MPEG-2 Transport Stream Research Articles

스테레오스코픽 비디오 방송 서비스를 위한 MPEG-2 전송스트림 구성 방안

본 논문은 3차원 입체효과를 나타내는 기술 중 가장 널리 쓰이고 있는 스테레오스코픽 영상을 방송 환경에서 활용되어지고 있는 MPEG-2 전송스트림(Transport Stream : TS)을 통해 효율적으로 서비스를 하기 위한 방법을 제안한다. 전통적인 MPEG-2 전송 스트림은 오디오, 비디오, 그리고 부가 데이터로 구성된 프로그램들을 각 데이터간의 동기 및 타이밍 기능을 제공하고 다중화함으로써 오디오, 비디오 및 데이터의 실시간 전송을 가능하게 한다. 그러나, 이러한 MPEG-2 전송스트림은 2D 영상을 기준으로 제정된 기술 규격으로 스테레오스코픽 영상을 위한 동기화, 다중화 처리 및 스테레오스코픽 동영상 식별이 어렵기에 스테레오스코픽 영상 서비스로 활용하기에는 어려움이 존재한다. 본 논문에서는 전통적인 MPEG-2 전송스트림의 2D 비디오 서비스와 호환성을 가지며 스테레오스코픽 비디오 서비스를 제공할 수 있도록 스테레오스코픽 비디오의 식별, 다중화 처리 및 동기화 처리를 위한 알고리듬을 제시하고 실제 구현을 통해 검증하였다. Stereoscopic video service is one of the most widely adopted technologies for 3-Dimensional technology. However, the stereoscopic video service has not been penetrated into the market since there does not exist a proper transmission method even though the conventional MPEG-2 Transport Stream (TS) has support for broadcast service of mono-scopic video contents. This is partly because the MPEG-2 TS does not support displaying a stereoscopic video content which consists of several composition types widely used in the market. This paper proposes the new MPEG-2 TS specification suitable for transmitting and displaying stereoscopic video contents. In order to show feasibility of the proposal, this paper describes an end-to-end system which is implemented with consideration of the method how to multiplex and de-multiplex stereoscopic video contents as well as keeping backward compatibility with the traditional MPEG-2 TS.

Scalable video coding over RTP and MPEG-2 transport stream in broadcast and IPTV channels

The ITU-T and ISO/IEC standard for scalable video coding was recently finalized. SVC allows for scalability of the video bitstream in the temporal, spatial, or fidelity domain, or any combination of those. Video scalability may be used for different purposes, such as saving bandwidth when the same media content is required to be sent simultaneously on a broadcast medium at different resolutions to support heterogeneous devices, when unequal error protection shall be used for coverage extension in wireless broadcasting, as well as for rate shaping in IPTV environments. Furthermore, it may also be useful in layered multicast transmission over the Internet or peer-to-peer networks, or in any transmission scenario where prioritized transmission for network flows is meaningful. In order to make usage of SVC in the aforementioned use cases, standards for defining the transport format and procedure are required. Therefore, we give a detailed overview of the recently finished SVC standards on transport over IP/RTP and the MPEG-2 transport stream. Both standards are important for IPTV and video on demand, where the first is important for SVC transport over mobile broadcast/multicast channels, and the latter is also important for SVC transport over traditional digital broadcast channels.

디지털방송 자막데이터를 이용한 멀티미디어 응용 서비스 연구

In this paper, we study on making a use of value-added services using DTV closed caption data. Note that Closed-Captioning service helps to bridge “digital divide” through extending broadcasting accessibility of a neglected class such as hearing-impaired person and foreigner. In Korea, DTV Closed Captioning standard was developed in June 2007, and Closed Captioning service should be provided by an enforcing law in all broadcasting services in April 2008. Here, we describe the method of extracting a caption data from MPEG-2 Transport Stream of ATSC-based digital TV signal and generating a caption file using the extracted caption data and time information. In addition, we present the segmentation method of broadcasting content using caption file. Experimental results show that implemented S/W tool provides the feasibility of the proposed methods and the usability of closed caption for a variety of data application service.

Towards an RTP Profile for IPTV

In this paper we analyze the two major transport protocols that are currently under discussion for IPTV and propose extensions to support convergence. On the one hand, the MPEG-2 transport stream (MPEG2-TS) is well established in the broadcast world, whereas the real-time transport protocol (RTP) is widely adopted in the Internet and mobile world. Our analysis focuses on the timing aspects such as synchronization and buffer management. It is shown that the system target decoder (STD) defined in MPEG2-TS provides a well defined decoder model that allows minimum buffering delay and predictable behavior. In contrast, an RTP receiver is intentionally defined less accurately to allow flexibility for multiple applications. However, this flexibility results in less clear receiver behavior which is a problem for the design and integration of IPTV systems. By comparing the main differences in the timing models of the two protocols we derive additional constraints and definitions for RTP that minimize the difference to MPEG2-TS. It is proposed to specify these changes as part of an RTP/IPTV profile which also defines an RTP system target decoder (RTP-STD). An example for the application of these extensions is a protocol gateway supporting the convergence of broadcast and the Internet world in IPTV.

Continuous Adaptation Enhancement in Linear Regression Algorithm for Clock Recovery in IPTV Environment

This paper presents a new technique for Clock Recovery in MPEG-2 Transport Streams in DVB-IPTV environment. The amount of jitter in this environment is very high compared to DVB-S or DVB-T. Use of linear regression algorithm for clock synchronization over packet switched networks is suggested by researchers in the similar field. However, the stability of such a filter in real time set top box application posed a problem. The continuous adaptive feedback loop used with the linear regression algorithm provides accurate clock synchronization even when subjected to jitter of half a second. This robustness is needed for IP environment because of the increasing network traffic. The algorithm was simulated and found to require low computation time, high jitter attenuation and fast response as needed for the IP network. It is auto-adaptive for both low and high jitter environments by analysing the jitter present in the stream. The algorithm was implemented for real time application in IPTV set top box and tested to be stable in the RTOS environment.

A QoS Monitoring System in a Heterogeneous Multi-Domain DVB-H Platform

The MobileTV, IPTV, and DVB standards (DVB-H/T) have been defined to offer mobile users interactive multimedia services with quality of service (QoS) consistency analogous to TV services. However, the market has yet to provide effective and economical solutions for the real-time delivery of such services to the corresponding transmitters over multi-domain IP networks. The monitoring system proposed in this paper enables the QoS in the IP networks involved in the delivery of real-time multimedia content to the transmitters to be ascertained. The system utilizes the QoS parameters defined in MPEG-2 Transport Streams to detect problems occurring in the heterogeneous multi-domain IP networks. The ability to detect problems having an adverse effect on QoS allows appropriate control actions to be determined to recover the QoS across the composite IP network. The design and implementation of the proposed QoS-Monitoring system (QoS-MS) is presented, followed by analysis of experimental results that demonstrate the feasibility of the system.

An Embedded and Programmable System Based FPGA for Real Time MPEG Stream Buffer Analysis

The MPEG transport stream is an extremely complex structure using interlinked tables and coded identifiers to separate the programs and the elementary streams. Quality management is therefore a complicated issue and the need to identify the degree of coding degradations in terms of coding and/or transmission errors or system failures is becoming an important criterion for the evaluation of the quality of the MPEG streams. A theoretical decoder (T-STD) defines the verification process based on the proper fill level of an MPEG decoder buffers whose size is defined by the standards in order to obtain an evaluation of the MPEG stream quality. This paper describes a new embedded and programmable solution capable of analysing MPEG streams in real time. The proposed hardware architecture provides a real time continuous buffer analysis of the MPEG stream components and is composed of several modules allowing for simultaneous modeling of the various buffers of the T-STD components (video, audio or system). Real time errors flags are generated when the buffers filling level becomes illegal (overflow, empty buffer, transfer delay, etc.). The architecture has been modeled, validated and simulated using the SystemC and VHDL languages in combination with real MPEG DVB-T streams. A VHDL synthesisable model of our architecture allows an implementation on an field-programmable gate array circuit based on Altera APEX20K1000. The hardware implementation of this configurable T-STD allows a data rate of 232 Mbps and requires only 9738 logical cells and 4,7 kB memory.

Novel Clock Recovery Module for MPEG-2 Transport Stream in Terrestrial Television

The paper presents an efficient low cost clock recovery module used in the Set Top Box for terrestrial television. It addresses the effects of jitter on MPEG-2 Transport streams for DVB-T. Lot of research on algorithms for IP and ATM networks has been carried out but the problem for terrestrial networks has not been sufficiently addressed. These algorithms require more CPU time and involve complex calculations which are difficult to implement in set top boxes due to the real time system constraints and cost considerations. A moving window FIR filter approach has been used in our module. This module is simulated in terrestrial and satellite environment and observed that it performs efficiently under high jittery conditions. This method was also implemented on a set top box SOC (system-on-chip) and it demonstrated high performance.

실시간 다시점 고화질 비디오 전송 시스템을 위한 동기화된 다중화 기법

This paper proposes a prototype realization of multi-view HD video transport system with synchronized multiplexing over IP networks. The proposed synchronized multiplexing considers the synchronization during video acquisition and the multiplexing for the interactive view-selection during transport. For the synchronized acquisition from multiple HDV camcorders through IEEE 1394 interface, we estimate the timeline differences among MPEG-2 compressed video streams by using global time of network between the cameras and a server and correct timelines of video streams by changing the time stamp of the MPEG-2 system stream. Also, we multiplex a selected number of acquired HD views at the MPEG-2 TS (transport stream) level for the interactive view-selection during transport. Thus, with the proposed synchronized multiplexing scheme, we can display synchronized HD view.

GSE: A flexible, yet efficient, encapsulation for IP over DVB‐S2 continuous generic streams

AbstractThe second‐generation specification for Digital Video Broadcast over Satellite, DVB‐S2, defines an improved and adaptive physical layer. A new framing structure at the link layer, known as the generic stream (GS), is introduced offering an alternative to the well‐known MPEG transport stream (TS). This paper presents the requirements for and the design of an encapsulation protocol to transport IPv4 and IPv6 datagrams and other network protocol packets directly over DVB‐S2 using this GS profile. The resulting generic stream encapsulation (GSE) is a novel method that provides flexible encapsulation with support for fragmentation. A flexible extension header format allows GSE to carry additional header information, suitable for enhanced features, such as link layer encryption and IP header compression. The paper assesses GSE performance by simulation using realistic traffic profiles and attenuation fading. This demonstrates considerable improvement in efficiency compared with the MPE/MPEG‐2 TS, especially when used with adaptive coding and modulation (ACM). Although designed for DVB‐S2 systems, the new encapsulation is suitable for other second‐generation physical layer standards. Copyright © 2008 John Wiley & Sons, Ltd.

Real-time synchronous multi-view video transport system over IP networks

This paper proposes a prototype realization of real-time synchronous multi-view video transport system which can extend the software-based multi-view video transport system over IP networks. This system supports H.264/MPEG-2 system based real-time multi-view video transport and considers the synchronization during acquisition and transport for the interactive view-selection during transport. For the synchronized acquisition from multiple multi-view cameras through IEEE 1394 interface, this system estimates the timeline differences among H.264 compressed video streams by using global time of network between the cameras and a server and corrects timelines of video streams by changing the timestamp of the MPEG-2 system stream. Then, it multiplexes a selected number of acquired views at the MPEG-2 TS (transport stream) level for the interactive view-selection during transport. Thus, with the proposed synchronized multiplexing scheme, this system can display synchronized view.

Remultiplexing Eureka-147 Ensemble With MPEG-2 TS for T-DMB Multimedia Service

This paper introduces a layered structure of the Korean terrestrial digital multimedia broadcasting (T-DMB) ensemble based on Eureka-147 and suggests a new transmission architecture for flexible T-DMB service. In the suggested transmission architecture, MPEG-2 transport stream (TS) for T-DMB multimedia service is inserted into the ensemble distributed via the ensemble transport interface (ETI). Thus, we propose a novel algorithm remultiplexing the ETI with MPEG-2 TS. The architecture of ETI remultiplexing algorithm is described as a four-step process, and implemented as real-time embedded software in an ETI remultiplexing system. The design and implementation of the interoperable ETI remultiplexing system available for a small-scale transmission system for T-DMB service is described in detail.

A Multi-channel HD Video Streaming in Digital Community Network: Video Rate Adaptation and Playback Adjustment

In this paper, we propose design and implementation of a practical end-to-end system which implements multi-channel digital television (DTV) multicasting to the home over broadband community access network using sender-driven video rate adaptation and receiver-driven playback adjustment. For video rate adaptation, we employ Program Clock Reference (PCR) embedded in the MPEG-2 transport stream (TS) to enhance packet timing control precision and regulate the transmission rate in a refined way. At the same time, we integrate Traffic Control (TC) ingeniously to separate streams of multiple channels at the network card of DTV server and thus avoid bandwidth competition between them. For playback adjustment, we introduce control theory and over-boundary strategy to reduce buffer jitter and maximize the visual quality of displayed video according to the receiver buffer occupancy. We test our system with standard definition (SD) and high definition (HD) MPEG-2 streams and find that the proposed algorithms can reduce overflow and underflow of sender and receiver buffer, and achieve better video quality and more concurrent channels than traditional solutions.

Design and Implementation of a Software Presenting Information in DVB Subtitles in Various Forms

Subtitle data carried by MPEG-2 transport stream includes texts of dialogs in bitmap graphics format. However, this data is not reachable for the blind audience. In addition, carrying extra compatible data with subtitle packets can be helpful for testing subtitle software automatically and for presenting live content (i.e. latest news or special advertisements). In this paper, we propose a method for converting subtitle data into text format by recognizing characters using neural networks. The converted data can be used for blind television users and testing subtitle software automatically. We also define a new subtitle segment for carrying test comparison data and a new subtitle page for representing live content. We present our implementation on Linux platform.

Monitoring and Troubleshooting in Operational IP-TV System

When deploying large scale of QoS guaranteed IP-TV services over Triple-Play networks, there is a growing demand to access to the live network services, monitor in real time the broadcast chains, acquire the dynamic network parameters to ensure an uninterrupted, high quality service. Considering the difference of IP packet switching network and the de-facto transport stream (TS) standard of broadcast digital video network, this paper describes design consideration and implementation of a powerful embedded IP-TV monitor for real time monitoring the IP-TV QoS and using this equipment to construct a payload-aware monitoring network for on-line troubleshooting when the QoS deteriorate. The monitor should be designed to have ability to monitor and troubleshoot the blended combination of network status and digital TV quality in real time and cross layer way, meanwhile the procedures of monitoring does not interfere with the on-line services. This monitor can be deployed in the networks to measure the three ultimate metrics: IP cumulative jitter, packet loss and MPEG TS errors and provide helpful information about the network and service when service failed. Implementation and experimental results show promise of being practical in the real IP-TV trial networks.

Adaptive Streaming of MPEG-based Audio/Video Content over Wireless Networks

This paper describes a method for robust streaming of combined MPEG audio/video content (encoded either with MPEG-2 or MPEG-4/H.264) over in-home wireless networks. We make use of currently used content distribution format (MPEG Transport Stream) and network protocols (either RTP or TCP/HTTP). The transmitted bit-rate is constantly adapted to the available network bandwidth, such that audio and video artifacts caused by packet loss are avoided. Bit-rate adaptation is achieved by using a packet scheduling technique called I-Frame Delay (IFD), which performs priority-based frame dropping upon insufficient bandwidth. We show an implementation using RTP and an implementation using TCP. Measurements on a real-life demonstrator set-up demonstrate the effectiveness of our approach.

Experience with SDI Contribution over IP Network

The growth of internet protocol (IP)-based service in the national and international markets has been tremendous, and the traffic volume is comparable to that of ordinary voice At the same time, the cost of IP technology is considerably less than that of asynchronous transfer mode (ATM) and synchronous digital hierarchy (SDH). The consequence is that telecom operators are constructing new IP networks with the intent to convert all services to IP. The result for the broadcaster is that it will be more economical to base new contribution and distribution networks on IP. The Pro-MPEG Forum has proposed two standards for transport of video over IP. Code of Practice #3 (CoP3) covers transport of compressed video in the form of MPEG-2 transport streams. This standard is well advanced, and equipment is available from several vendors, as was demonstrated at the SMPTE/VidTrans 2006 Conference. Code of Practice #4 (CoP4) is similar to CoP3 and concerns transport of uncompressed video¯serial digital interface (SDI). This paper describes the experience with using CoP4 for contribution. In addition, the paper includes a description of the standard and a tradeoff for SDI transport.

Method for providing live content during playback of recorded streams in personal video recorders

Personal video recorders provide the facility of recording live broadcast streams thus users can replay their recordings whenever they want. However, during playback of those recorded streams, users are isolated from the live content (e.g., latest news) in which they might be interested. In this paper, we propose an MPEG-2 transport stream compatible method which can inform the user about live content that's in user interest area during playback. The method-specific structures, algorithms and an implementation on a Linux platform are presented in the paper

Unidirectional Lightweight Encapsulation: Performance Evaluation and Application Perspectives

Multi Protocol Encapsulation (MPE) is today the dominant method for encapsulating IP data into the MPEG-2 Transport Stream so they can be conveyed over digital television platforms (DVB or ATSC). Unidirectional Lightweight Encapsulation (ULE) is a new alternative to MPE, providing simplicity, efficiency and configurability. This paper presents the principle and benefits of ULE, approaches the issues associated with its adoption and compares it against MPE through extensive performance measurements in a fully functional testbed under IPv4 and IPv6 load. It also proposes, implements and demonstrates the combination of MPEG-4/H.264, IPv6 and ULE as the most efficient and flexible protocol stack for next-generation transmission of digital video over Digital Television networks

A single receiving chip for DVB data broadcasting system

In this paper, we present a single-chip receiver for DVB data broadcasting system, which is designed using the methodology of hardware and software co-design. The hardware part filters out and descrambles the necessary MPEG-2 transport stream (TS) packets, then sends them to PC. Some useful peripheral functions such as PCI interface, I <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">2</sup> C bus controller and smart card 7816 interface are also developed to provide a cost-effective solution. Meanwhile, multi-protocol decapsulation related to data broadcasting protocol is analyzed and implemented mainly by software. In order to improve the throughput and hardware utilization, we adopt chaining mode direct memory access (DMA) scheme and a double buffer approach both in hardware and in software. Moreover, the parameters of chaining mode DMA and the depth of buffer are further optimized by theoretical analysis. The result indicates the throughput of chaining mode DMA is up to 475.8 Mb/s, which is 24 times as much as non-chaining mode DMA strategy