MPEG-2 TS Research Articles

Implementation of 4kUHD HEVC-content transmission

The Internet of things (IoT) has received a great deal of attention in recent years, and is still being approached with a wide range of views. At the same time, video data now accounts for over half of the internet traffic. With the current availability of beyond high definition, it is worth understanding the performance effects, especially for real-time applications. High Efficiency Video Coding (HEVC) aims to provide reduction in bandwidth utilisation while maintaining perceived video quality in comparison with its predecessor codecs. Its adoption aims to provide for areas such as television broadcast, multimedia streaming/storage, and mobile communications with significant improvements. Although there have been attempts at HEVC streaming, the literature/implementations offered do not take into consideration changes in the HEVC specifications. Beyond this point, it seems little research exists on real-time HEVC coded content live streaming. Our contribution fills this current gap in enabling compliant and real-time networked HEVC visual applications. This is done implementing a technique for real-time HEVC encapsulation in MPEG-2 Transmission Stream (MPEG-2 TS) and HTTP Live Streaming (HLS), thereby removing the need for multi-platform clients to receive and decode HEVC streams. It is taken further by evaluating the transmission of 4k UHDTV HEVC-coded content in a typical wireless environment using both computers and mobile devices, while considering well-known factors such as obstruction, interference and other unseen factors that affect the network performance and video quality. Our results suggest that 4kUHD can be streamed at 13.5 Mb/s, and can be delivered to multiple devices without loss in perceived quality.

The ATSC Link-layer Protocol (ALP): Design and Efficiency Evaluation

In this paper, a novel data link layer protocol adopted in the ATSC 3.0 terrestrial broadcast standard is described. The data link layer is a protocol layer between the physical layer and the network layer. The data link layer transports data from the network layer to the physical layer at the sending side and transports data from the physical layer to the network layer at the receiving side. The data link layer protocol in ATSC 3.0 is called the ATSC link-layer protocol (ALP). The major function of ALP is to take multiple input formats, not only IP packets but also legacy broadcast formats such as MPEG-2 TS packets, and output them in a common format to the physical layer. ALP provides overhead reduction mechanisms to achieve efficient transmission, particularly the compression of IP/UDP packet headers. The protocol and operation of ALP, as well as efficiency analysis and evaluation of ALP are presented.

Video Content-Based Bit Rate Estimation Scheme for Transcoding in IPTV Services

In this paper, a new bit rate estimation scheme is proposed to determine the bit rate for each subclass in an MPEG-2 TS to H.264/AVC transcoder after dividing an input MPEG-2 TS sequence into several subclasses. Video format transcoding in conventional IPTV and Smart TV services is a time-consuming process since the input sequence should be fully transcoded several times with different bit-rates to decide the bit-rate suitable for a service. The proposed scheme can automatically decide the bit-rate for the transcoded video sequence in those services which can be stored on a video streaming server as small as possible without losing any subject quality loss. In the proposed scheme, an input sequence to the transcoder is sub-classified by hierarchical clustering using a parameter value extracted from each frame. The candidate frames of each subclass are used to estimate the bit rate using a statistical analysis and a mathematical model. Experimental results show that the proposed scheme reduces the bit rate by, on an average approximately 52% in low-complexity video and 6% in high-complexity video with negligible degradation in subjective quality.

Modelling of H.264 MPEG2 TS Traffic Source

This paper deals with IPTV traffic source modelling. Traffic sources are used for simulation, emulation and real network testing. This model is made as a derivation of known recorded traffic sources that are analysed and statistically processed. As the results show the proposed model causes in comparison to the known traffic source very similar network traffic parameters when used in a simulated network.

Broadcast/multicast transmission of video signal in a WDM-PON using a low-cost injection-locked F-P LD with broadband ASE

A broadcast/multicast transmission method in a WDM-PON system using a low-cost Fabry-Perot laser diode with broadband ASE injection is proposed and experimentally demonstrated in this paper. Because of its low front-facet reflectivity (∼0.1%), the F-P LD exhibits a broadband lasing spectrum and reduced mode-partition noise. To demonstrate its performance, the bit error rate and packet loss rate were measured. The proposed optical source provides a cost-effective WDM-PON architecture for a broadcast/multicast service accommodating more than 32×622MBps. For the transmission of MPEG-2 TS packets, zero packet loss could be obtained at an optical power of -32dBm.

3DTV 방송을 위한 스테레오스코픽 비디오 영상 및 전송 포맷 기술규격의 검증

멀티미디어산업에 있어서 3DTV 서비스를 위한 스테레오스코픽 비디오는 Post-HD(post Hi호 Definition) 시대의 주요 콘텐츠로 부상하고 있으며, 콘텐츠 제작, 방송 및 가전 등 관련 산업 전반에 걸쳐 파급을 일으키며 3DTV 서비스 인프라를 구축시키고 있다. 이에 국내에서는 3DTV 방송 송수신 정합규격 표준화를 목표로 TTA에서 3DTV 송수신 규격 실무반(WG8061)을 구성하여, "3DTV 방송을 위한 스테레오스코픽 비디오 영상 및 전송 포맷 기술규격"의 표준안을 작성하고 있다. 본 논문에서는 현재 표준화 작업이 진행 중인 상기 표준규격을 적용한 test-bed를 설계하여, 규격의 기능을 검증하고 기존의 기술과의 역호환성에 대한 검증을 하며 향후 보완 및 검토되어야 하는 기술적 문제들을 도출하고자 한다. Stereoscopic video for 3DTV service is emerging as a main content of the Post-HD age in the multimedia industry, and also it starts a ripple effect to build infrastructure for 3DTV service throughout the related industries such as creating contents, consumer electronics and the broadcasting. Thus, "Technical Specification of Stereoscopic Video over MPEG-2 TS for 3DTV Service" for providing a standard for 3DTV broadcasting transmitter and receiver interface is under development by the 3DTV WG8061 in ITA, Korea. This paper provides verification of the functionalities, problems and future work of the above specification with designed test-bed. Through making a test bed, this paper try to verify not only the functionality of standard but also backward compatible for the present technology and abstract problems to would be reviewed and supplemented.

IP 망을 통한 MPEG-2 TS 기반의 프로그레시브 스트리밍을 위한 de-jitter 버퍼링 시간 추정 기법

In this paper, we propose an estimation of network jitter that occurs when transmitting TCP packets containing MPEG-2 TS in progressive streaming service over wired or wireless Internet networks. Based on the estimated network jitter size, we can calculate required de-jitter buffering time to absorb the network jitter at the receiver side. For this purpose, by exploiting the PCR timestamp existing in the TS packet header, we create a new timestamp information that is marked in the optional field of TCP packet header to estimate the network jitter. By using the proposed de-jitter buffering scheme, it is possible to employ the conventional T-STD buffer model without any modification in the progressive streaming service over IP networks. The proposed method can be applicable to the recently developed international standard, MPEG DASH (dynamic adaptive streaming over HTTP) technology.

1080p 60 Hz Intra-Frame Video CODEC Chip Design and Its Implementation

This paper presents a 1080p 60 Hz intra-frame CODEC system for zero delay AV streaming. For high quality streaming, the proposed CODEC employs an RGB-domain inter-color compensation algorithm using strong correlation between RGB color components which was previously presented by the authors of this paper. The proposed CODEC architecture is based on macroblock-level pipelining and parallel processing to handle a significant pixel rate of 1080p 60 Hz videos with RGB 4:4:4 format, i.e., 3 Gbps. Since syntax processing is a bottleneck to supporting speeds of up to 100 Mbps in real-time, a high performance context-adaptive variable length coding architecture exploiting the look-ahead technique is included in the proposed design. Also, the number of syntax symbols is adaptively restricted to accomplish zero end-to-end delay. Finally, by using MPEG-2 TS as the AV stream format, compatibility with general channel chips is guaranteed. Using TSMC 90 nm CMOS technology, the prototype chip is implemented with 1,208 K logic gates and 359 Kb internal SRAM. The chip can achieve real-time encoding and decoding of 1080p 60 Hz videos at 200 MHz execution speed in typical work conditions.

MPEG-2 TS로부터 IP 패킷을 구성하는 역다중화기 구현

This paper proposes an implementation of a hardware module for transmitting MPEG-2 TS data over the internet protocol (IP)-based network. This implementation consists of two modules; one is an encapsulation module which bridges between n TS packets, where , and an IP packets, the other is a packet conversion module which extracts an DSM-CC PS packet from consecutive TS packets and then reconstructing an IP packet. So, these IP packets are carried over 150 megabits per second. Although overall work flow of the proposed DeMUX is based on the reference design of ALTERA, the DeMUX is enhanced by modifying it and performs more functions by adding a packet conversion module. The DeMUX is described by Verilog-HDL (hardware description language) and shows the faithful functionality and throughput through the simulation.

선택적 암호화가 가능한 윈도우 미디어 보호 방법

IP 셋톱박스에 서비스되는 콘텐츠는 IP 네트워크를 통하여 전송되므로 네트워크 해킹 도구를 이용하면 스트리밍되는 콘텐츠를 저장후 불법이용이 가능하다. 따라서 안전한 스트리밍 서비스를 위해서는 스트리밍 환경에 적합한 콘텐츠 보호 방법이 요구된다. 최근까지 스트리밍 서비스되는 콘텐츠는 MPEG-2 TS를 이용하였지만, 낮은 압축률과 TS 패킷의 높은 오버헤드때문에 현재는 WMV, MPEG-4, H.264로 점차 옮겨가고 있다. 본 논문은 스트리밍되는 윈도우 미디어 콘텐츠를 보호하기 위한 DRM을 제안한다. 제안하는 방법은 기존의 WMV 스트리밍 시스템에 비의존적으로 설계되어 기존의 스트리밍 환경에 적용이 용이하고, 사용자 기기의 성능을 고려하여 DRM 처리 시간을 조절할 수 있도록 선택적 암호화 기법을 제공한다. 본 논문에서는 실험을 통하여 선택적 암호화를 이용하여 DRM 처리시간을 감소시킬 수 있음을 보인다. As content serviced for IP set-top boxes is streamed over IP network, the existing hacking tools for IP network can be used to capture the streamed content. Until recently, most of the content serviced on IP set-top boxes has been MPEG-2 TS. However, this content will be gradually moved to WMV, MPEG-4 or H.264 because of the relatively low compression efficiency and overhead of the TS packet. In this paper, we propose a DRM scheme other than WMRM for streamed WMV content. Our approach is to design a DRM scheme independent to the existing WMV streaming system. We also design this scheme in order to provide the feature for controlling the DRM processing time considering device performance. We verified it through the experiment.

Efficient Retransmission Architecture of Digital Broadcast Services over IPTV Networks

This paper describes architecture for supporting legacy digital broadcast services over IPTV networks. Legacy services refer to pre made real-time broadcast streams based on open standards such as MHP, ACAP, and OCAP. They use MPEG-2 TS as delivery protocol, but this is not a good choice for IPTV as it is defined in view of unidirectional media such as terrestrial or satellite networks. The proposed architecture retransmits digital broadcast services using IP-oriented protocols including service delivery, service discovery and selection, XML-based application signaling, and integrated application execution engines.

A cross-layered network-adaptive HD video streaming in digital A/V home network: channel monitoring and video rate adaptation

In this paper, we propose a cross-layered network-adaptive high definition (HD) video streaming system using a sender-driven wireless channel monitoring over the wireless local area network (WLAN) for the digital audio/visual (A/V) home network. For the wireless channel monitoring, an intelligent intermediate sender with an embedded wireless access point estimates the time-varying channel condition based on the observed medium access control (MAC) and physical (PHY) statistics. The observed channel condition is then utilized to dynamically guide the network-adaptive control of temporally-layered video flow (i.e., video frame rate control). For an implementation of the proposed system, a frame-based prioritized packetization using a real-time parsing module of the MPEG-2 TS stream is nicely provided. Various evaluation results over an IEEE 802.11a WLAN testbed are provided to verify the intended quality of service (QoS) adaptation capability. Experimental results showed that the proposed system can mitigate the quality degradation of video streaming due to the fluctuations of time-varying channel

An integrated home server for communication, broadcast reception, and home automation

This paper presents a home server integrating communication, digital broadcast reception, and home automation applications. The home server has a CPU of high performance, and an MPEG decoder as a PCI slave device, and a MPEG2 TS distributor to convey simultaneously three HDTV channels which may go their way from terrestrial, satellite, and cable tuners to the next destination equipment via Ethernet port and PCI bus. We also present the architecture of MPEG2 TS distributor.

지상파/케이블 데이터방송 재전송 서비스를 위한 PMCP 기반 PSIP 변환기

본 논문은 디지털방송 서비스의 필수 기술 요소이고 디지털방송 전환과 밀접하게 관련 있는 PSIP 응용 기술인 디지털방송 표준 인터페이스로 표준화된 PMCP(Program Metadata Control Protocol) 인터페이스 규격과, 이를 이용하여 구현된 지상파/케이블 데이터방송 재전송 서비스를 위한 PSIP 변환기의 구조 및 주요 기능에 대해서 기술한다. 본 논문의 PSIP 변환기는 지상파/케이블 데이터방송 재전송 서비스 환경하에서 디지털 지상파 데이터방송 콘텐츠를 디지털 케이블 데이터방송에 재사용하여 데이터방송 콘텐츠 활용도를 높일 뿐만 아니라 실시간 채널 정보 변경 및 방송 프로그램 이벤트 정보 변경만으로 지상파 방송프로그램 안내 정보를 쉽게 재활용/관리할 수 있다. 또한 저장/관리된 방송프로그램 안내 정보를 ATSC T3/Sl에서 표준화한 PMCP 규격을 이용하여 XML형태로 케이블 시스템(OOB SI, Out Of Band Service Information)으로 전송하거나, MPEG-2 전송스트림 형태로 전송할 수도 있다. In this paper, we implemented a terrestrial/cable PSIP converting system, so-called a PSIP converter, which is converting a terrestrial PSIP into a cable PSIP for a data broadcasting service in the interoperable network of terrestrial and cable, and define an interface between the PSIP converter and the OOB SI generator by using PMCP messages compliant to ATSC T3/Sl. The exiting PSIP converter just converts a terrestrial PSIP into a cable PSIP compliant to ATSC and OCAP standard and transmits by a MPEG-2 TS format. That is to say, it is not for the digital data broadcasting but for the digital broadcasting. In addition, the PSIP converter can support various types of PSIP information to the OOB SI generator by using PMCP messages defined by a hierarchical structure as per each channel, audio/video event, data event and so on.

複数搬送波分割伝送方式による高速ＭＰＥＧ‐２ ＴＳ信号のケーブルテレビ伝送

This paper proposes a novel division transmission method for using multiple carriers to send a highspeed MPEG-2 TS signal that cannot be sent by conventional methods within one cable TV channel (6 MHz bandwidth) transmission capacity.To meet the above requirement, we developed a flexible absorption technique of time lag between divided TS signals, and devised a new descriptor for the MPEG-2 TS. We made an experimental transmitter and a receiver, and evaluated transmission performance of these devices. Our tests show that the receiver can absorb the time lag between divided signals up to 1.36 ms, and this corresponds to a difference of more than 272 km in the transmission distance. Furthermore, experiments with prototype devices proved that the new descriptor enables the receiver to re-compose the original high-speed MPEG-2 TS automatically.

A lightweight encapsulation protocol for IP over MPEG-2 networks: design, implementation and analysis

Digital TV Broadcast standards (e.g., ATSC, DVB) offer a good solution to last mile Internet delivery. Satellites, terrestrial broadcast, and cable systems are well suited to the delivery of high bandwidth data to the end user. Low transport overhead and ready availability of MPEG-2 TS equipment make it very appealing as a bearer for connecting IP networks. The Multi-Protocol Encapsulation (MPE) is a standard for the transport of IP over DVB. However, MPE has a number of shortcomings and inefficiencies when supporting next generation Internetworking systems. This paper describes a new method for the encapsulation of IPv4/IPv6 datagrams, known as the Ultra Lightweight Encapsulation (ULE) that overcomes limitations of MPE. The paper provides the rationale for developing ULE and provides measurements and analysis to show that the new scheme can offer additional functionality while achieving increased transmission efficiency.

지상파 DMB용 Outer 인코더/리코더의 설계 및 구현

본 논문에서는 차세대 디지털 방송규격인 지상파 DM용 Outer 인코더/디코더를 설계하고 ALTERA의 FPGA를 이용하여 구현하고 검증하였다. 인코더 부분에서는 입력되는 MPEG-2 TS 패킷(188바이트)으로부터 비트 시리얼 알고리즘을 이용한 RS(Reed-Solomon) 인코더를 이용해 패리티 바이트(16바이트)를 생성하고 군집에러를 효과적으로 수정하기 위해 콘볼루션 인터리버를 구현해 데이터를 분산 출력 시켰다. 디코더 부분에서는 인코더에서 송신된 데이터에서 DMB에 적합한 동기 바이트 검출하는 알고리즘을 제시하였으며, RS디코더는 수정된 유클리드 알고리즘을 적용하여 회로구성을 간략화 하였다. 본 시스템은 하나의 패킷에서 최대 8바이트의 에러를 수정할 수 있고, C언어를 이용하여 알고리즘을 검증하고 VHDL로 작성하였으며, FPGA 칩 상에서 회로를 검증하였다. In this paper, we designed the outer encoder/decoder for the terrestrial DMB that is an advanced digital broadcasting standard, implemented, and verified by using ALTERA FPGA. In the encoder part, it was created the parity bytes (16 bytes) from the input packet (188by1e) of MPEG-2 TS and the encoded data was distributed output by the convolutional interleaver for Preventing burst errors. In the decoder part, It was proposed the algorithm that detects synchronous character suitable to DMB in transmitted data from the encoder. The circuit complexity in RS decoder was reduced by applying a modified Euclid's algorithm. This system has a capability to correct error of the maximum 8 bytes in a packet. After the outer encoder/decoder algorithm was verified by using C language, described in VHDL and implemented in the ALTERA FPGA chips.

サーバ型放送におけるコンテンツのアクセス制御方式

This paper presents an RMP (Rights Management and Protection) system for Digital Broadcasting Based on Home Servers to be used with receivers and home storage devices. We clarified the requirements required for rights protection of contents in Digital Broadcasting Based on Home Servers, and developed an RMP system that meets those requirements. This RMP system enables rights protection and advanced access control (term of validity, usage and charge condition, etc.) of content stored on home storage devices. We have also proposed a method for multiplexing scramble keys synchronous with MPEG-2 TS video packets, to achieve a non-linear playback function of contents encrypted by the developed RMP system. We confirmed that non-linear playback and trick-play function of contents encrypted by the developed RMP system can be processed in real time.

Service information for reception and re-composition of divided high-speed MPEG-2 TS transmitted by multiple carriers

Service information for reception and re-composition of divided high-speed MPEG-2 TS transmitted by multiple carriers

Scheduling algorithm for MPEG-2 TS multiplexers in CATV networks

We study the scheduling algorithm for multiplexers in CATV networks. First we propose the scheduling matrix concept, and then two kinds of scheduling algorithms are given, which are described with the scheduling matrix. Especially, we focus on the VBR scheduling algorithm being able to achieve a better display of the MPEG-2 program in CATV networks. The scheduler applying our VBR scheduling algorithm maintains, for each input stream, a counter to determine how many TS packets should be transmitted in the current scheduling cycle (one frame period) in order to prevent deadline violation. The real-time bit rate of each input stream is used to update the parameters of the scheduling matrix. Our VBR scheduling algorithm is compared numerically with the traditional CBR scheduling algorithm. The results of the simulation show that our VBR scheduling algorithm can achieve significantly lower delay, i.e., deadline violation probability, in comparison with the CBR scheduling algorithm. Also it is shown that a higher utilization of bandwidth can be achieved by means of our VBR scheduling algorithm than by the CBR scheduling algorithm.