A Century of Wireless
263	The studio J.J. Bottom and B. Marks (628 KB)
	In the embryonic days of sound broadcasting, the microphone and the transmitter were located in the same room. It was not long however before the radio studio developed as a separate entity to the transmitter. Over the decades, tremendous developments took place in the studio but, as the first century of wireless draws to a close, the trend now – at least with local-area radio stations – is to combine the studio once more with the transmitter site, in a space no larger than most living rooms. In this brief review, the authors have attempted to cover just some of the key technical developments which have taken place in the radio studio over the years. The topics considered include the microphone, sound mixers, recording and playback devices, stereo programming and studio acoustics.
263	The transmitter R.E. Fenn (730 KB)
	This article outlines the major developments which have taken place in sound transmitter equipment and techniques – from the first experiments before the invention of the thermionic valve, up to the present time. Over the years, high-power transmitters have developed into very sophisticated systems. Their progress was initially driven by a thirst for ever-increasing power and audibility. Today, like with many other things, their development is driven by the quest to reduce capital and operating costs.
263	The receiver J. Hill (565 KB)
	This article reviews the major trends and developments which have taken place in the design of wireless receivers, from the crystal set of the 1920s to the arrival in the 1960s of cheap transistorised radios, manufactured in the Far East. In the early days of sound broadcasting, a domestic wireless manufacturing industry evolved in many of the developed countries of the world. Thus, although this article relates mainly to British-made wireless sets, it is acknowledged that similar receivers were made in other countries.
263	International frequency regulation and planning F.M. Woolley (2.7 MB)
	At the origin of the regulation of radiocommunications was the problem of harmful interference, and the need to improve the safety of life at sea. As demand for spectrum grew, the need to use it more efficiently was a further incitation to regulation, which came to include frequency planning. This article describes the development of the regulation and planning of the use of the radio frequency spectrum, with emphasis on those aspects of special relevance to broadcasting. The adaptation of ITU structures to the changing requirements of this task is described, including the changes adopted in 1992, and the current moves to simplify the Radio Regulations.
263	From the coherer to DSP M. Lemme and R. Menicucci (996 KB)
	This article reviews the development of electronic devices used over the last century in wireless communication. It looks at early receiving devices such as the coherer, the magnetic detector and the cat's whisker, progressing to the thermionic valve, the semiconductor, the microchip and digital signal processing. On the transmission side, the early devices discussed include the spark-gap generator, the voltaic-arc generator and static frequency multipliers. This is followed by a brief description of more modern power devices, including thermionic valves and electron-velocity control tubes.
263	A century of trust in Mother Nature R. Levey (156 KB)
	The 100th anniversary of Marconi's first long-distance wireless transmissions is inevitably a pretext for many to claim their share of the credit for inventing sound broadcasting. While interest is centred on the pioneers and their creative experimentation, it is important to reflect also on the uses that have been made of this revolutionary technology, and perhaps to ponder on the last great mystery of wireless transmission.
263	Six great pioneers of wireless M. Meyer (403 KB)
	As stressed elsewhere in this issue, no one person was responsible for "inventing" wireless. It is broadly agreed that the works of Faraday, Maxwell and Hertz where vital in laying the foundations for Guglielmo Marconi, who was the first to exploit the practical applications of electromagnetic waves. To complement the portraits given here of these four great pioneers, the experimental works of Lodge and Popov are also reviewed. Some authorities feel that Lodge's important contribution to the history of wireless has been sadly neglected over the years while, in Russia, Popov is regarded as the inventor of radio communication. To complete this review, a brief chronology of important events in the history of wireless is given, from the discovery of static electrical charges around 600 B.C. to the European and International adoption of a single standard for tomorrow's wireless system – Digital Audio Broadcasting (DAB).
Access Services
300	Access Services for digital television Frans de Jong (738 KB)
	The number of disabled people in the European Union is growing. Currently 10% of the population is estimated to have a disability [1], including a large number of people with sensory disabilities. By the year 2020, it is estimated that 25% of the inhabitants will be over 60 [2], with the largest increase in the 75+ age band, where disability is most prevalent. This article outlines the choices available to broadcasters when starting access services over digital television platforms
Acoustics
274	Progress in concert hall design — developing an awareness of spatial sound and learning how to control it Robert. Essert (785 KB)
	The propagation of sound is a function of both time and space: our hearing and perception of sound are sensitive to spatial as well as temporal attributes. This article traces the development of spacial acoustics in the design of halls during the late 20th century, in relation to the advancement of acoustical knowledge and related technologies. An outline is given of current directions in modelling and measurement systems that may lead to a greater understanding of which spatial sound fields are preferred for different events, and how the geometrical form can influence them.
Application Programming Interface (API)
275	MHEG-5 and Java — the basis for a common European API? Allan Mornington-West (175 KB)
	The use of different proprietary APIs in digital television receivers is leading to a fragmented market in which the consumers are losing out, while the broadcasters battle to achieve exclusive ownership of a primary gateway to the viewer. The Author stresses the need for an open universal API and describes how this could be achieved using the MHEG-5 content decoder in conjunction with a Java-based Virtual Machine layer. He also describes a way forward to enable a practical migration from the use of existing proprietary APIs to the use of a single universal API.
Audio (levels & loudness)
310	Level and distortion in digital broadcasting Thomas Lund (653 KB)
	CD mastering – the most seasoned digital audio discipline – has turned into a loudness war rather than a quest for getting decent audio quality out of a potentially well-sounding media. Maximum loudness is also becoming a goal in itself in new movies, so that film operators are having to turn down the replay level to avoid complaints from the audience. In general, when audio normalization is based on peak-level detection, material with narrow dynamic range ends up the loudest. CD production not only relies on a peak-level measure (i.e. measurement scheme), it relies on a particularly bad and simplistic one, allowing massive amounts of distortion to be generated downstream of the studio in data-reduction systems and consumer equipment. The purpose of this article is to justify and recommend more fitting ways of measuring and controlling the audio level in digital broadcasting than looking at isolated samples or quasi-peak levels. The new ITU-R BS.1770 standard, specifying long-term loudness and peak-level detection, is evaluated and a centre of gravity approach to loudness control is suggested. Metadata associated with Dolby AC3 is shown to be insufficient at tackling the level and distortion issues across broadcast platforms, while legitimate control practices may be derived more cheaply and without ambiguity using statistical descriptors and real-time metering derived from BS.1770.
299	Loudness control — at the television playout stage John Emmett (300 KB)
	This article on Loudness control – while representing the views of the author – is based on a discussion paper submitted to the 5th Meeting of EBU Project Group P/AGA (Advisory Group on Audio), held at BBC R&D in December 2003.
297	Levelling and Loudness — in radio and television broadcasting Gerhard Spikofski and Siegfried Klar (810 KB)
	Sudden differences in loudness between – and even within – radio and television programmes has been well known for a long time. With the more-recent introduction of digital techniques, combined with the parallel transmission of digital and analogue broadcasts, this problem is again becoming highly significant. This article presents some solutions for avoiding loudness differences in radio and television broadcasting, based on levelling recommendations and a newly-developed loudness algorithm.
296	DAB and CD quality — reality or illusion Gerhard Spikofski and Siegfried Klar (652 KB)
	This article reports on the results of an investigation carried out into whether the transmitted sound quality offered by Digital Radio (DAB) stations in Germany is superior to that of FM radio. The tests revealed that not all is as it should be, with many stations not conforming with the relevant ARD recommendations for DAB broadcasters.
293	Audio levels — in the new world of digital systems John Emmett (330 KB)
	In this short article, the author describes some of the difficulties encountered with setting audio levels and loudness in the new digital environment.
293	Loudness and Dynamic Range in broadcast audio — the Dolby solution Tony Spath (582 KB)
	Digital delivery media offer a wider dynamic range for audio than their analogue predecessors. This entails adopting a larger difference between the average levels (and thus the implied loudness) and the signal peaks. Although it is possible to implement this larger difference in a TV station or media studio, problems will occur in the home – due to inconsistent loudness and electrical levels within the consumer receivers and audio equipment. One audio delivery system includes specific tools to overcome these problems, while allowing the full dynamic range of digital audio to be delivered. These tools – Dialogue Normalization and Dynamic Range Control – are described here with particular reference to digital TV.
Audio (multichannel)
2008-Q1	Microphone systems for Surround Sound pickup — and their use at Wimbledon tennis and The Proms
	Bill Whiston (246 kB)
	This article briefly describes some of the microphones developed specifically for Surround Sound pickup, along with several of the main Surround acquisition systems on which the majority of the dedicated Surround mics are based. It offers some personal advice on whether a particular system is suitable for use in this recording environment or that. Some microphone systems are obviously more intrusive “in shot” than others, depending on the location. The author also describes two major outside broadcasts that have involved Surround Sound mixes – the Wimbledon Tennis Championships and the BBC Proms Concerts from the Royal Albert Hall in London.
306	Audio in next-generation DVB broadcast systems Roland Vlaicu (196 KB)
	Broadcasters have significant new requirements for audio delivery in next-generation broadcast systems such as High-Definition Television. These include the capability to deliver soundtracks ranging from mono to 5.1 channels and beyond – with greater efficiency than with current systems, but also to maintain compatibility with existing consumer home cinema systems. A new audio delivery system, referred to as Enhanced AC-3 (marketing name: Dolby Digital Plus), has been developed to meet these requirements, and has been standardized in DVB and ATSC, referring to ETSI TS 102 366 V1.1.1 (2005-02).
297	The first European live radio broadcast in 5.1 surround Nikolaus Löwe (664 KB)
	Europe’s first satellite radio broadcast in “5.1 surround” took place from the Prix Europa competition in Berlin on 11 October 2003. This article outlines how the DTS 5.1 mix was produced in Berlin, distributed over the EBU’s Eurovision network, and delivered by Swedish Radio as a DVB-S satellite radio broadcast.
297	Multichannel audio — in the Digital Home John Emmett (498 KB)
	This article takes a light-hearted look at multichannel audio developments for the home, covering such technologies as Dolby Digital and DTS, and also looks at the current DVD format wars
292	The EBU's multichannel audio activities EBU Project Group P/MCA (251 KB)
	EBU Project Group P/MCA (Multichannel Audio) was set up to support the introduction of the 5.1 multichannel audio system for radio and television. The group has now reported and the results of its work are presented here.
292	Multichannel audio for television John Emmett (161 KB)
	Television sound can no longer be considered as a single entity. We will soon have viewers (listeners?) demanding “5.1” cinema-quality sound from every programme, whilst others – possibly the elderly or hard-of-hearing – wanting just the programme dialogue to be clearly reproduced from a tiny portable television. In this article, the author paints a picture of what he personally believes can be done with TV sound to cater for different user expectations – without making any fundamental changes to existing digital TV receivers, nor adding any significant costs at the production level.
292	How to get on-air with 5.1 audio — the Dolby® "5.1 Cookbook" for broadcasters Tony Spath (155 KB)
	This article is aimed at television broadcasters who want to go on-air with multichannel audio – using the Dolby® Digital (AC-3) audio delivery system. The necessary prerequisites to achieve this are described here, for different sections of the programme chain.
Broadcast Wave Format (BWF)
285	Five years in the history of audio files John. Emmett (197 KB)
	The EBU-developed Broadcast Wave Format has now been around for about five years. Here, the author takes a light-hearted look at this audio file format, which is now an AES standard.
274	The Broadcast Wave Format — an introduction Richard Chalmers (137 KB)
	This article provides a brief introduction to the new Broadcast Wave Format (BWF) file which has been developed by the EBU - in close collaboration with the audio industry - to facilitate the interchange of programme material between audio workstations.
274	The use of BWF files in Swedish radio Lars. Jonsson (212 KB)
	In this article, the author gives some background information on why the development of a common audio file format was essential for a radio broadcasting organization such as Swedish Radio (SR). BWF files will now be used in SR whenever audio workstations are interconnected via LANs and WANs.
Cable Distribution Networks
251	Noise and intermodulation in cable distribution networks K.N. Stokke (242 KB)
	This Tutorial looks at the problems associated with cable television network planning and, more especially, the calculation of noise and intermodulation ratios.
Coding (audio and video)
293	Windows Media 9 Series — a platform to deliver compressed audio and video for Internet and broadcast applications Jordi Ribas-Corbera (743 KB)
	Microsoft® Windows Media® 9 Series is a set of technologies that enables rich digital media experiences across all types of networks and devices. These technologies include an encoder to author the multimedia content, a server to distribute the content, a Digital Rights Management (DRM) system to let content owners set usage policies, and a variety of players to decode and render the content on personal computers and other consumer electronic devices. These components are built on top of a programmable and extensible platform that enables partners to build tailored applications and services. This article provides a high-level overview of the technologies in Windows Media 9 Series, with a particular focus on the different audio and video codecs available. Applications and services for broadcast (e.g., IP datacasting via DVB) are also discussed.
293	Broadcasting over the Web Kari Bulkley (684 KB)
	There are several different ways of distributing audio and video content over the Internet. You can encode it offline in any number of formats (Windows Media, Real, QuickTime etc) and host it on a web server for people to watch at their leisure. There may also arise a situation where you would want to do a live broadcast over the Internet, somewhat like a conventional television broadcast. There are many factors to consider when setting up for a live Internet broadcast – beginning with the available “live” encoding technologies. This article covers some of the many products available that will enable you to present a live audio and/or video broadcast over the Internet, with varying levels of complexity.
Coding (audio)
2008-Q3	Dolby Pulse — combining the merits of Dolby Digital and HE-AAC
	James Caselton (194 kB)
	In late 2007, Dolby Laboratories acquired Coding Technologies, the company which had developed techniques such as Spectral Band Replication (SBR) and Parametric Stereo (PS) for enhancing the efficiency of the Advanced Audio Coding (AAC) compression standard. This article outlines how Dolby Laboratories, Inc. has now integrated HE-AAC into the Dolby family to create a new audio coding system – called Dolby Pulse – for broadcasting and other applications where bandwidth is restricted.
306	Audio in next-generation DVB broadcast systems Roland Vlaicu (196 KB)
	Broadcasters have significant new requirements for audio delivery in next-generation broadcast systems such as High-Definition Television. These include the capability to deliver soundtracks ranging from mono to 5.1 channels and beyond – with greater efficiency than with current systems, but also to maintain compatibility with existing consumer home cinema systems. A new audio delivery system, referred to as Enhanced AC-3 (marketing name: Dolby Digital Plus), has been developed to meet these requirements, and has been standardized in DVB and ATSC, referring to ETSI TS 102 366 V1.1.1 (2005-02).
305	MPEG-4 HE-AAC v2 — audio coding for today's digital media world Stefan Meltzer and Gerald Moser (395 KB)
	Delivering broadcast-quality content to consumers is one of the most challenging tasks in the new world of digital broadcasting. One of the most critical aspects is the highly efficient use of the available transmission spectrum. Consequently, a careful choice of compression schemes for media content is essential – for both the technical and the economical feasibility of modern digital broadcasting systems. In the case of audio content, the MPEG-4 High Efficiency AAC v2 profile (HE-AAC v2) has proven, in several independent tests, to be the most efficient audio compression scheme available worldwide. It has recently been selected within DVB as part of its overall codec toolbox. HE-AAC v2 comprises a fully-featured tool set for the coding of audio signals in mono, stereo and multichannel modes (up to 48 channels) – at high quality levels using a wide range of bitrates.
304	Cascaded audio coding David Marston and Andrew Mason (527 KB)
	With the introduction of digital transmission, broadcasters have experienced significant problems with cascaded audio coding in the broadcast chain. It has been found that cascading different codecs can result in an overall degradation in the sound quality that many listeners find objectionable. A comprehensive investigation of this problem has been conducted by members of the EBU project group B/AIM. This article, based on a presentation given at IBC-2005, describes typical cascades of codecs found in radio broadcast chains, and aims to identify the most critical combinations. The intention is to guide broadcasters in deciding which codec combinations should be avoided in order to maximize the sound quality.
291	CT-aacPlus — a state-of-the-art audio coding system Martin Dietz and Stefan Meltzer (186 KB)
	CT-aacPlus is a combination of Spectral Band Replication (SBR) technology – a bandwidth-extension tool developed by Coding Technologies (CT) in Germany – with the MPEG Advanced Audio Coding (AAC) technology which, to date, has been one of the most efficient traditional perceptual audio-coding schemes. CT-aacPlus is able to deliver high-quality audio signals at bit-rates down to 24 kbit/s for mono and 48 kbit/s for stereo signals. The forthcoming Digital Radio Mondiale (DRM) broadcasting system, among others, will use CT-aacPlus for its audio-coding scheme. CT-aacPlus will enable DRM to deliver an audio quality, in the frequency range below 30 MHz, that is equivalent to – or even better than – that offered by today’s analogue FM services. This article describes the principles of traditional audio coders – and their limitations when used for low bit-rate applications. The second part describes the basic idea of SBR technology and demonstrates the improvements achieved through the combination of SBR technology with traditional audio coders such as AAC and MP3.
283	EBU listening tests on Internet audio codecs Gerhard Stoll and Franc Kozamernik (445 KB)
	The advent of Internet multimedia has stimulated the development of several advanced audio and video compression technologies. Although most of these developments have taken place outside the EBU, many members are using these low bit-rate codecs extensively for their webcasting activities, either for downloading or live streaming. To this end, the EBU Project Group, B/AIM (Audio in Multimedia), was asked to carry out some tests on several low bit-rate audio codecs that are now available on the commercial Internet market. This article gives the results of the subjective evaluations undertaken by B/AIM in late 1999 and early 2000. These EBU tests are the first international attempt at comparing the different audio compression schemes used on the Internet. In addition, prior to conducting these tests, no internationally-agreed subjective method was available for carrying out evaluations on very low bit-rate, intermediate-quality, codecs. In order to overcome this problem, the group was instrumental in devising a novel test method to evaluate specifically these low-quality audio codecs. The new method is now known as MUSHRA. Both the EBU and ITU-R have now adopted MUSHRA as a standard evaluation method.
283	An introduction to MPEG Layer-3 (MP3) K. Brandenburg and H. Popp (111 KB)
	MPEG Layer-3, otherwise known as MP3, has generated a phenomenal interest among Internet users, or at least among those who want to download highly-compressed digital audio files at near-CD quality. This article provides an introduction to the work of the MPEG group which was, and still is, responsible for bringing this open (i.e. non-proprietary) compression standard to the forefront of Internet audio downloads.
Coding (video)
2008-Q3	HDTV production codec tests
	Massimo Visca and Hans Hoffmann (808 kB)
	To address the need for more efficient HDTV studio compression systems, vendors have recently introduced new HDTV studio codecs. In 2007, an EBU project group investigated these codecs and this article describes the methodology used for the tests and summarizes the results obtained.
2008-Q2	SVC — a highly-scalable version of H.264/AVC
	Adi Kouadio, Maryline Clare, Ludovic Noblet and Vincent Bottreau (2.3 MB)
	Scalable Video Coding (SVC) is a recent amendment to the ISO/ITU Advanced Video Coding (H.264/AVC) standard, which provides optional but efficient scalability functionalities on top of the high coding efficiency of H.264/AVC. In addition to bringing a cost-efficient solution to the delivery of different formats of the same content to multiple users, it can be used to provide a better viewing experience (enhanced content portability, device power / content-quality adaptation, fast zapping times and fluid forward / rewind functions, efficient error retransmission, etc.). This article describes the potential of SVC, in terms of applications and performance. A brief overview of SVC functionalities, as well as practical use cases, are given in the following sections. Different performance evaluations, based on test results, are also described.
312	Multiple Description Coding — a new technology for video streaming over the Internet Andrea Vitali (593 KB)
	The Internet is growing quickly as a network of heterogeneous communication networks. The number of users is rapidly expanding and bandwidth-hungry services, such as video streaming, are becoming more and more popular by the day. However, heterogeneity and congestion cause three main problems: unpredictable throughput, losses and delays. The challenge is therefore to provide: (i) quality, even at low bitrates, (ii) reliability, independent of loss patterns and (iii) interactivity (low perceived latency) ... to many users simultaneously. In this article, we will discuss various well-known technologies for streaming video over the Internet. We will look at how these technologies partially solve the aforementioned problems. Then, we will present and explain Multiple Description Coding – which offers a very good solution – and how it has been implemented and tested at STMicroelectronics..
308	Prix Europa — results of the 2006 media streaming trial Franc Kozamernik and Marco de Giorgi (374 KB)
	The Prix Europa 2006 opening concert was given on 14 October 2006 in Berlin by a Portuguese World Music group called Gaiteros de Lisboa. On the occasion of this one-hour long concert, the EBU organized a technical experiment to distribute multichannel 5.1 audio – coded in HE AAC (High Efficiency Advanced Audio Coding) over the internet using a Peer-to-Peer (P2P) technology from Octoshape. This experiment is significant because, for the first time, an event was “broadcast” live in 5.1 multichannel format across the Internet, potentially addressing large audiences with high-quality surround sound.
304	MPEG-2 — high-compression technologies for HDTV Masaaki Kurozumi, Yukihiro Nishida and Eisuke Nakasu (451 KB)
	Digital video coding standards offer flexibility in their encoding techniques and enable coding efficiency improvements, in compliance with the standard, over a period of time. The MPEG-2 video coding standard [1] employs the adaptive DCT coding scheme with motion-compensated prediction. The amount of overhead information, including motion vector codes and coding modes, is often large for critical HDTV sequences at lower bitrates. NHK’s new coding method [2] – conforming to the MPEG-2 Main Profile – significantly reduces the amount of overhead information and makes digital HDTV services possible at lower bitrates, while maintaining compatibility with conventional digital broadcast receivers.
303	Dirac — video compression using open technology Tim Borer and Thomas Davies (527 KB)
	The distribution, delivery and storage of video are core activities for broadcasters. In the digital world, compression is used to exploit limited storage and transmission capacity as efficiently as possible. The BBC is developing a video compression technology, called Dirac, so that we can understand the technology and use it at reasonable cost and without restrictions. Dirac is a hybrid motion-compensated codec that uses modern techniques such as wavelet transforms and arithmetic coding. It is an open technology which means that it is freely available and can be used without the payment of licence fees. Open technology is well suited to the business model of public service broadcasters as it allows open collaboration by those interested in its future development.
302	AVC/H.264 — an advanced video coding system for SD and HD broadcasting Paola Sunna (77 KB)
	A bitrate of about 270 Mbit/s is needed to transmit uncompressed digital video that accords with ITU-R Rec. BT. 601 (i.e. standard-definition television). Digital HDTV, on the other hand, needs a considerably greater bitrate and – regardless of the modulation scheme adopted – transmission via traditional broadcast channels is impossible without the application of advanced video compression techniques. This article gives an overview of the current video coding technologies that are suitable for HDTV transmission; in particular, AVC/H.264.
295	Everything you wanted to know about video codecs — but were too afraid to ask David Wood (468 KB)
	Digital video compression technology continues to evolve, and the choice of systems presents a difficult challenge for broadcasters and web content providers. In this article, the author explains some of the factors shaping the evolution of video compression technology, and offers some insights into the comparative performance of video compression systems. The article is based on a presentation given in Spring 2003 to the EBU Technical Assembly in Moscow.
293	The emerging H.264/AVC standard Ralf Schäfer, Thomas Wiegand and Heiko Schwarz (544 KB)
	H.264/AVC is the current video standardization project of the ITU-T Video Coding Experts Group (VCEG) and the ISO/IEC Moving Picture Experts Group (MPEG). The main goals of this standardization effort are to develop a simple and straightforward video coding design, with enhanced compression performance, and to provide a “network-friendly” video representation which addresses “conversational” (video telephony) and “non-conversational” (storage, broadcast or streaming) applications. H.264/AVC has achieved a significant improvement in the rate-distortion efficiency – providing, typically, a factor of two in bit-rate savings when compared with existing standards such as MPEG-2 Video.
266	MPEG video — A simple introduction Bob Ely (89 KB)
	The core element of all DVB systems is the MPEG-2 vision coding standard, which is based upon a flexible toolkit of techniques for bit-rate reduction. The MPEG-2 specification only defines the bit-stream syntax and decoding process. The coding process is not specified, which means that compatible improvements in the picture quality will continue to be possible. In this article, the author provides a simple introduction to the technicalities of the MPEG-2 video coding standard.
COFDM / ODFM
295	OFDM receivers — impact on coverage of inter-symbol interference and window positioning Roland Brugger and David Hemingway (904 KB)
	This article offers a general overview of the possible strategies for FFT window synchronization in OFDM receivers. These strategies are equally applicable to the T-DAB and DVB-T broadcasting systems.
278	The how and why of COFDM Jonathan Stott (301 KB)
	Coded Orthogonal Frequency Division Multiplexing (COFDM) is a form of modulation which is particularly well-suited to the needs of the terrestrial broadcasting channel. COFDM can cope with high levels of multipath propagation, with a wide spread of delays between the received signals. This leads to the concept of single-frequency networks in which many transmitters send the same signal on the same frequency, generating “artificial multipath”. COFDM also copes well with co-channel narrowband interference, as may be caused by the carriers of existing analogue services. COFDM has therefore been chosen for two recent new standards for broadcasting – DAB and DVB-T, both of which have been optimized for their respective applications and have options to suit particular needs. The special performance of COFDM in respect of multipath and interference is only achieved by a careful choice of parameters and with attention to detail in the way in which the forward error-correction coding is applied.
276	The effects of phase noise in COFDM Jonathan Stott (571 KB)
	The reception of a COFDM signal is analyzed here for the case where phase noise has been added to the signal, e.g. by a receiver local oscillator. Two effects are distinguished: common phase error (a rotation of the signal constellation) and inter-carrier interference (similar to additive Gaussian noise). It is shown that the amounts of these effects can be deduced from the spectrum of the phase noise using a pair of weighting functions. Use of these weighting functions simplifies the process of computation; it also makes it easier to visualize the consequences of any modifications to the phase-noise spectrum. Some illustrations are given of the two phase-noise effects on the constellation of a DVB-T digital television signal, along with some practical observations on receiver implementation.
224	Principles of modulation and channel coding for digital broadcasting M. Alard and R. Lassalle (1159 KB)
	This article explains the benefits of using a system called Orthogonal Frequency Division Multiplexing to overcome the adverse effects of severe multipath propagation, such as occurs in mobile reception. The signal is demodulated with the aid of a Fast Fourier Transform technique. Consideration is given to the digital coding arrangement, and it is concluded that a concatenation of a convolutional code and a Reed-Solomon code gives excellent results. The feasibility of implementing such a system for the domestic market is briefly discussed.
Content Management
2004-Q1	ANTS — a complete system for automatic news programme annotation based on audiovisual content and text analysis
	Giorgio Dimino, Alberto Messina and Roberto Borgotallo (356 kB)
	This article describes an integrated system for the automatic annotation of television news programmes named ANTS (Automatic Newscast Transcription System). It consists of several analysis components, integrated within a unified architecture. Users have the possibility of accessing a large daily-growing database of news stories from the main national channels – all identified, categorised and published in a fully automatic way. The system identifies story boundaries, extracts texts from spoken content, classifies stories by subject and links external relevant information coming from the web. The system’s performance has been evaluated in a real-life scenario by a panel of professional users inside RAI. The strength of the approach behind ANTS is its ability to integrate several heterogeneous tools in a performant and ready-for-production environment. ANTS is capable of elaborating many hours of material per day, without significant service drops and with sufficiently good accuracy for industrial deployment in large broadcasting facilities.
302	"I want clips" — an introduction to the JIBS scheme for the exchange of educational video clips Nadège Boinnard (394 KB)
	The exchange of programme material in a digital world involves not only satellite distribution channels, but also important decisions over the best video compression format to use and, of course, the development of meaningful metadata to accompany the content material. This article provides an introduction to the JIBS platform that has been developed to enable the buying, selling and exchange of educational video clips among broadcasters and educational establishments.
293	Managing multimedia content for the Internet Pascal Dreer (773 KB)
	A news and information portal in nine languages ... streaming audio/video over the Internet ... content management systems ... data broadcasting ... data services for mobile phones ... a geographical information system … The changes have come thick and fast at swissinfo/Swiss Radio International over the past few years, with the introduction of a host of new services and applications. A traditional shortwave broadcaster has now turned into a multimedia venture, as described in this article.

European Broadcasting Union L'Ancienne Route 17A CH-1218 Grand-Saconnex Geneva Switzerland