV-like MPEG video streams on the Internet multicast by RTP-compliant servers. Unlike proprietary viewers, such as Cisco's IP/TV viewers, which run only on Windows platforms, our system runs on any Java Virtual Machine environments. We demonstrate the effectiveness of our design approach via actual experimentations. We also propose a Java-based relay agent architecture for relaying RTP streams across multicast-enabled routers. We show that our system is particularly suitable to run as relay agents in this architecture to tackle the cross-platform issue and to accommodate various audio or video media sources efficiently. By adhering to the Internet and multimedia standards, our system is generically applicable to various media streaming systems.

RQRP protocol for video delivery over IP

AbstractThe Digital Audio‐Visual Council has adopted the Request/Reply (RQRP) protocol, proposed by one of the authors, for the delivery mechanism of MPEG video streams. In this paper, we investigate the network characteristics of the RQRP protocol and analyse the protocol's performance via both simulation as well as actual experimental results. In this simulation, the RQRP protocol was mathematically an ideal model first and later, a feedback control system. Simulation results showed that, for both models, the network traffic is able to follow the playout rate at the receiver. In the case of the latter, which involves a simpler implementation, oscillation in network traffic is observed. A video‐on‐demand (VoD) system using the RQRP protocol was developed and experiments were performed. The results obtained are consistent with the simulation results. Copyright © 2001 John Wiley & Sons, Ltd.

Supporting MPEG video VBR traffic in wireless networks

In this paper, we analyse the statistical data sets of several MPEG encoded videos and propose a model of the elements of a scene. We also propose and evaluate a novel dynamic bandwidth allocation scheme for MPEG video sources suitable for wireless networks. The proposed model permits the characterisation of the elements of the stream scenes. The proposed scheme is dynamic and pro-active. It automatically adjusts the amount of reserved resources, while guarantying the required QoS. It exploits the structure of the MPEG video stream and allocates bandwidth on a scene basis. The dynamic bandwidth allocation algorithm which has been presented is evaluated using simulation and actual MPEG video data. The performance evaluations showed a major improvement in bandwidth utilisation as compared to other proposed schemes.

MPEG video streaming with VCR-functionality

With the proliferation of online multimedia content, the popularity of multimedia streaming technology, and the establishment of MPEG video coding standards, it is important to investigate how to efficiently implement an MPEG video streaming system. Digital video cassette recording (VCR) functionality (such as random access, fast forward, fast reverse, etc.) enables quick and user-friendly browsing of multimedia content, and thus is highly desirable in streaming video applications. The implementation of full VCR functionality, however, presents some technical challenges that have not yet been well resolved. We investigate the impacts of the VCR functionality on the network traffic and the video decoder complexity. We propose a least-cost scheme for the efficient implementation of MPEG streaming video system to provide full VCR functionality over a network with minimum requirements on the network bandwidth and the decoder complexity. We also discuss our implementation of an IP-based MPEG-4 video streaming platform which provides full VCR-functionality.

High-Quality Video Delivery over Local Area Networks with Application to Teaching at a Distance

Multimedia communication over networks is important in distance learning, computer-based multimedia and telecommunications. The quality of the video stream, the throughput of the communication and the time delay of the delivery must be considered. This paper describes some experimental work undertaken to optimise the issues that influence data loss and hence image quality, and transfer delay in MPEG video streams transferred over Local Area Networks (LAN).

AMISP: a complete content-based MPEG-2 error-resilient scheme

We address a new error-resilient scheme for broadcast quality MPEG-2 video streams to be transmitted over lossy packet networks. A new scene-complexity adaptive mechanism, namely Adaptive MPEG-2 Information Structuring (AMIS) is introduced. AMIS modulates the number of resynchronization points (i.e., slice headers and intra-coded macroblocks) in order to maximize the perceived video quality, assuming that the encoder is aware of the underlying packetization scheme, the packet loss probability (PLR), and the error-concealment technique implemented at the decoding side. The end-to-end video quality depends both on the encoding quality and the degradation due to data loss. Therefore, AMIS constantly determines the best compromise between the rate allocated to encode pure video information and the rate aiming at reducing the sensitivity to packet loss. Experimental results show that AMIS dramatically outperforms existing structuring techniques, thanks to its efficient adaptivity. We then extend AMIS with a forward-error-correction (FEC)-based protection algorithm to become AMISP. AMISP triggers the insertion of FEC packets in the MPEG-2 video packet stream. Finally, the performances of the AMISP scheme in an MPEG-2 over RTP/UDP/IP scenario are evaluated.

Scalable layered MPEG-2 video multicast architecture

Nowadays, multimedia systems disseminating real-time video information over the Internet are becoming increasingly popular. However, the heterogeneity of network components and end systems, and the lack of congestion control in the Internet pose a major challenge in delivering video. In this paper, a layered multicast architecture is presented to achieve efficient distribution of MPEG-2 video streams over the Internet. The video data is partitioned in a hierarchical manner and transmitted in different multicast channels. The participants automatically subscribe to a number of channels based upon the network congestion and the capabilities of the end systems. This architecture provides scalability on both bandwidth availability level and on the end system capability level.

Contention-free distributed dynamic reservation MAC protocol with deterministic scheduling (C-FD/sup 3/R MAC) for wireless ATM networks

This paper proposes a novel MAC protocol for wireless ATM networks, which is characterized by a contention-free mechanism of the reservation request and a deterministic nature of mobile-assisted (distributed) uplink scheduling under a framework of the dynamic reservation TDMA, as discussed in the current standardization activities of ETSI Project BRAN (broadband radio access network) and the wireless ATM working group in the ATM Forum. The design objective of the proposed MAC protocol is to guarantee the real-time constraint of the real-time VBR (rt-VBR) traffic class while maximizing the multiplexing gain among all ATM traffic classes, especially with a fixed length frame. The proposed deterministic scheduling scheme for the rt-VBR traffic class lends itself to implementing the minimal configuration of control data units for reservation request as desired under the limited wireless resources. Simulation experiments using statistically multiplexed MPEG-2 video streams are performed for a 25 Mbits/s wireless ATM access link scenario. It has been shown that the proposed framework guarantees the delay constraint of rt-VBR sessions along with its cell loss rate significantly reduced, while improving the average delay performance of the nrt-VBR in the range of 10%-30% without compromising the channel utilization as compared to the DSA++ system.