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Broadcast is a fundamental operation of wireless ad hoc networks (WANETs) and has widely been studied over the past few decades. However, most existing broadcasting strategies assume nonsleeping wireless nodes and thus are not suitable for uncoordinated duty-cycled WANETs, in which each node periodically switches on and off to save energy. In this paper, we study the minimum-transmission broadcast problem in uncoordinated duty-cycled WANETs (MTB-UD problem) and prove its NP-hardness. We show that modifications of existing broadcast approaches can only provide a linear approximation ratio of O(n) (where n is the number of nodes in the network). We propose a novel set-cover-based approximation (SCA) scheme with both centralized and distributed approximation algorithms. The centralized SCA (CSCA) algorithm has a logarithmic approximation ratio of 3(ln Δ + 1) and time complexity of O(n3) (Δ is the maximum degree of the network). The distributed SCA (DSCA) algorithm has a constant approximation ratio of at most 20 while keeping both linear time and message complexities. We have conducted both theoretical analysis and simulations to evaluate the performance of the proposed algorithms. Results show that both the CSCA and DSCA algorithms outperform the modified versions of existing broadcast approaches by at least 50%. [ABSTRACT FROM AUTHOR]

Two spanning trees rooted at vertex r in a graph G are called independent spanning trees (ISTs) if for each vertex v in G, v ≠ r, the paths from vertex v to vertex r in these two trees are internally distinct. If the connectivity of G is k, the IST problem is to construct k ISTs rooted at each vertex. The IST problem has found applications in fault-tolerant broadcasting, but it is still open for general graphs with connectivity greater than four. In this paper, we shall propose a very simple algorithm for solving the IST problem on multidimensional torus networks. In our algorithm, every vertex can determine its parent for a specific independent spanning tree only depending on its own label. Thus, our algorithm can also be implemented in parallel systems or distributed systems very easily. [ABSTRACT FROM AUTHOR]

A tunable internal antenna with an epsilon negative zeroth order resonator (ENG-ZOR) for digital video broadcasting-handheld service is proposed in this paper. The antenna consists of two helical radiators, an ENG ZOR with two inductors and a variable capacitor to adjust the resonant frequency. The resonant frequency of the designed antenna can be swept from 474-702 MHz by adjusting the capacitance of the variable capacitor by controlling the DC bias voltage. The proposed antenna has a very compact size of 40 mm X 5 mm X 5 mm. [ABSTRACT FROM AUTHOR]

Terrestrial or Handheld Digital Video Broadcasting (DVB-T/H) and Terrestrial Digital Multimedia Broadcasting (T-DMB) are two popular broadcasting standards that enable digital television transmissions to handheld receivers. This paper presents a comprehensive performance comparison between the physical layers of DVB-T/H and T-DMB when employed for mobile communications. By exploiting a recently proposed fast simulation model, we assess the BER of the two coded OFDM systems in several realistic scenarios, taking into account Rayleigh and Rice channels, different mobile speeds, inner and outer channel coding, channel estimation, and one or two receive an- tennas. Our comparison shows that the DVB-T/H physical layer performance highly depends on the delay spread of the channel, whereas T-DMB is less sensitive to the frequency selectivity of the channel. As a result, DVB-T/H yields better performance than T-DMB in typical Rayleigh channels with significant delay spread. On the contrary, at high SNR, T-DMB outperforms DVB-T/H in Rice channels with low delay spread. As a side result, we show the performance improvement of DVB-H produced by MPE-FEC at the data link layer. [ABSTRACT FROM AUTHOR]

Prior studies show that more than 70 percent of communication paths in a popular unstructured peer-to-peer (P2P) system (i.e., Gnutella) do not exploit the physical network topology, leading to the topology mismatch problem, and thus, lengthen communication between participating peers. While previous efforts in solving overlay topology matching problems do not guarantee the bounds of performance metrics (e.g., the communication delay between any two overlay peers and the broadcasting scope of any participating peer), in this paper, we present a novel topology matching algorithm that has provable performance qualities. In our proposal, each participating node creates and manages a constant number of overlay connections to other peers in a distributed manner. In rigorous performance analysis, we show that 1) the expected overlay communication delay between any two nodes in our P2P network is a constant; 2) in addition, any joining node has the exponential broadcasting scope in expectation; 3) furthermore, a participating node takes a polylogarithmic overhead to exploit the physical network locality and maintain its flooding scope. Together with extensive simulations, we present our proposal that significantly outperforms two recent solutions, i.e., THANCS and mOverlay, in terms of overlay communication latency and/or broadcasting scope. [ABSTRACT FROM AUTHOR]

In this paper, a downlink system in which a single-antenna base station communicates with K single antenna users over a time-correlated fading channel is considered. It is assumed that each receiver knows its own channel state, while the rate of the channel variation for all users and the corresponding initial fading gains are known to the base station. The average (per channel use throughput of the system is studied by applying various adaptive signaling schemes. Assuming a large number of users in the system, it is shown that using a scheduling scheme in which the base station transmits to the user with the maximum initial fading gain while using a fixed codeword length for all users, achieves the order of the maximum throughput. Moreover, an alternative scheduling scheme is proposed (by accounting for users' delays) and shown to achieve the optimum long-term fairness, while preserving the order of the maximum throughput. [ABSTRACT FROM AUTHOR]

This paper studies capacity bounds for discrete memoryless broadcast channels with confidential messages. Two private messages as well as a common message are transmitted; the common message is to be decoded by both receivers, while each private message is only for its intended receiver. In addition, each private message is to be kept secret from the unintended receiver where secrecy is measured by equivocation. Both inner and outer bounds are proposed to the rate equivocation region for broadcast channels with confidential messages. The proposed inner bound generalizes Csiszár and Körner's rate equivocation region for broadcast channels with a single confidential message, Liu et al.'s achievable rate region for broadcast channels with perfect secrecy Marton's and Gel'fand-Pinsker's achievable rate region for general broadcast channels. The proposed outer bounds, together with the inner bound, help establish the rate equivocation region of several classes of discrete memoryless broadcast channels with confidential messages, including the less noisy, deterministic, and semideterministic broadcast channels. Furthermore, specializing to the general broadcast channel by removing the confidentiality constraint, the proposed outer bounds reduce to new capacity outer bounds for the discrete memory broadcast channel. [ABSTRACT FROM AUTHOR]

Broadcast authentication is a critical security service in wireless sensor networks (WSNs), as it allows mobile users of WSNs to broadcast messages to multiple sensor nodes in a secure way. Although symmetric-key-based solutions such as μTESLA and multilevel μTESLA have been proposed, they all suffer from severe energy-depletion attacks resulting from the nature of delayed message authentication. This paper presents several efficient public-key-based schemes to achieve immediate broadcast authentication and thus avoid the security vulnerability that is intrinsic to μTESLA-like schemes. Our schemes are built upon the unique integration of several cryptographic techniques, including the Bloom filter, the partial message-recovery signature scheme, and the Merkle hash tree. We prove the effectiveness and efficiency of the proposed schemes by a comprehensive quantitative analysis of their energy consumption in both computation and communication. [ABSTRACT FROM AUTHOR]

Data broadcasting is an efficient way of delivering information over asymmetric wireless environments, and push systems can provide high scalability and client hardware simplicity. Many environments, however, are characterized by a priori unknown client demands and groups of clients that are located at the same region and have similar demands. The main disadvantage of the approaches in such cases is the lack of fairness, because groups with few members have much lower performance than groups with many clients, because the performance per group (directly) depends on the group size. In this paper, we propose a fair push system where the performance of each client is independent of the total number of clients that are located in the same region. This condition is achieved without the overall performance of the system being significantly affected. Moreover, the proposed fair system is extended to a priority-based system that multiplies the performance of a specific group, depending exclusively on its priority level in relevance to the rest of the groups. [ABSTRACT FROM AUTHOR]

Digital Video Broadcasting via Satellite-Second Generation (DVB-S2) was originally designed for fixed terminals, but it has recently been adopted by the DVB Return Channel via Satellite for Mobile scenario (DVB-RCS+M) under the singlecarrier-per-channel (SCPC) mode. For the purpose of adapting DVB-S2 for mobile reception, it has been agreed that, among others, forward error correction (FEC) at the link layer (LL) is a suitable means of achieving reliable reception in mobile environments. Prime candidates for LL-FEC had already been available schemes in the DVB family of standards: multiple-protocol encapsulation-FEC (MPE-FEC) and MPE interburst FEC (MPE-IFEC). Furthermore, different FEC codes may be applied within these schemes, i.e., Reed-Solomon (RS) or Raptor FEC codes. This paper introduces the integration of such schemes and codes on top of DVB-S2. In addition, we analyze the performance that can be achieved when applying these schemes with particular focus on two typical railway scenarios: 1) line of sight, together with the effect of railway power arches (LOS+PA), and 2) nonline of sight (nLOS). Both theoretical and simulation analyses reveal that LL-FEC can overcome typical fading effects in the railway scenario by selecting appropriate FEC codes and optimizing the coding parameters. We show that MPE-IFEC and extended MPE-FEC with Raptor codes - as finally specified in DVB-RCS+M-consistently show superior results over those of other LL-FEC for railway scenarios. We also indicate signaling updates to allow achievable performance. As for practical implementation, we propose two possible novel cross-layer architectures for unicast DVB-52 to provide quality of service (QoS). The architectures allow the migration from traditional packet encapsulation based on Moving Picture Experts Group 2-Transport Streams (MPEG2-TS) to new schemes, such as generic stream encapsulation (GSE). [ABSTRACT FROM AUTHOR]

Digital Video Broadcasting for Handheld (DVB-H) is one of the wireless technologies that provide modern digital television services. The main objective of this paper is to propose a method to evaluate the performance of mobile rebroadcaster in terms of symbol error rate, taking into account antenna effects and the Angle of Arrival statistics. A correction factor is derived from concepts as the Mean Effective Gain, which joined to a closed-form expression for the symbol error probability, allows predictions about the antenna impact on the broadcaster. The quantitative evaluation of the analysis identifies an important SNR reduction from 4 dB to 10 dB, depending on the environment. [ABSTRACT FROM AUTHOR]

A Gaussian multiple-input multiple-output broadcast channel (MIMO GBC) is considered. Throughout the paper it is assumed that 1) input signals are Gaussian and 2) perfect channel state information is available at the transmitter and at the receivers. By considering each data stream as a single user, the uplink-downlink signal-to-interference-plus-noise (SINR) duality is generalized to the MIMO case with general cross-talk matrix. The duality is subsequently applied to finding the solution for the SINR-balancing problem. The result serves as a tool for characterizing achievable rate regions of different coding strategies. Next, we investigate a superposition coding scheme proposed by Cover-van der Meulen-Hajek and Pursley (nicknamed CMHP [1]), where there is a common message to both users. We consider a MIMO broadcast channel with two users, each user has two antennas and the transmitter has four antennas. Assuming one common stream is sent by CMHP coding and successive decoding, a lower bound to the CMHP rate region is found. Behaviors of the CMHP rate region and sumrate are analyzed. We find the sumrate gaps between DPC, CMHP, and MMSE at high SNR for general 2-user multiple-input single-output (MISO) Gaussian broadcast channel. The result suggests when CMHP is beneficial for sumrate. [ABSTRACT FROM AUTHOR]

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]

Content or service delivered through IPTV is high quality and of high economic value accordingly. As digital technologies progress, illegal copy and redistribution of IPTV content become easier and simpler. Therefore it is required to protect IPTV content or service. In this paper, following the approaches taken by ITU-T IPTV Focus Group, we analyze the security threats and requirements. We also discuss related issues and solutions for IPTV. Specially, interoperability issues among different content and service protection systems are addressed. [ABSTRACT FROM AUTHOR]

This paper presents a digital rights management (DRM) scheme for mobile broadcasting that integrates management via, broadcasting and communications. The scheme provides access control and rights protection for subscription Mobile IPTV (Internet Protocol TV), enabling transmission of contract information to control broadcasting reception via both broadcasting wave and communications. It has transmission functions that reduce the amount of contract information that has to be transmitted through broadcasting as well as a function for transmitting contract information via communications. A PC simulation indicates that the DRM scheme is effective in terms of the transmission rate required to send contract information via broadcasting wave. We also demonstrated the feasibility of the DRM scheme by using a prototype system implemented on an SIM card. The DRM scheme enables advanced mobile receiver functions such as pay-per-view with an interactivity channel using communications and a subscription TV such as pay-per-month on mobile receivers without an interactivity channel by transmitting contract information via broadcasting wave. [ABSTRACT FROM AUTHOR]

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

With the increasing popularity of wireless networks and mobile computing, data broadcasting has emerged as an efficient way of delivering data to mobile clients having a high degree of commonality in their demand patterns. This paper proposes an adaptive wireless push system that operates efficiently in environments characterized by high broadcasting speeds and a-priori unknown client demands for data items. The proposed system adapts to the demand pattern of the client population in order to reflect the overall popularity of each data item. We propose a method for feedback collection by the server so that the client population can enjoy a performance increase in proportion to the broadcasting speed used by the server. Simulation results are presented which reveal satisfactory performance in environments with a-priori unknown client demands and under various high broadcasting speeds. [ABSTRACT FROM AUTHOR]

This paper presents a practical implementation of a quasigroup based multilevel, indexed scrambling transformation for use in signal authentication, encryption, and broadcasting applications in secret-key cryptography. Results of experiments with text and speech scrambling are presented. It is shown that the quasigroup transformation is very effective in destroying the structure of the input signal and, therefore, it can be an excellent encryption technique. [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]

The frequency assignment problem involves the assignment of discrete frequencies to the transmitters of a radio network, such as a radio broadcasting network. Frequency separation is necessary to avoid interference by other transmitters to the signal received from the wanted transmitter at the reception region. Here, it is of major importance to minimize the interference while at the same time using the spectrum efficiently. In this paper we present two original distributed algorithms implemented on clusters of PCs used to solve the frequency assignment problem in the field of radio broadcasting. The first one is based on the island distributed [1] implementation of our hybrid genetic algorithm [2]. The second one uses a distributed cooperative Tabu Search. Experimental results show that our algorithms, applied to several instances given by TDF-C2R, lead to important time performance improvements. [ABSTRACT FROM AUTHOR]

In the advanced generalized conservative staircase (AGCS) method, the video segments are arranged in groups. The first group contains first segment, second group contains next two segments, and pth group (p>2) contains 3ª2P-3 number of segments starting with kth segment S,. where k=3ª2p-3+1. Each segment of pth group (p>2) is divided into 3ª2P-3ª+mp+mp-1 subsegments, where mp determines the number of subsegments of a segment in pth group that are transmitted in a time slot. The subsegments are arranged in nip disjoint sets. The first set consists of subsegments Sq+i,2,..., Sq+i,q, Sq+i,q, where q 3ª2P-3 1⩽i⩽q, and zth set (1p) consists of subsegments Sq+i,qª(z-1)+z-1, Sq+i,qª(z-1)+z+1 Sq+i,qªz+z-1. In the AGCS method, an upper bound on mp has been derived to provide continuous data. If larger value of mp than its upper bound is taken, then the method will not provide continuous data. But, increasing the value of mp decreases the bandwidth requirement, which in turn gives better method. In this paper, we discuss a scheme that removes the bound on mp and still provides the continuous video data to users. [ABSTRACT FROM AUTHOR]

The problem of optimizing access and bandwidth sharing among Transmission Control Protocol (TCP) connections in the mobile digital video broadcasting return channel via satellite (DVB-RCS) is tackled in this paper. After sketching the general system architecture, we explicitly deal with the dynamic assignment of bandwidth to TCP connections on the return link, which is accomplished by a network control center (NCC) placed onboard the satellite. Mobile users access the satellite in multifrequency time-division multiple access (MF-TDMA), whereas they receive data from the NCC in time-division multiplexing (TDM). Two different techniques, based on deterministic and random access, are compared in terms of bandwidth usage and average completion time per connection, when the mobile user acts as both server and client. In the server case, to increase the TCP throughput, both packet-level forward error correction (FEC) on data sent by mobile users and a duplicated and delayed acknowledgment technique for TCP acknowledgment traffic from the NCC to the mobile users are applied. An analysis of the packet losses and a simulation campaign of file transfers by employing a realistic channel model has been carried out. The results of the analysis show the convenience of adopting a technique, in addition to the optimal data redundancy in different cases, such as the server or client role of users, their willingness to pay, the file size, and the environment type. [ABSTRACT FROM AUTHOR]

Cooperative communication systems have attracted much attention recently due to their desirable performance gain while using single antenna terminals. This paper addresses the joint timing and channel estimation problem, and furthermore the resynchronization of multiple timing offsets in a cooperative relay system. The estimations of timing and channel are conducted in two phases and the associated Cramér-Rao bounds (CRB) are derived for both phases. It is demonstrated that the conventional CRB is not valid for timing parameters under fading conditions, and a new bound called Weighted Bayesian CRB is proposed. With the timing and channel estimates, a general framework of the resynchronization filter design is developed in order to compensate the multiple timing offsets at the destination. The proposed methods are applied to different scenarios with varying degrees of timing misalignment and are numerically shown to provide excellent performances that approach the perfectly synchronized case. [ABSTRACT FROM AUTHOR]

This paper presents the development of Neural Network models to predict the likelihood of interference experienced by Broadcast users in the HF spectrum (3-30 MHz). The models are based upon several years of measurements recorded at Linköping (Sweden) across the HF band, covering a substantial part of a sunspot cycle. The dataset used for the model development is a result of a long-term project being undertaken jointly by the University of Manchester and by the Swedish Defence Research Establishment, to measure systematically and to analyze the occupancy of the entire HF spectrum. The measure of occupancy used is congestion, which is defined as the fraction of channels within a certain frequency allocation with signals exceeding a given threshold. The procedures for measuring and modeling congestion as a function of solar activity, time of day, day of year and incident field strength threshold are briefly presented. The accuracy of the predictions produced by the developed models demonstrate their ability to successfully capture the 24-hour, seasonal and long-term trend in the variability of congestion. These models can be used to advise operators on typical interference occupancy levels and assist the HF broadcast service in the planning of frequency usage and management by assessing the interference effect to short-wave, broadcasting in an effort to alleviate spectral congestion in the HF broadcast bands. [ABSTRACT FROM AUTHOR]

The MobileTV, IPTV, and DVB standards (DVB-HIT) have been defined to offer mobile users interactive multimedia services with quality of service (Q0S) consistency analogous to TV services. However, the market has yet to provide effective and economical solutions for the real-time delivery of such services to the corresponding transmitters over multi-domain IP networks. The monitoring system proposed in this paper enables the Q0S in the IP networks involved in the delivery of real-time multimedia content to the transmitters to be ascertained. The system utilizes the Q0S parameters defined in MPEG-2 Transport Streams to detect problems occurring in the heterogeneous multi-domain IP networks. The ability to detect problems having an adverse effect on Q0S allows appropriate control actions to be determined to recover the QoS across the composite IP network. The design and implementation of the proposed QoS-Monitoring system (QoS-MS) is presented, followed by analysis of experimental results that demonstrate the feasibility of the system. [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]

This paper describes a highly accurate carrier-frequency synchronization scheme for actively, automatically locking multiple, remotely located AM broadcast transmitters to a common frequency/timing reference source such as GPS. The extremely tight frequency lock (to ~1 part in 109 or better) permits the effective elimination of audible and even sub-audible beats between the local (desired) station's carrier signal and the distant stations' carriers, usually received via skywave propagation during the evening and nighttime hours. These carrier-beat components cause annoying modulations of the desired station's audio at the receiver and concurrent distortion of the audio modulation from the distant station(s) and often cause listeners to "tune out" due to the low reception quality. Significant reduction or elimination of the beats and related effects will greatly enlarge the effective (interference-limited) listening area of the desired station (from 4 to 10 times as indicated in our tests) and simultaneously reduce the corresponding interference of the local transmitter to the distant stations as well. In addition, AM stereo (CQUAM) reception will be particularly improved by minimizing the phase shifts induced by co-channel interfering signals; hybrid digital (HD) signals will also benefit via reduction in beats from analog signals. The automatic frequency-control hardware described is inexpensive ($1000-$2000), requires no periodic recalibration, has essentially zero long-term drift, and could employ alternate wide-area frequency references of suitable accuracy, including broadcasts from WWVB, LORAN-C, and equivalent sources. The basic configuration of the GPS-disciplined oscillator which solves this problem is extremely simple. The main oscillator is a conventional high-stability quartz-crystal type. To counter long- term drifts, the oscillator is slightly adjusted to track a high-precision source of standard frequency obtained from a specialized GPS receiver (or other source), usually at 10.000 MHz. This very stable local reference frequency is then used as a clock for a standard digitally implemented frequency synthesizer, which is programmed to generate the specific carrier frequency desired. The stability of the disciplining source, typically ~1 part in 109 to 1011,is thus transferred to the final AM transmitter carrier output frequency. [ABSTRACT FROM AUTHOR]

Commercial terrestrial digital multimedia broadcasting (T-DMB) service was launched in the Republic of Korea. Reflecting recent demands of high quality video, Advanced T-DMB(AT-DMB) adopting a hierarchical modulation is being developed as the way to increase the payload data. Based on the AT-DMB system, this paper proposes a method to provide increased video quality with high spatial resolution. By using the emerging H.264/MPEG-4 AVC amendment 3 scalable extension (SVC), the proposed method is able to support high spatial resolution as well as the currently serviced spatial resolution while retaining the backward compatibility to T-DMB. In the experimental results, the benefits in terms of coding efficiency and spatial scalability are proven in comparison with the conventional T-DMB and alternative methods to provide a high spatial resolution video. [ABSTRACT FROM AUTHOR]

In order to deliver video streams efficiently over WiFi to many thousands of consumer handheld devices, broadcast protocols must be employed. In this mode of operation the received video quality can deteriorate rapidly as a result of high application layer packet loss which occurs because MAC frame retransmission cannot be used. In this paper we develop a robust video delivery solution for broadcast transmission over 802.11a/g. Using a cross-layer WiFi simulator in combination with an accurate time-correlated fading channel the received video quality is evaluated for broadcast H.264 video sequences. Application layer cross-packet forward error correction is then used together with error concealment at the video client. Furthermore, the application of an external packet interleaver is considered. Combining a block size of two hundred packets (which introduces a 4.8 second delay) and an application layer FEC code rate of 0.75 our results demonstrate that video can be successfully broadcast over WiFi to many thousands of handheld terminals at large-scale spectator events. [ABSTRACT FROM AUTHOR]

This paper studies the two-hop relay broadcast channel (BC) for a relay-assisted wireless cellular network where the multiuser independent downlink signals from the base station (BS) are first transmitted to a fixed relay station (RS), and then forwarded by the RS to multiple mobile users. Assuming both the BS and RS are equipped with multiantennas, we study the joint optimization of linear beamforming and power control at the BS and RS so as to minimize their weighted sum-power consumption under the user minimum signal-to-interference-noise-ratio (SINR) - quality-of-service (QoS) - constraints. We apply two well-known criteria in the literature, namely, the "SINR balancing" and the "channel-inversion," for the design of linear precoding in the traditional nonrelay-assisted multiantenna BC to the relay-assisted multiantenna BC. First, a convergence-ensured iterative SINR-balancing algorithm is proposed to successively in turn optimize the transmit parameters at one station (BS or RS) with those at the other station being fixed. Second, a joint BS and RS channel-inversion algorithm is proposed together with a novel technique, termed "eigenmode switching," at the RS to reduce the power penalty of the channel inversion. Simulation results show that the proposed joint beamforming and power control schemes provide substantial power savings to achieve the assigned user QoS constraints. [ABSTRACT FROM AUTHOR]

In order to reduce the bit-rate of video signals, cur- rent coding standards apply hybrid coding with motion-compensated prediction and transform coding of the prediction error. In former publications, it has been shown that aliasing components contained in an image signal, as well as motion blur are limiting the prediction efficiency obtained by motion compensation. In this paper, we show that the analytical calculation of an optimal inter- polation filter at particular constraints is possible, resulting in total coding improvements of 20% at broadcast quality compared to the H.264/AVC High Profile. Furthermore, the spatial adaptation to local image characteristics enables further improvements of 0.15 dB for CIF sequences compared to globally adaptive filter or up to 0.6 dB, compared to the standard H.264/AVC. Additionally, we show that the presented approach is generally applicable, i.e., also motion blur can be exactly compensated, if particular constraints are fulfilled. [ABSTRACT FROM AUTHOR]

This paper presents technical requirements of on-channel repeaters (OCRs) to broadcast the terrestrial digital multimedia broadcasting (T-DMB) signals in single frequency network (SFN) environment. In order to meet such requirements, the configuration and implementation method of the equalization OCR are proposed. The proposed equalization OCR does not have only short processing delay, but also provides high quality transmitting signals with high power because of the rejection capability of a feedback signal introduced by incomplete antenna isolation and multipath signals existing between a main transmitter and the OCR. Computer simulations and laboratory tests results are provided to figure out performance of the proposed equalization OCR. [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]

Histogram equalization is a simple and effective method for contrast enhancement as it can automatically define the intensity transformation function based on statistical characteristics of the image. However, it tends to alter the brightness of the entire image, which it is not suitable for consumer electronic products, where preservation of the original brightness is essential to avoid annoying artifacts. This paper presents a new contrast enhancement method for generalization of the existing bihistogram equalization (BHE) and recursive mean-separate histogram equalization (RMSHE) methods. The proposed method is referred to gain-controllable clipped histogram equalization (GC-CHE) to provide both histogram equalization and brightness preservation. More specifically adaptive contrast enhancement is realized by using clipped histogram equalization with controllable gain. The clipping rate is determined based on the mean brightness, and the clipping threshold is determined based on the clipping rate. The clipping rate is adaptively controlled to enhance the contrast with preserving the mean brightness. It is mathematically proven that the mean brightness of the output image converges to that of the input image with adaptive controlled. Simulation results show that the proposed GC-CHE method outperforms existing histogram-based methods, such as HE, BHE, and RMSHE, in various situations. [ABSTRACT FROM AUTHOR]

This paper aims to combine adaptive modulation and coding with layered video coding to improve the quality of video services to users experiencing differing radio conditions, in the context of broadcast and multicast standards such as MBMS and BCMCS. We propose an optimal radio resource allocation algorithm which maximizes a general performance metric for a video session in polynomial time. We show that system-wide optimal resource allocation can be obtained by combining our algorithm with a simple two-step decomposition of the system. In some configurations frequent re-allocations of resource are required, so we also present a sub-optimal allocation algorithm which runs in near linear time. Simulation results show better video quality than existing resource allocation schemes over a range of conditions, and also suggest that the difference between the performance of optimal and sub-optimal solutions is less than 3%. [ABSTRACT FROM AUTHOR]

OFDMA is regarded as a promising solution for improving the performance of interactive wireless broadcasting systems. Recently, to enhance the performance of OFDMA systems, MIMO techniques have been investigated, and the OFDMA systems have been extended to MIMO architectures. Although transmit diversity is clearly advantageous on OFDMA uplink system in interactive broadcasting, it may not be possible that practical OFDMA system of wireless interactive broadcasting have more than two antennas due to size, cost, or hardware limitations. Cooperative diversity techniques are proposed to overcome the limitations. Cooperative networks aim to utilize the available terminals as a collection of distributed antennas for the transmission of signals from one end of the network to the other. Therefore, this paper focuses on design and performance evaluation of MIMO-OFDMA uplink system with cooperative diversity for wireless interactive broadcasting. For high-rate and reliable transmission, efficient transmission codes are proposed for two transmission cases (one-sided transmission case and two-sided transmission case). And, we present our investigations of the effects of line-of-sight. In the wireless communication environment, line-of-sight is very important to improve the system performance. Under the condition of utilizing the line-of-sight of cooperative partner, the performance of a terminal without line-of-sight can be improved. Taking into account the effects of utilizing line-of-sight of cooperative partner, the BER performance is investigated for a MIMO-OFDMA uplink scenario. [ABSTRACT FROM AUTHOR]

This paper addresses the automatic analysis of court-net sports video content. We extract information about the players, the playing-field in a bottom-up way until we reach scene-level semantic concepts. Each part of our framework is general, so that the system is applicable to several kinds of sports. A central point in our framework is a camera calibration module that relates the a-priori information of the geometric layout in the form of a court model to the input image. Exploiting this information, several novel algorithms are proposed, including playing-frame detection, players segmentation and tracking. To address the player-occlusion problem, we model the contour map of the player silhouettes using a nonlinear regression algorithm, which enables to locate the players during the occlusions caused by players in the same team. Additionally, a Bayesian-based classifier helps to recognize predefined key events, where the input is a number of real-world visual features. We illustrate the performance and efficiency of the proposed system by evaluating it for a variety of sports videos containing badminton, tennis and volleyball, and we show that our algorithm can operate with more than 91% feature detection accuracy and 90% event detection. [ABSTRACT FROM AUTHOR]

In this paper, we focus on the general relay channel. We investigate the application of the estimate-and-forward (EAF) relaying scheme to different scenarios. Specifically, we study assignments of the auxiliary random variable that always satisfy the feasibility constraints. We then consider the Gaussian relay channel with coded modulation, where we show that a three-level quantization outperforms the Gaussian quantization commonly used to evaluate the achievable EAF rates in this scenario. Last, we consider the cooperative general broadcast scenario with a multistep conference between the receivers. We first apply EAF to obtain a general achievable rate region with a multistep conference. We then use an explicit assignment for the auxiliary random variables to obtain an explicit rate expression for the single common message case with a two-step conference. [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]

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]

A new totaḻzeros decoding method based on efficient memory architecture for variable-length code tables (VLCTs) is proposed in this paper. In general, a large amount of memory accesses are required for context-based adaptive variable-length coding (CAVLC) decoding in the H.264/AVC standard. It is an important issue for low-power implementation of mobile video applications because the memory access function results in considerable power consumption. Some distinctive features in the VLCTs of the totaḻzeros have been found through a careful analysis of the codewords. Efficient memory architecture and an advanced decoding are developed in order to remove redundant memory access. The simulation results show that the proposed method achieves an approximately 80%-90% memory access saving, compared with conventional decoding. [ABSTRACT FROM AUTHOR]

In a multiple-antenna system with two transmitters and two receivers, a scenario of data communication, known as the X channel, is studied in which each receiver receives data from both transmitters. In this scenario, it is assumed that each transmitter is unaware of the other transmitter's data (noncooperative scenario). This system can be considered as a combination of two broadcast channels (from the transmitters' points of view) and two multiple-access channels (from the receivers' points of view). Taking advantage of both perspectives, two signaling schemes for such a scenario are developed. In these schemes, some linear filters are employed at the transmitters and at the receivers which decompose the system into either two noninterfering multiple-antenna broadcast subchannels or two noninterfering multiple-antenna multiple. access subchannels. The main objective in the design of the filters is to exploit the structure of the channel matrices to achieve the highest multiplexing gain (MG). It is shown that the proposed noncooperative signaling schemes outperform other known noncooperative schemes in terms of the achievable MG. In particular, it is shown that in some specific cases, the achieved MG is the same as the MG of the system if full cooperation is provided either between the transmitters or between the receivers. In the second part of the paper, it is shown that by using mixed design schemes, rather than decomposition schemes, and taking the statistical properties of the interference terms into account, the power offset of the system can be improved. The power offset represents the horizontal shift in the curve of the sum-rate versus the total power in decibels. [ABSTRACT FROM AUTHOR]

With the emergence of new TV media such as digital cable and satellite, the task of finding one's favorite program among hundreds of channels becomes a time consuming job. Furthermore, the channel zapping time of DTV is much longer than that of analogue one. To relieve the time overhead in channel navigation, this paper presents an intelligent channel navigation scheme for DTV channel selectors. The proposed navigation scheme analyzes each person's previous watching behavior in terms of recency and frequency, and then utilizes this information in the control of the up-down channel selector. Simulation studies show that the proposed scheme reduces the seek distance of the DTV channel navigation system by up to 68.7% under various conditions. [ABSTRACT FROM AUTHOR]

Convergence between broadcasting services and telecommunication services and provisioning of rich media services are two major requirements that recent mobile broadcast TV services try to satisfy. Although there exist independent technologies and international standards for each of the requirements, few efforts have been paid for providing solutions supporting these requirements together. In this paper, we present a standard-based implementation for mobile handsets in support of rich media enabled mobile broadcast TV services. [ABSTRACT FROM AUTHOR]

The aim of this paper is to present the concept for the development of an entirely speech-based user interface for a digital radio in combination with the possibility to specifically search for audio and data content. On the basis of a Digital Audio Broadcasting DAB device the embedded system utilizes a multichannel receiver with the possibility of simultaneously monitoring a variety of information sources with the purpose to generate spoken replies to verbal user requests. Furthermore, the system embraces an audio repository storing speech-based content elements to diminish the volatile property of broadcast content and to enable a memory function.' [ABSTRACT FROM AUTHOR]

This paper deals with cross-layer measurements for the efficient analysis of Digital Video Broadcasting-Terrestrial (DVB-T) system performance. In particular, valuable relation- ships between some cross-layer DVB-T system parameters are investigated through measurements conducted at the physical and application layers. Specifically, channel power, error vector magnitude, and modulation error ratio are assessed at the physical layer, whereas objective video quality indices are estimated at the application layer. To this aim, a number of real-life experiments are executed through a purposely developed testbed that enlists a suitable DVB-T transmission platform and real-life receivers. The ultimate goal is to highlight the importance of cross-layer measurements in the analysis and troubleshooting of DVB-T system operation and to show how to deduce from such measurements practical information and hints to be profitably used in the design and setup of any DVB-T broadcasting system. [ABSTRACT FROM AUTHOR]

One way to broadcast a popular video is to partition the video into segments, which are transmitted on several channels currently and periodically. The approach lets multiple users share channels, and thus obtains high bandwidth utilization. Many previous studies such as the recursive frequency-splitting (RFS) and harmonic broadcasting schemes focus on decreasing clients' waiting time. Another issue on reduction of clients' buffer demands is also important, especially for the goal to provide video services using limited-capability client devices such as PDAs and set-top boxes (STBs). Related work includes the skyscraper broadcasting (SkB), client-centric approach (CCA), greedy disk-conserving broadcasting (GDB), and reverse fast broadcasting (RFB) schemes. By combining RFS and RFB, this paper designs a hybrid broadcasting scheme (HyB), which exhibits a good tradeoff between client buffering spaces and waiting time. In comparison with SkB, CCA, GDB, RFB, and RFS, HyB requires smallest buffers, only 25% of video size. For waiting time, the scheme also outperforms all the schemes, except RFS. [ABSTRACT FROM AUTHOR]

After coarse synchronization in orthogonal frequency division multiplexing (OFDM) systems, there might still be a residual frequency offset (RFO) and a sampling clock frequency offset (SFO), which seriously degrade the performance of the systems. This paper suggests a simple way of estimating the SFO in OFDM-based digital radio mondiale (DRM) system. To demonstrate the efficiency of the proposed SFO estimator, comparisons are made with other existing estimators in terms of mean square error (MSE) performance, estimation range, and complexity. [ABSTRACT FROM AUTHOR]

In this paper we present a Base-matrix based decoder architecture for multi-rate QC-LDPC codes proposed in broadband broadcasting system. We use the Modified Min-Sum Algorithm (MMSA) as the decoding algorithm in this architecture, which lowers the complexity of the LDPC decoder while keeping almost the same performance or even better. Based on this algorithm, we designed a novel check node processing unit to reduce the complexity of the decoder and facilitate the multiplex of the processing units. The decoder designed with hardware constraints is not only scalable in throughput, but also easily configurable to support different QC-LDPC codes flexible in code rate and code length. [ABSTRACT FROM AUTHOR]

This paper presents a simple but effective IPTV channel control algorithm that keeps channel zapping time in the tolerable range with high network utilization. The proposed algorithm controls channel zapping time by adjusting the number of broadcasting channels that are located close to users over IP networks and the number of additional I-frames inserted into each channel, based on the user's channel preference information. Finally, experimental results are provided to show the performance of the proposed algorithm. [ABSTRACT FROM AUTHOR]