A survey of digital television broadcast transmission techniques

This paper is a survey of the transmission techniques used in digital television (TV) standards worldwide. With the increase in the demand for High-Definition (HD) TV, video-on-demand and mobile TV services, there was a real need for more bandwidth-efficient, flawless and crisp video quality, which motivated the migration from analogue to digital broadcasting. In this paper we present a brief history of the development of TV and then we survey the transmission technology used in different digital terrestrial, satellite, cable and mobile TV standards in different parts of the world. First, we present the Digital Video Broadcasting standards developed in Europe for terrestrial (DVB-T/T2), for satellite (DVB-S/S2), for cable (DVB-C) and for hand-held transmission (DVB-H). We then describe the Advanced Television System Committee standards developed in the USA both for terrestrial (ATSC) and for hand-held transmission (ATSC-M/H). We continue by describing the Integrated Services Digital Broadcasting standards developed in Japan for Terrestrial (ISDB-T) and Satellite (ISDB-S) transmission and then present the International System for Digital Television (ISDTV), which was developed in Brazil by adopteding the ISDB-T physical layer architecture. Following the ISDTV, we describe the Digital Terrestrial television Multimedia Broadcast (DTMB) standard developed in China. Finally, as a design example, we highlight the physical layer implementation of the DVB-T2 standar

Customized television: Standards compliant advanced digital television

This correspondence describes a European Union supported collaborative project called CustomTV based on the premise that future TV sets will provide all sorts of multimedia information and interactivity, as well as manage all such services according to each user’s or group of user’s preferences/profiles. We have demonstrated the potential of recent standards (MPEG-4 and MPEG-7) to implement such a scenario by building the following services: an advanced EPG, Weather Forecasting, and Stock Exchange/Flight Information

An Efficient Method to Improve the Audio Quality Using AAC Low Complexity Decoder

This paper presents a new approach to design a Digital Audio Broadcast (DAB) audio decoder is introduced to improve the superiority of audio. Countries all over the world use DAB broadcasting systems more prominently, in Europe. DAB+ is the upgraded version of digital audio broadcasting. DAB and DAB+ coexist in many countries, so receivers are essential to be compatible with both standards. DAB+ is approximately twice as efficient as DAB due to the adoption of the AAC+ audio codec, and DAB+ can provide high quality audio with bit rates as low as 64 kbit/s. Integrating an MPEG-1 Layer II (MP2) decoder and Advanced Audio Coding Low Complexity (AAC LC) decoder provides a fundamental audio decoding for DAB and DAB+. The generated audio frames data from the DAB channel decoders are stored in RAM. The bit stream demultiplexer parses the quantized spectrum data in the audio. The inverse quantization performs the inverse quantization computation and synthesis filter generates the time domain Pulse Code Modulation (PCM) samples, all the above operation results writes them back to the audio RAM. The existing system of this project uses HE AAC V2 decoder, that system consists has SBR and PS technologies. This two technologies are used to improve the sound quality in low bit rate program. The proposed scheme is uses AAC LC and MP2 decoder it improve the sound quality in high bit rate. The simulation of this project is carried out by using MATLAB R2011a and Xilinx ISE 9.2i. DOI: 10.17762/ijritcc2321-8169.15039

Transport of video over partial order connections

A Partial Order and partial reliable Connection (POC) is an end-to-end transport connection authorized to deliver objects in an order that can differ from the transmitted one. Such a connection is also authorized to lose some objects. The POC concept is motivated by the fact that heterogeneous best-effort networks such as Internet are plagued by unordered delivery of packets and losses, which tax the performances of current applications and protocols. It has been shown, in several research works, that out of order delivery is able to alleviate (with respect to CO service) the use of end systems’ communication resources. In this paper, the efficiency of out-of-sequence delivery on MPEG video streams processing is studied. Firstly, the transport constraints (in terms of order and reliability) that can be relaxed by MPEG video decoders, for improving video transport, are detailed. Then, we analyze the performance gain induced by this approach in terms of blocking times and recovered errors. We demonstrate that POC connections fill not only the conceptual gap between TCP and UDP but also provide real performance improvements for the transport of multimedia streams such MPEG video

Efficiency in audio processing : filter banks and transcoding

Audio transcoding is the conversion of digital audio from one compressed form A to another compressed form B, where A and B have different compression properties, such as a different bit-rate, sampling frequency or compression method. This is typically achieved by decoding A to an intermediate uncompressed form, and then encoding it to B. A significant portion of the involved computational effort pertains to operating the synthesis filter bank, which is an important processing block in the decoding stage, and the analysis filter bank, which is an important processing block in the encoding stage. This thesis presents methods for efficient implementations of filter banks and audio transcoders, and is separated into two main parts. In the first part, a new class of Frequency Response Masking (FRM) filter banks is introduced. These filter banks are usually characterized by comprising a tree-structured cascade of subfilters, which have small individual filter lengths. Methods of complexity reduction are proposed for the scenarios when the filter banks are operated in single-rate mode, and when they are operated in multirate mode; and for the scenarios when the input signal is real-valued, and when it is complex-valued. An efficient variable bandwidth FRM filter bank is designed by using signed-powers-of-two reduction of its subfilter coefficients. Our design has a complexity an order lower than that of an octave filter bank with the same specifications. In the second part, the audio transcoding process is analyzed. Audio transcoding is modeled as a cascaded quantization process, and the cascaded quantization of an input signal is analyzed under different conditions, for the MPEG 1 Layer 2 and MP3 compression methods. One condition is the input-to-output delay of the transcoder, which is known to have an impact on the audio quality of the transcoded material. Methods to reduce the error in a cascaded quantization process are also proposed. An ultra-fast MP3 transcoder that requires only integer operations is proposed and implemented in software. Our implementation shows an improvement by a factor of 5 to 16 over other best known transcoders in terms of execution speed

Multimedia delivery in the future internet

The term “Networked Media” implies that all kinds of media including text, image, 3D graphics, audio and video are produced, distributed, shared, managed and consumed on-line through various networks, like the Internet, Fiber, WiFi, WiMAX, GPRS, 3G and so on, in a convergent manner [1]. This white paper is the contribution of the Media Delivery Platform (MDP) cluster and aims to cover the Networked challenges of the Networked Media in the transition to the Future of the Internet. Internet has evolved and changed the way we work and live. End users of the Internet have been confronted with a bewildering range of media, services and applications and of technological innovations concerning media formats, wireless networks, terminal types and capabilities. And there is little evidence that the pace of this innovation is slowing. Today, over one billion of users access the Internet on regular basis, more than 100 million users have downloaded at least one (multi)media file and over 47 millions of them do so regularly, searching in more than 160 Exabytes1 of content. In the near future these numbers are expected to exponentially rise. It is expected that the Internet content will be increased by at least a factor of 6, rising to more than 990 Exabytes before 2012, fuelled mainly by the users themselves. Moreover, it is envisaged that in a near- to mid-term future, the Internet will provide the means to share and distribute (new) multimedia content and services with superior quality and striking flexibility, in a trusted and personalized way, improving citizens’ quality of life, working conditions, edutainment and safety. In this evolving environment, new transport protocols, new multimedia encoding schemes, cross-layer inthe network adaptation, machine-to-machine communication (including RFIDs), rich 3D content as well as community networks and the use of peer-to-peer (P2P) overlays are expected to generate new models of interaction and cooperation, and be able to support enhanced perceived quality-of-experience (PQoE) and innovative applications “on the move”, like virtual collaboration environments, personalised services/ media, virtual sport groups, on-line gaming, edutainment. In this context, the interaction with content combined with interactive/multimedia search capabilities across distributed repositories, opportunistic P2P networks and the dynamic adaptation to the characteristics of diverse mobile terminals are expected to contribute towards such a vision. Based on work that has taken place in a number of EC co-funded projects, in Framework Program 6 (FP6) and Framework Program 7 (FP7), a group of experts and technology visionaries have voluntarily contributed in this white paper aiming to describe the status, the state-of-the art, the challenges and the way ahead in the area of Content Aware media delivery platforms

A framework for multimedia playback and analysis of MPEG-2 videos with FFmpeg

Fast Forward Motion Pictures Expert Group (FFmpeg) is a well-known, high performance, cross platform open source library for recording, streaming, and playback of video and audio in various formats, namely, Motion Pictures Expert Group (MPEG), H.264, Audio Video Interleave (AVI), just to name a few. With FFmpeg current licensing options, it is also suitable for both open source and commercial software development. FFmpeg contains over 100 open source codecs for video encoding and decoding. Given the complexities of MPEG standards, FFmpeg still lacks a framework for (1) seeking to a particular image frame in a video, which is needed for accurate annotation at the frame level for applications in fields such as medical domain, digital communications and commercial video broadcasting and (2) motion vectors extraction for analysis of motion patterns in video content. Most importantly, FFmpeg code base is not well documented, which has raised a significant difficulty for developing an extension. As our contributions, we extended FFmpeg code base to include new APIs and libraries support accurate frame-level seek, motion vector extraction, and MPEG-2 video encoding/decoding. We documented FFmpeg MPEG-2 codec to facilitate future software development. We evaluated the performance of our implementation against a high-performance third-party commercial software development kit on videos captured from television broadcasts and from endoscopy procedures. To evaluate the usability of our libraries, we integrated them with some commercial applications. In the following sections, we will discuss our software architecture, important implementation details, performance evaluation results, and lessons learned

Polymorphic computing abstraction for heterogeneous architectures

Integration of multiple computing paradigms onto system on chip (SoC) has pushed the boundaries of design space exploration for hardware architectures and computing system software stack. The heterogeneity of computing styles in SoC has created a new class of architectures referred to as Heterogeneous Architectures. Novel applications developed to exploit the different computing styles are user centric for embedded SoC. Software and hardware designers are faced with several challenges to harness the full potential of heterogeneous architectures. Applications have to execute on more than one compute style to increase overall SoC resource utilization. The implication of such an abstraction is that application threads need to be polymorphic. Operating system layer is thus faced with the problem of scheduling polymorphic threads. Resource allocation is also an important problem to be dealt by the OS. Morphism evolution of application threads is constrained by the availability of heterogeneous computing resources. Traditional design optimization goals such as computational power and lower energy per computation are inadequate to satisfy user centric application resource needs. Resource allocation decisions at application layer need to permeate to the architectural layer to avoid conflicting demands which may affect energy-delay characteristics of application threads. We propose Polymorphic computing abstraction as a unified computing model for heterogeneous architectures to address the above issues. Simulation environment for polymorphic applications is developed and evaluated under various scheduling strategies to determine the effectiveness of polymorphism abstraction on resource allocation. User satisfaction model is also developed to complement polymorphism and used for optimization of resource utilization at application and network layer of embedded systems