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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]

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]

Underwater acoustic networks have recently emerged as a new area of research in wireless networking. These networks can support a large number of applications such as environmental and underwater equipment monitoring. In recent years, there has been substantial work on protocol design for these networks with most efforts focusing on MAC and network layer protocols. Despite being a fundamental networking primitive, data broadcasting has so far received little attention in the context of underwater networks. This paper proposes an adaptive push system for dissemination of data in underwater acoustic wireless networks. Besides achieving adaptation of its broadcast schedule according to the a priori unknown needs of the clients, the proposed system also efficiently combats the problem of high latency of the underwater acoustic wireless environment. Simulation results show superior performance of the proposed system in the underwater environment compared to adaptations of existing terrestrial push systems. [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]

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]

We consider the broadcasting of a scalar memoryless Gaussian source over a two-user Gaussian broadcast channel with bandwidth expansion. In concurrent with the source description, a message that should be decoded reliably just by the “higher quality” user is sent. Conditioned on the message rate, we derive a necessary condition for the achievability of a mean-squared-error distortion pair at the two receivers, which extends the Reznic–Feder–Zamir bound. This necessary condition and a sufficient condition that we derive establish the set of achievable energy-distortion exponents for this setting. [ABSTRACT FROM AUTHOR]

Recently, the use of a virtual backbone in various applications in mobile ad hoc networks (MANETs) has become popular. These applications include topology management, point and area coverage, and routing protocol design. In a MANET, one challenging issue is to construct a virtual backbone in a distributed and localized way while balancing several conflicting objectives: small approximation ratio, fast convergence, and low computation cost. Many existing distributed and localized algorithms select a virtual backbone without resorting to global or geographical information. However, these algorithms incur a high computation cost in a dense network. In this paper, we propose a distributed solution based on reducing the density of the network using two mechanisms: clustering and adjustable transmission range. By using adjustable transmission range, we also achieve another objective, energy-efficient design, as a by-product. As an application, we show an efficient broadcast scheme where nodes (and only nodes) in a virtual backbone are used to forward the broadcast message. The virtual backbone is constructed using Wu and Li's marking process [37] and the proposed density reduction process. The application of the density reduction process to other localized algorithms is also discussed. The efficiency of our approach is confirmed through both analytical and simulation study. [ABSTRACT FROM AUTHOR]

In this paper, we propose a family of structures, namely, κ-localized minimum spanning tree (LMSTκ) for topology control and broadcasting in wireless ad hoc networks. We give an efficient localized method to construct LMSTκ using only O(n) messages under the local-broadcast communication model, i.e., the signal sent by each node will be received by all nodes within the node's transmission range. We also analytically prove that the node degree of the structure LMSTκ is at most 6, LMSTκ is connected and planar and, more importantly, the total edge length of the LMSTk is within a constant factor of that of the minimum spanning tree when κ ≥ 2 (called low weighted hereafter). We then propose another low weighted structure, called Incident MST and RNG Graph (1MRG), that can be locally constructed using at most 13n messages under the local broadcast communication model. Test results are corroborated in the simulation study. We study the performance of our structures in terms of the total power consumption for broadcasting, the maximum node power needed to maintain the network connectivity. We theoretically prove that our structures are asymptotically the best possible for broadcasting among all locally constructed structures. Our simulations show that our new structures outperform previous locally constructed structures in terms of broadcasting and power assignment for connectivity. [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]

One of the unresolved problems in providing viable stereoscopic 3D video services based on terrestrial DTV is to carry additional video while maintaining existing 2D video quality within limited bandwidth. Mitigating this problem, our proposed 3D service incorporates ATSC Non- Real-Time (NRT) transmission scheme in which the additional video stream delivered in advance of use and stored for later consumption. At later time when transmitting an air AV program associating the additional video previously sent, a receiving device is able to play stereoscopic 3D video by combining real-time video and the stored additional video synchronously. In this paper, we propose backward compatible system for the 3D NRT service and show that our implementation successfully presents the new stereoscopic 3D video service based on Terrestrial DTV. [ABSTRACT FROM AUTHOR]

This paper presents a new TxID (Transmitter Identification) data insertion system for ATSC DTV auxiliary data transmission. Because the functional description of TxID was not included in the early ATSC transmission standard, most of the conventional DTV transmitters and translators cannot transmit TxID signal which should be generated during 8-VSB (Vestigial Side Band) symbol generation process. The proposed system generates RF signal of TxID and then combines it synchronously with the RF signal of the legacy ATSC transmitter. This combination enables the conventional DTV transmitters and translators to transmit TxID signal with auxiliary data based on RF watermark technique. The proposed system is implemented and its performance is evaluated through laboratory tests. [ABSTRACT FROM PUBLISHER]

There are two main approaches, static and dynamic, to broadcast algorithms in wireless ad hoc networks. In the static approach, local algorithms determine the status (forwarding/nonforwarding) of each node proactively based on local topology information and a globally known priority function. In this paper, we first show that local broadcast algorithms based on the static approach cannot achieve a good approximation factor to the optimum solution (an NP-hard problem). However, we show that a constant approximation factor is achievable if (relative) position information is available. In the dynamic approach, local algorithms determine the status of each node "on-the-fly” based on local topology information and broadcast state information. Using the dynamic approach, it was recently shown that local broadcast algorithms can achieve a constant approximation factor to the optimum solution when (approximate) position information is available. However, using position information can simplify the problem. Also, in some applications it may not be practical to have position information. Therefore, we wish to know whether local broadcast algorithms based on the dynamic approach can achieve a constant approximation factor without using position information. We answer this question in the positive—we design a local broadcast algorithm in which the status of each node is decided "on-the-fly” and prove that the algorithm can achieve both full delivery and a constant approximation to the optimum solution. [ABSTRACT FROM PUBLISHER]

This paper describes an interactive framework called LIME (Lightweight interactive multimedia environment). LIME provides a profile based on Web standards including XHTML, CSS and ECMAScript that addresses specific requirements and limitations of consumer electronics devices such as TV and IPTV. Since Web standards are typically developed for PC browsers, content rendering, navigation, and user interface considerations for TV require special attention. The LIME profile supports a subset of functions relevant to TV, as well as extensions to enable functions such as synchronized rendering of monomedia content including audio, video, graphics, and text, while providing interactivity. [ABSTRACT FROM AUTHOR]

In a dynamic wireless network where a source terminal communicates with a destination, in presence of other users in the network, it is worthwhile to consider oblivious relaying strategies of a relay in close proximity to the source transmitter. The source-relay channel is assumed to be a fixed gain additive white Gaussian noise (AWGN) channel due to source-relay colocation. The source-destination and the relay-destination transmissions are subject to block flat fading dynamics. A perfect channel state information (CSI) at the respective receivers only is assumed. With the expected throughput as a performance measure, we incorporate a two-layer broadcast approach into a cooperative strategy based on the decode-and-forward scheme - Sequential Decode-and Forward (SDF). The broadcast approach splits the transmitted rate into superimposed layers, corresponding to a "bad" and a "good" channel states, allowing better adaptation to the actual channel conditions. The achievable rate expressions for the SDF strategy are derived under the broadcast approach for multiple settings including single user, multiple-input single-output (MISO) and the general relay setting using successive decoding technique, both numerically and analytically. Continuous broadcasting lower bounds are derived for the MISO and for the oblivious cooperation scenarios. [ABSTRACT FROM AUTHOR]

The multicast capacity is determined for networks that have deterministic channels with broadcasting at the transmitters and no interference at the receivers. The multicast capacity is shown to have a cut-set interpretation. It is further shown that one cannot always layer channel and network coding in such networks. The proof of the latter result partially generalizes to discrete memoryless broadcast channels and is used to bound the common rate for problems where one achieves a cut bound on throughput. [ABSTRACT FROM AUTHOR]