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This paper describes a channel estimation algorithm for the advanced terrestrial digital multimedia broadcasting (T-DMB) transmission system which is fully backward compatible with the conventional T-DMB standard. The advanced T-DMB has enhanced features which provide higher transmission data rate for increasing program channel capacity and better multimedia A/V service quality than the conventional T-DMB system by adopting a hierarchical modulation and scalable audio/video CODEC. To support the higher data rate transmission for the advanced T-DMB, high performance channel estimation and equalization techniques should be adopted into the advanced T-DMB receivers. In this paper, we propose a new channel estimation technique for the Advanced T-DMB to compensate channel distortion under dynamic multipath environment. [ABSTRACT FROM AUTHOR]

As a class of high-performance forward error correction codes, turbo codes, which can approach the channel capacity, could become a candidate of the coding methods in future terrestrial broadcasting (TB) systems. Among all the demands of future TB system, high throughput and low latency are two basic requirements that need to be met. Parallel turbo decoding is a very effective method to reduce the latency and improve the throughput in the decoding stage. In this paper, a parallel turbo decoder is designed and implemented in field-programmable gate array (FPGA). A reverse address generator is proposed to reduce the complexity of interleaver and also the iteration time. A practical method of modulo operation is realized in FPGA which can save computing resources compared with using division operation. The latency of parallel turbo decoder after implementation can be as low as 23.2 us at a clock rate of 250 MHz and the throughput can reach up to 6.92 Gbps. [ABSTRACT FROM PUBLISHER]

This paper aims to improve the throughput of a broadcasting system that supports the transmission of multiple services with differentiated minimum signal-to-noise ratios (SNRs) required for successful receptions simultaneously. We propose a novel multiplexing method called bit division multiplexing (BDM), which outperforms the conventional time division multiplexing (TDM) counterpart by extending the multiplexing from symbol level to bit level. Benefiting from multiple error protection levels of bits within each high-order constellation symbol, BDM can provide so-called nonlinear allocation of the channel resources. Both average mutual information (AMI) analysis and simulation results demonstrate that, compared with TDM, BDM can significantly improve the overall transmission rate of multiple services subject to the differentiated minimum SNRs required for successful receptions, or decrease the minimum SNRs required for successful receptions subject to the transmission rate requirements of multiple services. [ABSTRACT FROM AUTHOR]

Long Term Evolution (LTE) standard enhances Multimedia Broadcast Multicast Services (MBMS) to provide highly efficient transmission of multicast/broadcast services. This paper aims at clarifying which is the best option for the Enhanced MBMS (E-MBMS) network configuration according to the operator needs. Besides, it describes the main technical features of the new E-MBMS LTE standard, for both streaming and file delivery, and derives optimum operation modes. Some relevant conclusions are drawn from a set of simulation results calibrated with International Telecommunication Union Radiocommunication (ITU-R) guidelines. Regarding streaming delivery, results show that increasing the MBMS over a Single Frequency Network (MBSFN) area size allows employing a higher modulation and coding scheme level, which entails more streaming channels per MBSFN area. Concerning file delivery, results show that the reduction in the mean throughput of unicast users matches exactly the percentage of stolen subframes. Besides, the hybrid approach suggested in this paper improves the file delivery efficiency while eliminating the dependency on a MBMS threshold. [ABSTRACT FROM PUBLISHER]

The requirements for future DVB-TIH networks demand that broadcasters design and deploy networks that provide ubiquitous reception in challenging indoors and other obstructed situations. It is essential that such networks are designed cost-effectively and with minimized environmental impact. The EC funded project PLUTO has since its start in 2006 explored the use of diversity to improve coverage in these difficult situations. The purpose of this paper is to investigate the performance of Transmit Delay Diversity (DD) with two antennas to improve the reception of DVB-TIH systems operating in different realistic propagation conditions through a series of tests using a SPIRENT SR5500 dual channel emulator. The relationship between correlation coefficient between channels, receiver velocity and diversity gain is investigated. It is shown that transmit delay diversity significantly improves the quality of reception particularly in simulated fast fading mobile broadcasting applications. This paper documents research conducted by Brunel University and Broadreach Systems. [ABSTRACT FROM AUTHOR]

The objective of this work is to research and analyse the performance of Cyclic Delay Diversity (CDD) with two transmit antenna on DVB-H systems operating in spatially correlated channel. It is shown in this paper that CDD can achieve desirable transmit diversity gain over uncorrelated channel with or without receiver diversity. However, in reality, the respective signal paths between spatially separated antennas and the mobile receiver is likely to be correlated because of insufficient antenna separation at the transmitter and the lack of scattering effect of the channel. Under this spatially correlated channel, it is apparent that CDD cannot achieve the same diversity gain as obtained under the uncorrelated channel. In this paper, a new upper bound on the pairwise error probability (PEP) of the CDD with spatial correlation of two transmit antennas is derived. The upper bound is used to study the CDD theoretical error performance and diversity gain losses over a generalized spatially correlated Rayleigh channel. This theoretical analysis is validated by the simulation of DVB-H systems with two transmit antennas and the CDD scheme. Both the theoretical and simulated results give the valuable insight that the CDD ability to perform well with a certain amount of channel correlation. [ABSTRACT FROM AUTHOR]

This paper considers the potential for an HDTV delivery format with 1920 x 1080 pixels progressive scanning and 50 frames per second in broadcast applications. The paper discusses the difficulties in characterizing the display to be assumed for reception. It elaborates on the required bit rate of the 1080p/50 format when critical content is coded in MPEG-4 H.264 AVC Part 10 and subjectively viewed on a large, flat panel display with 1920 x 1080 pixel resolution. The paper describes the initial subjective quality evaluations that have been made in these conditions. The results of these initial tests suggest that the required bit-rate for a 1080p/50 HDTV signal in emission could be kept equal or lower than that of 2nd generation HDTV formats, to achieve equal or better image quality. [ABSTRACT FROM AUTHOR]

Broadcasting is a prospective approach to support near video-on-demand services with light communication overhead. By letting clients share channels, such approaches involve partitioning a video into segments and repeatedly broadcasting these segments in multiple channels. An early paper proposed a broadcasting scheme called RFS (recursive frequency splitting), which can significantly reduce clients' waiting time. While efficient, RFS suffers from a high computational complexity of O(n log n), where n is the number of segments of the video, which is typically very large. This paper proposes an efficient segmentation scheme, which can significantly reduce the computational overhead by slightly sacrificing the number of segments that can be arranged as compared to RFS. [ABSTRACT FROM AUTHOR]

The DVB standard does not mandate the use of authentication and integrity protection for transport streams. This allows malicious third parties to replace legitimate broadcasts by overpowering terrestrial transmissions. The rogue signal can then deliver a malicious broadcast stream to exploit security vulnerabilities on Smart TVs (STVs) in range. We implemented a proof-of-concept attack based on a malicious Hybrid Broadcast Broadband TV app, able to acquire permanent system-level access to an STV over the air, in less than 10 s. These attacks, however, are severely limited in range due to required co-channel protection ratios (CCPRs), which is in direct contradiction to previous publications. We present evidence for these limitations in form of laboratory experiments, extensive simulations, and field measurements. To this end, we developed an automated, low-cost method for CCPR determination, as well as a method for non-disruptive attack range measurements based on a gap filler and the resulting channel impulse response. [ABSTRACT FROM PUBLISHER]

Due to the recent popularization of digital broadcasting systems, selective contents, i.e., watching contents selected by users themselves, have attracted great attention. For example, in a news program, after a user selects an interesting content, he/she watches it. In selective contents broadcasting, since the server needs to deliver many contents, the necessary bandwidth for continuously playing the data increases. Conventional schemes reduce the necessary bandwidth by producing an effective broadcast schedule. However, they do not consider the upper limit in the bandwidth. When an upper limit exists in the bandwidth, users may wait to play the data. In this paper, we propose a broadcasting scheme that reduces waiting time by considering available bandwidth. In our proposed scheme, by acquiring the channel bandwidth that is the same as the data consumption rate, we effectively reduce the waiting time. [ABSTRACT FROM AUTHOR]

LDPC codes are state-of-art error correcting codes, included in several standards for broadcast transmissions. Iterative soft-decision decoding algorithms for LDPC codes reach excellent error correction capability; their performance, however, is strongly affected by finite-precision issues in the representation of inner variables. Great attention has been paid, in recent literature, to the topic of quantization for LDPC decoders, but mostly focusing on binary modulations and analysing finite precision effects in a disaggregrated manner, i.e., considering separately each block of the receiver. Modern telecommunication standards, instead, often adopt high order modulation schemes, e.g. M-QAM, with the aim to achieve large spectral efficiency. This puts additional quantization problems, that have been poorly debated in previous literature. This paper discusses the choice of suitable quantization characteristics for both the decoder messages and the received samples in LDPC-coded systems using M-QAM schemes. The analysis involves also the demapper block, that provides initial likelihood values for the decoder, by relating its quantization strategy with that of the decoder. A new demapper version, based on approximate expressions, is also presented, that introduces a slight deviation from the ideal case but yields a low complexity hardware implementation. [ABSTRACT FROM AUTHOR]

In this paper we analyze the performance of a low complexity superimposed channel estimation technique for orthogonal frequency division multiplexing (OFDM). In particular, an analytical model of interference due to channel estimation errors and imperfect superimposed sequence cancellation at receiver is proposed, whose effectiveness is validated by simulations. Indeed, the significative length of OFDM symbols used in new wide area broadcasting applications makes the superimposed technique a viable alternative to the classical pilot aided technique. For the same computational complexity, the comparison between the two techniques is based on the achievable system throughput both for the current terrestrial digital video broadcasting (DVB-T) standard and for the proposed next generation DVB-T (DVB-T2). Our results show that superimposed technique provides higher bit-rates than the pilot aided technique, with a gain in the range of 4% to 10%. [ABSTRACT FROM AUTHOR]

In this paper, we consider a peer-to-peer (P2P) network, where multimedia streams are broadcast by matched peers based on their resource reciprocation profiles. We propose a new framework where each peer creates a coalition of matched peers with which it can exchange resources in order to improve its utility. The utility is determined based on explicit consideration of the peer's multimedia attributes and the quality derived by the peers' reciprocation behavior. We adopt the proportional bargaining solution to negotiate the upload bandwidth among the matched peers. Proportional bargaining allows to determine each peer's optimal (in a Pareto optimal sense) upload rates in a coalition in terms of its utility impact. The impact of an incoming peer on the coalition value, which represents the collective utility achieved by the peers in a coalition, is assessed by explicitly investigating the coalition value improvement. Finally, our results show that the proposed coalition-based resource reciprocation can improve the resource allocation/scheduling algorithms deployed in existing P2P systems such as BitTorrent and CoolStreaming. We also discuss how the proposed resource reciprocation approach can be implemented in other multimedia broadcasting applications. [ABSTRACT FROM AUTHOR]

Due to the recent popularization of mobile multi- media broadcasting, broadcasting continuous media data such as audio and video has attracted great deal of attention. In general continuous media data broadcasting, since clients have to wait to receive data before playing it, various schemes to reduce waiting time have been studied. Some reduce the waiting time by dividing the data into several segments and broadcasting preceding segments frequently with a single channel. However, by dividing the data into numerous segments and producing an effective broadcast schedule, the waiting time can be further reduced. In this paper, we propose a scheduling protocol to reduce waiting time with large-scale data segmentation. [ABSTRACT FROM AUTHOR]

Technical details of the recently announced Chinese Digital Terrestrial Television Broadcasting Standard which defines the physical layer transmission protocol including the frame structure, channel coding and modulation schemes are discussed in this paper. The major differences between Chinese DTV standard in multi-carrier working mode and DVB-T standard, all occupying the same 8 MHz baseband bandwidth, are addressed. The measurement results of several working modes showing satisfactory receiving performance of high and standard definition TV signals under both indoor and outdoor environments for the fixed and mobile reception are also presented. [ABSTRACT FROM AUTHOR]

It is well known that HPAs (High Power Amplifiers) are inherently nonlinear devices and many researches have focused on the predistortion for memoryless HPAs. However, memory effects of HPAs can no longer be ignored when communication systems have wider bandwidth. Memoryless predistortion techniques proposed previously seldom have satisfactory effectiveness for typical wideband applications such as OFDM (Orthogonal Frequency Division Multiplexing) systems. In this paper, an improved adaptive predistortion method called 2D LUT (2-dimension look-up table) with different accuracy levels is presented to linearize HPAs with memory effects. Simulation and experimental results show that 2D LUT implements excellent performance in mitigating the signal deterioration caused by memory effects, both rectifies the signal constellation distortion and suppresses the spectrum emission. Large scale matrix computation is also avoidable in these adaptive algorithms, which makes them feasible when a real-time system is necessary. [ABSTRACT FROM AUTHOR]

Periodic broadcasting is known as an efficient technique to support near Video-on-Demand services for delivering popular videos, since it can reduce the bandwidth requirement for transmitting streaming video to simultaneous viewers. The channel transition problem is an issue to be concerned about the variability of popularity of video. In this paper, we present a novel channel transition scheme based on the fast broadcasting. In most conditions, our proposed scheme has less bandwidth waste than the existing SFB scheme. In addition, no extra startup latency and client buffer are needed in our scheme. [ABSTRACT FROM AUTHOR]

This paper presents a new design and implementation method of distributed translator (DTxR), termed the equalization DTxR (EDTxR), for distributed translator network in ATSC systems. EDTxR has a simple structure and does not require any devices to be added to studio or transmitter facilities already deployed. EDTxRs which retransmit the same transmitter signal can have identical output symbol streams among them without additional synchronization information by utilizing the structure of the equalization digital on-channel repeater (EDOCR). Moreover, they can synchronize output frequency among them without global positioning system (GPS) clock receivers by adopting a crystal oscillator for each translator and compensating the frequency offset of the each crystal oscillator through the carrier and timing recovery. To verify the proposed method, multiple EDTxRs are implemented and tested in the laboratory and in the field. Through the tests, it is confirmed that EDTxR is a simple and economic distributed translator for distributed transmission system which does not need additional synchronization devices. Therefore, EDTxR can be a promising translator for coverage extension of digital terrestrial television broadcasting under spectrum deficient situations. [ABSTRACT FROM AUTHOR]

In providing video-on-demand (VoD) services to a number of clients through networks, the bandwidth requirements of video transmission restrict VoD services. For this reason, many significant broadcasting-based VoD schemes have been proposed to efficiently support services. However, the broadcasting-based VoD schemes approach needs frequency channel hopping, using many channels at the same time and managing many segments of a video. These make it difficult to implement. In this paper, we propose a harmonic staggered broadcasting scheme which has a simple structure and substantially improved VoD efficiency. The proposed scheme substantially reduces the number of segments required and the frequency for switching channels. Also this new scheme performs significantly better than the existing VoD schemes in viewer's waiting time and buffer requirements with simplicity. The numerical results demonstrate that the viewer's waiting time of the harmonic staggered broadcasting scheme is close to the harmonic broadcasting scheme and the buffer requirements of this can be adjusted by a video dividing coefficient h. [ABSTRACT FROM AUTHOR]

DVB-H networks allow high data rate broadcast access for hand-held terminals. A method to analyze the performance is developed. The influence of different MPE-FEC rates and modulation schemes on the performance of a DVB-H network is analyzed in this paper for 9 different reception conditions. Portable (indoor and outdoor) and mobile reception (car, train, tram, bus) are investigated. This analysis enables better estimations of the performance of DYB-H systems and an optimized DVB-H performance model. The percentage of valid reception, MPE-FEC gains, carrier to interference-plus-noise ratios, and minimal signal strengths for the different reception conditions and modulation schemes are presented. [ABSTRACT FROM AUTHOR]

This paper presents propagation measurement results in the UHF Digital Television (DTV) band, using an ATSC DTV signal as a channel sounding signal to investigate the characteristics of a wideband single-input multiple-output (SIMO) under mobile reception conditions. A series of DTV dual channel signal measurements were conducted in the Ottawa region to characterize the transmission of ATSC 8-VSB signals under different propagation environments. A dual channel receiver was used to record SIMO signals: these were used to provide channel impulse response estimates. From the recorded time-varying impulse responses and their correlation functions, average power delay profiles, scattering functions were computed to analyse the joint spatial fluctuations of the received signals. The DTV channel characterization study is aimed at broadcast system designers involved in data transmission of terrestrial digital television signals over mobile channels with diversity reception. [ABSTRACT FROM AUTHOR]

Manly essential multimedia applications rely on video-on-demand technology to deliver a video to different users. A number of periodic broadcast techniques have been proposed for the cost-effective implementation of such systems. Most of these techniques I would either try to minimize the server band- width, user bandwidth, user storage, user access latency to the video, or a combination of some of the aforementioned parameters. On the other hand, the implementation strategies of these broadcast schedules would necessitate a minimum bandwidth requirement for all users. Multi-resolution techniques address the heterogeneity problem by sacrificing user video quality. In this paper, we consider a different approach that does not possess this disadvantage. using an incremental channel design at the server side, and a specific broadcast schedule, users can choose among a range of band widths to use to download the video at the cost of their access latency and not to the video quality. We prove the correctness of the proposed solution; provide mathematical analysis to demonstrate its heterogeneous behavior, and present performance studies to illustrate its efficiency. [ABSTRACT FROM AUTHOR]

A new TV audio system based on the MPEG-H 3D audio standard has been designed, tested, and implemented for ATSC 3.0 broadcasting. The system offers immersive sound to increase the realism and immersion of programming, and offers audio objects that enable interactivity or personalization by viewers. Immersive sound may be broadcast using loudspeaker channel-based signals or scene-based components in combination with static or dynamic audio objects. Interactivity can be enabled through broadcaster-authored preset mixes or through user control of object gains and positions. Improved loudness and dynamic range control allows tailoring the sound for best reproduction on a variety of consumer devices and listening environments. The system includes features to allow operation in HD-SDI broadcast plants, storage, and editing of complex audio programs on existing video editor software or digital audio workstations, frame-accurate switching of programs, and new technologies to adapt current mixing consoles for live broadcast production of immersive and interactive sound. Field tests at live broadcast events were conducted during system design and a live demonstration test bed was constructed to prove the viability of the system design. The system also includes receiver-side components to enable interactivity, binaural rendering for headphone, or tablet computer listening, a “3D soundbar” for immersive playback without overhead speakers, and transport over HDMI 1.4 connections in consumer equipment. The system has been selected as a proposed standard of ATSC 3.0 and is the sole audio system of the UHD ATSC 3.0 broadcasting service currently being deployed in South Korea. [ABSTRACT FROM AUTHOR]

The International Telecommunications Union has defined the frequency band in which Broadcasting Satellite Systems (BSS) and Multi-channel Multipoint Distribution Systems (MMDS) should coexist. With the development of MMDS working on the same frequency band as the BSS, and theoretically with identical technical parameters, an important question arises of how to set up the parameters of terrestrial distribution systems so that their coexistence with satellite systems is enabled while at the same time terrestrial systems are not made technically too limited. In this article, we summarize the results of extensive research work that sought to answer the above question. Results of practical measurements performed on MMDS and BSS are presented. The calculations were performed according to the ITU-R Regulations. In addition to measuring professional commercial systems, we constructed a transmitting/receiving system to simulate a worst-case scenario-having the BSS and MMDS transmitters and receiver (measurement system) on the same line (Fig. 1). From the analysis of the empirical measurements and calculations, we determined how the parameters should be set up so as to enable the quality coexistence of BSS and MMDS. We focused on those parameters that significantly influence such a coexistence: the geographical position of BSS and MMDS transmitters, Symbol Rate, Code Rate, MMDS Transmitter Frequency, and MMDS Transmitter Power. [ABSTRACT FROM AUTHOR]

This work proposes a quality-oriented multimedia delivery framework that tackles the issue of optimizing video broadcasting and interactive video applications over packet networks with respect to both resource utilization and user perceived quality. Previous work showed that the quality of service requirements of multimedia applications can be optimally satisfied by pipeline forwarding of packets by keeping delay controlled and resource utilization high, while enabling highly scalable network devices. These properties are key in today networks to enable valuable (i.e., chargeable for) services and to avoid that the traffic increase due to broadband video either collapses existing networks or forces the deployment of high cost, cutting-edge technology to properly upgrade them. However, the current Internet is not based on such technology and its incremental introduction raises questions on how to handle video packets generated by pipeline-forwarding-unaware sources. This work proposes to use the perceptual importance of the carried video samples to determine which packets shall be transferred with pipeline forwarding-thus receiving deterministic service-and which with a traditional, e.g., best effort or differentiated service. Two new scheduling algorithms are proposed, with an extensive analysis and simulation results, which also investigate the impact of the encoding scheme. Performance bounds have been established by comparing the proposed algorithms with an exhaustive-search approach, showing that the performance is within 2 dB PSNR from the optimal solution in the worst case. [ABSTRACT FROM AUTHOR]

Raptor codes have been standardized as application layer forward error correction (FEC) codes for Multimedia Broadcast and Multicast Services (MBMS) and Digital Video Broadcast (DVB) due to their extraordinary advanced FEC protection and performance. Raptor codes are known to have characteristics very close to ideal and provide a wide range of operating flexibility and efficiency unmatched by other codes, and in particular they are an excellent implementation of fountain codes. Until now, investigations of the application of Raptor codes to UMTS and EPGRS have used an overall system model that does not accurately model the physical channel and user mobility, simplistically assuming independent random packet losses at the application layer. We investigate MBMS in UMTS in a much more realistic and complete simulation environment by considering advanced and complete channel models that simulate the physical channel and user mobility in a cellular network. We use this realisitic simulation environment to determine optimal system parameters under different mobility models, with different bearer parameters, and without and with selective combining. More specifically, we investigate joint settings of the Raptor code rate, the Turbo code rate, transmission power, etc., to find settings which provide reliable download delivery of files using minimal transmission energy. One of our main results is that optimal system-wide operating points use low transmission power and a modest amount of Turbo coding that results in relatively large radio packet loss rates that is compensated for by using a substantial amount of Raptor coding. These optimal operating points use far less transmission energy for download delivery of files than possible operating points without Raptor. [ABSTRACT FROM AUTHOR]

A popular video can be broadcast by partitioning the video into segments, which are broadcast on several channels simultaneously and periodically. This method allows multiple users to share channels, leading to higher bandwidth utilization. Previous studies mainly focus on reducing viewers' waiting time. This work studies another important issue, namely client buffer' savings. A reverse fast broadcasting (RFB) scheme is proposed to alleviate the buffer problem. There are two specific properties for RFB. First, this scheme arranges the segments in descending order of their numeric indices on each channel at the server end. Second, RFB requires a client to receive segments as late as possible, such that the client buffers the smallest number of segments. RFB has the same waiting time as the fast broadcasting (FB) scheme, but just needs a half of buffer spaces required by FB. RFB also requires smaller client buffers than the pyramid broadcasting (PB), skyscraper broadcasting (SkB), greedy disk-conserving broadcasting (GDB), BroadCatch, and recursive frequency-splitting schemes. Moreover, if we apply the same changes to the PB, SkB and GDB schemes, their buffer requirements decrease by 25% to 75%. [ABSTRACT FROM AUTHOR]

Space-frequency coded (SFC) and channel coded orthogonal frequency-division multiplexing (COFDM) are considered under narrow-band interference (NB!). Analytical expressions for the bit error probability (BEP) are derived for OFDM with SFC in a frequency-selective fading environment. It is shown that SFC increases the resistance of COFDM against the NB! and reduces the BEP considerably. Specific attention is paid to Digital Terrestrial Television Broadcasting (DTTB), and the associated coding gains are discussed. [ABSTRACT FROM AUTHOR]

To reduce the bandwidth of media-on-demand transmissions, different broadcasting techniques have been developed over the last years. For highly demanded media, proactive schemes have a lower bandwidth requirement compared to reactive ones. While some of the proactive schemes try to obtain an efficiency near to the theoretical limit, others have been developed to lower the bandwidth by partial preloading, additional playout delay, the use of breaks within the transmission or by exploiting bandwidth variations of variable-bit-rate media. In this article we describe how one of the most efficient proactive transmission schemes can further be improved and generalized so it supports all these additional enhancements and outperforms its predecessor and the specialized schemes. [ABSTRACT FROM AUTHOR]

Presents the results of a series of tests performed by the Communications Research Center Canada to investigate the possibility of improving reception in digital television broadcasting. Focus on the use of an On-Channel Repeater (OCR) in the areas covered by a digital transmitter; Analysis of the impact of OCR against reduced bit rate transmission; Conditions for the use of OCR over of bit rate reduction.